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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, "'Fredrik Bergholtz'" <fredrik.bergholtz@sverigesradio.se>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com>, <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com>, <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com>
In-Reply-To: <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com>
Date: Sat, 3 Jan 2015 15:57:33 +0530
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Archived-At: http://mailarchive.ietf.org/arch/msg/payload/7hn7VtvySHxseu9rM8XsmcpCrRU
Cc: payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hi all,

 

I  understand that bandwidth "b" parameter has lot of variants like AS, TIAS
and usage but all those parameters are media level and not codec specific.
The bandwidth allocation is going to be happen in the media level in the
network or end point based on the media level and not on the codec specific
in case the multiple codec is negotiated. In most of the implementation, the
multiple codec (Say Ex: G.711 & Opus) is going to be negotiated rather than
single codec. The codec specific bandwidth implementation complicates these
bandwidth calculation still further which is applicable to maxaveragebitrate
parameter in opus.

 

I could not understand why the generic parameter should not used or generic
parameter has to be enhanced which meets Opus requirement rather than opus
specific parameter. 

 

Thanks

Partha

 

From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Ali C. Begen
(abegen)
Sent: Wednesday, December 31, 2014 4:57 PM
To: Fredrik Bergholtz
Cc: payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate

 

Authors 

Please chime in and lets discuss s quick solution for this on the list. Once
agreed we can ship this draft. 

-acbegen

 

From: Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se>
Sent: Dec 31, 2014 11:42 AM
To: Ali C. Begen (abegen)
Cc: Jean-Marc Valin <jmvalin@mozilla.com>;Fredrik Bergholtz;payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate

 

Yes, I do see a strong need for specifying the bitrate. 

In the broadcasting use case we have to make sure that we get the best audio
quality that is possible for the network that is currently used but never
try to send more data than the particular link is capable of. For this
reason, specifying a maximum bit rate is required.

In addition we often have the requirement that the quality is constant
rather than the best possible. To avoid as many audible changes in the audio
as possible over time we will often use constant bit rate and then it
becomes even more important to be able to specify what bit rate should be. 

We will use the Opus codec over various network links with different
capabilities in terms of bit rate, delay variation, absolute delay, etc and
we will typically use the SDP to signal the parameters that best fit both
the current network and the current type of radio production. 

Best regards
Fredrik Bergholtz
Swedish Radio


> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)" <abegen@cisco.com>
wrote:
> 
> My understanding is that the b= line in the SDP is quite a subjective
parameter. I have seen people use it differently over the past several
years. So, it is a bit difficult to infer much accuracy from that line.
> 
> OTOH, if someone feels like Opus needs to specify some sort of bitrate
variability in the SDP, it can do so in an optional (opus) parameter.
Obviously rules around how to set that value and what to do with that value
need to be clearly specified.
> 
> Fredrik (or anyone else in the WG), do you see a strong need for this? If
so, now is the time to mention that.
> 
> -acbegen
> 
> 
>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin <jmvalin@mozilla.com> wrote:
>> 
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>> 
>> I have to say this is not something I understand really well. Does
>> anyone else here have an opinion on how to specify bit-rate?
>> 
>>    Jean-Marc
>> 
>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
>>> Hi.
>>> 
>>> 
>>> 
>>> I read in the latest draft that you have removed the maxaveragebitrate
>>> parameter from the SDP signalling for Opus. I think that this parameter
>>> is useful and that it provides added value over the bandwidth modifiers
>>> added by RFC 3556.
>>> 
>>> 
>>> 
>>> I must admit that I haven't read all of RFC 3556 but as I understand it,
>>> a b= line are not in any way related to an actual format of a media
>>> description. This means that with this mechanism, only one bitrate
>>> specification can be sent with each media description. Different coding
>>> algorithms have different constraints on the allowed bitrate. For
>>> example, I might want to send an SDP offer with both Enhanced apt-X at
>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now, specifying 576kbit/s via
>>> a b= line works for apt-X but makes no sense for Opus and vice versa. I
>>> think that the a=fmtp line is the correct place to specify the desired
>>> bitrate.
>>> 
>>> 
>>> 
>>> For the broadcasting use-case, it is often more important to have a
>>> constant and predictable audio quality and network usage than mere
>>> audibility at any cost.
>>> 
>>> 
>>> 
>>> Best regards,
>>> 
>>> Fredrik Bergholtz
>>> 
>>> Swedish Radio
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> payload mailing list
>>> payload@ietf.org
>>> https://www.ietf.org/mailman/listinfo/payload
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>> _______________________________________________
>> payload mailing list
>> payload@ietf.org
>> https://www.ietf.org/mailman/listinfo/payload
> 


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi all,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I&nbsp; understand that bandwidth &#8220;b&#8221; =
parameter has
lot of variants like AS, TIAS and usage but all those parameters are =
media
level and not codec specific. The bandwidth allocation is going to be =
happen in
the media level in the network or end point based on the media level and =
not on
the codec specific&nbsp; in case the multiple codec is negotiated. In =
most of
the implementation, the multiple codec (Say Ex: G.711 &amp; Opus) is =
going to
be negotiated rather than single codec. The codec specific bandwidth
implementation complicates these bandwidth calculation still further =
which is
applicable to maxaveragebitrate parameter in opus.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I could not understand why the generic parameter should =
not used
or generic parameter has to be enhanced which meets Opus requirement =
rather
than opus specific parameter. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
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color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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<div>

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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> payload
[mailto:payload-bounces@ietf.org] <b>On Behalf Of </b>Ali C. Begen =
(abegen)<br>
<b>Sent:</b> Wednesday, December 31, 2014 4:57 PM<br>
<b>To:</b> Fredrik Bergholtz<br>
<b>Cc:</b> payload@ietf.org<br>
<b>Subject:</b> Re: [payload] draft-ietf-payload-rtp-opus-05: =
maxaveragebitrate<o:p></o:p></span></p>

</div>

</div>

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

<div>

<div>

<p><span style=3D'font-family:"Calibri","sans-serif"'>Authors =
<o:p></o:p></span></p>

<p><span style=3D'font-family:"Calibri","sans-serif"'>Please chime in =
and lets
discuss s quick solution for this on the list. Once agreed we can ship =
this
draft. <o:p></o:p></span></p>

<div id=3D"x_signature-x">

<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>-acbegen<o:p></o:p></span></=
p>

</div>

</div>

<div id=3D"x_quoted_header">

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

<div style=3D'border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>From:</span=
></b><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'> Fredrik =
Bergholtz
&lt;fredrik.bergholtz@sverigesradio.se&gt;<br>
<b>Sent:</b> Dec 31, 2014 11:42 AM<br>
<b>To:</b> Ali C. Begen (abegen)<br>
<b>Cc:</b> Jean-Marc Valin &lt;jmvalin@mozilla.com&gt;;Fredrik
Bergholtz;payload@ietf.org<br>
<b>Subject:</b> Re: [payload] draft-ietf-payload-rtp-opus-05: =
maxaveragebitrate</span><o:p></o:p></p>

</div>

</div>

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

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt'>Yes, I do see a =
strong need
for specifying the bitrate. <br>
<br>
In the broadcasting use case we have to make sure that we get the best =
audio
quality that is possible for the network that is currently used but =
never try
to send more data than the particular link is capable of. For this =
reason,
specifying a maximum bit rate is required.<br>
<br>
In addition we often have the requirement that the quality is constant =
rather
than the best possible. To avoid as many audible changes in the audio as
possible over time we will often use constant bit rate and then it =
becomes even
more important to be able to specify what bit rate should be. <br>
<br>
We will use the Opus codec over various network links with different
capabilities in terms of bit rate, delay variation, absolute delay, etc =
and we
will typically use the SDP to signal the parameters that best fit both =
the
current network and the current type of radio production. <br>
<br>
Best regards<br>
Fredrik Bergholtz<br>
Swedish Radio<br>
<br>
<br>
&gt; On 31 dec 2014, at 05:57, &quot;Ali C. Begen (abegen)&quot;
&lt;abegen@cisco.com&gt; wrote:<br>
&gt; <br>
&gt; My understanding is that the b=3D line in the SDP is quite a =
subjective
parameter. I have seen people use it differently over the past several =
years.
So, it is a bit difficult to infer much accuracy from that line.<br>
&gt; <br>
&gt; OTOH, if someone feels like Opus needs to specify some sort of =
bitrate
variability in the SDP, it can do so in an optional (opus) parameter. =
Obviously
rules around how to set that value and what to do with that value need =
to be
clearly specified.<br>
&gt; <br>
&gt; Fredrik (or anyone else in the WG), do you see a strong need for =
this? If
so, now is the time to mention that.<br>
&gt; <br>
&gt; -acbegen<br>
&gt; <br>
&gt; <br>
&gt;&gt; On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
&lt;jmvalin@mozilla.com&gt; wrote:<br>
&gt;&gt; <br>
&gt;&gt; -----BEGIN PGP SIGNED MESSAGE-----<br>
&gt;&gt; Hash: SHA1<br>
&gt;&gt; <br>
&gt;&gt; I have to say this is not something I understand really well. =
Does<br>
&gt;&gt; anyone else here have an opinion on how to specify =
bit-rate?<br>
&gt;&gt; <br>
&gt;&gt;&nbsp;&nbsp;&nbsp; Jean-Marc<br>
&gt;&gt; <br>
&gt;&gt;&gt; On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:<br>
&gt;&gt;&gt; Hi.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; I read in the latest draft that you have removed the
maxaveragebitrate<br>
&gt;&gt;&gt; parameter from the SDP signalling for Opus. I think that =
this parameter<br>
&gt;&gt;&gt; is useful and that it provides added value over the =
bandwidth
modifiers<br>
&gt;&gt;&gt; added by RFC 3556.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; I must admit that I haven&#8217;t read all of RFC 3556 but =
as I
understand it,<br>
&gt;&gt;&gt; a b=3D line are not in any way related to an actual format =
of a
media<br>
&gt;&gt;&gt; description. This means that with this mechanism, only one =
bitrate<br>
&gt;&gt;&gt; specification can be sent with each media description. =
Different
coding<br>
&gt;&gt;&gt; algorithms have different constraints on the allowed =
bitrate. For<br>
&gt;&gt;&gt; example, I might want to send an SDP offer with both =
Enhanced
apt-X at<br>
&gt;&gt;&gt; 576 kbit/s and Opus at 510kbit/s with cbr. Now, specifying
576kbit/s via<br>
&gt;&gt;&gt; a b=3D line works for apt-X but makes no sense for Opus and =
vice
versa. I<br>
&gt;&gt;&gt; think that the a=3Dfmtp line is the correct place to =
specify the
desired<br>
&gt;&gt;&gt; bitrate.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; For the broadcasting use-case, it is often more important =
to have
a<br>
&gt;&gt;&gt; constant and predictable audio quality and network usage =
than mere<br>
&gt;&gt;&gt; audibility at any cost.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Best regards,<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Fredrik Bergholtz<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Swedish Radio<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; payload mailing list<br>
&gt;&gt;&gt; payload@ietf.org<br>
&gt;&gt;&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/payload">https://www.ietf.o=
rg/mailman/listinfo/payload</a><br>
&gt;&gt; -----BEGIN PGP SIGNATURE-----<br>
&gt;&gt; Version: GnuPG v1<br>
&gt;&gt; <br>
&gt;&gt; =
iQEcBAEBAgAGBQJUij/kAAoJEJ6/8sItn9q9V6EIALgEC6Iv+nZuu2BU8useaV4Z<br>
&gt;&gt; =
jLRz97vCEdcmj60jMzELC2gSKLpVW3AMcbP7gtPjm0odeLFEMvoweUFytnVt/cd6<br>
&gt;&gt; =
bCadjHMPAOSNXrPtbmS57JdlcmWHnsrJIVrsV/Ipn6o44FWL2BN5g/LnwlSTKs/N<br>
&gt;&gt; =
OtrjAdLj8b/GsmVS6VTTjTkygL1bp1Wsry5oAGXIPwPqFgQwuGMAxv/Wf6eEu7c3<br>
&gt;&gt; =
eoZzBKfBDM0bC8L+RCy6REYKHvSjPbyPAH0bQvkkKHfSc6I5ZLOZspvOHBmHvQFu<br>
&gt;&gt; =
DNj4DHCShHPnRRknVnXCXX9ddSC4/bLQSyZ9+WI9622N5lkre8+VnUIGx9iAr3I=3D<br>
&gt;&gt; =3DZuN4<br>
&gt;&gt; -----END PGP SIGNATURE-----<br>
&gt;&gt; <br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; payload mailing list<br>
&gt;&gt; payload@ietf.org<br>
&gt;&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/payload">https://www.ietf.o=
rg/mailman/listinfo/payload</a><br>
&gt; <o:p></o:p></span></p>

</div>

</div>

</div>

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From: Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se>
To: Parthasarathi R <partha@parthasarathi.co.in>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
Thread-Index: AdATp89X++FTuPj7QpWp8hHcyPj53AB992kAA8ORFIAADAt6MQABiR0AAJTPRIAACqfAww==
Date: Sat, 3 Jan 2015 14:32:38 +0000
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References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com>, <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com>, <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com>, <006201d0273f$e708a230$b519e690$@co.in>
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Cc: "payload@ietf.org" <payload@ietf.org>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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--_000_1CDC64C8B1304FB7B73A6F894CB729F0sverigesradiose_
Content-Type: text/plain; charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

Hi.

One thought om the general media level bit rate specification is that it ca=
n never be anything other than an approximate value. Different codecs defin=
e different constraints on the bit rate. I think that the easiest way to sp=
ecify exact bit rates would be via the fmtp attribute.

This can be compared with the ptime
attribute - as it is defined today, only approximate packet times can be si=
gnalled in SDP. For the broadcast use case this is not acceptable and for t=
hat reason, the European Broadcasting Union (EBU) has defined an SDP extens=
ion that allows signalling of a ptime value for each codec. There is no RFC=
 draft for this extension and it is not yet registered with IANA or IETF bu=
t it is starting to be used by manufacturers of broadcasting equipment. The=
 extension is known as "ACIP Profiles" and can be found at the EBU web site=
. But I digress.

I don't think that forcing encoders to approximate the nearest allowed valu=
e for the selected codec is less complicated than interpreting an exact val=
ue. In addition, it would add a non-deterministic aspect, thus the resultin=
g bit rate may not be what the user expected.

Best regards
Fredrik Bergholtz
Swedish Radio

On 3 jan 2015, at 11:27, "Parthasarathi R" <partha@parthasarathi.co.in<mail=
to:partha@parthasarathi.co.in>> wrote:

Hi all,

I  understand that bandwidth =93b=94 parameter has lot of variants like AS,=
 TIAS and usage but all those parameters are media level and not codec spec=
ific. The bandwidth allocation is going to be happen in the media level in =
the network or end point based on the media level and not on the codec spec=
ific  in case the multiple codec is negotiated. In most of the implementati=
on, the multiple codec (Say Ex: G.711 & Opus) is going to be negotiated rat=
her than single codec. The codec specific bandwidth implementation complica=
tes these bandwidth calculation still further which is applicable to maxave=
ragebitrate parameter in opus.

I could not understand why the generic parameter should not used or generic=
 parameter has to be enhanced which meets Opus requirement rather than opus=
 specific parameter.

Thanks
Partha

From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Ali C. Begen (=
abegen)
Sent: Wednesday, December 31, 2014 4:57 PM
To: Fredrik Bergholtz
Cc: payload@ietf.org<mailto:payload@ietf.org>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate


Authors

Please chime in and lets discuss s quick solution for this on the list. Onc=
e agreed we can ship this draft.
-acbegen

From: Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se<mailto:fredrik.=
bergholtz@sverigesradio.se>>
Sent: Dec 31, 2014 11:42 AM
To: Ali C. Begen (abegen)
Cc: Jean-Marc Valin <jmvalin@mozilla.com<mailto:jmvalin@mozilla.com>>;Fredr=
ik Bergholtz;payload@ietf.org<mailto:payload@ietf.org>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate

Yes, I do see a strong need for specifying the bitrate.

In the broadcasting use case we have to make sure that we get the best audi=
o quality that is possible for the network that is currently used but never=
 try to send more data than the particular link is capable of. For this rea=
son, specifying a maximum bit rate is required.

In addition we often have the requirement that the quality is constant rath=
er than the best possible. To avoid as many audible changes in the audio as=
 possible over time we will often use constant bit rate and then it becomes=
 even more important to be able to specify what bit rate should be.

We will use the Opus codec over various network links with different capabi=
lities in terms of bit rate, delay variation, absolute delay, etc and we wi=
ll typically use the SDP to signal the parameters that best fit both the cu=
rrent network and the current type of radio production.

Best regards
Fredrik Bergholtz
Swedish Radio


> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)" <abegen@cisco.com<mailt=
o:abegen@cisco.com>> wrote:
>
> My understanding is that the b=3D line in the SDP is quite a subjective p=
arameter. I have seen people use it differently over the past several years=
. So, it is a bit difficult to infer much accuracy from that line.
>
> OTOH, if someone feels like Opus needs to specify some sort of bitrate va=
riability in the SDP, it can do so in an optional (opus) parameter. Obvious=
ly rules around how to set that value and what to do with that value need t=
o be clearly specified.
>
> Fredrik (or anyone else in the WG), do you see a strong need for this? If=
 so, now is the time to mention that.
>
> -acbegen
>
>
>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin <jmvalin@mozilla.com<mailto=
:jmvalin@mozilla.com>> wrote:
>>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> I have to say this is not something I understand really well. Does
>> anyone else here have an opinion on how to specify bit-rate?
>>
>>    Jean-Marc
>>
>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
>>> Hi.
>>>
>>>
>>>
>>> I read in the latest draft that you have removed the maxaveragebitrate
>>> parameter from the SDP signalling for Opus. I think that this parameter
>>> is useful and that it provides added value over the bandwidth modifiers
>>> added by RFC 3556.
>>>
>>>
>>>
>>> I must admit that I haven=92t read all of RFC 3556 but as I understand =
it,
>>> a b=3D line are not in any way related to an actual format of a media
>>> description. This means that with this mechanism, only one bitrate
>>> specification can be sent with each media description. Different coding
>>> algorithms have different constraints on the allowed bitrate. For
>>> example, I might want to send an SDP offer with both Enhanced apt-X at
>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now, specifying 576kbit/s vi=
a
>>> a b=3D line works for apt-X but makes no sense for Opus and vice versa.=
 I
>>> think that the a=3Dfmtp line is the correct place to specify the desire=
d
>>> bitrate.
>>>
>>>
>>>
>>> For the broadcasting use-case, it is often more important to have a
>>> constant and predictable audio quality and network usage than mere
>>> audibility at any cost.
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Fredrik Bergholtz
>>>
>>> Swedish Radio
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> payload mailing list
>>> payload@ietf.org<mailto:payload@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/payload
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>>
>> _______________________________________________
>> payload mailing list
>> payload@ietf.org<mailto:payload@ietf.org>
>> https://www.ietf.org/mailman/listinfo/payload
>

--_000_1CDC64C8B1304FB7B73A6F894CB729F0sverigesradiose_
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252">
</head>
<body dir=3D"auto">
<div>Hi.&nbsp;</div>
<div><br>
</div>
<div>One thought om the general media level bit rate specification is that =
it can never be anything other than an approximate value. Different codecs =
define different constraints on the bit rate. I think that the easiest way =
to specify exact bit rates would
 be via the fmtp attribute.&nbsp;</div>
<div><br>
</div>
<div>This can be compared with the ptime<br>
attribute - as it is defined today, only approximate packet times can be si=
gnalled in SDP. For the broadcast use case this is not acceptable and for t=
hat reason, the European Broadcasting Union (EBU) has defined an SDP extens=
ion that allows signalling of a
 ptime value for each codec. There is no RFC draft for this extension and i=
t is not yet registered with IANA or IETF but it is starting to be used by =
manufacturers of broadcasting equipment. The extension is known as &quot;AC=
IP Profiles&quot; and can be found at the
 EBU web site. But I digress.&nbsp;<span id=3D"signature"></span></div>
<div><br>
</div>
<div>I don't think that forcing encoders to approximate the nearest allowed=
 value for the selected codec is less complicated than interpreting an exac=
t value. In addition, it would add a non-deterministic aspect, thus the res=
ulting bit rate may not be what
 the user expected.&nbsp;</div>
<div><br>
</div>
<div>Best regards</div>
<div>Fredrik Bergholtz</div>
<div>Swedish Radio</div>
<div><br>
On 3 jan 2015, at 11:27, &quot;Parthasarathi R&quot; &lt;<a href=3D"mailto:=
partha@parthasarathi.co.in">partha@parthasarathi.co.in</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
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<div class=3D"Section1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">Hi all,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">I&nbsp; understand that bandwidth =93b=94 parameter has lot =
of variants like AS, TIAS and usage but all those parameters are media leve=
l and not codec specific. The
 bandwidth allocation is going to be happen in the media level in the netwo=
rk or end point based on the media level and not on the codec specific&nbsp=
; in case the multiple codec is negotiated. In most of the implementation, =
the multiple codec (Say Ex: G.711 &amp; Opus)
 is going to be negotiated rather than single codec. The codec specific ban=
dwidth implementation complicates these bandwidth calculation still further=
 which is applicable to maxaveragebitrate parameter in opus.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">I could not understand why the generic parameter should not =
used or generic parameter has to be enhanced which meets Opus requirement r=
ather than opus specific
 parameter. <o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">Thanks<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D">Partha<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> payload =
[<a href=3D"mailto:payload-bounces@ietf.org">mailto:payload-bounces@ietf.or=
g</a>]
<b>On Behalf Of </b>Ali C. Begen (abegen)<br>
<b>Sent:</b> Wednesday, December 31, 2014 4:57 PM<br>
<b>To:</b> Fredrik Bergholtz<br>
<b>Cc:</b> <a href=3D"mailto:payload@ietf.org">payload@ietf.org</a><br>
<b>Subject:</b> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebit=
rate<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">A=
uthors <o:p></o:p></span></p>
<p><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">P=
lease chime in and lets discuss s quick solution for this on the list. Once=
 agreed we can ship this draft.
<o:p></o:p></span></p>
<div id=3D"x_signature-x">
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;">-acbegen<o:p></o:p></span></p>
</div>
</div>
<div id=3D"x_quoted_header">
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"> Fredri=
k Bergholtz &lt;<a href=3D"mailto:fredrik.bergholtz@sverigesradio.se">fredr=
ik.bergholtz@sverigesradio.se</a>&gt;<br>
<b>Sent:</b> Dec 31, 2014 11:42 AM<br>
<b>To:</b> Ali C. Begen (abegen)<br>
<b>Cc:</b> Jean-Marc Valin &lt;<a href=3D"mailto:jmvalin@mozilla.com">jmval=
in@mozilla.com</a>&gt;;Fredrik Bergholtz;<a href=3D"mailto:payload@ietf.org=
">payload@ietf.org</a><br>
<b>Subject:</b> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebit=
rate</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt">Yes, I do see a str=
ong need for specifying the bitrate.
<br>
<br>
In the broadcasting use case we have to make sure that we get the best audi=
o quality that is possible for the network that is currently used but never=
 try to send more data than the particular link is capable of. For this rea=
son, specifying a maximum bit rate
 is required.<br>
<br>
In addition we often have the requirement that the quality is constant rath=
er than the best possible. To avoid as many audible changes in the audio as=
 possible over time we will often use constant bit rate and then it becomes=
 even more important to be able
 to specify what bit rate should be. <br>
<br>
We will use the Opus codec over various network links with different capabi=
lities in terms of bit rate, delay variation, absolute delay, etc and we wi=
ll typically use the SDP to signal the parameters that best fit both the cu=
rrent network and the current type
 of radio production. <br>
<br>
Best regards<br>
Fredrik Bergholtz<br>
Swedish Radio<br>
<br>
<br>
&gt; On 31 dec 2014, at 05:57, &quot;Ali C. Begen (abegen)&quot; &lt;<a hre=
f=3D"mailto:abegen@cisco.com">abegen@cisco.com</a>&gt; wrote:<br>
&gt; <br>
&gt; My understanding is that the b=3D line in the SDP is quite a subjectiv=
e parameter. I have seen people use it differently over the past several ye=
ars. So, it is a bit difficult to infer much accuracy from that line.<br>
&gt; <br>
&gt; OTOH, if someone feels like Opus needs to specify some sort of bitrate=
 variability in the SDP, it can do so in an optional (opus) parameter. Obvi=
ously rules around how to set that value and what to do with that value nee=
d to be clearly specified.<br>
&gt; <br>
&gt; Fredrik (or anyone else in the WG), do you see a strong need for this?=
 If so, now is the time to mention that.<br>
&gt; <br>
&gt; -acbegen<br>
&gt; <br>
&gt; <br>
&gt;&gt; On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin &lt;<a href=3D"mailto=
:jmvalin@mozilla.com">jmvalin@mozilla.com</a>&gt; wrote:<br>
&gt;&gt; <br>
&gt;&gt; -----BEGIN PGP SIGNED MESSAGE-----<br>
&gt;&gt; Hash: SHA1<br>
&gt;&gt; <br>
&gt;&gt; I have to say this is not something I understand really well. Does=
<br>
&gt;&gt; anyone else here have an opinion on how to specify bit-rate?<br>
&gt;&gt; <br>
&gt;&gt;&nbsp;&nbsp;&nbsp; Jean-Marc<br>
&gt;&gt; <br>
&gt;&gt;&gt; On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:<br>
&gt;&gt;&gt; Hi.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; I read in the latest draft that you have removed the maxaverag=
ebitrate<br>
&gt;&gt;&gt; parameter from the SDP signalling for Opus. I think that this =
parameter<br>
&gt;&gt;&gt; is useful and that it provides added value over the bandwidth =
modifiers<br>
&gt;&gt;&gt; added by RFC 3556.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; I must admit that I haven=92t read all of RFC 3556 but as I un=
derstand it,<br>
&gt;&gt;&gt; a b=3D line are not in any way related to an actual format of =
a media<br>
&gt;&gt;&gt; description. This means that with this mechanism, only one bit=
rate<br>
&gt;&gt;&gt; specification can be sent with each media description. Differe=
nt coding<br>
&gt;&gt;&gt; algorithms have different constraints on the allowed bitrate. =
For<br>
&gt;&gt;&gt; example, I might want to send an SDP offer with both Enhanced =
apt-X at<br>
&gt;&gt;&gt; 576 kbit/s and Opus at 510kbit/s with cbr. Now, specifying 576=
kbit/s via<br>
&gt;&gt;&gt; a b=3D line works for apt-X but makes no sense for Opus and vi=
ce versa. I<br>
&gt;&gt;&gt; think that the a=3Dfmtp line is the correct place to specify t=
he desired<br>
&gt;&gt;&gt; bitrate.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; For the broadcasting use-case, it is often more important to h=
ave a<br>
&gt;&gt;&gt; constant and predictable audio quality and network usage than =
mere<br>
&gt;&gt;&gt; audibility at any cost.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Best regards,<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Fredrik Bergholtz<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Swedish Radio<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; payload mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:payload@ietf.org">payload@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/payload">http=
s://www.ietf.org/mailman/listinfo/payload</a><br>
&gt;&gt; -----BEGIN PGP SIGNATURE-----<br>
&gt;&gt; Version: GnuPG v1<br>
&gt;&gt; <br>
&gt;&gt; iQEcBAEBAgAGBQJUij/kAAoJEJ6/8sItn9q9V6EIALgEC6Iv&#43;nZuu2BU8useaV=
4Z<br>
&gt;&gt; jLRz97vCEdcmj60jMzELC2gSKLpVW3AMcbP7gtPjm0odeLFEMvoweUFytnVt/cd6<b=
r>
&gt;&gt; bCadjHMPAOSNXrPtbmS57JdlcmWHnsrJIVrsV/Ipn6o44FWL2BN5g/LnwlSTKs/N<b=
r>
&gt;&gt; OtrjAdLj8b/GsmVS6VTTjTkygL1bp1Wsry5oAGXIPwPqFgQwuGMAxv/Wf6eEu7c3<b=
r>
&gt;&gt; eoZzBKfBDM0bC8L&#43;RCy6REYKHvSjPbyPAH0bQvkkKHfSc6I5ZLOZspvOHBmHvQ=
Fu<br>
&gt;&gt; DNj4DHCShHPnRRknVnXCXX9ddSC4/bLQSyZ9&#43;WI9622N5lkre8&#43;VnUIGx9=
iAr3I=3D<br>
&gt;&gt; =3DZuN4<br>
&gt;&gt; -----END PGP SIGNATURE-----<br>
&gt;&gt; <br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; payload mailing list<br>
&gt;&gt; <a href=3D"mailto:payload@ietf.org">payload@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/payload">https://=
www.ietf.org/mailman/listinfo/payload</a><br>
&gt; <o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</blockquote>
</body>
</html>

--_000_1CDC64C8B1304FB7B73A6F894CB729F0sverigesradiose_--


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From: Jean-Marc Valin <jmvalin@jmvalin.ca>
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Setting the bitrate isn't exactly a new problem here. Opus isn't the
first codec to support more than one rate so how are the other codecs
(audio and video) handling bitrate signaling?

Cheers,

	Jean-Marc

On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
> Hi.
> 
> One thought om the general media level bit rate specification is
> that it can never be anything other than an approximate value.
> Different codecs define different constraints on the bit rate. I
> think that the easiest way to specify exact bit rates would be via
> the fmtp attribute.
> 
> This can be compared with the ptime attribute - as it is defined
> today, only approximate packet times can be signalled in SDP. For
> the broadcast use case this is not acceptable and for that reason,
> the European Broadcasting Union (EBU) has defined an SDP extension
> that allows signalling of a ptime value for each codec. There is no
> RFC draft for this extension and it is not yet registered with IANA
> or IETF but it is starting to be used by manufacturers of 
> broadcasting equipment. The extension is known as "ACIP Profiles"
> and can be found at the EBU web site. But I digress.
> 
> I don't think that forcing encoders to approximate the nearest
> allowed value for the selected codec is less complicated than
> interpreting an exact value. In addition, it would add a
> non-deterministic aspect, thus the resulting bit rate may not be
> what the user expected.
> 
> Best regards Fredrik Bergholtz Swedish Radio
> 
> On 3 jan 2015, at 11:27, "Parthasarathi R"
> <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>
> wrote:
> 
>> Hi all,
>> 
>> 
>> 
>> I  understand that bandwidth “b” parameter has lot of variants
>> like AS, TIAS and usage but all those parameters are media level
>> and not codec specific. The bandwidth allocation is going to be
>> happen in the media level in the network or end point based on
>> the media level and not on the codec specific  in case the
>> multiple codec is negotiated. In most of the implementation, the
>> multiple codec (Say Ex: G.711 & Opus) is going to be negotiated
>> rather than single codec. The codec specific bandwidth
>> implementation complicates these bandwidth calculation still
>> further which is applicable to maxaveragebitrate parameter in
>> opus.
>> 
>> 
>> 
>> I could not understand why the generic parameter should not used
>> or generic parameter has to be enhanced which meets Opus
>> requirement rather than opus specific parameter.
>> 
>> 
>> 
>> Thanks
>> 
>> Partha
>> 
>> 
>> 
>> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of
>> *Ali C. Begen (abegen) *Sent:* Wednesday, December 31, 2014 4:57
>> PM *To:* Fredrik Bergholtz *Cc:* payload@ietf.org
>> <mailto:payload@ietf.org> *Subject:* Re: [payload]
>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>> 
>> 
>> 
>> Authors
>> 
>> Please chime in and lets discuss s quick solution for this on
>> the list. Once agreed we can ship this draft.
>> 
>> -acbegen
>> 
>> 
>> 
>> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se 
>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31, 2014
>> 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc Valin
>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>;Fredrik
>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> *Subject:*
>> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>> 
>> 
>> 
>> Yes, I do see a strong need for specifying the bitrate.
>> 
>> In the broadcasting use case we have to make sure that we get the
>> best audio quality that is possible for the network that is
>> currently used but never try to send more data than the
>> particular link is capable of. For this reason, specifying a
>> maximum bit rate is required.
>> 
>> In addition we often have the requirement that the quality is
>> constant rather than the best possible. To avoid as many audible
>> changes in the audio as possible over time we will often use
>> constant bit rate and then it becomes even more important to be
>> able to specify what bit rate should be.
>> 
>> We will use the Opus codec over various network links with
>> different capabilities in terms of bit rate, delay variation,
>> absolute delay, etc and we will typically use the SDP to signal
>> the parameters that best fit both the current network and the
>> current type of radio production.
>> 
>> Best regards Fredrik Bergholtz Swedish Radio
>> 
>> 
>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>> 
>>> My understanding is that the b= line in the SDP is quite a
>>> subjective parameter. I have seen people use it differently
>>> over the past several years. So, it is a bit difficult to infer
>>> much accuracy from that line.
>>> 
>>> OTOH, if someone feels like Opus needs to specify some sort of
>>> bitrate variability in the SDP, it can do so in an optional
>>> (opus) parameter. Obviously rules around how to set that value
>>> and what to do with that value need to be clearly specified.
>>> 
>>> Fredrik (or anyone else in the WG), do you see a strong need
>>> for this? If so, now is the time to mention that.
>>> 
>>> -acbegen
>>> 
>>> 
>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>> wrote:
>>>> 
> I have to say this is not something I understand really well. Does 
> anyone else here have an opinion on how to specify bit-rate?
> 
> Jean-Marc
> 
>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> I read in the latest draft that you have removed the
>>>>>> maxaveragebitrate parameter from the SDP signalling for
>>>>>> Opus. I think that this parameter is useful and that it
>>>>>> provides added value over the bandwidth modifiers added
>>>>>> by RFC 3556.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> I must admit that I haven’t read all of RFC 3556 but as I
>>>>>> understand it, a b= line are not in any way related to an
>>>>>> actual format of a media description. This means that
>>>>>> with this mechanism, only one bitrate specification can
>>>>>> be sent with each media description. Different coding 
>>>>>> algorithms have different constraints on the allowed
>>>>>> bitrate. For example, I might want to send an SDP offer
>>>>>> with both Enhanced apt-X at 576 kbit/s and Opus at
>>>>>> 510kbit/s with cbr. Now, specifying 576kbit/s via a b=
>>>>>> line works for apt-X but makes no sense for Opus and vice
>>>>>> versa. I think that the a=fmtp line is the correct place
>>>>>> to specify the desired bitrate.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> For the broadcasting use-case, it is often more important
>>>>>> to have a constant and predictable audio quality and
>>>>>> network usage than mere audibility at any cost.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Best regards,
>>>>>> 
>>>>>> Fredrik Bergholtz
>>>>>> 
>>>>>> Swedish Radio
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________ payload
>>>>>> mailing list payload@ietf.org <mailto:payload@ietf.org> 
>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>> 
>>>> _______________________________________________ payload
>>>> mailing list payload@ietf.org <mailto:payload@ietf.org> 
>>>> https://www.ietf.org/mailman/listinfo/payload
>>> 
>> 
> 
> 
> _______________________________________________ payload mailing
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
> 
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From nobody Sun Jan  4 14:06:28 2015
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Jean-Marc Valin <jmvalin@jmvalin.ca>, Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se>, Parthasarathi R <partha@parthasarathi.co.in>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Date: Sun, 4 Jan 2015 22:05:30 +0000
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References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com>
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Some other codecs have fmtp parameters for maximum bitrate explicitly
(e.g. max-br in H264 / H264-SVC / H265) and/or implicitly (e.g.
profile-level-id in H264 / H264-SVC / H265 / AAC; config in AAC).

Other notable codecs that lack this include VP8 and VP9.

This is a general problem which may become more important with complex
applications that include simulcast and scalable media within the same
media description (m=3D line). The SDP b=3D parameter is clearly insufficie=
nt
for expressing multiple different maximum bitrates for each payload type,
unless we define a new bandwidth modifier which includes explicit PT
bindings. Payload formats that include fmtp parameters for maximum bitrate
can avoid this problem, so I think it is advisable to add this to Opus,
VP8 and VP9.

Mo

On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Setting the bitrate isn't exactly a new problem here. Opus isn't the
first codec to support more than one rate so how are the other codecs
(audio and video) handling bitrate signaling?

Cheers,

	Jean-Marc

On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
> Hi.
>=20
> One thought om the general media level bit rate specification is
> that it can never be anything other than an approximate value.
> Different codecs define different constraints on the bit rate. I
> think that the easiest way to specify exact bit rates would be via
> the fmtp attribute.
>=20
> This can be compared with the ptime attribute - as it is defined
> today, only approximate packet times can be signalled in SDP. For
> the broadcast use case this is not acceptable and for that reason,
> the European Broadcasting Union (EBU) has defined an SDP extension
> that allows signalling of a ptime value for each codec. There is no
> RFC draft for this extension and it is not yet registered with IANA
> or IETF but it is starting to be used by manufacturers of
> broadcasting equipment. The extension is known as "ACIP Profiles"
> and can be found at the EBU web site. But I digress.
>=20
> I don't think that forcing encoders to approximate the nearest
> allowed value for the selected codec is less complicated than
> interpreting an exact value. In addition, it would add a
> non-deterministic aspect, thus the resulting bit rate may not be
> what the user expected.
>=20
> Best regards Fredrik Bergholtz Swedish Radio
>=20
> On 3 jan 2015, at 11:27, "Parthasarathi R"
> <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>
> wrote:
>=20
>> Hi all,
>>=20
>>=20
>>=20
>> I  understand that bandwidth =B3b=B2 parameter has lot of variants
>> like AS, TIAS and usage but all those parameters are media level
>> and not codec specific. The bandwidth allocation is going to be
>> happen in the media level in the network or end point based on
>> the media level and not on the codec specific  in case the
>> multiple codec is negotiated. In most of the implementation, the
>> multiple codec (Say Ex: G.711 & Opus) is going to be negotiated
>> rather than single codec. The codec specific bandwidth
>> implementation complicates these bandwidth calculation still
>> further which is applicable to maxaveragebitrate parameter in
>> opus.
>>=20
>>=20
>>=20
>> I could not understand why the generic parameter should not used
>> or generic parameter has to be enhanced which meets Opus
>> requirement rather than opus specific parameter.
>>=20
>>=20
>>=20
>> Thanks
>>=20
>> Partha
>>=20
>>=20
>>=20
>> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of
>> *Ali C. Begen (abegen) *Sent:* Wednesday, December 31, 2014 4:57
>> PM *To:* Fredrik Bergholtz *Cc:* payload@ietf.org
>> <mailto:payload@ietf.org> *Subject:* Re: [payload]
>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>=20
>>=20
>>=20
>> Authors
>>=20
>> Please chime in and lets discuss s quick solution for this on
>> the list. Once agreed we can ship this draft.
>>=20
>> -acbegen
>>=20
>>=20
>>=20
>> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se
>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31, 2014
>> 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc Valin
>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>;Fredrik
>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> *Subject:*
>> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>=20
>>=20
>>=20
>> Yes, I do see a strong need for specifying the bitrate.
>>=20
>> In the broadcasting use case we have to make sure that we get the
>> best audio quality that is possible for the network that is
>> currently used but never try to send more data than the
>> particular link is capable of. For this reason, specifying a
>> maximum bit rate is required.
>>=20
>> In addition we often have the requirement that the quality is
>> constant rather than the best possible. To avoid as many audible
>> changes in the audio as possible over time we will often use
>> constant bit rate and then it becomes even more important to be
>> able to specify what bit rate should be.
>>=20
>> We will use the Opus codec over various network links with
>> different capabilities in terms of bit rate, delay variation,
>> absolute delay, etc and we will typically use the SDP to signal
>> the parameters that best fit both the current network and the
>> current type of radio production.
>>=20
>> Best regards Fredrik Bergholtz Swedish Radio
>>=20
>>=20
>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>=20
>>> My understanding is that the b=3D line in the SDP is quite a
>>> subjective parameter. I have seen people use it differently
>>> over the past several years. So, it is a bit difficult to infer
>>> much accuracy from that line.
>>>=20
>>> OTOH, if someone feels like Opus needs to specify some sort of
>>> bitrate variability in the SDP, it can do so in an optional
>>> (opus) parameter. Obviously rules around how to set that value
>>> and what to do with that value need to be clearly specified.
>>>=20
>>> Fredrik (or anyone else in the WG), do you see a strong need
>>> for this? If so, now is the time to mention that.
>>>=20
>>> -acbegen
>>>=20
>>>=20
>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>> wrote:
>>>>=20
> I have to say this is not something I understand really well. Does
> anyone else here have an opinion on how to specify bit-rate?
>=20
> Jean-Marc
>=20
>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> I read in the latest draft that you have removed the
>>>>>> maxaveragebitrate parameter from the SDP signalling for
>>>>>> Opus. I think that this parameter is useful and that it
>>>>>> provides added value over the bandwidth modifiers added
>>>>>> by RFC 3556.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> I must admit that I haven=B9t read all of RFC 3556 but as I
>>>>>> understand it, a b=3D line are not in any way related to an
>>>>>> actual format of a media description. This means that
>>>>>> with this mechanism, only one bitrate specification can
>>>>>> be sent with each media description. Different coding
>>>>>> algorithms have different constraints on the allowed
>>>>>> bitrate. For example, I might want to send an SDP offer
>>>>>> with both Enhanced apt-X at 576 kbit/s and Opus at
>>>>>> 510kbit/s with cbr. Now, specifying 576kbit/s via a b=3D
>>>>>> line works for apt-X but makes no sense for Opus and vice
>>>>>> versa. I think that the a=3Dfmtp line is the correct place
>>>>>> to specify the desired bitrate.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> For the broadcasting use-case, it is often more important
>>>>>> to have a constant and predictable audio quality and
>>>>>> network usage than mere audibility at any cost.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> Best regards,
>>>>>>=20
>>>>>> Fredrik Bergholtz
>>>>>>=20
>>>>>> Swedish Radio
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> _______________________________________________ payload
>>>>>> mailing list payload@ietf.org <mailto:payload@ietf.org>
>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>=20
>>>> _______________________________________________ payload
>>>> mailing list payload@ietf.org <mailto:payload@ietf.org>
>>>> https://www.ietf.org/mailman/listinfo/payload
>>>=20
>>=20
>=20
>=20
> _______________________________________________ payload mailing
> list payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20
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_______________________________________________
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

So would everyone be happy if I just revert to using maxaveragebitrate
as in draft 04?

	Jean-Marc


On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote:
> Some other codecs have fmtp parameters for maximum bitrate
> explicitly (e.g. max-br in H264 / H264-SVC / H265) and/or
> implicitly (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;
> config in AAC).
> 
> Other notable codecs that lack this include VP8 and VP9.
> 
> This is a general problem which may become more important with
> complex applications that include simulcast and scalable media
> within the same media description (m= line). The SDP b= parameter
> is clearly insufficient for expressing multiple different maximum
> bitrates for each payload type, unless we define a new bandwidth
> modifier which includes explicit PT bindings. Payload formats that
> include fmtp parameters for maximum bitrate can avoid this problem,
> so I think it is advisable to add this to Opus, VP8 and VP9.
> 
> Mo
> 
> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:
> 
> Setting the bitrate isn't exactly a new problem here. Opus isn't
> the first codec to support more than one rate so how are the other
> codecs (audio and video) handling bitrate signaling?
> 
> Cheers,
> 
> Jean-Marc
> 
> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
>> Hi.
> 
>> One thought om the general media level bit rate specification is 
>> that it can never be anything other than an approximate value. 
>> Different codecs define different constraints on the bit rate. I 
>> think that the easiest way to specify exact bit rates would be
>> via the fmtp attribute.
> 
>> This can be compared with the ptime attribute - as it is defined 
>> today, only approximate packet times can be signalled in SDP.
>> For the broadcast use case this is not acceptable and for that
>> reason, the European Broadcasting Union (EBU) has defined an SDP
>> extension that allows signalling of a ptime value for each codec.
>> There is no RFC draft for this extension and it is not yet
>> registered with IANA or IETF but it is starting to be used by
>> manufacturers of broadcasting equipment. The extension is known
>> as "ACIP Profiles" and can be found at the EBU web site. But I
>> digress.
> 
>> I don't think that forcing encoders to approximate the nearest 
>> allowed value for the selected codec is less complicated than 
>> interpreting an exact value. In addition, it would add a 
>> non-deterministic aspect, thus the resulting bit rate may not be 
>> what the user expected.
> 
>> Best regards Fredrik Bergholtz Swedish Radio
> 
>> On 3 jan 2015, at 11:27, "Parthasarathi R" 
>> <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>> 
>> wrote:
> 
>>> Hi all,
>>> 
>>> 
>>> 
>>> I  understand that bandwidth ³b² parameter has lot of variants 
>>> like AS, TIAS and usage but all those parameters are media
>>> level and not codec specific. The bandwidth allocation is going
>>> to be happen in the media level in the network or end point
>>> based on the media level and not on the codec specific  in case
>>> the multiple codec is negotiated. In most of the
>>> implementation, the multiple codec (Say Ex: G.711 & Opus) is
>>> going to be negotiated rather than single codec. The codec
>>> specific bandwidth implementation complicates these bandwidth
>>> calculation still further which is applicable to
>>> maxaveragebitrate parameter in opus.
>>> 
>>> 
>>> 
>>> I could not understand why the generic parameter should not
>>> used or generic parameter has to be enhanced which meets Opus 
>>> requirement rather than opus specific parameter.
>>> 
>>> 
>>> 
>>> Thanks
>>> 
>>> Partha
>>> 
>>> 
>>> 
>>> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of 
>>> *Ali C. Begen (abegen) *Sent:* Wednesday, December 31, 2014
>>> 4:57 PM *To:* Fredrik Bergholtz *Cc:* payload@ietf.org 
>>> <mailto:payload@ietf.org> *Subject:* Re: [payload] 
>>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>> 
>>> 
>>> 
>>> Authors
>>> 
>>> Please chime in and lets discuss s quick solution for this on 
>>> the list. Once agreed we can ship this draft.
>>> 
>>> -acbegen
>>> 
>>> 
>>> 
>>> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se 
>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31,
>>> 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc
>>> Valin <jmvalin@mozilla.com
>>> <mailto:jmvalin@mozilla.com>>;Fredrik 
>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
>>> maxaveragebitrate
>>> 
>>> 
>>> 
>>> Yes, I do see a strong need for specifying the bitrate.
>>> 
>>> In the broadcasting use case we have to make sure that we get
>>> the best audio quality that is possible for the network that
>>> is currently used but never try to send more data than the 
>>> particular link is capable of. For this reason, specifying a 
>>> maximum bit rate is required.
>>> 
>>> In addition we often have the requirement that the quality is 
>>> constant rather than the best possible. To avoid as many
>>> audible changes in the audio as possible over time we will
>>> often use constant bit rate and then it becomes even more
>>> important to be able to specify what bit rate should be.
>>> 
>>> We will use the Opus codec over various network links with 
>>> different capabilities in terms of bit rate, delay variation, 
>>> absolute delay, etc and we will typically use the SDP to
>>> signal the parameters that best fit both the current network
>>> and the current type of radio production.
>>> 
>>> Best regards Fredrik Bergholtz Swedish Radio
>>> 
>>> 
>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)" 
>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>> 
>>>> My understanding is that the b= line in the SDP is quite a 
>>>> subjective parameter. I have seen people use it differently 
>>>> over the past several years. So, it is a bit difficult to
>>>> infer much accuracy from that line.
>>>> 
>>>> OTOH, if someone feels like Opus needs to specify some sort
>>>> of bitrate variability in the SDP, it can do so in an
>>>> optional (opus) parameter. Obviously rules around how to set
>>>> that value and what to do with that value need to be clearly
>>>> specified.
>>>> 
>>>> Fredrik (or anyone else in the WG), do you see a strong need 
>>>> for this? If so, now is the time to mention that.
>>>> 
>>>> -acbegen
>>>> 
>>>> 
>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin 
>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>> wrote:
>>>>> 
>> I have to say this is not something I understand really well.
>> Does anyone else here have an opinion on how to specify
>> bit-rate?
> 
>> Jean-Marc
> 
>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> I read in the latest draft that you have removed the 
>>>>>>> maxaveragebitrate parameter from the SDP signalling
>>>>>>> for Opus. I think that this parameter is useful and
>>>>>>> that it provides added value over the bandwidth
>>>>>>> modifiers added by RFC 3556.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> I must admit that I haven¹t read all of RFC 3556 but as
>>>>>>> I understand it, a b= line are not in any way related
>>>>>>> to an actual format of a media description. This means
>>>>>>> that with this mechanism, only one bitrate
>>>>>>> specification can be sent with each media description.
>>>>>>> Different coding algorithms have different constraints
>>>>>>> on the allowed bitrate. For example, I might want to
>>>>>>> send an SDP offer with both Enhanced apt-X at 576
>>>>>>> kbit/s and Opus at 510kbit/s with cbr. Now, specifying
>>>>>>> 576kbit/s via a b= line works for apt-X but makes no
>>>>>>> sense for Opus and vice versa. I think that the a=fmtp
>>>>>>> line is the correct place to specify the desired
>>>>>>> bitrate.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> For the broadcasting use-case, it is often more
>>>>>>> important to have a constant and predictable audio
>>>>>>> quality and network usage than mere audibility at any
>>>>>>> cost.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Best regards,
>>>>>>> 
>>>>>>> Fredrik Bergholtz
>>>>>>> 
>>>>>>> Swedish Radio
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> payload mailing list payload@ietf.org
>>>>>>> <mailto:payload@ietf.org> 
>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>> 
>>>>> _______________________________________________ payload 
>>>>> mailing list payload@ietf.org <mailto:payload@ietf.org> 
>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>> 
>>> 
> 
> 
>> _______________________________________________ payload mailing 
>> list payload@ietf.org 
>> https://www.ietf.org/mailman/listinfo/payload
> 
> 
> _______________________________________________ payload mailing
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
> 
> _______________________________________________ payload mailing
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
> 
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From nobody Sun Jan  4 15:35:24 2015
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Mo Zanaty \(mzanaty\)'" <mzanaty@cisco.com>, "'Jean-Marc Valin'" <jmvalin@jmvalin.ca>, "'Fredrik Bergholtz'" <fredrik.bergholtz@sverigesradio.se>, "'Parthasarathi R'" <partha@parthasarathi.co.in>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com>
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Date: Mon, 5 Jan 2015 01:35:04 +0200
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hi,
For audio/voice there are RFC5577 and RFC4749 that has a bitrate =
parameter
while RFC5404 uses the b=3D parameter



Roni Even
As individual

> -----Original Message-----
> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Mo Zanaty
> (mzanaty)
> Sent: 05 January, 2015 12:06 AM
> To: Jean-Marc Valin; Fredrik Bergholtz; Parthasarathi R
> Cc: payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: =
maxaveragebitrate
>=20
> Some other codecs have fmtp parameters for maximum bitrate explicitly
(e.g.
> max-br in H264 / H264-SVC / H265) and/or implicitly (e.g.
> profile-level-id in H264 / H264-SVC / H265 / AAC; config in AAC).
>=20
> Other notable codecs that lack this include VP8 and VP9.
>=20
> This is a general problem which may become more important with complex
> applications that include simulcast and scalable media within the same
media
> description (m=3D line). The SDP b=3D parameter is clearly =
insufficient for
> expressing multiple different maximum bitrates for each payload type,
unless
> we define a new bandwidth modifier which includes explicit PT =
bindings.
> Payload formats that include fmtp parameters for maximum bitrate can =
avoid
> this problem, so I think it is advisable to add this to Opus,
> VP8 and VP9.
>=20
> Mo
>=20
> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> Setting the bitrate isn't exactly a new problem here. Opus isn't the =
first
codec
> to support more than one rate so how are the other codecs (audio and
video)
> handling bitrate signaling?
>=20
> Cheers,
>=20
> 	Jean-Marc
>=20
> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
> > Hi.
> >
> > One thought om the general media level bit rate specification is =
that
> > it can never be anything other than an approximate value.
> > Different codecs define different constraints on the bit rate. I =
think
> > that the easiest way to specify exact bit rates would be via the =
fmtp
> > attribute.
> >
> > This can be compared with the ptime attribute - as it is defined
> > today, only approximate packet times can be signalled in SDP. For =
the
> > broadcast use case this is not acceptable and for that reason, the
> > European Broadcasting Union (EBU) has defined an SDP extension that
> > allows signalling of a ptime value for each codec. There is no RFC
> > draft for this extension and it is not yet registered with IANA or
> > IETF but it is starting to be used by manufacturers of broadcasting
> > equipment. The extension is known as "ACIP Profiles"
> > and can be found at the EBU web site. But I digress.
> >
> > I don't think that forcing encoders to approximate the nearest =
allowed
> > value for the selected codec is less complicated than interpreting =
an
> > exact value. In addition, it would add a non-deterministic aspect,
> > thus the resulting bit rate may not be what the user expected.
> >
> > Best regards Fredrik Bergholtz Swedish Radio
> >
> > On 3 jan 2015, at 11:27, "Parthasarathi R"
> > <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>
> > wrote:
> >
> >> Hi all,
> >>
> >>
> >>
> >> I  understand that bandwidth =B3b=B2 parameter has lot of variants =
like
> >> AS, TIAS and usage but all those parameters are media level and not
> >> codec specific. The bandwidth allocation is going to be happen in =
the
> >> media level in the network or end point based on the media level =
and
> >> not on the codec specific  in case the multiple codec is =
negotiated.
> >> In most of the implementation, the multiple codec (Say Ex: G.711 &
> >> Opus) is going to be negotiated rather than single codec. The codec
> >> specific bandwidth implementation complicates these bandwidth
> >> calculation still further which is applicable to maxaveragebitrate
> >> parameter in opus.
> >>
> >>
> >>
> >> I could not understand why the generic parameter should not used or
> >> generic parameter has to be enhanced which meets Opus requirement
> >> rather than opus specific parameter.
> >>
> >>
> >>
> >> Thanks
> >>
> >> Partha
> >>
> >>
> >>
> >> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of *Ali
> >> C. Begen (abegen) *Sent:* Wednesday, December 31, 2014 4:57 PM =
*To:*
> >> Fredrik Bergholtz *Cc:* payload@ietf.org <mailto:payload@ietf.org>
> >> *Subject:* Re: [payload]
> >> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
> >>
> >>
> >>
> >> Authors
> >>
> >> Please chime in and lets discuss s quick solution for this on the
> >> list. Once agreed we can ship this draft.
> >>
> >> -acbegen
> >>
> >>
> >>
> >> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se
> >> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31, 2014
> >> 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc Valin
> >> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>;Fredrik
> >> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> *Subject:*
> >> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
> >>
> >>
> >>
> >> Yes, I do see a strong need for specifying the bitrate.
> >>
> >> In the broadcasting use case we have to make sure that we get the
> >> best audio quality that is possible for the network that is =
currently
> >> used but never try to send more data than the particular link is
> >> capable of. For this reason, specifying a maximum bit rate is
> >> required.
> >>
> >> In addition we often have the requirement that the quality is
> >> constant rather than the best possible. To avoid as many audible
> >> changes in the audio as possible over time we will often use =
constant
> >> bit rate and then it becomes even more important to be able to
> >> specify what bit rate should be.
> >>
> >> We will use the Opus codec over various network links with =
different
> >> capabilities in terms of bit rate, delay variation, absolute delay,
> >> etc and we will typically use the SDP to signal the parameters that
> >> best fit both the current network and the current type of radio
> >> production.
> >>
> >> Best regards Fredrik Bergholtz Swedish Radio
> >>
> >>
> >>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
> >>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
> >>>
> >>> My understanding is that the b=3D line in the SDP is quite a
> >>> subjective parameter. I have seen people use it differently over =
the
> >>> past several years. So, it is a bit difficult to infer much =
accuracy
> >>> from that line.
> >>>
> >>> OTOH, if someone feels like Opus needs to specify some sort of
> >>> bitrate variability in the SDP, it can do so in an optional
> >>> (opus) parameter. Obviously rules around how to set that value and
> >>> what to do with that value need to be clearly specified.
> >>>
> >>> Fredrik (or anyone else in the WG), do you see a strong need for
> >>> this? If so, now is the time to mention that.
> >>>
> >>> -acbegen
> >>>
> >>>
> >>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin <jmvalin@mozilla.com
> >>>> <mailto:jmvalin@mozilla.com>> wrote:
> >>>>
> > I have to say this is not something I understand really well. Does
> > anyone else here have an opinion on how to specify bit-rate?
> >
> > Jean-Marc
> >
> >>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> I read in the latest draft that you have removed the
> >>>>>> maxaveragebitrate parameter from the SDP signalling for Opus. I
> >>>>>> think that this parameter is useful and that it provides added
> >>>>>> value over the bandwidth modifiers added by RFC 3556.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> I must admit that I haven=B9t read all of RFC 3556 but as I
> >>>>>> understand it, a b=3D line are not in any way related to an =
actual
> >>>>>> format of a media description. This means that with this
> >>>>>> mechanism, only one bitrate specification can be sent with each
> >>>>>> media description. Different coding algorithms have different
> >>>>>> constraints on the allowed bitrate. For example, I might want =
to
> >>>>>> send an SDP offer with both Enhanced apt-X at 576 kbit/s and =
Opus
> >>>>>> at 510kbit/s with cbr. Now, specifying 576kbit/s via a b=3D =
line
> >>>>>> works for apt-X but makes no sense for Opus and vice versa. I
> >>>>>> think that the a=3Dfmtp line is the correct place to specify =
the
> >>>>>> desired bitrate.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> For the broadcasting use-case, it is often more important to =
have
> >>>>>> a constant and predictable audio quality and network usage than
> >>>>>> mere audibility at any cost.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Best regards,
> >>>>>>
> >>>>>> Fredrik Bergholtz
> >>>>>>
> >>>>>> Swedish Radio
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________ payload
> mailing
> >>>>>> list payload@ietf.org <mailto:payload@ietf.org>
> >>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>
> >>>> _______________________________________________ payload
> mailing
> >>>> list payload@ietf.org <mailto:payload@ietf.org>
> >>>> https://www.ietf.org/mailman/listinfo/payload
> >>>
> >>
> >
> >
> > _______________________________________________ payload mailing list
> > payload@ietf.org https://www.ietf.org/mailman/listinfo/payload
> >
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>=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


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From: Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se>
To: Jean-Marc Valin <jmvalin@jmvalin.ca>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Date: Mon, 5 Jan 2015 10:32:42 +0000
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References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com>,<54A9C9D0.8050002@mozilla.com>
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Cc: Jean-Marc Valin <jmvalin@jmvalin.ca>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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I think that re-instating maxaveragebitrate would do it for me.=20

/Fredrik


> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca> wrote:
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> So would everyone be happy if I just revert to using maxaveragebitrate
> as in draft 04?
>=20
>    Jean-Marc
>=20
>=20
>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote:
>> Some other codecs have fmtp parameters for maximum bitrate
>> explicitly (e.g. max-br in H264 / H264-SVC / H265) and/or
>> implicitly (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;
>> config in AAC).
>>=20
>> Other notable codecs that lack this include VP8 and VP9.
>>=20
>> This is a general problem which may become more important with
>> complex applications that include simulcast and scalable media
>> within the same media description (m=3D line). The SDP b=3D parameter
>> is clearly insufficient for expressing multiple different maximum
>> bitrates for each payload type, unless we define a new bandwidth
>> modifier which includes explicit PT bindings. Payload formats that
>> include fmtp parameters for maximum bitrate can avoid this problem,
>> so I think it is advisable to add this to Opus, VP8 and VP9.
>>=20
>> Mo
>>=20
>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:
>>=20
>> Setting the bitrate isn't exactly a new problem here. Opus isn't
>> the first codec to support more than one rate so how are the other
>> codecs (audio and video) handling bitrate signaling?
>>=20
>> Cheers,
>>=20
>> Jean-Marc
>>=20
>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
>>> Hi.
>>=20
>>> One thought om the general media level bit rate specification is=20
>>> that it can never be anything other than an approximate value.=20
>>> Different codecs define different constraints on the bit rate. I=20
>>> think that the easiest way to specify exact bit rates would be
>>> via the fmtp attribute.
>>=20
>>> This can be compared with the ptime attribute - as it is defined=20
>>> today, only approximate packet times can be signalled in SDP.
>>> For the broadcast use case this is not acceptable and for that
>>> reason, the European Broadcasting Union (EBU) has defined an SDP
>>> extension that allows signalling of a ptime value for each codec.
>>> There is no RFC draft for this extension and it is not yet
>>> registered with IANA or IETF but it is starting to be used by
>>> manufacturers of broadcasting equipment. The extension is known
>>> as "ACIP Profiles" and can be found at the EBU web site. But I
>>> digress.
>>=20
>>> I don't think that forcing encoders to approximate the nearest=20
>>> allowed value for the selected codec is less complicated than=20
>>> interpreting an exact value. In addition, it would add a=20
>>> non-deterministic aspect, thus the resulting bit rate may not be=20
>>> what the user expected.
>>=20
>>> Best regards Fredrik Bergholtz Swedish Radio
>>=20
>>> On 3 jan 2015, at 11:27, "Parthasarathi R"=20
>>> <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>=20
>>> wrote:
>>=20
>>>> Hi all,
>>>>=20
>>>>=20
>>>>=20
>>>> I  understand that bandwidth =B3b=B2 parameter has lot of variants=20
>>>> like AS, TIAS and usage but all those parameters are media
>>>> level and not codec specific. The bandwidth allocation is going
>>>> to be happen in the media level in the network or end point
>>>> based on the media level and not on the codec specific  in case
>>>> the multiple codec is negotiated. In most of the
>>>> implementation, the multiple codec (Say Ex: G.711 & Opus) is
>>>> going to be negotiated rather than single codec. The codec
>>>> specific bandwidth implementation complicates these bandwidth
>>>> calculation still further which is applicable to
>>>> maxaveragebitrate parameter in opus.
>>>>=20
>>>>=20
>>>>=20
>>>> I could not understand why the generic parameter should not
>>>> used or generic parameter has to be enhanced which meets Opus=20
>>>> requirement rather than opus specific parameter.
>>>>=20
>>>>=20
>>>>=20
>>>> Thanks
>>>>=20
>>>> Partha
>>>>=20
>>>>=20
>>>>=20
>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of=20
>>>> *Ali C. Begen (abegen) *Sent:* Wednesday, December 31, 2014
>>>> 4:57 PM *To:* Fredrik Bergholtz *Cc:* payload@ietf.org=20
>>>> <mailto:payload@ietf.org> *Subject:* Re: [payload]=20
>>>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>>>=20
>>>>=20
>>>>=20
>>>> Authors
>>>>=20
>>>> Please chime in and lets discuss s quick solution for this on=20
>>>> the list. Once agreed we can ship this draft.
>>>>=20
>>>> -acbegen
>>>>=20
>>>>=20
>>>>=20
>>>> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se=20
>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31,
>>>> 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc
>>>> Valin <jmvalin@mozilla.com
>>>> <mailto:jmvalin@mozilla.com>>;Fredrik=20
>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
>>>> maxaveragebitrate
>>>>=20
>>>>=20
>>>>=20
>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>=20
>>>> In the broadcasting use case we have to make sure that we get
>>>> the best audio quality that is possible for the network that
>>>> is currently used but never try to send more data than the=20
>>>> particular link is capable of. For this reason, specifying a=20
>>>> maximum bit rate is required.
>>>>=20
>>>> In addition we often have the requirement that the quality is=20
>>>> constant rather than the best possible. To avoid as many
>>>> audible changes in the audio as possible over time we will
>>>> often use constant bit rate and then it becomes even more
>>>> important to be able to specify what bit rate should be.
>>>>=20
>>>> We will use the Opus codec over various network links with=20
>>>> different capabilities in terms of bit rate, delay variation,=20
>>>> absolute delay, etc and we will typically use the SDP to
>>>> signal the parameters that best fit both the current network
>>>> and the current type of radio production.
>>>>=20
>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>=20
>>>>=20
>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"=20
>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>=20
>>>>> My understanding is that the b=3D line in the SDP is quite a=20
>>>>> subjective parameter. I have seen people use it differently=20
>>>>> over the past several years. So, it is a bit difficult to
>>>>> infer much accuracy from that line.
>>>>>=20
>>>>> OTOH, if someone feels like Opus needs to specify some sort
>>>>> of bitrate variability in the SDP, it can do so in an
>>>>> optional (opus) parameter. Obviously rules around how to set
>>>>> that value and what to do with that value need to be clearly
>>>>> specified.
>>>>>=20
>>>>> Fredrik (or anyone else in the WG), do you see a strong need=20
>>>>> for this? If so, now is the time to mention that.
>>>>>=20
>>>>> -acbegen
>>>>>=20
>>>>>=20
>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin=20
>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>> wrote:
>>> I have to say this is not something I understand really well.
>>> Does anyone else here have an opinion on how to specify
>>> bit-rate?
>>=20
>>> Jean-Marc
>>=20
>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I read in the latest draft that you have removed the=20
>>>>>>>> maxaveragebitrate parameter from the SDP signalling
>>>>>>>> for Opus. I think that this parameter is useful and
>>>>>>>> that it provides added value over the bandwidth
>>>>>>>> modifiers added by RFC 3556.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I must admit that I haven=B9t read all of RFC 3556 but as
>>>>>>>> I understand it, a b=3D line are not in any way related
>>>>>>>> to an actual format of a media description. This means
>>>>>>>> that with this mechanism, only one bitrate
>>>>>>>> specification can be sent with each media description.
>>>>>>>> Different coding algorithms have different constraints
>>>>>>>> on the allowed bitrate. For example, I might want to
>>>>>>>> send an SDP offer with both Enhanced apt-X at 576
>>>>>>>> kbit/s and Opus at 510kbit/s with cbr. Now, specifying
>>>>>>>> 576kbit/s via a b=3D line works for apt-X but makes no
>>>>>>>> sense for Opus and vice versa. I think that the a=3Dfmtp
>>>>>>>> line is the correct place to specify the desired
>>>>>>>> bitrate.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> For the broadcasting use-case, it is often more
>>>>>>>> important to have a constant and predictable audio
>>>>>>>> quality and network usage than mere audibility at any
>>>>>>>> cost.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> Best regards,
>>>>>>>>=20
>>>>>>>> Fredrik Bergholtz
>>>>>>>>=20
>>>>>>>> Swedish Radio
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> _______________________________________________
>>>>>>>> payload mailing list payload@ietf.org
>>>>>>>> <mailto:payload@ietf.org>=20
>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>=20
>>>>>> _______________________________________________ payload=20
>>>>>> mailing list payload@ietf.org <mailto:payload@ietf.org>=20
>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>>=20
>>> _______________________________________________ payload mailing=20
>>> list payload@ietf.org=20
>>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>>=20
>> _______________________________________________ payload mailing
>> list payload@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>> _______________________________________________ payload mailing
>> list payload@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/payload
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From: "Victor Demjanenko, Ph.D." <Victor.Demjanenko@vocal.com>
To: "'Roni Even'" <ron.even.tlv@gmail.com>, "'Mo Zanaty \(mzanaty\)'" <mzanaty@cisco.com>, "'Jean-Marc Valin'" <jmvalin@jmvalin.ca>, "'Fredrik Bergholtz'" <fredrik.bergholtz@sverigesradio.se>, "'Parthasarathi R'" <partha@parthasarathi.co.in>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com> <017401d02877$172e8b90$458ba2b0$@gmail.com>
In-Reply-To: <017401d02877$172e8b90$458ba2b0$@gmail.com>
Date: Mon, 5 Jan 2015 09:00:08 -0500
X-ASG-Orig-Subj: RE: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Cc: payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hi,

The set of permitted/preferred bit rate is also of interest for the MELP =
RTP
payload.  There are three permitted rates and some devices may only =
support
a specific subset.  (I know one device would only handle 2400 and 600
because the tables for 1200bps are huge.)

We currently suggested the following for MELP SDP:

    m=3Daudio 49120 RTP/AVP 97=20
    a=3Drtpmap:97 MELP/8000=20
    a=3Dfmtp:97 rate=3D2400,600,1200

Should we be considering a different way to express the =
permitted/preferred
rates?  We are not trying to invent something new and unique.  I believe =
I
had seen this with one other recent speech coder. =20

As an FYI, we have had a few reviewers external to the IETF process =
provide
comments.  Unfortunately they have been out-of-band but they have been
useful and supportive.  We would like to have our proposal move forward
towards an RFC in the next few months if possible.

Regards,

Victor Demjanenko


-----Original Message-----
From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Roni Even
Sent: Sunday, January 04, 2015 6:35 PM
To: 'Mo Zanaty (mzanaty)'; 'Jean-Marc Valin'; 'Fredrik Bergholtz';
'Parthasarathi R'
Cc: payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate

Hi,
For audio/voice there are RFC5577 and RFC4749 that has a bitrate =
parameter
while RFC5404 uses the b=3D parameter



Roni Even
As individual

> -----Original Message-----
> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Mo Zanaty
> (mzanaty)
> Sent: 05 January, 2015 12:06 AM
> To: Jean-Marc Valin; Fredrik Bergholtz; Parthasarathi R
> Cc: payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:=20
> maxaveragebitrate
>=20
> Some other codecs have fmtp parameters for maximum bitrate explicitly
(e.g.
> max-br in H264 / H264-SVC / H265) and/or implicitly (e.g.
> profile-level-id in H264 / H264-SVC / H265 / AAC; config in AAC).
>=20
> Other notable codecs that lack this include VP8 and VP9.
>=20
> This is a general problem which may become more important with complex =

> applications that include simulcast and scalable media within the same
media
> description (m=3D line). The SDP b=3D parameter is clearly =
insufficient=20
> for expressing multiple different maximum bitrates for each payload=20
> type,
unless
> we define a new bandwidth modifier which includes explicit PT =
bindings.
> Payload formats that include fmtp parameters for maximum bitrate can=20
> avoid this problem, so I think it is advisable to add this to Opus,
> VP8 and VP9.
>=20
> Mo
>=20
> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> Setting the bitrate isn't exactly a new problem here. Opus isn't the=20
> first
codec
> to support more than one rate so how are the other codecs (audio and
video)
> handling bitrate signaling?
>=20
> Cheers,
>=20
> 	Jean-Marc
>=20
> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
> > Hi.
> >
> > One thought om the general media level bit rate specification is=20
> > that it can never be anything other than an approximate value.
> > Different codecs define different constraints on the bit rate. I=20
> > think that the easiest way to specify exact bit rates would be via=20
> > the fmtp attribute.
> >
> > This can be compared with the ptime attribute - as it is defined=20
> > today, only approximate packet times can be signalled in SDP. For=20
> > the broadcast use case this is not acceptable and for that reason,=20
> > the European Broadcasting Union (EBU) has defined an SDP extension=20
> > that allows signalling of a ptime value for each codec. There is no=20
> > RFC draft for this extension and it is not yet registered with IANA=20
> > or IETF but it is starting to be used by manufacturers of=20
> > broadcasting equipment. The extension is known as "ACIP Profiles"
> > and can be found at the EBU web site. But I digress.
> >
> > I don't think that forcing encoders to approximate the nearest=20
> > allowed value for the selected codec is less complicated than=20
> > interpreting an exact value. In addition, it would add a=20
> > non-deterministic aspect, thus the resulting bit rate may not be =
what
the user expected.
> >
> > Best regards Fredrik Bergholtz Swedish Radio
> >
> > On 3 jan 2015, at 11:27, "Parthasarathi R"
> > <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>
> > wrote:
> >
> >> Hi all,
> >>
> >>
> >>
> >> I  understand that bandwidth =B3b=B2 parameter has lot of variants =
like=20
> >> AS, TIAS and usage but all those parameters are media level and not =

> >> codec specific. The bandwidth allocation is going to be happen in=20
> >> the media level in the network or end point based on the media=20
> >> level and not on the codec specific  in case the multiple codec is
negotiated.
> >> In most of the implementation, the multiple codec (Say Ex: G.711 &
> >> Opus) is going to be negotiated rather than single codec. The codec =

> >> specific bandwidth implementation complicates these bandwidth=20
> >> calculation still further which is applicable to maxaveragebitrate=20
> >> parameter in opus.
> >>
> >>
> >>
> >> I could not understand why the generic parameter should not used or =

> >> generic parameter has to be enhanced which meets Opus requirement=20
> >> rather than opus specific parameter.
> >>
> >>
> >>
> >> Thanks
> >>
> >> Partha
> >>
> >>
> >>
> >> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of *Ali =

> >> C. Begen (abegen) *Sent:* Wednesday, December 31, 2014 4:57 PM=20
> >> *To:* Fredrik Bergholtz *Cc:* payload@ietf.org=20
> >> <mailto:payload@ietf.org>
> >> *Subject:* Re: [payload]
> >> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
> >>
> >>
> >>
> >> Authors
> >>
> >> Please chime in and lets discuss s quick solution for this on the=20
> >> list. Once agreed we can ship this draft.
> >>
> >> -acbegen
> >>
> >>
> >>
> >> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se
> >> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31, 2014
> >> 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc Valin=20
> >> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>;Fredrik
> >> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> *Subject:*
> >> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
> >>
> >>
> >>
> >> Yes, I do see a strong need for specifying the bitrate.
> >>
> >> In the broadcasting use case we have to make sure that we get the=20
> >> best audio quality that is possible for the network that is=20
> >> currently used but never try to send more data than the particular=20
> >> link is capable of. For this reason, specifying a maximum bit rate=20
> >> is required.
> >>
> >> In addition we often have the requirement that the quality is=20
> >> constant rather than the best possible. To avoid as many audible=20
> >> changes in the audio as possible over time we will often use=20
> >> constant bit rate and then it becomes even more important to be=20
> >> able to specify what bit rate should be.
> >>
> >> We will use the Opus codec over various network links with=20
> >> different capabilities in terms of bit rate, delay variation,=20
> >> absolute delay, etc and we will typically use the SDP to signal the =

> >> parameters that best fit both the current network and the current=20
> >> type of radio production.
> >>
> >> Best regards Fredrik Bergholtz Swedish Radio
> >>
> >>
> >>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
> >>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
> >>>
> >>> My understanding is that the b=3D line in the SDP is quite a=20
> >>> subjective parameter. I have seen people use it differently over=20
> >>> the past several years. So, it is a bit difficult to infer much=20
> >>> accuracy from that line.
> >>>
> >>> OTOH, if someone feels like Opus needs to specify some sort of=20
> >>> bitrate variability in the SDP, it can do so in an optional
> >>> (opus) parameter. Obviously rules around how to set that value and =

> >>> what to do with that value need to be clearly specified.
> >>>
> >>> Fredrik (or anyone else in the WG), do you see a strong need for=20
> >>> this? If so, now is the time to mention that.
> >>>
> >>> -acbegen
> >>>
> >>>
> >>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin <jmvalin@mozilla.com =

> >>>> <mailto:jmvalin@mozilla.com>> wrote:
> >>>>
> > I have to say this is not something I understand really well. Does=20
> > anyone else here have an opinion on how to specify bit-rate?
> >
> > Jean-Marc
> >
> >>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> I read in the latest draft that you have removed the=20
> >>>>>> maxaveragebitrate parameter from the SDP signalling for Opus. I =

> >>>>>> think that this parameter is useful and that it provides added=20
> >>>>>> value over the bandwidth modifiers added by RFC 3556.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> I must admit that I haven=B9t read all of RFC 3556 but as I=20
> >>>>>> understand it, a b=3D line are not in any way related to an=20
> >>>>>> actual format of a media description. This means that with this =

> >>>>>> mechanism, only one bitrate specification can be sent with each =

> >>>>>> media description. Different coding algorithms have different=20
> >>>>>> constraints on the allowed bitrate. For example, I might want=20
> >>>>>> to send an SDP offer with both Enhanced apt-X at 576 kbit/s and =

> >>>>>> Opus at 510kbit/s with cbr. Now, specifying 576kbit/s via a =
b=3D=20
> >>>>>> line works for apt-X but makes no sense for Opus and vice=20
> >>>>>> versa. I think that the a=3Dfmtp line is the correct place to=20
> >>>>>> specify the desired bitrate.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> For the broadcasting use-case, it is often more important to=20
> >>>>>> have a constant and predictable audio quality and network usage =

> >>>>>> than mere audibility at any cost.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Best regards,
> >>>>>>
> >>>>>> Fredrik Bergholtz
> >>>>>>
> >>>>>> Swedish Radio
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________ payload
> mailing
> >>>>>> list payload@ietf.org <mailto:payload@ietf.org>=20
> >>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>
> >>>> _______________________________________________ payload
> mailing
> >>>> list payload@ietf.org <mailto:payload@ietf.org>=20
> >>>> https://www.ietf.org/mailman/listinfo/payload
> >>>
> >>
> >
> >
> > _______________________________________________ payload mailing list =

> > payload@ietf.org https://www.ietf.org/mailman/listinfo/payload
> >
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1
>=20
> iQEcBAEBAgAGBQJUqZhTAAoJEJ6/8sItn9q9NaAH/jTELxrPUMSmhWHIODeGOrJ
> X
> 67evbcn5cIsZY/8+yyofMNbaFq7DCqdR3BKX9uJ/dUJ+4yR/nbyxvw/msjd65dkv
> YS3HVibycnMOnZkFENr7by6vvMaZn7mIjbOWukxdBePDKHhQq7ZPec++j8gWVp
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> 6qy7BSMJle9TTH3nt/kzV8YbnP7Pt0iVUorMYH/JFUzHfwcPRROKlqmdgf8yAcIZ
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> =3DADn2
> -----END PGP SIGNATURE-----
>=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

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


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From: Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se>
To: "Victor Demjanenko, Ph.D." <Victor.Demjanenko@vocal.com>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Date: Mon, 5 Jan 2015 14:33:29 +0000
Message-ID: <A1B5079E-F9ED-4CD6-A40D-6BF56D3074EF@sverigesradio.se>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com> <017401d02877$172e8b90$458ba2b0$@gmail.com>, <184801d028ef$eacb9650$c062c2f0$@Demjanenko@vocal.com>
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Cc: "payload@ietf.org" <payload@ietf.org>, Jean-Marc Valin <jmvalin@jmvalin.ca>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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I may be off base here but if multiple bit-rates of the same codec are to b=
e specified, couldn't one simply put them as multiple formats in the m-line=
, all mapped to the same codec in the rtpmap-lines but having different fmt=
p-lines?

Best regards
Fredrik Bergholtz
Swedish Radio

> On 5 jan 2015, at 15:00, "Victor Demjanenko, Ph.D." <Victor.Demjanenko@vo=
cal.com> wrote:
>=20
> Hi,
>=20
> The set of permitted/preferred bit rate is also of interest for the MELP =
RTP
> payload.  There are three permitted rates and some devices may only suppo=
rt
> a specific subset.  (I know one device would only handle 2400 and 600
> because the tables for 1200bps are huge.)
>=20
> We currently suggested the following for MELP SDP:
>=20
>    m=3Daudio 49120 RTP/AVP 97=20
>    a=3Drtpmap:97 MELP/8000=20
>    a=3Dfmtp:97 rate=3D2400,600,1200
>=20
> Should we be considering a different way to express the permitted/preferr=
ed
> rates?  We are not trying to invent something new and unique.  I believe =
I
> had seen this with one other recent speech coder. =20
>=20
> As an FYI, we have had a few reviewers external to the IETF process provi=
de
> comments.  Unfortunately they have been out-of-band but they have been
> useful and supportive.  We would like to have our proposal move forward
> towards an RFC in the next few months if possible.
>=20
> Regards,
>=20
> Victor Demjanenko
>=20
>=20
> -----Original Message-----
> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Roni Even
> Sent: Sunday, January 04, 2015 6:35 PM
> To: 'Mo Zanaty (mzanaty)'; 'Jean-Marc Valin'; 'Fredrik Bergholtz';
> 'Parthasarathi R'
> Cc: payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>=20
> Hi,
> For audio/voice there are RFC5577 and RFC4749 that has a bitrate paramete=
r
> while RFC5404 uses the b=3D parameter
>=20
>=20
>=20
> Roni Even
> As individual
>=20
>> -----Original Message-----
>> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Mo Zanaty
>> (mzanaty)
>> Sent: 05 January, 2015 12:06 AM
>> To: Jean-Marc Valin; Fredrik Bergholtz; Parthasarathi R
>> Cc: payload@ietf.org
>> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:=20
>> maxaveragebitrate
>>=20
>> Some other codecs have fmtp parameters for maximum bitrate explicitly
> (e.g.
>> max-br in H264 / H264-SVC / H265) and/or implicitly (e.g.
>> profile-level-id in H264 / H264-SVC / H265 / AAC; config in AAC).
>>=20
>> Other notable codecs that lack this include VP8 and VP9.
>>=20
>> This is a general problem which may become more important with complex=20
>> applications that include simulcast and scalable media within the same
> media
>> description (m=3D line). The SDP b=3D parameter is clearly insufficient=
=20
>> for expressing multiple different maximum bitrates for each payload=20
>> type,
> unless
>> we define a new bandwidth modifier which includes explicit PT bindings.
>> Payload formats that include fmtp parameters for maximum bitrate can=20
>> avoid this problem, so I think it is advisable to add this to Opus,
>> VP8 and VP9.
>>=20
>> Mo
>>=20
>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:
>>=20
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>=20
>> Setting the bitrate isn't exactly a new problem here. Opus isn't the=20
>> first
> codec
>> to support more than one rate so how are the other codecs (audio and
> video)
>> handling bitrate signaling?
>>=20
>> Cheers,
>>=20
>>    Jean-Marc
>>=20
>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
>>> Hi.
>>>=20
>>> One thought om the general media level bit rate specification is=20
>>> that it can never be anything other than an approximate value.
>>> Different codecs define different constraints on the bit rate. I=20
>>> think that the easiest way to specify exact bit rates would be via=20
>>> the fmtp attribute.
>>>=20
>>> This can be compared with the ptime attribute - as it is defined=20
>>> today, only approximate packet times can be signalled in SDP. For=20
>>> the broadcast use case this is not acceptable and for that reason,=20
>>> the European Broadcasting Union (EBU) has defined an SDP extension=20
>>> that allows signalling of a ptime value for each codec. There is no=20
>>> RFC draft for this extension and it is not yet registered with IANA=20
>>> or IETF but it is starting to be used by manufacturers of=20
>>> broadcasting equipment. The extension is known as "ACIP Profiles"
>>> and can be found at the EBU web site. But I digress.
>>>=20
>>> I don't think that forcing encoders to approximate the nearest=20
>>> allowed value for the selected codec is less complicated than=20
>>> interpreting an exact value. In addition, it would add a=20
>>> non-deterministic aspect, thus the resulting bit rate may not be what
> the user expected.
>>>=20
>>> Best regards Fredrik Bergholtz Swedish Radio
>>>=20
>>> On 3 jan 2015, at 11:27, "Parthasarathi R"
>>> <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>
>>> wrote:
>>>=20
>>>> Hi all,
>>>>=20
>>>>=20
>>>>=20
>>>> I  understand that bandwidth =B3b=B2 parameter has lot of variants lik=
e=20
>>>> AS, TIAS and usage but all those parameters are media level and not=20
>>>> codec specific. The bandwidth allocation is going to be happen in=20
>>>> the media level in the network or end point based on the media=20
>>>> level and not on the codec specific  in case the multiple codec is
> negotiated.
>>>> In most of the implementation, the multiple codec (Say Ex: G.711 &
>>>> Opus) is going to be negotiated rather than single codec. The codec=20
>>>> specific bandwidth implementation complicates these bandwidth=20
>>>> calculation still further which is applicable to maxaveragebitrate=20
>>>> parameter in opus.
>>>>=20
>>>>=20
>>>>=20
>>>> I could not understand why the generic parameter should not used or=20
>>>> generic parameter has to be enhanced which meets Opus requirement=20
>>>> rather than opus specific parameter.
>>>>=20
>>>>=20
>>>>=20
>>>> Thanks
>>>>=20
>>>> Partha
>>>>=20
>>>>=20
>>>>=20
>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of *Ali=20
>>>> C. Begen (abegen) *Sent:* Wednesday, December 31, 2014 4:57 PM=20
>>>> *To:* Fredrik Bergholtz *Cc:* payload@ietf.org=20
>>>> <mailto:payload@ietf.org>
>>>> *Subject:* Re: [payload]
>>>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>>>=20
>>>>=20
>>>>=20
>>>> Authors
>>>>=20
>>>> Please chime in and lets discuss s quick solution for this on the=20
>>>> list. Once agreed we can ship this draft.
>>>>=20
>>>> -acbegen
>>>>=20
>>>>=20
>>>>=20
>>>> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se
>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31, 2014
>>>> 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc Valin=20
>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>;Fredrik
>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> *Subject:*
>>>> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>>>=20
>>>>=20
>>>>=20
>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>=20
>>>> In the broadcasting use case we have to make sure that we get the=20
>>>> best audio quality that is possible for the network that is=20
>>>> currently used but never try to send more data than the particular=20
>>>> link is capable of. For this reason, specifying a maximum bit rate=20
>>>> is required.
>>>>=20
>>>> In addition we often have the requirement that the quality is=20
>>>> constant rather than the best possible. To avoid as many audible=20
>>>> changes in the audio as possible over time we will often use=20
>>>> constant bit rate and then it becomes even more important to be=20
>>>> able to specify what bit rate should be.
>>>>=20
>>>> We will use the Opus codec over various network links with=20
>>>> different capabilities in terms of bit rate, delay variation,=20
>>>> absolute delay, etc and we will typically use the SDP to signal the=20
>>>> parameters that best fit both the current network and the current=20
>>>> type of radio production.
>>>>=20
>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>=20
>>>>=20
>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>=20
>>>>> My understanding is that the b=3D line in the SDP is quite a=20
>>>>> subjective parameter. I have seen people use it differently over=20
>>>>> the past several years. So, it is a bit difficult to infer much=20
>>>>> accuracy from that line.
>>>>>=20
>>>>> OTOH, if someone feels like Opus needs to specify some sort of=20
>>>>> bitrate variability in the SDP, it can do so in an optional
>>>>> (opus) parameter. Obviously rules around how to set that value and=20
>>>>> what to do with that value need to be clearly specified.
>>>>>=20
>>>>> Fredrik (or anyone else in the WG), do you see a strong need for=20
>>>>> this? If so, now is the time to mention that.
>>>>>=20
>>>>> -acbegen
>>>>>=20
>>>>>=20
>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin <jmvalin@mozilla.com=20
>>>>>> <mailto:jmvalin@mozilla.com>> wrote:
>>> I have to say this is not something I understand really well. Does=20
>>> anyone else here have an opinion on how to specify bit-rate?
>>>=20
>>> Jean-Marc
>>>=20
>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I read in the latest draft that you have removed the=20
>>>>>>>> maxaveragebitrate parameter from the SDP signalling for Opus. I=20
>>>>>>>> think that this parameter is useful and that it provides added=20
>>>>>>>> value over the bandwidth modifiers added by RFC 3556.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I must admit that I haven=B9t read all of RFC 3556 but as I=20
>>>>>>>> understand it, a b=3D line are not in any way related to an=20
>>>>>>>> actual format of a media description. This means that with this=20
>>>>>>>> mechanism, only one bitrate specification can be sent with each=20
>>>>>>>> media description. Different coding algorithms have different=20
>>>>>>>> constraints on the allowed bitrate. For example, I might want=20
>>>>>>>> to send an SDP offer with both Enhanced apt-X at 576 kbit/s and=20
>>>>>>>> Opus at 510kbit/s with cbr. Now, specifying 576kbit/s via a b=3D=20
>>>>>>>> line works for apt-X but makes no sense for Opus and vice=20
>>>>>>>> versa. I think that the a=3Dfmtp line is the correct place to=20
>>>>>>>> specify the desired bitrate.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> For the broadcasting use-case, it is often more important to=20
>>>>>>>> have a constant and predictable audio quality and network usage=20
>>>>>>>> than mere audibility at any cost.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> Best regards,
>>>>>>>>=20
>>>>>>>> Fredrik Bergholtz
>>>>>>>>=20
>>>>>>>> Swedish Radio
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> _______________________________________________ payload
>> mailing
>>>>>>>> list payload@ietf.org <mailto:payload@ietf.org>=20
>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>=20
>>>>>> _______________________________________________ payload
>> mailing
>>>>>> list payload@ietf.org <mailto:payload@ietf.org>=20
>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>=20
>>>=20
>>> _______________________________________________ payload mailing list=20
>>> payload@ietf.org https://www.ietf.org/mailman/listinfo/payload
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1
>>=20
>> iQEcBAEBAgAGBQJUqZhTAAoJEJ6/8sItn9q9NaAH/jTELxrPUMSmhWHIODeGOrJ
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>> bR
>> 6qy7BSMJle9TTH3nt/kzV8YbnP7Pt0iVUorMYH/JFUzHfwcPRROKlqmdgf8yAcIZ
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>> =3DADn2
>> -----END PGP SIGNATURE-----
>>=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
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20


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From: "Victor Demjanenko, Ph.D." <Victor.Demjanenko@vocal.com>
To: "'Fredrik Bergholtz'" <fredrik.bergholtz@sverigesradio.se>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com> <017401d02877$172e8b90$458ba2b0$@gmail.com>, <184801d028ef$eacb9650$c062c2f0$@Demjanenko@vocal.com> <A1B5079E-F9ED-4CD6-A40D-6BF56D3074EF@sverigesradio.se>
In-Reply-To: <A1B5079E-F9ED-4CD6-A40D-6BF56D3074EF@sverigesradio.se>
Date: Mon, 5 Jan 2015 09:54:46 -0500
X-ASG-Orig-Subj: RE: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Cc: payload@ietf.org, 'Jean-Marc Valin' <jmvalin@jmvalin.ca>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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I agree that this is viable (and may be preferred for some =
applications),
however, it does require distinct payload numbers for each rate.
Furthermore, it does not convey the preferred rate or permitted rates =
for
switching well.  A "rate" parameter with one, two or three rates shows =
both
the permitted rates and the preferred order.  Implied is also the =
permission
to switch rates on the fly without doing a REINVITE.

Victor

-----Original Message-----
From: Fredrik Bergholtz [mailto:fredrik.bergholtz@sverigesradio.se]=20
Sent: Monday, January 05, 2015 9:33 AM
To: Victor Demjanenko, Ph.D.
Cc: Roni Even; Mo Zanaty (mzanaty); Jean-Marc Valin; Fredrik Bergholtz;
Parthasarathi R; payload@ietf.org; Victor Demjanenko, Ph.D.
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate

I may be off base here but if multiple bit-rates of the same codec are =
to be
specified, couldn't one simply put them as multiple formats in the =
m-line,
all mapped to the same codec in the rtpmap-lines but having different
fmtp-lines?

Best regards
Fredrik Bergholtz
Swedish Radio

> On 5 jan 2015, at 15:00, "Victor Demjanenko, Ph.D."
<Victor.Demjanenko@vocal.com> wrote:
>=20
> Hi,
>=20
> The set of permitted/preferred bit rate is also of interest for the=20
> MELP RTP payload.  There are three permitted rates and some devices=20
> may only support a specific subset.  (I know one device would only=20
> handle 2400 and 600 because the tables for 1200bps are huge.)
>=20
> We currently suggested the following for MELP SDP:
>=20
>    m=3Daudio 49120 RTP/AVP 97=20
>    a=3Drtpmap:97 MELP/8000=20
>    a=3Dfmtp:97 rate=3D2400,600,1200
>=20
> Should we be considering a different way to express the=20
> permitted/preferred rates?  We are not trying to invent something new=20
> and unique.  I believe I had seen this with one other recent speech =
coder.
>=20
> As an FYI, we have had a few reviewers external to the IETF process=20
> provide comments.  Unfortunately they have been out-of-band but they=20
> have been useful and supportive.  We would like to have our proposal=20
> move forward towards an RFC in the next few months if possible.
>=20
> Regards,
>=20
> Victor Demjanenko
>=20
>=20
> -----Original Message-----
> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Roni Even
> Sent: Sunday, January 04, 2015 6:35 PM
> To: 'Mo Zanaty (mzanaty)'; 'Jean-Marc Valin'; 'Fredrik Bergholtz';=20
> 'Parthasarathi R'
> Cc: payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:=20
> maxaveragebitrate
>=20
> Hi,
> For audio/voice there are RFC5577 and RFC4749 that has a bitrate=20
> parameter while RFC5404 uses the b=3D parameter
>=20
>=20
>=20
> Roni Even
> As individual
>=20
>> -----Original Message-----
>> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Mo=20
>> Zanaty
>> (mzanaty)
>> Sent: 05 January, 2015 12:06 AM
>> To: Jean-Marc Valin; Fredrik Bergholtz; Parthasarathi R
>> Cc: payload@ietf.org
>> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:=20
>> maxaveragebitrate
>>=20
>> Some other codecs have fmtp parameters for maximum bitrate explicitly
> (e.g.
>> max-br in H264 / H264-SVC / H265) and/or implicitly (e.g.
>> profile-level-id in H264 / H264-SVC / H265 / AAC; config in AAC).
>>=20
>> Other notable codecs that lack this include VP8 and VP9.
>>=20
>> This is a general problem which may become more important with=20
>> complex applications that include simulcast and scalable media within =

>> the same
> media
>> description (m=3D line). The SDP b=3D parameter is clearly =
insufficient=20
>> for expressing multiple different maximum bitrates for each payload=20
>> type,
> unless
>> we define a new bandwidth modifier which includes explicit PT =
bindings.
>> Payload formats that include fmtp parameters for maximum bitrate can=20
>> avoid this problem, so I think it is advisable to add this to Opus,
>> VP8 and VP9.
>>=20
>> Mo
>>=20
>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca> wrote:
>>=20
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>=20
>> Setting the bitrate isn't exactly a new problem here. Opus isn't the=20
>> first
> codec
>> to support more than one rate so how are the other codecs (audio and
> video)
>> handling bitrate signaling?
>>=20
>> Cheers,
>>=20
>>    Jean-Marc
>>=20
>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote:
>>> Hi.
>>>=20
>>> One thought om the general media level bit rate specification is=20
>>> that it can never be anything other than an approximate value.
>>> Different codecs define different constraints on the bit rate. I=20
>>> think that the easiest way to specify exact bit rates would be via=20
>>> the fmtp attribute.
>>>=20
>>> This can be compared with the ptime attribute - as it is defined=20
>>> today, only approximate packet times can be signalled in SDP. For=20
>>> the broadcast use case this is not acceptable and for that reason,=20
>>> the European Broadcasting Union (EBU) has defined an SDP extension=20
>>> that allows signalling of a ptime value for each codec. There is no=20
>>> RFC draft for this extension and it is not yet registered with IANA=20
>>> or IETF but it is starting to be used by manufacturers of=20
>>> broadcasting equipment. The extension is known as "ACIP Profiles"
>>> and can be found at the EBU web site. But I digress.
>>>=20
>>> I don't think that forcing encoders to approximate the nearest=20
>>> allowed value for the selected codec is less complicated than=20
>>> interpreting an exact value. In addition, it would add a=20
>>> non-deterministic aspect, thus the resulting bit rate may not be=20
>>> what
> the user expected.
>>>=20
>>> Best regards Fredrik Bergholtz Swedish Radio
>>>=20
>>> On 3 jan 2015, at 11:27, "Parthasarathi R"
>>> <partha@parthasarathi.co.in <mailto:partha@parthasarathi.co.in>>
>>> wrote:
>>>=20
>>>> Hi all,
>>>>=20
>>>>=20
>>>>=20
>>>> I  understand that bandwidth =B3b=B2 parameter has lot of variants =
like=20
>>>> AS, TIAS and usage but all those parameters are media level and not =

>>>> codec specific. The bandwidth allocation is going to be happen in=20
>>>> the media level in the network or end point based on the media=20
>>>> level and not on the codec specific  in case the multiple codec is
> negotiated.
>>>> In most of the implementation, the multiple codec (Say Ex: G.711 &
>>>> Opus) is going to be negotiated rather than single codec. The codec =

>>>> specific bandwidth implementation complicates these bandwidth=20
>>>> calculation still further which is applicable to maxaveragebitrate=20
>>>> parameter in opus.
>>>>=20
>>>>=20
>>>>=20
>>>> I could not understand why the generic parameter should not used or =

>>>> generic parameter has to be enhanced which meets Opus requirement=20
>>>> rather than opus specific parameter.
>>>>=20
>>>>=20
>>>>=20
>>>> Thanks
>>>>=20
>>>> Partha
>>>>=20
>>>>=20
>>>>=20
>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On Behalf Of *Ali =

>>>> C. Begen (abegen) *Sent:* Wednesday, December 31, 2014 4:57 PM
>>>> *To:* Fredrik Bergholtz *Cc:* payload@ietf.org=20
>>>> <mailto:payload@ietf.org>
>>>> *Subject:* Re: [payload]
>>>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>>>=20
>>>>=20
>>>>=20
>>>> Authors
>>>>=20
>>>> Please chime in and lets discuss s quick solution for this on the=20
>>>> list. Once agreed we can ship this draft.
>>>>=20
>>>> -acbegen
>>>>=20
>>>>=20
>>>>=20
>>>> *From:*Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se
>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec 31, 2014
>>>> 11:42 AM *To:* Ali C. Begen (abegen) *Cc:* Jean-Marc Valin=20
>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>;Fredrik
>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> *Subject:*
>>>> Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>>>>=20
>>>>=20
>>>>=20
>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>=20
>>>> In the broadcasting use case we have to make sure that we get the=20
>>>> best audio quality that is possible for the network that is=20
>>>> currently used but never try to send more data than the particular=20
>>>> link is capable of. For this reason, specifying a maximum bit rate=20
>>>> is required.
>>>>=20
>>>> In addition we often have the requirement that the quality is=20
>>>> constant rather than the best possible. To avoid as many audible=20
>>>> changes in the audio as possible over time we will often use=20
>>>> constant bit rate and then it becomes even more important to be=20
>>>> able to specify what bit rate should be.
>>>>=20
>>>> We will use the Opus codec over various network links with=20
>>>> different capabilities in terms of bit rate, delay variation,=20
>>>> absolute delay, etc and we will typically use the SDP to signal the =

>>>> parameters that best fit both the current network and the current=20
>>>> type of radio production.
>>>>=20
>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>=20
>>>>=20
>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>=20
>>>>> My understanding is that the b=3D line in the SDP is quite a=20
>>>>> subjective parameter. I have seen people use it differently over=20
>>>>> the past several years. So, it is a bit difficult to infer much=20
>>>>> accuracy from that line.
>>>>>=20
>>>>> OTOH, if someone feels like Opus needs to specify some sort of=20
>>>>> bitrate variability in the SDP, it can do so in an optional
>>>>> (opus) parameter. Obviously rules around how to set that value and =

>>>>> what to do with that value need to be clearly specified.
>>>>>=20
>>>>> Fredrik (or anyone else in the WG), do you see a strong need for=20
>>>>> this? If so, now is the time to mention that.
>>>>>=20
>>>>> -acbegen
>>>>>=20
>>>>>=20
>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin <jmvalin@mozilla.com =

>>>>>> <mailto:jmvalin@mozilla.com>> wrote:
>>> I have to say this is not something I understand really well. Does=20
>>> anyone else here have an opinion on how to specify bit-rate?
>>>=20
>>> Jean-Marc
>>>=20
>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote: Hi.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I read in the latest draft that you have removed the=20
>>>>>>>> maxaveragebitrate parameter from the SDP signalling for Opus. I =

>>>>>>>> think that this parameter is useful and that it provides added=20
>>>>>>>> value over the bandwidth modifiers added by RFC 3556.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I must admit that I haven=B9t read all of RFC 3556 but as I=20
>>>>>>>> understand it, a b=3D line are not in any way related to an=20
>>>>>>>> actual format of a media description. This means that with this =

>>>>>>>> mechanism, only one bitrate specification can be sent with each =

>>>>>>>> media description. Different coding algorithms have different=20
>>>>>>>> constraints on the allowed bitrate. For example, I might want=20
>>>>>>>> to send an SDP offer with both Enhanced apt-X at 576 kbit/s and =

>>>>>>>> Opus at 510kbit/s with cbr. Now, specifying 576kbit/s via a =
b=3D=20
>>>>>>>> line works for apt-X but makes no sense for Opus and vice=20
>>>>>>>> versa. I think that the a=3Dfmtp line is the correct place to=20
>>>>>>>> specify the desired bitrate.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> For the broadcasting use-case, it is often more important to=20
>>>>>>>> have a constant and predictable audio quality and network usage =

>>>>>>>> than mere audibility at any cost.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> Best regards,
>>>>>>>>=20
>>>>>>>> Fredrik Bergholtz
>>>>>>>>=20
>>>>>>>> Swedish Radio
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> _______________________________________________ payload
>> mailing
>>>>>>>> list payload@ietf.org <mailto:payload@ietf.org>=20
>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>=20
>>>>>> _______________________________________________ payload
>> mailing
>>>>>> list payload@ietf.org <mailto:payload@ietf.org>=20
>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>=20
>>>=20
>>> _______________________________________________ payload mailing list =

>>> payload@ietf.org https://www.ietf.org/mailman/listinfo/payload
>> -----BEGIN PGP SIGNATURE-----
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>> =3DADn2
>> -----END PGP SIGNATURE-----
>>=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
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20


From nobody Mon Jan  5 09:47:57 2015
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From: Jean-Marc Valin <jmvalin@jmvalin.ca>
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To: "Victor Demjanenko, Ph.D." <Victor.Demjanenko@vocal.com>,  'Fredrik Bergholtz' <fredrik.bergholtz@sverigesradio.se>
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Cc: 'Jean-Marc Valin' <jmvalin@jmvalin.ca>, payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hash: SHA1

Listing multiple rates might make sense when the codec only supports a
handful of rates. In the case of Opus, it's a continuous scale so I
believe a max bitrate makes more sense.

	Jean-Marc

On 05/01/15 09:54 AM, Victor Demjanenko, Ph.D. wrote:
> I agree that this is viable (and may be preferred for some
> applications), however, it does require distinct payload numbers
> for each rate. Furthermore, it does not convey the preferred rate
> or permitted rates for switching well.  A "rate" parameter with
> one, two or three rates shows both the permitted rates and the
> preferred order.  Implied is also the permission to switch rates on
> the fly without doing a REINVITE.
> 
> Victor
> 
> -----Original Message----- From: Fredrik Bergholtz
> [mailto:fredrik.bergholtz@sverigesradio.se] Sent: Monday, January
> 05, 2015 9:33 AM To: Victor Demjanenko, Ph.D. Cc: Roni Even; Mo
> Zanaty (mzanaty); Jean-Marc Valin; Fredrik Bergholtz; Parthasarathi
> R; payload@ietf.org; Victor Demjanenko, Ph.D. Subject: Re:
> [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
> 
> I may be off base here but if multiple bit-rates of the same codec
> are to be specified, couldn't one simply put them as multiple
> formats in the m-line, all mapped to the same codec in the
> rtpmap-lines but having different fmtp-lines?
> 
> Best regards Fredrik Bergholtz Swedish Radio
> 
>> On 5 jan 2015, at 15:00, "Victor Demjanenko, Ph.D."
> <Victor.Demjanenko@vocal.com> wrote:
>> 
>> Hi,
>> 
>> The set of permitted/preferred bit rate is also of interest for
>> the MELP RTP payload.  There are three permitted rates and some
>> devices may only support a specific subset.  (I know one device
>> would only handle 2400 and 600 because the tables for 1200bps are
>> huge.)
>> 
>> We currently suggested the following for MELP SDP:
>> 
>> m=audio 49120 RTP/AVP 97 a=rtpmap:97 MELP/8000 a=fmtp:97
>> rate=2400,600,1200
>> 
>> Should we be considering a different way to express the 
>> permitted/preferred rates?  We are not trying to invent something
>> new and unique.  I believe I had seen this with one other recent
>> speech coder.
>> 
>> As an FYI, we have had a few reviewers external to the IETF
>> process provide comments.  Unfortunately they have been
>> out-of-band but they have been useful and supportive.  We would
>> like to have our proposal move forward towards an RFC in the next
>> few months if possible.
>> 
>> Regards,
>> 
>> Victor Demjanenko
>> 
>> 
>> -----Original Message----- From: payload
>> [mailto:payload-bounces@ietf.org] On Behalf Of Roni Even Sent:
>> Sunday, January 04, 2015 6:35 PM To: 'Mo Zanaty (mzanaty)';
>> 'Jean-Marc Valin'; 'Fredrik Bergholtz'; 'Parthasarathi R' Cc:
>> payload@ietf.org Subject: Re: [payload]
>> draft-ietf-payload-rtp-opus-05: maxaveragebitrate
>> 
>> Hi, For audio/voice there are RFC5577 and RFC4749 that has a
>> bitrate parameter while RFC5404 uses the b= parameter
>> 
>> 
>> 
>> Roni Even As individual
>> 
>>> -----Original Message----- From: payload
>>> [mailto:payload-bounces@ietf.org] On Behalf Of Mo Zanaty 
>>> (mzanaty) Sent: 05 January, 2015 12:06 AM To: Jean-Marc Valin;
>>> Fredrik Bergholtz; Parthasarathi R Cc: payload@ietf.org 
>>> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: 
>>> maxaveragebitrate
>>> 
>>> Some other codecs have fmtp parameters for maximum bitrate
>>> explicitly
>> (e.g.
>>> max-br in H264 / H264-SVC / H265) and/or implicitly (e.g. 
>>> profile-level-id in H264 / H264-SVC / H265 / AAC; config in
>>> AAC).
>>> 
>>> Other notable codecs that lack this include VP8 and VP9.
>>> 
>>> This is a general problem which may become more important with
>>>  complex applications that include simulcast and scalable media
>>> within the same
>> media
>>> description (m= line). The SDP b= parameter is clearly
>>> insufficient for expressing multiple different maximum bitrates
>>> for each payload type,
>> unless
>>> we define a new bandwidth modifier which includes explicit PT
>>> bindings. Payload formats that include fmtp parameters for
>>> maximum bitrate can avoid this problem, so I think it is
>>> advisable to add this to Opus, VP8 and VP9.
>>> 
>>> Mo
>>> 
>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
>>> wrote:
>>> 
> Setting the bitrate isn't exactly a new problem here. Opus isn't
> the first
>>> codec
> to support more than one rate so how are the other codecs (audio
> and
>>> video)
> handling bitrate signaling?
> 
> Cheers,
> 
> Jean-Marc
> 
>>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
>>>>> 
>>>>> One thought om the general media level bit rate
>>>>> specification is that it can never be anything other than
>>>>> an approximate value. Different codecs define different
>>>>> constraints on the bit rate. I think that the easiest way
>>>>> to specify exact bit rates would be via the fmtp
>>>>> attribute.
>>>>> 
>>>>> This can be compared with the ptime attribute - as it is
>>>>> defined today, only approximate packet times can be
>>>>> signalled in SDP. For the broadcast use case this is not
>>>>> acceptable and for that reason, the European Broadcasting
>>>>> Union (EBU) has defined an SDP extension that allows
>>>>> signalling of a ptime value for each codec. There is no RFC
>>>>> draft for this extension and it is not yet registered with
>>>>> IANA or IETF but it is starting to be used by manufacturers
>>>>> of broadcasting equipment. The extension is known as "ACIP
>>>>> Profiles" and can be found at the EBU web site. But I
>>>>> digress.
>>>>> 
>>>>> I don't think that forcing encoders to approximate the
>>>>> nearest allowed value for the selected codec is less
>>>>> complicated than interpreting an exact value. In addition,
>>>>> it would add a non-deterministic aspect, thus the resulting
>>>>> bit rate may not be what
>>> the user expected.
>>>>> 
>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>> 
>>>>> On 3 jan 2015, at 11:27, "Parthasarathi R" 
>>>>> <partha@parthasarathi.co.in
>>>>> <mailto:partha@parthasarathi.co.in>> wrote:
>>>>> 
>>>>>> Hi all,
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> I  understand that bandwidth ³b² parameter has lot of
>>>>>> variants like AS, TIAS and usage but all those parameters
>>>>>> are media level and not codec specific. The bandwidth
>>>>>> allocation is going to be happen in the media level in
>>>>>> the network or end point based on the media level and not
>>>>>> on the codec specific  in case the multiple codec is
>>> negotiated.
>>>>>> In most of the implementation, the multiple codec (Say
>>>>>> Ex: G.711 & Opus) is going to be negotiated rather than
>>>>>> single codec. The codec specific bandwidth implementation
>>>>>> complicates these bandwidth calculation still further
>>>>>> which is applicable to maxaveragebitrate parameter in
>>>>>> opus.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> I could not understand why the generic parameter should
>>>>>> not used or generic parameter has to be enhanced which
>>>>>> meets Opus requirement rather than opus specific
>>>>>> parameter.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Thanks
>>>>>> 
>>>>>> Partha
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
>>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
>>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
>>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
>>>>>> [payload] draft-ietf-payload-rtp-opus-05:
>>>>>> maxaveragebitrate
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Authors
>>>>>> 
>>>>>> Please chime in and lets discuss s quick solution for
>>>>>> this on the list. Once agreed we can ship this draft.
>>>>>> 
>>>>>> -acbegen
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> *From:*Fredrik Bergholtz
>>>>>> <fredrik.bergholtz@sverigesradio.se 
>>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
>>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
>>>>>> Jean-Marc Valin <jmvalin@mozilla.com
>>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik 
>>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
>>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
>>>>>> maxaveragebitrate
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>>> 
>>>>>> In the broadcasting use case we have to make sure that we
>>>>>> get the best audio quality that is possible for the
>>>>>> network that is currently used but never try to send more
>>>>>> data than the particular link is capable of. For this
>>>>>> reason, specifying a maximum bit rate is required.
>>>>>> 
>>>>>> In addition we often have the requirement that the
>>>>>> quality is constant rather than the best possible. To
>>>>>> avoid as many audible changes in the audio as possible
>>>>>> over time we will often use constant bit rate and then it
>>>>>> becomes even more important to be able to specify what
>>>>>> bit rate should be.
>>>>>> 
>>>>>> We will use the Opus codec over various network links
>>>>>> with different capabilities in terms of bit rate, delay
>>>>>> variation, absolute delay, etc and we will typically use
>>>>>> the SDP to signal the parameters that best fit both the
>>>>>> current network and the current type of radio
>>>>>> production.
>>>>>> 
>>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>>> 
>>>>>> 
>>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)" 
>>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>>> 
>>>>>>> My understanding is that the b= line in the SDP is
>>>>>>> quite a subjective parameter. I have seen people use it
>>>>>>> differently over the past several years. So, it is a
>>>>>>> bit difficult to infer much accuracy from that line.
>>>>>>> 
>>>>>>> OTOH, if someone feels like Opus needs to specify some
>>>>>>> sort of bitrate variability in the SDP, it can do so in
>>>>>>> an optional (opus) parameter. Obviously rules around
>>>>>>> how to set that value and what to do with that value
>>>>>>> need to be clearly specified.
>>>>>>> 
>>>>>>> Fredrik (or anyone else in the WG), do you see a strong
>>>>>>> need for this? If so, now is the time to mention that.
>>>>>>> 
>>>>>>> -acbegen
>>>>>>> 
>>>>>>> 
>>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
>>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
>>>>>>>> wrote:
>>>>> I have to say this is not something I understand really
>>>>> well. Does anyone else here have an opinion on how to
>>>>> specify bit-rate?
>>>>> 
>>>>> Jean-Marc
>>>>> 
>>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
>>>>>>>>>> Hi.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> I read in the latest draft that you have removed
>>>>>>>>>> the maxaveragebitrate parameter from the SDP
>>>>>>>>>> signalling for Opus. I think that this parameter
>>>>>>>>>> is useful and that it provides added value over
>>>>>>>>>> the bandwidth modifiers added by RFC 3556.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> I must admit that I haven¹t read all of RFC 3556
>>>>>>>>>> but as I understand it, a b= line are not in any
>>>>>>>>>> way related to an actual format of a media
>>>>>>>>>> description. This means that with this mechanism,
>>>>>>>>>> only one bitrate specification can be sent with
>>>>>>>>>> each media description. Different coding
>>>>>>>>>> algorithms have different constraints on the
>>>>>>>>>> allowed bitrate. For example, I might want to
>>>>>>>>>> send an SDP offer with both Enhanced apt-X at 576
>>>>>>>>>> kbit/s and Opus at 510kbit/s with cbr. Now,
>>>>>>>>>> specifying 576kbit/s via a b= line works for
>>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
>>>>>>>>>> I think that the a=fmtp line is the correct place
>>>>>>>>>> to specify the desired bitrate.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> For the broadcasting use-case, it is often more
>>>>>>>>>> important to have a constant and predictable
>>>>>>>>>> audio quality and network usage than mere
>>>>>>>>>> audibility at any cost.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> Best regards,
>>>>>>>>>> 
>>>>>>>>>> Fredrik Bergholtz
>>>>>>>>>> 
>>>>>>>>>> Swedish Radio
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> payload
> mailing
>>>>>>>>>> list payload@ietf.org <mailto:payload@ietf.org> 
>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> payload
> mailing
>>>>>>>> list payload@ietf.org <mailto:payload@ietf.org> 
>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>> 
>>>>> 
>>>>> _______________________________________________ payload
>>>>> mailing list payload@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/payload
>>> 
>>> _______________________________________________ payload mailing
>>> list payload@ietf.org 
>>> https://www.ietf.org/mailman/listinfo/payload
>>> 
>>> _______________________________________________ payload mailing
>>> list payload@ietf.org 
>>> https://www.ietf.org/mailman/listinfo/payload
>> 
>> _______________________________________________ payload mailing
>> list payload@ietf.org 
>> https://www.ietf.org/mailman/listinfo/payload
>> 
> 
> _______________________________________________ payload mailing
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
> 
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From nobody Mon Jan  5 23:35:10 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Audio/Video Transport Payloads Working Group of the IETF.

        Title           : RTP Payload Format for Opus Speech and Audio Codec
        Authors         : Julian Spittka
                          Koen Vos
                          Jean-Marc Valin
	Filename        : draft-ietf-payload-rtp-opus-06.txt
	Pages           : 17
	Date            : 2015-01-05

Abstract:
   This document defines the Real-time Transport Protocol (RTP) payload
   format for packetization of Opus encoded speech and audio data
   necessary to integrate the codec in the most compatible way.
   Further, it describes media type registrations for the RTP payload
   format.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-opus/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-payload-rtp-opus-06


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

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From nobody Mon Jan  5 23:38:43 2015
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From: Jean-Marc Valin <jmvalin@jmvalin.ca>
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hash: SHA1

Just posted a new version that brings back maxaveragebitrate exactly
as it was before.

Cheers,

	Jean-Marc

On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:
> I think that re-instating maxaveragebitrate would do it for me.
> 
> /Fredrik
> 
> 
>> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca>
>> wrote:
>> 
> So would everyone be happy if I just revert to using
> maxaveragebitrate as in draft 04?
> 
> Jean-Marc
> 
> 
>>>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some other
>>>> codecs have fmtp parameters for maximum bitrate explicitly
>>>> (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly
>>>> (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC; 
>>>> config in AAC).
>>>> 
>>>> Other notable codecs that lack this include VP8 and VP9.
>>>> 
>>>> This is a general problem which may become more important
>>>> with complex applications that include simulcast and scalable
>>>> media within the same media description (m= line). The SDP b=
>>>> parameter is clearly insufficient for expressing multiple
>>>> different maximum bitrates for each payload type, unless we
>>>> define a new bandwidth modifier which includes explicit PT
>>>> bindings. Payload formats that include fmtp parameters for
>>>> maximum bitrate can avoid this problem, so I think it is
>>>> advisable to add this to Opus, VP8 and VP9.
>>>> 
>>>> Mo
>>>> 
>>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
>>>> wrote:
>>>> 
>>>> Setting the bitrate isn't exactly a new problem here. Opus
>>>> isn't the first codec to support more than one rate so how
>>>> are the other codecs (audio and video) handling bitrate
>>>> signaling?
>>>> 
>>>> Cheers,
>>>> 
>>>> Jean-Marc
>>>> 
>>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
>>>> 
>>>>> One thought om the general media level bit rate
>>>>> specification is that it can never be anything other than
>>>>> an approximate value. Different codecs define different
>>>>> constraints on the bit rate. I think that the easiest way
>>>>> to specify exact bit rates would be via the fmtp
>>>>> attribute.
>>>> 
>>>>> This can be compared with the ptime attribute - as it is
>>>>> defined today, only approximate packet times can be
>>>>> signalled in SDP. For the broadcast use case this is not
>>>>> acceptable and for that reason, the European Broadcasting
>>>>> Union (EBU) has defined an SDP extension that allows
>>>>> signalling of a ptime value for each codec. There is no RFC
>>>>> draft for this extension and it is not yet registered with
>>>>> IANA or IETF but it is starting to be used by manufacturers
>>>>> of broadcasting equipment. The extension is known as "ACIP
>>>>> Profiles" and can be found at the EBU web site. But I 
>>>>> digress.
>>>> 
>>>>> I don't think that forcing encoders to approximate the
>>>>> nearest allowed value for the selected codec is less
>>>>> complicated than interpreting an exact value. In addition,
>>>>> it would add a non-deterministic aspect, thus the resulting
>>>>> bit rate may not be what the user expected.
>>>> 
>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>> 
>>>>> On 3 jan 2015, at 11:27, "Parthasarathi R" 
>>>>> <partha@parthasarathi.co.in
>>>>> <mailto:partha@parthasarathi.co.in>> wrote:
>>>> 
>>>>>> Hi all,
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> I  understand that bandwidth ³b² parameter has lot of
>>>>>> variants like AS, TIAS and usage but all those parameters
>>>>>> are media level and not codec specific. The bandwidth
>>>>>> allocation is going to be happen in the media level in
>>>>>> the network or end point based on the media level and not
>>>>>> on the codec specific  in case the multiple codec is
>>>>>> negotiated. In most of the implementation, the multiple
>>>>>> codec (Say Ex: G.711 & Opus) is going to be negotiated
>>>>>> rather than single codec. The codec specific bandwidth
>>>>>> implementation complicates these bandwidth calculation
>>>>>> still further which is applicable to maxaveragebitrate
>>>>>> parameter in opus.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> I could not understand why the generic parameter should
>>>>>> not used or generic parameter has to be enhanced which
>>>>>> meets Opus requirement rather than opus specific
>>>>>> parameter.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Thanks
>>>>>> 
>>>>>> Partha
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
>>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
>>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
>>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
>>>>>> [payload] draft-ietf-payload-rtp-opus-05:
>>>>>> maxaveragebitrate
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Authors
>>>>>> 
>>>>>> Please chime in and lets discuss s quick solution for
>>>>>> this on the list. Once agreed we can ship this draft.
>>>>>> 
>>>>>> -acbegen
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> *From:*Fredrik Bergholtz
>>>>>> <fredrik.bergholtz@sverigesradio.se 
>>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
>>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
>>>>>> Jean-Marc Valin <jmvalin@mozilla.com 
>>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik 
>>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org> 
>>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05: 
>>>>>> maxaveragebitrate
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>>> 
>>>>>> In the broadcasting use case we have to make sure that we
>>>>>> get the best audio quality that is possible for the
>>>>>> network that is currently used but never try to send more
>>>>>> data than the particular link is capable of. For this
>>>>>> reason, specifying a maximum bit rate is required.
>>>>>> 
>>>>>> In addition we often have the requirement that the
>>>>>> quality is constant rather than the best possible. To
>>>>>> avoid as many audible changes in the audio as possible
>>>>>> over time we will often use constant bit rate and then it
>>>>>> becomes even more important to be able to specify what
>>>>>> bit rate should be.
>>>>>> 
>>>>>> We will use the Opus codec over various network links
>>>>>> with different capabilities in terms of bit rate, delay
>>>>>> variation, absolute delay, etc and we will typically use
>>>>>> the SDP to signal the parameters that best fit both the
>>>>>> current network and the current type of radio
>>>>>> production.
>>>>>> 
>>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>>> 
>>>>>> 
>>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)" 
>>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>>> 
>>>>>>> My understanding is that the b= line in the SDP is
>>>>>>> quite a subjective parameter. I have seen people use it
>>>>>>> differently over the past several years. So, it is a
>>>>>>> bit difficult to infer much accuracy from that line.
>>>>>>> 
>>>>>>> OTOH, if someone feels like Opus needs to specify some
>>>>>>> sort of bitrate variability in the SDP, it can do so in
>>>>>>> an optional (opus) parameter. Obviously rules around
>>>>>>> how to set that value and what to do with that value
>>>>>>> need to be clearly specified.
>>>>>>> 
>>>>>>> Fredrik (or anyone else in the WG), do you see a strong
>>>>>>> need for this? If so, now is the time to mention that.
>>>>>>> 
>>>>>>> -acbegen
>>>>>>> 
>>>>>>> 
>>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin 
>>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
>>>>>>>> wrote:
>>>>> I have to say this is not something I understand really
>>>>> well. Does anyone else here have an opinion on how to
>>>>> specify bit-rate?
>>>> 
>>>>> Jean-Marc
>>>> 
>>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
>>>>>>>>>> Hi.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> I read in the latest draft that you have removed
>>>>>>>>>> the maxaveragebitrate parameter from the SDP
>>>>>>>>>> signalling for Opus. I think that this parameter
>>>>>>>>>> is useful and that it provides added value over
>>>>>>>>>> the bandwidth modifiers added by RFC 3556.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> I must admit that I haven¹t read all of RFC 3556
>>>>>>>>>> but as I understand it, a b= line are not in any
>>>>>>>>>> way related to an actual format of a media
>>>>>>>>>> description. This means that with this mechanism,
>>>>>>>>>> only one bitrate specification can be sent with
>>>>>>>>>> each media description. Different coding
>>>>>>>>>> algorithms have different constraints on the
>>>>>>>>>> allowed bitrate. For example, I might want to 
>>>>>>>>>> send an SDP offer with both Enhanced apt-X at
>>>>>>>>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now,
>>>>>>>>>> specifying 576kbit/s via a b= line works for
>>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
>>>>>>>>>> I think that the a=fmtp line is the correct place
>>>>>>>>>> to specify the desired bitrate.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> For the broadcasting use-case, it is often more 
>>>>>>>>>> important to have a constant and predictable
>>>>>>>>>> audio quality and network usage than mere
>>>>>>>>>> audibility at any cost.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> Best regards,
>>>>>>>>>> 
>>>>>>>>>> Fredrik Bergholtz
>>>>>>>>>> 
>>>>>>>>>> Swedish Radio
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________ 
>>>>>>>>>> payload mailing list payload@ietf.org 
>>>>>>>>>> <mailto:payload@ietf.org> 
>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> payload mailing list payload@ietf.org
>>>>>>>> <mailto:payload@ietf.org> 
>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>> 
>>>> 
>>>>> _______________________________________________ payload
>>>>> mailing list payload@ietf.org 
>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>> 
>>>> 
>>>> _______________________________________________ payload
>>>> mailing list payload@ietf.org 
>>>> https://www.ietf.org/mailman/listinfo/payload
>>>> 
>>>> _______________________________________________ payload
>>>> mailing list payload@ietf.org 
>>>> https://www.ietf.org/mailman/listinfo/payload
> 
> _______________________________________________ payload mailing
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
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From: Fredrik Bergholtz <fredrik.bergholtz@sverigesradio.se>
To: Jean-Marc Valin <jmvalin@jmvalin.ca>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Date: Tue, 6 Jan 2015 09:38:00 +0000
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References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com>,<54A9C9D0.8050002@mozilla.com> <06DA8517-E9D8-4726-8FAE-EB7D4B110674@sverigesradio.se>, <54AB90FA.6090900@mozilla.com>
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Excellent. Thanks!

/Fredrik


> On 6 jan 2015, at 08:38, "Jean-Marc Valin" <jmvalin@jmvalin.ca> wrote:
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> Just posted a new version that brings back maxaveragebitrate exactly
> as it was before.
>=20
> Cheers,
>=20
>    Jean-Marc
>=20
>> On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:
>> I think that re-instating maxaveragebitrate would do it for me.
>>=20
>> /Fredrik
>>=20
>>=20
>>> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca>
>>> wrote:
>> So would everyone be happy if I just revert to using
>> maxaveragebitrate as in draft 04?
>>=20
>> Jean-Marc
>>=20
>>=20
>>>>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some other
>>>>> codecs have fmtp parameters for maximum bitrate explicitly
>>>>> (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly
>>>>> (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;=20
>>>>> config in AAC).
>>>>>=20
>>>>> Other notable codecs that lack this include VP8 and VP9.
>>>>>=20
>>>>> This is a general problem which may become more important
>>>>> with complex applications that include simulcast and scalable
>>>>> media within the same media description (m=3D line). The SDP b=3D
>>>>> parameter is clearly insufficient for expressing multiple
>>>>> different maximum bitrates for each payload type, unless we
>>>>> define a new bandwidth modifier which includes explicit PT
>>>>> bindings. Payload formats that include fmtp parameters for
>>>>> maximum bitrate can avoid this problem, so I think it is
>>>>> advisable to add this to Opus, VP8 and VP9.
>>>>>=20
>>>>> Mo
>>>>>=20
>>>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
>>>>> wrote:
>>>>>=20
>>>>> Setting the bitrate isn't exactly a new problem here. Opus
>>>>> isn't the first codec to support more than one rate so how
>>>>> are the other codecs (audio and video) handling bitrate
>>>>> signaling?
>>>>>=20
>>>>> Cheers,
>>>>>=20
>>>>> Jean-Marc
>>>>>=20
>>>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
>>>>>=20
>>>>>> One thought om the general media level bit rate
>>>>>> specification is that it can never be anything other than
>>>>>> an approximate value. Different codecs define different
>>>>>> constraints on the bit rate. I think that the easiest way
>>>>>> to specify exact bit rates would be via the fmtp
>>>>>> attribute.
>>>>>=20
>>>>>> This can be compared with the ptime attribute - as it is
>>>>>> defined today, only approximate packet times can be
>>>>>> signalled in SDP. For the broadcast use case this is not
>>>>>> acceptable and for that reason, the European Broadcasting
>>>>>> Union (EBU) has defined an SDP extension that allows
>>>>>> signalling of a ptime value for each codec. There is no RFC
>>>>>> draft for this extension and it is not yet registered with
>>>>>> IANA or IETF but it is starting to be used by manufacturers
>>>>>> of broadcasting equipment. The extension is known as "ACIP
>>>>>> Profiles" and can be found at the EBU web site. But I=20
>>>>>> digress.
>>>>>=20
>>>>>> I don't think that forcing encoders to approximate the
>>>>>> nearest allowed value for the selected codec is less
>>>>>> complicated than interpreting an exact value. In addition,
>>>>>> it would add a non-deterministic aspect, thus the resulting
>>>>>> bit rate may not be what the user expected.
>>>>>=20
>>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>>=20
>>>>>> On 3 jan 2015, at 11:27, "Parthasarathi R"=20
>>>>>> <partha@parthasarathi.co.in
>>>>>> <mailto:partha@parthasarathi.co.in>> wrote:
>>>>>=20
>>>>>>> Hi all,
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> I  understand that bandwidth =B3b=B2 parameter has lot of
>>>>>>> variants like AS, TIAS and usage but all those parameters
>>>>>>> are media level and not codec specific. The bandwidth
>>>>>>> allocation is going to be happen in the media level in
>>>>>>> the network or end point based on the media level and not
>>>>>>> on the codec specific  in case the multiple codec is
>>>>>>> negotiated. In most of the implementation, the multiple
>>>>>>> codec (Say Ex: G.711 & Opus) is going to be negotiated
>>>>>>> rather than single codec. The codec specific bandwidth
>>>>>>> implementation complicates these bandwidth calculation
>>>>>>> still further which is applicable to maxaveragebitrate
>>>>>>> parameter in opus.
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> I could not understand why the generic parameter should
>>>>>>> not used or generic parameter has to be enhanced which
>>>>>>> meets Opus requirement rather than opus specific
>>>>>>> parameter.
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> Thanks
>>>>>>>=20
>>>>>>> Partha
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
>>>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
>>>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
>>>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
>>>>>>> [payload] draft-ietf-payload-rtp-opus-05:
>>>>>>> maxaveragebitrate
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> Authors
>>>>>>>=20
>>>>>>> Please chime in and lets discuss s quick solution for
>>>>>>> this on the list. Once agreed we can ship this draft.
>>>>>>>=20
>>>>>>> -acbegen
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> *From:*Fredrik Bergholtz
>>>>>>> <fredrik.bergholtz@sverigesradio.se=20
>>>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
>>>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
>>>>>>> Jean-Marc Valin <jmvalin@mozilla.com=20
>>>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik=20
>>>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>=20
>>>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:=20
>>>>>>> maxaveragebitrate
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>>>>=20
>>>>>>> In the broadcasting use case we have to make sure that we
>>>>>>> get the best audio quality that is possible for the
>>>>>>> network that is currently used but never try to send more
>>>>>>> data than the particular link is capable of. For this
>>>>>>> reason, specifying a maximum bit rate is required.
>>>>>>>=20
>>>>>>> In addition we often have the requirement that the
>>>>>>> quality is constant rather than the best possible. To
>>>>>>> avoid as many audible changes in the audio as possible
>>>>>>> over time we will often use constant bit rate and then it
>>>>>>> becomes even more important to be able to specify what
>>>>>>> bit rate should be.
>>>>>>>=20
>>>>>>> We will use the Opus codec over various network links
>>>>>>> with different capabilities in terms of bit rate, delay
>>>>>>> variation, absolute delay, etc and we will typically use
>>>>>>> the SDP to signal the parameters that best fit both the
>>>>>>> current network and the current type of radio
>>>>>>> production.
>>>>>>>=20
>>>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>>>>=20
>>>>>>>=20
>>>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"=20
>>>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>>>>=20
>>>>>>>> My understanding is that the b=3D line in the SDP is
>>>>>>>> quite a subjective parameter. I have seen people use it
>>>>>>>> differently over the past several years. So, it is a
>>>>>>>> bit difficult to infer much accuracy from that line.
>>>>>>>>=20
>>>>>>>> OTOH, if someone feels like Opus needs to specify some
>>>>>>>> sort of bitrate variability in the SDP, it can do so in
>>>>>>>> an optional (opus) parameter. Obviously rules around
>>>>>>>> how to set that value and what to do with that value
>>>>>>>> need to be clearly specified.
>>>>>>>>=20
>>>>>>>> Fredrik (or anyone else in the WG), do you see a strong
>>>>>>>> need for this? If so, now is the time to mention that.
>>>>>>>>=20
>>>>>>>> -acbegen
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin=20
>>>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
>>>>>>>>> wrote:
>>>>>> I have to say this is not something I understand really
>>>>>> well. Does anyone else here have an opinion on how to
>>>>>> specify bit-rate?
>>>>>=20
>>>>>> Jean-Marc
>>>>>=20
>>>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
>>>>>>>>>>> Hi.
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>> I read in the latest draft that you have removed
>>>>>>>>>>> the maxaveragebitrate parameter from the SDP
>>>>>>>>>>> signalling for Opus. I think that this parameter
>>>>>>>>>>> is useful and that it provides added value over
>>>>>>>>>>> the bandwidth modifiers added by RFC 3556.
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>> I must admit that I haven=B9t read all of RFC 3556
>>>>>>>>>>> but as I understand it, a b=3D line are not in any
>>>>>>>>>>> way related to an actual format of a media
>>>>>>>>>>> description. This means that with this mechanism,
>>>>>>>>>>> only one bitrate specification can be sent with
>>>>>>>>>>> each media description. Different coding
>>>>>>>>>>> algorithms have different constraints on the
>>>>>>>>>>> allowed bitrate. For example, I might want to=20
>>>>>>>>>>> send an SDP offer with both Enhanced apt-X at
>>>>>>>>>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now,
>>>>>>>>>>> specifying 576kbit/s via a b=3D line works for
>>>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
>>>>>>>>>>> I think that the a=3Dfmtp line is the correct place
>>>>>>>>>>> to specify the desired bitrate.
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>> For the broadcasting use-case, it is often more=20
>>>>>>>>>>> important to have a constant and predictable
>>>>>>>>>>> audio quality and network usage than mere
>>>>>>>>>>> audibility at any cost.
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>> Best regards,
>>>>>>>>>>>=20
>>>>>>>>>>> Fredrik Bergholtz
>>>>>>>>>>>=20
>>>>>>>>>>> Swedish Radio
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>>=20
>>>>>>>>>>> _______________________________________________=20
>>>>>>>>>>> payload mailing list payload@ietf.org=20
>>>>>>>>>>> <mailto:payload@ietf.org>=20
>>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>>>>=20
>>>>>>>>> _______________________________________________
>>>>>>>>> payload mailing list payload@ietf.org
>>>>>>>>> <mailto:payload@ietf.org>=20
>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>=20
>>>>>=20
>>>>>> _______________________________________________ payload
>>>>>> mailing list payload@ietf.org=20
>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>=20
>>>>>=20
>>>>> _______________________________________________ payload
>>>>> mailing list payload@ietf.org=20
>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>=20
>>>>> _______________________________________________ payload
>>>>> mailing list payload@ietf.org=20
>>>>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>> _______________________________________________ payload mailing
>> list payload@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/payload
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Jean-Marc Valin'" <jmvalin@jmvalin.ca>, "'Fredrik Bergholtz'" <fredrik.bergholtz@sverigesradio.se>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com>, <54A9C9D0.8050002@mozilla.com> <06DA8517-E9D8-4726-8FAE-EB7D4B110674@sverigesradio.se> <54AB90FA.6090900@mozilla.com>
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Date: Tue, 6 Jan 2015 23:18:26 +0530
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Archived-At: http://mailarchive.ietf.org/arch/msg/payload/M-NUR2xrMuJZO5CT0D6Fw57sQXU
Cc: payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hi Jean-Marc/Fredrik/all,

I understand that there is a need for the parameter in the codec level =
and
it is followed as fmtp parameter specific to the given codec=20

1) max-br (RFC 6185)  - H.264
2) maxbitrate (RFC 4749) - G.729.1
3) bitrate (RFC 5577) - G.722.1
3) maxaveragebitrate - Opus?

As per the below mail, this parameter is an open item for VP8 & VP9 and
those codecs will take their own naming convention for their maximum bit
rate.=20

My concern is that it is better to have some standard naming convention =
for
maximum bit rate for the given codec as

1) It is simple to implement in the generic fashion in case of multiple
codec handling in the given SDP. This is the common implementation.=20
2) Offer/answer behavior will be common across multiple codec.
3) Interaction with bandwidth ("b") parameter in the SDP level and these
codec specific max bit rate parameter will be clear in case both exists =
in
the same SDP media line.

Please let me know your opinion on the same.

Thanks
Partha


> -----Original Message-----
> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Jean-Marc
> Valin
> Sent: Tuesday, January 06, 2015 1:09 PM
> To: Fredrik Bergholtz
> Cc: payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:
> maxaveragebitrate
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> Just posted a new version that brings back maxaveragebitrate exactly
> as it was before.
>=20
> Cheers,
>=20
> 	Jean-Marc
>=20
> On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:
> > I think that re-instating maxaveragebitrate would do it for me.
> >
> > /Fredrik
> >
> >
> >> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca>
> >> wrote:
> >>
> > So would everyone be happy if I just revert to using
> > maxaveragebitrate as in draft 04?
> >
> > Jean-Marc
> >
> >
> >>>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some other
> >>>> codecs have fmtp parameters for maximum bitrate explicitly
> >>>> (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly
> >>>> (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;
> >>>> config in AAC).
> >>>>
> >>>> Other notable codecs that lack this include VP8 and VP9.
> >>>>
> >>>> This is a general problem which may become more important
> >>>> with complex applications that include simulcast and scalable
> >>>> media within the same media description (m=3D line). The SDP b=3D
> >>>> parameter is clearly insufficient for expressing multiple
> >>>> different maximum bitrates for each payload type, unless we
> >>>> define a new bandwidth modifier which includes explicit PT
> >>>> bindings. Payload formats that include fmtp parameters for
> >>>> maximum bitrate can avoid this problem, so I think it is
> >>>> advisable to add this to Opus, VP8 and VP9.
> >>>>
> >>>> Mo
> >>>>
> >>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
> >>>> wrote:
> >>>>
> >>>> Setting the bitrate isn't exactly a new problem here. Opus
> >>>> isn't the first codec to support more than one rate so how
> >>>> are the other codecs (audio and video) handling bitrate
> >>>> signaling?
> >>>>
> >>>> Cheers,
> >>>>
> >>>> Jean-Marc
> >>>>
> >>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
> >>>>
> >>>>> One thought om the general media level bit rate
> >>>>> specification is that it can never be anything other than
> >>>>> an approximate value. Different codecs define different
> >>>>> constraints on the bit rate. I think that the easiest way
> >>>>> to specify exact bit rates would be via the fmtp
> >>>>> attribute.
> >>>>
> >>>>> This can be compared with the ptime attribute - as it is
> >>>>> defined today, only approximate packet times can be
> >>>>> signalled in SDP. For the broadcast use case this is not
> >>>>> acceptable and for that reason, the European Broadcasting
> >>>>> Union (EBU) has defined an SDP extension that allows
> >>>>> signalling of a ptime value for each codec. There is no RFC
> >>>>> draft for this extension and it is not yet registered with
> >>>>> IANA or IETF but it is starting to be used by manufacturers
> >>>>> of broadcasting equipment. The extension is known as "ACIP
> >>>>> Profiles" and can be found at the EBU web site. But I
> >>>>> digress.
> >>>>
> >>>>> I don't think that forcing encoders to approximate the
> >>>>> nearest allowed value for the selected codec is less
> >>>>> complicated than interpreting an exact value. In addition,
> >>>>> it would add a non-deterministic aspect, thus the resulting
> >>>>> bit rate may not be what the user expected.
> >>>>
> >>>>> Best regards Fredrik Bergholtz Swedish Radio
> >>>>
> >>>>> On 3 jan 2015, at 11:27, "Parthasarathi R"
> >>>>> <partha@parthasarathi.co.in
> >>>>> <mailto:partha@parthasarathi.co.in>> wrote:
> >>>>
> >>>>>> Hi all,
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> I  understand that bandwidth =B3b=B2 parameter has lot of
> >>>>>> variants like AS, TIAS and usage but all those parameters
> >>>>>> are media level and not codec specific. The bandwidth
> >>>>>> allocation is going to be happen in the media level in
> >>>>>> the network or end point based on the media level and not
> >>>>>> on the codec specific  in case the multiple codec is
> >>>>>> negotiated. In most of the implementation, the multiple
> >>>>>> codec (Say Ex: G.711 & Opus) is going to be negotiated
> >>>>>> rather than single codec. The codec specific bandwidth
> >>>>>> implementation complicates these bandwidth calculation
> >>>>>> still further which is applicable to maxaveragebitrate
> >>>>>> parameter in opus.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> I could not understand why the generic parameter should
> >>>>>> not used or generic parameter has to be enhanced which
> >>>>>> meets Opus requirement rather than opus specific
> >>>>>> parameter.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Thanks
> >>>>>>
> >>>>>> Partha
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
> >>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
> >>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
> >>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
> >>>>>> [payload] draft-ietf-payload-rtp-opus-05:
> >>>>>> maxaveragebitrate
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Authors
> >>>>>>
> >>>>>> Please chime in and lets discuss s quick solution for
> >>>>>> this on the list. Once agreed we can ship this draft.
> >>>>>>
> >>>>>> -acbegen
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> *From:*Fredrik Bergholtz
> >>>>>> <fredrik.bergholtz@sverigesradio.se
> >>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
> >>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
> >>>>>> Jean-Marc Valin <jmvalin@mozilla.com
> >>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik
> >>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
> >>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
> >>>>>> maxaveragebitrate
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Yes, I do see a strong need for specifying the bitrate.
> >>>>>>
> >>>>>> In the broadcasting use case we have to make sure that we
> >>>>>> get the best audio quality that is possible for the
> >>>>>> network that is currently used but never try to send more
> >>>>>> data than the particular link is capable of. For this
> >>>>>> reason, specifying a maximum bit rate is required.
> >>>>>>
> >>>>>> In addition we often have the requirement that the
> >>>>>> quality is constant rather than the best possible. To
> >>>>>> avoid as many audible changes in the audio as possible
> >>>>>> over time we will often use constant bit rate and then it
> >>>>>> becomes even more important to be able to specify what
> >>>>>> bit rate should be.
> >>>>>>
> >>>>>> We will use the Opus codec over various network links
> >>>>>> with different capabilities in terms of bit rate, delay
> >>>>>> variation, absolute delay, etc and we will typically use
> >>>>>> the SDP to signal the parameters that best fit both the
> >>>>>> current network and the current type of radio
> >>>>>> production.
> >>>>>>
> >>>>>> Best regards Fredrik Bergholtz Swedish Radio
> >>>>>>
> >>>>>>
> >>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
> >>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
> >>>>>>>
> >>>>>>> My understanding is that the b=3D line in the SDP is
> >>>>>>> quite a subjective parameter. I have seen people use it
> >>>>>>> differently over the past several years. So, it is a
> >>>>>>> bit difficult to infer much accuracy from that line.
> >>>>>>>
> >>>>>>> OTOH, if someone feels like Opus needs to specify some
> >>>>>>> sort of bitrate variability in the SDP, it can do so in
> >>>>>>> an optional (opus) parameter. Obviously rules around
> >>>>>>> how to set that value and what to do with that value
> >>>>>>> need to be clearly specified.
> >>>>>>>
> >>>>>>> Fredrik (or anyone else in the WG), do you see a strong
> >>>>>>> need for this? If so, now is the time to mention that.
> >>>>>>>
> >>>>>>> -acbegen
> >>>>>>>
> >>>>>>>
> >>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
> >>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
> >>>>>>>> wrote:
> >>>>> I have to say this is not something I understand really
> >>>>> well. Does anyone else here have an opinion on how to
> >>>>> specify bit-rate?
> >>>>
> >>>>> Jean-Marc
> >>>>
> >>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
> >>>>>>>>>> Hi.
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> I read in the latest draft that you have removed
> >>>>>>>>>> the maxaveragebitrate parameter from the SDP
> >>>>>>>>>> signalling for Opus. I think that this parameter
> >>>>>>>>>> is useful and that it provides added value over
> >>>>>>>>>> the bandwidth modifiers added by RFC 3556.
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> I must admit that I haven=B9t read all of RFC 3556
> >>>>>>>>>> but as I understand it, a b=3D line are not in any
> >>>>>>>>>> way related to an actual format of a media
> >>>>>>>>>> description. This means that with this mechanism,
> >>>>>>>>>> only one bitrate specification can be sent with
> >>>>>>>>>> each media description. Different coding
> >>>>>>>>>> algorithms have different constraints on the
> >>>>>>>>>> allowed bitrate. For example, I might want to
> >>>>>>>>>> send an SDP offer with both Enhanced apt-X at
> >>>>>>>>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now,
> >>>>>>>>>> specifying 576kbit/s via a b=3D line works for
> >>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
> >>>>>>>>>> I think that the a=3Dfmtp line is the correct place
> >>>>>>>>>> to specify the desired bitrate.
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> For the broadcasting use-case, it is often more
> >>>>>>>>>> important to have a constant and predictable
> >>>>>>>>>> audio quality and network usage than mere
> >>>>>>>>>> audibility at any cost.
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> Best regards,
> >>>>>>>>>>
> >>>>>>>>>> Fredrik Bergholtz
> >>>>>>>>>>
> >>>>>>>>>> Swedish Radio
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> _______________________________________________
> >>>>>>>>>> payload mailing list payload@ietf.org
> >>>>>>>>>> <mailto:payload@ietf.org>
> >>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>>>
> >>>>>>>> _______________________________________________
> >>>>>>>> payload mailing list payload@ietf.org
> >>>>>>>> <mailto:payload@ietf.org>
> >>>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>
> >>>>
> >>>>> _______________________________________________ payload
> >>>>> mailing list payload@ietf.org
> >>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>
> >>>>
> >>>> _______________________________________________ payload
> >>>> mailing list payload@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>
> >>>> _______________________________________________ payload
> >>>> mailing list payload@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/payload
> >
> > _______________________________________________ payload mailing
> > list payload@ietf.org
> > https://www.ietf.org/mailman/listinfo/payload
> >
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1
>=20
> iQEcBAEBAgAGBQJUq5D3AAoJEJ6/8sItn9q9+XkH/27U71MfStYlp2WXoaiW9SmP
> YSpGdAbDipfnJ+7RZQ0bDf0GCRtxGxyVcAqsj9oYTFQAyMd67eRp88fzY0pRODVf
> VjgWm4VLzP7jbfemzxaZXMzO/MqoiMssIvmnDB2FCirZl03aZ0vcskRAqi/OXlCm
> hVJUWkf8MqxMJ1TKqbLn+WGlWHRk89wsyxYQkbfKTfQHVPmnno8eXGvDt71Gewi1
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> =3Dfe42
> -----END PGP SIGNATURE-----
>=20
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Date: Wed, 7 Jan 2015 12:55:18 +0000
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References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com> <,> <54A9C9D0.8050002@mozilla.com> <06DA8517-E9D8-4726-8FAE-EB7D4B110674@sverigesradio.se> <54AB90FA.6090900@mozilla.com> <002d01d029d8$fd0eba60$f72c2f20$@co.in>
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Cc: Jean-Marc Valin <jmvalin@jmvalin.ca>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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I think it would have been nice if b=3D attribute or something similar was =
used for all codecs in the same fashion, but that did not really work. I ag=
ree that a common payload parameter would seem ideal, however, I dont think=
 it would have been a lot easier to implement it. Most codecs may have exce=
ptions here and there and dealing with such complexities would likely be to=
ugh. So if someone wants to implement codecs a and b, he needs to understan=
d the payload parameters for both. Eliminating only one such parameter woul=
d not make much difference IMO.


> On Jan 6, 2015, at 7:48 PM, Parthasarathi R <partha@parthasarathi.co.in> =
wrote:
>=20
> Hi Jean-Marc/Fredrik/all,
>=20
> I understand that there is a need for the parameter in the codec level an=
d
> it is followed as fmtp parameter specific to the given codec=20
>=20
> 1) max-br (RFC 6185)  - H.264
> 2) maxbitrate (RFC 4749) - G.729.1
> 3) bitrate (RFC 5577) - G.722.1
> 3) maxaveragebitrate - Opus?
>=20
> As per the below mail, this parameter is an open item for VP8 & VP9 and
> those codecs will take their own naming convention for their maximum bit
> rate.=20
>=20
> My concern is that it is better to have some standard naming convention f=
or
> maximum bit rate for the given codec as
>=20
> 1) It is simple to implement in the generic fashion in case of multiple
> codec handling in the given SDP. This is the common implementation.=20
> 2) Offer/answer behavior will be common across multiple codec.
> 3) Interaction with bandwidth ("b") parameter in the SDP level and these
> codec specific max bit rate parameter will be clear in case both exists i=
n
> the same SDP media line.
>=20
> Please let me know your opinion on the same.
>=20
> Thanks
> Partha
>=20
>=20
>> -----Original Message-----
>> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Jean-Marc
>> Valin
>> Sent: Tuesday, January 06, 2015 1:09 PM
>> To: Fredrik Bergholtz
>> Cc: payload@ietf.org
>> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:
>> maxaveragebitrate
>>=20
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>=20
>> Just posted a new version that brings back maxaveragebitrate exactly
>> as it was before.
>>=20
>> Cheers,
>>=20
>> 	Jean-Marc
>>=20
>> On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:
>>> I think that re-instating maxaveragebitrate would do it for me.
>>>=20
>>> /Fredrik
>>>=20
>>>=20
>>>> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca>
>>>> wrote:
>>>>=20
>>> So would everyone be happy if I just revert to using
>>> maxaveragebitrate as in draft 04?
>>>=20
>>> Jean-Marc
>>>=20
>>>=20
>>>>>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some other
>>>>>> codecs have fmtp parameters for maximum bitrate explicitly
>>>>>> (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly
>>>>>> (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;
>>>>>> config in AAC).
>>>>>>=20
>>>>>> Other notable codecs that lack this include VP8 and VP9.
>>>>>>=20
>>>>>> This is a general problem which may become more important
>>>>>> with complex applications that include simulcast and scalable
>>>>>> media within the same media description (m=3D line). The SDP b=3D
>>>>>> parameter is clearly insufficient for expressing multiple
>>>>>> different maximum bitrates for each payload type, unless we
>>>>>> define a new bandwidth modifier which includes explicit PT
>>>>>> bindings. Payload formats that include fmtp parameters for
>>>>>> maximum bitrate can avoid this problem, so I think it is
>>>>>> advisable to add this to Opus, VP8 and VP9.
>>>>>>=20
>>>>>> Mo
>>>>>>=20
>>>>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
>>>>>> wrote:
>>>>>>=20
>>>>>> Setting the bitrate isn't exactly a new problem here. Opus
>>>>>> isn't the first codec to support more than one rate so how
>>>>>> are the other codecs (audio and video) handling bitrate
>>>>>> signaling?
>>>>>>=20
>>>>>> Cheers,
>>>>>>=20
>>>>>> Jean-Marc
>>>>>>=20
>>>>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
>>>>>>=20
>>>>>>> One thought om the general media level bit rate
>>>>>>> specification is that it can never be anything other than
>>>>>>> an approximate value. Different codecs define different
>>>>>>> constraints on the bit rate. I think that the easiest way
>>>>>>> to specify exact bit rates would be via the fmtp
>>>>>>> attribute.
>>>>>>=20
>>>>>>> This can be compared with the ptime attribute - as it is
>>>>>>> defined today, only approximate packet times can be
>>>>>>> signalled in SDP. For the broadcast use case this is not
>>>>>>> acceptable and for that reason, the European Broadcasting
>>>>>>> Union (EBU) has defined an SDP extension that allows
>>>>>>> signalling of a ptime value for each codec. There is no RFC
>>>>>>> draft for this extension and it is not yet registered with
>>>>>>> IANA or IETF but it is starting to be used by manufacturers
>>>>>>> of broadcasting equipment. The extension is known as "ACIP
>>>>>>> Profiles" and can be found at the EBU web site. But I
>>>>>>> digress.
>>>>>>=20
>>>>>>> I don't think that forcing encoders to approximate the
>>>>>>> nearest allowed value for the selected codec is less
>>>>>>> complicated than interpreting an exact value. In addition,
>>>>>>> it would add a non-deterministic aspect, thus the resulting
>>>>>>> bit rate may not be what the user expected.
>>>>>>=20
>>>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>>>=20
>>>>>>> On 3 jan 2015, at 11:27, "Parthasarathi R"
>>>>>>> <partha@parthasarathi.co.in
>>>>>>> <mailto:partha@parthasarathi.co.in>> wrote:
>>>>>>=20
>>>>>>>> Hi all,
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I  understand that bandwidth =B3b=B2 parameter has lot of
>>>>>>>> variants like AS, TIAS and usage but all those parameters
>>>>>>>> are media level and not codec specific. The bandwidth
>>>>>>>> allocation is going to be happen in the media level in
>>>>>>>> the network or end point based on the media level and not
>>>>>>>> on the codec specific  in case the multiple codec is
>>>>>>>> negotiated. In most of the implementation, the multiple
>>>>>>>> codec (Say Ex: G.711 & Opus) is going to be negotiated
>>>>>>>> rather than single codec. The codec specific bandwidth
>>>>>>>> implementation complicates these bandwidth calculation
>>>>>>>> still further which is applicable to maxaveragebitrate
>>>>>>>> parameter in opus.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I could not understand why the generic parameter should
>>>>>>>> not used or generic parameter has to be enhanced which
>>>>>>>> meets Opus requirement rather than opus specific
>>>>>>>> parameter.
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> Thanks
>>>>>>>>=20
>>>>>>>> Partha
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
>>>>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
>>>>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
>>>>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
>>>>>>>> [payload] draft-ietf-payload-rtp-opus-05:
>>>>>>>> maxaveragebitrate
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> Authors
>>>>>>>>=20
>>>>>>>> Please chime in and lets discuss s quick solution for
>>>>>>>> this on the list. Once agreed we can ship this draft.
>>>>>>>>=20
>>>>>>>> -acbegen
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> *From:*Fredrik Bergholtz
>>>>>>>> <fredrik.bergholtz@sverigesradio.se
>>>>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
>>>>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
>>>>>>>> Jean-Marc Valin <jmvalin@mozilla.com
>>>>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik
>>>>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
>>>>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
>>>>>>>> maxaveragebitrate
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> Yes, I do see a strong need for specifying the bitrate.
>>>>>>>>=20
>>>>>>>> In the broadcasting use case we have to make sure that we
>>>>>>>> get the best audio quality that is possible for the
>>>>>>>> network that is currently used but never try to send more
>>>>>>>> data than the particular link is capable of. For this
>>>>>>>> reason, specifying a maximum bit rate is required.
>>>>>>>>=20
>>>>>>>> In addition we often have the requirement that the
>>>>>>>> quality is constant rather than the best possible. To
>>>>>>>> avoid as many audible changes in the audio as possible
>>>>>>>> over time we will often use constant bit rate and then it
>>>>>>>> becomes even more important to be able to specify what
>>>>>>>> bit rate should be.
>>>>>>>>=20
>>>>>>>> We will use the Opus codec over various network links
>>>>>>>> with different capabilities in terms of bit rate, delay
>>>>>>>> variation, absolute delay, etc and we will typically use
>>>>>>>> the SDP to signal the parameters that best fit both the
>>>>>>>> current network and the current type of radio
>>>>>>>> production.
>>>>>>>>=20
>>>>>>>> Best regards Fredrik Bergholtz Swedish Radio
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
>>>>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
>>>>>>>>>=20
>>>>>>>>> My understanding is that the b=3D line in the SDP is
>>>>>>>>> quite a subjective parameter. I have seen people use it
>>>>>>>>> differently over the past several years. So, it is a
>>>>>>>>> bit difficult to infer much accuracy from that line.
>>>>>>>>>=20
>>>>>>>>> OTOH, if someone feels like Opus needs to specify some
>>>>>>>>> sort of bitrate variability in the SDP, it can do so in
>>>>>>>>> an optional (opus) parameter. Obviously rules around
>>>>>>>>> how to set that value and what to do with that value
>>>>>>>>> need to be clearly specified.
>>>>>>>>>=20
>>>>>>>>> Fredrik (or anyone else in the WG), do you see a strong
>>>>>>>>> need for this? If so, now is the time to mention that.
>>>>>>>>>=20
>>>>>>>>> -acbegen
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
>>>>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
>>>>>>>>>> wrote:
>>>>>>> I have to say this is not something I understand really
>>>>>>> well. Does anyone else here have an opinion on how to
>>>>>>> specify bit-rate?
>>>>>>=20
>>>>>>> Jean-Marc
>>>>>>=20
>>>>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
>>>>>>>>>>>> Hi.
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>> I read in the latest draft that you have removed
>>>>>>>>>>>> the maxaveragebitrate parameter from the SDP
>>>>>>>>>>>> signalling for Opus. I think that this parameter
>>>>>>>>>>>> is useful and that it provides added value over
>>>>>>>>>>>> the bandwidth modifiers added by RFC 3556.
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>> I must admit that I haven=B9t read all of RFC 3556
>>>>>>>>>>>> but as I understand it, a b=3D line are not in any
>>>>>>>>>>>> way related to an actual format of a media
>>>>>>>>>>>> description. This means that with this mechanism,
>>>>>>>>>>>> only one bitrate specification can be sent with
>>>>>>>>>>>> each media description. Different coding
>>>>>>>>>>>> algorithms have different constraints on the
>>>>>>>>>>>> allowed bitrate. For example, I might want to
>>>>>>>>>>>> send an SDP offer with both Enhanced apt-X at
>>>>>>>>>>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now,
>>>>>>>>>>>> specifying 576kbit/s via a b=3D line works for
>>>>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
>>>>>>>>>>>> I think that the a=3Dfmtp line is the correct place
>>>>>>>>>>>> to specify the desired bitrate.
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>> For the broadcasting use-case, it is often more
>>>>>>>>>>>> important to have a constant and predictable
>>>>>>>>>>>> audio quality and network usage than mere
>>>>>>>>>>>> audibility at any cost.
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>> Best regards,
>>>>>>>>>>>>=20
>>>>>>>>>>>> Fredrik Bergholtz
>>>>>>>>>>>>=20
>>>>>>>>>>>> Swedish Radio
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>>=20
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> payload mailing list payload@ietf.org
>>>>>>>>>>>> <mailto:payload@ietf.org>
>>>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>>>>>=20
>>>>>>>>>> _______________________________________________
>>>>>>>>>> payload mailing list payload@ietf.org
>>>>>>>>>> <mailto:payload@ietf.org>
>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>=20
>>>>>>=20
>>>>>>> _______________________________________________ payload
>>>>>>> mailing list payload@ietf.org
>>>>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>>>=20
>>>>>>=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
>>> _______________________________________________ payload mailing
>>> list payload@ietf.org
>>> https://www.ietf.org/mailman/listinfo/payload
>>>=20
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1
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>> =3Dfe42
>> -----END PGP SIGNATURE-----
>>=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


From nobody Wed Jan  7 16:42:23 2015
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>
References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com> <, > <54A9C9D0.8050002@mozilla.com> <06DA8517-E9D8-4726-8FAE-EB7D4B110674@sverigesradio.se> <54AB90FA.6090900@mozilla.com> <002d01d029d8$fd0eba60$f72c2f20$@co.in> <6F2D4628-1540-45B7-B31F-FAE7D841860F@cisco.com>
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Cc: 'Jean-Marc Valin' <jmvalin@jmvalin.ca>, payload@ietf.org
Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hi Ali,

I'm fine in case it is considered as implementers issue.

These bitrate related parameters are optional parameters for the given =
codec
and each implementor has its his own choice to implement or ignore.

Thanks
Partha

> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]
> Sent: Wednesday, January 07, 2015 6:25 PM
> To: Parthasarathi R
> Cc: Jean-Marc Valin; Fredrik Bergholtz; payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:
> maxaveragebitrate
>=20
> I think it would have been nice if b=3D attribute or something similar
> was used for all codecs in the same fashion, but that did not really
> work. I agree that a common payload parameter would seem ideal,
> however, I dont think it would have been a lot easier to implement it.
> Most codecs may have exceptions here and there and dealing with such
> complexities would likely be tough. So if someone wants to implement
> codecs a and b, he needs to understand the payload parameters for =
both.
> Eliminating only one such parameter would not make much difference =
IMO.
>=20
>=20
> > On Jan 6, 2015, at 7:48 PM, Parthasarathi R
> <partha@parthasarathi.co.in> wrote:
> >
> > Hi Jean-Marc/Fredrik/all,
> >
> > I understand that there is a need for the parameter in the codec
> level and
> > it is followed as fmtp parameter specific to the given codec
> >
> > 1) max-br (RFC 6185)  - H.264
> > 2) maxbitrate (RFC 4749) - G.729.1
> > 3) bitrate (RFC 5577) - G.722.1
> > 3) maxaveragebitrate - Opus?
> >
> > As per the below mail, this parameter is an open item for VP8 & VP9
> and
> > those codecs will take their own naming convention for their maximum
> bit
> > rate.
> >
> > My concern is that it is better to have some standard naming
> convention for
> > maximum bit rate for the given codec as
> >
> > 1) It is simple to implement in the generic fashion in case of
> multiple
> > codec handling in the given SDP. This is the common implementation.
> > 2) Offer/answer behavior will be common across multiple codec.
> > 3) Interaction with bandwidth ("b") parameter in the SDP level and
> these
> > codec specific max bit rate parameter will be clear in case both
> exists in
> > the same SDP media line.
> >
> > Please let me know your opinion on the same.
> >
> > Thanks
> > Partha
> >
> >
> >> -----Original Message-----
> >> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Jean-
> Marc
> >> Valin
> >> Sent: Tuesday, January 06, 2015 1:09 PM
> >> To: Fredrik Bergholtz
> >> Cc: payload@ietf.org
> >> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:
> >> maxaveragebitrate
> >>
> >> -----BEGIN PGP SIGNED MESSAGE-----
> >> Hash: SHA1
> >>
> >> Just posted a new version that brings back maxaveragebitrate =
exactly
> >> as it was before.
> >>
> >> Cheers,
> >>
> >> 	Jean-Marc
> >>
> >> On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:
> >>> I think that re-instating maxaveragebitrate would do it for me.
> >>>
> >>> /Fredrik
> >>>
> >>>
> >>>> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca>
> >>>> wrote:
> >>>>
> >>> So would everyone be happy if I just revert to using
> >>> maxaveragebitrate as in draft 04?
> >>>
> >>> Jean-Marc
> >>>
> >>>
> >>>>>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some other
> >>>>>> codecs have fmtp parameters for maximum bitrate explicitly
> >>>>>> (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly
> >>>>>> (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;
> >>>>>> config in AAC).
> >>>>>>
> >>>>>> Other notable codecs that lack this include VP8 and VP9.
> >>>>>>
> >>>>>> This is a general problem which may become more important
> >>>>>> with complex applications that include simulcast and scalable
> >>>>>> media within the same media description (m=3D line). The SDP =
b=3D
> >>>>>> parameter is clearly insufficient for expressing multiple
> >>>>>> different maximum bitrates for each payload type, unless we
> >>>>>> define a new bandwidth modifier which includes explicit PT
> >>>>>> bindings. Payload formats that include fmtp parameters for
> >>>>>> maximum bitrate can avoid this problem, so I think it is
> >>>>>> advisable to add this to Opus, VP8 and VP9.
> >>>>>>
> >>>>>> Mo
> >>>>>>
> >>>>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
> >>>>>> wrote:
> >>>>>>
> >>>>>> Setting the bitrate isn't exactly a new problem here. Opus
> >>>>>> isn't the first codec to support more than one rate so how
> >>>>>> are the other codecs (audio and video) handling bitrate
> >>>>>> signaling?
> >>>>>>
> >>>>>> Cheers,
> >>>>>>
> >>>>>> Jean-Marc
> >>>>>>
> >>>>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
> >>>>>>
> >>>>>>> One thought om the general media level bit rate
> >>>>>>> specification is that it can never be anything other than
> >>>>>>> an approximate value. Different codecs define different
> >>>>>>> constraints on the bit rate. I think that the easiest way
> >>>>>>> to specify exact bit rates would be via the fmtp
> >>>>>>> attribute.
> >>>>>>
> >>>>>>> This can be compared with the ptime attribute - as it is
> >>>>>>> defined today, only approximate packet times can be
> >>>>>>> signalled in SDP. For the broadcast use case this is not
> >>>>>>> acceptable and for that reason, the European Broadcasting
> >>>>>>> Union (EBU) has defined an SDP extension that allows
> >>>>>>> signalling of a ptime value for each codec. There is no RFC
> >>>>>>> draft for this extension and it is not yet registered with
> >>>>>>> IANA or IETF but it is starting to be used by manufacturers
> >>>>>>> of broadcasting equipment. The extension is known as "ACIP
> >>>>>>> Profiles" and can be found at the EBU web site. But I
> >>>>>>> digress.
> >>>>>>
> >>>>>>> I don't think that forcing encoders to approximate the
> >>>>>>> nearest allowed value for the selected codec is less
> >>>>>>> complicated than interpreting an exact value. In addition,
> >>>>>>> it would add a non-deterministic aspect, thus the resulting
> >>>>>>> bit rate may not be what the user expected.
> >>>>>>
> >>>>>>> Best regards Fredrik Bergholtz Swedish Radio
> >>>>>>
> >>>>>>> On 3 jan 2015, at 11:27, "Parthasarathi R"
> >>>>>>> <partha@parthasarathi.co.in
> >>>>>>> <mailto:partha@parthasarathi.co.in>> wrote:
> >>>>>>
> >>>>>>>> Hi all,
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> I  understand that bandwidth =B3b=B2 parameter has lot of
> >>>>>>>> variants like AS, TIAS and usage but all those parameters
> >>>>>>>> are media level and not codec specific. The bandwidth
> >>>>>>>> allocation is going to be happen in the media level in
> >>>>>>>> the network or end point based on the media level and not
> >>>>>>>> on the codec specific  in case the multiple codec is
> >>>>>>>> negotiated. In most of the implementation, the multiple
> >>>>>>>> codec (Say Ex: G.711 & Opus) is going to be negotiated
> >>>>>>>> rather than single codec. The codec specific bandwidth
> >>>>>>>> implementation complicates these bandwidth calculation
> >>>>>>>> still further which is applicable to maxaveragebitrate
> >>>>>>>> parameter in opus.
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> I could not understand why the generic parameter should
> >>>>>>>> not used or generic parameter has to be enhanced which
> >>>>>>>> meets Opus requirement rather than opus specific
> >>>>>>>> parameter.
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> Thanks
> >>>>>>>>
> >>>>>>>> Partha
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
> >>>>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
> >>>>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
> >>>>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
> >>>>>>>> [payload] draft-ietf-payload-rtp-opus-05:
> >>>>>>>> maxaveragebitrate
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> Authors
> >>>>>>>>
> >>>>>>>> Please chime in and lets discuss s quick solution for
> >>>>>>>> this on the list. Once agreed we can ship this draft.
> >>>>>>>>
> >>>>>>>> -acbegen
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> *From:*Fredrik Bergholtz
> >>>>>>>> <fredrik.bergholtz@sverigesradio.se
> >>>>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
> >>>>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
> >>>>>>>> Jean-Marc Valin <jmvalin@mozilla.com
> >>>>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik
> >>>>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
> >>>>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
> >>>>>>>> maxaveragebitrate
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> Yes, I do see a strong need for specifying the bitrate.
> >>>>>>>>
> >>>>>>>> In the broadcasting use case we have to make sure that we
> >>>>>>>> get the best audio quality that is possible for the
> >>>>>>>> network that is currently used but never try to send more
> >>>>>>>> data than the particular link is capable of. For this
> >>>>>>>> reason, specifying a maximum bit rate is required.
> >>>>>>>>
> >>>>>>>> In addition we often have the requirement that the
> >>>>>>>> quality is constant rather than the best possible. To
> >>>>>>>> avoid as many audible changes in the audio as possible
> >>>>>>>> over time we will often use constant bit rate and then it
> >>>>>>>> becomes even more important to be able to specify what
> >>>>>>>> bit rate should be.
> >>>>>>>>
> >>>>>>>> We will use the Opus codec over various network links
> >>>>>>>> with different capabilities in terms of bit rate, delay
> >>>>>>>> variation, absolute delay, etc and we will typically use
> >>>>>>>> the SDP to signal the parameters that best fit both the
> >>>>>>>> current network and the current type of radio
> >>>>>>>> production.
> >>>>>>>>
> >>>>>>>> Best regards Fredrik Bergholtz Swedish Radio
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
> >>>>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
> >>>>>>>>>
> >>>>>>>>> My understanding is that the b=3D line in the SDP is
> >>>>>>>>> quite a subjective parameter. I have seen people use it
> >>>>>>>>> differently over the past several years. So, it is a
> >>>>>>>>> bit difficult to infer much accuracy from that line.
> >>>>>>>>>
> >>>>>>>>> OTOH, if someone feels like Opus needs to specify some
> >>>>>>>>> sort of bitrate variability in the SDP, it can do so in
> >>>>>>>>> an optional (opus) parameter. Obviously rules around
> >>>>>>>>> how to set that value and what to do with that value
> >>>>>>>>> need to be clearly specified.
> >>>>>>>>>
> >>>>>>>>> Fredrik (or anyone else in the WG), do you see a strong
> >>>>>>>>> need for this? If so, now is the time to mention that.
> >>>>>>>>>
> >>>>>>>>> -acbegen
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
> >>>>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
> >>>>>>>>>> wrote:
> >>>>>>> I have to say this is not something I understand really
> >>>>>>> well. Does anyone else here have an opinion on how to
> >>>>>>> specify bit-rate?
> >>>>>>
> >>>>>>> Jean-Marc
> >>>>>>
> >>>>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
> >>>>>>>>>>>> Hi.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> I read in the latest draft that you have removed
> >>>>>>>>>>>> the maxaveragebitrate parameter from the SDP
> >>>>>>>>>>>> signalling for Opus. I think that this parameter
> >>>>>>>>>>>> is useful and that it provides added value over
> >>>>>>>>>>>> the bandwidth modifiers added by RFC 3556.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> I must admit that I haven=B9t read all of RFC 3556
> >>>>>>>>>>>> but as I understand it, a b=3D line are not in any
> >>>>>>>>>>>> way related to an actual format of a media
> >>>>>>>>>>>> description. This means that with this mechanism,
> >>>>>>>>>>>> only one bitrate specification can be sent with
> >>>>>>>>>>>> each media description. Different coding
> >>>>>>>>>>>> algorithms have different constraints on the
> >>>>>>>>>>>> allowed bitrate. For example, I might want to
> >>>>>>>>>>>> send an SDP offer with both Enhanced apt-X at
> >>>>>>>>>>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now,
> >>>>>>>>>>>> specifying 576kbit/s via a b=3D line works for
> >>>>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
> >>>>>>>>>>>> I think that the a=3Dfmtp line is the correct place
> >>>>>>>>>>>> to specify the desired bitrate.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> For the broadcasting use-case, it is often more
> >>>>>>>>>>>> important to have a constant and predictable
> >>>>>>>>>>>> audio quality and network usage than mere
> >>>>>>>>>>>> audibility at any cost.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> Best regards,
> >>>>>>>>>>>>
> >>>>>>>>>>>> Fredrik Bergholtz
> >>>>>>>>>>>>
> >>>>>>>>>>>> Swedish Radio
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> _______________________________________________
> >>>>>>>>>>>> payload mailing list payload@ietf.org
> >>>>>>>>>>>> <mailto:payload@ietf.org>
> >>>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>>>>>
> >>>>>>>>>> _______________________________________________
> >>>>>>>>>> payload mailing list payload@ietf.org
> >>>>>>>>>> <mailto:payload@ietf.org>
> >>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>
> >>>>>>
> >>>>>>> _______________________________________________ payload
> >>>>>>> mailing list payload@ietf.org
> >>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________ payload
> >>>>>> mailing list payload@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>
> >>>>>> _______________________________________________ payload
> >>>>>> mailing list payload@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>
> >>> _______________________________________________ payload mailing
> >>> list payload@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/payload
> >>>
> >> -----BEGIN PGP SIGNATURE-----
> >> Version: GnuPG v1
> >>
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> >> -----END PGP SIGNATURE-----
> >>
> >> _______________________________________________
> >> payload mailing list
> >> payload@ietf.org
> >> https://www.ietf.org/mailman/listinfo/payload
> >
> > _______________________________________________
> > payload mailing list
> > payload@ietf.org
> > https://www.ietf.org/mailman/listinfo/payload


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There wasn't much opportunity to talk about this at IETF 91, so I wanted to
mention that there is now a draft for RTP packetization for the VP9 video
codec, https://tools.ietf.org/html/draft-uberti-payload-vp9-00.

VP9 supports both temporal and spatial scalability features, and the
proposed packetization is designed to handle this. Please take a look and
send comments.

--001a1135f22c4b5079050c2772f3
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><span style=3D"font-size:12.8000001907349px">There wasn&#3=
9;t much opportunity to talk about this at IETF 91, so I wanted to mention =
that there is now a draft for RTP packetization for the VP9 video codec,=C2=
=A0</span><a href=3D"https://tools.ietf.org/html/draft-uberti-payload-vp9-0=
0" target=3D"_blank" style=3D"font-size:12.8000001907349px">https://tools.i=
etf.org/html/draft-uberti-payload-vp9-00</a><span style=3D"font-size:12.800=
0001907349px">.=C2=A0</span><div style=3D"font-size:12.8000001907349px"><br=
></div><div style=3D"font-size:12.8000001907349px">VP9 supports both tempor=
al and spatial scalability features, and the proposed packetization is desi=
gned to handle this. Please take a look and send comments.</div></div>

--001a1135f22c4b5079050c2772f3--


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There wasn't much opportunity to talk about this at IETF 91, so I wanted to
mention that there is now a draft for RTP packetization for the VP9 video
codec, https://tools.ietf.org/html/draft-uberti-payload-vp9-00.

VP9 supports both temporal and spatial scalability features, and the
proposed packetization is designed to handle this. Please take a look and
send comments.

--bcaec548a19d0483f6050c276a35
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<div dir=3D"ltr">There wasn&#39;t much opportunity to talk about this at IE=
TF 91, so I wanted to mention that there is now a draft for RTP packetizati=
on for the VP9 video codec, <a href=3D"https://tools.ietf.org/html/draft-ub=
erti-payload-vp9-00">https://tools.ietf.org/html/draft-uberti-payload-vp9-0=
0</a>.=C2=A0<div><br></div><div>VP9 supports both temporal and spatial scal=
ability features, and the proposed packetization is designed to handle this=
. Please take a look and send comments.<br></div></div>

--bcaec548a19d0483f6050c276a35--


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From: IKRAM UDDIN <ikramuddin205@yahoo.com>
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Subject: [payload]  draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Hi,
Check RFC5577 and RFC4749 for bitrate parameter (for voice)=C2=A0
Regards,
Ikram Ud Din
InterNetworks Research Laboratory,Universiti Utara Malaysia.

I'm fine in case it is considered as implementers issue.

These bitrate related parameters are optional parameters for the given code=
c
and each implementor has its his own choice to implement or ignore.

Thanks
Partha

> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]
> Sent: Wednesday, January 07, 2015 6:25 PM
> To: Parthasarathi R
> Cc: Jean-Marc Valin; Fredrik Bergholtz; payload@ietf.org
> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:
> maxaveragebitrate
>=20
> I think it would have been nice if b=3D attribute or something similar
> was used for all codecs in the same fashion, but that did not really
> work. I agree that a common payload parameter would seem ideal,
> however, I dont think it would have been a lot easier to implement it.
> Most codecs may have exceptions here and there and dealing with such
> complexities would likely be tough. So if someone wants to implement
> codecs a and b, he needs to understand the payload parameters for both.
> Eliminating only one such parameter would not make much difference IMO.
>=20
>=20
> > On Jan 6, 2015, at 7:48 PM, Parthasarathi R
> <partha@parthasarathi.co.in> wrote:
> >
> > Hi Jean-Marc/Fredrik/all,
> >
> > I understand that there is a need for the parameter in the codec
> level and
> > it is followed as fmtp parameter specific to the given codec
> >
> > 1) max-br (RFC 6185)=C2=A0 - H.264
> > 2) maxbitrate (RFC 4749) - G.729.1
> > 3) bitrate (RFC 5577) - G.722.1
> > 3) maxaveragebitrate - Opus?
> >
> > As per the below mail, this parameter is an open item for VP8 & VP9
> and
> > those codecs will take their own naming convention for their maximum
> bit
> > rate.
> >
> > My concern is that it is better to have some standard naming
> convention for
> > maximum bit rate for the given codec as
> >
> > 1) It is simple to implement in the generic fashion in case of
> multiple
> > codec handling in the given SDP. This is the common implementation.
> > 2) Offer/answer behavior will be common across multiple codec.
> > 3) Interaction with bandwidth ("b") parameter in the SDP level and
> these
> > codec specific max bit rate parameter will be clear in case both
> exists in
> > the same SDP media line.
> >
> > Please let me know your opinion on the same.
> >
> > Thanks
> > Partha
> >
> >
> >> -----Original Message-----
> >> From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Jean-
> Marc
> >> Valin
> >> Sent: Tuesday, January 06, 2015 1:09 PM
> >> To: Fredrik Bergholtz
> >> Cc: payload@ietf.org
> >> Subject: Re: [payload] draft-ietf-payload-rtp-opus-05:
> >> maxaveragebitrate
> >>
> >> -----BEGIN PGP SIGNED MESSAGE-----
> >> Hash: SHA1
> >>
> >> Just posted a new version that brings back maxaveragebitrate exactly
> >> as it was before.
> >>
> >> Cheers,
> >>
> >> =C2=A0=C2=A0=C2=A0 Jean-Marc
> >>
> >> On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:
> >>> I think that re-instating maxaveragebitrate would do it for me.
> >>>
> >>> /Fredrik
> >>>
> >>>
> >>>> On 5 jan 2015, at 00:16, "Jean-Marc Valin" <jmvalin@jmvalin.ca>
> >>>> wrote:
> >>>>
> >>> So would everyone be happy if I just revert to using
> >>> maxaveragebitrate as in draft 04?
> >>>
> >>> Jean-Marc
> >>>
> >>>
> >>>>>> On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some other
> >>>>>> codecs have fmtp parameters for maximum bitrate explicitly
> >>>>>> (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly
> >>>>>> (e.g. profile-level-id in H264 / H264-SVC / H265 / AAC;
> >>>>>> config in AAC).
> >>>>>>
> >>>>>> Other notable codecs that lack this include VP8 and VP9.
> >>>>>>
> >>>>>> This is a general problem which may become more important
> >>>>>> with complex applications that include simulcast and scalable
> >>>>>> media within the same media description (m=3D line). The SDP b=3D
> >>>>>> parameter is clearly insufficient for expressing multiple
> >>>>>> different maximum bitrates for each payload type, unless we
> >>>>>> define a new bandwidth modifier which includes explicit PT
> >>>>>> bindings. Payload formats that include fmtp parameters for
> >>>>>> maximum bitrate can avoid this problem, so I think it is
> >>>>>> advisable to add this to Opus, VP8 and VP9.
> >>>>>>
> >>>>>> Mo
> >>>>>>
> >>>>>> On 1/4/15, 2:45 PM, Jean-Marc Valin <jmvalin@jmvalin.ca>
> >>>>>> wrote:
> >>>>>>
> >>>>>> Setting the bitrate isn't exactly a new problem here. Opus
> >>>>>> isn't the first codec to support more than one rate so how
> >>>>>> are the other codecs (audio and video) handling bitrate
> >>>>>> signaling?
> >>>>>>
> >>>>>> Cheers,
> >>>>>>
> >>>>>> Jean-Marc
> >>>>>>
> >>>>>>> On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.
> >>>>>>
> >>>>>>> One thought om the general media level bit rate
> >>>>>>> specification is that it can never be anything other than
> >>>>>>> an approximate value. Different codecs define different
> >>>>>>> constraints on the bit rate. I think that the easiest way
> >>>>>>> to specify exact bit rates would be via the fmtp
> >>>>>>> attribute.
> >>>>>>
> >>>>>>> This can be compared with the ptime attribute - as it is
> >>>>>>> defined today, only approximate packet times can be
> >>>>>>> signalled in SDP. For the broadcast use case this is not
> >>>>>>> acceptable and for that reason, the European Broadcasting
> >>>>>>> Union (EBU) has defined an SDP extension that allows
> >>>>>>> signalling of a ptime value for each codec. There is no RFC
> >>>>>>> draft for this extension and it is not yet registered with
> >>>>>>> IANA or IETF but it is starting to be used by manufacturers
> >>>>>>> of broadcasting equipment. The extension is known as "ACIP
> >>>>>>> Profiles" and can be found at the EBU web site. But I
> >>>>>>> digress.
> >>>>>>
> >>>>>>> I don't think that forcing encoders to approximate the
> >>>>>>> nearest allowed value for the selected codec is less
> >>>>>>> complicated than interpreting an exact value. In addition,
> >>>>>>> it would add a non-deterministic aspect, thus the resulting
> >>>>>>> bit rate may not be what the user expected.
> >>>>>>
> >>>>>>> Best regards Fredrik Bergholtz Swedish Radio
> >>>>>>
> >>>>>>> On 3 jan 2015, at 11:27, "Parthasarathi R"
> >>>>>>> <partha@parthasarathi.co.in
> >>>>>>> <mailto:partha@parthasarathi.co.in>> wrote:
> >>>>>>
> >>>>>>>> Hi all,
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> I=C2=A0 understand that bandwidth =C2=B3b=C2=B2 parameter has lo=
t of
> >>>>>>>> variants like AS, TIAS and usage but all those parameters
> >>>>>>>> are media level and not codec specific. The bandwidth
> >>>>>>>> allocation is going to be happen in the media level in
> >>>>>>>> the network or end point based on the media level and not
> >>>>>>>> on the codec specific=C2=A0 in case the multiple codec is
> >>>>>>>> negotiated. In most of the implementation, the multiple
> >>>>>>>> codec (Say Ex: G.711 & Opus) is going to be negotiated
> >>>>>>>> rather than single codec. The codec specific bandwidth
> >>>>>>>> implementation complicates these bandwidth calculation
> >>>>>>>> still further which is applicable to maxaveragebitrate
> >>>>>>>> parameter in opus.
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> I could not understand why the generic parameter should
> >>>>>>>> not used or generic parameter has to be enhanced which
> >>>>>>>> meets Opus requirement rather than opus specific
> >>>>>>>> parameter.
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> Thanks
> >>>>>>>>
> >>>>>>>> Partha
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> *From:*payload [mailto:payload-bounces@ietf.org] *On
> >>>>>>>> Behalf Of *Ali C. Begen (abegen) *Sent:* Wednesday,
> >>>>>>>> December 31, 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*
> >>>>>>>> payload@ietf.org <mailto:payload@ietf.org> *Subject:* Re:
> >>>>>>>> [payload] draft-ietf-payload-rtp-opus-05:
> >>>>>>>> maxaveragebitrate
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> Authors
> >>>>>>>>
> >>>>>>>> Please chime in and lets discuss s quick solution for
> >>>>>>>> this on the list. Once agreed we can ship this draft.
> >>>>>>>>
> >>>>>>>> -acbegen
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> *From:*Fredrik Bergholtz
> >>>>>>>> <fredrik.bergholtz@sverigesradio.se
> >>>>>>>> <mailto:fredrik.bergholtz@sverigesradio.se>> *Sent:* Dec
> >>>>>>>> 31, 2014 11:42 AM *To:* Ali C. Begen (abegen) *Cc:*
> >>>>>>>> Jean-Marc Valin <jmvalin@mozilla.com
> >>>>>>>> <mailto:jmvalin@mozilla.com>>;Fredrik
> >>>>>>>> Bergholtz;payload@ietf.org <mailto:payload@ietf.org>
> >>>>>>>> *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:
> >>>>>>>> maxaveragebitrate
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> Yes, I do see a strong need for specifying the bitrate.
> >>>>>>>>
> >>>>>>>> In the broadcasting use case we have to make sure that we
> >>>>>>>> get the best audio quality that is possible for the
> >>>>>>>> network that is currently used but never try to send more
> >>>>>>>> data than the particular link is capable of. For this
> >>>>>>>> reason, specifying a maximum bit rate is required.
> >>>>>>>>
> >>>>>>>> In addition we often have the requirement that the
> >>>>>>>> quality is constant rather than the best possible. To
> >>>>>>>> avoid as many audible changes in the audio as possible
> >>>>>>>> over time we will often use constant bit rate and then it
> >>>>>>>> becomes even more important to be able to specify what
> >>>>>>>> bit rate should be.
> >>>>>>>>
> >>>>>>>> We will use the Opus codec over various network links
> >>>>>>>> with different capabilities in terms of bit rate, delay
> >>>>>>>> variation, absolute delay, etc and we will typically use
> >>>>>>>> the SDP to signal the parameters that best fit both the
> >>>>>>>> current network and the current type of radio
> >>>>>>>> production.
> >>>>>>>>
> >>>>>>>> Best regards Fredrik Bergholtz Swedish Radio
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>> On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"
> >>>>>>>>> <abegen@cisco.com <mailto:abegen@cisco.com>> wrote:
> >>>>>>>>>
> >>>>>>>>> My understanding is that the b=3D line in the SDP is
> >>>>>>>>> quite a subjective parameter. I have seen people use it
> >>>>>>>>> differently over the past several years. So, it is a
> >>>>>>>>> bit difficult to infer much accuracy from that line.
> >>>>>>>>>
> >>>>>>>>> OTOH, if someone feels like Opus needs to specify some
> >>>>>>>>> sort of bitrate variability in the SDP, it can do so in
> >>>>>>>>> an optional (opus) parameter. Obviously rules around
> >>>>>>>>> how to set that value and what to do with that value
> >>>>>>>>> need to be clearly specified.
> >>>>>>>>>
> >>>>>>>>> Fredrik (or anyone else in the WG), do you see a strong
> >>>>>>>>> need for this? If so, now is the time to mention that.
> >>>>>>>>>
> >>>>>>>>> -acbegen
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>> On Dec 12, 2014, at 3:07 AM, Jean-Marc Valin
> >>>>>>>>>> <jmvalin@mozilla.com <mailto:jmvalin@mozilla.com>>
> >>>>>>>>>> wrote:
> >>>>>>> I have to say this is not something I understand really
> >>>>>>> well. Does anyone else here have an opinion on how to
> >>>>>>> specify bit-rate?
> >>>>>>
> >>>>>>> Jean-Marc
> >>>>>>
> >>>>>>>>>>>> On 10/12/14 05:11 AM, Fredrik Bergholtz wrote:
> >>>>>>>>>>>> Hi.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> I read in the latest draft that you have removed
> >>>>>>>>>>>> the maxaveragebitrate parameter from the SDP
> >>>>>>>>>>>> signalling for Opus. I think that this parameter
> >>>>>>>>>>>> is useful and that it provides added value over
> >>>>>>>>>>>> the bandwidth modifiers added by RFC 3556.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> I must admit that I haven=C2=B9t read all of RFC 3556
> >>>>>>>>>>>> but as I understand it, a b=3D line are not in any
> >>>>>>>>>>>> way related to an actual format of a media
> >>>>>>>>>>>> description. This means that with this mechanism,
> >>>>>>>>>>>> only one bitrate specification can be sent with
> >>>>>>>>>>>> each media description. Different coding
> >>>>>>>>>>>> algorithms have different constraints on the
> >>>>>>>>>>>> allowed bitrate. For example, I might want to
> >>>>>>>>>>>> send an SDP offer with both Enhanced apt-X at
> >>>>>>>>>>>> 576 kbit/s and Opus at 510kbit/s with cbr. Now,
> >>>>>>>>>>>> specifying 576kbit/s via a b=3D line works for
> >>>>>>>>>>>> apt-X but makes no sense for Opus and vice versa.
> >>>>>>>>>>>> I think that the a=3Dfmtp line is the correct place
> >>>>>>>>>>>> to specify the desired bitrate.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> For the broadcasting use-case, it is often more
> >>>>>>>>>>>> important to have a constant and predictable
> >>>>>>>>>>>> audio quality and network usage than mere
> >>>>>>>>>>>> audibility at any cost.
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> Best regards,
> >>>>>>>>>>>>
> >>>>>>>>>>>> Fredrik Bergholtz
> >>>>>>>>>>>>
> >>>>>>>>>>>> Swedish Radio
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> _______________________________________________
> >>>>>>>>>>>> payload mailing list payload@ietf.org
> >>>>>>>>>>>> <mailto:payload@ietf.org>
> >>>>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>>>>>
> >>>>>>>>>> _______________________________________________
> >>>>>>>>>> payload mailing list payload@ietf.org
> >>>>>>>>>> <mailto:payload@ietf.org>
> >>>>>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>
> >>>>>>
> >>>>>>> _______________________________________________ payload
> >>>>>>> mailing list payload@ietf.org
> >>>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________ payload
> >>>>>> mailing list payload@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>>>>
> >>>>>> _______________________________________________ payload
> >>>>>> mailing list payload@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/payload
> >>>
> >>> _______________________________________________ payload mailing
> >>> list payload@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/payload
> >>>
> >> -----BEGIN PGP SIGNATURE-----
> >> Version: GnuPG v1
> >>
> >> iQEcBAEBAgAGBQJUq5D3AAoJEJ6/8sItn9q9+XkH/27U71MfStYlp2WXoaiW9SmP
> >> YSpGdAbDipfnJ+7RZQ0bDf0GCRtxGxyVcAqsj9oYTFQAyMd67eRp88fzY0pRODVf
> >> VjgWm4VLzP7jbfemzxaZXMzO/MqoiMssIvmnDB2FCirZl03aZ0vcskRAqi/OXlCm
> >> hVJUWkf8MqxMJ1TKqbLn+WGlWHRk89wsyxYQkbfKTfQHVPmnno8eXGvDt71Gewi1
> >> NYyycLisXOAAtvifE+OSGvE0PPw9GcxUgysV/fTRyzEsZV9KVzVQTa4Jj1x6ff8W
> >> 74ylYwYko5t93sRaEsQACVGjGYyXN/FfhpYZJNQ5S6j8gVrPc3Y5ZeFegx2jTEg=3D
> >> =3Dfe42
> >> -----END PGP SIGNATURE-----
> >>
> >> _______________________________________________
> >> payload mailing list
> >> payload@ietf.org
> >> https://www.ietf.org/mailman/listinfo/payload
> >
> > _______________________________________________
> > payload mailing list
> > payload@ietf.org
> > https://www.ietf.org/mailman/listinfo/payload

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

   
------=_Part_239794_959114072.1420963970805
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ve=
rdana, helvetica, sans-serif;font-size:16px"><div id=3D"yiv6010834525"><div=
 class=3D"qtdSeparateBR"><br><br></div><div class=3D"yiv6010834525yqt552529=
1268" id=3D"yiv6010834525yqtfd03162"><div id=3D"yui_3_16_0_1_1420963580365_=
17603"><div style=3D"color:#000;background-color:#fff;font-family:verdana, =
helvetica, sans-serif;font-size:16px;" id=3D"yui_3_16_0_1_1420963580365_176=
02"><div id=3D"yiv6010834525yui_3_16_0_1_1420963580365_11587"><span></span>=
</div><div></div><div id=3D"yiv6010834525yui_3_16_0_1_1420963580365_11585">=
Hi,<br></div><div class=3D"yiv6010834525qtdSeparateBR" id=3D"yui_3_16_0_1_1=
420963580365_19573"><div dir=3D"ltr" id=3D"yiv9593196870yui_3_16_0_1_142047=
8928816_6701" class=3D"" style=3D""><span class=3D"" id=3D"yui_3_16_0_1_142=
0963580365_17410" style=3D"font-family: 'Helvetica Neue', 'Segoe UI', Helve=
tica, Arial, 'Lucida Grande', sans-serif; font-size: 13px;">Check RFC5577 a=
nd RFC4749 for bitrate parameter (for voice)</span></div><div dir=3D"ltr" i=
d=3D"yiv9593196870yui_3_16_0_1_1420478928816_6701" class=3D"" style=3D"">&n=
bsp;<br clear=3D"none" class=3D"" style=3D""></div><div class=3D"" id=3D"yi=
v9593196870yui_3_16_0_1_1420478928816_6734" style=3D""><div dir=3D"ltr" id=
=3D"yiv9593196870yui_3_16_0_1_1420478928816_6733" class=3D"" style=3D""><sp=
an id=3D"yiv9593196870yui_3_16_0_1_1420478928816_6735" style=3D"color: rgb(=
255, 0, 127);" class=3D"">Regards,<br clear=3D"none" class=3D"" style=3D"">=
Ikram Ud Din<br clear=3D"none" class=3D"" style=3D"">InterNetworks Research=
 Laboratory,</span></div><div dir=3D"ltr" id=3D"yiv9593196870yui_3_16_0_1_1=
420478928816_6737" class=3D"" style=3D""><span id=3D"yiv9593196870yui_3_16_=
0_1_1420478928816_6736" style=3D"color: rgb(255, 0, 127);" class=3D"">Unive=
rsiti Utara Malaysia.</span></div><div class=3D"" style=3D"" id=3D"yui_3_16=
_0_1_1420963580365_21563"><span style=3D"color: rgb(255, 0, 127);" class=3D=
""><br class=3D"" style=3D""></span></div></div></div><div class=3D"yiv6010=
834525yahoo_quoted" id=3D"yiv6010834525yui_3_16_0_1_1420963580365_11601" st=
yle=3D"display: block;"><div id=3D"yiv6010834525yui_3_16_0_1_1420963580365_=
11600" style=3D"font-family:verdana, helvetica, sans-serif;font-size:16px;"=
><div id=3D"yiv6010834525yui_3_16_0_1_1420963580365_11599" style=3D"font-fa=
mily:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-s=
erif;font-size:16px;"><div class=3D"yiv6010834525y_msg_container" id=3D"yiv=
6010834525yui_3_16_0_1_1420963580365_11598"><br clear=3D"none">I'm fine in =
case it is considered as implementers issue.<br clear=3D"none"><br clear=3D=
"none">These bitrate related parameters are optional parameters for the giv=
en codec<br clear=3D"none">and each implementor has its his own choice to i=
mplement or ignore.<br clear=3D"none"><br clear=3D"none">Thanks<br clear=3D=
"none">Partha<br clear=3D"none"><div class=3D"yiv6010834525yqt3773208603" i=
d=3D"yiv6010834525yqtfd95839"><br clear=3D"none">&gt; -----Original Message=
-----<br clear=3D"none">&gt; From: Ali C. Begen (abegen) [mailto:<a rel=3D"=
nofollow" shape=3D"rect" ymailto=3D"mailto:abegen@cisco.com" target=3D"_bla=
nk" href=3D"mailto:abegen@cisco.com">abegen@cisco.com</a>]<br clear=3D"none=
">&gt; Sent: Wednesday, January 07, 2015 6:25 PM<br clear=3D"none">&gt; To:=
 Parthasarathi R<br clear=3D"none">&gt; Cc: Jean-Marc Valin; Fredrik Bergho=
ltz; <a rel=3D"nofollow" shape=3D"rect" id=3D"yiv6010834525yui_3_16_0_1_142=
0963580365_11607" ymailto=3D"mailto:payload@ietf.org" target=3D"_blank" hre=
f=3D"mailto:payload@ietf.org">payload@ietf.org</a><br clear=3D"none">&gt; S=
ubject: Re: [payload] draft-ietf-payload-rtp-opus-05:<br clear=3D"none">&gt=
; maxaveragebitrate<br clear=3D"none">&gt; <br clear=3D"none">&gt; I think =
it would have been nice if b=3D attribute or something similar<br clear=3D"=
none">&gt; was used for all codecs in the same fashion, but that did not re=
ally<br clear=3D"none">&gt; work. I agree that a common payload parameter w=
ould seem ideal,<br clear=3D"none">&gt; however, I dont think it would have=
 been a lot easier to implement it.<br clear=3D"none">&gt; Most codecs may =
have exceptions here and there and dealing with such<br clear=3D"none">&gt;=
 complexities would likely be tough. So if someone wants to implement<br cl=
ear=3D"none">&gt; codecs a and b, he needs to understand the payload parame=
ters for both.<br clear=3D"none">&gt; Eliminating only one such parameter w=
ould not make much difference IMO.<br clear=3D"none">&gt; <br clear=3D"none=
">&gt; <br clear=3D"none">&gt; &gt; On Jan 6, 2015, at 7:48 PM, Parthasarat=
hi R<br clear=3D"none">&gt; &lt;<a rel=3D"nofollow" shape=3D"rect" ymailto=
=3D"mailto:partha@parthasarathi.co.in" target=3D"_blank" href=3D"mailto:par=
tha@parthasarathi.co.in">partha@parthasarathi.co.in</a>&gt; wrote:<br clear=
=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt; Hi Jean-Marc/Fredrik/all,<b=
r clear=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt; I understand that th=
ere is a need for the parameter in the codec<br clear=3D"none">&gt; level a=
nd<br clear=3D"none">&gt; &gt; it is followed as fmtp parameter specific to=
 the given codec<br clear=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt; 1)=
 max-br (RFC 6185)&nbsp; - H.264<br clear=3D"none">&gt; &gt; 2) maxbitrate =
(RFC 4749) - G.729.1<br clear=3D"none">&gt; &gt; 3) bitrate (RFC 5577) - G.=
722.1<br clear=3D"none">&gt; &gt; 3) maxaveragebitrate - Opus?<br clear=3D"=
none">&gt; &gt;<br clear=3D"none">&gt; &gt; As per the below mail, this par=
ameter is an open item for VP8 &amp; VP9<br clear=3D"none">&gt; and<br clea=
r=3D"none">&gt; &gt; those codecs will take their own naming convention for=
 their maximum<br clear=3D"none">&gt; bit<br clear=3D"none">&gt; &gt; rate.=
<br clear=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt; My concern is that=
 it is better to have some standard naming<br clear=3D"none">&gt; conventio=
n for<br clear=3D"none">&gt; &gt; maximum bit rate for the given codec as<b=
r clear=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt; 1) It is simple to i=
mplement in the generic fashion in case of<br clear=3D"none">&gt; multiple<=
br clear=3D"none">&gt; &gt; codec handling in the given SDP. This is the co=
mmon implementation.<br clear=3D"none">&gt; &gt; 2) Offer/answer behavior w=
ill be common across multiple codec.<br clear=3D"none">&gt; &gt; 3) Interac=
tion with bandwidth ("b") parameter in the SDP level and<br clear=3D"none">=
&gt; these<br clear=3D"none">&gt; &gt; codec specific max bit rate paramete=
r will be clear in case both<br clear=3D"none">&gt; exists in<br clear=3D"n=
one">&gt; &gt; the same SDP media line.<br clear=3D"none">&gt; &gt;<br clea=
r=3D"none">&gt; &gt; Please let me know your opinion on the same.<br clear=
=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt; Thanks<br clear=3D"none">&g=
t; &gt; Partha<br clear=3D"none">&gt; &gt;<br clear=3D"none">&gt; &gt;<br c=
lear=3D"none">&gt; &gt;&gt; -----Original Message-----<br clear=3D"none">&g=
t; &gt;&gt; From: payload [mailto:<a rel=3D"nofollow" shape=3D"rect" ymailt=
o=3D"mailto:payload-bounces@ietf.org" target=3D"_blank" href=3D"mailto:payl=
oad-bounces@ietf.org">payload-bounces@ietf.org</a>] On Behalf Of Jean-<br c=
lear=3D"none">&gt; Marc<br clear=3D"none">&gt; &gt;&gt; Valin<br clear=3D"n=
one">&gt; &gt;&gt; Sent: Tuesday, January 06, 2015 1:09 PM<br clear=3D"none=
">&gt; &gt;&gt; To: Fredrik Bergholtz<br clear=3D"none">&gt; &gt;&gt; Cc: <=
a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:payload@ietf.org" targe=
t=3D"_blank" href=3D"mailto:payload@ietf.org">payload@ietf.org</a><br clear=
=3D"none">&gt; &gt;&gt; Subject: Re: [payload] draft-ietf-payload-rtp-opus-=
05:<br clear=3D"none">&gt; &gt;&gt; maxaveragebitrate<br clear=3D"none">&gt=
; &gt;&gt;<br clear=3D"none">&gt; &gt;&gt; -----BEGIN PGP SIGNED MESSAGE---=
--<br clear=3D"none">&gt; &gt;&gt; Hash: SHA1<br clear=3D"none">&gt; &gt;&g=
t;<br clear=3D"none">&gt; &gt;&gt; Just posted a new version that brings ba=
ck maxaveragebitrate exactly<br clear=3D"none">&gt; &gt;&gt; as it was befo=
re.<br clear=3D"none">&gt; &gt;&gt;<br clear=3D"none">&gt; &gt;&gt; Cheers,=
<br clear=3D"none">&gt; &gt;&gt;<br clear=3D"none">&gt; &gt;&gt; &nbsp;&nbs=
p;&nbsp; Jean-Marc<br clear=3D"none">&gt; &gt;&gt;<br clear=3D"none">&gt; &=
gt;&gt; On 05/01/15 05:32 AM, Fredrik Bergholtz wrote:<br clear=3D"none">&g=
t; &gt;&gt;&gt; I think that re-instating maxaveragebitrate would do it for=
 me.<br clear=3D"none">&gt; &gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt=
; /Fredrik<br clear=3D"none">&gt; &gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&=
gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt; On 5 jan 2015, at 00:16, "J=
ean-Marc Valin" &lt;<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:jm=
valin@jmvalin.ca" target=3D"_blank" href=3D"mailto:jmvalin@jmvalin.ca">jmva=
lin@jmvalin.ca</a>&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt; wrote:<br cl=
ear=3D"none">&gt; &gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt; So w=
ould everyone be happy if I just revert to using<br clear=3D"none">&gt; &gt=
;&gt;&gt; maxaveragebitrate as in draft 04?<br clear=3D"none">&gt; &gt;&gt;=
&gt;<br clear=3D"none">&gt; &gt;&gt;&gt; Jean-Marc<br clear=3D"none">&gt; &=
gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;<br clear=3D"none">&gt; &gt;=
&gt;&gt;&gt;&gt;&gt; On 04/01/15 05:05 PM, Mo Zanaty (mzanaty) wrote: Some =
other<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; codecs have fmtp para=
meters for maximum bitrate explicitly<br clear=3D"none">&gt; &gt;&gt;&gt;&g=
t;&gt;&gt; (e.g. max-br in H264 / H264-SVC / H265) and/or implicitly<br cle=
ar=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; (e.g. profile-level-id in H264 / =
H264-SVC / H265 / AAC;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; conf=
ig in AAC).<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"non=
e">&gt; &gt;&gt;&gt;&gt;&gt;&gt; Other notable codecs that lack this includ=
e VP8 and VP9.<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"=
none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; This is a general problem which may bec=
ome more important<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; with com=
plex applications that include simulcast and scalable<br clear=3D"none">&gt=
; &gt;&gt;&gt;&gt;&gt;&gt; media within the same media description (m=3D li=
ne). The SDP b=3D<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; parameter=
 is clearly insufficient for expressing multiple<br clear=3D"none">&gt; &gt=
;&gt;&gt;&gt;&gt;&gt; different maximum bitrates for each payload type, unl=
ess we<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; define a new bandwid=
th modifier which includes explicit PT<br clear=3D"none">&gt; &gt;&gt;&gt;&=
gt;&gt;&gt; bindings. Payload formats that include fmtp parameters for<br c=
lear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; maximum bitrate can avoid this =
problem, so I think it is<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; a=
dvisable to add this to Opus, VP8 and VP9.<br clear=3D"none">&gt; &gt;&gt;&=
gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; Mo<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;=
&gt;&gt;&gt; On 1/4/15, 2:45 PM, Jean-Marc Valin &lt;<a rel=3D"nofollow" sh=
ape=3D"rect" ymailto=3D"mailto:jmvalin@jmvalin.ca" target=3D"_blank" href=
=3D"mailto:jmvalin@jmvalin.ca">jmvalin@jmvalin.ca</a>&gt;<br clear=3D"none"=
>&gt; &gt;&gt;&gt;&gt;&gt;&gt; wrote:<br clear=3D"none">&gt; &gt;&gt;&gt;&g=
t;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; Setting the bitr=
ate isn't exactly a new problem here. Opus<br clear=3D"none">&gt; &gt;&gt;&=
gt;&gt;&gt;&gt; isn't the first codec to support more than one rate so how<=
br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; are the other codecs (audio=
 and video) handling bitrate<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt=
; signaling?<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt; Cheers,<br clear=3D"none">&gt; &gt;&gt;&g=
t;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; Jean-Marc<br=
 clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&g=
t;&gt;&gt;&gt;&gt;&gt; On 03/01/15 09:32 AM, Fredrik Bergholtz wrote: Hi.<b=
r clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&=
gt;&gt;&gt;&gt;&gt;&gt; One thought om the general media level bit rate<br =
clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; specification is that it c=
an never be anything other than<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;=
&gt;&gt; an approximate value. Different codecs define different<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; constraints on the bit rate. I =
think that the easiest way<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&=
gt; to specify exact bit rates would be via the fmtp<br clear=3D"none">&gt;=
 &gt;&gt;&gt;&gt;&gt;&gt;&gt; attribute.<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; This can =
be compared with the ptime attribute - as it is<br clear=3D"none">&gt; &gt;=
&gt;&gt;&gt;&gt;&gt;&gt; defined today, only approximate packet times can b=
e<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; signalled in SDP. For=
 the broadcast use case this is not<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;=
&gt;&gt;&gt; acceptable and for that reason, the European Broadcasting<br c=
lear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; Union (EBU) has defined an =
SDP extension that allows<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&g=
t; signalling of a ptime value for each codec. There is no RFC<br clear=3D"=
none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; draft for this extension and it is =
not yet registered with<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;=
 IANA or IETF but it is starting to be used by manufacturers<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; of broadcasting equipment. The extens=
ion is known as "ACIP<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; P=
rofiles" and can be found at the EBU web site. But I<br clear=3D"none">&gt;=
 &gt;&gt;&gt;&gt;&gt;&gt;&gt; digress.<br clear=3D"none">&gt; &gt;&gt;&gt;&=
gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; I don't thi=
nk that forcing encoders to approximate the<br clear=3D"none">&gt; &gt;&gt;=
&gt;&gt;&gt;&gt;&gt; nearest allowed value for the selected codec is less<b=
r clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; complicated than interpr=
eting an exact value. In addition,<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&=
gt;&gt;&gt; it would add a non-deterministic aspect, thus the resulting<br =
clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; bit rate may not be what t=
he user expected.<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; Best regards Fredrik Bergholtz =
Swedish Radio<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"n=
one">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; On 3 jan 2015, at 11:27, "Parthasara=
thi R"<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; &lt;<a rel=3D"no=
follow" shape=3D"rect" ymailto=3D"mailto:partha@parthasarathi.co.in" target=
=3D"_blank" href=3D"mailto:partha@parthasarathi.co.in">partha@parthasarathi=
.co.in</a><br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; &lt;mailto:<=
a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:partha@parthasarathi.co=
.in" target=3D"_blank" href=3D"mailto:partha@parthasarathi.co.in">partha@pa=
rthasarathi.co.in</a>&gt;&gt; wrote:<br clear=3D"none">&gt; &gt;&gt;&gt;&gt=
;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Hi all,<b=
r clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&g=
t; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;=
&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I&=
nbsp; understand that bandwidth =C2=B3b=C2=B2 parameter has lot of<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; variants like AS, TIAS and =
usage but all those parameters<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&=
gt;&gt;&gt; are media level and not codec specific. The bandwidth<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; allocation is going to be h=
appen in the media level in<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;=
&gt;&gt; the network or end point based on the media level and not<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; on the codec specific&nbsp;=
 in case the multiple codec is<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&=
gt;&gt;&gt; negotiated. In most of the implementation, the multiple<br clea=
r=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; codec (Say Ex: G.711 &amp;=
 Opus) is going to be negotiated<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt=
;&gt;&gt;&gt; rather than single codec. The codec specific bandwidth<br cle=
ar=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; implementation complicate=
s these bandwidth calculation<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&g=
t;&gt;&gt; still further which is applicable to maxaveragebitrate<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; parameter in opus.<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;=
&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt=
;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I could n=
ot understand why the generic parameter should<br clear=3D"none">&gt; &gt;&=
gt;&gt;&gt;&gt;&gt;&gt;&gt; not used or generic parameter has to be enhance=
d which<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; meets Opus =
requirement rather than opus specific<br clear=3D"none">&gt; &gt;&gt;&gt;&g=
t;&gt;&gt;&gt;&gt; parameter.<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&g=
t;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;=
&gt;&gt;&gt;&gt;&gt;&gt;&gt; Thanks<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;=
&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Pa=
rtha<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&g=
t;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&=
gt; *From:*payload [mailto:<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"ma=
ilto:payload-bounces@ietf.org" target=3D"_blank" href=3D"mailto:payload-bou=
nces@ietf.org">payload-bounces@ietf.org</a>] *On<br clear=3D"none">&gt; &gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Behalf Of *Ali C. Begen (abegen) *Sent:* Wedn=
esday,<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; December 31,=
 2014 4:57 PM *To:* Fredrik Bergholtz *Cc:*<br clear=3D"none">&gt; &gt;&gt;=
&gt;&gt;&gt;&gt;&gt;&gt; <a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mail=
to:payload@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org">pay=
load@ietf.org</a> &lt;mailto:<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"=
mailto:payload@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org"=
>payload@ietf.org</a>&gt; *Subject:* Re:<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt; [payload] draft-ietf-payload-rtp-opus-05:<br clear=3D=
"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; maxaveragebitrate<br clear=3D"=
none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;=
&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt=
;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Authors<br cl=
ear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &=
gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Please chime in and lets discuss s quick so=
lution for<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; this on =
the list. Once agreed we can ship this draft.<br clear=3D"none">&gt; &gt;&g=
t;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&=
gt;&gt; -acbegen<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br=
 clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt=
; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&=
gt;&gt;&gt;&gt; *From:*Fredrik Bergholtz<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt; &lt;<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mai=
lto:fredrik.bergholtz@sverigesradio.se" target=3D"_blank" href=3D"mailto:fr=
edrik.bergholtz@sverigesradio.se">fredrik.bergholtz@sverigesradio.se</a><br=
 clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; &lt;mailto:<a rel=3D"=
nofollow" shape=3D"rect" ymailto=3D"mailto:fredrik.bergholtz@sverigesradio.=
se" target=3D"_blank" href=3D"mailto:fredrik.bergholtz@sverigesradio.se">fr=
edrik.bergholtz@sverigesradio.se</a>&gt;&gt; *Sent:* Dec<br clear=3D"none">=
&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; 31, 2014 11:42 AM *To:* Ali C. Begen =
(abegen) *Cc:*<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Jean=
-Marc Valin &lt;<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:jmvali=
n@mozilla.com" target=3D"_blank" href=3D"mailto:jmvalin@mozilla.com">jmvali=
n@mozilla.com</a><br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; &=
lt;mailto:<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:jmvalin@mozi=
lla.com" target=3D"_blank" href=3D"mailto:jmvalin@mozilla.com">jmvalin@mozi=
lla.com</a>&gt;&gt;;Fredrik<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;=
&gt;&gt; Bergholtz;<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:pay=
load@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org">payload@i=
etf.org</a> &lt;mailto:<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto=
:payload@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org">paylo=
ad@ietf.org</a>&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;=
 *Subject:* Re: [payload] draft-ietf-payload-rtp-opus-05:<br clear=3D"none"=
>&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; maxaveragebitrate<br clear=3D"none">=
&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&g=
t;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<=
br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Yes, I do see a str=
ong need for specifying the bitrate.<br clear=3D"none">&gt; &gt;&gt;&gt;&gt=
;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I=
n the broadcasting use case we have to make sure that we<br clear=3D"none">=
&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; get the best audio quality that is po=
ssible for the<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; netw=
ork that is currently used but never try to send more<br clear=3D"none">&gt=
; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; data than the particular link is capable=
 of. For this<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; reaso=
n, specifying a maximum bit rate is required.<br clear=3D"none">&gt; &gt;&g=
t;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&=
gt;&gt; In addition we often have the requirement that the<br clear=3D"none=
">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; quality is constant rather than the=
 best possible. To<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; =
avoid as many audible changes in the audio as possible<br clear=3D"none">&g=
t; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; over time we will often use constant bi=
t rate and then it<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; =
becomes even more important to be able to specify what<br clear=3D"none">&g=
t; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; bit rate should be.<br clear=3D"none">&=
gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt=
;&gt;&gt;&gt;&gt; We will use the Opus codec over various network links<br =
clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; with different capabil=
ities in terms of bit rate, delay<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&g=
t;&gt;&gt;&gt; variation, absolute delay, etc and we will typically use<br =
clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; the SDP to signal the =
parameters that best fit both the<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&g=
t;&gt;&gt;&gt; current network and the current type of radio<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; production.<br clear=3D"none">&gt=
; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&=
gt;&gt;&gt;&gt; Best regards Fredrik Bergholtz Swedish Radio<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&g=
t;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&=
gt;&gt; On 31 dec 2014, at 05:57, "Ali C. Begen (abegen)"<br clear=3D"none"=
>&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; &lt;<a rel=3D"nofollow" shape=3D=
"rect" ymailto=3D"mailto:abegen@cisco.com" target=3D"_blank" href=3D"mailto=
:abegen@cisco.com">abegen@cisco.com</a> &lt;mailto:<a rel=3D"nofollow" shap=
e=3D"rect" ymailto=3D"mailto:abegen@cisco.com" target=3D"_blank" href=3D"ma=
ilto:abegen@cisco.com">abegen@cisco.com</a>&gt;&gt; wrote:<br clear=3D"none=
">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;=
&gt;&gt;&gt;&gt;&gt;&gt;&gt; My understanding is that the b=3D line in the =
SDP is<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; quite a =
subjective parameter. I have seen people use it<br clear=3D"none">&gt; &gt;=
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; differently over the past several years. S=
o, it is a<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; bit =
difficult to infer much accuracy from that line.<br clear=3D"none">&gt; &gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&g=
t;&gt;&gt;&gt;&gt; OTOH, if someone feels like Opus needs to specify some<b=
r clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; sort of bitrate =
variability in the SDP, it can do so in<br clear=3D"none">&gt; &gt;&gt;&gt;=
&gt;&gt;&gt;&gt;&gt;&gt; an optional (opus) parameter. Obviously rules arou=
nd<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; how to set t=
hat value and what to do with that value<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt;&gt; need to be clearly specified.<br clear=3D"none">&=
gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt;&gt; Fredrik (or anyone else in the WG), do you see a =
strong<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; need for=
 this? If so, now is the time to mention that.<br clear=3D"none">&gt; &gt;&=
gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;=
&gt;&gt;&gt;&gt; -acbegen<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&g=
t;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br c=
lear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; On Dec 12, 2014=
, at 3:07 AM, Jean-Marc Valin<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&g=
t;&gt;&gt;&gt;&gt; &lt;<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto=
:jmvalin@mozilla.com" target=3D"_blank" href=3D"mailto:jmvalin@mozilla.com"=
>jmvalin@mozilla.com</a> &lt;mailto:<a rel=3D"nofollow" shape=3D"rect" ymai=
lto=3D"mailto:jmvalin@mozilla.com" target=3D"_blank" href=3D"mailto:jmvalin=
@mozilla.com">jmvalin@mozilla.com</a>&gt;&gt;<br clear=3D"none">&gt; &gt;&g=
t;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; wrote:<br clear=3D"none">&gt; &gt;&gt;&g=
t;&gt;&gt;&gt;&gt; I have to say this is not something I understand really<=
br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; well. Does anyone else =
here have an opinion on how to<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&=
gt;&gt; specify bit-rate?<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<b=
r clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; Jean-Marc<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&g=
t;&gt;&gt;&gt;&gt;&gt;&gt;&gt; On 10/12/14 05:11 AM, Fredrik Bergholtz wrot=
e:<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; =
Hi.<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;=
<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br=
 clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br cl=
ear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I read i=
n the latest draft that you have removed<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; the maxaveragebitrate parameter from =
the SDP<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;=
&gt; signalling for Opus. I think that this parameter<br clear=3D"none">&gt=
; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; is useful and that it pr=
ovides added value over<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;=
&gt;&gt;&gt;&gt;&gt; the bandwidth modifiers added by RFC 3556.<br clear=3D=
"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none"=
>&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&g=
t; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I must admit that I hav=
en=C2=B9t read all of RFC 3556<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&=
gt;&gt;&gt;&gt;&gt;&gt;&gt; but as I understand it, a b=3D line are not in =
any<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;=
 way related to an actual format of a media<br clear=3D"none">&gt; &gt;&gt;=
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; description. This means that with =
this mechanism,<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;=
&gt;&gt;&gt; only one bitrate specification can be sent with<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; each media descri=
ption. Different coding<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;=
&gt;&gt;&gt;&gt;&gt; algorithms have different constraints on the<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; allowed bit=
rate. For example, I might want to<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&=
gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; send an SDP offer with both Enhanced apt-X =
at<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; =
576 kbit/s and Opus at 510kbit/s with cbr. Now,<br clear=3D"none">&gt; &gt;=
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; specifying 576kbit/s via a b=
=3D line works for<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&=
gt;&gt;&gt;&gt; apt-X but makes no sense for Opus and vice versa.<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; I think tha=
t the a=3Dfmtp line is the correct place<br clear=3D"none">&gt; &gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; to specify the desired bitrate.<br cl=
ear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D=
"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"no=
ne">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; For the broadcast=
ing use-case, it is often more<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&=
gt;&gt;&gt;&gt;&gt;&gt;&gt; important to have a constant and predictable<br=
 clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; audio=
 quality and network usage than mere<br clear=3D"none">&gt; &gt;&gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; audibility at any cost.<br clear=3D"none"=
>&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&g=
t; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; =
&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Best regards,<br clear=3D"non=
e">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">=
&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Fredrik Bergholtz<br =
clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br cle=
ar=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; Swedish R=
adio<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt=
;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<b=
r clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br c=
lear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clea=
r=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=
=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; ___________=
____________________________________<br clear=3D"none">&gt; &gt;&gt;&gt;&gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; payload mailing list <a rel=3D"nofollow" =
shape=3D"rect" ymailto=3D"mailto:payload@ietf.org" target=3D"_blank" href=
=3D"mailto:payload@ietf.org">payload@ietf.org</a><br clear=3D"none">&gt; &g=
t;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; &lt;mailto:<a rel=3D"nofollo=
w" shape=3D"rect" ymailto=3D"mailto:payload@ietf.org" target=3D"_blank" hre=
f=3D"mailto:payload@ietf.org">payload@ietf.org</a>&gt;<br clear=3D"none">&g=
t; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; <a rel=3D"nofollow" sha=
pe=3D"rect" target=3D"_blank" href=3D"https://www.ietf.org/mailman/listinfo=
/payload">https://www.ietf.org/mailman/listinfo/payload</a><br clear=3D"non=
e">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt=
;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; _____________________________________=
__________<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; =
payload mailing list <a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:p=
ayload@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org">payload=
@ietf.org</a><br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&g=
t; &lt;mailto:<a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:payload@=
ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org">payload@ietf.o=
rg</a>&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt;&gt; =
<a rel=3D"nofollow" shape=3D"rect" target=3D"_blank" href=3D"https://www.ie=
tf.org/mailman/listinfo/payload">https://www.ietf.org/mailman/listinfo/payl=
oad</a><br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&=
gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt=
;&gt; _______________________________________________ payload<br clear=3D"n=
one">&gt; &gt;&gt;&gt;&gt;&gt;&gt;&gt; mailing list <a rel=3D"nofollow" sha=
pe=3D"rect" ymailto=3D"mailto:payload@ietf.org" target=3D"_blank" href=3D"m=
ailto:payload@ietf.org">payload@ietf.org</a><br clear=3D"none">&gt; &gt;&gt=
;&gt;&gt;&gt;&gt;&gt; <a rel=3D"nofollow" shape=3D"rect" target=3D"_blank" =
href=3D"https://www.ietf.org/mailman/listinfo/payload">https://www.ietf.org=
/mailman/listinfo/payload</a><br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&g=
t;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt;<br clear=3D"none">&gt; &=
gt;&gt;&gt;&gt;&gt;&gt; _______________________________________________ pay=
load<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; mailing list <a rel=3D=
"nofollow" shape=3D"rect" ymailto=3D"mailto:payload@ietf.org" target=3D"_bl=
ank" href=3D"mailto:payload@ietf.org">payload@ietf.org</a><br clear=3D"none=
">&gt; &gt;&gt;&gt;&gt;&gt;&gt; <a rel=3D"nofollow" shape=3D"rect" target=
=3D"_blank" href=3D"https://www.ietf.org/mailman/listinfo/payload">https://=
www.ietf.org/mailman/listinfo/payload</a><br clear=3D"none">&gt; &gt;&gt;&g=
t;&gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; ____________=
___________________________________ payload<br clear=3D"none">&gt; &gt;&gt;=
&gt;&gt;&gt;&gt; mailing list <a rel=3D"nofollow" shape=3D"rect" ymailto=3D=
"mailto:payload@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.org=
">payload@ietf.org</a><br clear=3D"none">&gt; &gt;&gt;&gt;&gt;&gt;&gt; <a r=
el=3D"nofollow" shape=3D"rect" target=3D"_blank" href=3D"https://www.ietf.o=
rg/mailman/listinfo/payload">https://www.ietf.org/mailman/listinfo/payload<=
/a><br clear=3D"none">&gt; &gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt;&gt;=
 _______________________________________________ payload mailing<br clear=
=3D"none">&gt; &gt;&gt;&gt; list <a rel=3D"nofollow" shape=3D"rect" ymailto=
=3D"mailto:payload@ietf.org" target=3D"_blank" href=3D"mailto:payload@ietf.=
org">payload@ietf.org</a><br clear=3D"none">&gt; &gt;&gt;&gt; <a rel=3D"nof=
ollow" shape=3D"rect" target=3D"_blank" href=3D"https://www.ietf.org/mailma=
n/listinfo/payload">https://www.ietf.org/mailman/listinfo/payload</a><br cl=
ear=3D"none">&gt; &gt;&gt;&gt;<br clear=3D"none">&gt; &gt;&gt; -----BEGIN P=
GP SIGNATURE-----<br clear=3D"none">&gt; &gt;&gt; Version: GnuPG v1<br clea=
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=3D"none">&gt; &gt;&gt; _______________________________________________<br =
clear=3D"none">&gt; &gt;&gt; payload mailing list<br clear=3D"none">&gt; &g=
t;&gt; <a rel=3D"nofollow" shape=3D"rect" ymailto=3D"mailto:payload@ietf.or=
g" target=3D"_blank" href=3D"mailto:payload@ietf.org">payload@ietf.org</a><=
br clear=3D"none">&gt; &gt;&gt; <a rel=3D"nofollow" shape=3D"rect" target=
=3D"_blank" href=3D"https://www.ietf.org/mailman/listinfo/payload">https://=
www.ietf.org/mailman/listinfo/payload</a><br clear=3D"none">&gt; &gt;<br cl=
ear=3D"none">&gt; &gt; _______________________________________________<br c=
lear=3D"none">&gt; &gt; payload mailing list<br clear=3D"none">&gt; &gt; <a=
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ef=3D"https://www.ietf.org/mailman/listinfo/payload">https://www.ietf.org/m=
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_____________________________________<br clear=3D"none">payload mailing lis=
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ietf.org</a><br clear=3D"none"><a rel=3D"nofollow" shape=3D"rect" target=3D=
"_blank" href=3D"https://www.ietf.org/mailman/listinfo/payload">https://www=
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/html>
------=_Part_239794_959114072.1420963970805--


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Jean-Marc Valin <jmvalin@jmvalin.ca>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Date: Mon, 12 Jan 2015 10:42:15 +0000
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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--_000_EFC567A0F1CD43F0ADBFE94BE2626EE3ciscocom_
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Here goes my chair review on version 06:

Section 3.1.2:
1) "network capacity=94 is not the term you are looking for. You need to sa=
y something like =93available bandwidth=94.

Section 4.2:
1) I suppose in table 2, =93x=94 means not supported or N/A, which should b=
e mentioned in the document. What do the empty slots mean?

Section 6.1:
1) For "Published specification:", you should not say "none", rather say "R=
FC [XXXX]" with an editor note the RFC editor to fix the number upon public=
ation.
2) According to the template in rfc 6838, you are missing "Fragment identif=
ier considerations", which I believe simply say "N/A."

3) Change controller should be "IETF Payload Working Group delegated from t=
he IESG.=94

We should be ready to ship this draft once the above is addressed.
-acbegen


On Jan 6, 2015, at 9:38 AM, Jean-Marc Valin <jmvalin@jmvalin.ca<mailto:jmva=
lin@jmvalin.ca>> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Just posted a new version that brings back maxaveragebitrate exactly
as it was before.

Cheers,

Jean-Marc


--_000_EFC567A0F1CD43F0ADBFE94BE2626EE3ciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body dir=3D"auto" class=3D"ApplePlainTextBody" style=3D"word-wrap: break-w=
ord; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
Here goes my chair review on version 06: </div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
Section 3.1.2:</div>
<div><font face=3D"Consolas,sans-serif">1) &quot;<span style=3D"font-size: =
13px; line-height: 1.2em;">network capacity</span><span style=3D"font-size:=
 13px; line-height: 15.600000381469727px;">=94</span><span style=3D"font-si=
ze: 13px; line-height: 1.2em;">&nbsp;is not the term
 you are looking for. You need to say something like&nbsp;</span><span styl=
e=3D"font-size: 13px; line-height: 15.600000381469727px;">=93</span><span s=
tyle=3D"font-size: 13px; line-height: 1.2em;">available bandwidth</span><sp=
an style=3D"font-size: 13px; line-height: 15.600000381469727px;">=94</span>=
<span style=3D"font-size: 13px; line-height: 1.2em;">.</span></font></div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
Section 4.2:</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
1) I suppose in table 2, =93x=94 means not supported or N/A, which should b=
e mentioned in the document. What do the empty slots mean?</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
Section 6.1:</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
1) For &quot;Published specification:&quot;, you should not say &quot;none&=
quot;, rather say &quot;RFC [XXXX]&quot; with an editor note the RFC editor=
 to fix the number upon publication.</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
2) According to the template in rfc 6838, you are missing &quot;Fragment id=
entifier considerations&quot;, which I believe simply say &quot;N/A.&quot;<=
/div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<div><br>
</div>
3) Change controller should be &quot;IETF Payload Working Group delegated f=
rom the IESG.=94</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
We should be ready to ship this draft once the above is addressed.</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
-acbegen</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Consolas, sans-serif; font-=
size: 14px;">
<blockquote type=3D"cite">On Jan 6, 2015, at 9:38 AM, Jean-Marc Valin &lt;<=
a href=3D"mailto:jmvalin@jmvalin.ca">jmvalin@jmvalin.ca</a>&gt; wrote:<br>
<br>
-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
Just posted a new version that brings back maxaveragebitrate exactly<br>
as it was before.<br>
<br>
Cheers,<br>
<br>
<span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>Jean-Marc<b=
r>
</blockquote>
<br>
</div>
</body>
</html>

--_000_EFC567A0F1CD43F0ADBFE94BE2626EE3ciscocom_--


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From: Gunnar Persson N <gunnar.n.persson@ericsson.com>
To: "payload@ietf.org" <payload@ietf.org>
Thread-Topic: Question regarding RFC 4867
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Hi
I have a simple question regarding RFC 4867
In chapter...
4.3.1<https://tools.ietf.org/html/rfc4867#section-4.3.1>.  The Payload Head=
er

...it is stated:



Therefore, if a terminal continuously wishes to receive frames in the

   same mode X, it needs to set CMR=3DX for all its outbound payloads, and

   if a terminal has no preference in which mode to receive, it SHOULD

   set CMR=3D15 in all its outbound payloads.


Question:
Does "all its outbound payloads" mean "every RTP frame" (I e every 20ms)?

Comment:
On air interface CMR value is only communicated every second speech frame (=
I e every 40ms). I assume the idea is that BSS shall remember the last comm=
unicated CMR from mobile terminal and insert this CMR in all RTP frames. Pl=
ease comment if my assumption is correct.

Best Regards,
Gunnar Persson at Ericsson

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<p class=3D"MsoNormal">Hi <o:p></o:p></p>
<p class=3D"MsoNormal">I have a simple question regarding RFC 4867 <o:p></o=
:p></p>
<p class=3D"MsoNormal">In chapter&#8230;<o:p></o:p></p>
<h4><span style=3D"color:#C00000"></span><a name=3D"section-4.3.1"></a><a h=
ref=3D"https://tools.ietf.org/html/rfc4867#section-4.3.1"><span style=3D"fo=
nt-family:&quot;Courier New&quot;;color:#C00000">4.3.1</span></a><span styl=
e=3D"font-family:&quot;Courier New&quot;;color:#C00000">.&nbsp; The
 Payload Header<o:p></o:p></span></h4>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;">&#8230;it is stated:<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"color:#C00000">Therefore, if a terminal continuously wi=
shes to receive frames in the<o:p></o:p></span></pre>
<pre><span style=3D"color:#C00000">&nbsp;&nbsp; same mode X, it needs to se=
t CMR=3DX for all its outbound payloads, and<o:p></o:p></span></pre>
<pre><span style=3D"color:#C00000">&nbsp;&nbsp; if a terminal has no prefer=
ence in which mode to receive, it SHOULD<o:p></o:p></span></pre>
<pre><span style=3D"color:#C00000">&nbsp;&nbsp; set CMR=3D15 in all its out=
bound payloads.<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<p class=3D"MsoNormal">Question: <o:p></o:p></p>
<p class=3D"MsoNormal"><b>Does &#8220;all its outbound payloads&#8221; mean=
 &#8220;every RTP frame&#8221; (I e every 20ms)?
<o:p></o:p></b></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Comment:<o:p></o:p></p>
<p class=3D"MsoNormal">On air interface CMR value is only communicated ever=
y second speech frame (I e every 40ms). I assume the idea is that BSS shall=
 remember the last communicated CMR from mobile terminal and insert this CM=
R in all RTP frames. Please comment
 if my assumption is correct.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Best Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">Gunnar Persson at Ericsson <o:p></o:p></p>
</div>
</body>
</html>

--_000_7F3814818CC0B04E97C2ADBA62B5C9681C5345D8ESESSMB201erics_--


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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Ali,

On 12/01/15 05:42 AM, Ali C. Begen (abegen) wrote:
> Section 3.1.2: 1) "network capacity” is not the term you are 
> looking for. You need to say something like “available bandwidth”.

How about "available network bandwidth" since we use "bandwidth"
mainly for audio frequencies in this draft?

> Section 4.2: 1) I suppose in table 2, “x” means not supported or 
> N/A, which should be mentioned in the document. What do the empty 
> slots mean?

Actually "x" means supported and the ones without an "x" aren't
supported. I guess that wasn't clear. Maybe "o" for supported and "x"
for not supported? Or do you have other suggestions?

> Section 6.1: 1) For "Published specification:", you should not say 
> "none", rather say "RFC [XXXX]" with an editor note the RFC editor 
> to fix the number upon publication.

Not sure what you mean here.

> 2) According to the template in rfc 6838, you are missing "Fragment
> identifier considerations", which I believe simply say "N/A."

Will do.

> 3) Change controller should be "IETF Payload Working Group 
> delegated from the IESG.”

Will do.

	Jean-Marc

> We should be ready to ship this draft once the above is addressed. 
> -acbegen
> 
> 
>> On Jan 6, 2015, at 9:38 AM, Jean-Marc Valin <jmvalin@jmvalin.ca 
>> <mailto:jmvalin@jmvalin.ca>> wrote:
>> 
>> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
>> 
>> Just posted a new version that brings back maxaveragebitrate 
>> exactly as it was before.
>> 
>> Cheers,
>> 
>> Jean-Marc
> 
> 
> 
> _______________________________________________ payload mailing 
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
> 
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From nobody Mon Jan 12 22:42:38 2015
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <payload@ietf.org>
Date: Tue, 13 Jan 2015 08:42:28 +0200
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Subject: [payload] Call for adoption of RTP Payload Format for Non-Interleaved and Interleaved Parity FEC
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Hi,

The RTCweb FEC usage is looking for a solution the proposed RTP payload
format was presented and got support in the last IETF meeting. 

We would like to add a milestone for RTP Payload Format for Non-Interleaved
and Interleaved Parity FEC and adopt
draft-singh-payload-rtp-1d2d-parity-scheme-00 as the proposal for addressing
the mile stone.
 
Please provide feedback to the WG chairs by January 23rd
Thanks
Roni Even
Payload WG co-chair
 

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xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
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xmlns=3D"http://www.w3.org/TR/REC-html40"><head><META =
HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
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	font-family:"Calibri","sans-serif";
	color:windowtext;}
span.HTMLPreformattedChar
	{mso-style-name:"HTML Preformatted Char";
	mso-style-priority:99;
	mso-style-link:"HTML Preformatted";
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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>The RTCweb FEC =
usage is looking for a solution the proposed RTP payload format was =
presented and got support in the last IETF meeting. =
<o:p></o:p></p><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>We would =
like to add a milestone for <span style=3D'color:black'>RTP Payload =
Format for Non-Interleaved and Interleaved Parity FEC and adopt =
</span>draft-singh-payload-rtp-1d2d-parity-scheme-00 as the proposal for =
addressing the mile stone.<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Please =
provide feedback to the WG chairs by January =
23<sup>rd</sup><o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Thanks<o:p>=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Roni =
Even<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Payload WG =
co-chair<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre></div></body></html>
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From nobody Tue Jan 13 01:00:07 2015
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Jean-Marc Valin <jvalin@mozilla.com>, Jean-Marc Valin <jmvalin@jmvalin.ca>
Thread-Topic: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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References: <feed31dea4094e8b9cf3501fde502f86@STOEX21.sr.se> <548A3FEA.2070400@mozilla.com> <B13433EF-F6D2-4950-A0FA-9ECB70B3AF2C@cisco.com> <7E6F2FBF-3E29-4608-A9AE-2C5124BCE261@sverigesradio.se> <26b91d07-8e22-4790-b06e-564ee30b071d@cisco.com> <006201d0273f$e708a230$b519e690$@co.in> <1CDC64C8-B130-4FB7-B73A-6F894CB729F0@sverigesradio.se> <54A99858.1030102@mozilla.com> <D0CF1C53.41551%mzanaty@cisco.com> <54A9C9D0.8050002@mozilla.com> <06DA8517-E9D8-4726-8FAE-EB7D4B110674@sverigesradio.se> <54AB90FA.6090900@mozilla.com> <EFC567A0-F1CD-43F0-ADBF-E94BE2626EE3@cisco.com> <54B4A744.7080009@mozilla.com>
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Subject: Re: [payload] draft-ietf-payload-rtp-opus-05: maxaveragebitrate
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Subject: Re: [payload] Call for adoption of RTP Payload Format for Non-Interleaved and Interleaved Parity FEC
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Roni Even wrote:
> We would like to add a milestone forRTP Payload Format for Non-Interleaved and Interleaved Parity FEC and adoptdraft-singh-payload-rtp-1d2d-parity-scheme-00 as the proposal for addressing the mile stone.

I think we should do both of these things.


From nobody Tue Jan 13 05:28:58 2015
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Subject: [payload] Milestones changed for payload WG
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Changed milestone "Submit RTP Payload Format for Enhanced Variable
Rate Narrowband-Wideband Codec (EVRC-NW) for Proposed Standard",
resolved as "Done".

Changed milestone "Submit RTP profile for the carriage of SMPTE 336M
data for Proposed Standard", resolved as "Done".

Changed milestone "Submit How to Write an RTP Payload Format for
Informational", resolved as "Done".

Changed milestone "Submit RTP Payload Format for VP8 Video for
Proposed Standard", resolved as "Done".

Changed milestone "Submit RTP Payload Format for Standard apt-X and
Enhanced apt-X Codecs for Proposed Standard", resolved as "Done".

Changed milestone "Submit RTP Payload Format for G.711.0 for Proposed
Standard", resolved as "Done".

Changed milestone "Submit RTP Payload Format for High Efficiency Video
Coding for Proposed Standard.", resolved as "Done".

Changed milestone "Submit RTP Payload Format for Opus Speech and Audio
Codec as proposed standard", set due date to March 2015 from December
2012.

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


From nobody Tue Jan 13 06:21:03 2015
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Subject: [payload] I-D Action: draft-ietf-payload-rtp-opus-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Audio/Video Transport Payloads Working Group of the IETF.

        Title           : RTP Payload Format for Opus Speech and Audio Codec
        Authors         : Julian Spittka
                          Koen Vos
                          Jean-Marc Valin
	Filename        : draft-ietf-payload-rtp-opus-07.txt
	Pages           : 17
	Date            : 2015-01-13

Abstract:
   This document defines the Real-time Transport Protocol (RTP) payload
   format for packetization of Opus encoded speech and audio data
   necessary to integrate the codec in the most compatible way.
   Further, it describes media type registrations for the RTP payload
   format.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-opus/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-payload-rtp-opus-07


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

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


From nobody Tue Jan 13 06:22:53 2015
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Ali,

Just updated the draft with all the changes you requested.

Cheers,

	Jean-Marc

On 12/01/15 05:42 AM, Ali C. Begen (abegen) wrote:
> Here goes my chair review on version 06:
> 
> Section 3.1.2: 1) "network capacity” is not the term you are
> looking for. You need to say something like “available bandwidth”.
> 
> Section 4.2: 1) I suppose in table 2, “x” means not supported or
> N/A, which should be mentioned in the document. What do the empty
> slots mean?
> 
> Section 6.1: 1) For "Published specification:", you should not say
> "none", rather say "RFC [XXXX]" with an editor note the RFC editor
> to fix the number upon publication. 2) According to the template in
> rfc 6838, you are missing "Fragment identifier considerations",
> which I believe simply say "N/A."
> 
> 3) Change controller should be "IETF Payload Working Group
> delegated from the IESG.”
> 
> We should be ready to ship this draft once the above is addressed. 
> -acbegen
> 
> 
>> On Jan 6, 2015, at 9:38 AM, Jean-Marc Valin <jmvalin@jmvalin.ca 
>> <mailto:jmvalin@jmvalin.ca>> wrote:
>> 
>> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
>> 
>> Just posted a new version that brings back maxaveragebitrate
>> exactly as it was before.
>> 
>> Cheers,
>> 
>> Jean-Marc
> 
> 
> 
> _______________________________________________ payload mailing
> list payload@ietf.org 
> https://www.ietf.org/mailman/listinfo/payload
> 
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From: Justin Uberti <juberti@google.com>
Date: Tue, 13 Jan 2015 09:49:28 -0800
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To: "Timothy B. Terriberry" <tterribe@xiph.org>
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Subject: Re: [payload] Call for adoption of RTP Payload Format for Non-Interleaved and Interleaved Parity FEC
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--089e0115fc9ed09b25050c8c3ff5
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On Tue, Jan 13, 2015 at 4:51 AM, Timothy B. Terriberry <tterribe@xiph.org>
wrote:

> Roni Even wrote:
>
>> We would like to add a milestone forRTP Payload Format for
>> Non-Interleaved and Interleaved Parity FEC and adoptdraft-singh-payload-rtp-1d2d-parity-scheme-00
>> as the proposal for addressing the mile stone.
>>
>
> I think we should do both of these things.
>

I agree.

>
>

--089e0115fc9ed09b25050c8c3ff5
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Jan 13, 2015 at 4:51 AM, Timothy B. Terriberry <span dir=3D"ltr=
">&lt;<a href=3D"mailto:tterribe@xiph.org" target=3D"_blank">tterribe@xiph.=
org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Roni Even wrote=
:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
We would like to add a milestone forRTP Payload Format for Non-Interleaved =
and Interleaved Parity FEC and adoptdraft-singh-payload-rtp-<u></u>1d2d-par=
ity-scheme-00 as the proposal for addressing the mile stone.<br>
</blockquote>
<br>
I think we should do both of these things.<br></blockquote><div><br></div><=
div>I agree.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br></blockquote></d=
iv></div></div>

--089e0115fc9ed09b25050c8c3ff5--


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From: Thomas Edwards <Thomas.Edwards@fox.com>
To: "payload@ietf.org" <payload@ietf.org>
Thread-Topic: new I-D draft-edwards-rtp-ancillary RTP Payload for SMPTE ST 291 Ancillary Data
Thread-Index: AQHQL5quGtaZ7oBEiEudnxYs/ccOvg==
Date: Wed, 14 Jan 2015 01:37:39 +0000
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Subject: [payload] new I-D draft-edwards-rtp-ancillary RTP Payload for SMPTE ST 291 Ancillary Data
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--_000_D0DB04C22D344thomasedwardsfoxcom_
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Dear Payload WG,

I have submitted a new I-D (draft-edwards-rtp-ancillary) defining an RTP Pa=
yload for SMPTE ST 291 Ancillary Data.

SMPTE Ancillary data is used in today's professional serial digital video f=
ormats to carry a range of ancillary data types, including time code, KLV m=
etadata, Closed Captioning, and the Active Format Description (AFD).

The broadcast industry is preparing to move from the use of serial digital =
video over coax to "all-IP" professional networked media inside our broadca=
st plants.  Two of the needed standards have been there for years waiting f=
or affordable 10 GbE networks.  IETF has already standardized RTP carriage =
of uncompressed video in RFC 4175, and 24-bit uncompressed audio in RFC 319=
0 (as referenced by AES67).

The one remaining item needed is a way of carrying SMPTE ancillary data ove=
r RTP, which this I-D should address.

The I-D has already been through an informal review by SMPTE 32NF, and I ha=
ve worked in the comments received.  I do not expect that this process will=
 require formal liaison with between IETF and SMPTE.  The data model used f=
or the Ancillary data in RTP is based on an existing SMPTE standard to carr=
y Ancillary data over MPEG transport stream, so fortunately most of the "he=
avy lifting" on the data model design has already been done.

A new version of I-D, draft-edwards-rtp-ancillary-00.txt
has been successfully submitted by Thomas G. Edwards and posted to the
IETF repository.

Name: draft-edwards-rtp-ancillary
Revision: 00
Title: RTP Payload for SMPTE ST 291 Ancillary Data
Document date: 2015-01-13
Group: Individual Submission
Pages: 11
URL:            http://www.ietf.org/internet-drafts/draft-edwards-rtp-ancil=
lary-00.txt
Status:         https://datatracker.ietf.org/doc/draft-edwards-rtp-ancillar=
y/
Htmlized:       http://tools.ietf.org/html/draft-edwards-rtp-ancillary-00


Abstract:
   This memo describes an RTP Payload format for SMPTE Ancillary data,
   as defined by SMPTE ST 291-1.  SMPTE Ancillary data is generally used
   along with professional video formats to carry a range of ancillary
   data types, including time code, KLV metadata, Closed Captioning, and
   the Active Format Description (AFD).

-Thomas

--
Thomas Edwards
VP Engineering & Development
FOX Network Engineering and Operations
thomas.edwards@fox.com<mailto:thomas.edwards@fox.com>
Phone: +1.310.369.6696
10201 West Pico Blvd.
Los Angeles, CA 90035


--_000_D0DB04C22D344thomasedwardsfoxcom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; font-family: Calibri, sans-serif; font-size: 14=
px; color: rgb(0, 0, 0);">
<div>Dear Payload WG,</div>
<div><br>
</div>
<div>I have submitted a new I-D (draft-edwards-rtp-ancillary) defining an R=
TP Payload for SMPTE ST 291 Ancillary Data.</div>
<div><br>
</div>
<div>
<div>SMPTE Ancillary data is used in today's professional serial digital vi=
deo formats to carry a range of ancillary data types, including time code, =
KLV metadata, Closed Captioning, and the Active Format Description (AFD).</=
div>
<div><br>
</div>
<div>The broadcast industry is preparing to move from the use of serial dig=
ital video over coax to &quot;all-IP&quot; professional networked media ins=
ide our broadcast plants. &nbsp;Two of the needed standards have been there=
 for years waiting for affordable 10 GbE networks.
 &nbsp;IETF has already standardized RTP carriage of uncompressed video in =
RFC 4175, and 24-bit uncompressed audio in RFC 3190 (as referenced by AES67=
).&nbsp;</div>
<div><br>
</div>
<div>The one remaining item needed is a way of carrying SMPTE ancillary dat=
a over RTP, which this I-D should address.</div>
<div><br>
</div>
<div>The I-D has already been through an informal review by SMPTE 32NF, and=
 I have worked in the comments received. &nbsp;I do not expect that this pr=
ocess will require formal liaison with between IETF and SMPTE. &nbsp;The da=
ta model used for the Ancillary data in RTP
 is based on an existing SMPTE standard to carry Ancillary data over MPEG t=
ransport stream, so fortunately most of the &quot;heavy lifting&quot; on th=
e data model design has already been done.</div>
</div>
<div><br>
</div>
<div>A new version of I-D, draft-edwards-rtp-ancillary-00.txt</div>
<div>has been successfully submitted by Thomas G. Edwards and posted to the=
</div>
<div>IETF repository.</div>
<div><br>
</div>
<div>Name:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span>=
<span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>draft-edwar=
ds-rtp-ancillary</div>
<div>Revision:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </s=
pan>00</div>
<div>Title:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span=
><span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>RTP Payloa=
d for SMPTE ST 291 Ancillary Data</div>
<div>Document date:<span class=3D"Apple-tab-span" style=3D"white-space:pre"=
> </span>2015-01-13</div>
<div>Group:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span=
><span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>Individual=
 Submission</div>
<div>Pages:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span=
><span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>11</div>
<div>URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;<a href=3D"http://www.ietf.org/internet-drafts/draft-edwards-rtp-anci=
llary-00.txt">http://www.ietf.org/internet-drafts/draft-edwards-rtp-ancilla=
ry-00.txt</a></div>
<div>Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"htt=
ps://datatracker.ietf.org/doc/draft-edwards-rtp-ancillary/">
https://datatracker.ietf.org/doc/draft-edwards-rtp-ancillary/</a></div>
<div>Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://tools.=
ietf.org/html/draft-edwards-rtp-ancillary-00">
http://tools.ietf.org/html/draft-edwards-rtp-ancillary-00</a></div>
<div><br>
</div>
<div><br>
</div>
<div>Abstract:</div>
<div>&nbsp;&nbsp; This memo describes an RTP Payload format for SMPTE Ancil=
lary data,</div>
<div>&nbsp;&nbsp; as defined by SMPTE ST 291-1.&nbsp;&nbsp;SMPTE Ancillary =
data is generally used</div>
<div>&nbsp;&nbsp; along with professional video formats to carry a range of=
 ancillary</div>
<div>&nbsp;&nbsp; data types, including time code, KLV metadata, Closed Cap=
tioning, and</div>
<div>&nbsp;&nbsp; the Active Format Description (AFD).</div>
<div><br>
</div>
<div>-Thomas</div>
<div><br>
</div>
<div>-- </div>
<div>Thomas Edwards<span class=3D"Apple-tab-span" style=3D"white-space:pre"=
> </span></div>
<div>VP Engineering &amp; Development</div>
<div>FOX Network Engineering and Operations</div>
<div><a href=3D"mailto:thomas.edwards@fox.com">thomas.edwards@fox.com</a></=
div>
<div>Phone: &#43;1.310.369.6696</div>
<div>10201 West Pico Blvd.</div>
<div>Los Angeles, CA 90035</div>
<div><br>
</div>
</body>
</html>

--_000_D0DB04C22D344thomasedwardsfoxcom_--


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From: Thomas Edwards <Thomas.Edwards@fox.com>
To: "payload@ietf.org" <payload@ietf.org>
Thread-Topic: new I-D draft-edwards-rtp-ancillary RTP Payload for SMPTE ST 291 Ancillary Data
Thread-Index: AQHQL5quGtaZ7oBEiEudnxYs/ccOvpy/vE2A
Date: Wed, 14 Jan 2015 23:21:28 +0000
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Archived-At: <http://mailarchive.ietf.org/arch/msg/payload/NWsAguIQjOlw6s-h9TnC6Uxf-uE>
Subject: Re: [payload] new I-D draft-edwards-rtp-ancillary RTP Payload for SMPTE ST 291 Ancillary Data
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Dear Payload WG,

On the advice of the Payload WG chairs, I have re-submitted this I-D under =
the name "draft-edwards-payload-rtp-ancillary" so that it properly shows up=
 in the Payload WG list of individual submissions.

http://datatracker.ietf.org/doc/draft-edwards-payload-rtp-ancillary/

-Thomas

From: FNG <thomas.edwards@fox.com<mailto:thomas.edwards@fox.com>>
Date: Tuesday, January 13, 2015 at 5:37 PM
To: "payload@ietf.org<mailto:payload@ietf.org>" <payload@ietf.org<mailto:pa=
yload@ietf.org>>
Subject: new I-D draft-edwards-rtp-ancillary RTP Payload for SMPTE ST 291 A=
ncillary Data

Dear Payload WG,

I have submitted a new I-D (draft-edwards-rtp-ancillary) defining an RTP Pa=
yload for SMPTE ST 291 Ancillary Data.

SMPTE Ancillary data is used in today's professional serial digital video f=
ormats to carry a range of ancillary data types, including time code, KLV m=
etadata, Closed Captioning, and the Active Format Description (AFD).

The broadcast industry is preparing to move from the use of serial digital =
video over coax to "all-IP" professional networked media inside our broadca=
st plants.  Two of the needed standards have been there for years waiting f=
or affordable 10 GbE networks.  IETF has already standardized RTP carriage =
of uncompressed video in RFC 4175, and 24-bit uncompressed audio in RFC 319=
0 (as referenced by AES67).

The one remaining item needed is a way of carrying SMPTE ancillary data ove=
r RTP, which this I-D should address.

The I-D has already been through an informal review by SMPTE 32NF, and I ha=
ve worked in the comments received.  I do not expect that this process will=
 require formal liaison with between IETF and SMPTE.  The data model used f=
or the Ancillary data in RTP is based on an existing SMPTE standard to carr=
y Ancillary data over MPEG transport stream, so fortunately most of the "he=
avy lifting" on the data model design has already been done.

A new version of I-D, draft-edwards-rtp-ancillary-00.txt
has been successfully submitted by Thomas G. Edwards and posted to the
IETF repository.

Name: draft-edwards-rtp-ancillary
Revision: 00
Title: RTP Payload for SMPTE ST 291 Ancillary Data
Document date: 2015-01-13
Group: Individual Submission
Pages: 11
URL:            http://www.ietf.org/internet-drafts/draft-edwards-rtp-ancil=
lary-00.txt
Status:         https://datatracker.ietf.org/doc/draft-edwards-rtp-ancillar=
y/
Htmlized:       http://tools.ietf.org/html/draft-edwards-rtp-ancillary-00


Abstract:
   This memo describes an RTP Payload format for SMPTE Ancillary data,
   as defined by SMPTE ST 291-1.  SMPTE Ancillary data is generally used
   along with professional video formats to carry a range of ancillary
   data types, including time code, KLV metadata, Closed Captioning, and
   the Active Format Description (AFD).

-Thomas

--
Thomas Edwards
VP Engineering & Development
FOX Network Engineering and Operations
thomas.edwards@fox.com<mailto:thomas.edwards@fox.com>
Phone: +1.310.369.6696
10201 West Pico Blvd.
Los Angeles, CA 90035


--_000_D0DC397D2D475thomasedwardsfoxcom_
Content-Type: text/html; charset="us-ascii"
Content-ID: <95EC92D0CA85DE469A99285ADBF49B6F@namprd05.prod.outlook.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>
<div>
<div>Dear Payload WG,</div>
<div><br>
</div>
<div>On the advice of the Payload WG chairs, I have re-submitted this I-D u=
nder the name &quot;draft-edwards-payload-rtp-ancillary&quot; so that it pr=
operly shows up in the Payload WG list of individual submissions.</div>
<div><br>
</div>
<div><a href=3D"http://datatracker.ietf.org/doc/draft-edwards-payload-rtp-a=
ncillary">http://datatracker.ietf.org/doc/draft-edwards-payload-rtp-ancilla=
ry</a>/</div>
<div><br>
</div>
<div>-Thomas</div>
<div>
<div><br>
</div>
</div>
</div>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>FNG &lt;<a href=3D"mailto:tho=
mas.edwards@fox.com">thomas.edwards@fox.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Tuesday, January 13, 2015 at =
5:37 PM<br>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:payload=
@ietf.org">payload@ietf.org</a>&quot; &lt;<a href=3D"mailto:payload@ietf.or=
g">payload@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>new I-D draft-edwards-rtp-=
ancillary RTP Payload for SMPTE ST 291 Ancillary Data<br>
</div>
<div><br>
</div>
<div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; font-family: Calibri, sans-serif; font-size: 14p=
x; color: rgb(0, 0, 0);">
<div>Dear Payload WG,</div>
<div><br>
</div>
<div>I have submitted a new I-D (draft-edwards-rtp-ancillary) defining an R=
TP Payload for SMPTE ST 291 Ancillary Data.</div>
<div><br>
</div>
<div>
<div>SMPTE Ancillary data is used in today's professional serial digital vi=
deo formats to carry a range of ancillary data types, including time code, =
KLV metadata, Closed Captioning, and the Active Format Description (AFD).</=
div>
<div><br>
</div>
<div>The broadcast industry is preparing to move from the use of serial dig=
ital video over coax to &quot;all-IP&quot; professional networked media ins=
ide our broadcast plants. &nbsp;Two of the needed standards have been there=
 for years waiting for affordable 10 GbE networks.
 &nbsp;IETF has already standardized RTP carriage of uncompressed video in =
RFC 4175, and 24-bit uncompressed audio in RFC 3190 (as referenced by AES67=
).&nbsp;</div>
<div><br>
</div>
<div>The one remaining item needed is a way of carrying SMPTE ancillary dat=
a over RTP, which this I-D should address.</div>
<div><br>
</div>
<div>The I-D has already been through an informal review by SMPTE 32NF, and=
 I have worked in the comments received. &nbsp;I do not expect that this pr=
ocess will require formal liaison with between IETF and SMPTE. &nbsp;The da=
ta model used for the Ancillary data in RTP
 is based on an existing SMPTE standard to carry Ancillary data over MPEG t=
ransport stream, so fortunately most of the &quot;heavy lifting&quot; on th=
e data model design has already been done.</div>
</div>
<div><br>
</div>
<div>A new version of I-D, draft-edwards-rtp-ancillary-00.txt</div>
<div>has been successfully submitted by Thomas G. Edwards and posted to the=
</div>
<div>IETF repository.</div>
<div><br>
</div>
<div>Name:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span>=
<span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>draft-edwar=
ds-rtp-ancillary</div>
<div>Revision:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </s=
pan>00</div>
<div>Title:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span=
><span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>RTP Payloa=
d for SMPTE ST 291 Ancillary Data</div>
<div>Document date:<span class=3D"Apple-tab-span" style=3D"white-space:pre"=
> </span>2015-01-13</div>
<div>Group:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span=
><span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>Individual=
 Submission</div>
<div>Pages:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span=
><span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>11</div>
<div>URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;<a href=3D"http://www.ietf.org/internet-drafts/draft-edwards-rtp-anci=
llary-00.txt">http://www.ietf.org/internet-drafts/draft-edwards-rtp-ancilla=
ry-00.txt</a></div>
<div>Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"htt=
ps://datatracker.ietf.org/doc/draft-edwards-rtp-ancillary/">
https://datatracker.ietf.org/doc/draft-edwards-rtp-ancillary/</a></div>
<div>Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://tools.=
ietf.org/html/draft-edwards-rtp-ancillary-00">
http://tools.ietf.org/html/draft-edwards-rtp-ancillary-00</a></div>
<div><br>
</div>
<div><br>
</div>
<div>Abstract:</div>
<div>&nbsp;&nbsp; This memo describes an RTP Payload format for SMPTE Ancil=
lary data,</div>
<div>&nbsp;&nbsp; as defined by SMPTE ST 291-1.&nbsp;&nbsp;SMPTE Ancillary =
data is generally used</div>
<div>&nbsp;&nbsp; along with professional video formats to carry a range of=
 ancillary</div>
<div>&nbsp;&nbsp; data types, including time code, KLV metadata, Closed Cap=
tioning, and</div>
<div>&nbsp;&nbsp; the Active Format Description (AFD).</div>
<div><br>
</div>
<div>-Thomas</div>
<div><br>
</div>
<div>-- </div>
<div>Thomas Edwards<span class=3D"Apple-tab-span" style=3D"white-space:pre"=
> </span></div>
<div>VP Engineering &amp; Development</div>
<div>FOX Network Engineering and Operations</div>
<div><a href=3D"mailto:thomas.edwards@fox.com">thomas.edwards@fox.com</a></=
div>
<div>Phone: &#43;1.310.369.6696</div>
<div>10201 West Pico Blvd.</div>
<div>Los Angeles, CA 90035</div>
<div><br>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D0DC397D2D475thomasedwardsfoxcom_--


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From: Bo Burman <bo.burman@ericsson.com>
To: Roni Even <ron.even.tlv@gmail.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [payload] Call for adoption of RTP Payload Format for Non-Interleaved and Interleaved Parity FEC
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Subject: Re: [payload] Call for adoption of RTP Payload Format for Non-Interleaved and Interleaved Parity FEC
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I support this /Bo

From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Roni Even
Sent: den 13 januari 2015 07:42
To: payload@ietf.org
Subject: [payload] Call for adoption of RTP Payload Format for Non-Interlea=
ved and Interleaved Parity FEC

Hi,
The RTCweb FEC usage is looking for a solution the proposed RTP payload for=
mat was presented and got support in the last IETF meeting.

We would like to add a milestone for RTP Payload Format for Non-Interleaved=
 and Interleaved Parity FEC and adopt draft-singh-payload-rtp-1d2d-parity-s=
cheme-00 as the proposal for addressing the mile stone.



Please provide feedback to the WG chairs by January 23rd

Thanks

Roni Even

Payload WG co-chair



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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I support this /Bo<o:p=
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
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0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> payload =
[mailto:payload-bounces@ietf.org]
<b>On Behalf Of </b>Roni Even<br>
<b>Sent:</b> den 13 januari 2015 07:42<br>
<b>To:</b> payload@ietf.org<br>
<b>Subject:</b> [payload] Call for adoption of RTP Payload Format for Non-I=
nterleaved and Interleaved Parity FEC<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal">The RTCweb FEC usage is looking for a solution the p=
roposed RTP payload format was presented and got support in the last IETF m=
eeting.
<o:p></o:p></p>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;">We would like to add a milestone for <span style=3D"color=
:black">RTP Payload Format for Non-Interleaved and Interleaved Parity FEC a=
nd adopt </span>draft-singh-payload-rtp-1d2d-parity-scheme-00 as the propos=
al for addressing the mile stone.<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;"><o:p>&nbsp;</o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;">Please provide feedback to the WG chairs by January 23<su=
p>rd</sup><o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;">Thanks<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;">Roni Even<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;">Payload WG co-chair<o:p></o:p></span></pre>
<pre><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;"><o:p>&nbsp;</o:p></span></pre>
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Date: Thu, 29 Jan 2015 17:52:00 +0200
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Subject: [payload] Result on Call for adoption of RTP Payload Format for Non-Interleaved and Interleaved Parity FEC
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Hi,

There was support to adopt this subject as a work item and adopt
draft-singh-payload-rtp-1d2d-parity-scheme-00 as the initial document to
address the milestone.

 

I will create a milestone and the authors should submit the document as a WG
document

 

Roni Even

Payload co-chair

 

From: Roni Even [mailto:ron.even.tlv@gmail.com] 
Sent: 13 January, 2015 8:42 AM
To: 'payload@ietf.org'
Subject: Call for adoption of RTP Payload Format for Non-Interleaved and
Interleaved Parity FEC

 

Hi,

The RTCweb FEC usage is looking for a solution the proposed RTP payload
format was presented and got support in the last IETF meeting. 

We would like to add a milestone for RTP Payload Format for Non-Interleaved
and Interleaved Parity FEC and adopt
draft-singh-payload-rtp-1d2d-parity-scheme-00 as the proposal for addressing
the mile stone.
 
Please provide feedback to the WG chairs by January 23rd
Thanks
Roni Even
Payload WG co-chair
 

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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi,<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>There was support to =
adopt this subject as a work item and adopt =
</span>draft-singh-payload-rtp-1d2d-parity-scheme-00 as the initial =
document to address the milestone.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I will =
create a milestone and the authors should submit the document as a WG =
document<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Roni Even<o:p></o:p></p><p class=3DMsoNormal>Payload =
co-chair<span style=3D'color:#1F497D'><o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Roni Even [mailto:ron.even.tlv@gmail.com] <br><b>Sent:</b> 13 January, =
2015 8:42 AM<br><b>To:</b> 'payload@ietf.org'<br><b>Subject:</b> Call =
for adoption of RTP Payload Format for Non-Interleaved and Interleaved =
Parity FEC<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>The RTCweb FEC =
usage is looking for a solution the proposed RTP payload format was =
presented and got support in the last IETF meeting. =
<o:p></o:p></p><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>We would =
like to add a milestone for <span style=3D'color:black'>RTP Payload =
Format for Non-Interleaved and Interleaved Parity FEC and adopt =
</span>draft-singh-payload-rtp-1d2d-parity-scheme-00 as the proposal for =
addressing the mile stone.<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Please =
provide feedback to the WG chairs by January =
23<sup>rd</sup><o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Thanks<o:p>=
</o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Roni =
Even<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Payload WG =
co-chair<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre></div></div></body></html>
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From nobody Fri Jan 30 02:54:34 2015
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To: Gunnar Persson N <gunnar.n.persson@ericsson.com>, "payload@ietf.org" <payload@ietf.org>
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Archived-At: <http://mailarchive.ietf.org/arch/msg/payload/8cxOhHA6aYENtmHOKh6w2Wmwmsg>
Subject: Re: [payload] Question regarding RFC 4867
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Hi,

As author of this RFC, here is my answer.

On 2015-01-12 11:54, Gunnar Persson N wrote:
> Hi
> 
> I have a simple question regarding RFC 4867
> 
> In chapter…
> 
> 
>         4.3.1 <https://tools.ietf.org/html/rfc4867#section-4.3.1>.  The
>         Payload Header
> 
> …it is stated:
> 
>  
> 
> Therefore, if a terminal continuously wishes to receive frames in the
> 
>    same mode X, it needs to set CMR=X for all its outbound payloads, and
> 
>    if a terminal has no preference in which mode to receive, it SHOULD
> 
>    set CMR=15 in all its outbound payloads.
> 
>  
> 
> Question:
> 
> *Does “all its outbound payloads” mean “every RTP frame” (I e every 20ms)? *

payloads are RTP payloads, thus within each RTP packet sent for that
stream that contain an AMR payload.

> 
>  
> 
> Comment:
> 
> On air interface CMR value is only communicated every second speech
> frame (I e every 40ms). I assume the idea is that BSS shall remember the
> last communicated CMR from mobile terminal and insert this CMR in all
> RTP frames. Please comment if my assumption is correct.

Yes, that becomes the implication if one sends RTP packets which
contains only a single 20ms AMR frame.

Cheers

Magnus Westerlund

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Changed milestone "Submit RTP Payload Format for the iSAC codec for
Proposed Standard", set due date to June 2015 from April 2011.

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

