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Thread-Topic: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Jochen Issing" <iss@ieee.org>, <payload@ietf.org>, <mmusic@ietf.org>
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Subject: Re: [payload] [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Hi Jochen,

I am not that familiar with the details of the variation modes. If you
want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. and
HE-AACv2, it seems you could include multiple payload types in the
m-line, and then specify the corresponding fmtp parameters for each. Is
there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
within a single payload type?

Charles

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
Behalf Of Jochen Issing
> Sent: Wednesday, November 30, 2011 7:18 AM
> To: payload@ietf.org; mmusic@ietf.org
> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>=20
> Dear experts,
>=20
> I need to double-apologize before posting our findings:
> 1) for cross-posting, as I do not know, where this issue has its exact
home, which is hopefully not a
> response killer.
> 2) for the long mail, even though I tried to keep it short.
>=20
> Now, here's the actual request:
>=20
> We are referring to RFC3640 SDP fields only, as this is most often the
stumbling block. Let's use the
> following example:
>=20
>    m=3Daudio 49170 RTP/AVP 96
>    a=3Drtpmap:96 mpeg4-generic/48000/2
>    a=3Dfmtp:96 streamtype=3D5; profile-level-id=3D15; mode=3DAAC-hbr;
config=3D1190; sizeLength=3D13;
> indexLength=3D3; indexDeltaLength=3D3; constantDuration=3D1024
>=20
> The fields of interest are in the fmtp line: 'profile-level-id' and
'config'.
> In contrast to other codecs, e.g. G.XXX, AAC integrates many different
types of sub codecs, which are
> identified using the Audio Object Type (AOT). The AOT and some
additional (codec internal) information
> is stored in the Audio Specific Config (ASC), which is conveyed in the
'config' field.
> The 'profile-level-id' points to a profile with a certain level. The
profile comprises a set of one or
> more AOTs and the level restricts the stream to a maximum sample rate
and/or number channels. The
> profile-level-ids, as specified in ISO 14496-3, are partly
hierarchical and most often ambiguous.
>=20
> Suppose the classic SIP use case that two clients want to exchange
their codec capabilities.
> The first client specifies some payload types in the SDP part of SIP
INVITE which can differ in their
> profile-level-id and/or config.
> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
signal these codecs/AOTs he could
> either indicate all formats with the same profile-level-id (e.g., 48),
which includes all these AOTs
> or he could specify profile-level-ids, that are most 'narrow' to the
actual specified AOT or he could
> even mix this principle.
> On the other hand, the ASC might use implicit signaling, explicit
backwards-compatible, or explicit
> non-backwards-compatible.
> Bobs main problem would be to determine, whether the profile-level-id
was chosen to really identify
> the necessary (and applied) AAC AOTs or if he could rely on the ASC or
if the profile-level-id was
> just brawling about the encoders feature.
>=20
> To clean up this mess of opportunities, we would like to specify a
non-ambiguous way to utilize the
> profile-level-id and config fields appropriately and provide a guide
on how to accomplish clean and
> easier signalization and thus improve interoperability.
>=20
>=20
> But first of all we would like to make sure, that you agree to this
problem statement or do we miss
> something here?
> Comments are as always highly appreciated.
>=20
> Best Regards,
> --
> jochen
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From iss@IEEE.ORG  Mon Dec  5 14:50:55 2011
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From: Jochen Issing <iss@IEEE.ORG>
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To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Cc: mmusic@ietf.org, payload@ietf.org
Subject: Re: [payload] [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Hi Charles,

thanks for your response! However, the issue we identified is somewhat =
different:

The problem already takes place with a single fmtp line:
For example, one client indicates that it can (decode and) encode an AAC =
stream using
the config field in fmtp. The other side, receiving this information, =
then must
check its decoder, if it can decode the ASC and needs to setup its =
encoder,
so that it uses the very same ASC.

The problem occurs at the encoder of the caller (the receiver of the =
first SDP):
It might not be possible with reasonable effort to get an encoder =
creating the
same ASC as given in SDP (encoders are in general not initialized using =
an ASC).

I hope, this explained our problem much clearer.
Best,

jochen

On 05.12.2011, at 20:02, Charles Eckel (eckelcu) wrote:

> Hi Jochen,
>=20
> I am not that familiar with the details of the variation modes. If you
> want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. =
and
> HE-AACv2, it seems you could include multiple payload types in the
> m-line, and then specify the corresponding fmtp parameters for each. =
Is
> there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
> within a single payload type?
>=20
> Charles
>=20
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Jochen Issing
>> Sent: Wednesday, November 30, 2011 7:18 AM
>> To: payload@ietf.org; mmusic@ietf.org
>> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>>=20
>> Dear experts,
>>=20
>> I need to double-apologize before posting our findings:
>> 1) for cross-posting, as I do not know, where this issue has its =
exact
> home, which is hopefully not a
>> response killer.
>> 2) for the long mail, even though I tried to keep it short.
>>=20
>> Now, here's the actual request:
>>=20
>> We are referring to RFC3640 SDP fields only, as this is most often =
the
> stumbling block. Let's use the
>> following example:
>>=20
>>   m=3Daudio 49170 RTP/AVP 96
>>   a=3Drtpmap:96 mpeg4-generic/48000/2
>>   a=3Dfmtp:96 streamtype=3D5; profile-level-id=3D15; mode=3DAAC-hbr;
> config=3D1190; sizeLength=3D13;
>> indexLength=3D3; indexDeltaLength=3D3; constantDuration=3D1024
>>=20
>> The fields of interest are in the fmtp line: 'profile-level-id' and
> 'config'.
>> In contrast to other codecs, e.g. G.XXX, AAC integrates many =
different
> types of sub codecs, which are
>> identified using the Audio Object Type (AOT). The AOT and some
> additional (codec internal) information
>> is stored in the Audio Specific Config (ASC), which is conveyed in =
the
> 'config' field.
>> The 'profile-level-id' points to a profile with a certain level. The
> profile comprises a set of one or
>> more AOTs and the level restricts the stream to a maximum sample rate
> and/or number channels. The
>> profile-level-ids, as specified in ISO 14496-3, are partly
> hierarchical and most often ambiguous.
>>=20
>> Suppose the classic SIP use case that two clients want to exchange
> their codec capabilities.
>> The first client specifies some payload types in the SDP part of SIP
> INVITE which can differ in their
>> profile-level-id and/or config.
>> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
> signal these codecs/AOTs he could
>> either indicate all formats with the same profile-level-id (e.g., =
48),
> which includes all these AOTs
>> or he could specify profile-level-ids, that are most 'narrow' to the
> actual specified AOT or he could
>> even mix this principle.
>> On the other hand, the ASC might use implicit signaling, explicit
> backwards-compatible, or explicit
>> non-backwards-compatible.
>> Bobs main problem would be to determine, whether the profile-level-id
> was chosen to really identify
>> the necessary (and applied) AAC AOTs or if he could rely on the ASC =
or
> if the profile-level-id was
>> just brawling about the encoders feature.
>>=20
>> To clean up this mess of opportunities, we would like to specify a
> non-ambiguous way to utilize the
>> profile-level-id and config fields appropriately and provide a guide
> on how to accomplish clean and
>> easier signalization and thus improve interoperability.
>>=20
>>=20
>> But first of all we would like to make sure, that you agree to this
> problem statement or do we miss
>> something here?
>> Comments are as always highly appreciated.
>>=20
>> Best Regards,
>> --
>> jochen
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic

--=20
jochen


From iss@ieee.org  Mon Dec  5 15:50:14 2011
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Cc: "payload@ietf.org" <payload@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [payload] [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Hi Mike,

as far as I understand the AAC codecs, ASCs of the same AOT (and 
channels and sample rate) are not necessarily 'compatible' to each 
other, especially when it comes to more recent codecs, like AAC-ELD, 
where critical information might be included in the ASC. Thus, for a 
uniform usage of ASCs, it should be bit-exact. I'm afraid the caller 
needs to add a payload type if he does not manage to reproduce the ASC 
indicated by the callee, which would be then used in one direction only.

Regarding the signalization of SBR and PS (HE-AAC/HE-AACv2) you are 
right, that's another issue. However, with presence of SIP/SDP and 
offer/answer functionality, implicit and backwards-compatible 
signalization do not make sense at all IMHO. The legacy receiver can 
simply disregard the unsupported codecs. Anyhow, this is neither stated 
anywhere.

Best,
jochen

Severa, Michael J (Mike) wrote:
> My personal feeling is that the ASC need not be identical in syntax, just in semantics (one of the unfortunate aspects of AAC-related signaling is that there are a number of ways to signal the same bitstream). Another unfortunate aspect is that one of the options is to signal implicitly, which essentially means dont signal at all, rather 'detect' at decode time. That is obviously not useful for a capabilities exchange step.
>
> I think there might also be other open questions surrounding ambiguities in signaling AAC-LC/HE-AAC that should be worked through to generate a best or acceptable practices approach.
>
> Two of the ones that jump out at me are:
> - the question of what timescale from the rtpmap line means when you intend to be using HE-AAC in a 3640 environment (3GPP has explicitly indicated this should be sampling frequency of the underlying, 'core' AAC-LC portion of an HE-AAC stream. I assume this is the same semantic in 3640 (though I didnt notice it specified explicitly)). This question is probably only relevant in cases where the goal is to assume HE-AAC (something the multiple fmtp line suggestion might be implying).
> - how HE-AAC can be detected in implicit mode during negotiation (again, 3GPP added an SBR-Enabled flag to the SDP to allow HE-AAC to be detected ahead of time even with implicit signaling, im not sure there is an analogous solution for 3640).
>
> Both of these questions are probably irrelevant if implicit signaling is disallowed for HE-AAC (though in the first case that would require parsing the ASC during negotiation). Using multiple fmtp lines might be one way to 'disable' the implicit signaling requirement (since the backwards compatibility concern associated with implicit signaling can be dealt with by an explicit AAC-LC-only ftmp (and rtpmap)line).
>
> Sorry if that's a tangent.
>
> thanks,
>
> Mike
>
> ________________________________________
> From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] On Behalf Of Jochen Issing [iss@IEEE.ORG]
> Sent: Monday, December 05, 2011 2:50 PM
> To: Charles Eckel (eckelcu)
> Cc: mmusic@ietf.org; payload@ietf.org
> Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>
> Hi Charles,
>
> thanks for your response! However, the issue we identified is somewhat different:
>
> The problem already takes place with a single fmtp line:
> For example, one client indicates that it can (decode and) encode an AAC stream using
> the config field in fmtp. The other side, receiving this information, then must
> check its decoder, if it can decode the ASC and needs to setup its encoder,
> so that it uses the very same ASC.
>
> The problem occurs at the encoder of the caller (the receiver of the first SDP):
> It might not be possible with reasonable effort to get an encoder creating the
> same ASC as given in SDP (encoders are in general not initialized using an ASC).
>
> I hope, this explained our problem much clearer.
> Best,
>
> jochen
>
> On 05.12.2011, at 20:02, Charles Eckel (eckelcu) wrote:
>
>> Hi Jochen,
>>
>> I am not that familiar with the details of the variation modes. If you
>> want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. and
>> HE-AACv2, it seems you could include multiple payload types in the
>> m-line, and then specify the corresponding fmtp parameters for each. Is
>> there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
>> within a single payload type?
>>
>> Charles
>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Jochen Issing
>>> Sent: Wednesday, November 30, 2011 7:18 AM
>>> To: payload@ietf.org; mmusic@ietf.org
>>> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>>>
>>> Dear experts,
>>>
>>> I need to double-apologize before posting our findings:
>>> 1) for cross-posting, as I do not know, where this issue has its exact
>> home, which is hopefully not a
>>> response killer.
>>> 2) for the long mail, even though I tried to keep it short.
>>>
>>> Now, here's the actual request:
>>>
>>> We are referring to RFC3640 SDP fields only, as this is most often the
>> stumbling block. Let's use the
>>> following example:
>>>
>>>    m=audio 49170 RTP/AVP 96
>>>    a=rtpmap:96 mpeg4-generic/48000/2
>>>    a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr;
>> config=1190; sizeLength=13;
>>> indexLength=3; indexDeltaLength=3; constantDuration=1024
>>>
>>> The fields of interest are in the fmtp line: 'profile-level-id' and
>> 'config'.
>>> In contrast to other codecs, e.g. G.XXX, AAC integrates many different
>> types of sub codecs, which are
>>> identified using the Audio Object Type (AOT). The AOT and some
>> additional (codec internal) information
>>> is stored in the Audio Specific Config (ASC), which is conveyed in the
>> 'config' field.
>>> The 'profile-level-id' points to a profile with a certain level. The
>> profile comprises a set of one or
>>> more AOTs and the level restricts the stream to a maximum sample rate
>> and/or number channels. The
>>> profile-level-ids, as specified in ISO 14496-3, are partly
>> hierarchical and most often ambiguous.
>>> Suppose the classic SIP use case that two clients want to exchange
>> their codec capabilities.
>>> The first client specifies some payload types in the SDP part of SIP
>> INVITE which can differ in their
>>> profile-level-id and/or config.
>>> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
>> signal these codecs/AOTs he could
>>> either indicate all formats with the same profile-level-id (e.g., 48),
>> which includes all these AOTs
>>> or he could specify profile-level-ids, that are most 'narrow' to the
>> actual specified AOT or he could
>>> even mix this principle.
>>> On the other hand, the ASC might use implicit signaling, explicit
>> backwards-compatible, or explicit
>>> non-backwards-compatible.
>>> Bobs main problem would be to determine, whether the profile-level-id
>> was chosen to really identify
>>> the necessary (and applied) AAC AOTs or if he could rely on the ASC or
>> if the profile-level-id was
>>> just brawling about the encoders feature.
>>>
>>> To clean up this mess of opportunities, we would like to specify a
>> non-ambiguous way to utilize the
>>> profile-level-id and config fields appropriately and provide a guide
>> on how to accomplish clean and
>>> easier signalization and thus improve interoperability.
>>>
>>>
>>> But first of all we would like to make sure, that you agree to this
>> problem statement or do we miss
>>> something here?
>>> Comments are as always highly appreciated.
>>>
>>> Best Regards,
>>> --
>>> jochen
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>
> --
> jochen
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

-- 
jochen

From stewe@stewe.org  Wed Dec  7 08:09:11 2011
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Date: Wed, 07 Dec 2011 08:08:40 -0800
From: Stephan Wenger <stewe@stewe.org>
To: <mmusic@ietf.org>
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Thread-Topic: Incoming liaison statement from MMUSIC
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Hi all,
mmusic has received an incoming liaison from MPEG, indicating that they have
started on "codec independent media description code points".  My
understanding of this work is that these code points may be represented in
SDP.
My personal viewpoint is that this may well be sound architectural work, but
it may be better conducted here in mmusic, at least when it comes to SDP
representations and the related IANA work.  However, I know nothing about
this work beyond what's written in the statement, which is why I used "soft"
language.  Perhaps some of you can shed light into what's really going on.
The assorted avt WGs are probably also affected, as some of the proposed
code points sound like they are being described in payload formats, RTP
extensions and perhaps even in xrblock.  Let me suggest to use only the
music list for discussions, as only mmusic is being addressed in the
statement.
As it is short, I have attached the statement in word format to this email.
It will also appear shortly in the IETF liaison database.
Regards,
Stephan



--B_3406090145_2760267
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Hi all,</div><div>mmusic has=
 received an incoming liaison from MPEG, indicating that they have started o=
n "codec independent media description code points". &nbsp;My understanding =
of this work is that these code points may be represented in SDP.</div><div>=
My personal viewpoint is that this may well be sound architectural work, but=
 it may be better conducted here in mmusic, at least when it comes to SDP re=
presentations and the related IANA work. &nbsp;However,&nbsp;I know nothing =
about this work beyond what's written in the statement, which is why I used =
"soft" language. &nbsp;Perhaps some of you can shed light into what's really=
 going on.</div><div>The assorted avt WGs are probably also affected, as som=
e of the proposed code points sound like they are being described in payload=
 formats, RTP extensions and perhaps even in xrblock. &nbsp;Let me suggest t=
o use only the music list for discussions, as only mmusic is being addressed=
 in the statement.</div><div>As it is short, I have attached the statement i=
n word format to this email. &nbsp;It will also appear shortly in the IETF l=
iaison database.</div><div>Regards,</div><div>Stephan</div></body></html>

--B_3406090145_2760267--


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From: "Severa, Michael J (Mike)" <mike.severa@alcatel-lucent.com>
To: Jochen Issing <iss@IEEE.ORG>, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
Date: Mon, 5 Dec 2011 17:28:12 -0600
Thread-Topic: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [payload] [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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My personal feeling is that the ASC need not be identical in syntax, just i=
n semantics (one of the unfortunate aspects of AAC-related signaling is tha=
t there are a number of ways to signal the same bitstream). Another unfortu=
nate aspect is that one of the options is to signal implicitly, which essen=
tially means dont signal at all, rather 'detect' at decode time. That is ob=
viously not useful for a capabilities exchange step.=20

I think there might also be other open questions surrounding ambiguities in=
 signaling AAC-LC/HE-AAC that should be worked through to generate a best o=
r acceptable practices approach.=20

Two of the ones that jump out at me are:
- the question of what timescale from the rtpmap line means when you intend=
 to be using HE-AAC in a 3640 environment (3GPP has explicitly indicated th=
is should be sampling frequency of the underlying, 'core' AAC-LC portion of=
 an HE-AAC stream. I assume this is the same semantic in 3640 (though I did=
nt notice it specified explicitly)). This question is probably only relevan=
t in cases where the goal is to assume HE-AAC (something the multiple fmtp =
line suggestion might be implying).=20
- how HE-AAC can be detected in implicit mode during negotiation (again, 3G=
PP added an SBR-Enabled flag to the SDP to allow HE-AAC to be detected ahea=
d of time even with implicit signaling, im not sure there is an analogous s=
olution for 3640).=20

Both of these questions are probably irrelevant if implicit signaling is di=
sallowed for HE-AAC (though in the first case that would require parsing th=
e ASC during negotiation). Using multiple fmtp lines might be one way to 'd=
isable' the implicit signaling requirement (since the backwards compatibili=
ty concern associated with implicit signaling can be dealt with by an expli=
cit AAC-LC-only ftmp (and rtpmap)line).=20

Sorry if that's a tangent.=20

thanks,=20

Mike

________________________________________
From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] On Behalf Of Jochen=
 Issing [iss@IEEE.ORG]
Sent: Monday, December 05, 2011 2:50 PM
To: Charles Eckel (eckelcu)
Cc: mmusic@ietf.org; payload@ietf.org
Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues

Hi Charles,

thanks for your response! However, the issue we identified is somewhat diff=
erent:

The problem already takes place with a single fmtp line:
For example, one client indicates that it can (decode and) encode an AAC st=
ream using
the config field in fmtp. The other side, receiving this information, then =
must
check its decoder, if it can decode the ASC and needs to setup its encoder,
so that it uses the very same ASC.

The problem occurs at the encoder of the caller (the receiver of the first =
SDP):
It might not be possible with reasonable effort to get an encoder creating =
the
same ASC as given in SDP (encoders are in general not initialized using an =
ASC).

I hope, this explained our problem much clearer.
Best,

jochen

On 05.12.2011, at 20:02, Charles Eckel (eckelcu) wrote:

> Hi Jochen,
>
> I am not that familiar with the details of the variation modes. If you
> want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. and
> HE-AACv2, it seems you could include multiple payload types in the
> m-line, and then specify the corresponding fmtp parameters for each. Is
> there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
> within a single payload type?
>
> Charles
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Jochen Issing
>> Sent: Wednesday, November 30, 2011 7:18 AM
>> To: payload@ietf.org; mmusic@ietf.org
>> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>>
>> Dear experts,
>>
>> I need to double-apologize before posting our findings:
>> 1) for cross-posting, as I do not know, where this issue has its exact
> home, which is hopefully not a
>> response killer.
>> 2) for the long mail, even though I tried to keep it short.
>>
>> Now, here's the actual request:
>>
>> We are referring to RFC3640 SDP fields only, as this is most often the
> stumbling block. Let's use the
>> following example:
>>
>>   m=3Daudio 49170 RTP/AVP 96
>>   a=3Drtpmap:96 mpeg4-generic/48000/2
>>   a=3Dfmtp:96 streamtype=3D5; profile-level-id=3D15; mode=3DAAC-hbr;
> config=3D1190; sizeLength=3D13;
>> indexLength=3D3; indexDeltaLength=3D3; constantDuration=3D1024
>>
>> The fields of interest are in the fmtp line: 'profile-level-id' and
> 'config'.
>> In contrast to other codecs, e.g. G.XXX, AAC integrates many different
> types of sub codecs, which are
>> identified using the Audio Object Type (AOT). The AOT and some
> additional (codec internal) information
>> is stored in the Audio Specific Config (ASC), which is conveyed in the
> 'config' field.
>> The 'profile-level-id' points to a profile with a certain level. The
> profile comprises a set of one or
>> more AOTs and the level restricts the stream to a maximum sample rate
> and/or number channels. The
>> profile-level-ids, as specified in ISO 14496-3, are partly
> hierarchical and most often ambiguous.
>>
>> Suppose the classic SIP use case that two clients want to exchange
> their codec capabilities.
>> The first client specifies some payload types in the SDP part of SIP
> INVITE which can differ in their
>> profile-level-id and/or config.
>> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
> signal these codecs/AOTs he could
>> either indicate all formats with the same profile-level-id (e.g., 48),
> which includes all these AOTs
>> or he could specify profile-level-ids, that are most 'narrow' to the
> actual specified AOT or he could
>> even mix this principle.
>> On the other hand, the ASC might use implicit signaling, explicit
> backwards-compatible, or explicit
>> non-backwards-compatible.
>> Bobs main problem would be to determine, whether the profile-level-id
> was chosen to really identify
>> the necessary (and applied) AAC AOTs or if he could rely on the ASC or
> if the profile-level-id was
>> just brawling about the encoders feature.
>>
>> To clean up this mess of opportunities, we would like to specify a
> non-ambiguous way to utilize the
>> profile-level-id and config fields appropriately and provide a guide
> on how to accomplish clean and
>> easier signalization and thus improve interoperability.
>>
>>
>> But first of all we would like to make sure, that you agree to this
> problem statement or do we miss
>> something here?
>> Comments are as always highly appreciated.
>>
>> Best Regards,
>> --
>> jochen
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic

--
jochen

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

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Subject: Re: [payload] [AVTCORE] Incoming liaison statement from MMUSIC
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Hi Stephan

I can explain a bit.  We noticed in systems work at MPEG two things
a) we wanted to say something about the signal that was being compressed (audio or video) that was nothing to do with the compression, and we ended up having to 'pick' a codec document and point at it, because the codes we wanted were always documented with codecs.
b) the codes are also copied from spec. to spec. in many cases (so they happen to have values having the same meanings); in a few cases, each codec has 'invented its own' and the codes needlessly differ from spec. to spec.

Examples: the 'classic three' values used to describe video color: color_primaries, transfer_characteristics, matrix_coefficients.  These are copied from spec. to spec., with the more recent specs having more values. Recently there was a mistake noticed in an equation, and I think it had to be corrected in several places.  Another example is 'frame packing' (stereo packed into a mono video stream); the specs actually don't always align on the definitions here.

The variation of semantics for some points could be a problem; different values for the same semantics is a nuisance, e.g. when wanting to be codec-independent or when transcoding. Different numbers of values for the same code in different specs is a nuisance, and raises questions, and so on.

The basic idea is to collect a whole bunch of these, document them in one place, and remove the definitions from a whole bunch of specs.  So, for example, in a video spec. one might be able to say "this 8-bit field carries a color_primaries value as defined in [[this new spec]]; if absent, the default value is 23".  Correspondingly, in say SDP, one might be able to say "if this field is 0, the video is mono; otherwise, it carries a frame_packing values as defined in [[this new spec]]".

Makes sense?


On Dec 7, 2011, at 8:08 , Stephan Wenger wrote:

> Hi all,
> mmusic has received an incoming liaison from MPEG, indicating that they have started on "codec independent media description code points".  My understanding of this work is that these code points may be represented in SDP.
> My personal viewpoint is that this may well be sound architectural work, but it may be better conducted here in mmusic, at least when it comes to SDP representations and the related IANA work.  However, I know nothing about this work beyond what's written in the statement, which is why I used "soft" language.  Perhaps some of you can shed light into what's really going on.
> The assorted avt WGs are probably also affected, as some of the proposed code points sound like they are being described in payload formats, RTP extensions and perhaps even in xrblock.  Let me suggest to use only the music list for discussions, as only mmusic is being addressed in the statement.
> As it is short, I have attached the statement in word format to this email.  It will also appear shortly in the IETF liaison database.
> Regards,
> Stephan
> <29n124561.doc>_______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

David Singer
Multimedia and Software Standards, Apple Inc.


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi =
Stephan<div><br></div><div>I can explain a bit. &nbsp;We noticed in =
systems work at MPEG two things</div><div>a) we wanted to say something =
about the signal that was being compressed (audio or video) that was =
nothing to do with the compression, and we ended up having to 'pick' a =
codec document and point at it, because the codes we wanted were always =
documented with codecs.</div><div>b) the codes are also copied from =
spec. to spec. in many cases (so they happen to have values having the =
same meanings); in a few cases, each codec has 'invented its own' and =
the codes needlessly differ from spec. to =
spec.</div><div><br></div><div>Examples: the 'classic three' values used =
to describe video color: color_primaries, transfer_characteristics, =
matrix_coefficients. &nbsp;These are copied from spec. to spec., with =
the more recent specs having more values. Recently there was a mistake =
noticed in an equation, and I think it had to be corrected in several =
places. &nbsp;Another example is 'frame packing' (stereo packed into a =
mono video stream); the specs actually don't always align on the =
definitions here.</div><div><br></div><div>The variation of semantics =
for some points could be a problem; different values for the same =
semantics is a nuisance, e.g. when wanting to be codec-independent or =
when transcoding. Different numbers of values for the same code in =
different specs is a nuisance, and raises questions, and so =
on.</div><div><br></div><div>The basic idea is to collect a whole bunch =
of these, document them in one place, and remove the definitions from a =
whole bunch of specs. &nbsp;So, for example, in a video spec. one might =
be able to say "this 8-bit field carries a color_primaries value as =
defined in [[this new spec]]; if absent, the default value is 23". =
&nbsp;Correspondingly, in say SDP, one might be able to say "if this =
field is 0, the video is mono; otherwise, it carries a frame_packing =
values as defined in [[this new spec]]".</div><div><br></div><div>Makes =
sense?</div><div><br></div><div><br><div><div>On Dec 7, 2011, at 8:08 , =
Stephan Wenger wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: =
14px; font-family: Calibri, sans-serif; "><div>Hi all,</div><div>mmusic =
has received an incoming liaison from MPEG, indicating that they have =
started on "codec independent media description code points". &nbsp;My =
understanding of this work is that these code points may be represented =
in SDP.</div><div>My personal viewpoint is that this may well be sound =
architectural work, but it may be better conducted here in mmusic, at =
least when it comes to SDP representations and the related IANA work. =
&nbsp;However,&nbsp;I know nothing about this work beyond what's written =
in the statement, which is why I used "soft" language. &nbsp;Perhaps =
some of you can shed light into what's really going on.</div><div>The =
assorted avt WGs are probably also affected, as some of the proposed =
code points sound like they are being described in payload formats, RTP =
extensions and perhaps even in xrblock. &nbsp;Let me suggest to use only =
the music list for discussions, as only mmusic is being addressed in the =
statement.</div><div>As it is short, I have attached the statement in =
word format to this email. &nbsp;It will also appear shortly in the IETF =
liaison database.</div><div>Regards,</div><div>Stephan</div></div>
=
<span>&lt;29n124561.doc&gt;</span>________________________________________=
_______<br>Audio/Video Transport Core Maintenance<br><a =
href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>https://www.ietf.org/mail=
man/listinfo/avt<br></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
auto; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><div><div>David Singer</div><div>Multimedia and =
Software Standards, Apple =
Inc.</div></div></div></span></div></span></span>
</div>
<br></div></body></html>=

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

	Title           : RTP Payload Format for Bluetooth's SBC audio codec
	Author(s)       : Christian Hoene
                          Frans de Bont
	Filename        : draft-ietf-payload-rtp-sbc-01.txt
	Pages           : 24
	Date            : 2011-12-15

   This document specifies a Real-time Transport Protocol (RTP) payload
   format to be used for the low complexity subband codec (SBC), which
   is the mandatory audio codec of the Advanced Audio Distribution
   Profile (A2DP) Specification written by the Bluetooth(r) Special
   Interest Group (SIG). The payload format is designed to be able to
   interoperate with existing Bluetooth A2DP devices, to provide high
   streaming audio quality, interactive audio transmission over the
   internet, and ultra-low delay coding for jam sessions on the
   internet. This document contains also a media type registration
   which specifies the use of the RTP payload format.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-payload-rtp-sbc-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-payload-rtp-sbc-01.txt


From hoene@uni-tuebingen.de  Thu Dec 15 06:50:43 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: <payload@ietf.org>
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Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-sbc-01.txt
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Hello,

we would like to ask for WGLC if possible.

The Bluetooth SBC is a quite nice codec ranging from wideband speech, CD
quality up to distributed ensembles as it has ultra-low delay. The
complexity is very low and last not least, to the best of my knowledge it is
patent free. We have one implementation of this draft in libopal.

With best regards,

 Christian Hoene




> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
> Behalf Of internet-drafts@ietf.org
> Sent: Thursday, December 15, 2011 3:46 PM
> To: i-d-announce@ietf.org
> Cc: payload@ietf.org
> Subject: [payload] I-D Action: draft-ietf-payload-rtp-sbc-01.txt
> 
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
directories.
> This draft is a work item of the Audio/Video Transport Payloads Working
> Group of the IETF.
> 
> 	Title           : RTP Payload Format for Bluetooth's SBC audio codec
> 	Author(s)       : Christian Hoene
>                           Frans de Bont
> 	Filename        : draft-ietf-payload-rtp-sbc-01.txt
> 	Pages           : 24
> 	Date            : 2011-12-15
> 
>    This document specifies a Real-time Transport Protocol (RTP) payload
>    format to be used for the low complexity subband codec (SBC), which
>    is the mandatory audio codec of the Advanced Audio Distribution
>    Profile (A2DP) Specification written by the Bluetooth(r) Special
>    Interest Group (SIG). The payload format is designed to be able to
>    interoperate with existing Bluetooth A2DP devices, to provide high
>    streaming audio quality, interactive audio transmission over the
>    internet, and ultra-low delay coding for jam sessions on the
>    internet. This document contains also a media type registration
>    which specifies the use of the RTP payload format.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-payload-rtp-sbc-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-payload-rtp-sbc-01.txt
> 
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload


From csp@csperkins.org  Mon Dec 19 08:42:15 2011
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-sbc-01.txt
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Christian,

Some comments on this version:

- Section 3 is background, but is written in a way that could be =
interpreted as making a series of normative requirements (e.g., =93SBC =
can use four or eight subbands. The decoder shall support both; the =
encoder shall support at least 8 subbands=94). Be clear whether these =
are normative requirements made by this document, or by the SBC codec =
specification or the SBC A2DP profile.

- The draft uses =93CAN=94 throughout, but this is not an RFC 2119 =
keyword. What is meant?

- Need to mandate the behaviour if the fields in the frame header don=92t =
match the SDP negotiated values. It would be better if these parameters =
were only specified in the SDP, rather than duplicating them in-band and =
in the signalling.

- Media type registration: =93MIME type=94 -> =93Media type=94, since =
this is more general than Internet mail.

- The formatting of the =93optional parameters=94 section of the media =
type registration is unclear. Is this supposed to include the =
capabilities parameter? Is the capabilities parameter really optional? =
Section 7.2 seems to require it.

- The change controller cannot be the AVT working group, since that no =
longer exists

- Section 7.1.1 refers to the =93a=3Drtpmap:=94 attribute for the rate =
and channel parameters, but section 7.1 doesn=92t define these =
parameters. Need to add them in the required parameters section of the =
media type registration.

Colin


On 15 Dec 2011, at 14:51, Christian Hoene wrote:

> Hello,
>=20
> we would like to ask for WGLC if possible.
>=20
> The Bluetooth SBC is a quite nice codec ranging from wideband speech, =
CD
> quality up to distributed ensembles as it has ultra-low delay. The
> complexity is very low and last not least, to the best of my knowledge =
it is
> patent free. We have one implementation of this draft in libopal.
>=20
> With best regards,
>=20
> Christian Hoene
>=20
>=20
>=20
>=20
>> -----Original Message-----
>> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
>> Behalf Of internet-drafts@ietf.org
>> Sent: Thursday, December 15, 2011 3:46 PM
>> To: i-d-announce@ietf.org
>> Cc: payload@ietf.org
>> Subject: [payload] I-D Action: draft-ietf-payload-rtp-sbc-01.txt
>>=20
>>=20
>> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>> This draft is a work item of the Audio/Video Transport Payloads =
Working
>> Group of the IETF.
>>=20
>> 	Title           : RTP Payload Format for Bluetooth's SBC audio =
codec
>> 	Author(s)       : Christian Hoene
>>                          Frans de Bont
>> 	Filename        : draft-ietf-payload-rtp-sbc-01.txt
>> 	Pages           : 24
>> 	Date            : 2011-12-15
>>=20
>>   This document specifies a Real-time Transport Protocol (RTP) =
payload
>>   format to be used for the low complexity subband codec (SBC), which
>>   is the mandatory audio codec of the Advanced Audio Distribution
>>   Profile (A2DP) Specification written by the Bluetooth(r) Special
>>   Interest Group (SIG). The payload format is designed to be able to
>>   interoperate with existing Bluetooth A2DP devices, to provide high
>>   streaming audio quality, interactive audio transmission over the
>>   internet, and ultra-low delay coding for jam sessions on the
>>   internet. This document contains also a media type registration
>>   which specifies the use of the RTP payload format.
>>=20
>>=20
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-payload-rtp-sbc-01.txt
>>=20
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>=20
>> This Internet-Draft can be retrieved at:
>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-payload-rtp-sbc-01.txt
>>=20
>> _______________________________________________
>> payload mailing list
>> payload@ietf.org
>> https://www.ietf.org/mailman/listinfo/payload
>=20
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload



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




From wwwrun@rfc-editor.org  Fri Dec 23 14:31:15 2011
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Subject: [payload] RFC 6469 on RTP Payload Format for DV (IEC 61834) Video
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6469

        Title:      RTP Payload Format for DV 
                    (IEC 61834) Video 
        Author:     K. Kobayashi, K. Mishima,
                    S. Casner, C. Bormann
        Status:     Standards Track
        Stream:     IETF
        Date:       December 2011
        Mailbox:    ikob@riken.jp, 
                    three@sfc.wide.ad.jp, 
                    casner@acm.org,  
                    cabo@tzi.org
        Pages:      18
        Characters: 38041
        Obsoletes:  RFC3189

        I-D Tag:    draft-ietf-payload-rfc3189bis-03.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6469.txt

This document specifies the packetization scheme for encapsulating
the compressed digital video data streams commonly known as "DV" into
a payload format for the Real-Time Transport Protocol (RTP).  This
document obsoletes RFC 3189.  [STANDARDS-TRACK]

This document is a product of the Audio/Video Transport Payloads 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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The RFC Editor Team
Association Management Solutions, LLC



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From: Roni even <Even.roni@huawei.com>
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Hi,


draft-ietf-avt-rtp-evrc-nw-05 has passed Working Group Last Call in the Payload Working Group.





The new registrations are in Section 3.1 of the document. http://tools.ietf.org/html/draft-ietf-avt-rtp-evrc-nw-05#section-9.1


The information is also available below.




Comments on the registration are welcome.

Roni Even
Payload co-chair


This document introduces a new EVRC-NW 'audio' media subtypes.

9.1.  Media Type Registrations
   Following the guidelines in RFC 4855 [8] and RFC 4288 [9], this
   section registers new 'audio' media subtypes for EVRC-NW.

9.1.1.  Registration of Media Type audio/EVRCNW
   Type name: audio
   Subtype names: EVRCNW
   Required parameters: None
   Optional parameters:
   These parameters apply to RTP transfer only.
   mode-set-recv: A subset of EVRC-NW modes.  Possible values are a
   comma separated list of modes from the set {0,1,2,3,4,5,6,7} (see
   Table 2.6.1.2-4 in 3GPP2 C.S0014-D).  A decoder can use this
   attribute to inform an encoder of its preference to operate in a
   specified subset of modes.  Absence of this parameter signals the
   mode set {1,2,3,4,5,6,7}.
   ptime: see RFC 4566 [10].
   maxptime: see RFC 4566.
   maxinterleave: Maximum number for interleaving length (field LLL in
   the Interleaving Octet)[0..7].  The interleaving lengths used in the
   entire session MUST NOT exceed this maximum value.  If not signaled,
   the maxinterleave length MUST be 5.
   silencesupp: see Section 6.1 in RFC 4788.
   dtxmax: see Section 6.1 in RFC 4788.
   dtxmin: see Section 6.1 in RFC 4788.
   hangover: see Section 6.1 in RFC 4788.
   Encoding considerations:
   This media type is framed binary data (see RFC 4288, Section 4.8) and
   is defined for transfer of EVRC-NW encoded data via RTP using the
   Interleaved/Bundled packet format specified in RFC 3558 [6].
   Security considerations: See Section 15.
   Interoperability considerations: None
   Published specification:
   The EVRC-NW vocoder is specified in 3GPP2 C.S0014-D.  The transfer
   method with the Interleaved/Bundled packet format via RTP is
   specified in RFC 3558 [6].  See Section 6 for details for EVRC-NW.
   Applications that use this media type:
   It is expected that many VoIP applications (as well as mobile
   applications) will use this type.
   Additional information:
   The following applies to stored-file transfer methods:
   Magic number: #!EVRCNW\n (see Section 8)
   File extensions: enw, ENW
   Macintosh file type code: None
   Object identifier or OID: None
   EVRC-NW speech frames may also be stored in the file format "3g2"
   defined in 3GPP2 C.S0050-B, which is identified using the media types
   "audio/3gpp2" or "video/3gpp2" registered by RFC 4393 [11].
   Person & email address to contact for further information:
   Zheng Fang <zfang@qualcomm.com<mailto:zfang@qualcomm.com>>
   Intended usage: COMMON
   Restrictions on usage:
   When this media type is used in the context of transfer over RTP, the
   RTP payload format specified in Section 4.1 of RFC 3558 [6] SHALL be
   used.  In all other contexts, the file format defined in Section 8
   SHALL be used.  See Section 6 for details for EVRC-NW.
   Author:
   Zheng Fang <zfang@qualcomm.com<mailto:zfang@qualcomm.com>>
   Change controller:
   IETF Payload working group delegated from the IESG.

9.1.2.  Registration of Media Type audio/EVRCNW0
   Type name: audio
   Subtype names: EVRCNW0
   Required parameters: None
   Optional parameters:
   These parameters apply to RTP transfer only.
   mode-set-recv: A subset of EVRC-NW modes.  Possible values are a
   comma separated list of modes from the set {0,1,2,3,4,5,6,7} (see
   Table 2.6.1.2-4 in 3GPP2 C.S0014-D).  A decoder can use this
   attribute to inform an encoder of its preference to operate in a
   specified subset of modes.  Absence of this parameter signals the
   mode set {1,2,3,4,5,6,7}.
   ptime: see RFC 4566.
   silencesupp: see Section 6.1 in RFC 4788.
   dtxmax: see Section 6.1 in RFC 4788.
   dtxmin: see Section 6.1 in RFC 4788.
   hangover: see Section 6.1 in RFC 4788.
   Encoding considerations:
   This media type is framed binary data (see RFC 4288, Section 4.8) and
   is defined for transfer of EVRC-NW encoded data via RTP using the
   Header-Free packet format specified in RFC 3558 [6].
   Security considerations: See Section 15.
   Interoperability considertaions: None
   Published specification:
   The EVRC-NW vocoder is specified in 3GPP2 C.S0014-D.  The transfer
   method with the Header-Free packet format via RTP is specified in RFC
   3558 [6].
   Applications that use this media type:
   It is expected that many VoIP applications (as well as mobile
   applications) will use this type.
   Additional information: None
   Person & email address to contact for further information:
   Zheng Fang <zfang@qualcomm.com<mailto:zfang@qualcomm.com>>
   Intended usage: COMMON
   Restrictions on usage:
   This media type depends on RTP framing, and hence is only defined for
   transfer via RTP [5], the RTP payload format specified in Section 4.2
   of RFC 3558 [6] SHALL be used.  This media type SHALL NOT be used for
   storage or file transfer, instead audio/EVRCNW SHALL be used.
   Author:
   Zheng Fang <zfang@qualcomm.com<mailto:zfang@qualcomm.com>>
   Change controller:
   IETF Payload working group delegated from the IESG.


9.1.3.  Registration of Media Type audio/EVRCNW1
   Type name: audio
   Subtype names: EVRCNW1
   Required parameters: None
   Optional parameters:
   These parameters apply to RTP transfer only.
   mode-set-recv: A subset of EVRC-NW modes.  Possible values are a
   comma separated list of modes from the set {0,1} (see Table 2.6.1.2-4
   in 3GPP2 C.S0014-D).  A decoder can use this attribute to inform an
   encoder of its preference to operate in a specified subset of modes.
   A value of 0 signals the support for wideband fixed rate (full or
   half rate, depending on the value of 'fixedrate' parameter).  A value
   of 1 signals narroband fixed rate (full or half rate, depending on
   the value of 'fixedrate' parameter).  Absence of this parameter
   signals the mode 1.
   ptime: see RFC 4566.
   maxptime: see RFC 4566.
   fixedrate: Indicates the EVRC-NW rate of the session while in single
   rate operation.  Valid values include: 0.5 and 1, where a value of
   0.5 indicates the 1/2 rate while a value of 1 indicates the full
   rate.  If this parameter is not present, 1/2 rate is assumed.
   silencesupp: see Section 6.1 in RFC 4788.
   dtxmax: see Section 6.1 in RFC 4788.
   dtxmin: see Section 6.1 in RFC 4788.
   hangover: see Section 6.1 in RFC 4788.
   Encoding considerations:
   This media type is framed binary data (see RFC 4288, Section 4.8) and
   is defined for transfer of EVRC-NW encoded data via RTP using the
   Compact Bundled packet format specified in RFC 4788.
   Security considerations: See Section 15
   Interoperability considertaions: None
   Published specification:
   The EVRC-NW vocoder is specified in 3GPP2 C.S0014-D.  The transfer
   method with the Compact Bundled packet format via RTP is specified in
   RFC 4788.
   Applications that use this media type:
   It is expected that many VoIP applications (as well as mobile
   applications) will use this type.
   Additional information: None
   Person & email address to contact for further information:
   Zheng Fang <zfang@qualcomm.com<mailto:zfang@qualcomm.com>>
   Intended usage: COMMON
   Restrictions on usage:
   This media type depends on RTP framing, and hence is only defined for
   transfer via RTP [5], the RTP payload format specified in Section 4
   of RFC 4788 SHALL be used.  This media type SHALL NOT be used for
   storage or file transfer, instead audio/EVRCNW SHALL be used.
   Author:
   Zheng Fang <zfang@qualcomm.com<mailto:zfang@qualcomm.com>>
   Change controller:
   IETF Payload working group delegated from the IESG.

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<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><font size="3" face="Calibri">Hi,</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">
<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />
<o:p><font size="3" face="Calibri">&nbsp;</font></o:p></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><span style="FONT-FAMILY: 'Calibri','sans-serif'; FONT-SIZE: 11pt"><font size="3" face="Consolas">draft-ietf-avt-rtp-evrc-nw-05
</font></span><font size="3"><font face="Consolas">has passed Working Group Last Call in the Payload Working Group.</font></font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><o:p><font size="3" face="Consolas">&nbsp;</font></o:p></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><o:p><font size="3" face="Consolas">&nbsp;</font></o:p></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font size="3" face="Consolas">The new registrations are in Section 3.1 of the document.
</font><a href="http://tools.ietf.org/html/draft-ietf-avt-rtp-evrc-nw-05#section-9.1"><font size="2" face="Tahoma">http://tools.ietf.org/html/draft-ietf-avt-rtp-evrc-nw-05#section-9.1</font></a></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText">&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><font size="3" face="Calibri">The information is also available below.
</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><o:p><font size="3" face="Calibri">&nbsp;</font></o:p></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><span style="FONT-FAMILY: 'Calibri','sans-serif'; FONT-SIZE: 11pt"><o:p>&nbsp;</o:p></span></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><o:p><font size="3" face="Calibri">&nbsp;</font></o:p></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><font size="3" face="Calibri">Comments on the registration are welcome.</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><o:p><font size="3" face="Calibri">&nbsp;</font></o:p></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal" ass="MsoNo" cl!=""><font size="3" face="Calibri"><a></a><font size="3" face="Calibri">Roni</font>&nbsp;Even
</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><font size="3" face="Calibri">Payload co-chair</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><font size="3" face="Calibri"></font>&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><font size="3" face="Calibri"></font>&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">This document introduces a new EVRC-NW 'audio' media subtypes.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">9.1.&nbsp; Media Type Registrations</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Following the guidelines in RFC 4855 [8] and RFC 4288 [9], this<br>
&nbsp;&nbsp; section registers new 'audio' media subtypes for EVRC-NW.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">9.1.1.&nbsp; Registration of Media Type audio/EVRCNW</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Type name: audio</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Subtype names: EVRCNW</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Required parameters: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Optional parameters:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; These parameters apply to RTP transfer only.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; mode-set-recv<a></a>: A subset of EVRC-NW modes.&nbsp; Possible values are a<br>
&nbsp;&nbsp; comma separated list of modes from the set {0,1,2,3,4,5,6,7} (see<br>
&nbsp;&nbsp; Table 2.6.1.2-4 in 3GPP2 C.S0014-D).&nbsp; A decoder can use this<br>
&nbsp;&nbsp; attribute to inform an encoder of its preference to operate in a<br>
&nbsp;&nbsp; specified subset of modes.&nbsp; Absence of this parameter signals the<br>
&nbsp;&nbsp; mode set {1,2,3,4,5,6,7}.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; ptime<a></a>: see RFC 4566 [10].</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; maxptime<a></a>: see RFC 4566.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; maxinterleave<a></a>: Maximum number for interleaving length (field LLL in<br>
&nbsp;&nbsp; the Interleaving Octet)[0..7].&nbsp; The interleaving lengths used in the<br>
&nbsp;&nbsp; entire session MUST NOT exceed this maximum value.&nbsp; If not signaled,<br>
&nbsp;&nbsp; the&nbsp;maxinterleave<a></a> length MUST be 5.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; silencesupp<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; dtxmax<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; dtxmin<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; hangover: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Encoding considerations:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; This media type is framed binary data (see RFC 4288, Section 4.8) and</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; is defined for transfer of EVRC-NW encoded data via RTP using the<br>
&nbsp;&nbsp; Interleaved/Bundled packet format specified in RFC 3558 [6].</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Security considerations: See Section 15.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Interoperability considerations: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Published specification:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; The EVRC-NW&nbsp;vocoder<a></a> is specified in 3GPP2 C.S0014-D.&nbsp; The transfer<br>
&nbsp;&nbsp; method with the Interleaved/Bundled packet format via RTP is<br>
&nbsp;&nbsp; specified in RFC 3558 [6].&nbsp; See Section 6 for details for EVRC-NW.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Applications that use this media type:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; It is expected that many VoIP applications (as well as mobile<br>
&nbsp;&nbsp; applications) will use this type.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Additional information:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; The following applies to stored-file transfer methods:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Magic number: #!EVRCNW\n (see Section 8)</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; File extensions: enw<a></a>, ENW</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Macintosh file type code: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Object identifier or OID: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; EVRC-NW speech frames may also be stored in the file format &quot;3g2&quot;<br>
&nbsp;&nbsp; defined in 3GPP2 C.S0050-B, which is identified using the media types<br>
&nbsp;&nbsp; &quot;audio/3gpp2&quot; or &quot;video/3gpp2&quot; registered by RFC 4393 [11].</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Person &amp; email address to contact for further information:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp;&nbsp;Zheng<a></a> Fang &lt;<a href="mailto:zfang@qualcomm.com">zfang@qualcomm.com</a><a></a>&gt;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Intended usage: COMMON</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Restrictions on usage:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; When this media type is used in the context of transfer over RTP, the<br>
&nbsp;&nbsp; RTP payload format specified in Section 4.1 of RFC 3558 [6] SHALL be<br>
&nbsp;&nbsp; used.&nbsp; In all other contexts, the file format defined in Section 8<br>
&nbsp;&nbsp; SHALL be used.&nbsp; See Section 6 for details for EVRC-NW.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Author:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp;&nbsp;Zheng<a></a> Fang &lt;<a href="mailto:zfang@qualcomm.com">zfang@qualcomm.com</a><a></a>&gt;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Change controller:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; IETF Payload working group delegated from the IESG.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">9.1.2.&nbsp; Registration of Media Type audio/EVRCNW0</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Type name: audio</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Subtype names: EVRCNW0</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Required parameters: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Optional parameters:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; These parameters apply to RTP transfer only.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; mode-set-recv<a></a>: A subset of EVRC-NW modes.&nbsp; Possible values are a<br>
&nbsp;&nbsp; comma separated list of modes from the set {0,1,2,3,4,5,6,7} (see<br>
&nbsp;&nbsp; Table 2.6.1.2-4 in 3GPP2 C.S0014-D).&nbsp; A decoder can use this<br>
&nbsp;&nbsp; attribute to inform an encoder of its preference to operate in a<br>
&nbsp;&nbsp; specified subset of modes.&nbsp; Absence of this parameter signals the<br>
&nbsp;&nbsp; mode set {1,2,3,4,5,6,7}.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; ptime<a></a>: see RFC 4566.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; silencesupp<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; dtxmax<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; dtxmin<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; hangover: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Encoding considerations:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; This media type is framed binary data (see RFC 4288, Section 4.8) and<br>
&nbsp;&nbsp; is defined for transfer of EVRC-NW encoded data via RTP using the<br>
&nbsp;&nbsp; Header-Free packet format specified in RFC 3558 [6].</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Security considerations: See Section 15.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Interoperability considertaions<a></a>: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Published specification:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; The EVRC-NW&nbsp;vocoder<a></a> is specified in 3GPP2 C.S0014-D.&nbsp; The transfer</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; method with the Header-Free packet format via RTP is specified in RFC<br>
&nbsp;&nbsp; 3558 [6].</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Applications that use this media type:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; It is expected that many VoIP applications (as well as mobile<br>
&nbsp;&nbsp; applications) will use this type.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Additional information: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Person &amp; email address to contact for further information:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp;&nbsp;Zheng<a></a> Fang &lt;<a href="mailto:zfang@qualcomm.com">zfang@qualcomm.com</a><a></a>&gt;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Intended usage: COMMON</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Restrictions on usage:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; This media type depends on RTP framing, and hence is only defined for<br>
&nbsp;&nbsp; transfer via RTP [5], the RTP payload format specified in Section 4.2<br>
&nbsp;&nbsp; of RFC 3558 [6] SHALL be used.&nbsp; This media type SHALL NOT be used for<br>
&nbsp;&nbsp; storage or file transfer, instead audio/EVRCNW SHALL be used.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Author:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp;&nbsp;Zheng<a></a> Fang &lt;<a href="mailto:zfang@qualcomm.com">zfang@qualcomm.com</a><a></a>&gt;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Change controller:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; IETF Payload working group delegated from the IESG.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">9.1.3.&nbsp; Registration of Media Type audio/EVRCNW1</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Type name: audio</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Subtype names: EVRCNW1</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Required parameters: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Optional parameters:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; These parameters apply to RTP transfer only.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; mode-set-recv<a></a>: A subset of EVRC-NW modes.&nbsp; Possible values are a<br>
&nbsp;&nbsp; comma separated list of modes from the set {0,1} (see Table 2.6.1.2-4<br>
&nbsp;&nbsp; in 3GPP2 C.S0014-D).&nbsp; A decoder can use this attribute to inform an<br>
&nbsp;&nbsp; encoder of its preference to operate in a specified subset of modes.<br>
&nbsp;&nbsp; A value of 0 signals the support for wideband fixed rate (full or</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; half rate, depending on the value of 'fixedrate<a></a>' parameter).&nbsp; A value<br>
&nbsp;&nbsp; of 1 signals&nbsp;narroband<a></a> fixed rate (full or half rate, depending on<br>
&nbsp;&nbsp; the value of 'fixedrate<a></a>' parameter).&nbsp; Absence of this parameter<br>
&nbsp;&nbsp; signals the mode 1.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; ptime<a></a>: see RFC 4566.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; maxptime<a></a>: see RFC 4566.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; fixedrate<a></a>: Indicates the EVRC-NW rate of the session while in single<br>
&nbsp;&nbsp; rate operation.&nbsp; Valid values include: 0.5 and 1, where a value of<br>
&nbsp;&nbsp; 0.5 indicates the 1/2 rate while a value of 1 indicates the full<br>
&nbsp;&nbsp; rate.&nbsp; If this parameter is not present, 1/2 rate is assumed.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; silencesupp<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; dtxmax<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; dtxmin<a></a>: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; hangover: see Section 6.1 in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Encoding considerations:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; This media type is framed binary data (see RFC 4288, Section 4.8) and<br>
&nbsp;&nbsp; is defined for transfer of EVRC-NW encoded data via RTP using the<br>
&nbsp;&nbsp; Compact Bundled packet format specified in RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Security considerations: See Section 15</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Interoperability considertaions<a></a>: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Published specification:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; The EVRC-NW&nbsp;vocoder<a></a> is specified in 3GPP2 C.S0014-D.&nbsp; The transfer<br>
&nbsp;&nbsp; method with the Compact Bundled packet format via RTP is specified in<br>
&nbsp;&nbsp; RFC 4788.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Applications that use this media type:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; It is expected that many VoIP applications (as well as mobile<br>
&nbsp;&nbsp; applications) will use this type.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Additional information: None</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Person &amp; email address to contact for further information:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp;&nbsp;Zheng<a></a> Fang &lt;<a href="mailto:zfang@qualcomm.com">zfang@qualcomm.com</a><a></a>&gt;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Intended usage: COMMON</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Restrictions on usage:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; This media type depends on RTP framing, and hence is only defined for<br>
&nbsp;&nbsp; transfer via RTP [5], the RTP payload format specified in Section 4<br>
&nbsp;&nbsp; of RFC 4788 SHALL be used.&nbsp; This media type SHALL NOT be used for<br>
&nbsp;&nbsp; storage or file transfer, instead audio/EVRCNW SHALL be used.</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Author:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp;&nbsp;Zheng<a></a> Fang &lt;<a href="mailto:zfang@qualcomm.com">zfang@qualcomm.com</a><a></a>&gt;</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; Change controller:</p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal">&nbsp;&nbsp; IETF Payload working group delegated from the IESG.<br>
</p>
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