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From: "Ben Campbell" <ben@nostrum.com>
To: Huangyihong <rachel.huang@huawei.com>
Date: Tue, 01 Mar 2016 16:05:48 -0600
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Subject: Re: [avtext] New Version Notification for draft-ietf-avtext-splicing-notification-04.txt
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On 29 Feb 2016, at 20:27, Huangyihong (Rachel) wrote:

> Hi Ben,
>
> We author did receive IANA review. But there's no comments in that 
> review. It was just to confirm the actions that need IANA to complete. 
> And I think it requires us to respond only when any action they are 
> going to take is not accurate.

That's technically correct, but a confirmation email helps to let them 
know that you received and understood theirs.

>
> They also said these actions need to be evaluated by experts reviews 
> via a separate request. However, we haven't seen any expert review 
> comments from then on. Maybe they forgot to copy us or maybe there's 
> no comments at all. We're not sure about it.
>

They may also not be complete yet. (Hopefully you saw that a late 
Gen-ART review came out today.(

> It's okay for us to bring an update one before the IESG evaluation to 
> address the further issues you raised before. What do you think?

I suggest submitting an update to resolve my comments along with any 
change that may be needed from the gen-art review. The registration 
expert reviews may or may not suggest further changes, but we can handle 
those when they happen.

Thanks!

Ben.


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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: [avtext] New Version Notification for draft-ietf-avtext-splicing-notification-04.txt
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Date: Wed, 2 Mar 2016 01:31:09 +0000
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Subject: Re: [avtext] New Version Notification for draft-ietf-avtext-splicing-notification-04.txt
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Hi Ben,

Thanks for the suggestion. We'll do it next few days.

BR,
Rachel


> -----Original Message-----
> From: Ben Campbell [mailto:ben@nostrum.com]
> Sent: Wednesday, March 02, 2016 6:06 AM
> To: Huangyihong (Rachel)
> Cc: avtext@ietf.org
> Subject: Re: [avtext] New Version Notification for
> draft-ietf-avtext-splicing-notification-04.txt
>=20
> On 29 Feb 2016, at 20:27, Huangyihong (Rachel) wrote:
>=20
> > Hi Ben,
> >
> > We author did receive IANA review. But there's no comments in that
> > review. It was just to confirm the actions that need IANA to complete.
> > And I think it requires us to respond only when any action they are
> > going to take is not accurate.
>=20
> That's technically correct, but a confirmation email helps to let them kn=
ow that
> you received and understood theirs.
>=20
> >
> > They also said these actions need to be evaluated by experts reviews
> > via a separate request. However, we haven't seen any expert review
> > comments from then on. Maybe they forgot to copy us or maybe there's
> > no comments at all. We're not sure about it.
> >
>=20
> They may also not be complete yet. (Hopefully you saw that a late Gen-ART
> review came out today.(
>=20
> > It's okay for us to bring an update one before the IESG evaluation to
> > address the further issues you raised before. What do you think?
>=20
> I suggest submitting an update to resolve my comments along with any chan=
ge
> that may be needed from the gen-art review. The registration expert revie=
ws
> may or may not suggest further changes, but we can handle those when they
> happen.
>=20
> Thanks!
>=20
> Ben.


From nobody Wed Mar  2 04:50:38 2016
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To: "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, <draft-ietf-avtext-splicing-notification@ietf.org>
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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Hi,

This are some comments that came out of my review as an IANA expert for 
the RTP Header Extensions and RTCP Control Packet Types, and which of 
some is not related to the IANA expert review role. However, I have 
noted that several of these, issues 1,2,3, and 5, are related to 
protocol clearness as well as correct usage of RTP which are review 
criteria for the RTP header extension registry.


1. Section 1.1 and 3.1:

What the NTP timestamp included in the extensions represent are not 
being sufficiently made clear. I think it must be made clear that the 
NTP timestamp represents the Main RTP senders wall clock timestamp 
corresponding to these splicing in and out points. The main RTP senders 
time base may have nothing with any formal time basis distributed by the 
NTP protocol.

2. Section 3.1:

If the 7 octets splicing-out NTP timestamp is smaller
    than the lower 7 octets splicing-in NTP timestamp, it implies a wrap
    of the 64-bit splicing-out NTP timestamp which will then be
    calculated by the 7 octets splicing-out NTP timestamp plus
    0x100000000.

So I think "smaller" is a underspecified operation here. Although I 
think loading the 7 octets into one 64 bit unsigned integer respectively 
and perform a lesser than comparison will actually work.

3. Section 3.1:

This section is not clear on what is the data part of the header 
extension format. As it is not the header extension that can determine 
if there will be usage of one-byte vs two-byte format that will be used, 
this needs to be defined in a more un

4. Section 3.2:
The RTCP splicing notification message can be appended to RTCP SR the
    main RTP sender generates in compound RTCP packets, and hence follows
    the compound RTCP rules defined in Section 6.1 in [RFC3550].

I think the use of "append to RTCP SR" is unfortunate. I believe what 
would be the right messages is to say that the splicing notification 
messages is included in the same RTCP compound packet as the RTCP SR for 
the main RTP stream. Using the SSRC values to relate the two. This is 
important because there should be no formal requirement on ordering 
between individual RTCP packets within a compound packet.


5. Section 3.2:

This format is
       similar to the RTCP Sender Report (Section 6.4.1 of [RFC3550])

Why is it only similar and not the same as in RFC  3550?


6. Section 3.2:

    If the use of non-compound RTCP [RFC5506] was previously negotiated
    between the sender and the splicer, the RTCP splicing notification
    message may be sent as non-compound RTCP packets.

I wonder over this, I wonder if this RTCP packet type really should be 
sent in an RTCP non-compound packet without the corresponding RTCP SR 
packet? The reason I state this is that the RTP main sender has to use 
SR/RR report block LSR field to determine which is the latest received 
RTCP SR packet for the SSRC in the splicing RTCP packet. And due to 
timing of RTCP packet transmission that can be old information.  Thus, 
the splicer can operate on RTP to NTP Timestamps that are not current. 
Thus, I think it would be a recommendation to include the RTCP SR with 
the splicing packet. It can still be a non-compound as one could leave 
out the SDES CNAME items.

7. Section 6.2:

    Only codecs that are supported both by the main RTP stream and the
    substitutive RTP stream could be negotiated with SDP O/A. And the
    splicer MUST choose the same codec for both of these two streams.

So this payload type restriction is in my view in the wrong place and 
not clear enough. So, to be able to splice into one consistent outgoing 
stream there is a basic requirement that the RTP Payload Types used are 
having the same clock rate, otherwise RFC 7160 applies, as uncertainty 
about how RTP maps to NTP and splicing appears to be a bad mix.

Further if one is going to express this in this more restrictive way of 
demanding the same codes, then I do wonder if one actually don't need to 
go as far as saying that the same payload type configurations needs to 
be possible to us in both main and substitute streams.

8. Section 7:

I find a significant short coming in this security consideration. That 
is the failure to discuss the importance for the splicer to be able to 
trust the splicing interval information from the main RTP sender. This 
information really should be authenticated and integrity protected 
between the main RTP sender and the splicer.

Thus, I am quite worried about this paragraph:

    Since the mechanism introduced in this document is not dependent on
    any RTP payload-level hints for finding the splicing in and out
    points, Secure Real-Time Transport Protocol (SRTP) [3711] can be used
    to provide confidentiality of the RTP payload while leaving the RTP
    header in the clear so that the splicer can still carry out splicing
    without keying materials. However, for a malicious endpoint without
    the key, it can observe the splicing time in the RTP header, and it
    can intercept the substitutive content and even replace it with a
    different one if the substitutive stream does not use any security
    like SRTP and the splicer does not authenticate the main and
    substitutive content sources.

In the end it hints towards the need for authentication of the streams. 
But that really should be worded in a stronger way as a requirement for 
security.

I do understand that this is really a case where two layers of security 
would be good. One between the main and substitute sender and the 
intended receiver, and an outer layer for sender to splicer. However, I 
don't argue that such a solution needs to be created. I think having the 
splicer as a trusted entity is more likely to be a reasonable deployment 
model.

I do note that splicing do have the issue from key-management 
perspective that both main and substitute sender needs to use the same 
key if the security is from sender to receiver, rather than splicer to 
receiver.

In the end I think what needs to be changed are:

A. Clarify need for source authentication and integrity verification
B. The potential issues with having main and substitute stream sender 
use the same keys end-to-end for the same SSRC. This can be problematic 
from two-time pad perspective.
C. Likely be clear that a security model with a trusted splicer is the 
one that are easiest to realize without two layer security.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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From: Bo Burman <bo.burman@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-04.txt
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Subject: Re: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-04.txt
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Hi,

Thank you for the comments. I think they are all good, and I will shortly s=
ubmit an updated draft. I chose to make the new sub-registry creation secti=
on similar to RFC 5285 section 9.1 that creates the base registry, which I =
believe should be clearer than the previous prose text.

I checked the current IANA RTP Compact Header Extensions registry for MID (=
urn:ietf:params:rtp-hdrext:sdes:mid), and it seems it is no longer present =
there, so I removed the text that asks to move it into the new registry.

Cheers,
/Bo

> -----Original Message-----
> From: Ben Campbell [mailto:ben@nostrum.com]
> Sent: den 29 februari 2016 22:06
> To: Bo Burman
> Cc: avtext@ietf.org; draft-ietf-avtext-sdes-hdr-ext.all@ietf.org
> Subject: Re: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-04.txt
>=20
> Hi,
>=20
> I think this version is almost ready to progress, but I do have some mino=
r comments about the new IANA section. Since
> the IETF last call prompts IANA to start their review and plan their upda=
tes, and it might be confusing to have IANA
> consideration updates while this is in progress, we probably need to reso=
lve this before last call.
>=20
> -- 5.1, paragraph 5:
>=20
> The text says "IANA is requested to register the below in the RTP Compact=
 Header Extensions: Extensions, and to create
> the corresponding sub-registry:"
>=20
> But nothing in the rest of the section appears to describe the creation o=
f the sub-registry. Now, I think you've given most
> the information needed in 5.0, but that section reads like a summary rath=
er than an actual request for action. I propose a
> slight rearrangement, along the lines of the following:
>=20
> 5.0: Summary
> 5.1: registration of the URN.
> 5.2: Creation of the sub registry. (The sub-registry probably needs a hum=
an-readable name in addition to the URN.)
> 5.3: <current 5.2>
>=20
> -- 5.2, last paragraph
>=20
> The RFC editor does not move things in IANA registries. IANA does that.
> I suggest the following:
> OLD
>     RFC-editor note: Please move the MID SDES item already registered in
>     the main registry by [I-D.ietf-mmusic-sdp-bundle-negotiation] to the
>     newly formed RTP SDES Compact Header Extensions.
> NEW
>     IANA is also requested to move the MID SDES item already registered i=
n
>     the main registry by [I-D.ietf-mmusic-sdp-bundle-negotiation] to the
>     newly formed RTP SDES Compact Header Extensions.
>=20
>=20
>=20
> On 22 Feb 2016, at 2:44, Bo Burman wrote:
>=20
> > This update addresses the AD comment on explicitly requesting an IANA
> > sub-registry for the urn:ietf:params:rtp-hdrext:sdes sub-space, with
> > the additional RTCP SDES concerns described in this draft, compared to
> > what is already described for urn:ietf:params:rtp-hdrext. It also
> > addresses the AD comment on receiver actions for section 4.2.6, which
> > was accidentally omitted from the previous version.
> >
> > Cheers,
> > /Bo
> >
> >> -----Original Message-----
> >> From: avtext [mailto:avtext-bounces@ietf.org] On Behalf Of
> >> internet-drafts@ietf.org
> >> Sent: den 22 februari 2016 09:36
> >> To: i-d-announce@ietf.org
> >> Cc: avtext@ietf.org
> >> Subject: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-04.txt
> >>
> >>
> >> A New Internet-Draft is available from the on-line Internet-Drafts
> >> directories.
> >> This draft is a work item of the Audio/Video Transport Extensions of
> >> the IETF.
> >>
> >>         Title           : RTP Header Extension for RTCP Source
> >> Description Items
> >>         Authors         : Magnus Westerlund
> >>                           Bo Burman
> >>                           Roni Even
> >>                           Mo Zanaty
> >> 	Filename        : draft-ietf-avtext-sdes-hdr-ext-04.txt
> >> 	Pages           : 15
> >> 	Date            : 2016-02-22
> >>
> >> Abstract:
> >>    Source Description (SDES) items are normally transported in RTP
> >>    control protocol (RTCP).  In some cases it can be beneficial to
> >> speed
> >>    up the delivery of these items.  Mainly when a new source (SSRC)
> >>    joins an RTP session and the receivers needs this source's
> >> identity,
> >>    relation to other sources, or its synchronization context, all of
> >>    which may be fully or partially identified using SDES items.  To
> >>    enable this optimization, this document specifies a new RTP header
> >>    extension that can carry SDES items.
> >>
> >>
> >> The IETF datatracker status page for this draft is:
> >> https://datatracker.ietf.org/doc/draft-ietf-avtext-sdes-hdr-ext/
> >>
> >> There's also a htmlized version available at:
> >> https://tools.ietf.org/html/draft-ietf-avtext-sdes-hdr-ext-04
> >>
> >> A diff from the previous version is available at:
> >> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-sdes-hdr-ext-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/
> >>
> >> _______________________________________________
> >> avtext mailing list
> >> avtext@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avtext


From nobody Wed Mar  2 09:02:29 2016
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Subject: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Extensions of the IETF.

        Title           : RTP Header Extension for RTCP Source Description Items
        Authors         : Magnus Westerlund
                          Bo Burman
                          Roni Even
                          Mo Zanaty
	Filename        : draft-ietf-avtext-sdes-hdr-ext-05.txt
	Pages           : 15
	Date            : 2016-03-02

Abstract:
   Source Description (SDES) items are normally transported in RTP
   control protocol (RTCP).  In some cases it can be beneficial to speed
   up the delivery of these items.  Mainly when a new source (SSRC)
   joins an RTP session and the receivers needs this source's identity,
   relation to other sources, or its synchronization context, all of
   which may be fully or partially identified using SDES items.  To
   enable this optimization, this document specifies a new RTP header
   extension that can carry SDES items.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtext-sdes-hdr-ext/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-avtext-sdes-hdr-ext-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-avtext-sdes-hdr-ext-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.

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


From nobody Thu Mar  3 02:51:28 2016
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, "draft-ietf-avtext-splicing-notification@ietf.org" <draft-ietf-avtext-splicing-notification@ietf.org>
Thread-Topic: Some comments on draft-ietf-avtext-splicing-notification-04
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Date: Thu, 3 Mar 2016 10:51:10 +0000
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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Hi Magnus,

Thank you so much for your careful review and valuable comments. It helps u=
s to further improve the document quality.
Please see my replies inline.

BR,
Rachel


> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, March 02, 2016 8:50 PM
> To: avtext@ietf.org; Ben Campbell;
> draft-ietf-avtext-splicing-notification@ietf.org
> Subject: Some comments on draft-ietf-avtext-splicing-notification-04
>=20
> Hi,
>=20
> This are some comments that came out of my review as an IANA expert for t=
he
> RTP Header Extensions and RTCP Control Packet Types, and which of some is
> not related to the IANA expert review role. However, I have noted that se=
veral
> of these, issues 1,2,3, and 5, are related to protocol clearness as well =
as
> correct usage of RTP which are review criteria for the RTP header extensi=
on
> registry.
>=20
>=20
> 1. Section 1.1 and 3.1:
>=20
> What the NTP timestamp included in the extensions represent are not being
> sufficiently made clear. I think it must be made clear that the NTP times=
tamp
> represents the Main RTP senders wall clock timestamp corresponding to the=
se
> splicing in and out points. The main RTP senders time base may have nothi=
ng
> with any formal time basis distributed by the NTP protocol.

[Rachel]: You're right. So how about changing the "splicing interval" in se=
ction 1.1 to the following:

"
Splicing Interval:

The NTP-format timestamps, representing the main RTP sender wallclock time,=
 for the Splicing-In point and Splicing-Out point per [RFC6828] allowing th=
e splicer to know when to start and end the RTP splicing.
"
As for the ones in section 3.1, is it okay to just keep the "NTP-format tim=
estamps", since we clarify it in the terminology?

>=20
> 2. Section 3.1:
>=20
> If the 7 octets splicing-out NTP timestamp is smaller
>     than the lower 7 octets splicing-in NTP timestamp, it implies a wrap
>     of the 64-bit splicing-out NTP timestamp which will then be
>     calculated by the 7 octets splicing-out NTP timestamp plus
>     0x100000000.
>=20
> So I think "smaller" is a underspecified operation here. Although I think=
 loading
> the 7 octets into one 64 bit unsigned integer respectively and perform a =
lesser
> than comparison will actually work.

[Rachel]: So how about expressing it like this

OLD
"
The top 8 bits of the splicing-out NTP timestamp are inferred from the top =
8 bits of the splicing-in NTP timestamp. This is unambiguous, under the ass=
umption that the splicing-out time is after the splicing-in time, and the s=
plicing interval is less than 2^25 seconds. If the 7 octets splicing-out NT=
P timestamp is smaller than the lower 7 octets splicing-in NTP timestamp, i=
t implies a wrap of the 64-bit splicing-out NTP timestamp which will then b=
e calculated by the 7 octets splicing-out NTP timestamp plus 0x100000000. O=
therwise, the top 8 octets of splicing-out NTP timestamp is equal to the to=
p 8 octets of splicing-in NTP timestamp.
"

NEW
"
The top 8 bits of the splicing-out NTP timestamp are inferred from the top =
8 bits of the splicing-in NTP timestamp. This is unambiguous, under the ass=
umption that the splicing-out time is after the splicing-in time, and the s=
plicing interval is less than 2^25 seconds. Therefore, if the 7 octets spli=
cing-out NTP-format timestamp is smaller than 64-bit splicing-out NTP-forma=
t timestamp modulo 56 bits range, it implies a wrap of the 64-bit splicing-=
out NTP-format timestamp which will then be calculated by the 7 octets spli=
cing-out NTP timestamp plus 2^64. Otherwise, the top 8 octets of splicing-o=
ut NTP timestamp is equal to the top 8 octets of splicing-in NTP timestamp.
"

>=20
> 3. Section 3.1:
>=20
> This section is not clear on what is the data part of the header extensio=
n
> format. As it is not the header extension that can determine if there wil=
l be
> usage of one-byte vs two-byte format that will be used, this needs to be
> defined in a more un

[Rachel]: I don't understand this. The editing seems unfinished. The splici=
ng RTP header extension uses one-byte format with the fixed bit pattern 0xB=
EDE.

>=20
> 4. Section 3.2:
> The RTCP splicing notification message can be appended to RTCP SR the
>     main RTP sender generates in compound RTCP packets, and hence follows
>     the compound RTCP rules defined in Section 6.1 in [RFC3550].
>=20
> I think the use of "append to RTCP SR" is unfortunate. I believe what wou=
ld be
> the right messages is to say that the splicing notification messages is i=
ncluded
> in the same RTCP compound packet as the RTCP SR for the main RTP stream.
> Using the SSRC values to relate the two. This is important because there =
should
> be no formal requirement on ordering between individual RTCP packets with=
in
> a compound packet.

[Rachel]: Good point. How about changing to=20

"
The RTCP splicing notification message can be included in the RTCP compound=
 packet together with RTCP SR generated at the main RTP sender, and hence f=
ollows the compound RTCP rules defined in Section 6.1 in [RFC3550].
"

>=20
>=20
> 5. Section 3.2:
>=20
> This format is
>        similar to the RTCP Sender Report (Section 6.4.1 of [RFC3550])
>=20
> Why is it only similar and not the same as in RFC  3550?
>=20

[Rachel]: Actually it's the same. Bad wording:(.

>=20
> 6. Section 3.2:
>=20
>     If the use of non-compound RTCP [RFC5506] was previously negotiated
>     between the sender and the splicer, the RTCP splicing notification
>     message may be sent as non-compound RTCP packets.
>=20
> I wonder over this, I wonder if this RTCP packet type really should be se=
nt in an
> RTCP non-compound packet without the corresponding RTCP SR packet? The
> reason I state this is that the RTP main sender has to use SR/RR report b=
lock
> LSR field to determine which is the latest received RTCP SR packet for th=
e SSRC
> in the splicing RTCP packet. And due to timing of RTCP packet transmissio=
n that
> can be old information.  Thus, the splicer can operate on RTP to NTP
> Timestamps that are not current.
> Thus, I think it would be a recommendation to include the RTCP SR with th=
e
> splicing packet. It can still be a non-compound as one could leave out th=
e SDES
> CNAME items.

[Rachel]: Maybe I got it wrong. But are you saying that the accuracy of map=
ping RTP timestamp to NTP timestamp, which should be deduced by the informa=
tion in the RTCP SR that should be as close as the splicing starts? But the=
 RTCP splicing message is sent far ahead of the splicing happens, and shoul=
d be sent more than once to increase robustness. For example, may be severa=
l RTTs ahead. So in a non-compound case, it is possible that a RTCP SR will=
 be sent after a RTCP splicing message, hence it seems no meaning to includ=
e RTCP SR with RTCP splicing messages.=20

>=20
> 7. Section 6.2:
>=20
>     Only codecs that are supported both by the main RTP stream and the
>     substitutive RTP stream could be negotiated with SDP O/A. And the
>     splicer MUST choose the same codec for both of these two streams.
>=20
> So this payload type restriction is in my view in the wrong place and not=
 clear
> enough. So, to be able to splice into one consistent outgoing stream ther=
e is a
> basic requirement that the RTP Payload Types used are having the same clo=
ck
> rate, otherwise RFC 7160 applies, as uncertainty about how RTP maps to NT=
P
> and splicing appears to be a bad mix.

[Rachel]: Actually, this has been described in Section 2, 3rd paragraph. It=
 says

"
To ensure the Splicing Interval is valid for both the main RTP sender and t=
he substitutive RTP sender, the two senders MUST share a common reference c=
lock so that the splicer can achieve accurate splicing. The requirements fo=
r the common reference clock (e.g. resolution, skew) depend on the codec us=
ed by the media content.
"

>=20
> Further if one is going to express this in this more restrictive way of d=
emanding
> the same codes, then I do wonder if one actually don't need to go as far =
as
> saying that the same payload type configurations needs to be possible to =
us in
> both main and substitute streams.

[Rachel]: So you're suggesting that we delete this sentence?

>=20
> 8. Section 7:
>=20
> I find a significant short coming in this security consideration. That is=
 the failure
> to discuss the importance for the splicer to be able to trust the splicin=
g interval
> information from the main RTP sender. This information really should be
> authenticated and integrity protected between the main RTP sender and the
> splicer.
>=20
> Thus, I am quite worried about this paragraph:
>=20
>     Since the mechanism introduced in this document is not dependent on
>     any RTP payload-level hints for finding the splicing in and out
>     points, Secure Real-Time Transport Protocol (SRTP) [3711] can be used
>     to provide confidentiality of the RTP payload while leaving the RTP
>     header in the clear so that the splicer can still carry out splicing
>     without keying materials. However, for a malicious endpoint without
>     the key, it can observe the splicing time in the RTP header, and it
>     can intercept the substitutive content and even replace it with a
>     different one if the substitutive stream does not use any security
>     like SRTP and the splicer does not authenticate the main and
>     substitutive content sources.
>=20
> In the end it hints towards the need for authentication of the streams.
> But that really should be worded in a stronger way as a requirement for
> security.
>=20
> I do understand that this is really a case where two layers of security w=
ould be
> good. One between the main and substitute sender and the intended receive=
r,
> and an outer layer for sender to splicer. However, I don't argue that suc=
h a
> solution needs to be created. I think having the splicer as a trusted ent=
ity is
> more likely to be a reasonable deployment model.


[Rachel]: Okay. How about Add a new paragraph after the 3rd paragraph in Se=
ction 7?

"
To avoid above security issue, the authentication of the main and substitut=
ive content sources in the splicer SHOULD be considered. However, separatin=
g the security mechanism between senders and the splicer from the end-to-en=
d one may complicate the whole system deployment. In such a case, having th=
e splicer as a trusted entity is likely to be a reasonable deployment model=
.
"

>=20
> I do note that splicing do have the issue from key-management perspective
> that both main and substitute sender needs to use the same key if the sec=
urity
> is from sender to receiver, rather than splicer to receiver.

[Rachel]: Yes. I plan to add a sentence in the paragraph 3 like this:

OLD
"
Since the mechanism introduced in this document is not dependent on any RTP=
 payload-level hints for finding the splicing in and out points, Secure Rea=
l-Time Transport Protocol (SRTP) [RFC3711] can be used to provide confident=
iality of the RTP payload while leaving the RTP header in the clear so that=
 the splicer can still carry out splicing without keying materials.
"=20

NEW
"
Since the mechanism introduced in this document is not dependent on any RTP=
 payload-level hints for finding the splicing in and out points, Secure Rea=
l-Time Transport Protocol (SRTP) [RFC3711] can be used to provide confident=
iality of the RTP payload while leaving the RTP header in the clear so that=
 the splicer can still carry out splicing without keying materials. This im=
plies that either the main and substitutive sender use the same SRTP key, o=
r the splicer decrypts and re-encrypts with a single key.
"

>=20
> In the end I think what needs to be changed are:
>=20
> A. Clarify need for source authentication and integrity verification B. T=
he
> potential issues with having main and substitute stream sender use the sa=
me
> keys end-to-end for the same SSRC. This can be problematic from two-time =
pad
> perspective.
> C. Likely be clear that a security model with a trusted splicer is the on=
e that are
> easiest to realize without two layer security.

[Rachel]: See above proposed modifications.

>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


From nobody Thu Mar  3 14:00:38 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Extensions of the IETF.

        Title           : RTP Stream Identifier (RID) Source Description (SDES)
        Authors         : Adam Roach
                          Suhas Nandakumar
                          Peter Thatcher
	Filename        : draft-ietf-avtext-rid-01.txt
	Pages           : 7
	Date            : 2016-03-03

Abstract:
   This document defines and registers an RTCP SDES item, RID, for
   identification of RTP streams associated with Encoded Streams and
   Dependent Streams.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-avtext-rid-01

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-avtext-rid-01


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As Peter, Colin, and I discussed on-list, this version of RID proposes a 
second identifier, RRID, that is suitable for unique identification of 
Redundancy RTP Streams, should that become necessary for some 
application in the future. The only other changes are editorial. Please 
review the diffs (linked below) and comment.

Thanks!

/a

On 3/3/16 16:00, 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 Extensions of the IETF.
>
>          Title           : RTP Stream Identifier (RID) Source Description (SDES)
>          Authors         : Adam Roach
>                            Suhas Nandakumar
>                            Peter Thatcher
> 	Filename        : draft-ietf-avtext-rid-01.txt
> 	Pages           : 7
> 	Date            : 2016-03-03
>
> Abstract:
>     This document defines and registers an RTCP SDES item, RID, for
>     identification of RTP streams associated with Encoded Streams and
>     Dependent Streams.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtext-rid/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-avtext-rid-01
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-avtext-rid-01
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
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Subject: Re: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-04.txt
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Thanks! I have requested IETF Last Call for version 05.

Ben.

On 2 Mar 2016, at 10:59, Bo Burman wrote:

> Hi,
>
> Thank you for the comments. I think they are all good, and I will 
> shortly submit an updated draft. I chose to make the new sub-registry 
> creation section similar to RFC 5285 section 9.1 that creates the base 
> registry, which I believe should be clearer than the previous prose 
> text.
>
> I checked the current IANA RTP Compact Header Extensions registry for 
> MID (urn:ietf:params:rtp-hdrext:sdes:mid), and it seems it is no 
> longer present there, so I removed the text that asks to move it into 
> the new registry.
>
> Cheers,
> /Bo
>
>> -----Original Message-----
>> From: Ben Campbell [mailto:ben@nostrum.com]
>> Sent: den 29 februari 2016 22:06
>> To: Bo Burman
>> Cc: avtext@ietf.org; draft-ietf-avtext-sdes-hdr-ext.all@ietf.org
>> Subject: Re: [avtext] I-D Action: 
>> draft-ietf-avtext-sdes-hdr-ext-04.txt
>>
>> Hi,
>>
>> I think this version is almost ready to progress, but I do have some 
>> minor comments about the new IANA section. Since
>> the IETF last call prompts IANA to start their review and plan their 
>> updates, and it might be confusing to have IANA
>> consideration updates while this is in progress, we probably need to 
>> resolve this before last call.
>>
>> -- 5.1, paragraph 5:
>>
>> The text says "IANA is requested to register the below in the RTP 
>> Compact Header Extensions: Extensions, and to create
>> the corresponding sub-registry:"
>>
>> But nothing in the rest of the section appears to describe the 
>> creation of the sub-registry. Now, I think you've given most
>> the information needed in 5.0, but that section reads like a summary 
>> rather than an actual request for action. I propose a
>> slight rearrangement, along the lines of the following:
>>
>> 5.0: Summary
>> 5.1: registration of the URN.
>> 5.2: Creation of the sub registry. (The sub-registry probably needs a 
>> human-readable name in addition to the URN.)
>> 5.3: <current 5.2>
>>
>> -- 5.2, last paragraph
>>
>> The RFC editor does not move things in IANA registries. IANA does 
>> that.
>> I suggest the following:
>> OLD
>>     RFC-editor note: Please move the MID SDES item already registered 
>> in
>>     the main registry by [I-D.ietf-mmusic-sdp-bundle-negotiation] to 
>> the
>>     newly formed RTP SDES Compact Header Extensions.
>> NEW
>>     IANA is also requested to move the MID SDES item already 
>> registered in
>>     the main registry by [I-D.ietf-mmusic-sdp-bundle-negotiation] to 
>> the
>>     newly formed RTP SDES Compact Header Extensions.
>>
>>
>>
>> On 22 Feb 2016, at 2:44, Bo Burman wrote:
>>
>>> This update addresses the AD comment on explicitly requesting an 
>>> IANA
>>> sub-registry for the urn:ietf:params:rtp-hdrext:sdes sub-space, with
>>> the additional RTCP SDES concerns described in this draft, compared 
>>> to
>>> what is already described for urn:ietf:params:rtp-hdrext. It also
>>> addresses the AD comment on receiver actions for section 4.2.6, 
>>> which
>>> was accidentally omitted from the previous version.
>>>
>>> Cheers,
>>> /Bo
>>>
>>>> -----Original Message-----
>>>> From: avtext [mailto:avtext-bounces@ietf.org] On Behalf Of
>>>> internet-drafts@ietf.org
>>>> Sent: den 22 februari 2016 09:36
>>>> To: i-d-announce@ietf.org
>>>> Cc: avtext@ietf.org
>>>> Subject: [avtext] I-D Action: draft-ietf-avtext-sdes-hdr-ext-04.txt
>>>>
>>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts
>>>> directories.
>>>> This draft is a work item of the Audio/Video Transport Extensions 
>>>> of
>>>> the IETF.
>>>>
>>>>         Title           : RTP Header Extension for RTCP Source
>>>> Description Items
>>>>         Authors         : Magnus Westerlund
>>>>                           Bo Burman
>>>>                           Roni Even
>>>>                           Mo Zanaty
>>>> 	Filename        : draft-ietf-avtext-sdes-hdr-ext-04.txt
>>>> 	Pages           : 15
>>>> 	Date            : 2016-02-22
>>>>
>>>> Abstract:
>>>>    Source Description (SDES) items are normally transported in RTP
>>>>    control protocol (RTCP).  In some cases it can be beneficial to
>>>> speed
>>>>    up the delivery of these items.  Mainly when a new source (SSRC)
>>>>    joins an RTP session and the receivers needs this source's
>>>> identity,
>>>>    relation to other sources, or its synchronization context, all 
>>>> of
>>>>    which may be fully or partially identified using SDES items.  To
>>>>    enable this optimization, this document specifies a new RTP 
>>>> header
>>>>    extension that can carry SDES items.
>>>>
>>>>
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-ietf-avtext-sdes-hdr-ext/
>>>>
>>>> There's also a htmlized version available at:
>>>> https://tools.ietf.org/html/draft-ietf-avtext-sdes-hdr-ext-04
>>>>
>>>> A diff from the previous version is available at:
>>>> https://www.ietf.org/rfcdiff?url2=draft-ietf-avtext-sdes-hdr-ext-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/
>>>>
>>>> _______________________________________________
>>>> avtext mailing list
>>>> avtext@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/avtext


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Subject: [avtext] Last Call: <draft-ietf-avtext-sdes-hdr-ext-05.txt> (RTP Header Extension for RTCP Source Description Items) to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Extensions
WG (avtext) to consider the following document:
- 'RTP Header Extension for RTCP Source Description Items'
  <draft-ietf-avtext-sdes-hdr-ext-05.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2016-03-18. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   Source Description (SDES) items are normally transported in RTP
   control protocol (RTCP).  In some cases it can be beneficial to speed
   up the delivery of these items.  Mainly when a new source (SSRC)
   joins an RTP session and the receivers needs this source's identity,
   relation to other sources, or its synchronization context, all of
   which may be fully or partially identified using SDES items.  To
   enable this optimization, this document specifies a new RTP header
   extension that can carry SDES items.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-avtext-sdes-hdr-ext/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-avtext-sdes-hdr-ext/ballot/


No IPR declarations have been submitted directly on this I-D.



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Hi Jonathan,

This is not my preferred solution, but I don=E2=80=99t object to it =
going to WG last call.

Colin




> On 4 Mar 2016, at 22:47, Jonathan Lennox <jonathan@vidyo.com> wrote:
>=20
> Colin - can you send a note to the list as to whether this satisfies =
your comments on -00?  Yours are the only outstanding comments, so if so =
I=E2=80=99ll probably go ahead and issue a WGLC.
>=20
>> On Mar 3, 2016, at 5:03 PM, Adam Roach <adam@nostrum.com> wrote:
>>=20
>> As Peter, Colin, and I discussed on-list, this version of RID =
proposes a second identifier, RRID, that is suitable for unique =
identification of Redundancy RTP Streams, should that become necessary =
for some application in the future. The only other changes are =
editorial. Please review the diffs (linked below) and comment.
>>=20
>> Thanks!
>>=20
>> /a
>>=20
>> On 3/3/16 16:00, 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 Extensions of =
the IETF.
>>>=20
>>>        Title           : RTP Stream Identifier (RID) Source =
Description (SDES)
>>>        Authors         : Adam Roach
>>>                          Suhas Nandakumar
>>>                          Peter Thatcher
>>> 	Filename        : draft-ietf-avtext-rid-01.txt
>>> 	Pages           : 7
>>> 	Date            : 2016-03-03
>>>=20
>>> Abstract:
>>>   This document defines and registers an RTCP SDES item, RID, for
>>>   identification of RTP streams associated with Encoded Streams and
>>>   Dependent Streams.
>>>=20
>>>=20
>>> The IETF datatracker status page for this draft is:
>>> https://datatracker.ietf.org/doc/draft-ietf-avtext-rid/
>>>=20
>>> There's also a htmlized version available at:
>>> https://tools.ietf.org/html/draft-ietf-avtext-rid-01
>>>=20
>>> A diff from the previous version is available at:
>>> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-rid-01
>>>=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
>>> _______________________________________________
>>> avtext mailing list
>>> avtext@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avtext
>>=20
>> _______________________________________________
>> avtext mailing list
>> avtext@ietf.org
>> https://www.ietf.org/mailman/listinfo/avtext
>>=20
>=20



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





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On 3/6/16 06:14, Colin Perkins wrote:
> This is not my preferred solution

Can you expand on that?

The most recent version introduces two SDES items to allow unique 
identification of redundancy streams, which is an approach you described 
as "reasonable" back on February 17th.

Basically: the last round of edits were performed exclusively to address 
your concerns. If you think they don't, I'd like to either fix them or 
pull them out. Making the mechanism more complicated while also leaving 
you unsatisfied is a lose/lose situation.

/a


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> On 8 Mar 2016, at 20:48, Adam Roach <adam@nostrum.com> wrote:
>=20
> On 3/6/16 06:14, Colin Perkins wrote:
>> This is not my preferred solution
>=20
> Can you expand on that?
>=20
> The most recent version introduces two SDES items to allow unique =
identification of redundancy streams, which is an approach you described =
as "reasonable=E2=80=9D back on February 17th.

It=E2=80=99s a reasonable compromise, and I don=E2=80=99t object to the =
new approach, but I also don=E2=80=99t see any great benefit to the =
change.=20

> Basically: the last round of edits were performed exclusively to =
address your concerns. If you think they don't, I'd like to either fix =
them or pull them out. Making the mechanism more complicated while also =
leaving you unsatisfied is a lose/lose situation.

The introduction to the previous draft noted that none of the previous =
attempts to address this problem =E2=80=9Chave the proper ordinality to =
refer to an individual stream; all such identifiers can appear in more =
than one stream at a time=E2=80=9D, then defined a new identifier that =
can appear in more than one stream at a time. That is, it looked like =
the draft was aiming to solve the general problem, but it actually =
defined something much more narrow.=20

If you just want to solve the current narrowly defined problem, the =
approach taken in the previous draft makes more sense than the approach =
in the new draft (but the text suggesting it=E2=80=99s addressing the =
more general problem needs to be removed).

If you want to solve the general problem, then the new draft is a step =
in the right direction. It still retains some application specific =
semantics around redundancy, that I think will be problematic in future, =
though. Accordingly, I=E2=80=99m not sure the new draft goes far enough =
to be the general solution.

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




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Dear Rachel Huang,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 

avtext Session 1 (2:00:00)
    Thursday, Afternoon Session III 1730-1930
    Room Name: Buen Ayre A size: 125
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
Working Group Name: Audio/Video Transport Extensions
Area Name: Applications and Real-Time Area
Session Requester: Rachel Huang

Number of Sessions: 1
Length of Session(s):  2 Hours
Number of Attendees: 70
Conflicts to Avoid: 
 First Priority: avtcore payload xrblock mmusic perc rmcat dispatch netvc clue
 Second Priority: tsvwg sipcore lmap tram apparea



Special Requests:
  
---------------------------------------------------------


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
To: "Huangyihong (Rachel)" <rachel.huang@huawei.com>, "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, "draft-ietf-avtext-splicing-notification@ietf.org" <draft-ietf-avtext-splicing-notification@ietf.org>
References: <56D6E18F.8050603@ericsson.com> <51E6A56BD6A85142B9D172C87FC3ABBB86E8C95B@nkgeml513-mbs.china.huawei.com>
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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Hi,

Sorry for being a bit late in responding on this. I hope you have time 
to submit this before the cut-off on Monday.

Den 2016-03-03 kl. 11:51, skrev Huangyihong (Rachel):
> Hi Magnus,
>
> Thank you so much for your careful review and valuable comments. It
> helps us to further improve the document quality. Please see my
> replies inline.
>
> BR, Rachel
>
>
>> -----Original Message----- From: Magnus Westerlund
>> [mailto:magnus.westerlund@ericsson.com] Sent: Wednesday, March 02,
>> 2016 8:50 PM To: avtext@ietf.org; Ben Campbell;
>> draft-ietf-avtext-splicing-notification@ietf.org Subject: Some
>> comments on draft-ietf-avtext-splicing-notification-04
>>
>> Hi,
>>
>> This are some comments that came out of my review as an IANA expert
>> for the RTP Header Extensions and RTCP Control Packet Types, and
>> which of some is not related to the IANA expert review role.
>> However, I have noted that several of these, issues 1,2,3, and 5,
>> are related to protocol clearness as well as correct usage of RTP
>> which are review criteria for the RTP header extension registry.
>>
>>
>> 1. Section 1.1 and 3.1:
>>
>> What the NTP timestamp included in the extensions represent are not
>> being sufficiently made clear. I think it must be made clear that
>> the NTP timestamp represents the Main RTP senders wall clock
>> timestamp corresponding to these splicing in and out points. The
>> main RTP senders time base may have nothing with any formal time
>> basis distributed by the NTP protocol.
>
> [Rachel]: You're right. So how about changing the "splicing interval"
> in section 1.1 to the following:
>
> " Splicing Interval:
>
> The NTP-format timestamps, representing the main RTP sender wallclock
> time, for the Splicing-In point and Splicing-Out point per [RFC6828]
> allowing the splicer to know when to start and end the RTP splicing.
> " As for the ones in section 3.1, is it okay to just keep the
> "NTP-format timestamps", since we clarify it in the terminology?
>

Yes, I think so.

>>
>> 2. Section 3.1:
>>
>> If the 7 octets splicing-out NTP timestamp is smaller than the
>> lower 7 octets splicing-in NTP timestamp, it implies a wrap of the
>> 64-bit splicing-out NTP timestamp which will then be calculated by
>> the 7 octets splicing-out NTP timestamp plus 0x100000000.
>>
>> So I think "smaller" is a underspecified operation here. Although I
>> think loading the 7 octets into one 64 bit unsigned integer
>> respectively and perform a lesser than comparison will actually
>> work.
>
> [Rachel]: So how about expressing it like this
>
> OLD " The top 8 bits of the splicing-out NTP timestamp are inferred
> from the top 8 bits of the splicing-in NTP timestamp. This is
> unambiguous, under the assumption that the splicing-out time is after
> the splicing-in time, and the splicing interval is less than 2^25
> seconds. If the 7 octets splicing-out NTP timestamp is smaller than
> the lower 7 octets splicing-in NTP timestamp, it implies a wrap of
> the 64-bit splicing-out NTP timestamp which will then be calculated
> by the 7 octets splicing-out NTP timestamp plus 0x100000000.
> Otherwise, the top 8 octets of splicing-out NTP timestamp is equal to
> the top 8 octets of splicing-in NTP timestamp. "
>
> NEW " The top 8 bits of the splicing-out NTP timestamp are inferred
> from the top 8 bits of the splicing-in NTP timestamp. This is
> unambiguous, under the assumption that the splicing-out time is after
> the splicing-in time, and the splicing interval is less than 2^25
> seconds. Therefore, if the 7 octets splicing-out NTP-format timestamp
> is smaller than 64-bit splicing-out NTP-format timestamp modulo 56
> bits range, it implies a wrap of the 64-bit splicing-out NTP-format
> timestamp which will then be calculated by the 7 octets splicing-out
> NTP timestamp plus 2^64. Otherwise, the top 8 octets of splicing-out
> NTP timestamp is equal to the top 8 octets of splicing-in NTP
> timestamp. "

Hmm, this appears to be wrong about the wrapping. The wrapping that is 
the concern here that when the 56-bit value of the out is smaller than 
the 56-bit value of the splicing in point then a wrap of the 56-bit 
value has happened which will result in that one need to add 0x01 to the 
top 8-bit value copied from the splicing in point to the splicing out 
top 8-bit value.



>
>>
>> 3. Section 3.1:
>>
>> This section is not clear on what is the data part of the header
>> extension format. As it is not the header extension that can
>> determine if there will be usage of one-byte vs two-byte format
>> that will be used, this needs to be defined in a more un
>
> [Rachel]: I don't understand this. The editing seems unfinished. The
> splicing RTP header extension uses one-byte format with the fixed bit
> pattern 0xBEDE.

Sorry, what is missing is: The definition of the data field need to be 
done in a way that is agnostic to if 1 or 2-byte headers are being used 
in the header extension.

>
>>
>> 4. Section 3.2: The RTCP splicing notification message can be
>> appended to RTCP SR the main RTP sender generates in compound RTCP
>> packets, and hence follows the compound RTCP rules defined in
>> Section 6.1 in [RFC3550].
>>
>> I think the use of "append to RTCP SR" is unfortunate. I believe
>> what would be the right messages is to say that the splicing
>> notification messages is included in the same RTCP compound packet
>> as the RTCP SR for the main RTP stream. Using the SSRC values to
>> relate the two. This is important because there should be no formal
>> requirement on ordering between individual RTCP packets within a
>> compound packet.
>
> [Rachel]: Good point. How about changing to
>
> " The RTCP splicing notification message can be included in the RTCP
> compound packet together with RTCP SR generated at the main RTP
> sender, and hence follows the compound RTCP rules defined in Section
> 6.1 in [RFC3550]. "
>

Yes, that would work.

I realized now that you haven't expanded RTCP SR on its first use. 
Please do.

>>
>>
>> 5. Section 3.2:
>>
>> This format is similar to the RTCP Sender Report (Section 6.4.1 of
>> [RFC3550])
>>
>> Why is it only similar and not the same as in RFC  3550?
>>
>
> [Rachel]: Actually it's the same. Bad wording:(.
>
>>
>> 6. Section 3.2:
>>
>> If the use of non-compound RTCP [RFC5506] was previously
>> negotiated between the sender and the splicer, the RTCP splicing
>> notification message may be sent as non-compound RTCP packets.
>>
>> I wonder over this, I wonder if this RTCP packet type really should
>> be sent in an RTCP non-compound packet without the corresponding
>> RTCP SR packet? The reason I state this is that the RTP main sender
>> has to use SR/RR report block LSR field to determine which is the
>> latest received RTCP SR packet for the SSRC in the splicing RTCP
>> packet. And due to timing of RTCP packet transmission that can be
>> old information.  Thus, the splicer can operate on RTP to NTP
>> Timestamps that are not current. Thus, I think it would be a
>> recommendation to include the RTCP SR with the splicing packet. It
>> can still be a non-compound as one could leave out the SDES CNAME
>> items.
>
> [Rachel]: Maybe I got it wrong. But are you saying that the accuracy
> of mapping RTP timestamp to NTP timestamp, which should be deduced by
> the information in the RTCP SR that should be as close as the
> splicing starts? But the RTCP splicing message is sent far ahead of
> the splicing happens, and should be sent more than once to increase
> robustness. For example, may be several RTTs ahead. So in a
> non-compound case, it is possible that a RTCP SR will be sent after a
> RTCP splicing message, hence it seems no meaning to include RTCP SR
> with RTCP splicing messages.

So, yes things will work as long as the RTP to NTP mapping the in the 
splicer is correct. However, in some cases situations can occur that 
will modify the RTP to NTP mapping between the a previous sending of the 
Splicing message and the actual splicing start or stop.

So when does the RTP to NTP mapping change?

1. If the Payload Type changes and the clock rate changes and the system 
does not follow the recommendations in RFC 7160. However this is likely 
the least likely reason.

2. The more common reason will be clock drift. If the system wall clock 
progresses at a rate that is not matching the RTP timestamp rate, then 
periodically adjustments may occur. Thus changing the RTP to NTP mapping.

The issue with changing mapping is that one needs to know if it is the 
RTP timeline or the NTP wall clock that is the most relevant for when 
the splicing should occur. If it is the RTP one, due to that a specific 
amount of media samples needs to have been processed. In other words the 
boundary in the actual content is known and determines when the splice 
should happen. This I find likely for pre-recorded content.

But, I think I actually misrepresented the issue. I agree that it is not 
necessary that one have the SR and splicing messages in the same 
compound or reduced size RTCP packet. It is that one acknowledge what 
occurs if one do receive or not receive an update if the mapping has 
been updated between initial splicing information transmission and the 
implementation.

Different outcomes occur depending on which timescale is the primary one 
for when the splicing should occur.

If RTP timeline is the important. Then an update of the RTP to NTP 
mapping requires an updated NTP formated splicing point. In this case 
the best that one can do is send both Splicing info and RTCP SR. You 
need both to correctly splice.

If the NTP timeline is the one that matters, then an updated RTP to NTP 
mapping only needs the new values in the RTCP SR, no new splicing 
message is needed.

To conclude, what might be needed is actually rather a comment on that 
updated information may occur, and in certain cases both the splicing 
interval and the RTCP SR is needed to correctly splice.



>
>>
>> 7. Section 6.2:
>>
>> Only codecs that are supported both by the main RTP stream and the
>> substitutive RTP stream could be negotiated with SDP O/A. And the
>> splicer MUST choose the same codec for both of these two streams.
>>
>> So this payload type restriction is in my view in the wrong place
>> and not clear enough. So, to be able to splice into one consistent
>> outgoing stream there is a basic requirement that the RTP Payload
>> Types used are having the same clock rate, otherwise RFC 7160
>> applies, as uncertainty about how RTP maps to NTP and splicing
>> appears to be a bad mix.
>
> [Rachel]: Actually, this has been described in Section 2, 3rd
> paragraph. It says
>
> " To ensure the Splicing Interval is valid for both the main RTP
> sender and the substitutive RTP sender, the two senders MUST share a
> common reference clock so that the splicer can achieve accurate
> splicing. The requirements for the common reference clock (e.g.
> resolution, skew) depend on the codec used by the media content. "
>
>>
>> Further if one is going to express this in this more restrictive
>> way of demanding the same codes, then I do wonder if one actually
>> don't need to go as far as saying that the same payload type
>> configurations needs to be possible to us in both main and
>> substitute streams.
>
> [Rachel]: So you're suggesting that we delete this sentence?

Yes, the section you quote above already have a restriction using RFC 
2119 terms. Isn't that sufficient?

>
>>
>> 8. Section 7:
>>
>> I find a significant short coming in this security consideration.
>> That is the failure to discuss the importance for the splicer to be
>> able to trust the splicing interval information from the main RTP
>> sender. This information really should be authenticated and
>> integrity protected between the main RTP sender and the splicer.
>>
>> Thus, I am quite worried about this paragraph:
>>
>> Since the mechanism introduced in this document is not dependent
>> on any RTP payload-level hints for finding the splicing in and out
>> points, Secure Real-Time Transport Protocol (SRTP) [3711] can be
>> used to provide confidentiality of the RTP payload while leaving
>> the RTP header in the clear so that the splicer can still carry out
>> splicing without keying materials. However, for a malicious
>> endpoint without the key, it can observe the splicing time in the
>> RTP header, and it can intercept the substitutive content and even
>> replace it with a different one if the substitutive stream does not
>> use any security like SRTP and the splicer does not authenticate
>> the main and substitutive content sources.
>>
>> In the end it hints towards the need for authentication of the
>> streams. But that really should be worded in a stronger way as a
>> requirement for security.
>>
>> I do understand that this is really a case where two layers of
>> security would be good. One between the main and substitute sender
>> and the intended receiver, and an outer layer for sender to
>> splicer. However, I don't argue that such a solution needs to be
>> created. I think having the splicer as a trusted entity is more
>> likely to be a reasonable deployment model.
>
>
> [Rachel]: Okay. How about Add a new paragraph after the 3rd paragraph
> in Section 7?
>
> " To avoid above security issue, the authentication of the main and
> substitutive content sources in the splicer SHOULD be considered.
> However, separating the security mechanism between senders and the
> splicer from the end-to-end one may complicate the whole system
> deployment. In such a case, having the splicer as a trusted entity is
> likely to be a reasonable deployment model. "

No, I don't think this addresses the issue. The second paragraph already 
describes the trust model, of the trusted splicer. Especially as no 
alternative do yet exist. Although PERC WG exists and working on a model 
with end-to-end confidentiality, which could possibly work. That is not 
a great match either for what I think are the security requirements for 
services that uses splicing.

I think what needs to occur is to rewrite the third paragraph from the 
perspective that the splicer is trusted. Especially as the stream the 
splicer forwards will not match the received one from a perspective of 
number of packets and bytes, thus requiring RTCP changes. This is why a 
translator or mixer model is required. Thus, the packet forwarded from 
the splicer to the receiver will not be consistent with either the 
original or the substitution stream. For the authentication and 
integrity protection to not blow up the SRTP keys will be needed. A 
division of the roles of integrity versus confidentiality and you are 
back into PERC territory.

Thus, the discussion of header extension encryption is mote. Rather 
discuss that the splicer should use authenticated and integrity 
protected streams to prevent third parties from controlling the splicing 
as well as the substitution content.

>
>>
>> I do note that splicing do have the issue from key-management
>> perspective that both main and substitute sender needs to use the
>> same key if the security is from sender to receiver, rather than
>> splicer to receiver.
>
> [Rachel]: Yes. I plan to add a sentence in the paragraph 3 like
> this:
>
> OLD " Since the mechanism introduced in this document is not
> dependent on any RTP payload-level hints for finding the splicing in
> and out points, Secure Real-Time Transport Protocol (SRTP) [RFC3711]
> can be used to provide confidentiality of the RTP payload while
> leaving the RTP header in the clear so that the splicer can still
> carry out splicing without keying materials. "
>
> NEW " Since the mechanism introduced in this document is not
> dependent on any RTP payload-level hints for finding the splicing in
> and out points, Secure Real-Time Transport Protocol (SRTP) [RFC3711]
> can be used to provide confidentiality of the RTP payload while
> leaving the RTP header in the clear so that the splicer can still
> carry out splicing without keying materials. This implies that either
> the main and substitutive sender use the same SRTP key, or the
> splicer decrypts and re-encrypts with a single key. "
>

As I said above due to the SRTP keys for confidentiality and integrity 
are coming from a common master key, it is not possible to define a 
solution as indicated by the new text above.

I am sorry if I mislead you into the two layer security solution. It is 
a possibility, but require a lot of spec work. Because PERC will most 
likely not be sufficient for this usage.


Cheers


Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
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Färögatan 6                 | Mobile +46 73 0949079
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From nobody Mon Mar 21 03:16:00 2016
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, "draft-ietf-avtext-splicing-notification@ietf.org" <draft-ietf-avtext-splicing-notification@ietf.org>
Thread-Topic: Some comments on draft-ietf-avtext-splicing-notification-04
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Date: Mon, 21 Mar 2016 10:15:29 +0000
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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Hi Magnus,

Thanks for the response. I haven't checked the mail last weekend. Hope I ca=
n still have time to submit it^^. Please see inline.

BR,
Rachel

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, March 18, 2016 8:39 PM
> To: Huangyihong (Rachel); avtext@ietf.org; Ben Campbell;
> draft-ietf-avtext-splicing-notification@ietf.org
> Subject: Re: Some comments on draft-ietf-avtext-splicing-notification-04
>=20
> Hi,
>=20
> Sorry for being a bit late in responding on this. I hope you have time to=
 submit
> this before the cut-off on Monday.
>=20
> Den 2016-03-03 kl. 11:51, skrev Huangyihong (Rachel):
> > Hi Magnus,
> >
> > Thank you so much for your careful review and valuable comments. It
> > helps us to further improve the document quality. Please see my
> > replies inline.
> >
> > BR, Rachel
> >
> >
> >> -----Original Message----- From: Magnus Westerlund
> >> [mailto:magnus.westerlund@ericsson.com] Sent: Wednesday, March 02,
> >> 2016 8:50 PM To: avtext@ietf.org; Ben Campbell;
> >> draft-ietf-avtext-splicing-notification@ietf.org Subject: Some
> >> comments on draft-ietf-avtext-splicing-notification-04
> >>
> >> Hi,
> >>
> >> This are some comments that came out of my review as an IANA expert
> >> for the RTP Header Extensions and RTCP Control Packet Types, and
> >> which of some is not related to the IANA expert review role.
> >> However, I have noted that several of these, issues 1,2,3, and 5, are
> >> related to protocol clearness as well as correct usage of RTP which
> >> are review criteria for the RTP header extension registry.
> >>
> >>
> >> 1. Section 1.1 and 3.1:
> >>
> >> What the NTP timestamp included in the extensions represent are not
> >> being sufficiently made clear. I think it must be made clear that the
> >> NTP timestamp represents the Main RTP senders wall clock timestamp
> >> corresponding to these splicing in and out points. The main RTP
> >> senders time base may have nothing with any formal time basis
> >> distributed by the NTP protocol.
> >
> > [Rachel]: You're right. So how about changing the "splicing interval"
> > in section 1.1 to the following:
> >
> > " Splicing Interval:
> >
> > The NTP-format timestamps, representing the main RTP sender wallclock
> > time, for the Splicing-In point and Splicing-Out point per [RFC6828]
> > allowing the splicer to know when to start and end the RTP splicing.
> > " As for the ones in section 3.1, is it okay to just keep the
> > "NTP-format timestamps", since we clarify it in the terminology?
> >
>=20
> Yes, I think so.
>=20
> >>
> >> 2. Section 3.1:
> >>
> >> If the 7 octets splicing-out NTP timestamp is smaller than the lower
> >> 7 octets splicing-in NTP timestamp, it implies a wrap of the 64-bit
> >> splicing-out NTP timestamp which will then be calculated by the 7
> >> octets splicing-out NTP timestamp plus 0x100000000.
> >>
> >> So I think "smaller" is a underspecified operation here. Although I
> >> think loading the 7 octets into one 64 bit unsigned integer
> >> respectively and perform a lesser than comparison will actually work.
> >
> > [Rachel]: So how about expressing it like this
> >
> > OLD " The top 8 bits of the splicing-out NTP timestamp are inferred
> > from the top 8 bits of the splicing-in NTP timestamp. This is
> > unambiguous, under the assumption that the splicing-out time is after
> > the splicing-in time, and the splicing interval is less than 2^25
> > seconds. If the 7 octets splicing-out NTP timestamp is smaller than
> > the lower 7 octets splicing-in NTP timestamp, it implies a wrap of the
> > 64-bit splicing-out NTP timestamp which will then be calculated by the
> > 7 octets splicing-out NTP timestamp plus 0x100000000.
> > Otherwise, the top 8 octets of splicing-out NTP timestamp is equal to
> > the top 8 octets of splicing-in NTP timestamp. "
> >
> > NEW " The top 8 bits of the splicing-out NTP timestamp are inferred
> > from the top 8 bits of the splicing-in NTP timestamp. This is
> > unambiguous, under the assumption that the splicing-out time is after
> > the splicing-in time, and the splicing interval is less than 2^25
> > seconds. Therefore, if the 7 octets splicing-out NTP-format timestamp
> > is smaller than 64-bit splicing-out NTP-format timestamp modulo 56
> > bits range, it implies a wrap of the 64-bit splicing-out NTP-format
> > timestamp which will then be calculated by the 7 octets splicing-out
> > NTP timestamp plus 2^64. Otherwise, the top 8 octets of splicing-out
> > NTP timestamp is equal to the top 8 octets of splicing-in NTP
> > timestamp. "
>=20
> Hmm, this appears to be wrong about the wrapping. The wrapping that is th=
e
> concern here that when the 56-bit value of the out is smaller than the 56=
-bit
> value of the splicing in point then a wrap of the 56-bit value has happen=
ed
> which will result in that one need to add 0x01 to the top 8-bit value cop=
ied from
> the splicing in point to the splicing out top 8-bit value.


[Rachel]: You're right. The previous proposal is completely messed up. Here=
's the new proposal:

"
The top 8 bits of the splicing-out NTP timestamp are inferred
from the top 8 bits of the splicing-in NTP timestamp. This is
unambiguous, under the assumption that the splicing-out time is after
the splicing-in time, and the splicing interval is less than 2^25
seconds. Therefore, if the 7 octets splicing-out NTP-format timestamp
is smaller than the value of 7 low octets splicing-out NTP-format, it impli=
es a wrap of the 56-bit splicing-out NTP-format=09
timestamp which means the top 8-bit value of the 64-bit splicing-out is equ=
al to the top 8-bit value of splicing-in NTP
Timestamp plus 0x01. Otherwise, the top 8 bits of splicing-out
NTP timestamp is equal to the top 8 bits of splicing-in NTP
Timestamp.
"
>=20
>=20
>=20
> >
> >>
> >> 3. Section 3.1:
> >>
> >> This section is not clear on what is the data part of the header
> >> extension format. As it is not the header extension that can
> >> determine if there will be usage of one-byte vs two-byte format that
> >> will be used, this needs to be defined in a more un
> >
> > [Rachel]: I don't understand this. The editing seems unfinished. The
> > splicing RTP header extension uses one-byte format with the fixed bit
> > pattern 0xBEDE.
>=20
> Sorry, what is missing is: The definition of the data field need to be do=
ne in a
> way that is agnostic to if 1 or 2-byte headers are being used in the head=
er
> extension.
>

[Rachel]: I don't understand why should this extension to be agnostic for e=
xtension header types. There's no such limitation in [RFC5285] if I read it=
 correctly. And based on [RFC5285], it says:

"
   There are two variants of the extension: one-byte and two-byte
   headers.  Since it is expected that (a) the number of extensions in
   any given RTP session is small and (b) the extensions themselves are
   small, the one-byte header form is preferred and MUST be supported by
   all receivers.  A stream MUST contain only one-byte or two-byte
   headers: they MUST NOT be mixed within a stream.  Transmitters SHOULD
   NOT use the two-byte form when all extensions are small enough for
   the one-byte header form.
"

For splicing extension, it can be classified into a) case. That's why we us=
es one-byte header, and it is preferred and mandated by receivers. So I can=
't imagine any other cases to use 2-byte header. And by the way, [RFC6051] =
also only defines one-byte header.
=20
> >
> >>
> >> 4. Section 3.2: The RTCP splicing notification message can be
> >> appended to RTCP SR the main RTP sender generates in compound RTCP
> >> packets, and hence follows the compound RTCP rules defined in
> >> Section 6.1 in [RFC3550].
> >>
> >> I think the use of "append to RTCP SR" is unfortunate. I believe
> >> what would be the right messages is to say that the splicing
> >> notification messages is included in the same RTCP compound packet
> >> as the RTCP SR for the main RTP stream. Using the SSRC values to
> >> relate the two. This is important because there should be no formal
> >> requirement on ordering between individual RTCP packets within a
> >> compound packet.
> >
> > [Rachel]: Good point. How about changing to
> >
> > " The RTCP splicing notification message can be included in the RTCP
> > compound packet together with RTCP SR generated at the main RTP
> > sender, and hence follows the compound RTCP rules defined in Section
> > 6.1 in [RFC3550]. "
> >
>=20
> Yes, that would work.
>=20
> I realized now that you haven't expanded RTCP SR on its first use.
> Please do.

[Rachel]: Will do.

>=20
> >>
> >>
> >> 5. Section 3.2:
> >>
> >> This format is similar to the RTCP Sender Report (Section 6.4.1 of
> >> [RFC3550])
> >>
> >> Why is it only similar and not the same as in RFC  3550?
> >>
> >
> > [Rachel]: Actually it's the same. Bad wording:(.
> >
> >>
> >> 6. Section 3.2:
> >>
> >> If the use of non-compound RTCP [RFC5506] was previously
> >> negotiated between the sender and the splicer, the RTCP splicing
> >> notification message may be sent as non-compound RTCP packets.
> >>
> >> I wonder over this, I wonder if this RTCP packet type really should
> >> be sent in an RTCP non-compound packet without the corresponding
> >> RTCP SR packet? The reason I state this is that the RTP main sender
> >> has to use SR/RR report block LSR field to determine which is the
> >> latest received RTCP SR packet for the SSRC in the splicing RTCP
> >> packet. And due to timing of RTCP packet transmission that can be
> >> old information.  Thus, the splicer can operate on RTP to NTP
> >> Timestamps that are not current. Thus, I think it would be a
> >> recommendation to include the RTCP SR with the splicing packet. It
> >> can still be a non-compound as one could leave out the SDES CNAME
> >> items.
> >
> > [Rachel]: Maybe I got it wrong. But are you saying that the accuracy
> > of mapping RTP timestamp to NTP timestamp, which should be deduced by
> > the information in the RTCP SR that should be as close as the
> > splicing starts? But the RTCP splicing message is sent far ahead of
> > the splicing happens, and should be sent more than once to increase
> > robustness. For example, may be several RTTs ahead. So in a
> > non-compound case, it is possible that a RTCP SR will be sent after a
> > RTCP splicing message, hence it seems no meaning to include RTCP SR
> > with RTCP splicing messages.
>=20
> So, yes things will work as long as the RTP to NTP mapping the in the
> splicer is correct. However, in some cases situations can occur that
> will modify the RTP to NTP mapping between the a previous sending of the
> Splicing message and the actual splicing start or stop.
>=20
> So when does the RTP to NTP mapping change?
>=20
> 1. If the Payload Type changes and the clock rate changes and the system
> does not follow the recommendations in RFC 7160. However this is likely
> the least likely reason.
>=20
> 2. The more common reason will be clock drift. If the system wall clock
> progresses at a rate that is not matching the RTP timestamp rate, then
> periodically adjustments may occur. Thus changing the RTP to NTP mapping.
>=20
> The issue with changing mapping is that one needs to know if it is the
> RTP timeline or the NTP wall clock that is the most relevant for when
> the splicing should occur. If it is the RTP one, due to that a specific
> amount of media samples needs to have been processed. In other words the
> boundary in the actual content is known and determines when the splice
> should happen. This I find likely for pre-recorded content.
>=20
> But, I think I actually misrepresented the issue. I agree that it is not
> necessary that one have the SR and splicing messages in the same
> compound or reduced size RTCP packet. It is that one acknowledge what
> occurs if one do receive or not receive an update if the mapping has
> been updated between initial splicing information transmission and the
> implementation.
>=20
> Different outcomes occur depending on which timescale is the primary one
> for when the splicing should occur.
>=20
> If RTP timeline is the important. Then an update of the RTP to NTP
> mapping requires an updated NTP formated splicing point. In this case
> the best that one can do is send both Splicing info and RTCP SR. You
> need both to correctly splice.
>=20
> If the NTP timeline is the one that matters, then an updated RTP to NTP
> mapping only needs the new values in the RTCP SR, no new splicing
> message is needed.
>=20
> To conclude, what might be needed is actually rather a comment on that
> updated information may occur, and in certain cases both the splicing
> interval and the RTCP SR is needed to correctly splice.

[Rachel]: Okay. I get your point. So how about adding some texts as followi=
ng in Section 3.2, the last 2nd paragraph?

"
In cases of the mapping from RTP timestamp to NTP timestamp changes, e.g., =
clock drift happening before the splicing event, it may be required to send=
 RTCP SR or even updated Splicing Interval information timely to update the=
 timestamp mapping for accurate splicing.
"

>=20
>=20
>=20
> >
> >>
> >> 7. Section 6.2:
> >>
> >> Only codecs that are supported both by the main RTP stream and the
> >> substitutive RTP stream could be negotiated with SDP O/A. And the
> >> splicer MUST choose the same codec for both of these two streams.
> >>
> >> So this payload type restriction is in my view in the wrong place
> >> and not clear enough. So, to be able to splice into one consistent
> >> outgoing stream there is a basic requirement that the RTP Payload
> >> Types used are having the same clock rate, otherwise RFC 7160
> >> applies, as uncertainty about how RTP maps to NTP and splicing
> >> appears to be a bad mix.
> >
> > [Rachel]: Actually, this has been described in Section 2, 3rd
> > paragraph. It says
> >
> > " To ensure the Splicing Interval is valid for both the main RTP
> > sender and the substitutive RTP sender, the two senders MUST share a
> > common reference clock so that the splicer can achieve accurate
> > splicing. The requirements for the common reference clock (e.g.
> > resolution, skew) depend on the codec used by the media content. "
> >
> >>
> >> Further if one is going to express this in this more restrictive
> >> way of demanding the same codes, then I do wonder if one actually
> >> don't need to go as far as saying that the same payload type
> >> configurations needs to be possible to us in both main and
> >> substitute streams.
> >
> > [Rachel]: So you're suggesting that we delete this sentence?
>=20
> Yes, the section you quote above already have a restriction using RFC
> 2119 terms. Isn't that sufficient?
>=20
> >
> >>
> >> 8. Section 7:
> >>
> >> I find a significant short coming in this security consideration.
> >> That is the failure to discuss the importance for the splicer to be
> >> able to trust the splicing interval information from the main RTP
> >> sender. This information really should be authenticated and
> >> integrity protected between the main RTP sender and the splicer.
> >>
> >> Thus, I am quite worried about this paragraph:
> >>
> >> Since the mechanism introduced in this document is not dependent
> >> on any RTP payload-level hints for finding the splicing in and out
> >> points, Secure Real-Time Transport Protocol (SRTP) [3711] can be
> >> used to provide confidentiality of the RTP payload while leaving
> >> the RTP header in the clear so that the splicer can still carry out
> >> splicing without keying materials. However, for a malicious
> >> endpoint without the key, it can observe the splicing time in the
> >> RTP header, and it can intercept the substitutive content and even
> >> replace it with a different one if the substitutive stream does not
> >> use any security like SRTP and the splicer does not authenticate
> >> the main and substitutive content sources.
> >>
> >> In the end it hints towards the need for authentication of the
> >> streams. But that really should be worded in a stronger way as a
> >> requirement for security.
> >>
> >> I do understand that this is really a case where two layers of
> >> security would be good. One between the main and substitute sender
> >> and the intended receiver, and an outer layer for sender to
> >> splicer. However, I don't argue that such a solution needs to be
> >> created. I think having the splicer as a trusted entity is more
> >> likely to be a reasonable deployment model.
> >
> >
> > [Rachel]: Okay. How about Add a new paragraph after the 3rd paragraph
> > in Section 7?
> >
> > " To avoid above security issue, the authentication of the main and
> > substitutive content sources in the splicer SHOULD be considered.
> > However, separating the security mechanism between senders and the
> > splicer from the end-to-end one may complicate the whole system
> > deployment. In such a case, having the splicer as a trusted entity is
> > likely to be a reasonable deployment model. "
>=20
> No, I don't think this addresses the issue. The second paragraph already
> describes the trust model, of the trusted splicer. Especially as no
> alternative do yet exist. Although PERC WG exists and working on a model
> with end-to-end confidentiality, which could possibly work. That is not
> a great match either for what I think are the security requirements for
> services that uses splicing.
>=20
> I think what needs to occur is to rewrite the third paragraph from the
> perspective that the splicer is trusted. Especially as the stream the
> splicer forwards will not match the received one from a perspective of
> number of packets and bytes, thus requiring RTCP changes. This is why a
> translator or mixer model is required. Thus, the packet forwarded from
> the splicer to the receiver will not be consistent with either the
> original or the substitution stream. For the authentication and
> integrity protection to not blow up the SRTP keys will be needed. A
> division of the roles of integrity versus confidentiality and you are
> back into PERC territory.
>=20
> Thus, the discussion of header extension encryption is mote. Rather
> discuss that the splicer should use authenticated and integrity
> protected streams to prevent third parties from controlling the splicing
> as well as the substitution content.

[Rachel]: Here's my proposal to rewrite 3rd paragraph to discuss the authen=
tication between splicer and receiver. Is it what you're looking for?
"
Since splicer works as a trusted entity, any RTP-level or outside security =
mechanism, such IPsec[RFC4301] or Datagram Transport Layer Security [RFC634=
7], will use a security association between the splicer and the receiver. W=
hen using the Secure Real-Time Transport Protocol (SRTP) [RFC3711], the spl=
icer could be provisioned with the same security association as the main RT=
P sender.
"

>=20
> >
> >>
> >> I do note that splicing do have the issue from key-management
> >> perspective that both main and substitute sender needs to use the
> >> same key if the security is from sender to receiver, rather than
> >> splicer to receiver.
> >
> > [Rachel]: Yes. I plan to add a sentence in the paragraph 3 like
> > this:
> >
> > OLD " Since the mechanism introduced in this document is not
> > dependent on any RTP payload-level hints for finding the splicing in
> > and out points, Secure Real-Time Transport Protocol (SRTP) [RFC3711]
> > can be used to provide confidentiality of the RTP payload while
> > leaving the RTP header in the clear so that the splicer can still
> > carry out splicing without keying materials. "
> >
> > NEW " Since the mechanism introduced in this document is not
> > dependent on any RTP payload-level hints for finding the splicing in
> > and out points, Secure Real-Time Transport Protocol (SRTP) [RFC3711]
> > can be used to provide confidentiality of the RTP payload while
> > leaving the RTP header in the clear so that the splicer can still
> > carry out splicing without keying materials. This implies that either
> > the main and substitutive sender use the same SRTP key, or the
> > splicer decrypts and re-encrypts with a single key. "
> >
>=20
> As I said above due to the SRTP keys for confidentiality and integrity
> are coming from a common master key, it is not possible to define a
> solution as indicated by the new text above.
>=20
> I am sorry if I mislead you into the two layer security solution. It is
> a possibility, but require a lot of spec work. Because PERC will most
> likely not be sufficient for this usage.
>=20
>=20
> Cheers
>=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> 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: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'avtext@ietf.org'" <avtext@ietf.org>, "'Ben Campbell'" <ben@nostrum.com>, "'draft-ietf-avtext-splicing-notification@ietf.org'" <draft-ietf-avtext-splicing-notification@ietf.org>
Thread-Topic: Some comments on draft-ietf-avtext-splicing-notification-04
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Date: Mon, 21 Mar 2016 13:52:05 +0000
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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From nobody Mon Mar 21 07:28:58 2016
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To: "Huangyihong (Rachel)" <rachel.huang@huawei.com>, "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, "draft-ietf-avtext-splicing-notification@ietf.org" <draft-ietf-avtext-splicing-notification@ietf.org>
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Date: Mon, 21 Mar 2016 15:27:49 +0100
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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Hi,

Please see inline. I have removed issues that I think are all fine and
you can simply incorporate them. From my perspective, I think it is best 
that you submit an update prior to the dead-line, then we all can review 
the changes prior to the meeting. And if any changes are needed, you can 
update in 2 weeks time.

Please see inline.


Den 2016-03-21 kl. 11:15, skrev Huangyihong (Rachel):
>> -----Original Message----- From: Magnus Westerlund
>> [mailto:magnus.westerlund@ericsson.com] Sent: Friday, March 18,
>> 2016 8:39 PM To: Huangyihong (Rachel); avtext@ietf.org; Ben
>> Campbell; draft-ietf-avtext-splicing-notification@ietf.org Subject:
>> Re: Some comments on draft-ietf-avtext-splicing-notification-04
>>
>> Hi,
>>
>> Sorry for being a bit late in responding on this. I hope you have
>> time to submit this before the cut-off on Monday.
>>
>> Den 2016-03-03 kl. 11:51, skrev Huangyihong (Rachel):
>>>>
>>>> 2. Section 3.1:
>>>>
> " The top 8 bits of the splicing-out NTP timestamp are inferred from
> the top 8 bits of the splicing-in NTP timestamp. This is unambiguous,
> under the assumption that the splicing-out time is after the
> splicing-in time, and the splicing interval is less than 2^25
> seconds. Therefore, if the 7 octets splicing-out NTP-format
> timestamp is smaller than the value of 7 low octets splicing-out
> NTP-format, it implies a wrap of the 56-bit splicing-out NTP-format
> timestamp which means the top 8-bit value of the 64-bit splicing-out
> is equal to the top 8-bit value of splicing-in NTP Timestamp plus
> 0x01. Otherwise, the top 8 bits of splicing-out NTP timestamp is
> equal to the top 8 bits of splicing-in NTP Timestamp. "

Yes, with the correction you added I think this is fine.

>>
>>
>>
>>>
>>>>
>>>> 3. Section 3.1:
>>>>
>>>> This section is not clear on what is the data part of the
>>>> header extension format. As it is not the header extension that
>>>> can determine if there will be usage of one-byte vs two-byte
>>>> format that will be used, this needs to be defined in a more
>>>> un
>>>
>>> [Rachel]: I don't understand this. The editing seems unfinished.
>>> The splicing RTP header extension uses one-byte format with the
>>> fixed bit pattern 0xBEDE.
>>
>> Sorry, what is missing is: The definition of the data field need to
>> be done in a way that is agnostic to if 1 or 2-byte headers are
>> being used in the header extension.
>>
>
> [Rachel]: I don't understand why should this extension to be agnostic
> for extension header types. There's no such limitation in [RFC5285]
> if I read it correctly. And based on [RFC5285], it says:
>
> " There are two variants of the extension: one-byte and two-byte
> headers.  Since it is expected that (a) the number of extensions in
> any given RTP session is small and (b) the extensions themselves are
> small, the one-byte header form is preferred and MUST be supported
> by all receivers.  A stream MUST contain only one-byte or two-byte
> headers: they MUST NOT be mixed within a stream.  Transmitters
> SHOULD NOT use the two-byte form when all extensions are small enough
> for the one-byte header form. "
>
> For splicing extension, it can be classified into a) case. That's why
> we uses one-byte header, and it is preferred and mandated by
> receivers. So I can't imagine any other cases to use 2-byte header.
> And by the way, [RFC6051] also only defines one-byte header.

So this debate do gets us into RFC5285 update country. However the issue 
is that although a particular RTP header extension has a preference or 
requirement for which format to use. The actual format used in an actual 
RTP packet will be depending on the union of all the requirement for 
header extension format. For example the SDES header extension is one 
that can require to use the 2-byte format due to the SDES item's length 
(>16 bytes). If that header extension and yours needs to be in the same 
packet, then also yours need to use the 2-byte header format in this 
particular packet.

This is issue that needs to be clarified in RFC 5285 update, and 
hopefully to allow a particular SSRC to switch between formats on a 
packet by packet basis. If not then we end up in a situation where hard 
choices will have to be done when conflicting requirements occur.

The above issue is from my perspective that if one defines a header 
extension in such a way that it is clear on what is the data part and 
don't lock down the format to use, then it will work with the updated spec.


>>>>
>>>> 6. Section 3.2:
>>>>
>
> [Rachel]: Okay. I get your point. So how about adding some texts as
> following in Section 3.2, the last 2nd paragraph?
>
> " In cases of the mapping from RTP timestamp to NTP timestamp
> changes, e.g., clock drift happening before the splicing event, it
> may be required to send RTCP SR or even updated Splicing Interval
> information timely to update the timestamp mapping for accurate
> splicing. "
>

Yes, this at least raises the issue.


>>>
>>>>
>>>> 8. Section 7:
>>>>
>>>> I find a significant short coming in this security
>>>> consideration. That is the failure to discuss the importance
>>>> for the splicer to be able to trust the splicing interval
>>>> information from the main RTP sender. This information really
>>>> should be authenticated and integrity protected between the
>>>> main RTP sender and the splicer.
>>>>
>>>> Thus, I am quite worried about this paragraph:
>>>>
>>>> Since the mechanism introduced in this document is not
>>>> dependent on any RTP payload-level hints for finding the
>>>> splicing in and out points, Secure Real-Time Transport Protocol
>>>> (SRTP) [3711] can be used to provide confidentiality of the RTP
>>>> payload while leaving the RTP header in the clear so that the
>>>> splicer can still carry out splicing without keying materials.
>>>> However, for a malicious endpoint without the key, it can
>>>> observe the splicing time in the RTP header, and it can
>>>> intercept the substitutive content and even replace it with a
>>>> different one if the substitutive stream does not use any
>>>> security like SRTP and the splicer does not authenticate the
>>>> main and substitutive content sources.
>>>>
>>>> In the end it hints towards the need for authentication of the
>>>> streams. But that really should be worded in a stronger way as
>>>> a requirement for security.
>>>>
>>>> I do understand that this is really a case where two layers of
>>>> security would be good. One between the main and substitute
>>>> sender and the intended receiver, and an outer layer for sender
>>>> to splicer. However, I don't argue that such a solution needs
>>>> to be created. I think having the splicer as a trusted entity
>>>> is more likely to be a reasonable deployment model.
>>>
>>>
>>> [Rachel]: Okay. How about Add a new paragraph after the 3rd
>>> paragraph in Section 7?
>>>
>>> " To avoid above security issue, the authentication of the main
>>> and substitutive content sources in the splicer SHOULD be
>>> considered. However, separating the security mechanism between
>>> senders and the splicer from the end-to-end one may complicate
>>> the whole system deployment. In such a case, having the splicer
>>> as a trusted entity is likely to be a reasonable deployment
>>> model. "
>>
>> No, I don't think this addresses the issue. The second paragraph
>> already describes the trust model, of the trusted splicer.
>> Especially as no alternative do yet exist. Although PERC WG exists
>> and working on a model with end-to-end confidentiality, which could
>> possibly work. That is not a great match either for what I think
>> are the security requirements for services that uses splicing.
>>
>> I think what needs to occur is to rewrite the third paragraph from
>> the perspective that the splicer is trusted. Especially as the
>> stream the splicer forwards will not match the received one from a
>> perspective of number of packets and bytes, thus requiring RTCP
>> changes. This is why a translator or mixer model is required. Thus,
>> the packet forwarded from the splicer to the receiver will not be
>> consistent with either the original or the substitution stream. For
>> the authentication and integrity protection to not blow up the SRTP
>> keys will be needed. A division of the roles of integrity versus
>> confidentiality and you are back into PERC territory.
>>
>> Thus, the discussion of header extension encryption is mote.
>> Rather discuss that the splicer should use authenticated and
>> integrity protected streams to prevent third parties from
>> controlling the splicing as well as the substitution content.
>
> [Rachel]: Here's my proposal to rewrite 3rd paragraph to discuss the
> authentication between splicer and receiver. Is it what you're
> looking for? " Since splicer works as a trusted entity, any RTP-level
> or outside security mechanism, such IPsec[RFC4301] or Datagram
> Transport Layer Security [RFC6347], will use a security association
> between the splicer and the receiver. When using the Secure Real-Time
> Transport Protocol (SRTP) [RFC3711], the splicer could be provisioned
> with the same security association as the main RTP sender. "

Yes, it is at least what I believe to be most important. This as we are 
currently lacking the tools for doing a model where the splicer is only 
semi-trusted.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Mon Mar 21 07:36:41 2016
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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On 21 Mar 2016, at 9:27, Magnus Westerlund wrote:

> From my perspective, I think it is best that you submit an update 
> prior to the dead-line, then we all can review the changes prior to 
> the meeting. And if any changes are needed, you can update in 2 weeks 
> time.

I agree.

Thanks,

Ben.


From nobody Mon Mar 21 09:00:48 2016
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Subject: [avtext] I-D Action: draft-ietf-avtext-lrr-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Extensions of the IETF.

        Title           : The Layer Refresh Request (LRR) RTCP Feedback Message
        Authors         : Jonathan Lennox
                          Danny Hong
                          Justin Uberti
                          Stefan Holmer
                          Magnus Flodman
	Filename        : draft-ietf-avtext-lrr-02.txt
	Pages           : 14
	Date            : 2016-03-21

Abstract:
   This memo describes the RTCP Payload-Specific Feedback Message "Layer
   Refresh Request" (LRR), which can be used to request a state refresh
   of one or more substreams of a layered media stream.  It also defines
   its use with several RTP payloads for scalable media formats.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-avtext-lrr-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-avtext-lrr-02


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 nobody Mon Mar 21 09:07:04 2016
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, "draft-ietf-avtext-splicing-notification@ietf.org" <draft-ietf-avtext-splicing-notification@ietf.org>
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From nobody Mon Mar 21 09:08:01 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Extensions of the IETF.

        Title           : RTP/RTCP extension for RTP Splicing Notification
        Authors         : Jinwei Xia
                          Roni Even
                          Rachel Huang
                          Lingli Deng
	Filename        : draft-ietf-avtext-splicing-notification-05.txt
	Pages           : 19
	Date            : 2016-03-21

Abstract:
   Content splicing is a process that replaces the content of a main
   multimedia stream with other multimedia content, and delivers the
   substitutive multimedia content to the receivers for a period of
   time. The splicer is designed to handle RTP splicing and needs to
   know when to start and end the splicing.

   This memo defines two RTP/RTCP extensions to indicate the splicing
   related information to the splicer: an RTP header extension that
   conveys the information in-band and an RTCP packet that conveys the
   information out-of-band.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtext-splicing-notification/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-avtext-splicing-notification-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-avtext-splicing-notification-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.

Internet-Drafts are also available by anonymous FTP at:
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From nobody Mon Mar 21 09:16:16 2016
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Thread-Topic: [avtext] I-D Action: draft-ietf-avtext-lrr-02.txt
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This version of the LRR draft takes out the VP9 definitions, as we discusse=
d in Yokohama, to move them to the VP9 payload spec.

It also formally defines the message semantics, and adds SDP definitions an=
d IANA registrations.=20

I believe this document is now complete.  Comments welcome.

> On Mar 21, 2016, at 6:00 PM, Internet-Drafts@ietf.org wrote:
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
> This draft is a work item of the Audio/Video Transport Extensions of the =
IETF.
>=20
>        Title           : The Layer Refresh Request (LRR) RTCP Feedback Me=
ssage
>        Authors         : Jonathan Lennox
>                          Danny Hong
>                          Justin Uberti
>                          Stefan Holmer
>                          Magnus Flodman
> 	Filename        : draft-ietf-avtext-lrr-02.txt
> 	Pages           : 14
> 	Date            : 2016-03-21
>=20
> Abstract:
>   This memo describes the RTCP Payload-Specific Feedback Message "Layer
>   Refresh Request" (LRR), which can be used to request a state refresh
>   of one or more substreams of a layered media stream.  It also defines
>   its use with several RTP payloads for scalable media formats.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtext-lrr/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-avtext-lrr-02
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtext-lrr-02
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>=20


From nobody Mon Mar 21 16:17:52 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Extensions of the IETF.

        Title           : Frame Marking RTP Header Extension
        Authors         : Espen Berger
                          Suhas Nandakumar
                          Mo Zanaty
	Filename        : draft-ietf-avtext-framemarking-01.txt
	Pages           : 8
	Date            : 2016-03-21

Abstract:
   This document describes a Frame Marking RTP header extension used to
   convey information about video frames that is critical for error
   recovery and packet forwarding in RTP middleboxes or network nodes.
   It is most useful when media is encrypted, and essential when the
   middlebox or node has no access to the media encryption keys.  It is
   also useful for codec-agnostic processing of encrypted or unencrypted
   media, while it also supports extensions for codec-specific
   information.


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

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From nobody Mon Mar 21 19:14:35 2016
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From nobody Tue Mar 22 03:35:44 2016
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To: "Huangyihong (Rachel)" <rachel.huang@huawei.com>, "avtext@ietf.org" <avtext@ietf.org>, Ben Campbell <ben@nostrum.com>, "draft-ietf-avtext-splicing-notification@ietf.org" <draft-ietf-avtext-splicing-notification@ietf.org>
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Date: Tue, 22 Mar 2016 11:35:37 +0100
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Den 2016-03-21 kl. 17:06, skrev Huangyihong (Rachel):
> Hi Magus,
>
> Thank you for the quick response. I think below is the only issue
> that we're discussing. I'll submit a new version to reflect the
> issues you have.
>
> BR, Rachel
>
>>>>>
>>>>> 3. Section 3.1:
>>>>>
>>>>> This section is not clear on what is the data part of the
>>>>> header extension format. As it is not the header extension
>>>>> that can determine if there will be usage of one-byte vs
>>>>> two-byte format that will be used, this needs to be defined
>>>>> in a more un
>>>>
>>>> [Rachel]: I don't understand this. The editing seems
>>>> unfinished. The splicing RTP header extension uses one-byte
>>>> format with the fixed bit pattern 0xBEDE.
>>>
>>> Sorry, what is missing is: The definition of the data field need
>>> to be done in a way that is agnostic to if 1 or 2-byte headers
>>> are being used in the header extension.
>>>
>>
>> [Rachel]: I don't understand why should this extension to be
>> agnostic for extension header types. There's no such limitation in
>> [RFC5285] if I read it correctly. And based on [RFC5285], it says:
>>
>> " There are two variants of the extension: one-byte and two-byte
>> headers.  Since it is expected that (a) the number of extensions
>> in any given RTP session is small and (b) the extensions themselves
>> are small, the one-byte header form is preferred and MUST be
>> supported by all receivers.  A stream MUST contain only one-byte or
>> two-byte headers: they MUST NOT be mixed within a stream.
>> Transmitters SHOULD NOT use the two-byte form when all extensions
>> are small enough for the one-byte header form. "
>>
>> For splicing extension, it can be classified into a) case. That's
>> why we uses one-byte header, and it is preferred and mandated by
>> receivers. So I can't imagine any other cases to use 2-byte
>> header. And by the way, [RFC6051] also only defines one-byte
>> header.
>
> So this debate do gets us into RFC5285 update country. However the
> issue is that although a particular RTP header extension has a
> preference or requirement for which format to use. The actual format
> used in an actual RTP packet will be depending on the union of all
> the requirement for header extension format. For example the SDES
> header extension is one that can require to use the 2-byte format due
> to the SDES item's length (>16 bytes). If that header extension and
> yours needs to be in the same packet, then also yours need to use the
> 2-byte header format in this particular packet.
>
> [Rachel]: The RFC5285 update already permits to use 1-byte and 2-byte
> format in the same RTP stream. You're saying the case they're used in
> one RTP packet. For splicing header extension, it only appears a
> number of RTP packet instead of all the RTP packets. So I'm just
> curious the possibility of the 2 kinds of extensions must be in the
> same packet.

No, that is not what I try to say. I am saying that within one RTP 
stream, i.e. under one SSRC. Then in one packet one might need to use 
the 2-byte format due to that an additional RTP header extension is 
added to the one with Splicing information. In a later packet where 
there is need to send a splicing RTP header extension, there might not 
be any other RTP header extension, nor one that requires the two-byte 
header extension. Thus, this packet can be sent with the 1-byte format.

>
> This is issue that needs to be clarified in RFC 5285 update, and
> hopefully to allow a particular SSRC to switch between formats on a
> packet by packet basis. If not then we end up in a situation where
> hard choices will have to be done when conflicting requirements
> occur.
>
> The above issue is from my perspective that if one defines a header
> extension in such a way that it is clear on what is the data part and
> don't lock down the format to use, then it will work with the updated
> spec.
>
> [Rachel]: It's no harm to include a 2-byte format in this version to
> avoid the case that you mentioned. And I'll do it right now.

Yes, that is one way of doing it. The point I tried to made is that one 
can define ones RTP header extension is such a way that it becomes 
agnostic to if 1-byte or 2-byte format is used for the header 
extensions. This should be the recommended practice unless the format 
requires the 2-byte format.

I will have to review the RFC 5285 update to see if this has sufficient 
clarifications in this area. I would recommend that also the rest of the 
WG do take a look on the document update.

cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FĂ¤rĂ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Wed Mar 23 01:18:52 2016
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Thread-Topic: WGLC: draft-ietf-avtext-rid-01.txt
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This is to announce a 1.5 week Working Group Last Call for

	draft-ietf-avtext-rid-01

as proposed standard.

Please review and provide any comments you may have on the document by Sund=
ay, April 3 (the beginning of the Buenos Aires IETF meeting). Comments shou=
ld be sent to the document authors and the AVTEXT WG list.

If you review the document but do not have any comments, please send a note=
 to that effect as well.

Thank you!

        Jonathan (AVTEXT co-chair)


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From nobody Wed Mar 23 23:26:32 2016
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Thread-Topic: [avtext] Draft agenda for AVTEXT meeting in Buenos Aires
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Date: Thu, 24 Mar 2016 06:26:20 +0000
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Cc: Jonathan Lennox <jonathan@vidyo.com>
Subject: [avtext]  Draft agenda for AVTEXT meeting in Buenos Aires
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Dear all,

Here's the draft agenda for AVTEXT meeting in Buenos Aires. And I already u=
ploaded it.
https://www.ietf.org/proceedings/95/agenda/agenda-95-avtext
So if you have any changes or other things to discuss, please let Jonathan =
or I know. Thanks.

BR,
Rachel

AVTEXT Audio/Video Transport Extensions WG Meeting
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D
Date and Time: THURSDAY 4/7/16, Afternoon Session III, 17:30-19:30
Room: Buen Ayre A
Chairs: Jonathan Lennox / Rachel Huang

1. Note Well, Note Takers, Agenda Bashing - (Chairs, 17:30, 5 min)

2. WG Status - (Chairs, 17:35, 5 min)

   Status of working group draft:

      draft-ieft-avtext-lrr-02            [ready for WGLC]
      draft-ietf-avtext-rid-01            [ready for WGLC]
      draft-ietf-avtext-sdes-hdr-ext-05   [IETF LC]
      draft-ietf-avtext-splicing-notification-05 [IANA review]

3. Chartered work:

   Frame Marking RTP Header Extension (Mo Zanaty, 17:40, 30 min)

      draft-ietf-avtext-framemarking-01

   Codec Control Messages for Layered (Stephan Wenger, 18:10, 20 min)

      draft-wenger-avtext-avpf-ccm-layered-00

4. Possible topic from RMCAT - (tbd, 18:30, 20 min)

5. Next steps and open mic - (Chairs, 18:50, 10 min)



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<p class=3D"MsoNormal"><span lang=3D"EN-US">Dear all,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Here&#8217;s the draft agenda f=
or AVTEXT meeting
</span><span lang=3D"EN-US" style=3D"color:black">in Buenos Aires</span><sp=
an lang=3D"EN-US">. And I already uploaded it.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><a href=3D"https://www.ietf.org=
/proceedings/95/agenda/agenda-95-avtext">https://www.ietf.org/proceedings/9=
5/agenda/agenda-95-avtext</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">So if you have any changes or o=
ther things to discuss, please let Jonathan or I know. Thanks.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">BR,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Rachel<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">AVTEXT A=
udio/Video Transport Extensions WG Meeting<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">Date and=
 Time: THURSDAY 4/7/16, Afternoon Session III, 17:30-19:30
<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">Room: Bu=
en Ayre A<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">Chairs: =
Jonathan Lennox / Rachel Huang<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">1. Note =
Well, Note Takers, Agenda Bashing - (Chairs, 17:30, 5 min)<o:p></o:p></span=
></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">2. WG St=
atus - (Chairs, 17:35, 5 min)<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp; Status of working group draft:<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; draft-ieft-avtext-lrr-02&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [ready for WGLC]<o:p></o:p></span></=
p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; draft-ietf-avtext-rid-01&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [ready for WGLC]<o:p></o:p></span></=
p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; draft-ietf-avtext-sdes-hdr-ext-05&nbsp;&nbsp; [IETF =
LC]<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; draft-ietf-avtext-splicing-notification-05 [IANA rev=
iew]<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">3. Chart=
ered work:<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp; Frame Marking RTP Header Extension (Mo Zanaty, 17:40, 30 min)<o:p></o:=
p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; draft-ietf-avtext-framemarking-01<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp; Codec Control Messages for Layered (Stephan Wenger, 18:10, 20 min)<o:p=
></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; draft-wenger-avtext-avpf-ccm-layered-00<o:p></o:p></=
span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">4. Possi=
ble topic from RMCAT - (tbd, 18:30, 20 min)<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;"><o:p>&nb=
sp;</o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:12.0pt;font-family:&#23435;&#20307;">5. Next =
steps and open mic - (Chairs, 18:50, 10 min)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
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From: Bo Burman <bo.burman@ericsson.com>
To: Jonathan Lennox <jonathan@vidyo.com>, "avtext@ietf.org" <avtext@ietf.org>
Thread-Topic: WGLC: draft-ietf-avtext-rid-01.txt
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Subject: Re: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
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I've reviewed this and have the following comments:

1) In section 3 you use capitalized versions of terms from RFC7656. One of =
the late changes in the publication process of RFC7656 was to not use capit=
alized terms, so you might want to align to non-capitalized versions. No st=
rong opinion on that one, though.

2) Would it be correct, semantically, to add the following text at the end =
of the second paragraph in section 3; "...to refer to this construct of a s=
ource RTP stream that originates either from an encoded stream or a depende=
nt stream, but is not a redundancy RTP stream"? If so, I suggest to make th=
at clarification, or otherwise change text in that paragraph to make it cle=
ar.

3) To align with a recent update of draft-ietf-avtext-sdes-hdr-ext, IANA re=
gistration of RID and RRID in RTP SDES Compact Header Extensions registry s=
hould also be added to sections 5.1 and 5.2, respectively, like:

5.1:
...
   It is requested that the following SDES item is registered in the
   RTP SDES Compact Header Extensions registry:

   Extension URI: urn:ietf:params:rtp-hdrext:sdes:rid
   Description:   Source Description: RTP Stream Identifier
                  (SDES RID)
   Contact:       Authors of [RFCXXXX]
   Reference:     [RFCXXXX]

5.2:
...
   It is requested that the following SDES item is registered in the
   RTP SDES Compact Header Extensions registry:

   Extension URI: urn:ietf:params:rtp-hdrext:sdes:rrid
   Description:   Source Description: Redundancy RTP Stream Identifier
                  (SDES RRID)
   Contact:       Authors of [RFCXXXX]
   Reference:     [RFCXXXX]

Cheers,
Bo

> -----Original Message-----
> From: avtext [mailto:avtext-bounces@ietf.org] On Behalf Of Jonathan Lenno=
x
> Sent: den 23 mars 2016 09:19
> To: avtext@ietf.org
> Subject: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
>=20
> This is to announce a 1.5 week Working Group Last Call for
>=20
> 	draft-ietf-avtext-rid-01
>=20
> as proposed standard.
>=20
> Please review and provide any comments you may have on the document by Su=
nday, April 3 (the beginning of the
> Buenos Aires IETF meeting). Comments should be sent to the document autho=
rs and the AVTEXT WG list.
>=20
> If you review the document but do not have any comments, please send a no=
te to that effect as well.
>=20
> Thank you!
>=20
>         Jonathan (AVTEXT co-chair)
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext


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Date: Tue, 29 Mar 2016 09:25:52 -0700
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--94eb2c123344c56b4c052f327c53
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I have reviewed the document and have the following comment:

Need for RRID.  Currently, draft-ietf-mmusic-rid does not make use of the
RRID.  As I understand it, no WebRTC browsers are currently planning to
implement the RRID.  Furthermore, neither the WebRTC 1.0 API nor ORTC
supports the RRID.

Given this, do we really need to include the RRID in this document?





Minor editorial issues below



On Wed, Mar 23, 2016 at 1:18 AM, Jonathan Lennox <jonathan@vidyo.com> wrote:

> This is to announce a 1.5 week Working Group Last Call for
>
>         draft-ietf-avtext-rid-01
>
> as proposed standard.
>
> Please review and provide any comments you may have on the document by
> Sunday, April 3 (the beginning of the Buenos Aires IETF meeting). Comments
> should be sent to the document authors and the AVTEXT WG list.
>
> If you review the document but do not have any comments, please send a
> note to that effect as well.
>
> Thank you!
>
>         Jonathan (AVTEXT co-chair)
>
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>

--94eb2c123344c56b4c052f327c53
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<div dir=3D"ltr">I have reviewed the document and have the following commen=
t:=C2=A0<div><br></div><div>Need for RRID.=C2=A0 Currently, draft-ietf-mmus=
ic-rid does not make use of the RRID.=C2=A0 As I understand it, no WebRTC b=
rowsers are currently planning to implement the RRID.=C2=A0 Furthermore, ne=
ither the WebRTC 1.0 API nor ORTC supports the RRID. =C2=A0=C2=A0</div><div=
><br></div><div>Given this, do we really need to include the RRID in this d=
ocument? =C2=A0</div><div><div><br></div><div><br><div><br></div><div><br><=
/div><div><br></div><div>Minor editorial issues below<br><div><br></div><di=
v><br></div></div></div></div></div><div class=3D"gmail_extra"><br><div cla=
ss=3D"gmail_quote">On Wed, Mar 23, 2016 at 1:18 AM, Jonathan Lennox <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:jonathan@vidyo.com" target=3D"_blank">jona=
than@vidyo.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">This=
 is to announce a 1.5 week Working Group Last Call for<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 draft-ietf-avtext-rid-01<br>
<br>
as proposed standard.<br>
<br>
Please review and provide any comments you may have on the document by Sund=
ay, April 3 (the beginning of the Buenos Aires IETF meeting). Comments shou=
ld be sent to the document authors and the AVTEXT WG list.<br>
<br>
If you review the document but do not have any comments, please send a note=
 to that effect as well.<br>
<br>
Thank you!<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Jonathan (AVTEXT co-chair)<br>
<br>
_______________________________________________<br>
avtext mailing list<br>
<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avtext" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/avtext</a><br>
</blockquote></div><br></div>

--94eb2c123344c56b4c052f327c53--


From nobody Tue Mar 29 14:35:07 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: Huangyihong <rachel.huang@huawei.com>, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [avtext] Some comments on draft-ietf-avtext-splicing-notification-04
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Hi,

Where have we landed with this? Will a new revision be required?

Thanks!

Ben.

On 23 Mar 2016, at 4:29, Huangyihong (Rachel) wrote:

> Hi Magnus,
>
> Inline please.
>
> BR,
> Rachel
>
>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Tuesday, March 22, 2016 6:36 PM
>> To: Huangyihong (Rachel); avtext@ietf.org; Ben Campbell;
>> draft-ietf-avtext-splicing-notification@ietf.org
>> Subject: Re: ç­”ĺ¤Ť: Some comments on 
>> draft-ietf-avtext-splicing-notification-04
>>
>> Den 2016-03-21 kl. 17:06, skrev Huangyihong (Rachel):
>>> Hi Magus,
>>>
>>> Thank you for the quick response. I think below is the only issue 
>>> that
>>> we're discussing. I'll submit a new version to reflect the issues 
>>> you
>>> have.
>>>
>>> BR, Rachel
>>>
>>>>>>>
>>>>>>> 3. Section 3.1:
>>>>>>>
>>>>>>> This section is not clear on what is the data part of the header
>>>>>>> extension format. As it is not the header extension that can
>>>>>>> determine if there will be usage of one-byte vs two-byte format
>>>>>>> that will be used, this needs to be defined in a more un
>>>>>>
>>>>>> [Rachel]: I don't understand this. The editing seems unfinished.
>>>>>> The splicing RTP header extension uses one-byte format with the
>>>>>> fixed bit pattern 0xBEDE.
>>>>>
>>>>> Sorry, what is missing is: The definition of the data field need 
>>>>> to
>>>>> be done in a way that is agnostic to if 1 or 2-byte headers are
>>>>> being used in the header extension.
>>>>>
>>>>
>>>> [Rachel]: I don't understand why should this extension to be 
>>>> agnostic
>>>> for extension header types. There's no such limitation in [RFC5285]
>>>> if I read it correctly. And based on [RFC5285], it says:
>>>>
>>>> " There are two variants of the extension: one-byte and two-byte
>>>> headers.  Since it is expected that (a) the number of extensions in
>>>> any given RTP session is small and (b) the extensions themselves 
>>>> are
>>>> small, the one-byte header form is preferred and MUST be supported 
>>>> by
>>>> all receivers.  A stream MUST contain only one-byte or two-byte
>>>> headers: they MUST NOT be mixed within a stream.
>>>> Transmitters SHOULD NOT use the two-byte form when all extensions 
>>>> are
>>>> small enough for the one-byte header form. "
>>>>
>>>> For splicing extension, it can be classified into a) case. That's 
>>>> why
>>>> we uses one-byte header, and it is preferred and mandated by
>>>> receivers. So I can't imagine any other cases to use 2-byte header.
>>>> And by the way, [RFC6051] also only defines one-byte header.
>>>
>>> So this debate do gets us into RFC5285 update country. However the
>>> issue is that although a particular RTP header extension has a
>>> preference or requirement for which format to use. The actual format
>>> used in an actual RTP packet will be depending on the union of all 
>>> the
>>> requirement for header extension format. For example the SDES header
>>> extension is one that can require to use the 2-byte format due to 
>>> the
>>> SDES item's length (>16 bytes). If that header extension and yours
>>> needs to be in the same packet, then also yours need to use the 
>>> 2-byte
>>> header format in this particular packet.
>>>
>>> [Rachel]: The RFC5285 update already permits to use 1-byte and 
>>> 2-byte
>>> format in the same RTP stream. You're saying the case they're used 
>>> in
>>> one RTP packet. For splicing header extension, it only appears a
>>> number of RTP packet instead of all the RTP packets. So I'm just
>>> curious the possibility of the 2 kinds of extensions must be in the
>>> same packet.
>>
>> No, that is not what I try to say. I am saying that within one RTP 
>> stream, i.e.
>> under one SSRC. Then in one packet one might need to use the 2-byte 
>> format
>> due to that an additional RTP header extension is added to the one 
>> with Splicing
>> information. In a later packet where there is need to send a splicing 
>> RTP header
>> extension, there might not be any other RTP header extension, nor one 
>> that
>> requires the two-byte header extension. Thus, this packet can be sent 
>> with the
>> 1-byte format.
>
> [Rachel]: Yes, I understand you point. And that's all what RFC5285 
> update is talking about, see below
>
> "
> A stream MUST contain only one-byte or two-byte
> headers unless it is known that all recipients support mixing, either
> by offer/answer negotiation (see section 6) or by out-of-band
> knowledge.  One-byte and two-byte headers MUST NOT be mixed in a
> single RTP packet.
> "
>
>>
>>>
>>> This is issue that needs to be clarified in RFC 5285 update, and
>>> hopefully to allow a particular SSRC to switch between formats on a
>>> packet by packet basis. If not then we end up in a situation where
>>> hard choices will have to be done when conflicting requirements 
>>> occur.
>>>
>>> The above issue is from my perspective that if one defines a header
>>> extension in such a way that it is clear on what is the data part 
>>> and
>>> don't lock down the format to use, then it will work with the 
>>> updated
>>> spec.
>>>
>>> [Rachel]: It's no harm to include a 2-byte format in this version to
>>> avoid the case that you mentioned. And I'll do it right now.
>>
>> Yes, that is one way of doing it. The point I tried to made is that 
>> one can define
>> ones RTP header extension is such a way that it becomes agnostic to 
>> if 1-byte
>> or 2-byte format is used for the header extensions. This should be 
>> the
>> recommended practice unless the format requires the 2-byte format.
>
> [Rachel]: We design the splicing header extension in a bit-saving way, 
> because the length of data field is fixed.
>
>>
>> I will have to review the RFC 5285 update to see if this has 
>> sufficient
>> clarifications in this area. I would recommend that also the rest of 
>> the WG do
>> take a look on the document update.
>>
>> cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> FĂ¤rĂ¶gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------


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Subject: Re: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
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> On 23 Mar 2016, at 08:18, Jonathan Lennox <jonathan@vidyo.com> wrote:
>=20
> This is to announce a 1.5 week Working Group Last Call for
>=20
> 	draft-ietf-avtext-rid-01
>=20
> as proposed standard.
>=20
> Please review and provide any comments you may have on the document by =
Sunday, April 3 (the beginning of the Buenos Aires IETF meeting). =
Comments should be sent to the document authors and the AVTEXT WG list.

The introduction to this draft still says:

>    To address this situation, we define a new RTCP SDES identifier =
that
>    uniquely identifies a single stream.=20

shortly followed by:

>                                                       when this new
>    identifier is in use, it appears (and contains the same value) in
>    both in the Redundancy RTP Stream as well as the stream it is
>    correcting.

That seems contradictory to me.=20

I understand that there=E2=80=99s no consensus to make the RID a unique =
per-stream identifier, and I=E2=80=99m not asking to revisit that =
discussion. However, the draft does need to be revised to not claim that =
the RID uniquely identifies a stream, since it doesn=E2=80=99t in all =
cases.


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





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On 3/29/16 11:25, Bernard Aboba wrote:
> I have reviewed the document and have the following comment:
>
> Need for RRID.  Currently, draft-ietf-mmusic-rid does not make use of 
> the RRID.  As I understand it, no WebRTC browsers are currently 
> planning to implement the RRID.  Furthermore, neither the WebRTC 1.0 
> API nor ORTC supports the RRID.
>
> Given this, do we really need to include the RRID in this document?
>

...and then, on 3/30/16 10:38, Colin Perkins wrote:
> The introduction to this draft still says:
>
>>     To address this situation, we define a new RTCP SDES identifier that
>>     uniquely identifies a single stream.
> shortly followed by:
>
>>                                                        when this new
>>     identifier is in use, it appears (and contains the same value) in
>>     both in the Redundancy RTP Stream as well as the stream it is
>>     correcting.
> That seems contradictory to me.
>
> I understand that thereâ€™s no consensus to make the RID a unique per-stream identifier, and Iâ€™m not asking to revisit that discussion. However, the draft does need to be revised to not claim that the RID uniquely identifies a stream, since it doesnâ€™t in all cases.
>

I'm finding this a bit perplexing. RRID provides the means to uniquely 
identify a stream (making the introductory text true). The introductory 
text motivates the inclusion of RRID, and the presence of RRID satisfies 
the goal stated in the introduction.

It seems to me we can proceed in one of two ways:

 1. Strip RRID from the document, thereby eliminating the ability to
    uniquely identify streams, and remove the consequently erroneous
    statement from the introduction, or

 2. Leave RRID in the document, which makes the cited statement in the
    introduction actually true -- effectively, leave the current
    document as-is.


I'd be interested in hearing Bernard and Colin's take on this, and even 
*more* interested in having additional parties weigh in.

/a


--------------000706090304030600070704
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: 8bit

<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 3/29/16 11:25, Bernard Aboba wrote:<br>
    </div>
    <blockquote
cite="mid:CAOW+2dtMwwF0WfMx6A3hSquLRsdx4BCHjsVfQRFfYm+he2VaNA@mail.gmail.com"
      type="cite">
      <div dir="ltr">I have reviewed the document and have the following
        comment:Â 
        <div><br>
        </div>
        <div>Need for RRID.Â  Currently, draft-ietf-mmusic-rid does not
          make use of the RRID.Â  As I understand it, no WebRTC browsers
          are currently planning to implement the RRID.Â  Furthermore,
          neither the WebRTC 1.0 API nor ORTC supports the RRID. Â Â </div>
        <div><br>
        </div>
        <div>Given this, do we really need to include the RRID in this
          document? Â </div>
        <br>
      </div>
    </blockquote>
    <br>
    ...and then, on 3/30/16 10:38, Colin Perkins wrote:<br>
    <blockquote
      cite="mid:E2EDC271-3C9E-41CD-8CFA-3B8FADA9E8A3@csperkins.org"
      type="cite">
      <pre wrap="">
The introduction to this draft still says:

</pre>
      <blockquote type="cite">
        <pre wrap="">   To address this situation, we define a new RTCP SDES identifier that
   uniquely identifies a single stream. 
</pre>
      </blockquote>
      <pre wrap="">shortly followed by:

</pre>
      <blockquote type="cite">
        <pre wrap="">                                                      when this new
   identifier is in use, it appears (and contains the same value) in
   both in the Redundancy RTP Stream as well as the stream it is
   correcting.
</pre>
      </blockquote>
      <pre wrap="">That seems contradictory to me. 

I understand that thereâ€™s no consensus to make the RID a unique per-stream identifier, and Iâ€™m not asking to revisit that discussion. However, the draft does need to be revised to not claim that the RID uniquely identifies a stream, since it doesnâ€™t in all cases.

</pre>
    </blockquote>
    <br>
    I'm finding this a bit perplexing. RRID provides the means to
    uniquely identify a stream (making the introductory text true). The
    introductory text motivates the inclusion of RRID, and the presence
    of RRID satisfies the goal stated in the introduction.<br>
    <br>
    It seems to me we can proceed in one of two ways:<br>
    <br>
    <ol>
      <li>Strip RRID from the document, thereby eliminating the ability
        to uniquely identify streams, and remove the consequently
        erroneous statement from the introduction, or<br>
        <br>
      </li>
      <li>Leave RRID in the document, which makes the cited statement in
        the introduction actually true -- effectively, leave the current
        document as-is.</li>
    </ol>
    <p><br>
      I'd be interested in hearing Bernard and Colin's take on this, and
      even *more* interested in having additional parties weigh in.<br>
    </p>
    <p>/a<br>
    </p>
  </body>
</html>

--------------000706090304030600070704--


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Date: Wed, 30 Mar 2016 15:12:32 -0700
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Subject: Re: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
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Adam said:

"It seems to me we can proceed in one of two ways:


   1. Strip RRID from the document, thereby eliminating the ability to
   uniquely identify streams, and remove the consequently erroneous stateme=
nt
   from the introduction, or

   2. Leave RRID in the document, which makes the cited statement in the
   introduction actually true -- effectively, leave the current document as=
-is.


I'd be interested in hearing Bernard and Colin's take on this, and even
*more* interested in having additional parties weigh in."

[BA] I agree with your assessment of the choices.  My only request is that
the document actually reflect what implementers intend to support, in the
absence of the  additional requirements language provided in RFC 6919 which
describes the apparent status of the RRID quite well (but alas is only an
April 1 RFC):

1 <https://tools.ietf.org/html/rfc6919#section-1>.  MUST (BUT WE KNOW YOU W=
ON'T)

   The phrase "MUST (BUT WE KNOW YOU WON'T)" is used to indicate
   requirements that are needed to meet formal review criteria (e.g.,
   mandatory-to-implement security mechanisms), when these mechanisms
   are too inconvenient for implementers to actually implement.


On Wed, Mar 30, 2016 at 12:07 PM, Adam Roach <adam@nostrum.com> wrote:

> On 3/29/16 11:25, Bernard Aboba wrote:
>
> I have reviewed the document and have the following comment:
>
> Need for RRID.  Currently, draft-ietf-mmusic-rid does not make use of the
> RRID.  As I understand it, no WebRTC browsers are currently planning to
> implement the RRID.  Furthermore, neither the WebRTC 1.0 API nor ORTC
> supports the RRID.
>
> Given this, do we really need to include the RRID in this document?
>
>
> ...and then, on 3/30/16 10:38, Colin Perkins wrote:
>
> The introduction to this draft still says:
>
>
>    To address this situation, we define a new RTCP SDES identifier that
>    uniquely identifies a single stream.
>
> shortly followed by:
>
>
>                                                       when this new
>    identifier is in use, it appears (and contains the same value) in
>    both in the Redundancy RTP Stream as well as the stream it is
>    correcting.
>
> That seems contradictory to me.
>
> I understand that there=E2=80=99s no consensus to make the RID a unique p=
er-stream identifier, and I=E2=80=99m not asking to revisit that discussion=
. However, the draft does need to be revised to not claim that the RID uniq=
uely identifies a stream, since it doesn=E2=80=99t in all cases.
>
>
>
> I'm finding this a bit perplexing. RRID provides the means to uniquely
> identify a stream (making the introductory text true). The introductory
> text motivates the inclusion of RRID, and the presence of RRID satisfies
> the goal stated in the introduction.
>
> It seems to me we can proceed in one of two ways:
>
>
>    1. Strip RRID from the document, thereby eliminating the ability to
>    uniquely identify streams, and remove the consequently erroneous state=
ment
>    from the introduction, or
>
>    2. Leave RRID in the document, which makes the cited statement in the
>    introduction actually true -- effectively, leave the current document =
as-is.
>
>
> I'd be interested in hearing Bernard and Colin's take on this, and even
> *more* interested in having additional parties weigh in.
>
> /a
>

--001a1142702c5f2f60052f4b72fc
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<div dir=3D"ltr">Adam said:=C2=A0<div><br></div><div>&quot;<span style=3D"f=
ont-size:12.8px">It seems to me we can proceed in one of two ways:</span></=
div><br style=3D"font-size:12.8px"><ol style=3D"font-size:12.8px"><li style=
=3D"margin-left:15px">Strip RRID from the document, thereby eliminating the=
 ability to uniquely identify streams, and remove the consequently erroneou=
s statement from the introduction, or<br><br></li><li style=3D"margin-left:=
15px">Leave RRID in the document, which makes the cited statement in the in=
troduction actually true -- effectively, leave the current document as-is.<=
/li></ol><p style=3D"font-size:12.8px"><br></p><div><span style=3D"font-siz=
e:12.8px">I&#39;d be interested in hearing Bernard and Colin&#39;s take on =
this, and even *more* interested in having additional parties weigh in.</sp=
an>&quot;</div><div><br></div><div>[BA] I agree with your assessment of the=
 choices.=C2=A0 My only request is that the document actually reflect what =
implementers intend to support, in the absence of the =C2=A0additional requ=
irements language provided in RFC 6919 which describes the apparent status =
of the RRID quite well (but alas is only an April 1 RFC):=C2=A0</div><div><=
br></div><div><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;m=
argin-bottom:0px;color:rgb(0,0,0)"><span class=3D"" style=3D"line-height:0p=
t;display:inline;font-size:1em;font-weight:bold"><h2 style=3D"line-height:0=
pt;display:inline;font-size:1em"><a class=3D"" name=3D"section-1" href=3D"h=
ttps://tools.ietf.org/html/rfc6919#section-1" style=3D"color:black;text-dec=
oration:none">1</a>.  MUST (BUT WE KNOW YOU WON&#39;T)</h2></span>

   The phrase &quot;MUST (BUT WE KNOW YOU WON&#39;T)&quot; is used to indic=
ate
   requirements that are needed to meet formal review criteria (e.g.,
   mandatory-to-implement security mechanisms), when these mechanisms
   are too inconvenient for implementers to actually implement.</pre></div>=
</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Mar=
 30, 2016 at 12:07 PM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:a=
dam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><span class=3D"">
    <div>On 3/29/16 11:25, Bernard Aboba wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">I have reviewed the document and have the following
        comment:=C2=A0
        <div><br>
        </div>
        <div>Need for RRID.=C2=A0 Currently, draft-ietf-mmusic-rid does not
          make use of the RRID.=C2=A0 As I understand it, no WebRTC browser=
s
          are currently planning to implement the RRID.=C2=A0 Furthermore,
          neither the WebRTC 1.0 API nor ORTC supports the RRID. =C2=A0=C2=
=A0</div>
        <div><br>
        </div>
        <div>Given this, do we really need to include the RRID in this
          document? =C2=A0</div>
        <br>
      </div>
    </blockquote>
    <br></span><span class=3D"">
    ...and then, on 3/30/16 10:38, Colin Perkins wrote:<br>
    <blockquote type=3D"cite">
      <pre>The introduction to this draft still says:

</pre>
      <blockquote type=3D"cite">
        <pre>   To address this situation, we define a new RTCP SDES identi=
fier that
   uniquely identifies a single stream.=20
</pre>
      </blockquote>
      <pre>shortly followed by:

</pre>
      <blockquote type=3D"cite">
        <pre>                                                      when thi=
s new
   identifier is in use, it appears (and contains the same value) in
   both in the Redundancy RTP Stream as well as the stream it is
   correcting.
</pre>
      </blockquote>
      <pre>That seems contradictory to me.=20

I understand that there=E2=80=99s no consensus to make the RID a unique per=
-stream identifier, and I=E2=80=99m not asking to revisit that discussion. =
However, the draft does need to be revised to not claim that the RID unique=
ly identifies a stream, since it doesn=E2=80=99t in all cases.

</pre>
    </blockquote>
    <br></span>
    I&#39;m finding this a bit perplexing. RRID provides the means to
    uniquely identify a stream (making the introductory text true). The
    introductory text motivates the inclusion of RRID, and the presence
    of RRID satisfies the goal stated in the introduction.<br>
    <br>
    It seems to me we can proceed in one of two ways:<br>
    <br>
    <ol>
      <li>Strip RRID from the document, thereby eliminating the ability
        to uniquely identify streams, and remove the consequently
        erroneous statement from the introduction, or<br>
        <br>
      </li>
      <li>Leave RRID in the document, which makes the cited statement in
        the introduction actually true -- effectively, leave the current
        document as-is.</li>
    </ol>
    <p><br>
      I&#39;d be interested in hearing Bernard and Colin&#39;s take on this=
, and
      even *more* interested in having additional parties weigh in.<span cl=
ass=3D"HOEnZb"><font color=3D"#888888"><br>
    </font></span></p><span class=3D"HOEnZb"><font color=3D"#888888">
    <p>/a<br>
    </p>
  </font></span></div>

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

--001a1142702c5f2f60052f4b72fc--


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Subject: Re: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
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On 3/30/16 17:12, Bernard Aboba wrote:
> [BA] I agree with your assessment of the choices.  My only request is 
> that the document actually reflect what implementers intend to 
> support, in the absence of the  additional requirements language 
> provided in RFC 6919 which describes the apparent status of the RRID 
> quite well (but alas is only an April 1 RFC):

Ah, I thought it was clear that support for either or both mechanisms is 
completely optional and independent of each other. Would it help to add 
text to that effect?

/a


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Subject: Re: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
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> On 30 Mar 2016, at 20:07, Adam Roach <adam@nostrum.com> wrote:
> On 3/29/16 11:25, Bernard Aboba wrote:
>> I have reviewed the document and have the following comment:=20
>>=20
>> Need for RRID.  Currently, draft-ietf-mmusic-rid does not make use of =
the RRID.  As I understand it, no WebRTC browsers are currently planning =
to implement the RRID.  Furthermore, neither the WebRTC 1.0 API nor ORTC =
supports the RRID.  =20
>>=20
>> Given this, do we really need to include the RRID in this document? =20=

>=20
> ...and then, on 3/30/16 10:38, Colin Perkins wrote:
>> The introduction to this draft still says:
>>=20
>>=20
>>>    To address this situation, we define a new RTCP SDES identifier =
that
>>>    uniquely identifies a single stream.=20
>>>=20
>> shortly followed by:
>>=20
>>=20
>>>                                                       when this new
>>>    identifier is in use, it appears (and contains the same value) in
>>>    both in the Redundancy RTP Stream as well as the stream it is
>>>    correcting.
>>>=20
>> That seems contradictory to me.=20
>>=20
>> I understand that there=E2=80=99s no consensus to make the RID a =
unique per-stream identifier, and I=E2=80=99m not asking to revisit that =
discussion. However, the draft does need to be revised to not claim that =
the RID uniquely identifies a stream, since it doesn=E2=80=99t in all =
cases.
>>=20
>>=20
>=20
> I'm finding this a bit perplexing. RRID provides the means to uniquely =
identify a stream (making the introductory text true). The introductory =
text motivates the inclusion of RRID, and the presence of RRID satisfies =
the goal stated in the introduction.
>=20
> It seems to me we can proceed in one of two ways:
>=20
> 	=E2=80=A2 Strip RRID from the document, thereby eliminating the =
ability to uniquely identify streams, and remove the consequently =
erroneous statement from the introduction, or
>=20
> 	=E2=80=A2 Leave RRID in the document, which makes the cited =
statement in the introduction actually true =E2=80=94 effectively, leave =
the current document as-is.

I read the Introduction as saying *both* that the RID has to be unique =
for each stream, *and* that the RID has to have the same value in the =
main stream and in the redundant stream. This seems like a =
contradiction, irrespective of what you do with the RRID. The fix might =
be as simple as changing =E2=80=9CTo address this situation, we define a =
new RTCP SDES identifier that uniquely identifies a single stream=E2=80=9D=
 to =E2=80=9C=E2=80=A6that identifies a stream and any redundant =
encoding of that stream=E2=80=9D.

> I'd be interested in hearing Bernard and Colin's take on this, and =
even *more* interested in having additional parties weigh in.



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





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Subject: Re: [avtext] WGLC: draft-ietf-avtext-rid-01.txt
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Hi,

Here are my comments on the draft-ietf-avtext-rid-01. I think it is very 
good that we have an upcoming meeting where we can discuss the below issues.

1. Abstract:

    This document defines and registers an RTCP SDES item, RID, for
    identification of RTP streams associated with Encoded Streams and
    Dependent Streams.

Should this really use plural of RTP streams as well as Encoded Streams. 
A particular RID value is the identifier for one Encoded or Dependent 
Stream in its RTP packetized form plus. Thus I think this abstract could 
be clearer.  However, I think it may need to addressed anyway based on 
comment 3 and 4.



2. Section 1:

RTP sessions frequently consist of multiple streams, each of which is
    identified at any given time by its SSRC; however, the SSRC
    associated with a stream is not guaranteed to be stable over its
    lifetime.  Within a session, these streams can be tagged with a
    number of identifiers, including CNAMEs and MSIDs
    [I-D.ietf-mmusic-msid].  Unfortunately, none of these have the proper
    ordinality to refer to an individual stream; all such identifiers can
    appear in more than one stream at a time.  While approaches that use
    unique Payload Types (PTs) per stream have been used in some
    applications,

I think the usage of Stream should at least on the first usage say "RTP 
Stream" and with a reference to RFC7656 to make it clear what entity you 
are discussing in this paragraph.

3. Section 3:

    For Encoded Streams, the RID refers to the "Source RTP Stream" as
    defined by [RFC7656] Section 2.1.10.  For Dependent Streams, it
    refers to the RTP Stream that, like the Source RTP Stream of an
    Encoded Stream, is the RTP Stream that is not a Redundancy RTP
    Stream.  For conciseness, we define the term "RID RTP Stream" to
    refer to this construct.

I think this definition is not correct. From my perspective it might be 
better to talk about the RTP stream produced by the media packetizer 
from a encoded or set of dependent streams. I write a set of dependent 
stream as on RTP stream can be produced by packetizing multiple 
dependent streams in the same. This would occur if one is not producing 
different RTP streams for all produced scalability layers.

However, the main issue is that the definition does not correctly 
identifies what the RID refers to. To my understanding the RID refers to 
one RTP stream [A] packetized from either one encoded stream and zero or 
more dependent streams, or one or more dependent streams and the one or 
more redundancy streams protecting the RTP stream [A].

This also has the consequence that RID RTP Stream is not a suitable term 
as the RID identifies an possible aggregate of RTP streams, one source 
and zero or more redundancy RTP streams.

The above changed definition could be used, however I am not certain 
that it is what we really like to use here. For that purpose see next 
comment.

4. Section 3:

    For clarity, when RID is used, Redundancy RTP Streams that can be
    used to repair Received RTP Streams will use the same RID value as
    the Received RTP Stream they are intended to be combined with.  If
    applications want to identify individual redundancy streams, they can
    add an RRID to them instead of or in addition to the RID.

So the intention with RRID, is to identify Redundancy RTP streams. 
Considering that RID doesn't actually only represent the source RTP 
stream for the media, there is a mismatch. Also, the needs for RRID is 
also not clear, however, the basics of identifying the RTP stream only 
has the advantage of high likelihood of re-use in other mechanisms.

However, I would like to take a collected consideration of the RID and 
RRID definitions and actually propose that we change the RID definition. 
To be clear on that this is another definition this new identifier I 
will refer to as RTP Stream IDentifier (SID).

The definition of SID would be: "An identifier for a single RTP stream."

This of course have implication on the RID mechanism expressing the 
payload format constraints. However, I think it fulfil the basic needs 
very nicely. The a=rid lines points at a SID for which is the RTP stream 
for which the set of constraints apply.

This is the basic part of the RID proposal. And this removes the need 
for RRID, as also Redundancy RTP streams can have a SID, if needed by 
the application. We get a generic basic identification that is likely to 
be re-usable while solving the current needs.

Some may protest, hey but we are loosing the coupling method between the 
source RTP stream and the Redundancy stream. Yes, I have a proposal 
below for how to resolve that, but at the same time, it is this coupling 
that creates the complicated definition of the RID SDES item. And this 
is actually quite a separate problem. One which has existed for some 
time, and may actually benefit from being taken on a separate work item. 
Especially as we may want to retrofit the outcome onto some existing 
mechanisms such as RTP Retransmission and Generic XOR FEC to enable 
single RTP session usage efficiently for both.

So below is a possible solution if one desire to have a way of coupling 
the redundancy stream to the source RTP stream already in signalling and 
identifying it with the same properties as we get with RID for the 
Source RTP streams.

The solution is simple, we simply have a=RID lines also for the 
Redundancy RTP streams. Then we define a new constraint which I will 
call "protect" which is a list of SIDs that points at one or more RTP 
streams for which this Redundancy RTP Stream is constrained to protect 
and provide redundancy for.

An basic SDP example for this would be the following:

    m=video 10000 RTP/SAVPF 100 101 108 109
    a=rtpmap:100 H264/90000
    a=fmtp:100 profile-level-id=42401f; packetization-mode=0
    a=rtpmap:101 H264/90000
    a=fmtp:101 profile-level-id=42401f; packetization-mode=1
    a=rtpmap:108 rtx/90000
    a=fmtp:108 apt=100
    a=rtpmap:109 rtx/90000
    a=fmtp:109 apt=101
    a=sendrecv
    a=mid:v1 (max resolution)
    a=rid:1 send max-width=1280;max-height=720;max-fps=30
    a=rid:2 recv max-width=1280;max-height=720;max-fps=30
    a=rid:A send pt=108,109;protect=1
    a=rid:B recv pt=108,109;protect=2

So at RTP session startup when the offerer of the above starts sending 
it will send two RTP streams from the above block. One for the video 
using SID=1 and another for the RTX packets using SID=A.


5. Section 4 and 6.

What I am missing in this is a discussion of privacy and the potential 
for tracking the streams across an RTP system. This is usually fine, but 
may reveal things to a third party, thus their value should be protected 
towards third parties.

The next part of this issue is the generation of these values should 
avoid leaking any information about the user and implementation. This 
leads to the question if the values should be using random values, with 
tag lengths as needed by the application.


Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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