
From Even.roni@huawei.com  Mon Jul  4 11:32:35 2011
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Date: Mon, 04 Jul 2011 21:30:14 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <EDC0A1AE77C57744B664A310A0B23AE21FB7801B@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com>
To: "'DRAGE, Keith (Keith)'" <keith.drage@alcatel-lucent.com>, avtext@ietf.org
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Subject: Re: [avtext] Consensus call on RTP Splicing direction forward
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Hi,
I think that Dave and Colin who had interest in this work both said that =
if
non detectability is not an issue than the mixer approach make sense, =
the
CSRC can help with monitoring while on the other side the splicer can =
make
it harder to detect by not sending the CSRC.
I think that it will be good to update the current draft with the
requirements based on the discussion and do the mixer as first step.

We can later look if and how to address the best way to support a good
enough non detectable case.

Roni

> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On
> Behalf Of DRAGE, Keith (Keith)
> Sent: Thursday, June 23, 2011 12:25 PM
> To: avtext@ietf.org
> Subject: Re: [avtext] Consensus call on RTP Splicing direction forward
>=20
> [As WG chair]
>=20
> Magnus sent out this call a couple of weeks back and we haven't seen
> very many responses.
>=20
> If you have not already responded, please do so.
>=20
> Regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On
> Behalf
> > Of Magnus Westerlund
> > Sent: 10 June 2011 17:00
> > To: avtext@ietf.org
> > Subject: [avtext] Consensus call on RTP Splicing direction forward
> >
> > Hi,
> >
> > We have had this ongoing discussion on the mailing list around the
> RTP
> > Splicing and the pro-and-cons for two major solutions. To try to get
> to
> > some decision on what the WG should focus on I would like to ask the
> WG
> > to indicate their preference for what should be done here. The
> options
> > are three on a high level. And this is for an informational document
> > that explains how you use the existing RTP features to accomplish
> > splicing.
> >
> > A) Focus only on RTP Translator based solution
> >
> > B) Focus only on RTP Mixer based solution
> >
> > C) Define both Mixer and Translator based solutions.
> >
> > A) is pretty well described in the individual draft:
> > https://datatracker.ietf.org/doc/draft-xia-avtext-splicing-for-rtp/
> >
> > The trade-offs and limiations of A and B has been fairly well
> discussed
> > in the following email list discussion.
> > http://www.ietf.org/mail-archive/web/avtext/current/msg00046.html
> >
> > So if you aren't familiar with the topic already please review the
> above
> > and then indicate your position no later than the 26th of June.
> >
> > Independently of the choice that solution will have to discuss the
> pros
> > and cons of that method and what properties it has.
> >
> > 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
> > =
---------------------------------------------------------------------
> -
> >
> > _______________________________________________
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> > avtext@ietf.org
> > https://www.ietf.org/mailman/listinfo/avtext
> _______________________________________________
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From internet-drafts@ietf.org  Mon Jul  4 15:44:36 2011
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Subject: [avtext] I-D Action: draft-ietf-avtext-mixer-to-client-audio-level-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport 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-03.txt
	Pages           : 16
	Date            : 2011-07-04

   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-03.txt

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This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avtext-mixer-to-client-audio-=
level-03.txt

From internet-drafts@ietf.org  Mon Jul  4 15:44:37 2011
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Subject: [avtext] I-D Action: draft-ietf-avtext-client-to-mixer-audio-level-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Extensions Work=
ing Group of the IETF.

	Title           : A Real-Time Transport Protocol (RTP) Header Extension fo=
r Client-to- Mixer Audio Level Indication
	Author(s)       : Jonathan Lennox
                          Emil Ivov
                          Enrico Marocco
	Filename        : draft-ietf-avtext-client-to-mixer-audio-level-03.txt
	Pages           : 11
	Date            : 2011-07-04

   This document defines a mechanism by which packets of Real-Time
   Transport Protocol (RTP) audio streams can indicate, in an RTP header
   extension, the audio level of the audio sample carried in the RTP
   packet.  In large conferences, this can reduce the load on an audio
   mixer or other middlebox which wants to forward only a few of the
   loudest audio streams, without requiring it to decode and measure
   every stream that is received.


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http://www.ietf.org/internet-drafts/draft-ietf-avtext-client-to-mixer-audio=
-level-03.txt

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This Internet-Draft can be retrieved at:
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level-03.txt

From emil@sip-communicator.org  Mon Jul  4 15:47:47 2011
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Cc: "avtext@ietf.org" <avtext@ietf.org>, draft-ietf-avtext-mixer-to-client-audio-level@tools.ietf.org
Subject: Re: [avtext] Comments on	draft-ietf-avtext-mixer-to-client-audio-level-02
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Hey Magnus, Keith, all,

We have just submitted new versions of the level drafts that we hope
would address most of the comments expressed on the list.

Diff2 available here:
http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtext-mixer-to-client-audio-level-03.txt
http://tools.ietf.org/rfcdiff?url2=draft-ietf-avtext-client-to-mixer-audio-level-03.txt


Comments inline:

Magnus Westerlund commented on mixer-to-client:
> Hi,
> 
> I reviewed the draft and have some comments.

Thanks!

> 1. The intended status says Informational. This document is intended for
> standards track.

Fixed.

> 2. Section 3, paqe 4:
> "   Note that in some cases a mixer may be sending an RTP audio stream
>    that only contains audio level information and no actual audio.
>    Updating a (web) interface conference module may be one reason for
>    this to happen."
> 
> My question would what value you stick into the payload type field if
> you have no payload only header? It is not obvious that you can do this
> without configuring a payload format that either is valid for a zero
> length payload or define a specific for that case.

Agreed. Removed.

> 3. Section 4 and 5. I find the definition confusing, especially as
> important rules of the actual format comes in later sections.
>  a) First of all it is not written in generic engough way to support
> both the short and long format.
>  b) The actual extension formats that is the payload is in the next
> section. Including important rules, like that the CSRC list shall be
> mapped against the level list.
> 
> I would recommend that Section 4 and 5 is merged and that a more ordered
> structure to the definition of the header extensions is made. Where
> field values are clearly defined including explicit length in bits. And
> where the rules around it is separated from the basic field definitions.

OK, we've merged sections 4 and 5 into a single "Audio Levels" section.
We've described use of the extension with both the one-byte and the
two-byte header formats and added example for both of them.

We've also modified the corresponding section in the client-to-mixer
draft so that both are now almost identical, which is also what Keith
advised in an earlier exchange.

> 4. Section 5. I think the dBov definition needs to be really clear. Yes,
> the convention is amplitudes are squared when doing dB calculation. But
> it is not clear if this is done for the single highest value in the
> measured range, or if it is done as an average over the measured range.
> Please provide a definition or a reference.

Well, we've mostly been using it as an average. A root mean square to be
precise. We've modified one of the paragraphs to read as follows:

    The audio level header extension only carries the level of the
    audio in the RTP payload of the packet it is associated with, with
    no long-term averaging or smoothing applied. That level is measured
    as a root mean square of all the samples in the measured range.

We hope that this text, combined with the reference implementation
should clear all doubts. However, to be perfectly honest, I am not
completely convinced that we should specify this. Maybe we could have
this as RECOMMENDED leaving the door open to alternative measurement
mechanisms?

> 5. Section 5 and Section 3:
> 
> I would note that the folllowing text in section 5:
>    The audio level header extension only carries the level of the audio
>    in the RTP payload of the packet it is associated with, with no long-
>    term averaging or smoothing applied.
> 
> Makes it impossible for the following to be done:
> Note that in some cases a mixer may be sending an RTP audio stream
>    that only contains audio level information and no actual audio.
>    Updating a (web) interface conference module may be one reason for
>    this to happen.
> 
> This as there is no duration of audio indicated, only a first sample.

Agreed. Text already removed as per comment 2.

> 6. Section 5:
>   "The audio level for digital silence, for example for a muted audio
>    source, MAY be represented as 127 (-127 dBov), regardless of the
>    dynamic range of the encoded audio format."
> 
> I don't like this MAY. To me it appears to only complicate
> implementations. Either you mandate the behavior so that one can trust
> sources muted by the application to always carry value 127. Or one
> anyway need to have noise level detector so that one can determine when
> there is signal above the noise level. One may have to have that anyway,
> but I think we should consider if we want clear MUTED indication here or
> not.

You are right. Mandating it would probably be best. Changed to MUST.

> 7. Section 6.
> "Presence of the above attribute in the SDP description of a media
>    stream indicates that some or all RTP packets in that stream would
>    contain the audio level information RTP extension header."
> 
> To me this sentence is a mix between a usage rule:
> "some or all packets may contain the header extension" and a signaling
> rule: The presence of the attribute indicates the usage of the audio
> level header extension.
> 
> In addition it not is the attribute per say that indicate it, it is the
> attribute with the given URI is used to indicate that a particular value
> is bound to identify the header extension defined in this doc.

OK. Changed to:

    Presence of the above attribute in the SDP description of a media
    stream indicates that RTP packets in that stream, which contain the
    level extension defined in this document, will be carrying them
    with an ID of 7.

If this doesn't work then we can simply remove it. This is, after all,
only comfort text meant to assist implementors. Rules of use and
signalling are inherited from 5285 anyway.

> 8. Section 6.
>   "This speicification does not define use of the audio level extensions
>    in video streams.  Therefore, the extension defined in this document
>    SHOULD NOT be advertised in anything but audio streams."
> 
> First spelling error of specification.

Fixed.

> Secondly, we also have media types like text, where I also don't see a
> logical mapping. Thus I wonder if this should be formulated in a way
> that avoids being media specific. I also wonder if there is any point in
> leaving the hole with "SHOULD NOT"? Either skip using that and formulate
> that it is only defined for audio streams, or some other way?

OK, so how about:
    This specification only defines use of the audio level
    extensions in audio streams. They MUST NOT be advertised with
    other media types such as video or text for example.

> 9. Section 7:
> I am missing a reference to the VBR consideration that I think contains
> a much better and complete explanation to the issue with just energy
> level or similar phonetic structure indicators.

Added.

Magnus Westerlund commented on mixer-to-client:
> 1. Section 3.
> "The two-byte header defined in RFC 5285 [RFC5285] may also be used."
> 
> I think that it should be stated up front that header extensions per
> definitions can be used with either of the header formats. In addition I
> think one needs to separate a bit clearer what is framework and what is
> the new parts.

Addressed as with the mixer-to-client draft.

> 2. Section 3. I think the dBov definition needs to be really clear. Yes,
> the convention is amplitudes are squared when doing dB calculation. But
> it is not clear if this is done for the single highest value in the
> measured range, or if it is done as an average over the measured range.
> Please provide a definition or a reference.

Addressed as with the mixer-to-client draft.

> 3. Section 5:
>   "The audio level for digital silence, for example for a muted audio
>    source, MAY be represented as 127 (-127 dBov), regardless of the
>    dynamic range of the encoded audio format."
> 
> I don't like this MAY. To me it appears to only complicate
> implementations. Either you mandate the behavior so that one can trust
> sources muted by the application to always carry value 127. Or one
> anyway need to have noise level detector so that one can determine when
> there is signal above the noise level. One may have to have that anyway,
> but I think we should consider if we want clear MUTED indication here or
> not.

Mandated as with the mixer-to-client draft.

> 4. Section 3:
>    In addition, a flag bit (labeled V) indicates whether the encoder
>    believes the audio packet contains voice activity (1) or does not
>    (0).  The voice activity detection algorithm is unspecified and left
>    implementation-specific.
> 
> Would it be of interest to have an signalling parameter for an
> implementation to indicate its algorithm?

My personal opinion is that, if this is deemed necessary, it can
probably be handled in a separate draft at a later stage. This would
also allow some level of negotiation, again, if people find this
necessary. Any other thoughts?

> 5. Section 8.2:
> I think both the encrypted header extension and the VBR audio is likely
> normative references as they either is recommend solution or a
> description of the issue.

I don't know. Header encryption is only one way of handling this.
There's also DTLS, IPsec and even VPN. So in that respect it seems to me
that this method is best seen here as an example.


Keith commented on mixer-to-client
> Section 6.
> 
>    This speicification does not define use of the audio level extensions
>    in video streams.  Therefore, the extension defined in this document
>    SHOULD NOT be advertised in anything but audio streams.
> 
> Typo: "specification"

Fixed.

Keith commented on client-to-mixer
> Section 3.
> 
>    Implementations MAY choose to measure audio levels prior to encoding
>    them in the payload carried in the RTP payload, e.g. on raw linear
>    PCM input.
> 
> "MAY" defines an option. Is this really defining an option or just indicating a possibility, as in "can".

Also raised by Magnus. Reply above.

Cheers,
Emil

From xiajinwei@huawei.com  Mon Jul  4 18:34:31 2011
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Hi,

I am also eager to know the direction of next step work.

Thank your understanding.


Jinwei

> -----Original Message-----
> From: avtext-bounces@ietf.org=20
> [mailto:avtext-bounces@ietf.org] On Behalf Of Roni Even
> Sent: Tuesday, July 05, 2011 2:30 AM
> To: 'DRAGE, Keith (Keith)'; avtext@ietf.org
> Subject: Re: [avtext] Consensus call on RTP Splicing direction forward
>=20
> Hi,
> I think that Dave and Colin who had interest in this work=20
> both said that if non detectability is not an issue than the=20
> mixer approach make sense, the CSRC can help with monitoring=20
> while on the other side the splicer can make it harder to=20
> detect by not sending the CSRC.
> I think that it will be good to update the current draft with=20
> the requirements based on the discussion and do the mixer as=20
> first step.
>=20
> We can later look if and how to address the best way to=20
> support a good enough non detectable case.
>=20
> Roni
>=20
> > -----Original Message-----
> > From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On=20
> > Behalf Of DRAGE, Keith (Keith)
> > Sent: Thursday, June 23, 2011 12:25 PM
> > To: avtext@ietf.org
> > Subject: Re: [avtext] Consensus call on RTP Splicing=20
> direction forward
> >=20
> > [As WG chair]
> >=20
> > Magnus sent out this call a couple of weeks back and we=20
> haven't seen=20
> > very many responses.
> >=20
> > If you have not already responded, please do so.
> >=20
> > Regards
> >=20
> > Keith
> >=20
> > > -----Original Message-----
> > > From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On
> > Behalf
> > > Of Magnus Westerlund
> > > Sent: 10 June 2011 17:00
> > > To: avtext@ietf.org
> > > Subject: [avtext] Consensus call on RTP Splicing direction forward
> > >
> > > Hi,
> > >
> > > We have had this ongoing discussion on the mailing list around the
> > RTP
> > > Splicing and the pro-and-cons for two major solutions. To=20
> try to get
> > to
> > > some decision on what the WG should focus on I would like=20
> to ask the
> > WG
> > > to indicate their preference for what should be done here. The
> > options
> > > are three on a high level. And this is for an=20
> informational document=20
> > > that explains how you use the existing RTP features to accomplish=20
> > > splicing.
> > >
> > > A) Focus only on RTP Translator based solution
> > >
> > > B) Focus only on RTP Mixer based solution
> > >
> > > C) Define both Mixer and Translator based solutions.
> > >
> > > A) is pretty well described in the individual draft:
> > >=20
> https://datatracker.ietf.org/doc/draft-xia-avtext-splicing-for-rtp/
> > >
> > > The trade-offs and limiations of A and B has been fairly well
> > discussed
> > > in the following email list discussion.
> > > http://www.ietf.org/mail-archive/web/avtext/current/msg00046.html
> > >
> > > So if you aren't familiar with the topic already please review the
> > above
> > > and then indicate your position no later than the 26th of June.
> > >
> > > Independently of the choice that solution will have to discuss the
> > pros
> > > and cons of that method and what properties it has.
> > >
> > > Cheers
> > >
> > > Magnus Westerlund
> > >
> > >=20
> --------------------------------------------------------------------
> > > -
> > -
> > > Multimedia Technologies, Ericsson Research EAB/TVM
> > >=20
> --------------------------------------------------------------------
> > > -
> > -
> > > Ericsson AB                | Phone  +46 10 7148287
> > > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > > SE-164 80 Stockholm, Sweden| mailto:=20
> magnus.westerlund@ericsson.com
> > >=20
> --------------------------------------------------------------------
> > > -
> > -
> > >
> > > _______________________________________________
> > > avtext mailing list
> > > avtext@ietf.org
> > > https://www.ietf.org/mailman/listinfo/avtext
> > _______________________________________________
> > avtext mailing list
> > avtext@ietf.org
> > https://www.ietf.org/mailman/listinfo/avtext
>=20
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext


From keith.drage@alcatel-lucent.com  Wed Jul  6 07:23:57 2011
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(As WG cochair)

This is to announce a Working Group Last Call on=20

draft-ietf-avtext-mixer-to-client-audio-level-03

(Do note this is two parallel calls on two related drafts)

Please review and supply comments by Wednesday 20th July 2011.

It is helpful if you make some attempt to categorise uour comments, e.g. Ma=
jor, Minor technical, Editorial.

If you review the document and find no issue with the current version, then=
 it is helpful to send a mail to the WG chairs saying so as this helps us a=
ssess the level of review.

Regards

Keith

P.S. The intent is to allow any identified show stoppers to be discussed at=
 the face to face in Quebec, so the earlier such comments are made the bett=
er.

From keith.drage@alcatel-lucent.com  Wed Jul  6 07:24:05 2011
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(As WG cochair)

This is to announce a Working Group Last Call on=20

draft-ietf-avtext-client-to-mixer-audio-level-03

(Do note this is two parallel calls on two related drafts)

Please review and supply comments by Wednesday 20th July 2011.

It is helpful if you make some attempt to categorise uour comments, e.g. Ma=
jor, Minor technical, Editorial.

If you review the document and find no issue with the current version, then=
 it is helpful to send a mail to the WG chairs saying so as this helps us a=
ssess the level of review.

Regards

Keith

P.S. The intent is to allow any identified show stoppers to be discussed at=
 the face to face in Quebec, so the earlier such comments are made the bett=
er.

From internet-drafts@ietf.org  Mon Jul 11 13:42:10 2011
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Subject: [avtext] I-D Action: draft-ietf-avtext-multiple-clock-rates-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Extensions Work=
ing Group of the IETF.

	Title           : Support for multiple clock rates in an RTP session
	Author(s)       : Marc Petit-Huguenin
	Filename        : draft-ietf-avtext-multiple-clock-rates-01.txt
	Pages           : 10
	Date            : 2011-07-11

   This document clarifies the RTP specification when different clock
   rates are used in an RTP session.  It also provides guidance on how
   to interoperate with legacy RTP implementations that use multiple
   clock rates.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avtext-multiple-clock-rates-=
01.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-multiple-clock-rates-0=
1.txt

From petithug@acm.org  Mon Jul 11 13:46:50 2011
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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [avtext] Comments on draft-ietf-avtext-multiple-clock-rates-00.txt
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I just released a new version of this draft that incorporate the two first
points below.  Unfortunately, and I apologize for that, I did not had the time
to work on the two remaining points.  I will have more time to work on that
before Taipei and we should still be on time for the submission in December as
per the charter.

On 06/13/2011 06:07 AM, Magnus Westerlund wrote:
> Hi,
> 
> I have reviewed this WG 00 version, and thanks the other for taking my
> comments into consideration in the previous version.
> 
> I have some additional comments and suggestions for improvements.
> However I think it is important that we get input from the WG on these
> questions.
> 
> The below are my views as an individual, not WG chair.
> 
> 1. Section 1:
> "This document first tries to list in Section 2 and subsections all
>    the various algorithms used by existing RTP implementations."
> 
> Wouldn't it better to state that this section "lists all known
> algorithms used in RTP implementations of the time of writing." Or
> something like this. To avoid using "all" without a qualifier.
> 
> 2. Section 4.1:
>> Each time the clock rate changes, the start_offset and capture_start
>>    values are calculated with the following formulas:
>>
>>    start_offset = (capture_time - capture_start) * previous_clock_rate
>>    capture_start = capture_time
> 
> I think it is important to elaborate on what "capture_time" is in this
> formula. That it is the capture time of the first instant the new
> timestamp rate is going to be used.
> 
> 3. Section 4.1:
> 
> I am still not quite happy about these recommendations:
> 
>    An RTP Sender with RTCP turned on MUST use a different SSRC for each
>    different clock rate.
> 
> Because in more advanced applications like video conferencing with
> centralized mixing the impact of switching SSRC is so big. It can
> involved the following mechanisms that will be effected:
> 
> - Any connection to application logic, side boards etc.
> - The issue that who a SSRC becomes unknown until an CNAME is
> distributed to everyone that needs to know, that include speaker
> indication using CSRC.
> - RTP retransmission, that needs to use another SSRC and then acquire
> the binding using CNAME is SSRC multiplexed.
> - The fact that switching and "BYE" an SSRC prevents an receiver to
> request retransmission of the packet that has been lost but not yet
> delivered when the SSRC is given up.
> - TMMBR limiations set on a SSRC value that needs to be transfered to
> the new one.
> 
> So I think "MUST" is way to strict as rule. Especially as there are
> rules that make this problem much less an issue as soon as
> implementations has been updated. But that will be a bit more work as we
> should also include clarifications and fixes on the found issues around
> RTCP clock rate switches. And recommendations for how to avoid the issue
> in the future.
> 
> 4. Section 4.2:
> 
>  D(i,j) = (arrival_time_j * clock_rate_i - timestamp_j) -
>    (arrival_time_i * clock_rate_i - timestamp_i)
> 
> 
> I think it is important that this is expanded a bit.
> 
> First of all that we will only avoid an error if "j" is the first packet
> for the new Timestamp Rate. Thus we should likely discuss how one gets
> this value to be a sane one in any RTCP report. Likely by doing the
> calculation twice, one up to the switch and then a second time from the
> switch forward.
> 
> If one doesn't get the packet which is first of the switch, then I think
> we should consider if we need to discuss how one use the two nearest SRs
> to calculate the actual point of the switch.
> 
> If my math hasn't gone wrong I think
> 
> NTP_of_Switch = (TS_j-TS_i+ R_i*N_i - R_j*N_j)/(R_i-R_j)
> 
> Where i is an RTCP SR prior to the switch and j is an RTCP SR after the
> switch and TS is the timestamp value, N the NTP value and R the
> timestamp rate at that moment.
> 
> What I don't know is how robust this is to a non-consistent method of
> timestamp value switch.

- -- 
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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Subject: Re: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03
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On 07/06/2011 09:22 AM, DRAGE, Keith (Keith) wrote:
> (As WG cochair)
>
> This is to announce a Working Group Last Call on
>
> draft-ietf-avtext-client-to-mixer-audio-level-03
>
> (Do note this is two parallel calls on two related drafts)
>
> Please review and supply comments by Wednesday 20th July 2011.
>
> It is helpful if you make some attempt to categorise uour comments, e.g. Major, Minor technical, Editorial.
>
> If you review the document and find no issue with the current version, then it is helpful to send a mail to the WG chairs saying so as this helps us assess the level of review.

Editorial:

Paragraph 2 of Section 3 - "using the one or the two-byte header" seems 
to be a bit hard to parse. Maybe "using either the one-byte or the 
two-byte header format" would be better?

Last paragraph of Section 3 - 'noice' should be 'noise'

Minor technical:

Should there be any text included to indicate that a mixer receiving RTP 
packets with these header extensions should recognize that an SSRC 
change means that the client's behavior regarding this extension may 
also change? For example, when the session is first established, the 
client may be using the 'V' flag to reliably indicate the presence of 
voice in the audio payload... but after an SSRC change to a new client 
source, the 'V' flag may no longer be used at all. The 'design guidance' 
section does warn implementors that they may need to double-check the 
audio level (and 'V' flags) reported by the client, but a specific 
warning that this will almost always be necessary after an SSRC change 
might be useful as well.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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On 07/06/2011 09:23 AM, DRAGE, Keith (Keith) wrote:
> (As WG cochair)
>
> This is to announce a Working Group Last Call on
>
> draft-ietf-avtext-mixer-to-client-audio-level-03
>
> (Do note this is two parallel calls on two related drafts)
>
> Please review and supply comments by Wednesday 20th July 2011.
>
> It is helpful if you make some attempt to categorise uour comments, e.g. Major, Minor technical, Editorial.
>
> If you review the document and find no issue with the current version, then it is helpful to send a mail to the WG chairs saying so as this helps us assess the level of review.

Editorial:

Paragraph 1 of Section 3: no comma necessary after "More advanced 
mechanisms"

Paragraph 2 of Section 4 - "using the one or the two-byte header" seems 
to be a bit hard to parse. Maybe "using either the one-byte or the 
two-byte header format" would be better?

Paragraph 3 of Section 4 - 'adn' should be 'and'

Paragraph 4 of Section 4 - 'noice' should be 'noise'

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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The agenda is currently as follows (please note that times may be adjusted =
between now and the meeting, also please note that the slot is always subje=
ct to change):

Audio/Video Transport Extensions (avtext)
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

WEDNESDAY, July 27, 2011
1510-1610  Afternoon Session II
205 ABC

Personnel =20
Chairs: Magnus Westerlund <magnus.westerlund@ericsson.com>
Keith Drage <keith.drage@alcatel-lucent.com>
=20
Area Director: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com> =20

Mailing List =20
Address: avtext@ietf.org=20
To Subscribe: https://www.ietf.org/mailman/listinfo/avtext=20
Archive: http://www.ietf.org/mail-archive/web/avtext/=20

Jabber Chat =20
Room Address: xmpp:avtext@jabber.ietf.org=20
Logs: http://jabber.ietf.org/logs/avtext/=20

Agenda
------

15:10 - 15:20	Agenda Bash and Status=09
WG chairs

15:20 - 15:40	Audio Levels	=09
Emil Ivov / Jonathan Lennox
Issues arising from WG last call
draft-ietf-avtext-client-to-mixer-audio-level-03
draft-ietf-avtext-mixer-to-client-audio-level-03

15:40 - 15:50	Splicing	=09
Jinwei Xia
Direction forward
draft-xia-avtext-splicing-for-rtp-00

15:50 - 16:05	IEEE 1588/802.1AS Synchronisation for RTP Streams
Aidan Williams
Of interest to working group?
draft-williams-avtext-avbsync-01

16:05 - 16:10	Not allocated

16:10		Close


The following documents will not be separately discussed although comments =
are always welcome on the mailing list. We need reviewers for draft-ietf-av=
text-rams-scenarios-00
draft-ietf-avtext-multiple-clock-rates-01
draft-ietf-avtext-rams-scenarios-00


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Subject: Re: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03
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A reminder of this WGLC - please review as we'd like to get these out of th=
e door soon after IETF#81

Regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith)
> Sent: 06 July 2011 15:22
> To: 'avtext@ietf.org'
> Cc: 'draft-ietf-avtext-client-to-mixer-audio-level@tools.ietf.org'
> Subject: WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03
>=20
> (As WG cochair)
>=20
> This is to announce a Working Group Last Call on
>=20
> draft-ietf-avtext-client-to-mixer-audio-level-03
>=20
> (Do note this is two parallel calls on two related drafts)
>=20
> Please review and supply comments by Wednesday 20th July 2011.
>=20
> It is helpful if you make some attempt to categorise uour comments, e.g.
> Major, Minor technical, Editorial.
>=20
> If you review the document and find no issue with the current version,
> then it is helpful to send a mail to the WG chairs saying so as this help=
s
> us assess the level of review.
>=20
> Regards
>=20
> Keith
>=20
> P.S. The intent is to allow any identified show stoppers to be discussed
> at the face to face in Quebec, so the earlier such comments are made the
> better.

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A reminder of this WGLC - please review as we'd like to get these out of th=
e door soon after IETF#81

Regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith)
> Sent: 06 July 2011 15:18
> To: avtext@ietf.org
> Cc: 'draft-ietf-avtext-mixer-to-client-audio-level@tools.ietf.org'
> Subject: WGLC for draft-ietf-avtext-mixer-to-client-audio-level-03
>=20
> (As WG cochair)
>=20
> This is to announce a Working Group Last Call on
>=20
> draft-ietf-avtext-mixer-to-client-audio-level-03
>=20
> (Do note this is two parallel calls on two related drafts)
>=20
> Please review and supply comments by Wednesday 20th July 2011.
>=20
> It is helpful if you make some attempt to categorise uour comments, e.g.
> Major, Minor technical, Editorial.
>=20
> If you review the document and find no issue with the current version,
> then it is helpful to send a mail to the WG chairs saying so as this help=
s
> us assess the level of review.
>=20
> Regards
>=20
> Keith
>=20
> P.S. The intent is to allow any identified show stoppers to be discussed
> at the face to face in Quebec, so the earlier such comments are made the
> better.

From magnus.westerlund@ericsson.com  Wed Jul 13 07:38:57 2011
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Hi,

I have reviewed this version and I think it is in good shape and ready
for publication. I have one editorial comment.

1. Editorial Section 4.
       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |  ID   |  len=2  |0|  level 1    |0|  level 2    |0|  level 3  |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The bit numbering isn't aligned and I guess that may have helped masking
an issue in that the ID+Len fields are 9 bits, and that level 3 is only 6.

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 magnus.westerlund@ericsson.com  Wed Jul 13 08:22:22 2011
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Subject: Re: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03
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Hi,

I have reviewed this new version and have the following comments:

>From my previous review that I don't believe got answered.

1. Section 3:
   In addition, a flag bit (labeled V) indicates whether the encoder
   believes the audio packet contains voice activity (1) or does not
   (0).  The voice activity detection algorithm is unspecified and left
   implementation-specific.

Would it be of interest to have an signalling parameter for an
implementation to indicate its algorithm?

The reason I am coming back to this, is that if you don't know if it is
used, how can you then use the V bit. Is V=0 because there is no voice
activity or because there is no VAD implemented.

One can also expand the question to what a media sender without VAD
should set as value for V.

2. Section 6.
"One solution is header
   extension encryption [I-D.lennox-avtcore-srtp-encrypted-header-ext]."

I think the formulation of also pointing to lower layer solution is
appropriate. If one uses DTLS (without SRTP based media protection),
TLS, IPsec etc then this information would not be visible.

Cheers

Magnus Westerlund

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


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Subject: Re: [avtext] WGLC for draft-ietf-avtext-mixer-to-client-audio-level-03
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HI,

I have read the latest copy and do not have any issues.

Regards,

Peter Musgrave

On Wed, Jul 13, 2011 at 9:31 AM, DRAGE, Keith (Keith) <
keith.drage@alcatel-lucent.com> wrote:

> A reminder of this WGLC - please review as we'd like to get these out of
> the door soon after IETF#81
>
> Regards
>
> Keith
>
> > -----Original Message-----
> > From: DRAGE, Keith (Keith)
> > Sent: 06 July 2011 15:18
> > To: avtext@ietf.org
> > Cc: 'draft-ietf-avtext-mixer-to-client-audio-level@tools.ietf.org'
> > Subject: WGLC for draft-ietf-avtext-mixer-to-client-audio-level-03
> >
> > (As WG cochair)
> >
> > This is to announce a Working Group Last Call on
> >
> > draft-ietf-avtext-mixer-to-client-audio-level-03
> >
> > (Do note this is two parallel calls on two related drafts)
> >
> > Please review and supply comments by Wednesday 20th July 2011.
> >
> > It is helpful if you make some attempt to categorise uour comments, e.g.
> > Major, Minor technical, Editorial.
> >
> > If you review the document and find no issue with the current version,
> > then it is helpful to send a mail to the WG chairs saying so as this
> helps
> > us assess the level of review.
> >
> > Regards
> >
> > Keith
> >
> > P.S. The intent is to allow any identified show stoppers to be discussed
> > at the face to face in Quebec, so the earlier such comments are made the
> > better.
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>

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HI,=A0<div><br></div><div>I have read the latest copy and do not have any i=
ssues.=A0</div><div><br></div><div>Regards,=A0</div><div><br></div><div>Pet=
er Musgrave<br><br><div class=3D"gmail_quote">On Wed, Jul 13, 2011 at 9:31 =
AM, DRAGE, Keith (Keith) <span dir=3D"ltr">&lt;<a href=3D"mailto:keith.drag=
e@alcatel-lucent.com">keith.drage@alcatel-lucent.com</a>&gt;</span> wrote:<=
br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">A reminder of this WGLC - please review as =
we&#39;d like to get these out of the door soon after IETF#81<br>
<br>
Regards<br>
<font color=3D"#888888"><br>
Keith<br>
</font><div class=3D"im"><br>
&gt; -----Original Message-----<br>
&gt; From: DRAGE, Keith (Keith)<br>
&gt; Sent: 06 July 2011 15:18<br>
&gt; To: <a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><br>
&gt; Cc: &#39;<a href=3D"mailto:draft-ietf-avtext-mixer-to-client-audio-lev=
el@tools.ietf.org">draft-ietf-avtext-mixer-to-client-audio-level@tools.ietf=
.org</a>&#39;<br>
&gt; Subject: WGLC for draft-ietf-avtext-mixer-to-client-audio-level-03<br>
&gt;<br>
&gt; (As WG cochair)<br>
&gt;<br>
&gt; This is to announce a Working Group Last Call on<br>
&gt;<br>
&gt; draft-ietf-avtext-mixer-to-client-audio-level-03<br>
&gt;<br>
&gt; (Do note this is two parallel calls on two related drafts)<br>
&gt;<br>
&gt; Please review and supply comments by Wednesday 20th July 2011.<br>
&gt;<br>
&gt; It is helpful if you make some attempt to categorise uour comments, e.=
g.<br>
&gt; Major, Minor technical, Editorial.<br>
&gt;<br>
&gt; If you review the document and find no issue with the current version,=
<br>
&gt; then it is helpful to send a mail to the WG chairs saying so as this h=
elps<br>
&gt; us assess the level of review.<br>
&gt;<br>
</div><div class=3D"im">&gt; Regards<br>
&gt;<br>
&gt; Keith<br>
&gt;<br>
&gt; P.S. The intent is to allow any identified show stoppers to be discuss=
ed<br>
&gt; at the face to face in Quebec, so the earlier such comments are made t=
he<br>
&gt; better.<br>
_______________________________________________<br>
</div><div><div></div><div class=3D"h5">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>

--000e0cd3ff928cbf3204a80b489b--

From jonathan@vidyo.com  Thu Jul 14 15:49:54 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Thu, 14 Jul 2011 18:49:47 -0400
Thread-Topic: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03
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Subject: Re: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03
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On Jul 13, 2011, at 11:22 AM, Magnus Westerlund wrote:

> Hi,
>=20
> I have reviewed this new version and have the following comments:
>=20
> From my previous review that I don't believe got answered.
>=20
> 1. Section 3:
>   In addition, a flag bit (labeled V) indicates whether the encoder
>   believes the audio packet contains voice activity (1) or does not
>   (0).  The voice activity detection algorithm is unspecified and left
>   implementation-specific.
>=20
> Would it be of interest to have an signalling parameter for an
> implementation to indicate its algorithm?

I'm not sure whether the algorithm, specifically, is useful (it'd require e=
stablishing an IANA registry of specific VAD algorithms), but I could imagi=
ne that some way of indicating VAD used/not-used could be useful.

An alternative would be to mandate that use of the extension implies that y=
ou're sending meaningful VAD.

> 2. Section 6.
> "One solution is header
>   extension encryption [I-D.lennox-avtcore-srtp-encrypted-header-ext]."
>=20
> I think the formulation of also pointing to lower layer solution is
> appropriate. If one uses DTLS (without SRTP based media protection),
> TLS, IPsec etc then this information would not be visible.

Okay.  The intention is to follow the same general pattern as srtp-not-mand=
atory, so we'll see if we can re-use some language from there, and also ref=
erence it.

--
Jonathan Lennox
jonathan@vidyo.com



From magnus.westerlund@ericsson.com  Fri Jul 15 07:01:58 2011
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On 2011-07-15 00:49, Jonathan Lennox wrote:
> 
> On Jul 13, 2011, at 11:22 AM, Magnus Westerlund wrote:
> 
>> Hi,
>> 
>> I have reviewed this new version and have the following comments:
>> 
>> From my previous review that I don't believe got answered.
>> 
>> 1. Section 3: In addition, a flag bit (labeled V) indicates whether
>> the encoder believes the audio packet contains voice activity (1)
>> or does not (0).  The voice activity detection algorithm is
>> unspecified and left implementation-specific.
>> 
>> Would it be of interest to have an signalling parameter for an 
>> implementation to indicate its algorithm?
> 
> I'm not sure whether the algorithm, specifically, is useful (it'd
> require establishing an IANA registry of specific VAD algorithms),
> but I could imagine that some way of indicating VAD used/not-used
> could be useful.
> 
> An alternative would be to mandate that use of the extension implies
> that you're sending meaningful VAD.

Yes, I think signalling if a client use VAD or not would be most
appropriate solution.


> 
>> 2. Section 6. "One solution is header extension encryption
>> [I-D.lennox-avtcore-srtp-encrypted-header-ext]."
>> 
>> I think the formulation of also pointing to lower layer solution
>> is appropriate. If one uses DTLS (without SRTP based media
>> protection), TLS, IPsec etc then this information would not be
>> visible.
> 
> Okay.  The intention is to follow the same general pattern as
> srtp-not-mandatory, so we'll see if we can re-use some language from
> there, and also reference it.

In this case you could just copy the corresponding sentence from the
mixer to client draft.

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 keith.drage@alcatel-lucent.com  Tue Jul 19 15:00:17 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Tue, 19 Jul 2011 23:59:41 +0200
Thread-Topic: WGLC comments on draft-ietf-avtext-client-to-mixer-audio-level-03
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Subject: [avtext] WGLC comments on draft-ietf-avtext-client-to-mixer-audio-level-03
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No technical issues identifies. These are primarily editorial.

1)	Normative references to RFC 3550. At the moment the only reference in th=
e text is in section 1, which is really meant to be a description section, =
and the reference here cannot really be inferred to be normative. I have no=
 doubt you probably do need this as a normative reference, but it is curren=
tly carried implicitly in the normative reference to RFC 5285 made at the s=
tart of section 3. Why not add an explicit reference to RFC 3550 to the fir=
st paragraph there.

2)	Section 1

Need to remove non RFC 2119 usage of lower case "should".

   In a centralized Real-Time Transport Protocol (RTP) [RFC3550] audio
   conference, an audio mixer or forwarder receives audio streams from
   many or all of the conference participants.  It then selectively
   forwards some of them to other participants in the conference.  In
   large conferences, it is possible that such a server might be
   receiving a large number of streams, of which only a few should be
   forwarded to the other conference participants.

Believed what is meant here is "...of which only a few are intended to be f=
orwarded...".

3)	Section 5

This appears to be intended as an informative section. Therefore use of low=
er case "should" and "must" that might be mistaken for normative language a=
re best avoided.

   Mixers and forwarders generally should not base audio forwarding
   decisions directly on packet-by-packet audio level information, but
   rather should apply some analysis of the audio levels and trends.
   This general rule applies whether audio levels are provided by
   endpoints (as defined in this document), or are calculated at a
   server, as would be done in the absence of this information.  This
   section discusses several issues that mixers and forwarders may wish
   to take into account.  (Note that this section provides design
   guidance only, and is not normative.)

   Additionally, different participants may have their audio input set
   differently.  It may be useful to apply some sort of automatic gain
   control to the audio levels.  There are a number of possible
   approaches to acheiving this, e.g. by measuring peak audio levels, by
   average audio levels during speech, or by measuring background audio
   levels (average audio level levels during non-speech).

Is best rendered as:

   It is not desirable that mixers and forwarders base audio forwarding
   decisions directly on packet-by-packet audio level information, and=20
   instead, better to apply some analysis of the audio levels and trends.
   This general rule applies whether audio levels are provided by
   endpoints (as defined in this document), or are calculated at a
   server, as would be done in the absence of this information.  This
   section discusses several issues that mixers and forwarders may wish
   to take into account.  (Note that this section provides design
   guidance only, and is not normative.)

   Additionally, different participants can have their audio input set
   differently.  It can be useful to apply some sort of automatic gain
   control to the audio levels.  There are a number of possible
   approaches to achieving this, e.g. by measuring peak audio levels, by
   average audio levels during speech, or by measuring background audio
   levels (average audio level levels during non-speech).

Note also typo in "acheiving"

4)	Section 6

   A malicious endpoint could choose to set the values in this header
   extension falsely, so as to falsely claim that audio or voice is or
   is not present.  It is not clear what could be gained by falsely
   claiming that audio is not present, but an endpoint falsely claiming
   that audio is present could perform a denial-of-service attack on an
   audio conference, so as to send silence to suppress other conference
   members' audio.  Thus, a device relying on audio level data from
   untrusted endpoints SHOULD periodically audit the level information
   transmitted, taking appropriate corrective action if endpoints appear
   to be sending incorrect data. =20

Suggest a minor rewording of the requirement as follows as " if endpoints a=
ppear to be sending incorrect data" cannot be part of the condition.

   A malicious endpoint could choose to set the values in this header
   extension falsely, so as to falsely claim that audio or voice is or
   is not present.  It is not clear what could be gained by falsely
   claiming that audio is not present, but an endpoint falsely claiming
   that audio is present could perform a denial-of-service attack on an
   audio conference, so as to send silence to suppress other conference
   members' audio.  If a device relies on audio level data from
   untrusted endpoints, then the device SHOULD periodically audit the level=
=20
   information transmitted, taking appropriate corrective action. This
   provides a level of protection against endpoints sending incorrect data.=
 =20

5)	Section 6

(Note that as it is valid for an
   endpoint to choose to measure audio levels prior to encoding, some
   degree of discrepancy SHOULD be tolerated.)

Preceding a "SHOULD" with "Note..." and putting it in parentheses is liable=
 to lead to lack of clarity as to whether it is normative or informative. I=
 suggest it is probably informative and it should be reworded accordingly.

Regards

Keith

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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Tue, 19 Jul 2011 23:59:40 +0200
Thread-Topic: WGLC comments on draft-ietf-avtext-mixer-to-client-audio-level-03
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Cc: "draft-ietf-avtext-mixer-to-client-audio-level@tools.ietf.org" <draft-ietf-avtext-mixer-to-client-audio-level@tools.ietf.org>
Subject: [avtext] WGLC comments on draft-ietf-avtext-mixer-to-client-audio-level-03
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No technical issues identifies. These are editorial.

1)	Section 3:

   According to RFC 3550 [RFC3550] a mixer is expected to include in
   outgoing RTP packets a list of identifiers (CSRC IDs) indicating the
   sources that contributed to the resulting stream.  The presence of
   such CSRC IDs allows RTP clients to determine, in a binary way, the
   active speaker(s) in any given moment.  RTCP also provides a basic
   mechanism to map the CSRC IDs to user identities through the CNAME
   field.  More advanced mechanisms, may exist depending on the
   signaling protocol used to establish and control a conference.  In
   the case of the Session Initiation Protocol [RFC3261] for example,
   the Event Package for Conference State [RFC4575] defines a <src-id>
   tag which binds CSRC IDs to media streams and SIP URIs.

Change "may" to "can" to avoid using RFC 2119 language.

   According to RFC 3550 [RFC3550] a mixer is expected to include in
   outgoing RTP packets a list of identifiers (CSRC IDs) indicating the
   sources that contributed to the resulting stream.  The presence of
   such CSRC IDs allows RTP clients to determine, in a binary way, the
   active speaker(s) in any given moment.  RTCP also provides a basic
   mechanism to map the CSRC IDs to user identities through the CNAME
   field.  More advanced mechanisms, can exist depending on the
   signaling protocol used to establish and control a conference.  In
   the case of the Session Initiation Protocol [RFC3261] for example,
   the Event Package for Conference State [RFC4575] defines a <src-id>
   tag which binds CSRC IDs to media streams and SIP URIs.

2)	Section 3:

   It may sometimes happen that a conference involves more than a single
   mixer.  In such cases each of the mixers MAY choose to relay the CSRC
   list and audio-level information they receive from peer mixers (as
   long as the total CSRC count remains below 16).  Given that the
   maximum audio level is not precisely defined by this specification,
   it is likely that in such situations average audio levels would be
   perceptibly different for the participants located behind the
   different mixers.

Change "may" to "can" to avoid using RFC 2119 language.

   It can sometimes happen that a conference involves more than a single
   mixer.  In such cases each of the mixers MAY choose to relay the CSRC
   list and audio-level information they receive from peer mixers (as
   long as the total CSRC count remains below 16).  Given that the
   maximum audio level is not precisely defined by this specification,
   it is likely that in such situations average audio levels would be
   perceptibly different for the participants located behind the
   different mixers.

3)	Section 4

   Audio levels in this document are defined in the same manner as is
   audio noise level in the RTP Payload Comfort  Noise specification
   [RFC3389].  In the comfort noice specification, the overall magnitude
   of the noise level in comfort noise is encoded into the first byte of
   the payload, with spectral information about the noise in subsequent
   bytes.  This specification's audio level parameter is defined so as
   to be identical to the comfort noise payload's noise-level byte.

Typo "noice" --> "noise"

4)	Section 8

Presumably "dispatch" --> "avt" or "avtext".

5)	Section A.1

"singal" --> "single"

6)	Section A.1

     * @param overload the overload (point) of <tt>signal</tt>.
     * For example, <tt>overload</tt> may be {@link Byte#MAX_VALUE}
     * for 8-bit signed samples or {@link Short#MAX_VALUE} for
     * 16-bit signed samples.
     *

Change "may" to "can" to avoid using RFC 2119 language.

     * @param overload the overload (point) of <tt>signal</tt>.
     * For example, <tt>overload</tt> can be {@link Byte#MAX_VALUE}
     * for 8-bit signed samples or {@link Short#MAX_VALUE} for
     * 16-bit signed samples.
     *

Regards

Keith

From keith.drage@alcatel-lucent.com  Wed Jul 20 03:21:47 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Wed, 20 Jul 2011 12:21:39 +0200
Thread-Topic: WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03 and draft-ietf-avtext-mixer-to-client-audio-level-03
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Subject: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03 and draft-ietf-avtext-mixer-to-client-audio-level-03
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(As WG cochair)

We set a deadline for working group last call comments on the above two dra=
fts of today. So if you still have comments in the works, please get them p=
osted to the list with all due speed.

A day or two is not going to make much difference, but if you have any show=
 stoppers, it would be nice to know about them earlier.

It would also be good if a few more people who had read the draft and had n=
o comments would identify this to the chairs.

Regards

Keith

From stewe@stewe.org  Wed Jul 20 03:38:59 2011
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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03 and draft-ietf-avtext-mixer-to-client-audio-level-03
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Subject: Re: [avtext] WGLC for draft-ietf-avtext-client-to-mixer-audio-level-03 and draft-ietf-avtext-mixer-to-client-audio-level-03
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I scanned over these docs during a long flight last week, and was happy
with them.  Had not enough time (due to ongoing JCT-VC/MPEG meeting) to
create minor/editorial comments.
Stephan



On 7.20.2011 03:21 , "DRAGE, Keith (Keith)"
<keith.drage@alcatel-lucent.com> wrote:

>(As WG cochair)
>
>We set a deadline for working group last call comments on the above two
>drafts of today. So if you still have comments in the works, please get
>them posted to the list with all due speed.
>
>A day or two is not going to make much difference, but if you have any
>show stoppers, it would be nice to know about them earlier.
>
>It would also be good if a few more people who had read the draft and had
>no comments would identify this to the chairs.
>
>Regards
>
>Keith
>_______________________________________________
>avtext mailing list
>avtext@ietf.org
>https://www.ietf.org/mailman/listinfo/avtext



From keith.drage@alcatel-lucent.com  Thu Jul 21 09:08:53 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Thu, 21 Jul 2011 18:08:48 +0200
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Subject: [avtext] Revised agenda for AVTEXT
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(As WG cochair)

Please note the revised agenda at=20

http://www.ietf.org/proceedings/81/agenda/avtext.txt

The change is to reduce the time for the audio level drafts (as we have had=
 very few WGLC comments) and to host the PAYLOAD discussion which would oth=
erwise have been done by AVTCORE on Wednesday morning.

That gives AVTCORE a bit more flexibility to deal with other urgent issues.

No item has been removed from the agenda.

Regards

Keith

From john.elwell@siemens-enterprise.com  Fri Jul 22 06:37:45 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Fri, 22 Jul 2011 15:37:39 +0200
Thread-Topic: WGLC on client-to-mixer
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Looks ready to go.

John



John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

From john.elwell@siemens-enterprise.com  Fri Jul 22 06:37:45 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Fri, 22 Jul 2011 15:37:39 +0200
Thread-Topic: WGLC on mixer to client
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Looks ready to go apart from a couple of minor comment:

"   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."

Perhaps "every conference participant" should be "every contributing confer=
ence participant", since it is only those participants whose CSRCs are incl=
uded.

"   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"."
The words "to content" do not parse. Perhaps it should say "to provide cont=
ent".

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
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[Apologies for cross posting AVT lists - reason below]

=20

All,

=20

I will be discussing issues contained in draft-ramalho-payload g7110-00
at the AVT extensions timeslot on Wednesday afternoon.

=20

This draft is a payload WG draft, but there isn't a payload meeting at
IETF 81. And this draft was previously scheduled to be discussed  the
AVT core timeslot prior to its being moved to AVTEXT (thus the reason
for the cross post to all AVT groups). Roni and Ali encouraged me to say
a few words prior to the discussion this week.

=20

G.711.0 is a stateless and lossless compression of G.711 VoIP RTP
payloads. Thus it is a compression algorithm tailored for G.711. When
negotiated end-to-end it is negotiated "as if it were a codec".

=20

However, being lossless and stateless, G.711.0 can be applied anywhere
within an end-to-end G.711 negotiated session. When applied in this
manner, it suffers no voice quality degradation relative to G.711 (duh,
its lossless).

=20

Most of the open issues in the draft relate to the use of G.711.0 within
the context of an and-to-end G.711 negotiated session.

=20

I look forward to comments generated from the discussion of this draft
for those cases.

=20

Regards,

=20

Michael Ramalho

=20

=20

Michael Ramalho
Technical Leader
Product Development
mramalho@cisco.com <mailto:mramalho@cisco.com>=20
Phone: +1 919 476 2038
Mobile: +1 941 544 2844



Cisco Systems, Inc.
4564 Tuscana Drive
Sarasota, FL 34241-4201
United States
http://ramalho.webhop.info
Skype: mramalho_mar42

=20

=20

=09
=09

=20

=20


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<p class=3DMsoNormal>[Apologies for cross posting AVT lists &#8211; =
reason below]<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>All,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I will be discussing issues contained in =
draft-ramalho-payload
g7110-00 at the AVT extensions timeslot on Wednesday =
afternoon.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>This draft is a payload WG draft, but there =
isn&#8217;t a
payload meeting at IETF 81. And this draft was previously scheduled to =
be
discussed &nbsp;the AVT core timeslot prior to its being moved to AVTEXT =
(thus the
reason for the cross post to all AVT groups). Roni and Ali encouraged me =
to say
a few words prior to the discussion this week.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>G.711.0 is a stateless and lossless compression of =
G.711 VoIP
RTP payloads. Thus it is a compression algorithm tailored for G.711. =
When
negotiated end-to-end it is negotiated &#8220;as if it were a =
codec&#8221;.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>However, being lossless and stateless, G.711.0 can =
be
applied anywhere within an end-to-end G.711 negotiated session. When =
applied in
this manner, it suffers no voice quality degradation relative to G.711 =
(duh,
its lossless).<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Most of the open issues in the draft relate to the =
use of
G.711.0 within the context of an and-to-end G.711 negotiated =
session.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I look forward to comments generated from the =
discussion of
this draft for those cases.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regards,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Michael Ramalho<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

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    <p class=3DMsoNormal><span =
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    width=3D110 height=3D73 id=3D"Picture_x0020_1"
    src=3D"cid:image001.gif@01CC4AD1.08107BA0"
    alt=3D"http://www.cisco.com/swa/i/logo.gif"><o:p></o:p></span></p>
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    =
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Michael
    Ramalho</span></b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'><br>
    </span><b><span =
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    color:#666666'>Technical Leader</span></b><span =
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    font-family:"Arial","sans-serif";color:#666666'><br>
    </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>Product Development</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    <a href=3D"mailto:mramalho@cisco.com"><span =
style=3D'color:#666666'>mramalho@cisco.com</span></a><br>
    Phone: </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>+1 919 476 2038</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    Mobile: </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>+1 941 544 2844</span></b><span =
style=3D'font-size:8.5pt;
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Cisco
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    4564 Tuscana Drive<br>
    Sarasota, FL 34241-4201<br>
    United States<br>
    <b>http://ramalho.webhop.info</b><br>
    <b>Skype:</b> mramalho_mar42<o:p></o:p></span></p>
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From keith.drage@alcatel-lucent.com  Mon Jul 25 15:48:46 2011
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Subject: [avtext] Calling all AVTEXT presenters
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(As AVTEXT cochair)

Can we please have any slide presentations for AVTEXT by lunchtime local ti=
me tomorrow so that we can get them uploaded to the meeting materials site.

Keith

From keith.drage@alcatel-lucent.com  Tue Jul 26 08:24:51 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Tue, 26 Jul 2011 17:24:44 +0200
Thread-Topic: Presentation slides for Wednesday session
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(As AVTEXT cochair)

These should now all be uploaded.=20

For those participating remotely we have included the audio streaming and j=
abber links on the status slides - that information is also available elsew=
here.

Any issues let the chairs know as soon as possible.

We will also be looking for volunteer notetakers and a jabber scribe. If no=
one volunteers early we will compel volunteers at the start of the session!

Regards

Keith

From mramalho@cisco.com  Tue Jul 26 10:02:03 2011
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Thread-Topic: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, <draft-ramalho-payload-g7110@tools.ietf.org>
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Subject: Re: [avtext] [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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Hadriel,

Thank you for asking your question.

Re: "Have any IPR disclosures been submitted to the IETF for this?"

For the G.711.0 IETF payload draft, no.

In the ITU-T there are four IPR disclosures for G.711.0 - one for each
of the collaborating companies.

However, Cisco had hoped for a disclosures to be filed at either the
IETF or the ITU-T by IETF 81 by all four collaborating companies
explicitly stating that G.711.0 for "consumer software terminals" (e.g.,
softclients) be royalty-free. We hope such a statement will be released
in the near future.

RE: "For one, just to be able to do this requires sniffing SDP "

Not true. Virtually every implementation using G.711 I have found in
practice uses the STATIC payload type of 0 or 8 for G.711.

Thus every RTP packet with PT =3D [ 0 | 8 ] can only have a G.711 =
payload
inside of it.

When G.711.0 is negotiated end to end, it will have its own media line.
The suggested "hint" in the G.711 SDP is not to "signal G.711.0", but
rather for middleboxes like NATs/Firewalls/etc. to "expect G.711" in
either uncompressed (real G.711 with PT =3D [0 | 8]) or compressed (with
PT =3D Q) form as equivalents.

Part of the reason for the "hint in the G.711 SDP" is because SBCs will
likely strip the G.711.0 m line from the SDP because it doesn't know
about G.711.0 and only let the m line for G.711 to pass ... it might
just let an unrecognized G.711 attribute through. I would like to get
opinions for the group and appreciate your question.

Re: SRTP

Yep, the raw G.711 needs to be in the clear for the compression to work.
Note that the PT is in the clear with SRTP (because the RTP header is in
the clear) ... and thus SRTP should be able to encrypt G.711.0 without
issues.

Re: "Cut out the "In the Middle" section".

That is a possibility. However I can say that we have explicit customer
requests that IF such a compression is likely to be used by "in the
middle entities" (i.e., service providers or administrative domains
within a service provider), that we provide guidance on how to
accomplish that compression so that everyone does it the same way.

I also agree that keeping it in the draft will cause heartburn relative
to its absence ;-). But I've got customers to please. And I have thick
skin ;-).

Re: "Middleboxes are a pain"

Yes they are; but the are also a fact of life. I have a longer talk in
which I enumerate the issues related to SBCs ... but that talk is
inappropriate for the IETF slot (at least that is what I have been
told).

Thanks again,

Michael Ramalho


-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
Behalf Of Hadriel Kaplan
Sent: Tuesday, July 26, 2011 10:12 AM
To: draft-ramalho-payload-g7110@tools.ietf.org
Cc: avtext@ietf.org; payload@ietf.org
Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussed
inAVTEXT

[sorry for cross-posting, but this draft is going to be presented in
avtext]

Hi,
I read this draft and I have a couple comments, as follows:

1) Have any IPR disclosures been submitted to the IETF for this?  I ask
because I believe there are several submitted to ITU for this codec.

2) The stuff in section 6 on "In the Middle" is really controversial,
and arguably a very bad idea. =20

For one, just to be able to do this requires sniffing SDP which is only
possible if it's in cleartext, and if the sniffer can reassemble
fragments or stream segments, and if the sniffer is guaranteed to be in
the path for all SDP exchanges between the parties for ever.  And to be
able to really do it you have to *modify* the SDP which is a whole
different ballgame. =20

And then there's the way in which the draft describes doing it, by
inserting an fmtp attribute line of the new payload type, which is wrong
in my opinion - it should either replace the payload type in the m-line
and add the original 711a/u-law info as a new attribute, or better yet
add the new payload type as another option in the m-line and wait for an
SDP answer to choose (for protocols that have an offer/answer model).
Otherwise the middlebox has to know another middlebox is in the call
path a priori.

And then there's the problem of SRTP: if SRTP prevents the middlebox
from doing the compression, the middlebox needs to know when SRTP is
going to be used by inspecting SDP, which isn't as straightforward as
looking for "SAVP" in the m-line... e.g. the endpoints could be using
RFC 5939. =20

In summary: I would remove this idea/section from the draft.  Publishing
a draft defining the payload format and media type for G.711.0 is fairly
straight-forward and non-controversial (I think), but going beyond that
to "support" this man-in-the-middle thing is going to cause you
heartburn and delay getting the basic stuff published as an RFC.
Just my 2 cents.

-hadriel


On Jul 25, 2011, at 12:55 PM, Michael Ramalho wrote:

> This draft is a payload WG draft, but there isn't a payload meeting at
IETF 81. And this draft was previously scheduled to be discussed  the
AVT core timeslot prior to its being moved to AVTEXT (thus the reason
for the cross post to all AVT groups). Roni and Ali encouraged me to say
a few words prior to the discussion this week.
>=20
> G.711.0 is a stateless and lossless compression of G.711 VoIP RTP
payloads. Thus it is a compression algorithm tailored for G.711. When
negotiated end-to-end it is negotiated "as if it were a codec".
>=20
> However, being lossless and stateless, G.711.0 can be applied anywhere
within an end-to-end G.711 negotiated session. When applied in this
manner, it suffers no voice quality degradation relative to G.711 (duh,
its lossless).
>=20
> Most of the open issues in the draft relate to the use of G.711.0
within the context of an and-to-end G.711 negotiated session.
>=20
> I look forward to comments generated from the discussion of this draft
for those cases.
>=20

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

From HKaplan@acmepacket.com  Tue Jul 26 07:11:54 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "draft-ramalho-payload-g7110@tools.ietf.org" <draft-ramalho-payload-g7110@tools.ietf.org>
Date: Tue, 26 Jul 2011 10:11:49 -0400
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Cc: "avtext@ietf.org" <avtext@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [avtext] draft-ramalho-payload-g7110-00 to be discussed in AVTEXT
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[sorry for cross-posting, but this draft is going to be presented in avtext=
]

Hi,
I read this draft and I have a couple comments, as follows:

1) Have any IPR disclosures been submitted to the IETF for this?  I ask bec=
ause I believe there are several submitted to ITU for this codec.

2) The stuff in section 6 on "In the Middle" is really controversial, and a=
rguably a very bad idea. =20

For one, just to be able to do this requires sniffing SDP which is only pos=
sible if it's in cleartext, and if the sniffer can reassemble fragments or =
stream segments, and if the sniffer is guaranteed to be in the path for all=
 SDP exchanges between the parties for ever.  And to be able to really do i=
t you have to *modify* the SDP which is a whole different ballgame. =20

And then there's the way in which the draft describes doing it, by insertin=
g an fmtp attribute line of the new payload type, which is wrong in my opin=
ion - it should either replace the payload type in the m-line and add the o=
riginal 711a/u-law info as a new attribute, or better yet add the new paylo=
ad type as another option in the m-line and wait for an SDP answer to choos=
e (for protocols that have an offer/answer model).  Otherwise the middlebox=
 has to know another middlebox is in the call path a priori.

And then there's the problem of SRTP: if SRTP prevents the middlebox from d=
oing the compression, the middlebox needs to know when SRTP is going to be =
used by inspecting SDP, which isn't as straightforward as looking for "SAVP=
" in the m-line... e.g. the endpoints could be using RFC 5939. =20

In summary: I would remove this idea/section from the draft.  Publishing a =
draft defining the payload format and media type for G.711.0 is fairly stra=
ight-forward and non-controversial (I think), but going beyond that to "sup=
port" this man-in-the-middle thing is going to cause you heartburn and dela=
y getting the basic stuff published as an RFC.
Just my 2 cents.

-hadriel


On Jul 25, 2011, at 12:55 PM, Michael Ramalho wrote:

> This draft is a payload WG draft, but there isn=92t a payload meeting at =
IETF 81. And this draft was previously scheduled to be discussed  the AVT c=
ore timeslot prior to its being moved to AVTEXT (thus the reason for the cr=
oss post to all AVT groups). Roni and Ali encouraged me to say a few words =
prior to the discussion this week.
>=20
> G.711.0 is a stateless and lossless compression of G.711 VoIP RTP payload=
s. Thus it is a compression algorithm tailored for G.711. When negotiated e=
nd-to-end it is negotiated =93as if it were a codec=94.
>=20
> However, being lossless and stateless, G.711.0 can be applied anywhere wi=
thin an end-to-end G.711 negotiated session. When applied in this manner, i=
t suffers no voice quality degradation relative to G.711 (duh, its lossless=
).
>=20
> Most of the open issues in the draft relate to the use of G.711.0 within =
the context of an and-to-end G.711 negotiated session.
>=20
> I look forward to comments generated from the discussion of this draft fo=
r those cases.
>=20


From keith.drage@alcatel-lucent.com  Tue Jul 26 10:37:54 2011
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Subject: [avtext] Crossposting PAYLOAD
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(As AVTEXT cochair)

While the PAYLOAD presentation is piggybacking on the AVTEXT session, pleas=
e only post comments on PAYLOAD drafts to the PAYLOAD mailing list.

Regards

Keith

From mramalho@cisco.com  Tue Jul 26 13:32:22 2011
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Thread-Topic: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
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Subject: Re: [avtext] [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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Hadriel,

Thanks for the volley.

Comments in-line below (w/ MAR:).

Michael

-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
Sent: Tuesday, July 26, 2011 1:51 PM
To: Michael Ramalho (mramalho)
Cc: draft-ramalho-payload-g7110@tools.ietf.org; avtext@ietf.org;
payload@ietf.org
Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussed
inAVTEXT


On Jul 26, 2011, at 1:01 PM, Michael Ramalho (mramalho) wrote:

> RE: "For one, just to be able to do this requires sniffing SDP "
>=20
> Not true. Virtually every implementation using G.711 I have found in
> practice uses the STATIC payload type of 0 or 8 for G.711.
>=20
> Thus every RTP packet with PT =3D [ 0 | 8 ] can only have a G.711
payload
> inside of it.

Sorry I should have been explicit.  It needs to sniff to detect the PT
of the G.711.0 payload.  I.e., for the example diagram figure 4 in the
draft, device E needs to sniff the SDP offer to determine what the "Q"
value is, and device C needs to do the same for the SDP answer for the
reverse media stream.

MAR: Thanks for the clarification. Yes, you have got it right (one
exception shortly). SBCs normally "sniff SDP", so that might not be an
issue for them. But in the case where there are multiple SBCs in the
end-to-end path, one or more of them may strip out the G.711.0 m line
(as you have it below) before a given SBC sees the end-to-end SDP.

MAR: As an aside, some network providers may "configure PT =3D Q" in =
their
networks to make things easier (at the downside of not allowing their
applications to use PT =3D Q), although that would not be recommended =
(in
the IETF or in general).

MAR: Lastly, the compression can be applied in specific topologies -
when there are no middleboxes that deny flows based on RTP PT changes -
by tunnel endpoints that know how to compress/decompress G.711/G.711.0.
This tunneling is unspecified magic in the draft. It can also be used in
many of the same topologies where multi-hop header compression is
employed.

>=20
> When G.711.0 is negotiated end to end, it will have its own media
line.

Ummm... I'm confused.  The example in the draft is:
   m=3Daudio RTP/AVP 0
   a=3Drtpmap: 0 PCMU
   a=3Dfmtp:0 G7110 =3D Q   <<< the G.711 SDP hint

Are you saying that it's really this?:
   m=3Daudio 49120 RTP/AVP 0
   a=3Drtpmap:0 PCMU
   a=3Dfmtp:0 G7110 =3D Q
   m=3Daudio 49120 RTP/AVP 99
   a=3Drtpmap:99 G7110/8000
   a=3Dptime:20
   a=3Dfmtp:98 complaw=3Dmu

MAR: Precisely ... one m-line for G.711 (in your case mu-law) and one
m-line for G.711.0 (complaw=3Dmu, PT =3D 99). The "hint" is in the G.711
m-line (and the draft says so ... but you example makes it more clear ..
thanks).

MAR: I want to know if anyone "objects" to a new fmtp parameter
"G7110/8000". Worse case, if an SBC doesn't like that parameter, it
could strip it with no harm to setting up a G.711 flow.

> The suggested "hint" in the G.711 SDP is not to "signal G.711.0", but
> rather for middleboxes like NATs/Firewalls/etc. to "expect G.711" in
> either uncompressed (real G.711 with PT =3D [0 | 8]) or compressed =
(with
> PT =3D Q) form as equivalents.

But they're NOT equivalent.  For all intents and purposes G.711.0 is a
different codec from G.711, afaict.

MAR: OK .. we are even now .. I should have been more explicit here ;-).

MAR: While I believe the best "in the middle" use cases are when SBCs
are involved in the (lossless) transcode ... there are SOME LIMITED
cases in which SPs desire to switch from G.711 to G.711.0 though such
boxes without the transcode and without signaling (so-called "in-media
methods"). Such SPs account for the bandwidth savings relative to G.711
in their traffic engineering. Like I said earlier, I have a longer talk
on the subject and would be happy to discuss further. Perhaps I should
have asked for a slot in the behave WG for these issues?

That it can be implemented as a bump-in-the-wire without full DSPs
matters not.  The most logical thing for a middlebox to do would be to
handle it exactly like it handles middlebox-provided
transcoding/transrating cases today, with normal SDP
modifications/behavior, no? (and by that I mean act as a first-class
participant in SDP, by inserting/adding codecs into the same m-line, and
handling offer/answer appropriately)

MAR: See my last comment; there may be exceptions for specific customers
and topologies. Indeed, the reason for the "hint" is precisely to inform
devices such as SBCs that the switch from PT=3D0 to PT=3D99 (your =
example)
could occur naturally, so please don't kill the flow.

MAR: Thanks again for your commentary, I appreciate it.

-hadriel


From mramalho@cisco.com  Wed Jul 27 10:36:13 2011
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
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Subject: Re: [avtext] [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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Hadriel,

Re: Put G.711.0 on one m-line.

If others believe, as you do, that having G.711.0 on it's own m-line is
actually worse (in the sense of the probability of it being stripped
out) than having in on the same m-line as G.711 ... I am OK with that.

Do others have an opinion?

I am inclined to go with your recommendation if no one objects.

Re: "Are you saying that there's a GRE or MPLS tunnel between C and E,
and that C knows a priori (without SIP/SDP) that RTP it sends goes
through E, and that C knows a priori that E is capable of G.711.0
decompression to 711a/u-law?"

A tunnel of some sort ... including WAAS-like tunnels in enterprise use
cases.

The question of how the end tunnels (Box C and E) "find themselves" is
purposely left outside the scope of the draft.

What I am hoping for is that we can specify some principals for how you
compress/decompress so that the flow to the "G.711.0 unaware" endpoints
don't know it occurred. Better text is always welcome.

Thanks again,

Michael


-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
Sent: Wednesday, July 27, 2011 1:42 AM
To: Michael Ramalho (mramalho)
Cc: draft-ramalho-payload-g7110@tools.ietf.org; avtext@ietf.org;
payload@ietf.org
Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussed
inAVTEXT




On Jul 26, 2011, at 4:32 PM, Michael Ramalho (mramalho) wrote:

> MAR: Thanks for the clarification. Yes, you have got it right (one
> exception shortly). SBCs normally "sniff SDP", so that might not be an
> issue for them. But in the case where there are multiple SBCs in the
> end-to-end path, one or more of them may strip out the G.711.0 m line
> (as you have it below) before a given SBC sees the end-to-end SDP.

I think the odds of an SBC stripping out something would be far less if
this weren't encoded as two m-lines for the same media type (audio).
I.e., just do normal SDP like this:
  m=3Daudio 49120 RTP/AVP 0 99
  a=3Dptime:20
  a=3Drtpmap:99 G7110/8000
  a=3Dfmtp:99 complaw=3Dmu


> MAR: As an aside, some network providers may "configure PT =3D Q" in
their
> networks to make things easier (at the downside of not allowing their
> applications to use PT =3D Q), although that would not be recommended
(in
> the IETF or in general).

But even if you fixed a PT for G.711.0, wouldn't you still need to sniff
the SDP to learn the IP/port of the media?  Or are you just going to
guess that if the packet looks like RTP it is RTP?=20


> MAR: Lastly, the compression can be applied in specific topologies -
> when there are no middleboxes that deny flows based on RTP PT changes
-
> by tunnel endpoints that know how to compress/decompress
G.711/G.711.0.
> This tunneling is unspecified magic in the draft. It can also be used
in
> many of the same topologies where multi-hop header compression is
> employed.

I don't understand what you mean, or why that matters.  All I have to go
by is the diagram in the draft:


                                                **********************
                                                *                    *
      |------------|    |--------------------|  *  |---------------| *
      |     A:     |    |         B:         |  *  |  C: G.711.0   | *
      |  SENDING   |--->|    ZERO OR MORE    |---->|  Compression  | *
      |   G.711    |    |   ROUTING AND/OR   |  *  |   PT=3D[0|8]    | *
      |  ENDPOINT  |    |   SWITCHING HOPS   |  *  |  CHANGED TO   | *
      |            |    | AND/OR MIDDLEBOXES |  *  |    PT =3D Q     | *
      |____________|    |____________________|  *  |_______________| *
                                                *         |          *
                                               *          |          *
                                    ********* *           |          *
                                    *                     |          *
                                    *                     V          *
                                    *        |--------------------|  *
                                    *        |         D:         |  *
                          G.711.0   *        |    ZERO OR MORE    |  *
                        COMPRESSION *        |   ROUTING AND/OR   |  *
                          SEGMENT   *        |   SWITCHING HOPS   |  *
                         (payload   *        | AND/OR MIDDLEBOXES |  *
                           only)    *        |____________________|  *
                                    *                     |          *
                                    *                     |          *
                                    ***********           |          *
                                               *          |          *
                                                *         V          *
      |------------|    |--------------------|  *  |---------------| *
      |     G:     |    |         F:         |  *  |  E: G.711.0   | *
      | RECEIVING  |<---|    ZERO OR MORE    |<----| DECOMPRESSION | *
      |   G.711    |    |   ROUTING AND/OR   |  *  |    PT =3D Q     | *
      |  ENDPOINT  |    |   SWITCHING HOPS   |  *  | CHANGED BACK  | *
      |            |    | AND/OR MIDDLEBOXES |  *  |  TO PT=3D[0|8]  | *
      |____________|    |____________________|  *  |_______________| *
                                                *                    *
                                                **********************


Are you saying that there's a GRE or MPLS tunnel between C and E, and
that C knows a priori (without SIP/SDP) that RTP it sends goes through
E, and that C knows a priori that E is capable of G.711.0 decompression
to 711a/u-law?

-hadriel


From magnus.westerlund@ericsson.com  Wed Jul 27 13:59:55 2011
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Subject: [avtext] Confirming WG consensus on draft-xia-avtext-splicing-for-rtp-00
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WG,

In todays AVTEXT WG session we had some discussion on the way forward
for the splicing solution. We did ask two consensus questions.

There was consensus to use mixer in user detectable case.

There was consensus to use mixer also in the user non-detectable case.

In other words using mixers in all cases of splicing.

This email is to confirm this WG consensus decision please provide any
comments within 2 weeks.

Magnus Westerlund
WG Chair


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


From keith.drage@alcatel-lucent.com  Fri Jul 29 10:51:34 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtext@ietf.org" <avtext@ietf.org>
Date: Fri, 29 Jul 2011 19:51:29 +0200
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We're starting to think about the publication request on the audio level dr=
afts:

draft-ietf-avtext-client-to-mixer-audio-level
draft-ietf-avtext-mixer-to-client-audio-level

For that it would be useful to know if anyone has implementation experience=
 of these documents. Either send a message to the chairs (avtext-chairs@too=
ls.ietf.org) or to the list if you consider you have some useful work in th=
is area.

Note that this information does not have to be specific. What we are intere=
sted in is has implementation been performed that would have identified err=
ors in the protocol specification. Text along the lines of "I know of 3 com=
mercial deployments" or "I know of a bench model" are perfectly adequate if=
 you don't want to advertise.

Regards

Keith

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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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For:

draft-ietf-avtext-mixer-to-client-audio-level

The WGLC has been held and the comments made are mainly editorial. The docu=
ment will be updated shortly to address these comments, and the document is=
 ready for publication request. A poll in the meeting identified a signific=
ant number of people had read the WGLC version, and that it was ready for p=
ublication.

draft-ietf-avtext-client-to-mixer-audio-level

The WGLC has been held.=20

There is one significant technical enhancement to be made, where the propos=
al is to define an SDP extension to Indicate Voice Activity Detection (VAD)=
 support (or lack of). This was the mechanism that had support in the face =
to face meeting, and I'll use this message to seek confirmation of that dec=
ision on the list.

We'll try and get this extension text to the list. We will get the next ver=
sion of the draft reviewed by MMUSIC experts for this SDP content.

The document will be updated shortly. A poll in the meeting identified a si=
gnificant number of people had read the WGLC version, and that it was ready=
 for publication.

Apart from this MMUSIC review we will be looking to do the publication requ=
est shortly.

And if you have issues with the face-to-face meeting decision to use an SDP=
 extension to Indicate Voice Activity Detection (VAD) support then please p=
ost to the list as soon as possible.

Regards

Keith

From kpfleming@digium.com  Fri Jul 29 13:58:31 2011
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Subject: Re: [avtext] Current status of audio level drafts after face to face meeting (and confirmation of decisions)
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On 07/29/2011 02:19 PM, DRAGE, Keith (Keith) wrote:
> For:
>
> draft-ietf-avtext-mixer-to-client-audio-level
>
> The WGLC has been held and the comments made are mainly editorial. The document will be updated shortly to address these comments, and the document is ready for publication request. A poll in the meeting identified a significant number of people had read the WGLC version, and that it was ready for publication.
>
> draft-ietf-avtext-client-to-mixer-audio-level
>
> The WGLC has been held.
>
> There is one significant technical enhancement to be made, where the proposal is to define an SDP extension to Indicate Voice Activity Detection (VAD) support (or lack of). This was the mechanism that had support in the face to face meeting, and I'll use this message to seek confirmation of that decision on the list.
>
> We'll try and get this extension text to the list. We will get the next version of the draft reviewed by MMUSIC experts for this SDP content.
>
> The document will be updated shortly. A poll in the meeting identified a significant number of people had read the WGLC version, and that it was ready for publication.
>
> Apart from this MMUSIC review we will be looking to do the publication request shortly.
>
> And if you have issues with the face-to-face meeting decision to use an SDP extension to Indicate Voice Activity Detection (VAD) support then please post to the list as soon as possible.

I presume you mean SDP 'parameter' here, right?

Would this parameter indicate:

* that the offerer has VAD ability and will employ it for reporting 
audio levels

* that the offerer wishes the answerer to employ VAD if it can when it 
reports audio levels

* that the offerer *requires* the answerer to employ VAD when it reports 
audio levels

* something else?

Sorry, I missed the meeting yesterday and I don't think the minutes have 
been posted yet or I'd have consulted them before replying :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From emil@sip-communicator.org  Fri Jul 29 14:06:48 2011
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Date: Fri, 29 Jul 2011 23:06:43 +0200
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Subject: Re: [avtext] Current status of audio level drafts after face to face meeting (and confirmation of decisions)
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Hey Kevin,

=D0=9D=D0=B0 29.07.11 22:58, Kevin P. Fleming =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> On 07/29/2011 02:19 PM, DRAGE, Keith (Keith) wrote:
>> For:
>>
>> draft-ietf-avtext-mixer-to-client-audio-level
>>
>> The WGLC has been held and the comments made are mainly editorial. The=
 document will be updated shortly to address these comments, and the docu=
ment is ready for publication request. A poll in the meeting identified a=
 significant number of people had read the WGLC version, and that it was =
ready for publication.
>>
>> draft-ietf-avtext-client-to-mixer-audio-level
>>
>> The WGLC has been held.
>>
>> There is one significant technical enhancement to be made, where the p=
roposal is to define an SDP extension to Indicate Voice Activity Detectio=
n (VAD) support (or lack of). This was the mechanism that had support in =
the face to face meeting, and I'll use this message to seek confirmation =
of that decision on the list.
>>
>> We'll try and get this extension text to the list. We will get the nex=
t version of the draft reviewed by MMUSIC experts for this SDP content.
>>
>> The document will be updated shortly. A poll in the meeting identified=
 a significant number of people had read the WGLC version, and that it wa=
s ready for publication.
>>
>> Apart from this MMUSIC review we will be looking to do the publication=
 request shortly.
>>
>> And if you have issues with the face-to-face meeting decision to use a=
n SDP extension to Indicate Voice Activity Detection (VAD) support then p=
lease post to the list as soon as possible.
>=20
> I presume you mean SDP 'parameter' here, right?

Jonathan will be posting a detailed description here but basically it
will be a parameter in the line that advertises support for the level
extension.

> Would this parameter indicate:
>=20
> * that the offerer has VAD ability and will employ it for reporting=20
> audio levels

If set accordingly - yes. It could also be set to mean that the offerer
is not planning on using VAD.

> * that the offerer wishes the answerer to employ VAD if it can when it =

> reports audio levels
>=20
> * that the offerer *requires* the answerer to employ VAD when it report=
s=20
> audio levels

This is not the intention and we should probably make it clear in the
text that using the vad param is only meant for extension senders. (Good
point btw!)

Cheers,
Emil

--=20
http://jitsi.org


From jonathan@vidyo.com  Fri Jul 29 14:32:38 2011
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To: "avtext@ietf.org" <avtext@ietf.org>
Date: Fri, 29 Jul 2011 17:32:31 -0400
Thread-Topic: Proposed resolution of the presence of VAD in client-to-mixer audio levels
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Subject: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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In the working group session on Wednesday, there was a conclusion that it w=
ould be best to signal in SDP whether an endpoint is putting meaningful VAD=
 data into the client-to-mixer audio level extension.

My proposal is as follows.  Please give feedback on whether this satisfies =
everyone's objections and requirements.

There is an optional "vad" parameter defined as the extension attributes fi=
eld for the "urn:ietf:params:rtp-hdrext:ssrc-audio-level" header extension =
element. It takes the form "vad=3Don" if the creator of the audio level ext=
ension is providing VAD information, and "vad=3Doff" if it is not.

If vad=3Don is specified, the V bit in the audio header extension element i=
s defined as in the current draft.  If vad=3Doff is specified, the value of=
 the V bit is unspecified and MUST be ignored by receivers.

If the extension attribute is not specified, the default is vad=3Don.

This parameter is signaled in the extension attributes; it matches RFC 5285=
's <extensionattributes> BNF in SDP.=20

An SDP answer MUST include the same value of the vad=3D parameter as was co=
ntained in the offer (for the same extension-ID value), if the answer accep=
ted the extension.

An endpoint MAY negotiate either or both variants of the extension, with th=
e two forms having different extension ID values, using the standard RFC 52=
85 mechanisms.

Example SDP (showing both the syntax and the double-negotiation):

a=3Dextmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=3Don
a=3Dextmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=3Doff

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: [avtext] RFC 6332 on Multicast Acquisition Report Block Type for RTP Control Protocol (RTCP) Extended Reports (XRs)
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6332

        Title:      Multicast Acquisition Report Block Type 
                    for RTP Control Protocol (RTCP) Extended 
                    Reports (XRs) 
        Author:     A. Begen, E. Friedrich
        Status:     Standards Track
        Stream:     IETF
        Date:       July 2011
        Mailbox:    abegen@cisco.com, 
                    efriedri@cisco.com
        Pages:      16
        Characters: 35217
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-avtext-multicast-acq-rtcp-xr-06.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6332.txt

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 diagnostic 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 the
Multicast Acquisition (MA) report block, within the framework of RTP
Control Protocol (RTCP) Extended Reports (XRs) (RFC 3611).  This
document also defines the necessary signaling of the new MA report
block type in the Session Description Protocol (SDP).  [STANDARDS-TRACK]

This document is a product of the Audio/Video Transport Extensions Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

This announcement is sent to the IETF-Announce and rfc-dist lists.
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specifically noted otherwise on the RFC itself, all RFCs are for
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The RFC Editor Team
Association Management Solutions, LLC



From stephen.botzko@gmail.com  Fri Jul 29 15:47:14 2011
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Date: Fri, 29 Jul 2011 18:47:10 -0400
Message-ID: <CAMC7SJ70Ercc=6aP_fPNyYpLuXKp9nWwsNEf_CMbggq2Q1fhYw@mail.gmail.com>
From: Stephen Botzko <stephen.botzko@gmail.com>
To: Jonathan Lennox <jonathan@vidyo.com>
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Cc: "avtext@ietf.org" <avtext@ietf.org>
Subject: Re: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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--bcaec5171e2dda1a4604a93d0fdf
Content-Type: text/plain; charset=ISO-8859-1

It might make sense to have senders set V=0 when vad=off, rather than leave
it unspecified.

Also, if the far-end doesn't accept the extension (doesn't return vad= in
the answer), I would assume that V should be set to 0 anyway, and VAD is
off.  So I would question the default being VAD=on.  Having the default
value of on, when then far-end doesn't understand / accept the extension
seems incorrect.

Regards,
Stephen Botzko

On Fri, Jul 29, 2011 at 5:32 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

> In the working group session on Wednesday, there was a conclusion that it
> would be best to signal in SDP whether an endpoint is putting meaningful VAD
> data into the client-to-mixer audio level extension.
>
> My proposal is as follows.  Please give feedback on whether this satisfies
> everyone's objections and requirements.
>
> There is an optional "vad" parameter defined as the extension attributes
> field for the "urn:ietf:params:rtp-hdrext:ssrc-audio-level" header extension
> element. It takes the form "vad=on" if the creator of the audio level
> extension is providing VAD information, and "vad=off" if it is not.
>
> If vad=on is specified, the V bit in the audio header extension element is
> defined as in the current draft.  If vad=off is specified, the value of the
> V bit is unspecified and MUST be ignored by receivers.
>
> If the extension attribute is not specified, the default is vad=on.
>
> This parameter is signaled in the extension attributes; it matches RFC
> 5285's <extensionattributes> BNF in SDP.
>
> An SDP answer MUST include the same value of the vad= parameter as was
> contained in the offer (for the same extension-ID value), if the answer
> accepted the extension.
>
> An endpoint MAY negotiate either or both variants of the extension, with
> the two forms having different extension ID values, using the standard RFC
> 5285 mechanisms.
>
> Example SDP (showing both the syntax and the double-negotiation):
>
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on
> a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=off
>
> --
> Jonathan Lennox
> jonathan@vidyo.com
>
>
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext
>

--bcaec5171e2dda1a4604a93d0fdf
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

It might make sense to have senders set V=3D0 when vad=3Doff, rather than l=
eave it unspecified.<br><br>Also, if the far-end doesn&#39;t accept the ext=
ension (doesn&#39;t return vad=3D in the answer), I would assume that V sho=
uld be set to 0 anyway, and VAD is off.=A0 So I would question the default =
being VAD=3Don.=A0 Having the default value of on, when then far-end doesn&=
#39;t understand / accept the extension seems incorrect.<br>
<br>Regards,<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Fri, Ju=
l 29, 2011 at 5:32 PM, Jonathan Lennox <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:jonathan@vidyo.com">jonathan@vidyo.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex;">
In the working group session on Wednesday, there was a conclusion that it w=
ould be best to signal in SDP whether an endpoint is putting meaningful VAD=
 data into the client-to-mixer audio level extension.<br>
<br>
My proposal is as follows. =A0Please give feedback on whether this satisfie=
s everyone&#39;s objections and requirements.<br>
<br>
There is an optional &quot;vad&quot; parameter defined as the extension att=
ributes field for the &quot;urn:ietf:params:rtp-hdrext:ssrc-audio-level&quo=
t; header extension element. It takes the form &quot;vad=3Don&quot; if the =
creator of the audio level extension is providing VAD information, and &quo=
t;vad=3Doff&quot; if it is not.<br>

<br>
If vad=3Don is specified, the V bit in the audio header extension element i=
s defined as in the current draft. =A0If vad=3Doff is specified, the value =
of the V bit is unspecified and MUST be ignored by receivers.<br>
<br>
If the extension attribute is not specified, the default is vad=3Don.<br>
<br>
This parameter is signaled in the extension attributes; it matches RFC 5285=
&#39;s &lt;extensionattributes&gt; BNF in SDP.<br>
<br>
An SDP answer MUST include the same value of the vad=3D parameter as was co=
ntained in the offer (for the same extension-ID value), if the answer accep=
ted the extension.<br>
<br>
An endpoint MAY negotiate either or both variants of the extension, with th=
e two forms having different extension ID values, using the standard RFC 52=
85 mechanisms.<br>
<br>
Example SDP (showing both the syntax and the double-negotiation):<br>
<br>
a=3Dextmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=3Don<br>
a=3Dextmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=3Doff<br>
<br>
--<br>
Jonathan Lennox<br>
<a href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.com</a><br>
<br>
<br>
_______________________________________________<br>
avtext mailing list<br>
<a href=3D"mailto:avtext@ietf.org">avtext@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avtext" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/avtext</a><br>
</blockquote></div><br>

--bcaec5171e2dda1a4604a93d0fdf--

From jonathan@vidyo.com  Fri Jul 29 16:18:11 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Stephen Botzko <stephen.botzko@gmail.com>
Date: Fri, 29 Jul 2011 19:18:05 -0400
Thread-Topic: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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Subject: Re: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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On Jul 29, 2011, at 6:47 PM, Stephen Botzko wrote:

> It might make sense to have senders set V=3D0 when vad=3Doff, rather than=
 leave it unspecified.

The motivation for leaving it unspecified was to allow a no-op conversion f=
rom vad=3Don to vad=3Doff.  To be sure, (byte & 0x7f) or (byte | 0x80) isn'=
t a very computationally-expensive operation, though.=20

If we do want to set the bit to a fixed value, I think it'd be better to se=
t V=3D1, since if someone's somehow confused about the setting of the vad=
=3D flag, they're better off incorrectly interpreting a packet as having vo=
ice when it doesn't, rather than incorrectly interpreting it as not having =
voice when it does. =20

> Also, if the far-end doesn't accept the extension (doesn't return vad=3D =
in the answer), I would assume that V should be set to 0 anyway, and VAD is=
 off.  So I would question the default being VAD=3Don.  Having the default =
value of on, when then far-end doesn't understand / accept the extension se=
ems incorrect.

An answer with an extmap attribute whose vad=3D value -- specified as the e=
xtensionattributes on urn:ietf:params:rtp-hdrext:ssrc-audio-level -- doesn'=
t match the corresponding extmap attribute in the offer would be an invalid=
 answer, under the rules I've proposed. (It's not clear whether this is tru=
e in general for extmaps' extensionattributes values in RFC 5285, but it se=
ems like it would be a good idea for it to be).

If you don't understand or don't want to use a extension header element, or=
 you don't like the vad=3D value offered, you simply don't include the rele=
vant extmap attribute in the SDP answer.  (This is the standard RFC 5285 of=
fer/answer procedure.)


--
Jonathan Lennox
jonathan@vidyo.com



From stephen.botzko@gmail.com  Fri Jul 29 16:24:24 2011
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Jonathan Lennox <jonathan@vidyo.com>
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Subject: Re: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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in-line
Stephen Botzko

On Fri, Jul 29, 2011 at 7:18 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>
> On Jul 29, 2011, at 6:47 PM, Stephen Botzko wrote:
>
> > It might make sense to have senders set V=0 when vad=off, rather than
> leave it unspecified.
>
> The motivation for leaving it unspecified was to allow a no-op conversion
> from vad=on to vad=off.  To be sure, (byte & 0x7f) or (byte | 0x80) isn't a
> very computationally-expensive operation, though.
>
> If we do want to set the bit to a fixed value, I think it'd be better to
> set V=1, since if someone's somehow confused about the setting of the vad=
> flag, they're better off incorrectly interpreting a packet as having voice
> when it doesn't, rather than incorrectly interpreting it as not having voice
> when it does.
>

that is reasonable.

>
> > Also, if the far-end doesn't accept the extension (doesn't return vad= in
> the answer), I would assume that V should be set to 0 anyway, and VAD is
> off.  So I would question the default being VAD=on.  Having the default
> value of on, when then far-end doesn't understand / accept the extension
> seems incorrect.
>
> An answer with an extmap attribute whose vad= value -- specified as the
> extensionattributes on urn:ietf:params:rtp-hdrext:ssrc-audio-level --
> doesn't match the corresponding extmap attribute in the offer would be an
> invalid answer, under the rules I've proposed. (It's not clear whether this
> is true in general for extmaps' extensionattributes values in RFC 5285, but
> it seems like it would be a good idea for it to be).
>
> If you don't understand or don't want to use a extension header element, or
> you don't like the vad= value offered, you simply don't include the relevant
> extmap attribute in the SDP answer.  (This is the standard RFC 5285
> offer/answer procedure.)
>
> Yes  So if vad is not included in the SDP answer, than vad is off?

>
> --
> Jonathan Lennox
> jonathan@vidyo.com
>
>
>

--20cf307cfc80ea007804a93d9402
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in-line<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Fri, Jul 29,=
 2011 at 7:18 PM, Jonathan Lennox <span dir=3D"ltr">&lt;<a href=3D"mailto:j=
onathan@vidyo.com">jonathan@vidyo.com</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex;">
<div class=3D"im"><br>
On Jul 29, 2011, at 6:47 PM, Stephen Botzko wrote:<br>
<br>
&gt; It might make sense to have senders set V=3D0 when vad=3Doff, rather t=
han leave it unspecified.<br>
<br>
</div>The motivation for leaving it unspecified was to allow a no-op conver=
sion from vad=3Don to vad=3Doff. =A0To be sure, (byte &amp; 0x7f) or (byte =
| 0x80) isn&#39;t a very computationally-expensive operation, though.<br>
<br>
If we do want to set the bit to a fixed value, I think it&#39;d be better t=
o set V=3D1, since if someone&#39;s somehow confused about the setting of t=
he vad=3D flag, they&#39;re better off incorrectly interpreting a packet as=
 having voice when it doesn&#39;t, rather than incorrectly interpreting it =
as not having voice when it does.<br>
</blockquote><div>=A0</div><div>that is reasonable. <br></div><blockquote c=
lass=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px s=
olid rgb(204, 204, 204); padding-left: 1ex;">
<div class=3D"im"><br>
&gt; Also, if the far-end doesn&#39;t accept the extension (doesn&#39;t ret=
urn vad=3D in the answer), I would assume that V should be set to 0 anyway,=
 and VAD is off. =A0So I would question the default being VAD=3Don. =A0Havi=
ng the default value of on, when then far-end doesn&#39;t understand / acce=
pt the extension seems incorrect.<br>

<br>
</div>An answer with an extmap attribute whose vad=3D value -- specified as=
 the extensionattributes on urn:ietf:params:rtp-hdrext:ssrc-audio-level -- =
doesn&#39;t match the corresponding extmap attribute in the offer would be =
an invalid answer, under the rules I&#39;ve proposed. (It&#39;s not clear w=
hether this is true in general for extmaps&#39; extensionattributes values =
in RFC 5285, but it seems like it would be a good idea for it to be).<br>

<br>
If you don&#39;t understand or don&#39;t want to use a extension header ele=
ment, or you don&#39;t like the vad=3D value offered, you simply don&#39;t =
include the relevant extmap attribute in the SDP answer. =A0(This is the st=
andard RFC 5285 offer/answer procedure.)<br>

<font color=3D"#888888"><br></font></blockquote><div>Yes=A0 So if vad is no=
t included in the SDP answer, than vad is off?<br></div><blockquote class=
=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid=
 rgb(204, 204, 204); padding-left: 1ex;">
<font color=3D"#888888">
<br>
--<br>
</font><div><div></div><div class=3D"h5">Jonathan Lennox<br>
<a href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.com</a><br>
<br>
<br>
</div></div></blockquote></div><br>

--20cf307cfc80ea007804a93d9402--

From jonathan@vidyo.com  Fri Jul 29 16:36:38 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Stephen Botzko <stephen.botzko@gmail.com>
Date: Fri, 29 Jul 2011 19:36:35 -0400
Thread-Topic: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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Subject: Re: [avtext] Proposed resolution of the presence of VAD in client-to-mixer audio levels
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On Jul 29, 2011, at 7:24 PM, Stephen Botzko wrote:

On Fri, Jul 29, 2011 at 7:18 PM, Jonathan Lennox <jonathan@vidyo.com<mailto=
:jonathan@vidyo.com>> wrote:

An answer with an extmap attribute whose vad=3D value -- specified as the e=
xtensionattributes on urn:ietf:params:rtp-hdrext:ssrc-audio-level -- doesn'=
t match the corresponding extmap attribute in the offer would be an invalid=
 answer, under the rules I've proposed. (It's not clear whether this is tru=
e in general for extmaps' extensionattributes values in RFC 5285, but it se=
ems like it would be a good idea for it to be).

If you don't understand or don't want to use a extension header element, or=
 you don't like the vad=3D value offered, you simply don't include the rele=
vant extmap attribute in the SDP answer.  (This is the standard RFC 5285 of=
fer/answer procedure.)

Yes  So if vad is not included in the SDP answer, than vad is off?

If the extmap isn't included in the SDP answer, then the whole audio level =
extension header is off, i.e. must not be used.

If the extmap is included in the SDP answer, then whether to set the V bit =
in the header extension element is controlled by the value of the parameter=
 in the answer, which must match its value in the offer.

This parameter only controls the contents of the extension header; this doe=
sn't in any way signal or control "vad" in the sense of whether either endp=
oint should use silence suppression or discontinuous transmission or comfor=
t noise or anything like that.

Would it be clearer to name the attribute something like "vbit" rather than=
 "vad", to emphasize that this is only specifying the contents of the heade=
r extension element, not anything else?

--
Jonathan Lennox
jonathan@vidyo.com<mailto:jonathan@vidyo.com>



--_000_6ADD3479AE6B43C89705484DE78026F4vidyocom_
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode:=
 space; -webkit-line-break: after-white-space; "><br><div><div>On Jul 29, 2=
011, at 7:24 PM, Stephen Botzko wrote:</div><blockquote type=3D"cite"><font=
 class=3D"Apple-style-span" color=3D"#000000"><br></font><div class=3D"gmai=
l_quote">On Fri, Jul 29, 2011 at 7:18 PM, Jonathan Lennox <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.com</a>&gt;</span=
> wrote:<blockquote class=3D"gmail_quote" style=3D"margin-top: 0pt; margin-=
right: 0pt; margin-bottom: 0pt; margin-left: 0.8ex; border-left-width: 1px;=
 border-left-style: solid; border-left-color: rgb(204, 204, 204); padding-l=
eft: 1ex; position: static; z-index: auto; "><div class=3D"im">

<br>
</div>An answer with an extmap attribute whose vad=3D value -- specified as=
 the extensionattributes on urn:ietf:params:rtp-hdrext:ssrc-audio-level -- =
doesn't match the corresponding extmap attribute in the offer would be an i=
nvalid answer, under the rules I've proposed. (It's not clear whether this =
is true in general for extmaps' extensionattributes values in RFC 5285, but=
 it seems like it would be a good idea for it to be).<br>

<br>
If you don't understand or don't want to use a extension header element, or=
 you don't like the vad=3D value offered, you simply don't include the rele=
vant extmap attribute in the SDP answer. &nbsp;(This is the standard RFC 52=
85 offer/answer procedure.)<br>

<font color=3D"#888888"><br></font></blockquote><div>Yes&nbsp; So if vad is=
 not included in the SDP answer, than vad is off?</div></div>
</blockquote><br></div><div>If the extmap isn't included in the SDP answer,=
 then the whole audio level extension header is off, i.e. must not be used.=
</div><div><br></div><div>If the extmap is included in the SDP answer, then=
 whether to set the V bit in the header extension element is controlled by =
the value of the parameter in the answer, which must match its value in the=
 offer.</div><div><br></div><div>This parameter only controls the contents =
of the extension header; this doesn't in any way signal or control "vad" in=
 the sense of whether either endpoint should use silence suppression or dis=
continuous transmission or comfort noise or anything like that.</div><div><=
br></div><div>Would it be clearer to name the attribute something like "vbi=
t" rather than "vad", to emphasize that this is only specifying the content=
s of the header extension element, not anything else?</div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; color:=
 rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: no=
rmal; font-weight: normal; letter-spacing: normal; line-height: normal; orp=
hans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-sp=
ace: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacin=
g: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-e=
ffect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px=
; font-size: medium; "><span class=3D"Apple-style-span" style=3D"border-col=
lapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-style: n=
ormal; font-variant: normal; font-weight: normal; letter-spacing: normal; l=
ine-height: normal; orphans: 2; text-indent: 0px; text-transform: none; whi=
te-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-s=
pacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations=
-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width=
: 0px; font-size: medium; "><div style=3D"word-wrap: break-word; -webkit-nb=
sp-mode: space; -webkit-line-break: after-white-space; ">--</div><div style=
=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: af=
ter-white-space; ">Jonathan Lennox<br><a href=3D"mailto:jonathan@vidyo.com"=
>jonathan@vidyo.com</a><br><br></div></span></span>
</div>
<br></body></html>=

--_000_6ADD3479AE6B43C89705484DE78026F4vidyocom_--

From keith.drage@alcatel-lucent.com  Fri Jul 29 17:51:47 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "Kevin P. Fleming" <kpfleming@digium.com>, "avtext@ietf.org" <avtext@ietf.org>
Date: Sat, 30 Jul 2011 02:51:38 +0200
Thread-Topic: [avtext] Current status of audio level drafts after face to face meeting (and confirmation of decisions)
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Subject: Re: [avtext] Current status of audio level drafts after face to face meeting (and confirmation of decisions)
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More specific details have been posted by Jonathan.

Yes it means a SDP parameter.

The notes are not yet available but I suspect they will not give a great de=
al of detail behind the decision. I think the general opinion was that peop=
le wanted something explicit.

Regards

Keith

> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf
> Of Kevin P. Fleming
> Sent: 29 July 2011 21:58
> To: avtext@ietf.org
> Subject: Re: [avtext] Current status of audio level drafts after face to
> face meeting (and confirmation of decisions)
>=20
> On 07/29/2011 02:19 PM, DRAGE, Keith (Keith) wrote:
> > For:
> >
> > draft-ietf-avtext-mixer-to-client-audio-level
> >
> > The WGLC has been held and the comments made are mainly editorial. The
> document will be updated shortly to address these comments, and the
> document is ready for publication request. A poll in the meeting
> identified a significant number of people had read the WGLC version, and
> that it was ready for publication.
> >
> > draft-ietf-avtext-client-to-mixer-audio-level
> >
> > The WGLC has been held.
> >
> > There is one significant technical enhancement to be made, where the
> proposal is to define an SDP extension to Indicate Voice Activity
> Detection (VAD) support (or lack of). This was the mechanism that had
> support in the face to face meeting, and I'll use this message to seek
> confirmation of that decision on the list.
> >
> > We'll try and get this extension text to the list. We will get the next
> version of the draft reviewed by MMUSIC experts for this SDP content.
> >
> > The document will be updated shortly. A poll in the meeting identified =
a
> significant number of people had read the WGLC version, and that it was
> ready for publication.
> >
> > Apart from this MMUSIC review we will be looking to do the publication
> request shortly.
> >
> > And if you have issues with the face-to-face meeting decision to use an
> SDP extension to Indicate Voice Activity Detection (VAD) support then
> please post to the list as soon as possible.
>=20
> I presume you mean SDP 'parameter' here, right?
>=20
> Would this parameter indicate:
>=20
> * that the offerer has VAD ability and will employ it for reporting
> audio levels
>=20
> * that the offerer wishes the answerer to employ VAD if it can when it
> reports audio levels
>=20
> * that the offerer *requires* the answerer to employ VAD when it reports
> audio levels
>=20
> * something else?
>=20
> Sorry, I missed the meeting yesterday and I don't think the minutes have
> been posted yet or I'd have consulted them before replying :-)
>=20
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpflemin=
g
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> _______________________________________________
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> avtext@ietf.org
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From keith.drage@alcatel-lucent.com  Fri Jul 29 17:58:44 2011
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Date: Sat, 30 Jul 2011 02:58:37 +0200
Thread-Topic: [avtext] RFC 6332 on Multicast Acquisition Report Block Type for	RTP Control Protocol (RTCP) Extended Reports (XRs)
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Subject: Re: [avtext] RFC 6332 on Multicast Acquisition Report Block Type for	RTP Control Protocol (RTCP) Extended Reports (XRs)
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Thanks to Ali Begen and Eric Friedrich for progressing this through to a pu=
blished document.

Keith

> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf
> Of rfc-editor@rfc-editor.org
> Sent: 29 July 2011 23:20
> To: ietf-announce@ietf.org; rfc-dist@rfc-editor.org
> Cc: avtext@ietf.org; rfc-editor@rfc-editor.org
> Subject: [avtext] RFC 6332 on Multicast Acquisition Report Block Type for
> RTP Control Protocol (RTCP) Extended Reports (XRs)
>=20
>=20
> A new Request for Comments is now available in online RFC libraries.
>=20
>=20
>         RFC 6332
>=20
>         Title:      Multicast Acquisition Report Block Type
>                     for RTP Control Protocol (RTCP) Extended
>                     Reports (XRs)
>         Author:     A. Begen, E. Friedrich
>         Status:     Standards Track
>         Stream:     IETF
>         Date:       July 2011
>         Mailbox:    abegen@cisco.com,
>                     efriedri@cisco.com
>         Pages:      16
>         Characters: 35217
>         Updates/Obsoletes/SeeAlso:   None
>=20
>         I-D Tag:    draft-ietf-avtext-multicast-acq-rtcp-xr-06.txt
>=20
>         URL:        http://www.rfc-editor.org/rfc/rfc6332.txt
>=20
> 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 diagnostic 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 the
> Multicast Acquisition (MA) report block, within the framework of RTP
> Control Protocol (RTCP) Extended Reports (XRs) (RFC 3611).  This
> document also defines the necessary signaling of the new MA report
> block type in the Session Description Protocol (SDP).  [STANDARDS-TRACK]
>=20
> This document is a product of the Audio/Video Transport Extensions Workin=
g
> Group of the IETF.
>=20
> This is now a Proposed Standard Protocol.
>=20
> STANDARDS TRACK: This document specifies an Internet standards track
> protocol for the Internet community,and requests discussion and
> suggestions
> for improvements.  Please refer to the current edition of the Internet
> Official Protocol Standards (STD 1) for the standardization state and
> status of this protocol.  Distribution of this memo is unlimited.
>=20
> This announcement is sent to the IETF-Announce and rfc-dist lists.
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>=20
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>=20
> Requests for special distribution should be addressed to either the
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>=20
>=20
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>=20
>=20
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