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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: [avtext] draft-petithuguenin-avtext-multiple-clock-rates-01
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I just published a new version of the draft, which now use the formulas provided
by Magnus:

http://tools.ietf.org/html/draft-petithuguenin-avtext-multiple-clock-rates-01

As discussed in the meeting in Prague, I am now proposing this draft to be
adopted as the WG document for the "Submit Support for multiple clock rates in
an RTP session for Proposed Standard" milestone.

Changelog:

   o  Clarified the goals for this documents
   o  Removed the non-monotonic method (replaced by Magnus formula).
   o  Moved the "RTP Sender and RTP Receiver section inside a new
      "Recommendations" section.
   o  Inserted the new Sender formula inside the Recommendation section.
   o  Inserted the new jitter formula in the RTP Receiver section.
   o  Emptied the Analysis sections.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>, <avtext@ietf.org>
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Subject: Re: [avtext] draft-begen-avtext-rams-scenarios-00 : comments
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Hi Tom,

Thanks for the review.=20

> -----Original Message-----
> From: Van Caenegem, Tom (Tom) =
[mailto:tom.van_caenegem@alcatel-lucent.com]
> Sent: Tuesday, April 19, 2011 12:11 PM
> To: avtext@ietf.org
> Cc: Ali C. Begen (abegen)
> Subject: draft-begen-avtext-rams-scenarios-00 : comments
>=20
> Hi Ali,
>=20
> I read through the draft draft-begen-avtext-rams-scenarios-00.
>=20
> -One general comment I have is that you mention multiple times "RAMS =
session", while "RAMS session" is not defined in the
> RAMS draft. Maybe it would be good to clarify this.

I am actually looking for a better term here. Any suggestions? Or yes, =
we will need to define it.
=20
> E.g. in the Intro section, you say:
> "In
>    scenarios where multiple RAMS sessions will be simultaneously run =
by
>    an RTP receiver, they need to be coordinated."
> I suggest to say:
>=20
> "In
>    scenarios where multiple RAMS sessions, each initiated with an =
individual RAMS-R message to a different feedback target
> FT-ap,
> will be simultaneously run by an RTP receiver, they need to be =
coordinated by the RTP receiver."
>=20
>=20
> -because the draft focuses on co-ordinated RAMS sessions/scenarios, =
maybe add this in the title of the draft?

Good idea.
=20
> -I do not identify easily what are the guidelines this draft proposes =
(as mentioned in the intro/abstract). Is this about which
> multiplexing scenario is best in terms of combination with RAMS, in =
terms of flexibility ?.. Perhaps just stick to describing
> different scenarios, without mentioning guidelines?

Maybe, it was supposed to be more on the guidelines side. Any =
suggestions to improve the doc in that direction?
=20
> These are more detailed comments:
>=20
> Section introduction
> "However, the specification has mostly focused
>    on the simplest case, which is when the RTP receiver acquires only
>    one multicast stream at a time, to explain the protocol details."
>=20
> I would rephrase this as follows:
>=20
> The specification does not discuss scenarios where an RTP receiver =
makes use of RAMS to rapidly acquire multiple and
> associated multicast streams in parallel, or where different RTP =
sessions are part of the same SSM. E.g. the example section
> 8.3 in the specification addresses only the simple case of an RTP =
receiver rapidly acquiring only one multicast stream based
> on an SDP which describes a single SSM composed of a single RTP =
stream."

OK, some rewording is in order here.
=20
> section 4.3 : see my comments for section 6.3.. I do not understand =
scenario 3
>=20
> section 4.4:
>=20
> "and the
>    retransmission and RAMS operations will not be able to be applied
>    based on the payload type"
> DELETE: "retransmission"  (..because retransmission is based on RTP =
SN.. no need to distinguish among PT?)

OK.
=20
> section 5:
> "However, since this is not an easy requirement to satisfy,
>    RAMS specification forbids to have more than one RTP session to be
>    associated with a specific feedback target"
>=20
> ADD: ..with a specific FB target on a specific port.

Better said "with a specific FTAp" =20
=20
>=20
> section 6.1
>=20
> "This is the preferred deployment model for FEC.  Having FEC in a
>    different multicast group provides flexibility for not only the RTP
>    receivers that are not FEC capable but also the ones that are FEC
>    capable but are not willing to receive FEC during the rapid
>    acquisition."
>=20
> SUGGEST to DELETE:
> This is the preferred deployment model for FEC.

This is not much subjective AFAICT. Any strong reason why we should =
delete it? This is also what we "recommend" in FEC Framework.
=20
> REPHRASE : Having FEC in a
>    different multicast group provides two flexibility points:  RTP
>    receivers that are not FEC capable can receive the primary video =
stream without FEC, and RTP receivers that are FEC
>    capable can decide to not receive FEC during the rapid
>    acquisition (but still join the FEC stream after RAMS for the =
primary video stream has finished).

OK.=20
=20
> "In this case, the RTP receiver can request a Max
>    Receive Bitrate of 22 Mbps in the RAMS Request message"
>=20
> ADD: ..in the RAMS request message for the prmary video stream.

Good idea.=20
=20
> section 6.2:
>=20
> "The
>    only difference from scenario #1 is that here the join time is the
>    same for both the primary video and FEC streams".
>=20
> REPHRASE: .. is that here the join time is default the same for the =
primary video and FEC stream.
>=20
> (editor note: also in  scenario 1 the join time can be the same)

I see you point. I will make this clearer.=20
=20
> section 6.3
>=20
> "This is similar to scenario #2.  However, since we cannot explicitly
>    specify the bitrates for the primary and FEC streams, the RTP
>    receiver can request to rapidly acquire both streams in parallel.  =
In
>    this case, two separate RAMS Request messages have to be sent by =
the
>    RTP receiver to the feedback target."
>=20
>=20
> ..I am a bit confused here. Fact is that, following the RAMS =
specification, normally a single RAMS-R message is sent. If we

Right, a single RAMS-R asking for all SSRCs would work. When you =
indicate the max receive bitrate, individual streams will be accelerated =
relatively. If that is what one wants, it is fine.

> know the SSRCs of each stream, an RTP receiver can just request the =
primary video stream. However, a single RAMS-R does
> not allow to specify e.g. "max receive bit rate" per stream.. but I do =
not think this is desired anyhow, as it will be the source
> of the two sub streams that determine the relative bursting of the two =
streams.

So, looks like some revision is needed here.=20

> If we do not know the SSRC values, we cannot request separately the =
SSRC streams using a single RAMS-R.
> ..but why do we need to send two separate RAMS-R messages... and how =
will you do that when there is only a single FT-ap?

Yeah, something is not worded accurately here.  I will try to make this =
clearer.

Thanks.
-acbegen=20

From emil@sip-communicator.org  Wed May  4 05:37:44 2011
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Subject: Re: [avtext] Comments on draft-ietf-avtext-mixer-to-client-audio-level
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Hey Keith, all,

We are about to submit a new version and have addressed most of the
comments raised by you and others on this list. We just wanted to
clarify a couple of things before submitting:

=D0=9D=D0=B0 28.03.11 13:29, DRAGE, Keith (Keith) =D0=BD=D0=B0=D0=BF=D0=B8=
=D1=81=D0=B0:
> Again a brief scan and identified some comments:
>=20
> 1)	Section 3:
>=20
> This document describes an RTP header extension that allows mixers
> to indicate the audio-level of every conference participant (CSRC)
> in addition to simply indicating their on/off status.  This new
> header extension is based on the "General Mechanism for RTP Header=20
> Extensions" [RFC5285].
>=20
> RFC 5285 is a normative reference, and therefore the use of the term
> "is based" is inappropriate. Need to be much more precise about how
> RFC 5285 is followed (although some of that is covered by later
> references.

Does the following work?

   This document describes an RTP header extension that allows mixers to
   indicate the audio-level of every conference participant (CSRC) in
   addition to simply indicating their on/off status.  This new header
   extension uses "General Mechanism for RTP Header Extensions"
   described in [RFC5285].


> 2)	Section 4:
>=20
> The 4-bit len field is the number minus one of data bytes (i.e.
> audio level values) transported in this header extension element
> following the one-byte header.  Therefore, the value zero in this
> field indicates that one byte of data follows.  A value of 15 is not=20
> allowed by this specification and it MUST NOT be used as the RTP=20
> header can carry a maximum of 15 CSRC IDs.  The maximum value
> allowed is therefore 14 indicating a following sequence of 15 audio
> level values.
>=20
> I suspect that any RFC 2119 language here actually belongs in a
> procedural section, but I was also unclear what the real conformance
> requirement was here.=20

I am not sure I understand this comment.

> Surely the main conformance requirement is to
> carry 15 or less audio levels. This results in a maximum encoding of
> 14.

Yes this is what we had in mind and what the text currently says. Is
there a problem with it? Could you maybe suggest a better wording?


> 3)	Section 6:
>=20
> Conferencing clients that support audio level indicators and have no=20
> mixing capabilities SHOULD always include the direction parameter in=20
> the "extmap" attribute setting it to "recvonly".  Conference focus=20
> entities with mixing capabilities MAY omit the direction or set it
> to "sendrecv" in SDP offers.  Such entities SHOULD set it to
> "sendonly" in SDP answers to offers with a "recvonly" parameter and
> to "sendrecv" when answering other "sendrecv" offers.
>=20
> Is such a requirement in scope of this document? I suspect it ends up
> updating an existing MMUSIC RFC.

We don't really change or override anything in here and it's mostly
a clarification. The thinking is that if a conferencing client does not
have conferencing capabilities, then they won't be sending the header
extension and would only be able to receive and process those coming
from the mixer. This translates to

    SHOULD always include the direction parameter in the "extmap"
    attribute setting it to "recvonly"

would it help if we simply removed the 2119 language and otherwise kept
the text as it is?

> 4)	Section 6:
>=20
> An example SDP offer/answer exchange between two conference focus=20
> entities with mixing capabilities negotiating an audio stream with=20
> bidirectional flwo of audio level information.
>=20
> Typo "flow"

Fixed.

> 5)	Section 7:
>=20
> 2.  Furthermore, the fact that audio level values would not be=20
> protected even in an SRTP session may be of concern in some cases=20
> where the activity of a particular participant in a conference is=20
> confidential.
>=20
> Change "may" to "can" or "might" to avoid RFC 2119 style sentences.

Changed.

> 6)	Section 8:
>=20
> Add a note to RFC editor saying replace RFC XXXX by the number of
> this RFC.

Added.

> 7)	Section 12.2
>=20
> RFC 3920, RFC 3551 appear to not be used as references.

Removed.

> 8)	General:
>=20
> I suspect the editors could do with putting the two drafts side by
> side and aligning the text from one to the other in the areas where
> they are both trying to say the same thing. This is not a call for a
> single document - I think they are distinct in that respect, but
> there are similarities in the requirements and we should only say
> similar things in the same way.

OK.

> Finally I notice that the summary of changes since the previous
> version has been restricted to the last version. Could I ask that this
> material is reinstated, and we will delete it at the publication
> stage. In pursuance of this, please add a note to the RFC editor to
> delete this material during final editing.

Done. Whatever changelogs existed are back into the document.

We also removed the level renderer and added an overload parameter to
the AudioLevelCalculator as suggested by Peter Musgrave.



Cheers,
Emil

>=20
>=20
> Regards
>=20
> Keith _______________________________________________ avtext mailing
> list avtext@ietf.org https://www.ietf.org/mailman/listinfo/avtext
>=20

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31



From emil@sip-communicator.org  Wed May  4 06:20:59 2011
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Adding to my previous post:

=D0=9D=D0=B0 04.05.11 15:37, Emil Ivov =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0=
:
>> Conferencing clients that support audio level indicators and have no=20
>> mixing capabilities SHOULD always include the direction parameter in=20
>> the "extmap" attribute setting it to "recvonly".  Conference focus=20
>> entities with mixing capabilities MAY omit the direction or set it
>> to "sendrecv" in SDP offers.  Such entities SHOULD set it to
>> "sendonly" in SDP answers to offers with a "recvonly" parameter and
>> to "sendrecv" when answering other "sendrecv" offers.
>>
>> Is such a requirement in scope of this document? I suspect it ends up
>> updating an existing MMUSIC RFC.
>=20
> We don't really change or override anything in here and it's mostly
> a clarification. The thinking is that if a conferencing client does not=

> have conferencing capabilities, then they won't be sending the header
> extension and would only be able to receive and process those coming
> from the mixer. This translates to
>=20
>     SHOULD always include the direction parameter in the "extmap"
>     attribute setting it to "recvonly"
>=20
> would it help if we simply removed the 2119 language and otherwise kept=

> the text as it is?

Like this for example:

  Conferencing clients that support audio level indicators and
  have no mixing capabilities would not be able to content for
  this audio level extension and would hence have to always
  include the direction parameter in the "extmap" attribute
  with a value of "recvonly".  Conference focus entities with
  mixing capabilities can omit the direction or set it to
  "sendrecv" in SDP offers. Such entities would need to
  set it to "sendonly" in SDP answers to offers with a
  "recvonly" parameter and to "sendrecv" when answering other
  "sendrecv" offers.

Emil


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To: Emil Ivov <emcho@jitsi.org>
Date: Wed, 4 May 2011 19:11:03 +0200
Thread-Topic: [avtext] Comments on draft-ietf-avtext-mixer-to-client-audio-level
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See below.

I've removed the fixed comments.

Keith

> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: 04 May 2011 13:38
> To: DRAGE, Keith (Keith)
> Cc: avtext@ietf.org
> Subject: Re: [avtext] Comments on draft-ietf-avtext-mixer-to-client-audio=
-
> level
>=20
> Hey Keith, all,
>=20
> We are about to submit a new version and have addressed most of the
> comments raised by you and others on this list. We just wanted to
> clarify a couple of things before submitting:
>=20
> =EE=C1 28.03.11 13:29, DRAGE, Keith (Keith) =CE=C1=D0=C9=D3=C1:
> > Again a brief scan and identified some comments:
> >
> > 1)	Section 3:
> >
> > This document describes an RTP header extension that allows mixers
> > to indicate the audio-level of every conference participant (CSRC)
> > in addition to simply indicating their on/off status.  This new
> > header extension is based on the "General Mechanism for RTP Header
> > Extensions" [RFC5285].
> >
> > RFC 5285 is a normative reference, and therefore the use of the term
> > "is based" is inappropriate. Need to be much more precise about how
> > RFC 5285 is followed (although some of that is covered by later
> > references.
>=20
> Does the following work?
>=20
>    This document describes an RTP header extension that allows mixers to
>    indicate the audio-level of every conference participant (CSRC) in
>    addition to simply indicating their on/off status.  This new header
>    extension uses "General Mechanism for RTP Header Extensions"
>    described in [RFC5285].
>=20
That works.

>=20
> > 2)	Section 4:
> >
> > The 4-bit len field is the number minus one of data bytes (i.e.
> > audio level values) transported in this header extension element
> > following the one-byte header.  Therefore, the value zero in this
> > field indicates that one byte of data follows.  A value of 15 is not
> > allowed by this specification and it MUST NOT be used as the RTP
> > header can carry a maximum of 15 CSRC IDs.  The maximum value
> > allowed is therefore 14 indicating a following sequence of 15 audio
> > level values.
> >
> > I suspect that any RFC 2119 language here actually belongs in a
> > procedural section, but I was also unclear what the real conformance
> > requirement was here.
>=20
> I am not sure I understand this comment.
>=20
As written, I found it very different to think of a test for the requiremen=
t based only on the MUST.


> > Surely the main conformance requirement is to
> > carry 15 or less audio levels. This results in a maximum encoding of
> > 14.
>=20
> Yes this is what we had in mind and what the text currently says. Is
> there a problem with it? Could you maybe suggest a better wording?
>=20
Without investigating deeper I was unclear what the limitation is here. Is =
the value 15 possible but reserved elsewhere for extension or what.

If the coding already doesn't allow the value 15, but only the values 0 thr=
ough 14, then I would assume we just need an informative statement pointing=
 out the consequences or the normative requirement.

In the value 15 or more is allowed, then I would suggest just writing a sta=
tement that says "Implementations MUST NOT include more than 15 audio level=
 values."

>=20
> > 3)	Section 6:
> >
> > Conferencing clients that support audio level indicators and have no
> > mixing capabilities SHOULD always include the direction parameter in
> > the "extmap" attribute setting it to "recvonly".  Conference focus
> > entities with mixing capabilities MAY omit the direction or set it
> > to "sendrecv" in SDP offers.  Such entities SHOULD set it to
> > "sendonly" in SDP answers to offers with a "recvonly" parameter and
> > to "sendrecv" when answering other "sendrecv" offers.
> >
> > Is such a requirement in scope of this document? I suspect it ends up
> > updating an existing MMUSIC RFC.
>=20
> We don't really change or override anything in here and it's mostly
> a clarification. The thinking is that if a conferencing client does not
> have conferencing capabilities, then they won't be sending the header
> extension and would only be able to receive and process those coming
> from the mixer. This translates to
>=20
>     SHOULD always include the direction parameter in the "extmap"
>     attribute setting it to "recvonly"
>=20
> would it help if we simply removed the 2119 language and otherwise kept
> the text as it is?
>=20
Yes.



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Fine by me

Keith

> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: 04 May 2011 14:21
> To: DRAGE, Keith (Keith)
> Cc: avtext@ietf.org
> Subject: Re: [avtext] Comments on draft-ietf-avtext-mixer-to-client-audio=
-
> level
>=20
> Adding to my previous post:
>=20
> =EE=C1 04.05.11 15:37, Emil Ivov =CE=C1=D0=C9=D3=C1:
> >> Conferencing clients that support audio level indicators and have no
> >> mixing capabilities SHOULD always include the direction parameter in
> >> the "extmap" attribute setting it to "recvonly".  Conference focus
> >> entities with mixing capabilities MAY omit the direction or set it
> >> to "sendrecv" in SDP offers.  Such entities SHOULD set it to
> >> "sendonly" in SDP answers to offers with a "recvonly" parameter and
> >> to "sendrecv" when answering other "sendrecv" offers.
> >>
> >> Is such a requirement in scope of this document? I suspect it ends up
> >> updating an existing MMUSIC RFC.
> >
> > We don't really change or override anything in here and it's mostly
> > a clarification. The thinking is that if a conferencing client does not
> > have conferencing capabilities, then they won't be sending the header
> > extension and would only be able to receive and process those coming
> > from the mixer. This translates to
> >
> >     SHOULD always include the direction parameter in the "extmap"
> >     attribute setting it to "recvonly"
> >
> > would it help if we simply removed the 2119 language and otherwise kept
> > the text as it is?
>=20
> Like this for example:
>=20
>   Conferencing clients that support audio level indicators and
>   have no mixing capabilities would not be able to content for
>   this audio level extension and would hence have to always
>   include the direction parameter in the "extmap" attribute
>   with a value of "recvonly".  Conference focus entities with
>   mixing capabilities can omit the direction or set it to
>   "sendrecv" in SDP offers. Such entities would need to
>   set it to "sendonly" in SDP answers to offers with a
>   "recvonly" parameter and to "sendrecv" when answering other
>   "sendrecv" offers.
>=20
> Emil


From emil@sip-communicator.org  Thu May  5 02:52:44 2011
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Hey Keith,

(inline)

=EE=C1 04.05.11 20:11, DRAGE, Keith (Keith) =CE=C1=D0=C9=D3=C1:
>>> 2)	Section 4:
>>>=20
>>> The 4-bit len field is the number minus one of data bytes (i.e.=20
>>> audio level values) transported in this header extension element=20
>>> following the one-byte header.  Therefore, the value zero in
>>> this field indicates that one byte of data follows.  A value of
>>> 15 is not allowed by this specification and it MUST NOT be used
>>> as the RTP header can carry a maximum of 15 CSRC IDs.  The
>>> maximum value allowed is therefore 14 indicating a following
>>> sequence of 15 audio level values.
>>>=20
>>> I suspect that any RFC 2119 language here actually belongs in a=20
>>> procedural section, but I was also unclear what the real
>>> conformance requirement was here.
>>=20
>> I am not sure I understand this comment.
>>=20
> As written, I found it very different to think of a test for the
> requirement based only on the MUST.
>=20
>=20
>>> Surely the main conformance requirement is to carry 15 or less
>>> audio levels. This results in a maximum encoding of 14.
>>=20
>> Yes this is what we had in mind and what the text currently says.
>> Is there a problem with it? Could you maybe suggest a better
>> wording?
>>=20
> Without investigating deeper I was unclear what the limitation is
> here. Is the value 15 possible but reserved elsewhere for extension
> or what.
>=20
> If the coding already doesn't allow the value 15, but only the values
> 0 through 14, then I would assume we just need an informative
> statement pointing out the consequences or the normative
> requirement.
>=20
> In the value 15 or more is allowed, then I would suggest just writing
> a statement that says "Implementations MUST NOT include more than 15
> audio level values."

OK I see. So how about this:

   The 4-bit len field is the number minus one of data bytes (i.e. audio
   level values) transported in this header extension element following
   the one-byte header.  Therefore, the value zero in this field
   indicates that one byte of data follows.  RFC 3550 [RFC3550] only
   allows RTP packets to carry a maximum of 15 CSRC IDs.  Given that
   audio levels directly refer to CSRC IDs, implementations MUST NOT
   include more than 15 audio level values.  The maximum value allowed
   in the len field is therefore 14.

Emil


From keith.drage@alcatel-lucent.com  Thu May  5 02:54:10 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Emil Ivov <emcho@jitsi.org>
Date: Thu, 5 May 2011 11:53:42 +0200
Thread-Topic: [avtext] Comments on draft-ietf-avtext-mixer-to-client-audio-level
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Fine by me.

Keith

> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: 05 May 2011 10:53
> To: DRAGE, Keith (Keith)
> Cc: avtext@ietf.org
> Subject: Re: [avtext] Comments on draft-ietf-avtext-mixer-to-client-audio=
-
> level
>=20
> Hey Keith,
>=20
> (inline)
>=20
> =EE=C1 04.05.11 20:11, DRAGE, Keith (Keith) =CE=C1=D0=C9=D3=C1:
> >>> 2)	Section 4:
> >>>
> >>> The 4-bit len field is the number minus one of data bytes (i.e.
> >>> audio level values) transported in this header extension element
> >>> following the one-byte header.  Therefore, the value zero in
> >>> this field indicates that one byte of data follows.  A value of
> >>> 15 is not allowed by this specification and it MUST NOT be used
> >>> as the RTP header can carry a maximum of 15 CSRC IDs.  The
> >>> maximum value allowed is therefore 14 indicating a following
> >>> sequence of 15 audio level values.
> >>>
> >>> I suspect that any RFC 2119 language here actually belongs in a
> >>> procedural section, but I was also unclear what the real
> >>> conformance requirement was here.
> >>
> >> I am not sure I understand this comment.
> >>
> > As written, I found it very different to think of a test for the
> > requirement based only on the MUST.
> >
> >
> >>> Surely the main conformance requirement is to carry 15 or less
> >>> audio levels. This results in a maximum encoding of 14.
> >>
> >> Yes this is what we had in mind and what the text currently says.
> >> Is there a problem with it? Could you maybe suggest a better
> >> wording?
> >>
> > Without investigating deeper I was unclear what the limitation is
> > here. Is the value 15 possible but reserved elsewhere for extension
> > or what.
> >
> > If the coding already doesn't allow the value 15, but only the values
> > 0 through 14, then I would assume we just need an informative
> > statement pointing out the consequences or the normative
> > requirement.
> >
> > In the value 15 or more is allowed, then I would suggest just writing
> > a statement that says "Implementations MUST NOT include more than 15
> > audio level values."
>=20
> OK I see. So how about this:
>=20
>    The 4-bit len field is the number minus one of data bytes (i.e. audio
>    level values) transported in this header extension element following
>    the one-byte header.  Therefore, the value zero in this field
>    indicates that one byte of data follows.  RFC 3550 [RFC3550] only
>    allows RTP packets to carry a maximum of 15 CSRC IDs.  Given that
>    audio levels directly refer to CSRC IDs, implementations MUST NOT
>    include more than 15 audio level values.  The maximum value allowed
>    in the len field is therefore 14.
>=20
> Emil


From magnus.westerlund@ericsson.com  Fri May  6 07:18:12 2011
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Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
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On 2011-04-13 16:49, Jinwei Xia wrote:

>> Can we conclude that:
>>
>> Translator:
>>     pros:  allow primary source know the situation of full path.
>>     cons: more considertaion for media adaptation; changing RTP SN and/or
>> TS, and that RTCP.
>>
>> Mixer:
>>     pros: straight forward approach for media adaptation.
>>     cons: primary source only know the situation between itself and mixer;
>> changing RTP SN, TS, SSRC and CSRC, and that RTCP.
>>
>> Is there any missing pros & cons of translator and mixer on splicing?

Sorry for dropping out of the discussion.

I think that the document needs to have a little bit of discussion of
the two main approaches and the pro and con list wit a little more
detail than the above. That as I think it is important it is clear under
what premises and reasoning the path was selected.

Then I think we should form a consensus on which approach we think is
the more suitable to describe in all its gory details.

But if minimal detectability is gone, then it might be that translator
becomes more palatable for me.

I am still not certain how one can avoid continues offset rewriting of
rtp sequence number, timestamp etc unless one uses multiple SSRC in
either a mixer or a translator usage.

Cheers

Magnus Westerlund

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


From xiajinwei@huawei.com  Mon May  9 01:22:08 2011
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Magnus,=20

Thank you for picking up the discussion.  Let's categorize RTP splicing =
to
detectability and non-detectability.=20

In non-detectable case, The RTP header parameters must remain consistent
regardless of the media content the RTP conveys.  We can analyse the
pron&cons of two approaches as following four aspects:

RTP: translator should rewrite the SN and TS when the amount of
substitutived packets is not equal to amount of main packets replaced.
Translator must execute transcoding if the media type are different =
between
the two streams; Mixer should initiate its own SN and TS for the new =
media
stream, or transcoding, additionally mixer should change the CSRC for =
loop
detection. But the CSRC difference can allow downstream client to be =
able to
detect the switch.

RTCP: translator should rewrite RTCP traffic passing through it to =
reflect
the real characteristics of each other; Mixer divides RTCP flow between
source and user into two separate RTCP loops, hence media source has no =
idea
about the situation on user. Compared to translator, mixer is a more
straight forward approach on RTCP.=20

Security:  If encryption is employed, the media translator and mixer =
need to
be able to decrypt and re-encrypt the media stream. The difference is
translator use one cryptographic key per user, mixer use key per domain.
There is not strong requirement to hide translator or mixer from =
receiver,
but just there is one potential security issue: mixer exposes its =
address to
receivers and may get attacks from vicious receivers.

Overhead: neither translaotr nor mixer can avoid the overhead while no
splicing occur. When security mechanisms are used, there is more =
overhead on
translator since it needs to prepare multiple different keying material =
for
different receiver.


In non-detectable case, I personally incline to translator since =
translator
can totally eliminate the difference between the two stream and erase =
the
potential security issue.


In detectable case, the thing becomes simpler. translator is suitable to
handle detectable splicing. When splicing ends, translator resumes main
stream and rewrites the sequence number one higher than the last packet =
sent
to receiver in the main stream. There is a little overhead on translator =
to
rewrite RTP SN. If media sender support SSRC multiplexing, the thing =
becomes
simplest, the splicer can be a normal "switch node" which swith between
multiple streams with different SSRCs to realize zero overhead. The
appropriate solution is decided on whether the headend uses SSRC
multiplexing mechanism.=20


BR
Jinwei


> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Friday, May 06, 2011 10:18 PM
> To: Jinwei Xia
> Cc: DRAGE, Keith (Keith); avtext@ietf.org
> Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
>=20
> On 2011-04-13 16:49, Jinwei Xia wrote:
>=20
> >> Can we conclude that:
> >>
> >> Translator:
> >>     pros:  allow primary source know the situation of full path.
> >>     cons: more considertaion for media adaptation; changing RTP SN=20
> >> and/or TS, and that RTCP.
> >>
> >> Mixer:
> >>     pros: straight forward approach for media adaptation.
> >>     cons: primary source only know the situation between=20
> itself and=20
> >> mixer; changing RTP SN, TS, SSRC and CSRC, and that RTCP.
> >>
> >> Is there any missing pros & cons of translator and mixer=20
> on splicing?
>=20
> Sorry for dropping out of the discussion.
>=20
> I think that the document needs to have a little bit of=20
> discussion of the two main approaches and the pro and con=20
> list wit a little more detail than the above. That as I think=20
> it is important it is clear under what premises and reasoning=20
> the path was selected.
>=20
> Then I think we should form a consensus on which approach we=20
> think is the more suitable to describe in all its gory details.
>=20
> But if minimal detectability is gone, then it might be that=20
> translator becomes more palatable for me.
>=20
> I am still not certain how one can avoid continues offset=20
> rewriting of rtp sequence number, timestamp etc unless one=20
> uses multiple SSRC in either a mixer or a translator usage.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20


From internet-drafts@ietf.org  Mon May  9 03:05:51 2011
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Subject: [avtext] I-D Action: draft-ietf-avtext-mixer-to-client-audio-level-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Extensions Work=
ing Group of the IETF.

	Title           : A Real-Time Transport Protocol (RTP) Header Extension fo=
r Mixer-to- Client Audio Level Indication
	Author(s)       : Emil Ivov
                          Enrico Marocco
                          Jonathan Lennox
	Filename        : draft-ietf-avtext-mixer-to-client-audio-level-02.txt
	Pages           : 15
	Date            : 2011-05-09

   This document describes a mechanism for RTP-level mixers in audio
   conferences to deliver information about the audio level of
   individual participants.  Such audio level indicators are transported
   in the same RTP packets as the audio data they pertain to.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-mixer-to-client-audio=
-level-02.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtext-mixer-to-client-audio-=
level-02.txt

From emil@sip-communicator.org  Mon May  9 03:15:23 2011
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Subject: [avtext] mixer-to-client audio levels ready (Was: I-D Action: draft-ietf-avtext-mixer-to-client-audio-level-02.txt)
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Hey all,

The authors are confident that all open issues and mailing list comments
have now been addressed and that this document is now ready to move forwa=
rd.

We will be submitting the client-to-mixer draft in the following days.

Cheers,
Emil

=D0=9D=D0=B0 09.05.11 12:05, internet-drafts@ietf.org =D0=BD=D0=B0=D0=BF=D0=
=B8=D1=81=D0=B0:
> A New Internet-Draft is available from the on-line Internet-Drafts dire=
ctories. This draft is a work item of the Audio/Video Transport Extension=
s Working Group of the IETF.
>=20
> 	Title           : A Real-Time Transport Protocol (RTP) Header Extensio=
n for Mixer-to- Client Audio Level Indication
> 	Author(s)       : Emil Ivov
>                           Enrico Marocco
>                           Jonathan Lennox
> 	Filename        : draft-ietf-avtext-mixer-to-client-audio-level-02.txt=

> 	Pages           : 15
> 	Date            : 2011-05-09
>=20
>    This document describes a mechanism for RTP-level mixers in audio
>    conferences to deliver information about the audio level of
>    individual participants.  Such audio level indicators are transporte=
d
>    in the same RTP packets as the audio data they pertain to.
>=20
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avtext-mixer-to-client-a=
udio-level-02.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtext-mixer-to-client-au=
dio-level-02.txt
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>=20

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31


From magnus.westerlund@ericsson.com  Thu May 12 02:19:04 2011
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Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
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On 2011-05-09 10:21, Xiajinwei wrote:
> 
> Magnus, 
> 
> Thank you for picking up the discussion.  Let's categorize RTP splicing to
> detectability and non-detectability. 
> 
> In non-detectable case, The RTP header parameters must remain consistent
> regardless of the media content the RTP conveys.  We can analyse the
> pron&cons of two approaches as following four aspects:

First of all non-detectable is still detectable just by looking at the
media stream. For example a non-detectable splicing operation with a
video stream using H.264 will still be detectable just because that
streams encoded by different encoders will have different parameter
sets. Only if you do full media transcoding in the splicer can you avoid
this. This is both complex and reduces quality thus I hope this is not
expected.

> 
> RTP: translator should rewrite the SN and TS when the amount of
> substitutived packets is not equal to amount of main packets replaced.
> Translator must execute transcoding if the media type are different between
> the two streams; Mixer should initiate its own SN and TS for the new media
> stream, or transcoding, additionally mixer should change the CSRC for loop
> detection. But the CSRC difference can allow downstream client to be able to
> detect the switch.

Hmmm, I am not certain I agree with this description or usage of the
models.

Lets start with the picture

Source1 ---> Splicer ----> receiver
source2 -------^

In a translator case source1 view of the session is that it sees RTCP
feedback from the receiver, at least when its content is forwarded. When
it splices in source 2, then it creates a consistent continuation of
source1 stream by setting SN and TS, while the splicer provide feedback
on delivery to source 1 while it splices instead of the receiver. When
it goes back to source1 it still will need to change SN and TS to
maintain previous consistency.

In a mixer case the splicer will use its own SSRC always and just select
which of the source it uses for the media in the stream. Which means
that the mixer owns the SN and TS space it forwards the media using.
Also feedback will always be separated for the leg to the mixer and the
leg between the mixer and the receiver. If the sources needs to react to
downstream events additional RTCP extensions are needed.
In the mixer case the mixer can select to use CSRC and forward the
corresponding SDES items or not.



> 
> RTCP: translator should rewrite RTCP traffic passing through it to reflect
> the real characteristics of each other; Mixer divides RTCP flow between
> source and user into two separate RTCP loops, hence media source has no idea
> about the situation on user. Compared to translator, mixer is a more
> straight forward approach on RTCP. 

Yes, except that if response to other leg is needed, then codec control
messages or similar are needed.

> 
> Security:  If encryption is employed, the media translator and mixer need to
> be able to decrypt and re-encrypt the media stream. The difference is
> translator use one cryptographic key per user, mixer use key per domain.
> There is not strong requirement to hide translator or mixer from receiver,
> but just there is one potential security issue: mixer exposes its address to
> receivers and may get attacks from vicious receivers.

I don't understand why you come to the conclusion about why translators
and mixers uses different keys. That is up to the implementation.

And if we are talking about IPTV we are anyway talking about payload
internal encryption which neither the boxes will touch unless one has
decided to to media transcoding which seems expensive for this.

> 
> Overhead: neither translaotr nor mixer can avoid the overhead while no
> splicing occur. When security mechanisms are used, there is more overhead on
> translator since it needs to prepare multiple different keying material for
> different receiver.

I don't see the need for having different security overhead between
translator and mixer.

> 
> 
> In non-detectable case, I personally incline to translator since translator
> can totally eliminate the difference between the two stream and erase the
> potential security issue.

I think both can realize the non-detectable case.

> 
> 
> In detectable case, the thing becomes simpler. translator is suitable to
> handle detectable splicing. When splicing ends, translator resumes main
> stream and rewrites the sequence number one higher than the last packet sent
> to receiver in the main stream. There is a little overhead on translator to
> rewrite RTP SN. 

What about the TS? I think that may become inconsistent and thus needs
rewriting also.

If media sender support SSRC multiplexing, the thing becomes
> simplest, the splicer can be a normal "switch node" which swith between
> multiple streams with different SSRCs to realize zero overhead. The
> appropriate solution is decided on whether the headend uses SSRC
> multiplexing mechanism. 

I think the question is not if the sender supports SSRC multiplexing,
but if the receivers do. Sending two different RTP streams and hoping
that they are nicely inserted in the gap is possible but not necessarily
trivial.

Cheers

Magnus Westerlund

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


From stewe@stewe.org  Thu May 12 07:54:33 2011
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Hi Magnus,

Inline...

On 5.12.2011 02:18 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>On 2011-05-09 10:21, Xiajinwei wrote:
>>=20
>> Magnus,=20
>>=20
>> Thank you for picking up the discussion.  Let's categorize RTP splicing
>>to
>> detectability and non-detectability.
>>=20
>> In non-detectable case, The RTP header parameters must remain consistent
>> regardless of the media content the RTP conveys.  We can analyse the
>> pron&cons of two approaches as following four aspects:
>
>First of all non-detectable is still detectable just by looking at the
>media stream. For example a non-detectable splicing operation with a
>video stream using H.264 will still be detectable just because that
>streams encoded by different encoders will have different parameter
>sets. Only if you do full media transcoding in the splicer can you avoid
>this. This is both complex and reduces quality thus I hope this is not
>expected.


I don't think your example is a good one.  In H.264 (as well as in most
other video compression standards) there are certain points in the
bitstream which break bitstream prediction.  The data (after decoding)
between those points is called a sequence.  A decoder cannot, without
relying on heuristic means, determine whether a bitstream composed of more
than one sequence was produced by one or more than one encoder.

It's equally possible that two decoders use the same parameter sets.  It's
not even unlikely that they do, especially if you are using a "standard"
set of tools and image formats, probably mandated by a profiling
organization such as 3GPP.

At the risk of being proven wrong in an exotic use case, I would even
argue that, with most media codecs, it is very hard or even impossible, to
identify properly implemented splicing without side information of some
kind.

I believe I agree with the rest of Magnus' comments below (without having
thought about them in much detail).

Stephan


>
>>=20
>> RTP: translator should rewrite the SN and TS when the amount of
>> substitutived packets is not equal to amount of main packets replaced.
>> Translator must execute transcoding if the media type are different
>>between
>> the two streams; Mixer should initiate its own SN and TS for the new
>>media
>> stream, or transcoding, additionally mixer should change the CSRC for
>>loop
>> detection. But the CSRC difference can allow downstream client to be
>>able to
>> detect the switch.
>
>Hmmm, I am not certain I agree with this description or usage of the
>models. =20
>
>Lets start with the picture
>
>Source1 ---> Splicer ----> receiver
>source2 -------^
>
>In a translator case source1 view of the session is that it sees RTCP
>feedback from the receiver, at least when its content is forwarded. When
>it splices in source 2, then it creates a consistent continuation of
>source1 stream by setting SN and TS, while the splicer provide feedback
>on delivery to source 1 while it splices instead of the receiver. When
>it goes back to source1 it still will need to change SN and TS to
>maintain previous consistency.
>
>In a mixer case the splicer will use its own SSRC always and just select
>which of the source it uses for the media in the stream. Which means
>that the mixer owns the SN and TS space it forwards the media using.
>Also feedback will always be separated for the leg to the mixer and the
>leg between the mixer and the receiver. If the sources needs to react to
>downstream events additional RTCP extensions are needed.
>In the mixer case the mixer can select to use CSRC and forward the
>corresponding SDES items or not.
>
>
>
>>=20
>> RTCP: translator should rewrite RTCP traffic passing through it to
>>reflect
>> the real characteristics of each other; Mixer divides RTCP flow between
>> source and user into two separate RTCP loops, hence media source has no
>>idea
>> about the situation on user. Compared to translator, mixer is a more
>> straight forward approach on RTCP.
>
>Yes, except that if response to other leg is needed, then codec control
>messages or similar are needed.
>
>>=20
>> Security:  If encryption is employed, the media translator and mixer
>>need to
>> be able to decrypt and re-encrypt the media stream. The difference is
>> translator use one cryptographic key per user, mixer use key per domain.
>> There is not strong requirement to hide translator or mixer from
>>receiver,
>> but just there is one potential security issue: mixer exposes its
>>address to
>> receivers and may get attacks from vicious receivers.
>
>I don't understand why you come to the conclusion about why translators
>and mixers uses different keys. That is up to the implementation.
>
>And if we are talking about IPTV we are anyway talking about payload
>internal encryption which neither the boxes will touch unless one has
>decided to to media transcoding which seems expensive for this.
>
>>=20
>> Overhead: neither translaotr nor mixer can avoid the overhead while no
>> splicing occur. When security mechanisms are used, there is more
>>overhead on
>> translator since it needs to prepare multiple different keying material
>>for
>> different receiver.
>
>I don't see the need for having different security overhead between
>translator and mixer.
>
>>=20
>>=20
>> In non-detectable case, I personally incline to translator since
>>translator
>> can totally eliminate the difference between the two stream and erase
>>the
>> potential security issue.
>
>I think both can realize the non-detectable case.
>
>>=20
>>=20
>> In detectable case, the thing becomes simpler. translator is suitable to
>> handle detectable splicing. When splicing ends, translator resumes main
>> stream and rewrites the sequence number one higher than the last packet
>>sent
>> to receiver in the main stream. There is a little overhead on
>>translator to
>> rewrite RTP SN.=20
>
>What about the TS? I think that may become inconsistent and thus needs
>rewriting also.
>
>If media sender support SSRC multiplexing, the thing becomes
>> simplest, the splicer can be a normal "switch node" which swith between
>> multiple streams with different SSRCs to realize zero overhead. The
>> appropriate solution is decided on whether the headend uses SSRC
>> multiplexing mechanism.
>
>I think the question is not if the sender supports SSRC multiplexing,
>but if the receivers do. Sending two different RTP streams and hoping
>that they are nicely inserted in the gap is possible but not necessarily
>trivial.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>avtext mailing list
>avtext@ietf.org
>https://www.ietf.org/mailman/listinfo/avtext



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Magnus,

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Thursday, May 12, 2011 5:19 PM
> To: Xiajinwei
> Cc: 'DRAGE, Keith (Keith)'; avtext@ietf.org
> Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
>=20
> On 2011-05-09 10:21, Xiajinwei wrote:
> >=20
> > Magnus,
> >=20
> > Thank you for picking up the discussion.  Let's categorize RTP=20
> > splicing to detectability and non-detectability.
> >=20
> > In non-detectable case, The RTP header parameters must remain=20
> > consistent regardless of the media content the RTP conveys.  We can=20
> > analyse the pron&cons of two approaches as following four aspects:
>=20
> First of all non-detectable is still detectable just by=20
> looking at the media stream. For example a non-detectable=20
> splicing operation with a video stream using H.264 will still=20
> be detectable just because that streams encoded by different=20
> encoders will have different parameter sets. Only if you do=20
> full media transcoding in the splicer can you avoid this.=20
> This is both complex and reduces quality thus I hope this is=20
> not expected.
>=20

I agree minimum-detectable is more appropriate. In
draft-xia-avtext-splicing-for-rtp-00, I mentioned the RTP header =
parameters
MUST remain consistent except that meida type SHOULD be same since
transcoding is prohibitively expensive and complex, so it  is =
RECOMMENDED to
use same media type in insertion stream as that in main stream. If an
operator want to realize complete non-detectable on client, he must take =
the
tranacoding cost in account.=20

Moreover, Splicer SHOULD also realize minimum-detectable in application
layer, for example, reusing existing PIDs after a splicing described in
[SMPTE-321M]. How this is done is an implementation detail and not =
discussed
in this draft.

So how about above change?=20

> >=20
> > RTP: translator should rewrite the SN and TS when the amount of=20
> > substitutived packets is not equal to amount of main=20
> packets replaced.
> > Translator must execute transcoding if the media type are different=20
> > between the two streams; Mixer should initiate its own SN=20
> and TS for=20
> > the new media stream, or transcoding, additionally mixer=20
> should change=20
> > the CSRC for loop detection. But the CSRC difference can allow=20
> > downstream client to be able to detect the switch.
>=20
> Hmmm, I am not certain I agree with this description or usage=20
> of the models.
>=20
> Lets start with the picture
>=20
> Source1 ---> Splicer ----> receiver
> source2 -------^
>=20
> In a translator case source1 view of the session is that it=20
> sees RTCP feedback from the receiver, at least when its=20
> content is forwarded. When it splices in source 2, then it=20
> creates a consistent continuation of
> source1 stream by setting SN and TS, while the splicer=20
> provide feedback on delivery to source 1 while it splices=20
> instead of the receiver. When it goes back to source1 it=20
> still will need to change SN and TS to maintain previous consistency.
>=20
> In a mixer case the splicer will use its own SSRC always and=20
> just select which of the source it uses for the media in the=20
> stream. Which means that the mixer owns the SN and TS space=20
> it forwards the media using.
> Also feedback will always be separated for the leg to the=20
> mixer and the leg between the mixer and the receiver. If the=20
> sources needs to react to downstream events additional RTCP=20
> extensions are needed.
> In the mixer case the mixer can select to use CSRC and=20
> forward the corresponding SDES items or not.
>=20
>=20

Your description is more detailed, also including RTCP processing. =
Compared
to translator, mixer change the CSRC field in RTP header during =
splicing.

>=20
> >=20
> > RTCP: translator should rewrite RTCP traffic passing through it to=20
> > reflect the real characteristics of each other; Mixer divides RTCP=20
> > flow between source and user into two separate RTCP loops,=20
> hence media=20
> > source has no idea about the situation on user. Compared to=20
> > translator, mixer is a more straight forward approach on RTCP.
>=20
> Yes, except that if response to other leg is needed, then=20
> codec control messages or similar are needed.
>=20

True. RFC5104 can repair it.

> >=20
> > Security:  If encryption is employed, the media translator=20
> and mixer=20
> > need to be able to decrypt and re-encrypt the media stream. The=20
> > difference is translator use one cryptographic key per=20
> user, mixer use key per domain.
> > There is not strong requirement to hide translator or mixer from=20
> > receiver, but just there is one potential security issue: mixer=20
> > exposes its address to receivers and may get attacks from=20
> vicious receivers.
>=20
> I don't understand why you come to the conclusion about why=20
> translators and mixers uses different keys. That is up to the=20
> implementation.
>=20
> And if we are talking about IPTV we are anyway talking about=20
> payload internal encryption which neither the boxes will=20
> touch unless one has decided to to media transcoding which=20
> seems expensive for this.

an translator change the payload of RTP during splicing with SSRC =
intact,
should it use different keys for different segments?=20

>=20
> >=20
> > Overhead: neither translaotr nor mixer can avoid the=20
> overhead while no=20
> > splicing occur. When security mechanisms are used, there is more=20
> > overhead on translator since it needs to prepare multiple different=20
> > keying material for different receiver.
>=20
> I don't see the need for having different security overhead=20
> between translator and mixer.
>=20
> >=20
> >=20
> > In non-detectable case, I personally incline to translator since=20
> > translator can totally eliminate the difference between the=20
> two stream=20
> > and erase the potential security issue.
>=20
> I think both can realize the non-detectable case.
>=20

Yes, both can do that. The problem is which one is better. What's your
prefer in the two candidates or otherwise?

> >=20
> >=20
> > In detectable case, the thing becomes simpler. translator=20
> is suitable=20
> > to handle detectable splicing. When splicing ends,=20
> translator resumes=20
> > main stream and rewrites the sequence number one higher=20
> than the last=20
> > packet sent to receiver in the main stream. There is a=20
> little overhead=20
> > on translator to rewrite RTP SN.
>=20
> What about the TS? I think that may become inconsistent and=20
> thus needs rewriting also.

Sorry for missing it again, the TS is higher than the last inserted =
packet
to maintain consistency.

>=20
> If media sender support SSRC multiplexing, the thing becomes
> > simplest, the splicer can be a normal "switch node" which swith=20
> > between multiple streams with different SSRCs to realize zero=20
> > overhead. The appropriate solution is decided on whether=20
> the headend=20
> > uses SSRC multiplexing mechanism.
>=20
> I think the question is not if the sender supports SSRC=20
> multiplexing, but if the receivers do. Sending two different=20
> RTP streams and hoping that they are nicely inserted in the=20
> gap is possible but not necessarily trivial.

Yes, this is inner implementation to fill up the gap. Using SSRC
multiplexing may be the most straight forward approach for detectable =
case.


Thank you


Jinwei

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


From magnus.westerlund@ericsson.com  Fri May 13 01:19:00 2011
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On 2011-05-13 01:54, Stephan Wenger wrote:
> Hi Magnus,
> 
> Inline...
> 
> On 5.12.2011 02:18 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
> wrote:
> 
>> On 2011-05-09 10:21, Xiajinwei wrote:
>>>
>>> Magnus, 
>>>
>>> Thank you for picking up the discussion.  Let's categorize RTP splicing
>>> to
>>> detectability and non-detectability.
>>>
>>> In non-detectable case, The RTP header parameters must remain consistent
>>> regardless of the media content the RTP conveys.  We can analyse the
>>> pron&cons of two approaches as following four aspects:
>>
>> First of all non-detectable is still detectable just by looking at the
>> media stream. For example a non-detectable splicing operation with a
>> video stream using H.264 will still be detectable just because that
>> streams encoded by different encoders will have different parameter
>> sets. Only if you do full media transcoding in the splicer can you avoid
>> this. This is both complex and reduces quality thus I hope this is not
>> expected.
> 
> 
> I don't think your example is a good one.  In H.264 (as well as in most
> other video compression standards) there are certain points in the
> bitstream which break bitstream prediction.  The data (after decoding)
> between those points is called a sequence.  A decoder cannot, without
> relying on heuristic means, determine whether a bitstream composed of more
> than one sequence was produced by one or more than one encoder.
> 
> It's equally possible that two decoders use the same parameter sets.  It's
> not even unlikely that they do, especially if you are using a "standard"
> set of tools and image formats, probably mandated by a profiling
> organization such as 3GPP.
> 
> At the risk of being proven wrong in an exotic use case, I would even
> argue that, with most media codecs, it is very hard or even impossible, to
> identify properly implemented splicing without side information of some
> kind.

I agree that it is not certain that you can detect, but in certain case
you will be able to detect, at least using heuristics.

Yes, if you ensure a certain level of harmonization between the encoding
I agree you can make it difficult to detect.

Cheers

Magnus Westerlund

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


From csp@csperkins.org  Mon May 16 03:47:21 2011
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On 12 May 2011, at 10:18, Magnus Westerlund wrote:
> On 2011-05-09 10:21, Xiajinwei wrote:
>>=20
>> Magnus,=20
>>=20
>> Thank you for picking up the discussion.  Let's categorize RTP =
splicing to
>> detectability and non-detectability.=20
>>=20
>> In non-detectable case, The RTP header parameters must remain =
consistent
>> regardless of the media content the RTP conveys.  We can analyse the
>> pron&cons of two approaches as following four aspects:
>=20
> First of all non-detectable is still detectable just by looking at the
> media stream. For example a non-detectable splicing operation with a
> video stream using H.264 will still be detectable just because that
> streams encoded by different encoders will have different parameter
> sets. Only if you do full media transcoding in the splicer can you =
avoid
> this. This is both complex and reduces quality thus I hope this is not
> expected.
>=20
>>=20
>> RTP: translator should rewrite the SN and TS when the amount of
>> substitutived packets is not equal to amount of main packets =
replaced.
>> Translator must execute transcoding if the media type are different =
between
>> the two streams; Mixer should initiate its own SN and TS for the new =
media
>> stream, or transcoding, additionally mixer should change the CSRC for =
loop
>> detection. But the CSRC difference can allow downstream client to be =
able to
>> detect the switch.
>=20
> Hmmm, I am not certain I agree with this description or usage of the
> models.
>=20
> Lets start with the picture
>=20
> Source1 ---> Splicer ----> receiver
> source2 -------^
>=20
> In a translator case source1 view of the session is that it sees RTCP
> feedback from the receiver, at least when its content is forwarded. =
When
> it splices in source 2, then it creates a consistent continuation of
> source1 stream by setting SN and TS, while the splicer provide =
feedback
> on delivery to source 1 while it splices instead of the receiver. When
> it goes back to source1 it still will need to change SN and TS to
> maintain previous consistency.
>=20
> In a mixer case the splicer will use its own SSRC always and just =
select
> which of the source it uses for the media in the stream. Which means
> that the mixer owns the SN and TS space it forwards the media using.
> Also feedback will always be separated for the leg to the mixer and =
the
> leg between the mixer and the receiver. If the sources needs to react =
to
> downstream events additional RTCP extensions are needed.
> In the mixer case the mixer can select to use CSRC and forward the
> corresponding SDES items or not.
>=20
>=20
>=20
>>=20
>> RTCP: translator should rewrite RTCP traffic passing through it to =
reflect
>> the real characteristics of each other; Mixer divides RTCP flow =
between
>> source and user into two separate RTCP loops, hence media source has =
no idea
>> about the situation on user. Compared to translator, mixer is a more
>> straight forward approach on RTCP.=20
>=20
> Yes, except that if response to other leg is needed, then codec =
control
> messages or similar are needed.
>=20
>>=20
>> Security:  If encryption is employed, the media translator and mixer =
need to
>> be able to decrypt and re-encrypt the media stream. The difference is
>> translator use one cryptographic key per user, mixer use key per =
domain.
>> There is not strong requirement to hide translator or mixer from =
receiver,
>> but just there is one potential security issue: mixer exposes its =
address to
>> receivers and may get attacks from vicious receivers.
>=20
> I don't understand why you come to the conclusion about why =
translators
> and mixers uses different keys. That is up to the implementation.
>=20
> And if we are talking about IPTV we are anyway talking about payload
> internal encryption which neither the boxes will touch unless one has
> decided to to media transcoding which seems expensive for this.
>=20
>>=20
>> Overhead: neither translaotr nor mixer can avoid the overhead while =
no
>> splicing occur. When security mechanisms are used, there is more =
overhead on
>> translator since it needs to prepare multiple different keying =
material for
>> different receiver.
>=20
> I don't see the need for having different security overhead between
> translator and mixer.
>=20
>>=20
>>=20
>> In non-detectable case, I personally incline to translator since =
translator
>> can totally eliminate the difference between the two stream and erase =
the
>> potential security issue.
>=20
> I think both can realize the non-detectable case.
>=20
>>=20
>>=20
>> In detectable case, the thing becomes simpler. translator is suitable =
to
>> handle detectable splicing. When splicing ends, translator resumes =
main
>> stream and rewrites the sequence number one higher than the last =
packet sent
>> to receiver in the main stream. There is a little overhead on =
translator to
>> rewrite RTP SN.=20
>=20
> What about the TS? I think that may become inconsistent and thus needs
> rewriting also.

I'd think it reasonable to require that the inserted content MUST be the =
same duration as the original content. Otherwise, not only will you need =
to rewrite the timestamps after the splicing has ended, you'll either =
have a gap or extra data to be buffered somewhere. You'll only have to =
rewrite timestamps during the splicing, to normalise the random offset.

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




From magnus.westerlund@ericsson.com  Mon May 16 04:41:31 2011
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WG,

AVTEXT has one milestone:

Dec 2011 Submit Support for multiple clock rates in an RTP session
         for Proposed Standard

We have one candidate document:

https://datatracker.ietf.org/doc/draft-petithuguenin-avtext-multiple-clock-rates/

This is the call to the WG if we want to adopt this document as the
basis for meeting that milestone? Please provide comments even if it is
just "Yes".

If you think there is some aspect of the issue or the solution that
hasn't been discussed and you want to ensure is covered, please comment
on that.

The WG chairs will make a decision on 31st of May based on your comments.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Tue May 17 00:14:36 2011
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Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
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On 2011-05-13 05:54, Xiajinwei wrote:
> 
> Magnus,
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
>> Sent: Thursday, May 12, 2011 5:19 PM
>> To: Xiajinwei
>> Cc: 'DRAGE, Keith (Keith)'; avtext@ietf.org
>> Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
>>
>> On 2011-05-09 10:21, Xiajinwei wrote:
>>>
>>> Magnus,
>>>
>>> Thank you for picking up the discussion.  Let's categorize RTP 
>>> splicing to detectability and non-detectability.
>>>
>>> In non-detectable case, The RTP header parameters must remain 
>>> consistent regardless of the media content the RTP conveys.  We can 
>>> analyse the pron&cons of two approaches as following four aspects:
>>
>> First of all non-detectable is still detectable just by 
>> looking at the media stream. For example a non-detectable 
>> splicing operation with a video stream using H.264 will still 
>> be detectable just because that streams encoded by different 
>> encoders will have different parameter sets. Only if you do 
>> full media transcoding in the splicer can you avoid this. 
>> This is both complex and reduces quality thus I hope this is 
>> not expected.
>>
> 
> I agree minimum-detectable is more appropriate. In
> draft-xia-avtext-splicing-for-rtp-00, I mentioned the RTP header parameters
> MUST remain consistent except that meida type SHOULD be same since
> transcoding is prohibitively expensive and complex, so it  is RECOMMENDED to
> use same media type in insertion stream as that in main stream. If an
> operator want to realize complete non-detectable on client, he must take the
> tranacoding cost in account. 
> 
> Moreover, Splicer SHOULD also realize minimum-detectable in application
> layer, for example, reusing existing PIDs after a splicing described in
> [SMPTE-321M]. How this is done is an implementation detail and not discussed
> in this draft.
> 
> So how about above change? 
> 


I guess PIDs are protocol IDs. I think one needs to have a more detailed
discussion on what IDs can be reused and when they can't.

>>>
>>> RTP: translator should rewrite the SN and TS when the amount of 
>>> substitutived packets is not equal to amount of main 
>> packets replaced.
>>> Translator must execute transcoding if the media type are different 
>>> between the two streams; Mixer should initiate its own SN 
>> and TS for 
>>> the new media stream, or transcoding, additionally mixer 
>> should change 
>>> the CSRC for loop detection. But the CSRC difference can allow 
>>> downstream client to be able to detect the switch.
>>
>> Hmmm, I am not certain I agree with this description or usage 
>> of the models.
>>
>> Lets start with the picture
>>
>> Source1 ---> Splicer ----> receiver
>> source2 -------^
>>
>> In a translator case source1 view of the session is that it 
>> sees RTCP feedback from the receiver, at least when its 
>> content is forwarded. When it splices in source 2, then it 
>> creates a consistent continuation of
>> source1 stream by setting SN and TS, while the splicer 
>> provide feedback on delivery to source 1 while it splices 
>> instead of the receiver. When it goes back to source1 it 
>> still will need to change SN and TS to maintain previous consistency.
>>
>> In a mixer case the splicer will use its own SSRC always and 
>> just select which of the source it uses for the media in the 
>> stream. Which means that the mixer owns the SN and TS space 
>> it forwards the media using.
>> Also feedback will always be separated for the leg to the 
>> mixer and the leg between the mixer and the receiver. If the 
>> sources needs to react to downstream events additional RTCP 
>> extensions are needed.
>> In the mixer case the mixer can select to use CSRC and 
>> forward the corresponding SDES items or not.
>>
>>
> 
> Your description is more detailed, also including RTCP processing. Compared
> to translator, mixer change the CSRC field in RTP header during splicing.
> 

It may do that, it is not a strict requirement to use CSRC in mixers.
And in a splicer there appears to be benefits of not including the CRSC.

>>>
>>> Security:  If encryption is employed, the media translator 
>> and mixer 
>>> need to be able to decrypt and re-encrypt the media stream. The 
>>> difference is translator use one cryptographic key per 
>> user, mixer use key per domain.
>>> There is not strong requirement to hide translator or mixer from 
>>> receiver, but just there is one potential security issue: mixer 
>>> exposes its address to receivers and may get attacks from 
>> vicious receivers.
>>
>> I don't understand why you come to the conclusion about why 
>> translators and mixers uses different keys. That is up to the 
>> implementation.
>>
>> And if we are talking about IPTV we are anyway talking about 
>> payload internal encryption which neither the boxes will 
>> touch unless one has decided to to media transcoding which 
>> seems expensive for this.
> 
> an translator change the payload of RTP during splicing with SSRC intact,
> should it use different keys for different segments? 

For the payload internal stuff people are using, it can likely handle
such key changes using security payload internal mechanism. The whole
point with these payload internal mechanisms is that no entity between
the content source and the decoder actually needs to have the key.

If one uses regular transport security like SRTP or DTLS, IPsec, etc.
then you need to re-process all the packets if you change something in
them. Thus different key-context for the different legs makes sense.

> 
>>
>>>
>>> Overhead: neither translaotr nor mixer can avoid the 
>> overhead while no 
>>> splicing occur. When security mechanisms are used, there is more 
>>> overhead on translator since it needs to prepare multiple different 
>>> keying material for different receiver.
>>
>> I don't see the need for having different security overhead 
>> between translator and mixer.
>>
>>>
>>>
>>> In non-detectable case, I personally incline to translator since 
>>> translator can totally eliminate the difference between the 
>> two stream 
>>> and erase the potential security issue.
>>
>> I think both can realize the non-detectable case.
>>
> 
> Yes, both can do that. The problem is which one is better. What's your
> prefer in the two candidates or otherwise?

I think it comes down to how one wants to organize the splicer and the
upstream content sources. To which degree they should aware of the
splicer and if requiring changes to them are acceptable, specifically
the original content source.

If modifying the original content source is out of the picture then I
think a translator is the right way to go, although I believe it will
require a bit more text to describe all the aspects one needs to think
about.

If one can modify the original content source, then I think a mixer is
more straight forward and thus easier to describe.

What the answer is to the modifying question I don't know. Input is
desirable.


>>
>> If media sender support SSRC multiplexing, the thing becomes
>>> simplest, the splicer can be a normal "switch node" which swith 
>>> between multiple streams with different SSRCs to realize zero 
>>> overhead. The appropriate solution is decided on whether 
>> the headend 
>>> uses SSRC multiplexing mechanism.
>>
>> I think the question is not if the sender supports SSRC 
>> multiplexing, but if the receivers do. Sending two different 
>> RTP streams and hoping that they are nicely inserted in the 
>> gap is possible but not necessarily trivial.
> 
> Yes, this is inner implementation to fill up the gap. Using SSRC
> multiplexing may be the most straight forward approach for detectable case.
> 

Yes, but also least likely to ensure glitch free media processing, at
least before you have updated your receivers to include a few mechanisms
to ensure that they can quickly and correctly create a common timeline.

Thus I think SSRC multiplexing is to quite a large degree a red herring
in this case. It will not work well in legacy case, and even with
updates it is still creating potential issues. Thus I think using a
single SSRC and inserting the content into the stream is the most likely
to work well.

Cheers

Magnus Westerlund

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On 2011-05-16 12:47, Colin Perkins wrote:

>>>
>>> In detectable case, the thing becomes simpler. translator is suitable to
>>> handle detectable splicing. When splicing ends, translator resumes main
>>> stream and rewrites the sequence number one higher than the last packet sent
>>> to receiver in the main stream. There is a little overhead on translator to
>>> rewrite RTP SN. 
>>
>> What about the TS? I think that may become inconsistent and thus needs
>> rewriting also.
> 
> I'd think it reasonable to require that the inserted content MUST be
> the same duration as the original content. Otherwise, not only will
> you need to rewrite the timestamps after the splicing has ended,
> you'll either have a gap or extra data to be buffered somewhere.
> You'll only have to rewrite timestamps during the splicing, to
> normalise the random offset.
> 

I agree that you need to have content to insert that is very close to
the space you are replacing. The question is what happens when things
are off with a video frame or less, or an audio frame or less.

One thing to worry about here is what happens if the frame rate varies
between the content, then you may have a small diff to take care of.
Yes, letting that be a bit extra delay between content in the point of
switching is back is likely fine, but it needs to be discussed in the
document.

I might have made the impression that this issue is bigger than it
really is.

Cheers

Magnus Westerlund

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Magnus,

Inline~~

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Tuesday, May 17, 2011 3:19 PM
> To: Colin Perkins
> Cc: Xiajinwei; avtext@ietf.org
> Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
>=20
> On 2011-05-16 12:47, Colin Perkins wrote:
>=20
> >>>
> >>> In detectable case, the thing becomes simpler. translator is=20
> >>> suitable to handle detectable splicing. When splicing ends,=20
> >>> translator resumes main stream and rewrites the sequence=20
> number one=20
> >>> higher than the last packet sent to receiver in the main stream.=20
> >>> There is a little overhead on translator to rewrite RTP SN.
> >>
> >> What about the TS? I think that may become inconsistent and thus=20
> >> needs rewriting also.
> >=20
> > I'd think it reasonable to require that the inserted=20
> content MUST be=20
> > the same duration as the original content. Otherwise, not only will=20
> > you need to rewrite the timestamps after the splicing has ended,=20
> > you'll either have a gap or extra data to be buffered somewhere.
> > You'll only have to rewrite timestamps during the splicing, to=20
> > normalise the random offset.
> >=20
>=20
> I agree that you need to have content to insert that is very=20
> close to the space you are replacing. The question is what=20
> happens when things are off with a video frame or less, or an=20
> audio frame or less.
>=20

Your concern make senses, one of splicing requirements is Splicer SHOULD =
NOT
cause perceptible media clipping at the splicing joint.  Media source =
must
schedule the splicing point which usually is the boundary of two
independently decodable frames. When audio and video are not =
interleaved,
each audio frame is usually decodable for common audio codecs. For =
video,
some predicted frames need to depent on other frames, so the splicing =
point
is usually at the Group of Pictures (GoP) boundary instead.

> One thing to worry about here is what happens if the frame=20
> rate varies between the content, then you may have a small=20
> diff to take care of.
> Yes, letting that be a bit extra delay between content in the=20
> point of switching is back is likely fine, but it needs to be=20
> discussed in the document.

Media encoding bitrates of the two contents may be different but the
difference should be minor, since both of them should match the user's =
media
comsuption rate to ensure that the user's buffer remains at a stable =
level
over time. If the encoding bitrate of insertion content is drastically
higher or lower than that of main content when splicing begins, user =
might
rapidly flood or drain its buffer, and eventually result in a buffer
overflow or underflow. So the insertion content is recommended to be =
encoded
at an appropriate bitrate to match main content's. If the insertion =
content
comes from another media source, the source may have knowledge about the
media encoding bitrate of main content in advance. How does it know that =
is
out of scope in this draft.

I will add above content in next version.

Thank you again!


Jinwei

>=20
> I might have made the impression that this issue is bigger=20
> than it really is.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20


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From: Jinwei Xia <xiajinwei@huawei.com>
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Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
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Hi Magnus=EF=BC=8C

Sorry for my late response.

> > I agree minimum-detectable is more appropriate. In=20
> > draft-xia-avtext-splicing-for-rtp-00, I mentioned the RTP header=20
> > parameters MUST remain consistent except that meida type SHOULD be=20
> > same since transcoding is prohibitively expensive and=20
> complex, so it =20
> > is RECOMMENDED to use same media type in insertion stream=20
> as that in=20
> > main stream. If an operator want to realize complete=20
> non-detectable on=20
> > client, he must take the tranacoding cost in account.
> >=20
> > Moreover, Splicer SHOULD also realize minimum-detectable in=20
> > application layer, for example, reusing existing PIDs after=20
> a splicing=20
> > described in [SMPTE-321M]. How this is done is an implementation=20
> > detail and not discussed in this draft.
> >=20
> > So how about above change?=20
> >=20
>=20
>=20
> I guess PIDs are protocol IDs. I think one needs to have a=20
> more detailed discussion on what IDs can be reused and when=20
> they can't.
>=20

No, PID is packet identifier, a 13-bit ID to to identify elementary =
stream of a program in MPEG2 transport stream.For each program, there is =
one unique Program Map Table (PMT), each program includes multiple =
streams (video, audio and metadata), each stream has its unique packet =
identifier (PID).=20

Since splicing changes the contents of a transport stream, and the PIDs =
(i.e., new PIDs in insertion stream), then the changes must be reflected =
in valid PMT packets. Splicer are responsible for
sending any changes to the PMT required to accommodate changes in the =
number of PIDs after a splicing.

However, in order to prevent commercial killing devices from taking =
advantage of changes in PMT, systems with splicing are encouraged to =
avoid changes to the PMT by reusing existing video and audio PIDs (i.e., =
PIDs in original stream) after a splicing.

> > an translator change the payload of RTP during splicing with SSRC=20
> > intact, should it use different keys for different segments?
>=20
> For the payload internal stuff people are using, it can=20
> likely handle such key changes using security payload=20
> internal mechanism. The whole point with these payload=20
> internal mechanisms is that no entity between the content=20
> source and the decoder actually needs to have the key.
>=20
> If one uses regular transport security like SRTP or DTLS, IPsec, etc.
> then you need to re-process all the packets if you change=20
> something in them. Thus different key-context for the=20
> different legs makes sense.

Right, we should narrow the security scope into transport security since =
the we are discussing splicing in media transport area.  The security =
considerations will give the SRTP based security contexts in future =
version.

> >>>
> >>> In non-detectable case, I personally incline to translator since=20
> >>> translator can totally eliminate the difference between the
> >> two stream
> >>> and erase the potential security issue.
> >>
> >> I think both can realize the non-detectable case.
> >>
> >=20
> > Yes, both can do that. The problem is which one is better.=20
> What's your=20
> > prefer in the two candidates or otherwise?
>=20
> I think it comes down to how one wants to organize the=20
> splicer and the upstream content sources. To which degree=20
> they should aware of the splicer and if requiring changes to=20
> them are acceptable, specifically the original content source.
>=20
> If modifying the original content source is out of the=20
> picture then I think a translator is the right way to go,=20
> although I believe it will require a bit more text to=20
> describe all the aspects one needs to think about.
>=20
> If one can modify the original content source, then I think a=20
> mixer is more straight forward and thus easier to describe.
>=20
> What the answer is to the modifying question I don't know.=20
> Input is desirable.
>=20

I think this is a key point, but feeling a little confusion about above =
analysis. Based on our previous discussion, whether translator or mixer =
can act as splicer needs to take two aspects into account when =
minimum-detectable requirement is supported. One aspect is bitrate =
adaption to handle varying network condition, another is overhead on =
splicer while splicing is not active. But neither seems to be related to =
original content source changing requirement, do you mean the original =
content source should reserve some gap in the content using some sort of =
RTP empty packets during the gap time, or otherwise?

Another confusing point is the mixer definition in RFC3550: "An =
intermediate system that receives RTP packets from one or more sources, =
possibly changes the data format, combines the packets in some manner =
and then forwards a new RTP packet." I am not sure whether combine =
implies a full mix, but a splicer does not combine packets in some =
manner, it merely decides which of two streams to forward.

> > Yes, this is inner implementation to fill up the gap. Using SSRC=20
> > multiplexing may be the most straight forward approach for=20
> detectable case.
> >=20
>=20
> Yes, but also least likely to ensure glitch free media=20
> processing, at least before you have updated your receivers=20
> to include a few mechanisms to ensure that they can quickly=20
> and correctly create a common timeline.
>=20
> Thus I think SSRC multiplexing is to quite a large degree a=20
> red herring in this case. It will not work well in legacy=20
> case, and even with updates it is still creating potential=20
> issues. Thus I think using a single SSRC and inserting the=20
> content into the stream is the most likely to work well.

Agree, let's simplify the case and focus on a single SSRC.


Thank you


Jinwei

>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20


From xiajinwei@huawei.com  Mon May 23 01:26:37 2011
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As an complement to my previous feedback.

> > 
> > I agree that you need to have content to insert that is 
> very close to 
> > the space you are replacing. The question is what happens 
> when things 
> > are off with a video frame or less, or an audio frame or less.
> > 
> 
> Your concern make senses, one of splicing requirements is 
> Splicer SHOULD NOT cause perceptible media clipping at the 
> splicing joint.  Media source must schedule the splicing 
> point which usually is the boundary of two independently 
> decodable frames. When audio and video are not interleaved, 
> each audio frame is usually decodable for common audio 
> codecs. For video, some predicted frames need to depent on 
> other frames, so the splicing point is usually at the Group 
> of Pictures (GoP) boundary instead.
> 

Meanwhile, the insertion media source or splicer must schedule that the
duration of insertion content  properly matches the gap time, i.e., starting
with an independently decodable frame and ending with another one.

Moreover, to reduce the playout latency, splicer may fill some specific
information (e.g., preamble information for rapid demultiplexing and
decoding insertion content) up the user's buffer with serval seconds earlier
before the presentation time of the insertion content.


Jinwei



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On 17 May 2011, at 08:18, Magnus Westerlund wrote:
> On 2011-05-16 12:47, Colin Perkins wrote:
>>>> In detectable case, the thing becomes simpler. translator is =
suitable to handle detectable splicing. When splicing ends, translator =
resumes main stream and rewrites the sequence number one higher than the =
last packet sent to receiver in the main stream. There is a little =
overhead on translator to rewrite RTP SN.=20
>>>=20
>>> What about the TS? I think that may become inconsistent and thus =
needs rewriting also.
>>=20
>> I'd think it reasonable to require that the inserted content MUST be =
the same duration as the original content. Otherwise, not only will you =
need to rewrite the timestamps after the splicing has ended, you'll =
either have a gap or extra data to be buffered somewhere. You'll only =
have to rewrite timestamps during the splicing, to normalise the random =
offset.
>=20
> I agree that you need to have content to insert that is very close to =
the space you are replacing. The question is what happens when things =
are off with a video frame or less, or an audio frame or less.
>=20
> One thing to worry about here is what happens if the frame rate varies =
between the content, then you may have a small diff to take care of. =
Yes, letting that be a bit extra delay between content in the point of =
switching is back is likely fine, but it needs to be discussed in the =
document.

Okay, this makes sense. I guess providing the inserted content is not =
longer than the gap, there's no real problem, and the draft just needs =
to mention the issue. If the inserted content might be longer, then =
things get a little more complex.

> I might have made the impression that this issue is bigger than it =
really is.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext



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




From csp@csperkins.org  Mon May 23 03:58:13 2011
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On 23 May 2011, at 09:26, Jinwei Xia wrote:
> As an complement to my previous feedback.
>=20
>>> I agree that you need to have content to insert that is very close =
to the space you are replacing. The question is what happens when things =
are off with a video frame or less, or an audio frame or less.
>>=20
>> Your concern make senses, one of splicing requirements is Splicer =
SHOULD NOT cause perceptible media clipping at the splicing joint.  =
Media source must schedule the splicing point which usually is the =
boundary of two independently decodable frames. When audio and video are =
not interleaved, each audio frame is usually decodable for common audio =
codecs. For video, some predicted frames need to depent on other frames, =
so the splicing point is usually at the Group of Pictures (GoP) boundary =
instead.
>=20
> Meanwhile, the insertion media source or splicer must schedule that =
the duration of insertion content properly matches the gap time, i.e., =
starting with an independently decodable frame and ending with another =
one.
>=20
> Moreover, to reduce the playout latency, splicer may fill some =
specific information (e.g., preamble information for rapid =
demultiplexing and decoding insertion content) up the user's buffer with =
serval seconds earlier before the presentation time of the insertion =
content.


That will need careful specification, since the requirements for =
rewriting the RTP headers in this case are tricky.

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




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Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
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On 23 May 2011, at 08:02, Jinwei Xia wrote:
> Hi Magnus=EF=BC=8C
>=20
> Sorry for my late response.
>=20
>>> I agree minimum-detectable is more appropriate. In =
draft-xia-avtext-splicing-for-rtp-00, I mentioned the RTP header =
parameters MUST remain consistent except that meida type SHOULD be same =
since transcoding is prohibitively expensive and complex, so it  is =
RECOMMENDED to use same media type in insertion stream as that in main =
stream. If an operator want to realize complete non-detectable on =
client, he must take the tranacoding cost in account.
>>>=20
>>> Moreover, Splicer SHOULD also realize minimum-detectable in =
application layer, for example, reusing existing PIDs after a splicing =
described in [SMPTE-321M]. How this is done is an implementation detail =
and not discussed in this draft.
>>>=20
>>> So how about above change?=20
>>=20
>> I guess PIDs are protocol IDs. I think one needs to have a more =
detailed discussion on what IDs can be reused and when they can't.
>=20
> No, PID is packet identifier, a 13-bit ID to to identify elementary =
stream of a program in MPEG2 transport stream.For each program, there is =
one unique Program Map Table (PMT), each program includes multiple =
streams (video, audio and metadata), each stream has its unique packet =
identifier (PID).=20
>=20
> Since splicing changes the contents of a transport stream, and the =
PIDs (i.e., new PIDs in insertion stream), then the changes must be =
reflected in valid PMT packets. Splicer are responsible for sending any =
changes to the PMT required to accommodate changes in the number of PIDs =
after a splicing.
>=20
> However, in order to prevent commercial killing devices from taking =
advantage of changes in PMT, systems with splicing are encouraged to =
avoid changes to the PMT by reusing existing video and audio PIDs (i.e., =
PIDs in original stream) after a splicing.

This sounds like something that should be specified by MPEG, not by =
IETF. Is there an MPEG document that this draft can reference to discuss =
how MPEG-specific payload data is manipulated?

>>> an translator change the payload of RTP during splicing with SSRC =
intact, should it use different keys for different segments?
>>=20
>> For the payload internal stuff people are using, it can likely handle =
such key changes using security payload internal mechanism. The whole =
point with these payload internal mechanisms is that no entity between =
the content source and the decoder actually needs to have the key.
>>=20
>> If one uses regular transport security like SRTP or DTLS, IPsec, etc. =
then you need to re-process all the packets if you change something in =
them. Thus different key-context for the different legs makes sense.
>=20
> Right, we should narrow the security scope into transport security =
since the we are discussing splicing in media transport area.  The =
security considerations will give the SRTP based security contexts in =
future version.

People are using payload internal security, so it needs to be mentioned =
in the draft, but (as this the PID stuff), it should probably be by =
reference to some other document.

>>>>> In non-detectable case, I personally incline to translator since =
translator can totally eliminate the difference between the two stream =
and erase the potential security issue.
>>>>=20
>>>> I think both can realize the non-detectable case.
>>>>=20
>>>=20
>>> Yes, both can do that. The problem is which one is better. What's =
your prefer in the two candidates or otherwise?
>>=20
>> I think it comes down to how one wants to organize the=20
>> splicer and the upstream content sources. To which degree=20
>> they should aware of the splicer and if requiring changes to=20
>> them are acceptable, specifically the original content source.
>>=20
>> If modifying the original content source is out of the=20
>> picture then I think a translator is the right way to go,=20
>> although I believe it will require a bit more text to=20
>> describe all the aspects one needs to think about.
>>=20
>> If one can modify the original content source, then I think a=20
>> mixer is more straight forward and thus easier to describe.
>>=20
>> What the answer is to the modifying question I don't know.=20
>> Input is desirable.
>>=20
>=20
> I think this is a key point, but feeling a little confusion about =
above analysis. Based on our previous discussion, whether translator or =
mixer can act as splicer needs to take two aspects into account when =
minimum-detectable requirement is supported. One aspect is bitrate =
adaption to handle varying network condition, another is overhead on =
splicer while splicing is not active. But neither seems to be related to =
original content source changing requirement, do you mean the original =
content source should reserve some gap in the content using some sort of =
RTP empty packets during the gap time, or otherwise?
>=20
> Another confusing point is the mixer definition in RFC3550: "An =
intermediate system that receives RTP packets from one or more sources, =
possibly changes the data format, combines the packets in some manner =
and then forwards a new RTP packet." I am not sure whether combine =
implies a full mix, but a splicer does not combine packets in some =
manner, it merely decides which of two streams to forward.
>=20
>>> Yes, this is inner implementation to fill up the gap. Using SSRC =
multiplexing may be the most straight forward approach for detectable =
case.
>>=20
>> Yes, but also least likely to ensure glitch free media processing, at =
least before you have updated your receivers to include a few mechanisms =
to ensure that they can quickly and correctly create a common timeline.
>>=20
>> Thus I think SSRC multiplexing is to quite a large degree a red =
herring in this case. It will not work well in legacy case, and even =
with updates it is still creating potential issues. Thus I think using a =
single SSRC and inserting the content into the stream is the most likely =
to work well.
>=20
> Agree, let's simplify the case and focus on a single SSRC.


I think the draft would be much clearer if it focussed on either =
non-detectable (e.g., using a translator) or detectable splicing, and =
didn't try to discuss both.

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




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Subject: Re: [avtext] Splicing: Confirmation of decisions at IETF#80
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Hi Colin,

> >> 
> >> I guess PIDs are protocol IDs. I think one needs to have a 
> more detailed discussion on what IDs can be reused and when 
> they can't.
> > 
> > No, PID is packet identifier, a 13-bit ID to to identify 
> elementary stream of a program in MPEG2 transport stream.For 
> each program, there is one unique Program Map Table (PMT), 
> each program includes multiple streams (video, audio and 
> metadata), each stream has its unique packet identifier (PID). 
> > 
> > Since splicing changes the contents of a transport stream, 
> and the PIDs (i.e., new PIDs in insertion stream), then the 
> changes must be reflected in valid PMT packets. Splicer are 
> responsible for sending any changes to the PMT required to 
> accommodate changes in the number of PIDs after a splicing.
> > 
> > However, in order to prevent commercial killing devices 
> from taking advantage of changes in PMT, systems with 
> splicing are encouraged to avoid changes to the PMT by 
> reusing existing video and audio PIDs (i.e., PIDs in original 
> stream) after a splicing.
> 
> This sounds like something that should be specified by MPEG, 
> not by IETF. Is there an MPEG document that this draft can 
> reference to discuss how MPEG-specific payload data is manipulated?

Right, above description is about MPEG splicing, I just interprets how
non-detectable splicing is done in application layer herein, but would not
include above description in draft. There are several MPEG-splicing
references in SCTE and SMPTE STDs.

> 
> >>> an translator change the payload of RTP during splicing 
> with SSRC intact, should it use different keys for different segments?
> >> 
> >> For the payload internal stuff people are using, it can 
> likely handle such key changes using security payload 
> internal mechanism. The whole point with these payload 
> internal mechanisms is that no entity between the content 
> source and the decoder actually needs to have the key.
> >> 
> >> If one uses regular transport security like SRTP or DTLS, 
> IPsec, etc. then you need to re-process all the packets if 
> you change something in them. Thus different key-context for 
> the different legs makes sense.
> > 
> > Right, we should narrow the security scope into transport 
> security since the we are discussing splicing in media 
> transport area.  The security considerations will give the 
> SRTP based security contexts in future version.
> 
> People are using payload internal security, so it needs to be 
> mentioned in the draft, but (as this the PID stuff), it 
> should probably be by reference to some other document.
> 

Accept

> >> Thus I think SSRC multiplexing is to quite a large degree 
> a red herring in this case. It will not work well in legacy 
> case, and even with updates it is still creating potential 
> issues. Thus I think using a single SSRC and inserting the 
> content into the stream is the most likely to work well.
> > 
> > Agree, let's simplify the case and focus on a single SSRC.
> 
> 
> I think the draft would be much clearer if it focussed on 
> either non-detectable (e.g., using a translator) or 
> detectable splicing, and didn't try to discuss both.
> 

Both cases can be discussed in the maillist with wide scope, but if there
should be only one solution in a single draft, I prefer to address
minimum-detectable case. 

Thank you.


Jinwei

> --
> Colin Perkins
> http://csperkins.org/
> 
> 
> 


From internet-drafts@ietf.org  Thu May 26 09:37:01 2011
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Subject: [avtext] I-D Action: draft-ietf-avtext-multicast-acq-rtcp-xr-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Extensions Work=
ing Group of the IETF.

	Title           : Multicast Acquisition Report Block Type for RTP Control =
Protocol (RTCP) Extended Reports (XRs)
	Author(s)       : Ali Begen
                          Eric Friedrich
	Filename        : draft-ietf-avtext-multicast-acq-rtcp-xr-05.txt
	Pages           : 22
	Date            : 2011-05-26

   In most RTP-based multicast applications, the RTP source sends inter-
   related data.  Due to this interdependency, randomly joining RTP
   receivers usually cannot start consuming the multicast data right
   after they join the session.  Thus, they often experience a random
   acquisition delay.  An RTP receiver can use one or more different
   approaches to achieve rapid acquisition.  Yet, due to various
   factors, performance of the rapid acquisition methods usually varies.
   Furthermore, in some cases the RTP receiver can do a simple multicast
   join (in other cases it is compelled to do so).  For quality
   reporting, monitoring and diagnostics purposes, it is important to
   collect detailed information from the RTP receivers about their
   acquisition and presentation experiences.  This document addresses
   this issue by defining a new report block type, called Multicast
   Acquisition (MA) Report Block, within the framework of RTP Control
   Protocol (RTCP) Extended Reports (XR) (RFC 3611).  This document also
   defines the necessary signaling of the new MA report block type in
   the Session Description Protocol (SDP).


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-multicast-acq-rtcp-xr=
-05.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtext-multicast-acq-rtcp-xr-=
05.txt

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Subject: Re: [avtext] Call for adopting document for mile stone "Support for multiple clock rates in an RTP session for Proposed Standard"
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On 16 May 2011, at 12:41, Magnus Westerlund wrote:
> AVTEXT has one milestone:
>=20
> Dec 2011 Submit Support for multiple clock rates in an RTP session
>         for Proposed Standard
>=20
> We have one candidate document:
>=20
> =
https://datatracker.ietf.org/doc/draft-petithuguenin-avtext-multiple-clock=
-rates/
>=20
> This is the call to the WG if we want to adopt this document as the
> basis for meeting that milestone? Please provide comments even if it =
is
> just "Yes".

I've read this, and think that it's useful, and should be adopted.=20

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




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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Extensions Work=
ing Group of the IETF.

	Title           : Multicast Acquisition Report Block Type for RTP Control =
Protocol (RTCP) Extended Reports (XRs)
	Author(s)       : Ali Begen
                          Eric Friedrich
	Filename        : draft-ietf-avtext-multicast-acq-rtcp-xr-06.txt
	Pages           : 21
	Date            : 2011-05-29

   In most RTP-based multicast applications, the RTP source sends inter-
   related data.  Due to this interdependency, randomly joining RTP
   receivers usually cannot start consuming the multicast data right
   after they join the session.  Thus, they often experience a random
   acquisition delay.  An RTP receiver can use one or more different
   approaches to achieve rapid acquisition.  Yet, due to various
   factors, performance of the rapid acquisition methods usually varies.
   Furthermore, in some cases the RTP receiver can do a simple multicast
   join (in other cases it is compelled to do so).  For quality
   reporting, monitoring and diagnostics purposes, it is important to
   collect detailed information from the RTP receivers about their
   acquisition and presentation experiences.  This document addresses
   this issue by defining a new report block type, called Multicast
   Acquisition (MA) Report Block, within the framework of RTP Control
   Protocol (RTCP) Extended Reports (XR) (RFC 3611).  This document also
   defines the necessary signaling of the new MA report block type in
   the Session Description Protocol (SDP).


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-multicast-acq-rtcp-xr=
-06.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtext-multicast-acq-rtcp-xr-=
06.txt

From magnus.westerlund@ericsson.com  Tue May 31 01:23:56 2011
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [avtext] Call for adopting document for mile stone "Support for multiple clock rates in an RTP session for Proposed Standard"
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As Individual I would like to support this adoption.

Magnus

On 2011-05-16 13:41, Magnus Westerlund wrote:
> WG,
> 
> AVTEXT has one milestone:
> 
> Dec 2011 Submit Support for multiple clock rates in an RTP session
>          for Proposed Standard
> 
> We have one candidate document:
> 
> https://datatracker.ietf.org/doc/draft-petithuguenin-avtext-multiple-clock-rates/
> 
> This is the call to the WG if we want to adopt this document as the
> basis for meeting that milestone? Please provide comments even if it is
> just "Yes".
> 
> If you think there is some aspect of the issue or the solution that
> hasn't been discussed and you want to ensure is covered, please comment
> on that.
> 
> The WG chairs will make a decision on 31st of May based on your comments.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
> 


-- 

Magnus Westerlund

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


From musgravepj@gmail.com  Tue May 31 04:03:01 2011
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From: Peter Musgrave <musgravepj@gmail.com>
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Subject: Re: [avtext] Call for adopting document for mile stone "Support for multiple clock rates in an RTP session for Proposed Standard"
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--20cf3054a8c1d21bbe04a49058f4
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+1

Peter Musgrave

On Tue, May 31, 2011 at 4:23 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> As Individual I would like to support this adoption.
>
> Magnus
>
> On 2011-05-16 13:41, Magnus Westerlund wrote:
> > WG,
> >
> > AVTEXT has one milestone:
> >
> > Dec 2011 Submit Support for multiple clock rates in an RTP session
> >          for Proposed Standard
> >
> > We have one candidate document:
> >
> >
> https://datatracker.ietf.org/doc/draft-petithuguenin-avtext-multiple-cloc=
k-rates/
> >
> > This is the call to the WG if we want to adopt this document as the
> > basis for meeting that milestone? Please provide comments even if it is
> > just "Yes".
> >
> > If you think there is some aspect of the issue or the solution that
> > hasn't been discussed and you want to ensure is covered, please comment
> > on that.
> >
> > The WG chairs will make a decision on 31st of May based on your comment=
s.
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > avtext mailing list
> > avtext@ietf.org
> > https://www.ietf.org/mailman/listinfo/avtext
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>

--20cf3054a8c1d21bbe04a49058f4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

+1=A0<div><br></div><div>Peter Musgrave<br><br><div class=3D"gmail_quote">O=
n Tue, May 31, 2011 at 4:23 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a =
href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.c=
om</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">As Individual I would like to support this =
adoption.<br>
<br>
Magnus<br>
<br>
On 2011-05-16 13:41, Magnus Westerlund wrote:<br>
&gt; WG,<br>
<div class=3D"im">&gt;<br>
&gt; AVTEXT has one milestone:<br>
&gt;<br>
&gt; Dec 2011 Submit Support for multiple clock rates in an RTP session<br>
&gt; =A0 =A0 =A0 =A0 =A0for Proposed Standard<br>
&gt;<br>
&gt; We have one candidate document:<br>
&gt;<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-petithuguenin-avtext=
-multiple-clock-rates/" target=3D"_blank">https://datatracker.ietf.org/doc/=
draft-petithuguenin-avtext-multiple-clock-rates/</a><br>
&gt;<br>
&gt; This is the call to the WG if we want to adopt this document as the<br=
>
&gt; basis for meeting that milestone? Please provide comments even if it i=
s<br>
&gt; just &quot;Yes&quot;.<br>
&gt;<br>
</div>&gt; If you think there is some aspect of the issue or the solution t=
hat<br>
&gt; hasn&#39;t been discussed and you want to ensure is covered, please co=
mment<br>
&gt; on that.<br>
&gt;<br>
&gt; The WG chairs will make a decision on 31st of May based on your commen=
ts.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus Westerlund<br>
&gt;<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%=
2B46%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel=
:%2B46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlu=
nd@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt; ----------------------------------------------------------------------=
<br>
<div class=3D"im">&gt;<br>
&gt; _______________________________________________<br>
&gt; avtext mailing list<br>
&gt; <a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/avtext" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/avtext</a><br>
&gt;<br>
<br>
<br>
</div>--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<div><div></div><div class=3D"h5"><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" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/avtext</a><br>
</div></div></blockquote></div><br></div>

--20cf3054a8c1d21bbe04a49058f4--

From abegen@cisco.com  Tue May 31 07:42:09 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <avtext@ietf.org>
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I will repeat my earlier support here again for adopting this.

-acbegen

> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
> Sent: Tuesday, May 31, 2011 4:24 AM
> To: avtext@ietf.org
> Subject: Re: [avtext] Call for adopting document for mile stone =
"Support for multiple clock rates in an RTP session for
> Proposed Standard"
>=20
> As Individual I would like to support this adoption.
>=20
> Magnus
>=20
> On 2011-05-16 13:41, Magnus Westerlund wrote:
> > WG,
> >
> > AVTEXT has one milestone:
> >
> > Dec 2011 Submit Support for multiple clock rates in an RTP session
> >          for Proposed Standard
> >
> > We have one candidate document:
> >
> > =
https://datatracker.ietf.org/doc/draft-petithuguenin-avtext-multiple-cloc=
k-rates/
> >
> > This is the call to the WG if we want to adopt this document as the
> > basis for meeting that milestone? Please provide comments even if it =
is
> > just "Yes".
> >
> > If you think there is some aspect of the issue or the solution that
> > hasn't been discussed and you want to ensure is covered, please =
comment
> > on that.
> >
> > The WG chairs will make a decision on 31st of May based on your =
comments.
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > =
----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > =
----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > =
----------------------------------------------------------------------
> >
> > _______________________________________________
> > avtext mailing list
> > avtext@ietf.org
> > https://www.ietf.org/mailman/listinfo/avtext
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

From ray.vanbrandenburg@tno.nl  Tue May 31 07:53:51 2011
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: "avtext@ietf.org" <avtext@ietf.org>
Thread-Topic: [avtext] Call for adopting document for mile stone "Support for multiple clock rates in an RTP session for Proposed Standard"
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Date: Tue, 31 May 2011 14:53:44 +0000
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Subject: Re: [avtext] Call for adopting document for mile stone "Support for multiple clock rates in an RTP session for Proposed Standard"
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I support this document and am willing to review future versions.=20

Ray van Brandenburg


> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On=20
> Behalf Of Magnus Westerlund
> Sent: Tuesday, May 31, 2011 4:24 AM
> To: avtext@ietf.org
> Subject: Re: [avtext] Call for adopting document for mile stone=20
> "Support for multiple clock rates in an RTP session for Proposed Standard"
>=20
> As Individual I would like to support this adoption.
>=20
> Magnus
>=20
> On 2011-05-16 13:41, Magnus Westerlund wrote:
> > WG,
> >
> > AVTEXT has one milestone:
> >
> > Dec 2011 Submit Support for multiple clock rates in an RTP session
> >          for Proposed Standard
> >
> > We have one candidate document:
> >
> > https://datatracker.ietf.org/doc/draft-petithuguenin-avtext-multiple
> > -clock-rates/
> >
> > This is the call to the WG if we want to adopt this document as the=20
> > basis for meeting that milestone? Please provide comments even if it=20
> > is just "Yes".
> >
> > If you think there is some aspect of the issue or the solution that=20
> > hasn't been discussed and you want to ensure is covered, please=20
> > comment on that.
> >
> > The WG chairs will make a decision on 31st of May based on your comment=
s.
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > --------------------------------------------------------------------
> > -- Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > --------------------------------------------------------------------
> > --
> >
> > _______________________________________________
> > avtext mailing list
> > avtext@ietf.org
> > https://www.ietf.org/mailman/listinfo/avtext
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
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