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From: "Julian Spittka" <julian.spittka@skype.net>
To: <payload@ietf.org>
Date: Tue, 5 Jul 2011 02:10:48 +0200
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Cc: 'Koen Vos' <koen.vos@skype.net>, 'Jean-Marc Valin' <jmvalin@jmvalin.ca>
Subject: [payload] RTP Payload Format and File Storage Format for Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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All,

Please find a first draft "draft-spittka-payload-rtp-opus-00" for the "RTP
Payload Format and File Storage Format for Opus Speech and Audio Codec"
uploaded at
https://datatracker.ietf.org/doc/draft-spittka-payload-rtp-opus/. 

Any feedback, questions, or comments are highly welcome.

Best Regards,

Julian.

---
Julian Spittka
Skype

see & chat: julian_spittka
write: julian.spittka@skype.net



From ron.even.tlv@gmail.com  Tue Jul  5 14:01:25 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <julian.spittka@skype.net>, <codec@ietf.org>
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Cc: payload@ietf.org
Subject: Re: [payload] [codec] RTP Payload Format and File Storage Format for Opus Speech	and Audio Codec - draft-spittka-payload-rtp-opus-00
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Hi,
I noticed the discussion that started on the codec mailing list. 
Since this work will be handled in the payload WG please send the comments
to payload@ietf.org 
Thanks
Roni Even
Payload co-chair

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Julian Spittka
> Sent: Tuesday, July 05, 2011 3:06 AM
> To: codec@ietf.org
> Subject: [codec] RTP Payload Format and File Storage Format for Opus
> Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
> 
> All,
> 
> Please find a first draft "draft-spittka-payload-rtp-opus-00" for the
> "RTP
> Payload Format and File Storage Format for Opus Speech and Audio Codec"
> uploaded at
> https://datatracker.ietf.org/doc/draft-spittka-payload-rtp-opus/.
> 
> Any feedback, questions, or comments are highly welcome.
> 
> Best Regards,
> 
> Julian.
> 
> ---
> Julian Spittka
> Skype
> 
> see & chat: julian_spittka
> write: julian.spittka@skype.net
> 
> 
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From ron.even.tlv@gmail.com  Tue Jul  5 14:17:44 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <bens@alum.mit.edu>, <julian.spittka@skype.net>
References: <00f501cc3aa7$551264c0$ff372e40$@spittka@skype.net> <4E133E21.3050001@fas.harvard.edu>
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Cc: codec@ietf.org, payload@ietf.org
Subject: Re: [payload] [codec] RTP Payload Format and File Storage Format for Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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Hi,
I find it difficult to address all you comments being in XML patch format.
In general the payload specification should provide enough information that
will enable the reader to create or parse the RTP payload without
understanding the codec payload based on the payload specification. As such
it is useful to provide information that will explain the general concepts
of the codec and the optional parameters. So I find sections 1 and 3 useful
and they should be in the document
Roni Even
   

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Benjamin M. Schwartz
> Sent: Tuesday, July 05, 2011 7:39 PM
> To: julian.spittka@skype.net
> Cc: codec@ietf.org
> Subject: Re: [codec] RTP Payload Format and File Storage Format for
> Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
> 
> On 07/04/2011 08:06 PM, Julian Spittka wrote:
> > Any feedback, questions, or comments are highly welcome.
> 
> Thank you for doing this work.  I think your proposed RTP and SDP
> formats are very nice and a good step for us.
> 
> I have a few comments.
> 
> 1.  I think the description of Opus should be removed.  There's no need
> to duplicate the contents of the Opus RFC.
> 
> 2.  I think the payload specification should make (almost) no mention
> of Opus frames, modes, etc.  These aspects of Opus are entirely
> internal to the payload, which is a "black box".  As far as the RTP
> specification is concerned, Opus consists of (almost-)arbitrary-
> duration "packets" and that's it.
> 
> 3.  To avoid confusion, I think we should be very careful and clear
> about the distinction between audio bandwidth and samplerate, as their
> relationship in Opus is far from simple.
> 
> I have implemented these ideas, along with a few minor changes, as a
> patch to the draft .xml, attached.  Please feel free to discard this
> patch; I mean it only as a way to communicate my suggestions more
> clearly.
> 
> --Ben


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Thread-Topic: WGLC for draft-ietf-payload-rtp-klv-01
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: <payload@ietf.org>
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Subject: [payload] WGLC for draft-ietf-payload-rtp-klv-01
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Hi everyone,

This is to announce a 2-week WGLC for the following draft (ends July =
20th):
http://datatracker.ietf.org/doc/draft-ietf-payload-rtp-klv/

Please review the document and post any questions/concerns you have to =
the mailing list.

Thanks.
-acbegen

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From: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [payload] [codec] RTP Payload Format and File Storage Format for Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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I would prefer to see the file format separated into a separate draft =
for two reason. One, it allows an implementation to say they do RFC XXXX =
and have that be clear if the mean the RTP payload or the file format. =
Two, I suspect the file format conversation might take longer to =
resolve.=20

Cullen

PS - this discussion and draft needs to happen in the payloads WG, feel =
free to CC the rtcweb@ietf.org mailing list.

On Jul 4, 2011, at 6:06 PM, Julian Spittka wrote:

> All,
>=20
> Please find a first draft "draft-spittka-payload-rtp-opus-00" for the =
"RTP
> Payload Format and File Storage Format for Opus Speech and Audio =
Codec"
> uploaded at
> https://datatracker.ietf.org/doc/draft-spittka-payload-rtp-opus/.=20
>=20
> Any feedback, questions, or comments are highly welcome.
>=20
> Best Regards,
>=20
> Julian.
>=20
> ---
> Julian Spittka
> Skype
>=20
> see & chat: julian_spittka
> write: julian.spittka@skype.net
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From magnus.westerlund@ericsson.com  Thu Jul  7 03:53:35 2011
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Subject: Re: [payload] [codec] RTP Payload Format and File Storage Format for Opus	Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On 2011-07-06 18:43, Cullen Jennings wrote:
> 
> I would prefer to see the file format separated into a separate draft for two reason. One, it allows an implementation to say they do RFC XXXX and have that be clear if the mean the RTP payload or the file format. Two, I suspect the file format conversation might take longer to resolve. 
> 
> Cullen
> 
> PS - this discussion and draft needs to happen in the payloads WG, feel free to CC the rtcweb@ietf.org mailing list.

Cullen, I assume you mean CODEC WG rather than the RTCWEB WG.

Cheer

Magnus Westerlund

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Subject: Re: [payload] [codec] RTP Payload Format and File Storage Format for Opus	Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On Jul 7, 2011, at 4:53 AM, Magnus Westerlund wrote:

> On 2011-07-06 18:43, Cullen Jennings wrote:
>>=20
>> I would prefer to see the file format separated into a separate draft =
for two reason. One, it allows an implementation to say they do RFC XXXX =
and have that be clear if the mean the RTP payload or the file format. =
Two, I suspect the file format conversation might take longer to =
resolve.=20
>>=20
>> Cullen
>>=20
>> PS - this discussion and draft needs to happen in the payloads WG, =
feel free to CC the rtcweb@ietf.org mailing list.
>=20
> Cullen, I assume you mean CODEC WG rather than the RTCWEB WG.

yes - thank you. too much RTCweb on my mind and was just typing while on =
a call. Always bad to do


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Subject: Re: [payload] [codec] RTP Payload Format and File Storage Format for Opus	Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On 07/05/2011 05:15 PM, Roni Even wrote:
> In general the payload specification should provide enough information =
that
> will enable the reader to create or parse the RTP payload without
> understanding the codec payload based on the payload specification. As =
such
> it is useful to provide information that will explain the general conce=
pts
> of the codec and the optional parameters. So I find sections 1 and 3 us=
eful
> and they should be in the document

Despite having removed Section 3 in my suggested edit, I agree.  The
payload document should present relevant information about Opus, such as
the available frame sizes and the significance of the low-bitrate
redundancy feature.  However, it should not discuss non-normative
behaviors of the reference coder, or internal details such as subsampling=

and coding modes.

The most important thing to understand about Opus coding modes is that th=
e
encoder can, and does, dynamically switch among them on a packet-by-packe=
t
basis in order to optimize quality.  The draft seems to have been written=

under the assumption that a stream, once started in a particular mode, ma=
y
not be able to switch to other modes.   This is no longer true, as
explained in the latest Opus draft.

--Ben


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From Even.roni@huawei.com  Mon Jul 11 07:27:30 2011
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Date: Mon, 11 Jul 2011 17:24:43 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [payload] RTP Payload Format and File Storage Format for Opus	Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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Hi,
I reviewed the draft and it is a good start. Based on Cullen recommendation
I am posting to both codec and payload mailing lists.

I did not review the storage format since I understand that it will be
removed from this draft and discussed separately.

I have some comments

1. In the introduction "thus, making it the codec of choice for various
applications using the Internet or similar networks" maybe change to "thus,
making it useful for various applications using the Internet or similar
networks". The first sentence have a market share declaration.

2. In section 3.2 when you talk about network bandwidth what does it mean.
Do you calculate it based on the whole IP packet, the RTP packet or just the
opus payload. I assume it is the last one so please define the term network
bandwidth.

3. in section 3.2.1 you talk about "average bit rate". How do you measure
average bit rate, is it average in 3 minutes, in 1 minute?

4. In section 3.4 "The decision to send "in-band" FEC information is
entirely controlled by the encoder ". This is true only if the decoder
signaled support for FEC.

5. In section 3.5 "If a decoder can not take advantage of the benefits of a
stereo  signal this SHOULD  be indicated at the time a session is set up.
In  that case the sending side SHOULD  NOT send stereo signals as it leads
to an inefficient usage of network bandwidth". I think that the SHOULD NOT
need to be MUST NOT. Is there a reason for the sender to ignore the receiver
request?

6. In section 4.1 is there a specific reason for stating "The receiving side
MUST be prepared to receive duplicates of RTP packets.  Only one of those
payloads MUST be provided to the Opus decoder for decoding and others MUST
be discarded" since this is a standard RTP behavior. The duplicate packets
can be reported in RTCP.

7. In section 4.2 "The maximum packet length is limited to the amount of
encoded data representing 120 ms of speech or audio data". I think that is
should also be limted by the MTU size if you do not want to fragment the RTP
packets.

8.In section 6 the whole section talks about what an encoder can do and not
about how to find and report congestion I think that maybe instead of "It is
RECOMMENDED that congestion control is applied during the transmission of
Opus encoded data ." say " that "receivers MUST monitor the packet loss rate
to ensure congestion is not caused, following the guidelines in Section 2 of
RFC 3551." . 

9. In section 7.1 you mention a required parameter "Rate". Is this a
parameter that need to be specified in the SDP. You only need to say in the
mapping to SDP that The RTP clock rate in "a=rtpmap" MUST be  48000. You
example 1 does not have the rate parameter just a=rtpmap:101 opus/48000

10. In section 7.1 maxcodedaudiobandwidth. Is this relay an issue to signal
the sampling rate. Isn't it enough to have the maximum bandwidth which will
affect the sampling rate according to table 1.

11. In section 7.1 I noticed that ptime, maxptime and minptime signaled by
the receiver are just recommendations and not the desired value as in RFC
3264, and the sender can send any size it wishes upto 120. So what is the
benfit for the receiver to signal it. I thought that this is a receive
capability.

12. In section 71. Why define maxaveragebitrate and not use the SDP b=
parameter.

13. In section 7.1 Why need the stereo parameter there is the number of
channels in the rtpmap. It should say a=rtpmap:101 opus/48000/2 instead of
a=rtpmap:101 opus/48000 and stereo=1.

14.In 7.1 the CBR parameter let the decoder specify if to use CBR or VBR.
Yet the CBR does not specify a value making it look a bit like VBR. I think
that there should be an option for the receiver to ask for a specific CBR
rate.

15.About fec and useinbandfec. I did not see if there an overhead for using
fec that should be reflected in the bandwidth considerations.


Some nits

1. In the introduction " The codec has a very low algorithmic delay and is
is"
2. In 4.1 " according to Table 2 to determine the RTP timestamp" maybe "
according to Table 2 to determine the increment in RTP timestamp"


Roni Even as individual



I would like to have two comment as Payload WG co-chair

1. Person & email address to contact for further information:

      SILK  Support silksupport@skype.net
I think IANA will prefer a person and not a general email address

2. The change controller Should be "IETF Payload working group delegated
from the IESG.". I think this was a conclusion on previous discussion





> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
> Behalf Of Julian Spittka
> Sent: Tuesday, July 05, 2011 3:11 AM
> To: payload@ietf.org
> Cc: 'Koen Vos'; 'Jean-Marc Valin'
> Subject: [payload] RTP Payload Format and File Storage Format for Opus
> Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
> 
> All,
> 
> Please find a first draft "draft-spittka-payload-rtp-opus-00" for the
> "RTP
> Payload Format and File Storage Format for Opus Speech and Audio Codec"
> uploaded at
> https://datatracker.ietf.org/doc/draft-spittka-payload-rtp-opus/.
> 
> Any feedback, questions, or comments are highly welcome.
> 
> Best Regards,
> 
> Julian.
> 
> ---
> Julian Spittka
> Skype
> 
> see & chat: julian_spittka
> write: julian.spittka@skype.net
> 
> 
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload


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Date: Mon, 11 Jul 2011 10:53:44 -0400
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To: Roni Even <Even.roni@huawei.com>
References: <00f601cc3aa7$ff4c9460$fde5bd20$%spittka@skype.net> <008901cc3fd6$4d18e490$e74aadb0$%roni@huawei.com>
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Subject: Re: [payload] RTP Payload Format and File Storage Format for	Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On 07/11/2011 10:24 AM, Roni Even wrote:
> 5. In section 3.5 "If a decoder can not take advantage of the benefits =
of a
> stereo  signal this SHOULD  be indicated at the time a session is set u=
p.
> In  that case the sending side SHOULD  NOT send stereo signals as it le=
ads
> to an inefficient usage of network bandwidth". I think that the SHOULD =
NOT
> need to be MUST NOT. Is there a reason for the sender to ignore the rec=
eiver
> request?

If a stream is being sent to many recipients, most of whom prefer stereo,=

then it may be appropriate to send a single stereo stream to all
recipients, rather than increase the encoder CPU requirements by encoding=

multiple versions of the stream.

> 14.In 7.1 the CBR parameter let the decoder specify if to use CBR or VB=
R.
> Yet the CBR does not specify a value making it look a bit like VBR. I t=
hink
> that there should be an option for the receiver to ask for a specific C=
BR
> rate.

I think the implication was that in CBR mode, "maxaveragebitrate" also
sets a hard ceiling on the size of an individual packet.  I agree that
this is not a perfect solution.  Personally, I would prefer that the
receiver indicate both a hard limit and a soft target (or similar).

--Ben


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From Even.roni@huawei.com  Mon Jul 11 11:57:10 2011
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Date: Mon, 11 Jul 2011 21:53:58 +0300
From: Roni Even <Even.roni@huawei.com>
To: ietf-types@iana.org
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Cc: draft-ietf-payload-rfc3189bis.all@tools.ietf.org, payload@ietf.org
Subject: [payload] Updating the registration of audio and video DV media subtype
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Hi,

 

draft-ietf-payload-rfc3189bis-01 has passed Working Group Last Call in the
Payload Working Group.

 

Note that it an update to existing RFC3189 registrations
http://www.iana.org/assignments/rtp-parameters for DV media subtype.

 

The new registrations are in Section 3.1 of the document.
http://tools.ietf.org/html/draft-ietf-payload-rfc3189bis-01#section-3.1 

 

The information is also available below. 

 

I noticed that there is a need to change the change controller from " IETF
Audio/Video Transport working group delegated from the   IESG"  to " IETF
Payload  working group delegated from the   IESG"

 

 

Other comments on the registration are welcome.

 

Thanks

Roni Even 

Payload co-chair

 

 

3.1.1.  Media Type Registration for DV Video

   

 

Type name:  video

 

   Subtype name:  DV

 

   Required parameters:

 

      encode:  type of DV format.  Permissible values for encode are

         SD-VCR/525-60,

         SD-VCR/625-50,

         HD-VCR/1125-60,

         HD-VCR/1250-50,

         SDL-VCR/525-60,

         SDL-VCR/625-50,

         314M-25/525-60,

         314M-25/625-50,

         314M-50/525-60,

         314M-50/625-50,

         370M/1080-60i,

         370M/1080-50i,

         370M/720-60p,

         370M/720-50p,

         306M/525-60 (for backward compatibility),

         and 306M/625-50 (for backward compatibility).

 

   Optional parameters:

 

      audio:  whether the DV stream includes audio data or not.

         Permissible values for audio are bundled and none.  Defaults to
none.

 

   Encoding considerations:

 

         DV video can be transmitted with RTP as specified in RFCXXXX(This
document).  Other transport methods are not specified.

 

   Security considerations:

 

         See Section 4 of RFCXXXX (This document).

 

   Interoperability considerations:  NONE

 

   Public specification:

 

         IEC 61834 Standard

         SMPTE 314M

         SMPTE 370M

         RFCXXXX (This document)

         SMPTE 306M (for backward compatibility).

 

   Applications that use this media type:  Audio and video streaming and
conferencing tools.

 

   Additional information:  NONE

 

   Person & email address to contact for further information:

 

         Katsushi Kobayashi

 

         e-mail: ikob@ni.aist.go.jp

 

   Intended usage:  COMMON

 

   Restrictions on usage:  This media type depends on RTP framing, and hence
is only defined for transfer via RTP (RFC 3550).  Transfer within other
framing protocols is not defined at this time.

 

   Author:

 

         Katsushi Kobayashi

 

   Change controller:

 

         IETF Audio/Video Transport working group delegated from the

         IESG

 

 

3.1.2.  Media Type Registration for DV Audio

 

   Type name:  audio

 

   Subtype name:  DV

 

   Required parameters:

 

      encode:  type of DV format.  Permissible values for encode are

         SD-VCR/525-60,

         SD-VCR/625-50,

         HD-VCR/1125-60,

         HD-VCR/1250-50,

         SDL-VCR/525-60,

         SDL-VCR/625-50,

         314M-25/525-60,

         314M-25/625-50,

         314M-50/525-60,

         314M-50/625-50,

         370M/1080-60i,

         370M/1080-50i,

         370M/720-60p,

         370M/720-50p,

         306M/525-60 (for backward compatibility),

         and 306M/625-50 (for backward compatibility).

 

   Optional parameters:

 

      audio:  whether the DV stream includes audio data or not.

         Permissible values for audio are bundled and none.  Defaults to
none.

 

   Encoding considerations:

 

         DV audio can be transmitted with RTP as specified in RFCXXXX(This
document).  Other transport methods are not specified.

 

   Security considerations:

 

         See Section 4 of RFCXXXX (This document).

 

   Interoperability considerations:  NONE

 

   Published specification:

 

         IEC 61834 Standard

         SMPTE 314M

         SMPTE 370M

         RFCXXXX (This document)

         SMPTE 306M (for backward compatibility).

 

   Applications that use this media type:  Audio and video streaming and
conferencing tools.

 

   Additional information:  NONE

 

   Person & email address to contact for further information:

 

         Katsushi Kobayashi

 

         e-mail: ikob@ni.aist.go.jp

 

   Intended usage:  COMMON

 

   Restrictions on usage:  This media type depends on RTP framing, and hence
is only defined for transfer via RTP (RFC 3550).  Transfer      within other
framing protocols is not defined at this time.

 

 

   Author:

 

         Katsushi Kobayashi

 

   Change controller:

 

         IETF Audio/Video Transport working group delegated from the

         IESG

 

 


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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>Hi,<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoPlainText><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>draft-ietf-payload-rfc3189bis-01</span><span style='font-family:"Courier New"'> </span>has passed Working Group Last Call in the Payload Working Group.<o:p></o:p></p><p class=MsoPlainText><o:p>&nbsp;</o:p></p><p class=MsoPlainText>Note that it an update to existing RFC3189 registrations <a href="http://www.iana.org/assignments/rtp-parameters">http://www.iana.org/assignments/rtp-parameters</a> for DV media subtype.<o:p></o:p></p><p class=MsoPlainText><o:p>&nbsp;</o:p></p><p class=MsoPlainText>The new registrations are in Section 3.1 of the document. <a href="http://tools.ietf.org/html/draft-ietf-payload-rfc3189bis-01#section-3.1">http://tools.ietf.org/html/draft-ietf-payload-rfc3189bis-01#section-3.1</a> <o:p></
 o:p></p>
nbsp;</o:p></p><p class=MsoNormal>The information is also available below. <o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoPlainText><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>I noticed that there is a need to change the change controller from &quot;</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> IETF Audio/Video Transport working group delegated from the&nbsp;&nbsp; IESG&quot; </span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>&nbsp;to &quot;</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> IETF Payload &nbsp;working group delegated from the&nbsp;&nbsp; IESG&quot;<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>Other comments on the registration are welcome.<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p cla
 ss=MsoNo
><p class=MsoNormal>Roni Even <o:p></o:p></p><p class=MsoNormal>Payload co-chair<o:p></o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>3.1.1.&nbsp; Media Type Registration for DV Video<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; <o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>Type name:&nbsp; video<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Subtype name:&nbsp; DV<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-
 family:"
; Required parameters:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; encode:&nbsp; type of DV format.&nbsp; Permissible values for encode are<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SD-VCR/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SD-VCR/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; HD-VCR/1125-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; HD-VCR/1250-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp
 ;&nbsp;&
DL-VCR/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SDL-VCR/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-25/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-25/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-50/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-50/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/1080-60i,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp
 ;&nbsp;&
bsp;&nbsp; 370M/1080-50i,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/720-60p,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/720-50p,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 306M/525-60 (for backward compatibility),<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and 306M/625-50 (for backward compatibility).<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Optional parameters:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p
  class=M
font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; audio:&nbsp; whether the DV stream includes audio data or not.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Permissible values for audio are bundled and none.&nbsp; Defaults to&nbsp; none.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Encoding considerations:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; DV video can be transmitted with RTP as specified in RFCXXXX(This document).&nbsp; Other transport methods are not specified.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o
 :p></spa
xt><span style='font-family:"Courier New"'>&nbsp;&nbsp; Security considerations:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; See Section 4 of RFCXXXX (This document).<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Interoperability considerations:&nbsp; NONE<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Public specification:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs
 p;&nbsp;
FR style='font-family:"Courier New"'>IEC 61834 Standard<o:p></o:p></span></p><p class=MsoPlainText><span lang=FR style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SMPTE 314M<o:p></o:p></span></p><p class=MsoPlainText><span lang=FR style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SMPTE 370M<o:p></o:p></span></p><p class=MsoPlainText><span lang=FR style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span style='font-family:"Courier New"'>RFCXXXX (This document)<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SMPTE 306M (for backward compatibility).<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Applications that use this media type:&nbsp; Audio and video s
 treaming
o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Additional information:&nbsp; NONE<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Person &amp; email address to contact for further information:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Katsushi Kobayashi<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; e-mail: ikob@ni.aist.go.jp<o:p></o:p></span></p><p class=
 MsoPlain
mily:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Intended usage:&nbsp; COMMON<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Restrictions on usage:&nbsp; This media type depends on RTP framing, and hence is only defined for transfer via RTP (RFC 3550).&nbsp; Transfer within other framing protocols is not defined at this time.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Author:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Katsushi Kobayashi<o:p></o:p></sp
 an></p><
n style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Change controller:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IETF Audio/Video Transport working group delegated from the<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IESG<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>3.1.2.&nbsp; Media Type Registration for DV Audio<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p>
 </span><
<span style='font-family:"Courier New"'>&nbsp;&nbsp; Type name:&nbsp; audio<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Subtype name:&nbsp; DV<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Required parameters:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; encode:&nbsp; type of DV format.&nbsp; Permissible values for encode are<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SD-VCR/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Couri
 er New"'
&nbsp;&nbsp;&nbsp;&nbsp; SD-VCR/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; HD-VCR/1125-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; HD-VCR/1250-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SDL-VCR/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SDL-VCR/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-25/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-25/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-f
 amily:"C
&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;314M-50/525-60,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 314M-50/625-50,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/1080-60i,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/1080-50i,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/720-60p,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 370M/720-50p,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 306M/525-60 (for backward compatibility),<o:p></o:p></span></p><p class
 =MsoPlai
amily:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and 306M/625-50 (for backward compatibility).<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Optional parameters:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; audio:&nbsp; whether the DV stream includes audio data or not.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Permissible values for audio are bundled and none.&nbsp; Defaults to none.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Encodin
 g consid
n></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; DV audio can be transmitted with RTP as specified in RFCXXXX(This document).&nbsp; Other transport methods are not specified.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Security considerations:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; See Section 4 of RFCXXXX (This document).<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nb
 sp; Inte
ns:&nbsp; NONE<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Published specification:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IEC 61834 Standard<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;SMPTE 314M<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span lang=FR style='font-family:"Courier New"'>SMPTE 370M<o:p></o:p></span></p><p class=MsoPlainText><span lang=FR style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFCXXXX (This document)<o:p></o:p></span><
 /p><p cl
ng=FR style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span style='font-family:"Courier New"'>SMPTE 306M (for backward compatibility).<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Applications that use this media type:&nbsp; Audio and video streaming and conferencing tools.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Additional information:&nbsp; NONE<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Person &amp; email address to contact for further information:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-f
 amily:"C
/o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Katsushi Kobayashi<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; e-mail: ikob@ni.aist.go.jp<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Intended usage:&nbsp; COMMON<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Restrictions on usage:&nbsp; This media type depends on RTP framing, and hence is only defined for transfer via RTP (RFC 3550).&nbsp; Transfer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; within other framing 
 protocol
time.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Author:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Katsushi Kobayashi<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp; Change controller:<o:p></o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IETF Audio/Video Transport worki
 ng group
/o:p></span></p><p class=MsoPlainText><span style='font-family:"Courier New"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IESG<o:p></o:p></span></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal><o:p>&nbsp;</o:p></p></div></body></html>

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Subject: Re: [payload] RTP Payload Format and File Storage Format for	Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On 7/11/2011 10:24 AM, Roni Even wrote:
> 11. In section 7.1 I noticed that ptime, maxptime and minptime signaled by
> the receiver are just recommendations and not the desired value as in RFC
> 3264, and the sender can send any size it wishes upto 120. So what is the
> benfit for the receiver to signal it. I thought that this is a receive
> capability.

ptime is I think desired but not required.  maxptime and minptime should 
be hard receiver limits.
Per Colin on AVT on 30 Mar 2006, ptime is a "hint", but the encoder can 
vary up to maxptime (and I
assume down to minptime).  (I didn't bother researching further).

-- 
Randell Jesup
randell-ietf@jesup.org


From stephen.botzko@gmail.com  Mon Jul 11 13:57:05 2011
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Subject: Re: [payload] RTP Payload Format and File Storage Format for Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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--bcaec5486088ea92f304a7d16c11
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ptime is described in both RFC3264 and RFC4566, which is a bit confusing.
But the normative text for ptime is in RFC3264, I think that should be
referenced in 7.1. (maxptime and minptime only appear in RFC4566).

ptime, maxptime, and minptime are media attributes, so they apply to all
codecs on the m= line.  This is not that clear in this text, as it seems to
mandate specific values for this OPUS.  For instance:

>>> maxptime

Possible values are 3, 5,
      10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
      rounded up to the next full integer value up to a maximum value of
      120 as defined in Section 4 and Section 5 of this document.  If no
      value is specified, 120 is assumed as default.
>>>

The signaled maxptime certainly might exceed 120 ms, since the offerer might
be quite willing to receive some other codec on the m-line at that interval.

Regards,
Stephen Botzko

On Mon, Jul 11, 2011 at 4:23 PM, Randell Jesup <randell-ietf@jesup.org>wrote:

> On 7/11/2011 10:24 AM, Roni Even wrote:
>
>> 11. In section 7.1 I noticed that ptime, maxptime and minptime signaled by
>> the receiver are just recommendations and not the desired value as in RFC
>> 3264, and the sender can send any size it wishes upto 120. So what is the
>> benfit for the receiver to signal it. I thought that this is a receive
>> capability.
>>
>
> ptime is I think desired but not required.  maxptime and minptime should be
> hard receiver limits.
> Per Colin on AVT on 30 Mar 2006, ptime is a "hint", but the encoder can
> vary up to maxptime (and I
> assume down to minptime).  (I didn't bother researching further).
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
>
> ______________________________**_________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/**listinfo/payload<https://www.ietf.org/mailman/listinfo/payload>
>

--bcaec5486088ea92f304a7d16c11
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Content-Transfer-Encoding: quoted-printable

ptime is described in both RFC3264 and RFC4566, which is a bit=20
confusing.=A0 But the normative text for ptime is in RFC3264, I think that
 should be referenced in 7.1. (maxptime and minptime only appear in RFC4566=
).<br>
<br>
ptime, maxptime, and minptime are media attributes, so they apply to all
 codecs on the m=3D line.=A0 This is not that clear in this text, as it=20
seems to mandate specific values for this OPUS.=A0 For instance:<br>
<br>
&gt;&gt;&gt; maxptime<br>
<pre><font style=3D"font-family: arial,helvetica,sans-serif;" size=3D"2">Po=
ssible values are 3, 5,
      10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
      rounded up to the next full integer value up to a maximum value of
      120 as defined in Section 4 and Section 5 of this document.  If no
      value is specified, 120 is assumed as default.<br>&gt;&gt;&gt;</font>=
</pre>
The signaled maxptime certainly might exceed 120 ms, since the offerer migh=
t be quite willing to receive some other codec on the m-line at that interv=
al.<br><br>Regards,<br>Stephen Botzko<br><br><div class=3D"gmail_quote">
On Mon, Jul 11, 2011 at 4:23 PM, Randell Jesup <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"im">On 7/11/2011 10:24 AM, 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">
11. In section 7.1 I noticed that ptime, maxptime and minptime signaled by<=
br>
the receiver are just recommendations and not the desired value as in RFC<b=
r>
3264, and the sender can send any size it wishes upto 120. So what is the<b=
r>
benfit for the receiver to signal it. I thought that this is a receive<br>
capability.<br>
</blockquote>
<br></div>
ptime is I think desired but not required. =A0maxptime and minptime should =
be hard receiver limits.<br>
Per Colin on AVT on 30 Mar 2006, ptime is a &quot;hint&quot;, but the encod=
er can vary up to maxptime (and I<br>
assume down to minptime). =A0(I didn&#39;t bother researching further).<br>=
<font color=3D"#888888">
<br>
-- <br>
Randell Jesup<br>
<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@je=
sup.org</a></font><div><div></div><div class=3D"h5"><br>
<br>
______________________________<u></u>_________________<br>
payload mailing list<br>
<a href=3D"mailto:payload@ietf.org" target=3D"_blank">payload@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/payload" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/payload</a><br>
</div></div></blockquote></div><br>

--bcaec5486088ea92f304a7d16c11--

From internet-drafts@ietf.org  Mon Jul 11 14:24:16 2011
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Subject: [payload] I-D Action: draft-ietf-payload-rtp-howto-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Payloads Workin=
g Group of the IETF.

	Title           : How to Write an RTP Payload Format
	Author(s)       : Magnus Westerlund
	Filename        : draft-ietf-payload-rtp-howto-01.txt
	Pages           : 57
	Date            : 2011-07-11

   This document contains information on how to best write an RTP
   payload format.  It provides reading tips, design practices, and
   practical tips on how to produce an RTP payload format specification
   quickly and with good results.  A template is also included with
   instructions that can be used when writing an RTP payload format.


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

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-payload-rtp-howto-01.txt

From magnus.westerlund@ericsson.com  Mon Jul 11 14:27:56 2011
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Subject: [payload] Issues in RTP Payload HOWTO (draft-ietf-payload-rtp-howto-01)
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Hi,

In the process of updating the I-D to draft-ietf-payload-rtp-howto-01 I
found some issues that I think should be discusses by the WG.

These issues have not be changed yet from prior version of the draft.

1. Section 4.1.3 in the I-D discuss the WG draft names of the RTP
payload formats. It recommends a form that is
draft-ietf-wg-rtp-<desriptive-name>-<version>. Which made more sense in
the old AVT when there was more need to separate payload formats from
other drafts. In the payload WG I think we can remove the recommendation
for "-rtp-" in the file name. Any one against?

2. Section 7.4 and the IANA template indicates that RTP payload format
Media types shall be registered also in the RTP Payload Format media
types found on page: http://www.iana.org/assignments/rtp-parameters
My question to the WG. Should we continue to do this?

The registry is not needed for collision prevention. Its sole purpose is
as a quick way of finding all the RTP payload format media types.

If we think the later is good enough then the WG chairs needs to work to
update that registry to be correct and complete as it currently are
missing some format. They also need to ensure that this bit of procedure
really is followed in all the payload formats going forward. The
alternative I see it to declare the registry as discontinued and add a
note that it will not any longer be maintained and that people have to
look in the main registry.

3. The draft has an open issue that says:
Should any procedure for the future when the Payload WG is closed be
described?

I think we should remove this open issue and continue without specifying
that procedure. The reason is that we can't really predict the situation
when there no longer are a WG that is chartered for RTP payload formats
development. Several possible reasons for that can occur, and based on
the situation the procedures to handle that needs to be developed at
that time. Any one having a different view?

Cheers

Magnus Westerlund

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


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Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-howto-01.txt
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Hi,

This updates main change is a media scalability section that has been
contributed by Thomas Schierl, then banged up a bit by me.

The rest is an editing pass and an attempt to resolve some open issues.
See other email for issues requiring discussion.

Diff:
http://tools.ietf.org/rfcdiff?url2=draft-ietf-payload-rtp-howto-01

I would very much appreciate some feedback so we can conclude this I-D
and request publication of it.

Cheers

Magnus

On 2011-07-11 23:24, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Audio/Video Transport Payloads Working Group of the IETF.
> 
> 	Title           : How to Write an RTP Payload Format
> 	Author(s)       : Magnus Westerlund
> 	Filename        : draft-ietf-payload-rtp-howto-01.txt
> 	Pages           : 57
> 	Date            : 2011-07-11
> 
>    This document contains information on how to best write an RTP
>    payload format.  It provides reading tips, design practices, and
>    practical tips on how to produce an RTP payload format specification
>    quickly and with good results.  A template is also included with
>    instructions that can be used when writing an RTP payload format.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-payload-rtp-howto-01.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-payload-rtp-howto-01.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From randell-ietf@jesup.org  Mon Jul 11 14:58:14 2011
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Subject: Re: [payload] RTP Payload Format and File Storage Format for Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On 7/11/2011 4:57 PM, Stephen Botzko wrote:
> ptime is described in both RFC3264 and RFC4566, which is a bit 
> confusing.  But the normative text for ptime is in RFC3264, I think 
> that should be referenced in 7.1. (maxptime and minptime only appear 
> in RFC4566).
>
> ptime, maxptime, and minptime are media attributes, so they apply to 
> all codecs on the m= line.  This is not that clear in this text, as it 
> seems to mandate specific values for this OPUS.  For instance:
>
> >>> maxptime
> Possible values are 3, 5,
>        10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
>        rounded up to the next full integer value up to a maximum value of
>        120 as defined in Section 4 and Section 5 of this document.  If no
>        value is specified, 120 is assumed as default.
> >>>
> The signaled maxptime certainly might exceed 120 ms, since the offerer 
> might be quite willing to receive some other codec on the m-line at 
> that interval.

Right.  You could define a fmtp parameter of maxptime (or equivalent) 
that give the maximum for that
codec/payload.  iLBC uses fmtp to negotiate between 20 and 30ms frame 
sizes, though that's not
exactly the same as the ptime issues here.

ptime and maxptime are a bit problematic due to the global nature (and 
that's part of why ptime is a hint).
Also, they can be any arbitrary value (due to the shared nature), not 
just multiples of your codec's frame size.

-- 
Randell Jesup
randell-ietf@jesup.org


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From: Koen Vos <koen.vos@skype.net>
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Subject: Re: [payload] RTP Payload Format and File Storage Format for	Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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Randell Jesup wrote:
> ptime is I think desired but not required.  maxptime and minptime should 
> be hard receiver limits.

The goal has always been that every Opus decoder is able to decode any valid Opus bitstream.  Advantages are:
- In a multi-party call you can send the same bitstream to all end points.  If maxptime and minptime were hard limits, then two end points may not have any overlap in their ranges of allowed ptime, making it impossible to share bitstreams.
- Less reliance on parameter negotiation.  Keep in mind that bitstreams may be saved to file and played out later.

Note that the cost in storage requirements of maxptime being a soft limit is modest: payloads are stored in compressed format, and payloads with ptime > 20 ms can always be decoded in chunks of 20 ms (or less).

If Opus' use of minptime and maxptime differs from what is generally accepted, then perhaps we should use different names for these parameters?

Roni Even wrote:
> So what is the benfit for the receiver to signal it.

See section 7.2.1: performance (e.g. quality) may be better for some ptime values than for others.

best,
koen.


----- Original Message -----
From: "Randell Jesup" <randell-ietf@jesup.org>
To: payload@ietf.org
Sent: Monday, July 11, 2011 1:23:41 PM
Subject: Re: [payload] RTP Payload Format and File Storage Format for	Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00

On 7/11/2011 10:24 AM, Roni Even wrote:
> 11. In section 7.1 I noticed that ptime, maxptime and minptime signaled by
> the receiver are just recommendations and not the desired value as in RFC
> 3264, and the sender can send any size it wishes upto 120. So what is the
> benfit for the receiver to signal it. I thought that this is a receive
> capability.

ptime is I think desired but not required.  maxptime and minptime should 
be hard receiver limits.
Per Colin on AVT on 30 Mar 2006, ptime is a "hint", but the encoder can 
vary up to maxptime (and I
assume down to minptime).  (I didn't bother researching further).

-- 
Randell Jesup
randell-ietf@jesup.org

_______________________________________________
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From internet-drafts@ietf.org  Mon Jul 11 15:26:47 2011
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Subject: [payload] I-D Action: draft-ietf-payload-vp8-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Audio/Video Transport Payloads Workin=
g Group of the IETF.

	Title           : RTP Payload Format for VP8 Video
	Author(s)       : Patrik Westin
                          Henrik F Lundin
                          Michael Glover
                          Justin Uberti
                          Frank Galligan
	Filename        : draft-ietf-payload-vp8-01.txt
	Pages           : 27
	Date            : 2011-07-11

   This memo describes an RTP payload format for the VP8 video codec.
   The payload format has wide applicability, as it supports
   applications from low bit-rate peer-to-peer usage, to high bit-rate
   video conferences.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-payload-vp8-01.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-payload-vp8-01.txt

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> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
> Sent: Monday, July 11, 2011 2:27 PM
> To: payload@ietf.org
> Subject: [payload] Issues in RTP Payload =
HOWTO(draft-ietf-payload-rtp-howto-01)
>=20
> Hi,
>=20
> In the process of updating the I-D to draft-ietf-payload-rtp-howto-01 =
I
> found some issues that I think should be discusses by the WG.
>=20
> These issues have not be changed yet from prior version of the draft.
>=20
> 1. Section 4.1.3 in the I-D discuss the WG draft names of the RTP
> payload formats. It recommends a form that is
> draft-ietf-wg-rtp-<desriptive-name>-<version>. Which made more sense =
in
> the old AVT when there was more need to separate payload formats from
> other drafts. In the payload WG I think we can remove the =
recommendation
> for "-rtp-" in the file name. Any one against?

I guess we can drop it as "wg" will be payload already.
=20
> 2. Section 7.4 and the IANA template indicates that RTP payload format
> Media types shall be registered also in the RTP Payload Format media
> types found on page: http://www.iana.org/assignments/rtp-parameters
> My question to the WG. Should we continue to do this?

Why should not we continue?
=20
> The registry is not needed for collision prevention. Its sole purpose =
is
> as a quick way of finding all the RTP payload format media types.

I personally find it useful.
=20
> If we think the later is good enough then the WG chairs needs to work =
to
> update that registry to be correct and complete as it currently are
> missing some format. They also need to ensure that this bit of =
procedure
> really is followed in all the payload formats going forward. The
> alternative I see it to declare the registry as discontinued and add a
> note that it will not any longer be maintained and that people have to
> look in the main registry.

Which registry are you referring to?
=20
> 3. The draft has an open issue that says:
> Should any procedure for the future when the Payload WG is closed be
> described?
>=20
> I think we should remove this open issue and continue without =
specifying
> that procedure. The reason is that we can't really predict the =
situation
> when there no longer are a WG that is chartered for RTP payload =
formats
> development. Several possible reasons for that can occur, and based on
> the situation the procedures to handle that needs to be developed at
> that time. Any one having a different view?

Agreed. If payload shuts down, then I guess RAI will take care of it.
=20
-acbegen (individual)

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

From csp@csperkins.org  Tue Jul 12 07:56:29 2011
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To: julian.spittka@skype.net
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Cc: 'Jean-Marc Valin' <jmvalin@jmvalin.ca>, payload@ietf.org
Subject: Re: [payload] RTP Payload Format and File Storage Format for Opus Speech and Audio Codec - draft-spittka-payload-rtp-opus-00
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On 5 Jul 2011, at 01:10, Julian Spittka wrote:
> Please find a first draft "draft-spittka-payload-rtp-opus-00" for the =
"RTP Payload Format and File Storage Format for Opus Speech and Audio =
Codec" uploaded at =
https://datatracker.ietf.org/doc/draft-spittka-payload-rtp-opus/.=20
>=20
> Any feedback, questions, or comments are highly welcome.


A few quick comments:

- The codec supports a VBR mode, so it would be useful to reference the =
discussion in draft-ietf-avtcore-srtp-vbr-audio in the Security =
Considerations

- Section 4.1 states that the Marker bit in the RTP header is not used. =
Any reason why the standard audio usage of setting the M bit to one on =
the first packet on a talk-spurt isn't appropriate?=20

- The media type registration defines a "stereo" parameter. Rather than =
define something codec-specific, can't this use the encoding parameter =
in the SDP "a=3Drtpmap:" line, line other stereo codecs?

Cheers,
Colin

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




From magnus.westerlund@ericsson.com  Wed Jul 13 01:26:51 2011
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Subject: Re: [payload] Issues in RTP Payload HOWTO(draft-ietf-payload-rtp-howto-01)
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On 2011-07-12 00:39, Ali C. Begen (abegen) wrote:
>  
>> 2. Section 7.4 and the IANA template indicates that RTP payload format
>> Media types shall be registered also in the RTP Payload Format media
>> types found on page: http://www.iana.org/assignments/rtp-parameters
>> My question to the WG. Should we continue to do this?
> 
> Why should not we continue?

The main reason is to ensure that this registration actually happens.
There is formats missing from it.

>  
>> The registry is not needed for collision prevention. Its sole purpose is
>> as a quick way of finding all the RTP payload format media types.
> 
> I personally find it useful.

Yes, I also find it a useful concept. But as it is incomplete I can't
rely on it.

>  
>> If we think the later is good enough then the WG chairs needs to work to
>> update that registry to be correct and complete as it currently are
>> missing some format. They also need to ensure that this bit of procedure
>> really is followed in all the payload formats going forward. The
>> alternative I see it to declare the registry as discontinued and add a
>> note that it will not any longer be maintained and that people have to
>> look in the main registry.
> 
> Which registry are you referring to?

"RTP Payload Format media types" found on page:
http://www.iana.org/assignments/rtp-parameter

Someone needs to ensure that this registry actually contains all the RTP
payload types.

Secondly, which definitely will be the WG chairs job, is to ensure that
the IANA sections do request that any future registrations actually do
this.

If we look at the current set of PAYLOAD WG documents the status of this
registration request is as follows when it comes to request of IANA to
actually add or modify the "RTP Payload Format media types"

draft-ietf-avt-rtp-evrc-nw-03: Don't request registration
draft-ietf-avt-rtp-ipmr-15.txt: Don't request registration
draft-ietf-payload-rfc3016bis-01.txt: Doesn't update the registration
draft-ietf-payload-rfc3189bis-01: Doesn't update the registration
draft-ietf-payload-rtp-g718-00: Don't request registration
draft-ietf-payload-rtp-klv-01: Requests Registration!!!
draft-ietf-payload-rtp-mvc-00: Don't request registration
draft-ietf-payload-rtp-sbc-00: Don't request registration
draft-ietf-payload-vp8-01: Don't request registration

In other words, only one out of nine drafts actually ensures that their
drafts get added to this registry.

That is why I raised this question, if we should continue to use it or
declare it dead.

Cheers

Magnus Westerlund

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


From magnus.westerlund@ericsson.com  Wed Jul 13 04:42:30 2011
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Subject: Re: [payload] Issues in RTP Payload HOWTO(draft-ietf-payload-rtp-howto-01)
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To clarify as I have received one email pointing out that draft x do
update this registry despite what I wrote.

But, that is misunderstanding. There are two regestries.

The Media Types registry:
http://www.iana.org/assignments/media-types/index.html

This is the registry that ensure that there are no colliding media types
etc.

The registry I am discussing is the
"RTP Payload Format media types"
http://www.iana.org/assignments/rtp-parameters

This is a courtesy registry to track which of the media types that exist
in the previous registry that are defined for RTP usage.

I hope that helps setting what I want discussed.

Cheers

Magnus

On 2011-07-13 10:26, Magnus Westerlund wrote:
> On 2011-07-12 00:39, Ali C. Begen (abegen) wrote:
>>  
>>> 2. Section 7.4 and the IANA template indicates that RTP payload format
>>> Media types shall be registered also in the RTP Payload Format media
>>> types found on page: http://www.iana.org/assignments/rtp-parameters
>>> My question to the WG. Should we continue to do this?
>>
>> Why should not we continue?
> 
> The main reason is to ensure that this registration actually happens.
> There is formats missing from it.
> 
>>  
>>> The registry is not needed for collision prevention. Its sole purpose is
>>> as a quick way of finding all the RTP payload format media types.
>>
>> I personally find it useful.
> 
> Yes, I also find it a useful concept. But as it is incomplete I can't
> rely on it.
> 
>>  
>>> If we think the later is good enough then the WG chairs needs to work to
>>> update that registry to be correct and complete as it currently are
>>> missing some format. They also need to ensure that this bit of procedure
>>> really is followed in all the payload formats going forward. The
>>> alternative I see it to declare the registry as discontinued and add a
>>> note that it will not any longer be maintained and that people have to
>>> look in the main registry.
>>
>> Which registry are you referring to?
> 
> "RTP Payload Format media types" found on page:
> http://www.iana.org/assignments/rtp-parameter
> 
> Someone needs to ensure that this registry actually contains all the RTP
> payload types.
> 
> Secondly, which definitely will be the WG chairs job, is to ensure that
> the IANA sections do request that any future registrations actually do
> this.
> 
> If we look at the current set of PAYLOAD WG documents the status of this
> registration request is as follows when it comes to request of IANA to
> actually add or modify the "RTP Payload Format media types"
> 
> draft-ietf-avt-rtp-evrc-nw-03: Don't request registration
> draft-ietf-avt-rtp-ipmr-15.txt: Don't request registration
> draft-ietf-payload-rfc3016bis-01.txt: Doesn't update the registration
> draft-ietf-payload-rfc3189bis-01: Doesn't update the registration
> draft-ietf-payload-rtp-g718-00: Don't request registration
> draft-ietf-payload-rtp-klv-01: Requests Registration!!!
> draft-ietf-payload-rtp-mvc-00: Don't request registration
> draft-ietf-payload-rtp-sbc-00: Don't request registration
> draft-ietf-payload-vp8-01: Don't request registration
> 
> In other words, only one out of nine drafts actually ensures that their
> drafts get added to this registry.
> 
> That is why I raised this question, if we should continue to use it or
> declare it dead.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
> 


-- 

Magnus Westerlund

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


From dpetrie@sipez.com  Mon Jul 18 17:52:05 2011
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From: Daniel Petrie <dpetrie@sipez.com>
To: payload@ietf.org
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Subject: [payload] AAC_LC mime type and RTP/SDP binding
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Hi:
Sorry for the broadcast on this.

I am looking for the RFC for RTP streaming of audio/AAC_LC.

Normally I can pretty easily find the codec profile with RTP and SDP binding for a mime type.  However I am having difficulty finding the RFC or draft for this.  Iana has no registration for the audio/AAC_LC MIME type, yet I know of several vendors who are advertising this MIME type in their SDP.  I have scanned through RFC 3016 & 3640, but less I am being dense, these do not define it either.

I would appreciate a pointer.

Cheers,
Dan


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From: "Bont, Frans de" <frans.de.bont@philips.com>
To: "dpetrie@sipez.com" <dpetrie@sipez.com>, "payload@ietf.org" <payload@ietf.org>
Date: Tue, 19 Jul 2011 09:10:31 +0200
Thread-Topic: [payload] AAC_LC mime type and RTP/SDP binding
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Subject: Re: [payload] AAC_LC mime type and RTP/SDP binding
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Hi Dan,

RFC 3016 & 3640 define two different payload formats for MPEG-4 AAC LC.

RFC 3016 defines the MIME subtype audio/MP4A-LATM, and support all MPEG-4 A=
udio coders, so also MPEG-4 AAC LC. The payload format is based on MPEG-4 L=
ATM (Low-overhead MPEG-4 Audio Transport Multiplex), a format which is also=
 used by DVB services using MPEG-2 TS.
An example of deployment of RFC 3016 is the 3GPP Transparent end-to-end Pac=
ket-switched Streaming Service (PSS) (3GPP TS 26.234).

RFC 3640 defines the MIME subtype mpeg4-generic, which has defined audio mo=
des for MPEG-4 AAC and MPEG-4 CELP. The AAC modes are AAC-lbr and AAC-hbr a=
nd the payload carries elementary AAC frames. RFC 3640 is referenced in ETS=
I TS 102 005 (DVB services delivered directly over IP protocols) and the AT=
SC Mobile DTV Standard (ATSC A/153 Part 8).

I am not aware of a audio/AAC_LC MIME type.

Best regards,
Frans

> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
> Behalf Of Daniel Petrie
> Sent: Tuesday 19 July 2011 2:52
> To: payload@ietf.org
> Subject: [payload] AAC_LC mime type and RTP/SDP binding
>
> Hi:
> Sorry for the broadcast on this.
>
> I am looking for the RFC for RTP streaming of audio/AAC_LC.
>
> Normally I can pretty easily find the codec profile with RTP and SDP
> binding for a mime type.  However I am having difficulty finding the
> RFC or draft for this.  Iana has no registration for the audio/AAC_LC
> MIME type, yet I know of several vendors who are advertising this MIME
> type in their SDP.  I have scanned through RFC 3016 & 3640, but less I
> am being dense, these do not define it either.
>
> I would appreciate a pointer.
>
> Cheers,
> Dan
>
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload

The information contained in this message may be confidential and legally p=
rotected under applicable law. The message is intended solely for the addre=
ssee(s). If you are not the intended recipient, you are hereby notified tha=
t any use, forwarding, dissemination, or reproduction of this message is st=
rictly prohibited and may be unlawful. If you are not the intended recipien=
t, please contact the sender by return e-mail and destroy all copies of the=
 original message.


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From: Daniel Petrie <dpetrie@sipez.com>
To: "payload@ietf.org" <payload@ietf.org>, Frans deBont <frans.de.bont@philips.com>
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Subject: Re: [payload] AAC_LC mime type and RTP/SDP binding
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Hi Frans:=0AThank you for the response and references to AAC LC.  I am defi=
nitely seeing mime type audio/AAC_LC from several vendors.  See the followi=
ng rtpmap from the SDP:=0A=0Aa=3Drtpmap:116 AAC_LC/32000=0A=0ACan anyone pr=
ovide a pointer to a spec. which defines what AAC profile and parameters ar=
e implied by this?=0A=0ACheers,=0ADan=0A=0A--- On Tue, 7/19/11, Bont, Frans=
 de <frans.de.bont@philips.com> wrote:=0A=0A> From: Bont, Frans de <frans.d=
e.bont@philips.com>=0A> Subject: RE: [payload] AAC_LC mime type and RTP/SDP=
 binding=0A> To: "dpetrie@sipez.com" <dpetrie@sipez.com>, "payload@ietf.org=
" <payload@ietf.org>=0A> Date: Tuesday, July 19, 2011, 3:10 AM=0A> Hi Dan,=
=0A> =0A> RFC 3016 & 3640 define two different payload formats=0A> for MPEG=
-4 AAC LC.=0A> =0A> RFC 3016 defines the MIME subtype audio/MP4A-LATM, and=
=0A> support all MPEG-4 Audio coders, so also MPEG-4 AAC LC. The=0A> payloa=
d format is based on MPEG-4 LATM (Low-overhead MPEG-4=0A> Audio Transport M=
ultiplex), a format which is also used by=0A> DVB services using MPEG-2 TS.=
=0A> An example of deployment of RFC 3016 is the 3GPP=0A> Transparent end-t=
o-end Packet-switched Streaming Service=0A> (PSS) (3GPP TS 26.234).=0A> =0A=
> RFC 3640 defines the MIME subtype mpeg4-generic, which has=0A> defined au=
dio modes for MPEG-4 AAC and MPEG-4 CELP. The AAC=0A> modes are AAC-lbr and=
 AAC-hbr and the payload carries=0A> elementary AAC frames. RFC 3640 is ref=
erenced in ETSI TS 102=0A> 005 (DVB services delivered directly over IP pro=
tocols) and=0A> the ATSC Mobile DTV Standard (ATSC A/153 Part 8).=0A> =0A> =
I am not aware of a audio/AAC_LC MIME type.=0A> =0A> Best regards,=0A> Fran=
s=0A> =0A> > -----Original Message-----=0A> > From: payload-bounces@ietf.or=
g=0A> [mailto:payload-bounces@ietf.org]=0A> On=0A> > Behalf Of Daniel Petri=
e=0A> > Sent: Tuesday 19 July 2011 2:52=0A> > To: payload@ietf.org=0A> > Su=
bject: [payload] AAC_LC mime type and RTP/SDP=0A> binding=0A> >=0A> > Hi:=
=0A> > Sorry for the broadcast on this.=0A> >=0A> > I am looking for the RF=
C for RTP streaming of=0A> audio/AAC_LC.=0A> >=0A> > Normally I can pretty =
easily find the codec profile=0A> with RTP and SDP=0A> > binding for a mime=
 type.=A0 However I am having=0A> difficulty finding the=0A> > RFC or draft=
 for this.=A0 Iana has no registration=0A> for the audio/AAC_LC=0A> > MIME =
type, yet I know of several vendors who are=0A> advertising this MIME=0A> >=
 type in their SDP.=A0 I have scanned through RFC=0A> 3016 & 3640, but less=
 I=0A> > am being dense, these do not define it either.=0A> >=0A> > I would=
 appreciate a pointer.=0A> >=0A> > Cheers,=0A> > Dan=0A> >=0A> > __________=
_____________________________________=0A> > payload mailing list=0A> > payl=
oad@ietf.org=0A> > https://www.ietf.org/mailman/listinfo/payload=0A> =0A> T=
he information contained in this message may be=0A> confidential and legall=
y protected under applicable law. The=0A> message is intended solely for th=
e addressee(s). If you are=0A> not the intended recipient, you are hereby n=
otified that any=0A> use, forwarding, dissemination, or reproduction of thi=
s=0A> message is strictly prohibited and may be unlawful. If you=0A> are no=
t the intended recipient, please contact the sender by=0A> return e-mail an=
d destroy all copies of the original=0A> message.=0A> =0A> 

From kaz.mishima@gmail.com  Wed Jul 20 01:03:04 2011
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Cc: draft-ietf-payload-rfc3189bis.all@tools.ietf.org, ietf-types@iana.org, payload@ietf.org
Subject: Re: [payload] Updating the registration of audio and video DV media subtype
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Hi

Thanks for your comments.
I'll update our draft with working group's comments.

Best regards,
Kazuhiro


2011/7/12 Roni Even <Even.roni@huawei.com>:
> Hi,
>
>
>
> draft-ietf-payload-rfc3189bis-01 has passed Working Group Last Call in th=
e
> Payload Working Group.
>
>
>
> Note that it an update to existing RFC3189 registrations
> http://www.iana.org/assignments/rtp-parameters for DV media subtype.
>
>
>
> The new registrations are in Section 3.1 of the document.
> http://tools.ietf.org/html/draft-ietf-payload-rfc3189bis-01#section-3.1
>
> nbsp;
>
> The information is also available below.
>
>
>
> I noticed that there is a need to change the change controller from " IET=
F
> Audio/Video Transport working group delegated from the=A0=A0 IESG" =A0to =
" IETF
> Payload =A0working group delegated from the=A0=A0 IESG"
>
>
>
>
>
> Other comments on the registration are welcome.
>
>
>
> Roni Even
>
> Payload co-chair
>
>
>
>
>
> 3.1.1.=A0 Media Type Registration for DV Video
>
>
>
>
>
> Type name:=A0 video
>
>
>
> =A0=A0 Subtype name:=A0 DV
>
>
>
>
>
> =A0=A0=A0=A0=A0 encode:=A0 type of DV format.=A0 Permissible values for e=
ncode are
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SD-VCR/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SD-VCR/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 HD-VCR/1125-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 HD-VCR/1250-50,
>
> =A0=A0&nbsp ;=A0& DL-VCR/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SDL-VCR/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-25/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-25/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-50/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-50/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/1080-60i,
>
> &nbsp ;=A0& bsp;=A0 370M/1080-50i,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/720-60p,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/720-50p,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 306M/525-60 (for backward compatibility),
>
> =A0=A0=A0=A0=A0=A0=A0=A0 and 306M/625-50 (for backward compatibility).
>
>
>
> =A0=A0 Optional parameters:
>
>
>
> =A0=A0=A0=A0=A0 audio:=A0 whether the DV stream includes audio data or no=
t.
>
> =A0=A0=A0=A0=A0=A0=A0=A0 Permissible values for audio are bundled and non=
e.=A0 Defaults to
> none.
>
>
>
> =A0=A0 Encoding considerations:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 DV video can be transmitted with RTP as specifie=
d in RFCXXXX(This
> document).=A0 Other transport methods are not specified.
>
> =A0=A0=A0 Security considerations:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 See Section 4 of RFCXXXX (This document).
>
>
>
> =A0=A0 Interoperability considerations:=A0 NONE
>
>
>
> =A0=A0 Public specification:
>
>
>
> =A0=A0=A0=A0=A0&nbs p;=A0 FR style=3D'font-family:"Courier New"'>IEC 6183=
4 Standard
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SMPTE 314M
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SMPTE 370M
>
> =A0=A0=A0=A0=A0=A0=A0=A0 RFCXXXX (This document)
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SMPTE 306M (for backward compatibility).
>
>
>
> =A0=A0 Applications that use this media type:=A0 Audio and video s treami=
ng o:p>
>
>
>
> =A0=A0 Additional information:=A0 NONE
>
>
>
> =A0=A0 Person & email address to contact for further information:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 Katsushi Kobayashi
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 e-mail: ikob@ni.aist.go.jp
>
>
>
> =A0=A0 Intended usage:=A0 COMMON
>
>
>
> =A0=A0 Restrictions on usage:=A0 This media type depends on RTP framing, =
and hence
> is only defined for transfer via RTP (RFC 3550).=A0 Transfer within other
> framing protocols is not defined at this time.
>
>
>
> =A0=A0 Author:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 Katsushi Kobayashi
>
> < n style=3D'font-family:"Courier New"'>
>
> =A0=A0 Change controller:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 IETF Audio/Video Transport working group delegat=
ed from the
>
> =A0=A0=A0=A0=A0=A0=A0=A0 IESG
>
>
>
>
>
> 3.1.2.=A0 Media Type Registration for DV Audio
>
> =A0 < =A0=A0 Type name:=A0 audio
>
>
>
> =A0=A0 Subtype name:=A0 DV
>
>
>
> =A0=A0 Required parameters:
>
>
>
> =A0=A0=A0=A0=A0 encode:=A0 type of DV format.=A0 Permissible values for e=
ncode are
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SD-VCR/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 HD-VCR/1125-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 HD-VCR/1250-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SDL-VCR/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SDL-VCR/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-25/525-60,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-25/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 314M-50/625-50,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/1080-60i,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/1080-50i,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/720-60p,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 370M/720-50p,
>
> =A0=A0=A0=A0=A0=A0=A0=A0 306M/525-60 (for backward compatibility),
>
> =A0=A0=A0=A0=A0=A0=A0=A0 and 306M/625-50 (for backward compatibility).
>
>
>
> =A0=A0 Optional parameters:
>
>
>
> =A0=A0=A0=A0=A0 audio:=A0 whether the DV stream includes audio data or no=
t.
>
> =A0=A0=A0=A0=A0=A0=A0=A0 Permissible values for audio are bundled and non=
e.=A0 Defaults to
> none.
>
>
>
> =A0=A0 Encodin g consid n>
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 DV audio can be transmitted with RTP as specifie=
d in RFCXXXX(This
> document).=A0 Other transport methods are not specified.
>
>
>
> =A0=A0 Security considerations:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 See Section 4 of RFCXXXX (This document).
>
>
>
> =A0&nb sp; Inte ns:=A0 NONE
>
>
>
> =A0=A0 Published specification:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 IEC 61834 Standard
>
> =A0=A0=A0=A0 =A0=A0=A0=A0SMPTE 314M
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SMPTE 370M
>
> =A0=A0=A0=A0=A0=A0=A0=A0 RFCXXXX (This document)< /p>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 SMPTE 306M (for backward compatibility).
>
>
>
> =A0=A0 Applications that use this media type:=A0 Audio and video streamin=
g and
> conferencing tools.
>
>
>
> =A0=A0 Additional information:=A0 NONE
>
>
>
> =A0=A0 Person & email address to contact for further information:
>
> =A0=A0=A0=A0=A0=A0=A0=A0 Katsushi Kobayashi
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 e-mail: ikob@ni.aist.go.jp
>
>
>
> =A0=A0 Intended usage:=A0 COMMON
>
>
>
> =A0=A0 Restrictions on usage:=A0 This media type depends on RTP framing, =
and hence
> is only defined for transfer via RTP (RFC 3550).=A0 Transfer=A0=A0=A0=A0=
=A0 within other
> framing protocol time.
>
>
>
>
>
> =A0=A0 Author:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 Katsushi Kobayashi
>
>
>
> =A0=A0 Change controller:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 IETF Audio/Video Transport worki ng group /o:p>
>
> =A0=A0=A0=A0=A0=A0=A0=A0 IESG
>
>
>
>

From Even.roni@huawei.com  Thu Jul 21 06:23:52 2011
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Date: Thu, 21 Jul 2011 16:21:01 +0300
From: Roni Even <Even.roni@huawei.com>
To: avt@ietf.org, rtcweb@ietf.org
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Cc: rtcweb-chairs@tools.ietf.org, payload@ietf.org, payload-chairs@tools.ietf.org
Subject: [payload] Updated AVTcore agenda - new slot for multiplexing RTP for broweser based RTC
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Hi,

We apologize for the last minute change. 

The topic of "Multiplexing of Real-Time Transport Protocol (RTP) Traffic for
Browser based Real-Time Communications (RTC)" proposed in
draft-rosenberg-rtcweb-rtpmux-00 caused a long discussion in RTCweb mailing
list and the feeling is that it should be also discussed in AVTcore.  Magnus
has started a thread on the AVTcore list.
In order to have a time slot we moved the payload WG status presentation to
the AVText slot which will be on Wednesday afternoon. 
We had to remove the mprtp presentation based on low discussion on the
mailing list. We may have one slide on mprtp during the chair status
 
Please review the above draft, I recommend also to look at
http://tools.ietf.org/html/draft-perkins-rtcweb-rtp-usage-02 
 
Thanks
Roni Even

 

 


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vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>We apologize =
for the last minute change. <o:p></o:p></p><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>The topic =
of &quot;Multiplexing of Real-Time Transport Protocol (RTP) Traffic for =
Browser based Real-Time Communications (RTC)&quot; proposed in =
&nbsp;&nbsp;draft-rosenberg-rtcweb-rtpmux-00 caused a long discussion in =
RTCweb mailing list and the feeling is that it should be also discussed =
in AVTcore. &nbsp;Magnus has started a thread on the AVTcore =
list.<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>In order =
to have a time slot we moved the payload WG status presentation to the =
AVText slot which will be on Wednesday afternoon. =
<o:p></o:p></span></pre><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>We had to =
remove the mprtp presentation based on low discussion on the mailing =
list. We may have one slide on mprtp during the chair =
status<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 =
review the above draft, I recommend also to look at <a =
href=3D"http://tools.ietf.org/html/draft-perkins-rtcweb-rtp-usage-02">htt=
p://tools.ietf.org/html/draft-perkins-rtcweb-rtp-usage-02</a> =
<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"'>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><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>=

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Audio/Video Transport Core Maintenance  (AVTCore) Working Group
===============================================================

CHAIRS:  Magnus Westerlund       <magnus.westerlund@ericsson.com>
         Roni Even         <even.roni@huawei.com>

AGENDA

Wednesday, 27 July 2011 at 9:00-11:30 (208 AB)
----------------------------------------------
09:00   AVTCore Status Update                               (Chairs, 20)

09:20   Encryption of Header Extensions in SRTP             (Lennox, 15)
        draft-ietf-avtcore-srtp-encrypted-header-ext-00

09:35   RTCP Extension for Feedback Suppression Indication  (Qin Wu, 15)
        draft-ietf-avtcore-feedback-supression-rtp-05

09:50   Monitoring Architecture for RTP                     (Qin Wu, 15)
        draft-ietf-avtcore-monarch-03

10:05   RTP Multiple Stream Sessions and Simulcast(Magnus Westerlund , 30)
        draft-westerlund-avtcore-multistream-and-simulcast-00

10:35   Delayed Duplication Attribute and Duplication Grouping in SDP(Begen, 15)
        draft-begen-mmusic-temporal-interleaving-02
        draft-begen-mmusic-redundancy-grouping-01

10:50   Retransmission for SSM Sessions                      (Begen, 10)
        draft-vancaenegem-avtcore-retransmission-for-ssm-00

11:00   Multiplexing of RTP trafic for Browser based RTC (Rosenberg, 30)
        draft-rosenberg-rtcweb-rtpmux-00

11:30   End

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<html>
<head>
<title>Audio/Video Transport Core Maintenance  (AVTCore) Working Group Agenda</title>
</head>
<body bgcolor="#ffffff"><h1>Audio/Video Transport Core Maintenance  (AVTCore) Working Group Agenda</h1>
<hr>
<p>
<h3>
Chairs: Magnus Westerlund       <magnus.westerlund@ericsson.com>, Roni Even         <even.roni@huawei.com>
</h3>
<p>
<h2>Wednesday, 27 July 2011 at 9:00-11:30 (208 AB)</h2>
<p>
<table border="0" cellpadding="5">
<tr>
<th align="left">09:00
<th align="left">AVTCore Status Update<th align="left">Chairs
<tr>
<th align="left">09:20
<th align="left">Encryption of Header Extensions in SRTP<th align="left">Lennox
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-ietf-avtcore-srtp-encrypted-header-ext-00">draft-ietf-avtcore-srtp-encrypted-header-ext-00</a>
<tr>
<th align="left">09:35
<th align="left">RTCP Extension for Feedback Suppression Indication<th align="left">Qin Wu
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-ietf-avtcore-feedback-supression-rtp-05">draft-ietf-avtcore-feedback-supression-rtp-05</a>
<tr>
<th align="left">09:50
<th align="left">Monitoring Architecture for RTP<th align="left">Qin Wu
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-ietf-avtcore-monarch-03">draft-ietf-avtcore-monarch-03</a>
<tr>
<th align="left">10:05
<th align="left">RTP Multiple Stream Sessions and Simulcast<th align="left">Magnus Westerlund 
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-westerlund-avtcore-multistream-and-simulcast-00">draft-westerlund-avtcore-multistream-and-simulcast-00</a>
<tr>
<th align="left">10:35
<th align="left">Delayed Duplication Attribute and Duplication Grouping in SDP<th align="left">Begen
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-begen-mmusic-temporal-interleaving-02">draft-begen-mmusic-temporal-interleaving-02</a>
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-begen-mmusic-redundancy-grouping-01">draft-begen-mmusic-redundancy-grouping-01</a>
<tr>
<th align="left">10:50
<th align="left">Retransmission for SSM Sessions<th align="left">Begen
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-vancaenegem-avtcore-retransmission-for-ssm-00">draft-vancaenegem-avtcore-retransmission-for-ssm-00</a>
<tr>
<th align="left">11:00
<th align="left">Multiplexing of RTP trafic for Browser based RTC<th align="left">Rosenberg
<tr>
<td align="left">
<td align="left"><a href="http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00">draft-rosenberg-rtcweb-rtpmux-00</a>
<tr>
<th align="left">11:30
<th align="left">End</table>

--Boundary_(ID_JK5DzPDoqh6oPJ3Nh+Btzw)--

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Subject: [payload] Far End Camera Control in SIP
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I must implement FECC over SIP as a part of my work project.
I need more details about  RFC 4573.
would you please introduce me some reference about it?
PLZ help me.
thanks very much.

--001485f87eb4f43a3204a8945854
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I must implement FECC over SIP as a part of my work=20
project.<br>I need more details about=A0 RFC 4573.<br>would you please=20
introduce me some reference about it?<br>PLZ help me.<br>
thanks very much.


--001485f87eb4f43a3204a8945854--

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Date: Thu, 21 Jul 2011 18:18:14 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [payload] Far End Camera Control in SIP
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Hi,

You should look at H.281 http://www.itu.int/rec/T-REC-H.281-199411-I

and H.224 http://www.itu.int/rec/T-REC-H.224-200501-I  with the fix in
http://www.itu.int/rec/T-REC-H.224-200708-I!Cor1 

 

H.224 is the encapsulation and H.281 is FECC

 

ITU-T recommendations are available free of charge

Roni Even

 

From: Mina Salehi [mailto:mina.s67@gmail.com] 
Sent: Thursday, July 21, 2011 4:42 PM
To: Even.roni@huawei.com
Subject: Far End Camera Control in SIP

 

Mr Roni,
I must implement FECC over SIP as a part of my work project.
I need more details about  RFC 4573.
would you please introduce me some reference about it?
PLZ help me.
thanks very much. 


--Boundary_(ID_mkZqgi5E0znng9/pZ5hECQ)
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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<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><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>You should look at H.281 <a =
href=3D"http://www.itu.int/rec/T-REC-H.281-199411-I">http://www.itu.int/r=
ec/T-REC-H.281-199411-I</a><o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>and H.224 <a =
href=3D"http://www.itu.int/rec/T-REC-H.224-200501-I">http://www.itu.int/r=
ec/T-REC-H.224-200501-I</a>&nbsp; with the fix in <a =
href=3D"http://www.itu.int/rec/T-REC-H.224-200708-I!Cor1">http://www.itu.=
int/rec/T-REC-H.224-200708-I!Cor1</a> <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>H.224 is the encapsulation and H.281 is FECC<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>ITU-T recommendations are available free of =
charge<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni Even<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><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"'> =
Mina Salehi [mailto:mina.s67@gmail.com] <br><b>Sent:</b> Thursday, July =
21, 2011 4:42 PM<br><b>To:</b> Even.roni@huawei.com<br><b>Subject:</b> =
Far End Camera Control in SIP<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Mr =
Roni,<br>I must implement FECC over SIP as a part of my work =
project.<br>I need more details about&nbsp; RFC 4573.<br>would you =
please introduce me some reference about it?<br>PLZ help me.<br>thanks =
very much. <o:p></o:p></p></div></div></body></html>=

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From hoene@uni-tuebingen.de  Mon Jul 25 09:10:47 2011
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Subject: [payload] Frames to Packets in draft-ietf-codec-opus-07?
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Hello,

I was wondering why the draft-ietf-codec-opus-07 draft has feature to
concatenate multiple codec frames to packets. At the same time,
draft-spittka-payload-rtp-opus-00 does not support to place multiple =
codec
entities in on RTP packet.

As a consequence, the maximal size of an Opus packet is limited to =
120ms.
Especially, on low rate links this might be too low. Also, on links with =
a
high packet overhead (such as IPsec or IEEE 802.11b WLAN), this limits =
seems
to be too.

I would suggest to either

1) Remove the concatenation from draft-ietf-codec-opus-07 and add it to
draft-spittka-payload-rtp-opus-00.

Or

2) Add a second concatenation feature to the payload spec.

With best regards,

 Christian








-------------------------------------------------------------------------=
---
-----
Dr.-Ing. Christian Hoene, University of T=FCbingen, Computer Science, =
Chair of
Communication Networks, Research Group Interactive Communication Systems
(ICS)
Sand 13, 72076 T=FCbingen, Germany, Tel +49 7071 2970532,
www.net.uni-tuebingen.de




From hoene@uni-tuebingen.de  Mon Jul 25 09:11:18 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: <payload@ietf.org>, <codec@ietf.org>
Date: Mon, 25 Jul 2011 10:33:27 -0400
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Subject: [payload] Padding in draft-ietf-codec-opus-07?
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Hello,

I am wondering about a padding feature in draft-ietf-codec-opus-07. It
allows you to extend the frame size by filling it up with zeros.

 I just spoke with the Tim, Jean-Marc and Gregory about this feature. As =
far
as I understood, this feature is required at transcoding gateways to =
enforce
a stream of packets having a constant size (while using UDP). Otherwise,
attackers could potentially follow the conversation by observing packet
sizes.=20

To my understanding, padding is a feature that can be easily skipped =
from
the codec specification. It violates the layering principle and is =
intended
only for a very specific use cases. On the other side, constant sized =
frames
can be easily supported by keeping the codec parameters fixed.

With best regards,

 Christian Hoene



-------------------------------------------------------------------------=
---
-----
Dr.-Ing. Christian Hoene, University of T=FCbingen, Computer Science, =
Chair of
Communication Networks, Research Group Interactive Communication Systems
(ICS)
Sand 13, 72076 T=FCbingen, Germany, Tel +49 7071 2970532,
www.net.uni-tuebingen.de




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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
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Subject: [payload] draft-ramalho-payload-g7110-00 to be discussed in AVTEXT
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[Apologies for cross posting AVT lists - reason below]

=20

All,

=20

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

=20

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

=20

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

=20

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

=20

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

=20

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

=20

Regards,

=20

Michael Ramalho

=20

=20

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



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

=20

=20

=09
=09

=20

=20


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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

<table class=3DMsoNormalTable border=3D0 cellspacing=3D0 cellpadding=3D0 =
width=3D543
 style=3D'width:407.25pt'>
 <tr>
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    width=3D110 height=3D73 id=3D"Picture_x0020_1"
    src=3D"cid:image001.gif@01CC4AD1.08107BA0"
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    <td nowrap valign=3Dtop style=3D'padding:0in 0in 11.25pt .25in'>
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    =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";color:#666666'>=
Michael
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style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'><br>
    </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>Technical Leader</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>Product Development</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    <a href=3D"mailto:mramalho@cisco.com"><span =
style=3D'color:#666666'>mramalho@cisco.com</span></a><br>
    Phone: </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>+1 919 476 2038</span></b><span =
style=3D'font-size:8.5pt;
    font-family:"Arial","sans-serif";color:#666666'><br>
    Mobile: </span><b><span =
style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'>+1 941 544 2844</span></b><span =
style=3D'font-size:8.5pt;
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Cisco
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style=3D'font-size:8.5pt;font-family:"Arial","sans-serif";
    color:#666666'><br>
    4564 Tuscana Drive<br>
    Sarasota, FL 34241-4201<br>
    United States<br>
    <b>http://ramalho.webhop.info</b><br>
    <b>Skype:</b> mramalho_mar42<o:p></o:p></span></p>
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<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

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

</div>

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Date: Mon, 25 Jul 2011 13:56:55 -0400
From: "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu>
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Cc: codec@ietf.org, payload@ietf.org
Subject: Re: [payload] Frames to Packets in draft-ietf-codec-opus-07?
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On 07/25/2011 10:23 AM, Christian Hoene wrote:
> I was wondering why the draft-ietf-codec-opus-07 draft has feature to
> concatenate multiple codec frames to packets. At the same time,
> draft-spittka-payload-rtp-opus-00 does not support to place multiple co=
dec
> entities in on RTP packet.

I think we are in agreement that for many packet transports, a single
frame per packet (max duration of 20~ms for general audio, minimum of
2.5ms or 400 pps) would be inappropriate, and so a concatenation scheme i=
s
required.  The question is then: where should the concatenation scheme be=

specified?

The motivation for specifying concatenation inside the codec is to promot=
e
interoperability and adoption by reducing the total amount of engineering=

work required to implement Opus.  This happens in two ways:

1. Avoiding duplicative software work

Given the structure of existing networked audio software, I expect that
software using Opus will typically rely on a pre-existing Opus library,
and there will be only a small number of widely used Opus implementations=
=2E
 However, there will likely be many implementations of the Opus-RTP
specification, due to engineering integration requirements.  As such, we
can reduce implementation effort by minimizing the complexity of the RTP
specification.

2. Avoiding duplicative specification work

Opus will be distributed over many transports other than RTP, and a
concatenation scheme may be desirable in each case.  Specifying
concatenation inside the codec avoids the need to specify a new
concatenation scheme for every transport.  Having a single concatenation
scheme should therefore accelerate adoption, minimize confusion when
moving packets between transports, and minimize the amount of software
required to make Opus work.

Personally, I've found H.264 to be an example of how specifying framing
outside the codec can create problems. H.264's packet/framing schemes (th=
e
"network abstraction layer", "access units", bytestream alignment, etc.)
are specified outside the codec, so that they must be handled by the
stream processing layer.  Correctly distinguishing and converting between=

the various possibilities is so confusing that, until the latest release
last month, gstreamer's H.264 parser could not correctly depayload H.264
from the popular AVCHD format.


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Subject: Re: [payload] [codec] Frames to Packets in draft-ietf-codec-opus-07?
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> 1) Remove the concatenation from draft-ietf-codec-opus-07 and add it to
> draft-spittka-payload-rtp-opus-00.

We have to this point bent over backwards to keep the bitstream from 
changing since we declared a (soft) freeze in February, because people 
specifically asked for this in order to begin testing. While the working 
group can still agree to change it if a _serious_ issue arises with the 
format, I personally do not think this qualifies.

From abegen@cisco.com  Mon Jul 25 21:34:59 2011
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: <payload@ietf.org>
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-klv-01
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The WGLC has ended a few days ago but I have not seen any comments on =
the list. This is a short draft. We would appreciate to get some =
comments on the list so that we can proceed with publication.

Thanks.

> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On =
Behalf Of Ali C. Begen (abegen)
> Sent: Wednesday, July 06, 2011 11:35 AM
> To: payload@ietf.org
> Subject: [payload] WGLC for draft-ietf-payload-rtp-klv-01
>=20
> Hi everyone,
>=20
> This is to announce a 2-week WGLC for the following draft (ends July =
20th):
> http://datatracker.ietf.org/doc/draft-ietf-payload-rtp-klv/
>=20
> Please review the document and post any questions/concerns you have to =
the mailing list.
>=20
> Thanks.
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload

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

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

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

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

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

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

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

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

-hadriel


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

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


From mramalho@cisco.com  Tue Jul 26 10:02:03 2011
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, <draft-ramalho-payload-g7110@tools.ietf.org>
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Hadriel,

Thank you for asking your question.

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

For the G.711.0 IETF payload draft, no.

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

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

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

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

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

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

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

Re: SRTP

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

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

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

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

Re: "Middleboxes are a pain"

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

Thanks again,

Michael Ramalho


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

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

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

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

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

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

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

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

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

-hadriel


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

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

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

From HKaplan@acmepacket.com  Tue Jul 26 10:51:17 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
Date: Tue, 26 Jul 2011 13:51:13 -0400
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Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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On Jul 26, 2011, at 1:01 PM, Michael Ramalho (mramalho) wrote:

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

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


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

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

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


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

But they're NOT equivalent.  For all intents and purposes G.711.0 is a diff=
erent codec from G.711, afaict.  That it can be implemented as a bump-in-th=
e-wire without full DSPs matters not.  The most logical thing for a middleb=
ox to do would be to handle it exactly like it handles middlebox-provided t=
ranscoding/transrating cases today, with normal SDP modifications/behavior,=
 no? (and by that I mean act as a first-class participant in SDP, by insert=
ing/adding codecs into the same m-line, and handling offer/answer appropria=
tely)

-hadriel


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On 07/26/2011 01:51 PM, Hadriel Kaplan wrote:

>> The suggested "hint" in the G.711 SDP is not to "signal G.711.0", but
>> rather for middleboxes like NATs/Firewalls/etc. to "expect G.711" in
>> either uncompressed (real G.711 with PT = [0 | 8]) or compressed (with
>> PT = Q) form as equivalents.
>
> But they're NOT equivalent.  For all intents and purposes G.711.0 is a different codec from G.711, afaict.  That it can be implemented as a bump-in-the-wire without full DSPs matters not.  The most logical thing for a middlebox to do would be to handle it exactly like it handles middlebox-provided transcoding/transrating cases today, with normal SDP modifications/behavior, no? (and by that I mean act as a first-class participant in SDP, by inserting/adding codecs into the same m-line, and handling offer/answer appropriately)

I tend to agree with Hadriel's opinion here; this really isn't much 
different from a pair of middleboxes in the path of a G.711 call 
choosing to transcode the audio stream to/from G.729 in order to save 
bandwidth (except that does require DSP-type processing, but that 
shouldn't affect how the transformation is signaled).

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

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

Thanks for the volley.

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

Michael

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


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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

-hadriel


From mramalho@cisco.com  Tue Jul 26 15:43:51 2011
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Kevin P. Fleming" <kpfleming@digium.com>, <payload@ietf.org>
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Kevin,

Re: "this really isn't much  different from a pair of middleboxes in the
path of a G.711 call choosing to transcode the audio stream to/from
G.729 in order to save bandwidth "

I respectfully disagree.

While a transcode from G.711 to G.729 "saves bandwidth" ... it also
DEGRADES VOICE QUALITY. Thus adjacent signaling elements and/or
endpoints and/or service providers SHOULD be informed that a transcode
that degraded voice quality occurred. A natural way to inform the "other
parties" is via signaling.

A transcode from G.711 to G711.0 "saves bandwidth" (for zero-mean
acoustic signals) ... but it does NOT degrade voice quality. Indeed,
after the decompression the IDENTICAL G.711 PAYLOAD is reproduced. Thus
adjacent signaling elements and/or endpoints and/or service providers
NEED NOT be informed that it even occurred!

So while I would heartily agree that lossy transcodes be signaled to the
endpoints, I disagree that they need to be informed of a transformation
which is truly invisible to the endpoints.

Do we signal to the endpoints that RTP header compression occurred
somewhere on the end-to-end path? I don't think so.

Michael Ramalho

-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
Behalf Of Kevin P. Fleming
Sent: Tuesday, July 26, 2011 2:02 PM
To: payload@ietf.org
Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be
discussedinAVTEXT

On 07/26/2011 01:51 PM, Hadriel Kaplan wrote:

>> The suggested "hint" in the G.711 SDP is not to "signal G.711.0", but
>> rather for middleboxes like NATs/Firewalls/etc. to "expect G.711" in
>> either uncompressed (real G.711 with PT =3D [0 | 8]) or compressed
(with
>> PT =3D Q) form as equivalents.
>
> But they're NOT equivalent.  For all intents and purposes G.711.0 is a
different codec from G.711, afaict.  That it can be implemented as a
bump-in-the-wire without full DSPs matters not.  The most logical thing
for a middlebox to do would be to handle it exactly like it handles
middlebox-provided transcoding/transrating cases today, with normal SDP
modifications/behavior, no? (and by that I mean act as a first-class
participant in SDP, by inserting/adding codecs into the same m-line, and
handling offer/answer appropriately)

I tend to agree with Hadriel's opinion here; this really isn't much=20
different from a pair of middleboxes in the path of a G.711 call=20
choosing to transcode the audio stream to/from G.729 in order to save=20
bandwidth (except that does require DSP-type processing, but that=20
shouldn't affect how the transformation is signaled).

--=20
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype:
kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
_______________________________________________
payload mailing list
payload@ietf.org
https://www.ietf.org/mailman/listinfo/payload

From HKaplan@acmepacket.com  Tue Jul 26 22:42:24 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
Date: Wed, 27 Jul 2011 01:42:16 -0400
Thread-Topic: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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On Jul 26, 2011, at 4:32 PM, Michael Ramalho (mramalho) wrote:

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

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


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

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


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

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


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


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

-hadriel


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
Date: Wed, 27 Jul 2011 01:49:18 -0400
Thread-Topic: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussed inAVTEXT
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On Jul 26, 2011, at 4:32 PM, Michael Ramalho (mramalho) wrote:
>  Like I said earlier, I have a longer talk
> on the subject and would be happy to discuss further. Perhaps I should
> have asked for a slot in the behave WG for these issues?

Yes a presentation discussion will help, but I don't think BEHAVE would be =
the right WG.
IMHO, most of this draft belongs in PAYLOAD, but all of section 6 belongs s=
omewhere else, maybe MMUSIC.  Just my 2 cents though.

-hadriel



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Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussedinAVTEXT
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On 07/26/2011 06:43 PM, Michael Ramalho (mramalho) wrote:
> Kevin,
>
> Re: "this really isn't much  different from a pair of middleboxes in the
> path of a G.711 call choosing to transcode the audio stream to/from
> G.729 in order to save bandwidth "
>
> I respectfully disagree.
>
> While a transcode from G.711 to G.729 "saves bandwidth" ... it also
> DEGRADES VOICE QUALITY. Thus adjacent signaling elements and/or
> endpoints and/or service providers SHOULD be informed that a transcode
> that degraded voice quality occurred. A natural way to inform the "other
> parties" is via signaling.
>
> A transcode from G.711 to G711.0 "saves bandwidth" (for zero-mean
> acoustic signals) ... but it does NOT degrade voice quality. Indeed,
> after the decompression the IDENTICAL G.711 PAYLOAD is reproduced. Thus
> adjacent signaling elements and/or endpoints and/or service providers
> NEED NOT be informed that it even occurred!
>
> So while I would heartily agree that lossy transcodes be signaled to the
> endpoints, I disagree that they need to be informed of a transformation
> which is truly invisible to the endpoints.

Fair enough. I probably shouldn't have tried to read the draft yesterday 
in the middle of an IETF WG meeting. Mea culpa.

> Do we signal to the endpoints that RTP header compression occurred
> somewhere on the end-to-end path? I don't think so.

No, but there is still signaling between the middleboxes that decide to 
compress/uncompress RTP headers along the path, and the endpoints don't 
even have to be aware that this is a possibility. Granted, this draft 
also allows for such 'endpoint ignorance' if the middleboxes use some 
signaling mechanism (defined elsewhere), but it just feels a bit strange 
to me to have endpoints use a mechanism to indicate to some elements 
that might be in the path how an optional behavior should be employed.

Is it at all realistic to get a static RTP payload number assigned for 
G.711.0? If that was done, then such middleboxes wouldn't need any help 
from the endpoints to select a payload number to use for the transformed 
media.


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

From stephen.botzko@gmail.com  Wed Jul 27 06:35:30 2011
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Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussedinAVTEXT
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--90e6ba475d4fd607da04a90d1e3e
Content-Type: text/plain; charset=ISO-8859-1

static payload types are a scarce resource, I wouldn't assign one to this
(even if it were practical), as this is a very specialized use case for a
rather specialized codec.

Stephen Botzko

On Wed, Jul 27, 2011 at 9:29 AM, Kevin P. Fleming <kpfleming@digium.com>wrote:

> On 07/26/2011 06:43 PM, Michael Ramalho (mramalho) wrote:
>
>> Kevin,
>>
>> Re: "this really isn't much  different from a pair of middleboxes in the
>> path of a G.711 call choosing to transcode the audio stream to/from
>> G.729 in order to save bandwidth "
>>
>> I respectfully disagree.
>>
>> While a transcode from G.711 to G.729 "saves bandwidth" ... it also
>> DEGRADES VOICE QUALITY. Thus adjacent signaling elements and/or
>> endpoints and/or service providers SHOULD be informed that a transcode
>> that degraded voice quality occurred. A natural way to inform the "other
>> parties" is via signaling.
>>
>> A transcode from G.711 to G711.0 "saves bandwidth" (for zero-mean
>> acoustic signals) ... but it does NOT degrade voice quality. Indeed,
>> after the decompression the IDENTICAL G.711 PAYLOAD is reproduced. Thus
>> adjacent signaling elements and/or endpoints and/or service providers
>> NEED NOT be informed that it even occurred!
>>
>> So while I would heartily agree that lossy transcodes be signaled to the
>> endpoints, I disagree that they need to be informed of a transformation
>> which is truly invisible to the endpoints.
>>
>
> Fair enough. I probably shouldn't have tried to read the draft yesterday in
> the middle of an IETF WG meeting. Mea culpa.
>
>
>  Do we signal to the endpoints that RTP header compression occurred
>> somewhere on the end-to-end path? I don't think so.
>>
>
> No, but there is still signaling between the middleboxes that decide to
> compress/uncompress RTP headers along the path, and the endpoints don't even
> have to be aware that this is a possibility. Granted, this draft also allows
> for such 'endpoint ignorance' if the middleboxes use some signaling
> mechanism (defined elsewhere), but it just feels a bit strange to me to have
> endpoints use a mechanism to indicate to some elements that might be in the
> path how an optional behavior should be employed.
>
> Is it at all realistic to get a static RTP payload number assigned for
> G.711.0? If that was done, then such middleboxes wouldn't need any help from
> the endpoints to select a payload number to use for the transformed media.
>
>
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> ______________________________**_________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/**listinfo/payload<https://www.ietf.org/mailman/listinfo/payload>
>

--90e6ba475d4fd607da04a90d1e3e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

static payload types are a scarce resource, I wouldn&#39;t assign one to th=
is (even if it were practical), as this is a very specialized use case for =
a rather specialized codec.<br><br>Stephen Botzko<br><br><div class=3D"gmai=
l_quote">
On Wed, Jul 27, 2011 at 9:29 AM, Kevin P. Fleming <span dir=3D"ltr">&lt;<a =
href=3D"mailto:kpfleming@digium.com">kpfleming@digium.com</a>&gt;</span> wr=
ote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"im">On 07/26/2011 06:43 PM, Michael Ramalho (mramalho) wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Kevin,<br>
<br>
Re: &quot;this really isn&#39;t much =A0different from a pair of middleboxe=
s in the<br>
path of a G.711 call choosing to transcode the audio stream to/from<br>
G.729 in order to save bandwidth &quot;<br>
<br>
I respectfully disagree.<br>
<br>
While a transcode from G.711 to G.729 &quot;saves bandwidth&quot; ... it al=
so<br>
DEGRADES VOICE QUALITY. Thus adjacent signaling elements and/or<br>
endpoints and/or service providers SHOULD be informed that a transcode<br>
that degraded voice quality occurred. A natural way to inform the &quot;oth=
er<br>
parties&quot; is via signaling.<br>
<br>
A transcode from G.711 to G711.0 &quot;saves bandwidth&quot; (for zero-mean=
<br>
acoustic signals) ... but it does NOT degrade voice quality. Indeed,<br>
after the decompression the IDENTICAL G.711 PAYLOAD is reproduced. Thus<br>
adjacent signaling elements and/or endpoints and/or service providers<br>
NEED NOT be informed that it even occurred!<br>
<br>
So while I would heartily agree that lossy transcodes be signaled to the<br=
>
endpoints, I disagree that they need to be informed of a transformation<br>
which is truly invisible to the endpoints.<br>
</blockquote>
<br></div>
Fair enough. I probably shouldn&#39;t have tried to read the draft yesterda=
y in the middle of an IETF WG meeting. Mea culpa.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Do we signal to the endpoints that RTP header compression occurred<br>
somewhere on the end-to-end path? I don&#39;t think so.<br>
</blockquote>
<br></div>
No, but there is still signaling between the middleboxes that decide to com=
press/uncompress RTP headers along the path, and the endpoints don&#39;t ev=
en have to be aware that this is a possibility. Granted, this draft also al=
lows for such &#39;endpoint ignorance&#39; if the middleboxes use some sign=
aling mechanism (defined elsewhere), but it just feels a bit strange to me =
to have endpoints use a mechanism to indicate to some elements that might b=
e in the path how an optional behavior should be employed.<br>

<br>
Is it at all realistic to get a static RTP payload number assigned for G.71=
1.0? If that was done, then such middleboxes wouldn&#39;t need any help fro=
m the endpoints to select a payload number to use for the transformed media=
.<div>
<div></div><div class=3D"h5"><br>
<br>
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href=3D"mailto:kfleming@digium.com" target=3D"_blank">kfleming@d=
igium.com</a> | SIP: <a href=3D"mailto:kpfleming@digium.com" target=3D"_bla=
nk">kpfleming@digium.com</a> | Skype: kpfleming<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">www.dig=
ium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank">www=
.asterisk.org</a><br>
______________________________<u></u>_________________<br>
payload mailing list<br>
<a href=3D"mailto:payload@ietf.org" target=3D"_blank">payload@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/payload" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/payload</a><br>
</div></div></blockquote></div><br>

--90e6ba475d4fd607da04a90d1e3e--

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To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
Date: Wed, 27 Jul 2011 09:56:35 -0400
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On Jul 26, 2011, at 6:43 PM, Michael Ramalho (mramalho) wrote:

> While a transcode from G.711 to G.729 "saves bandwidth" ... it also
> DEGRADES VOICE QUALITY.

It doesn't matter whether a particular codec or payload transform is lossle=
ss or not, from the perspective of SDP and RTP.  SRTP encrypt/decrypt and a=
uthentication is also completely "lossless".  You could, in theory, do SRTP=
 between two middleboxes.  All you'd need is a way to exchange keys between=
 the two middleboxes, just like for G.711.0 you need a way to exchange payl=
oad types and companding.  Obviously SBCs and other middleboxes can and do =
SRTP termination/origination, but we would never have done it the way this =
draft is describing to do G.711.0.  I must be missing something about your =
use-case that makes it different.


> Do we signal to the endpoints that RTP header compression occurred
> somewhere on the end-to-end path? I don't think so.
>=20

ROHC for RTP only works in extremely constrained deployment scenarios, and =
as far as I'm aware can only be done either by the devices generating the R=
TP so they know what IP/UDP combo stream is RTP before-hand, or by channeli=
zation/separation and signaling for a context.  And as far as I know it has=
 signaling in ROHC itself - it's not just blindly starting to compress with=
out knowing the far-end is capable of decompressing and what modes.=20

-hadriel


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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It is literally impossible to assign one, because RFC 3551 establishes a
policy that no more static payload type values would be assigned for
RTP/AVP:
=20
<snip>
   This specification establishes the policy that no additional static
   payload types will be assigned beyond the ones defined in this
   document.  Establishing this policy avoids the problem of trying to
   create a set of criteria for accepting static assignments and
   encourages the implementation and deployment of the dynamic payload
   type mechanisms.
=20
   The final set of static payload type assignments is provided in
   Tables 4 and 5.
</snip>
=20
Muthu
=20
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
Behalf Of Stephen Botzko
Sent: Wednesday, July 27, 2011 7:05 PM
To: Kevin P. Fleming
Cc: payload@ietf.org
Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be
discussedinAVTEXT
=20
static payload types are a scarce resource, I wouldn't assign one to
this (even if it were practical), as this is a very specialized use case
for a rather specialized codec.

Stephen Botzko
On Wed, Jul 27, 2011 at 9:29 AM, Kevin P. Fleming <kpfleming@digium.com>
wrote:
On 07/26/2011 06:43 PM, Michael Ramalho (mramalho) wrote:
Kevin,

Re: "this really isn't much  different from a pair of middleboxes in the
path of a G.711 call choosing to transcode the audio stream to/from
G.729 in order to save bandwidth "

I respectfully disagree.

While a transcode from G.711 to G.729 "saves bandwidth" ... it also
DEGRADES VOICE QUALITY. Thus adjacent signaling elements and/or
endpoints and/or service providers SHOULD be informed that a transcode
that degraded voice quality occurred. A natural way to inform the "other
parties" is via signaling.

A transcode from G.711 to G711.0 "saves bandwidth" (for zero-mean
acoustic signals) ... but it does NOT degrade voice quality. Indeed,
after the decompression the IDENTICAL G.711 PAYLOAD is reproduced. Thus
adjacent signaling elements and/or endpoints and/or service providers
NEED NOT be informed that it even occurred!

So while I would heartily agree that lossy transcodes be signaled to the
endpoints, I disagree that they need to be informed of a transformation
which is truly invisible to the endpoints.
=20
Fair enough. I probably shouldn't have tried to read the draft yesterday
in the middle of an IETF WG meeting. Mea culpa.
	=20
	Do we signal to the endpoints that RTP header compression
occurred
	somewhere on the end-to-end path? I don't think so.
=20
No, but there is still signaling between the middleboxes that decide to
compress/uncompress RTP headers along the path, and the endpoints don't
even have to be aware that this is a possibility. Granted, this draft
also allows for such 'endpoint ignorance' if the middleboxes use some
signaling mechanism (defined elsewhere), but it just feels a bit strange
to me to have endpoints use a mechanism to indicate to some elements
that might be in the path how an optional behavior should be employed.

Is it at all realistic to get a static RTP payload number assigned for
G.711.0? If that was done, then such middleboxes wouldn't need any help
from the endpoints to select a payload number to use for the transformed
media.



--=20
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype:
kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
_______________________________________________
payload mailing list
payload@ietf.org
https://www.ietf.org/mailman/listinfo/payload
=20

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</head>

<body lang=3DEN-US link=3Dblue vlink=3Dpurple =
style=3D'tab-interval:.5in'>

<div class=3DWordSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'>It is
literally impossible to assign one, because <span =
class=3DSpellE>RFC</span> 3551 establishes
a policy that no more static payload type values would be assigned for =
<span
class=3DSpellE>RTP</span>/<span =
class=3DSpellE>AVP</span>:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'>&lt;snip&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>This specification =
establishes the policy
that no additional static<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>payload types will be =
assigned beyond the
ones defined in this<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>document.<span =
style=3D'mso-spacerun:yes'>&nbsp;
</span>Establishing this policy avoids the problem of trying =
to<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>create a set of criteria =
for accepting
static assignments and<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>encourages the =
implementation and deployment
of the dynamic payload<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>type =
mechanisms.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>The final set of static =
payload type
assignments is provided in<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New Roman"'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>Tables 4 and =
5.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'>&lt;/snip&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'>Muthu<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman"'><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";
mso-fareast-font-family:"Times New Roman"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";mso-fareast-f=
ont-family:
"Times New Roman"'> payload-bounces@ietf.org =
[mailto:payload-bounces@ietf.org] <b>On
Behalf Of </b>Stephen Botzko<br>
<b>Sent:</b> Wednesday, July 27, 2011 7:05 PM<br>
<b>To:</b> Kevin P. Fleming<br>
<b>Cc:</b> payload@ietf.org<br>
<b>Subject:</b> Re: [payload] draft-ramalho-payload-g7110-00 to be
discussedinAVTEXT<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>static payload types =
are a
scarce resource, I wouldn't assign one to this (even if it were =
practical), as
this is a very specialized use case for a rather specialized codec.<br>
<br>
Stephen Botzko<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On Wed, Jul 27, 2011 at 9:29 AM, Kevin P. Fleming =
&lt;<a
href=3D"mailto:kpfleming@digium.com">kpfleming@digium.com</a>&gt; =
wrote:<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On 07/26/2011 06:43 PM, Michael Ramalho (mramalho) =
wrote:<o:p></o:p></p>

<p class=3DMsoNormal>Kevin,<br>
<br>
Re: &quot;this really isn't much &nbsp;different from a pair of =
middleboxes in
the<br>
path of a G.711 call choosing to transcode the audio stream to/from<br>
G.729 in order to save bandwidth &quot;<br>
<br>
I respectfully disagree.<br>
<br>
While a transcode from G.711 to G.729 &quot;saves bandwidth&quot; ... it =
also<br>
DEGRADES VOICE QUALITY. Thus adjacent signaling elements and/or<br>
endpoints and/or service providers SHOULD be informed that a =
transcode<br>
that degraded voice quality occurred. A natural way to inform the =
&quot;other<br>
parties&quot; is via signaling.<br>
<br>
A transcode from G.711 to G711.0 &quot;saves bandwidth&quot; (for =
zero-mean<br>
acoustic signals) ... but it does NOT degrade voice quality. Indeed,<br>
after the decompression the IDENTICAL G.711 PAYLOAD is reproduced. =
Thus<br>
adjacent signaling elements and/or endpoints and/or service =
providers<br>
NEED NOT be informed that it even occurred!<br>
<br>
So while I would heartily agree that lossy transcodes be signaled to =
the<br>
endpoints, I disagree that they need to be informed of a =
transformation<br>
which is truly invisible to the endpoints.<o:p></o:p></p>

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

</div>

<p class=3DMsoNormal>Fair enough. I probably shouldn't have tried to =
read the
draft yesterday in the middle of an IETF WG meeting. Mea =
culpa.<o:p></o:p></p>

<div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;mso-border-left-alt:
solid #CCCCCC .75pt;padding:0in 0in 0in =
6.0pt;margin-left:4.8pt;margin-right:
0in'>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Do we signal to the endpoints that RTP header =
compression
occurred<br>
somewhere on the end-to-end path? I don't think so.<o:p></o:p></p>

</blockquote>

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

</div>

<p class=3DMsoNormal>No, but there is still signaling between the =
middleboxes
that decide to compress/uncompress RTP headers along the path, and the
endpoints don't even have to be aware that this is a possibility. =
Granted, this
draft also allows for such 'endpoint ignorance' if the middleboxes use =
some
signaling mechanism (defined elsewhere), but it just feels a bit strange =
to me
to have endpoints use a mechanism to indicate to some elements that =
might be in
the path how an optional behavior should be employed.<br>
<br>
Is it at all realistic to get a static RTP payload number assigned for =
G.711.0?
If that was done, then such middleboxes wouldn't need any help from the
endpoints to select a payload number to use for the transformed =
media.<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><br>
<br>
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href=3D"mailto:kfleming@digium.com" =
target=3D"_blank">kfleming@digium.com</a>
| SIP: <a href=3D"mailto:kpfleming@digium.com" =
target=3D"_blank">kpfleming@digium.com</a>
| Skype: kpfleming<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at <a href=3D"http://www.digium.com" =
target=3D"_blank">www.digium.com</a>
&amp; <a href=3D"http://www.asterisk.org" =
target=3D"_blank">www.asterisk.org</a><br>
_______________________________________________<br>
payload mailing list<br>
<a href=3D"mailto:payload@ietf.org" =
target=3D"_blank">payload@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/payload" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/payload</a><o:p><=
/o:p></p>

</div>

</div>

</div>

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

</div>

</div>

</body>

</html>

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

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

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

Do others have an opinion?

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

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

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

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

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

Thanks again,

Michael


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




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

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

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


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

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


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

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


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


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

-hadriel


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Thread-Topic: [payload] draft-ramalho-payload-g7110-00 to be discussedinAVTEXT
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References: <DCD4D03D-93A3-4EC9-A099-F22D92DE4BAC@acmepacket.com> <999109E6BC528947A871CDEB5EB908A0044E2EEB@XMB-RCD-209.cisco.com><CDB75F50-9298-4711-8C0B-968B47D3EE10@acmepacket.com> <4E2F0103.7020108@digium.com> <999109E6BC528947A871CDEB5EB908A0044E30DA@XMB-RCD-209.cisco.com> <ED81C4E8-8666-42D4-80DF-424A186D145B@acmepacket.com>
From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
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Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be discussedinAVTEXT
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Hadriel,

In-line below.

Michael

-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
Sent: Wednesday, July 27, 2011 9:57 AM
To: Michael Ramalho (mramalho)
Cc: Kevin P. Fleming; payload@ietf.org
Subject: Re: [payload] draft-ramalho-payload-g7110-00 to be
discussedinAVTEXT


On Jul 26, 2011, at 6:43 PM, Michael Ramalho (mramalho) wrote:

> While a transcode from G.711 to G.729 "saves bandwidth" ... it also
> DEGRADES VOICE QUALITY.

It doesn't matter whether a particular codec or payload transform is
lossless or not, from the perspective of SDP and RTP.  SRTP
encrypt/decrypt and authentication is also completely "lossless".  You
could, in theory, do SRTP between two middleboxes.

MAR: Not even in theory .. but also in practice!

MAR: SRTP was designed (as I understand it) from day one to encrypt
between "session endpoints".

MAR: Insofar as the UAs facing each other in two consenting SBCs are SIP
dialog endpoints (i.e., "in the middle" of a longer end-to-end "call")
... they can do SRTP between them!

MAR: I guess I don't get the "in theory" part.

All you'd need is a way to exchange keys between the two middleboxes,
just like for G.711.0 you need a way to exchange payload types and
companding.

MAR: The companding for the "G.711.0 compression segment" (as defined in
the draft) is assumed to be the companding used in the true user
endpoints ... which can be determined from the incoming PT (0 or 8).

MAR: The communication/negotiation/exchange between the "G.711.0 tunnel
endpoints" is purposely unspecified in the draft. Indeed, how the
G.711.0 compression/decompression endpoints "find themselves" is also
purposely unspecified in the draft.

MAR: In the absence of middleboxes and NAT/PAT and in certain topologies
... there are a variety of simple "in-media" methods by which
compression/decompression endpoints can find themselves. There are many
"gotchas" for those use cases ... but again ... these cases are
purposely unspecified in the draft. Unfortunately, I was not allotted
enough time to go through those cases in the avtext meeting today.

MAR: My comment about various forms of header compression was to point
out, via example, that SBCs do not always know what is going on "within"
the flows that traverse them (I could put a friendly comment that SBCs
being "control freaks" ... but I'll refrain ;-) ).

Obviously SBCs and other middleboxes can and do SRTP
termination/origination, but we would never have done it the way this
draft is describing to do G.711.0.  I must be missing something about
your use-case that makes it different.

MAR: I wouldn't use per-flow encryption in your example either ... but
that is a different story.

> Do we signal to the endpoints that RTP header compression occurred
> somewhere on the end-to-end path? I don't think so.
>=20

ROHC for RTP only works in extremely constrained deployment scenarios,
and as far as I'm aware can only be done either by the devices
generating the RTP so they know what IP/UDP combo stream is RTP
before-hand, or by channelization/separation and signaling for a
context.  And as far as I know it has signaling in ROHC itself - it's
not just blindly starting to compress without knowing the far-end is
capable of decompressing and what modes.=20

-hadriel


From jonathan@vidyo.com  Wed Jul 27 12:11:17 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "codec@ietf.org" <codec@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Date: Wed, 27 Jul 2011 15:10:34 -0400
Thread-Topic: [payload] Frames to Packets in draft-ietf-codec-opus-07?
Thread-Index: AcxMkNzxKrhe3c1/SY+h3AcPKT1kOQ==
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On Jul 25, 2011, at 1:56 PM, Benjamin M. Schwartz wrote:

> On 07/25/2011 10:23 AM, Christian Hoene wrote:
>> I was wondering why the draft-ietf-codec-opus-07 draft has feature to
>> concatenate multiple codec frames to packets. At the same time,
>> draft-spittka-payload-rtp-opus-00 does not support to place multiple cod=
ec
>> entities in on RTP packet.
>=20
> I think we are in agreement that for many packet transports, a single
> frame per packet (max duration of 20~ms for general audio, minimum of
> 2.5ms or 400 pps) would be inappropriate, and so a concatenation scheme i=
s
> required.  The question is then: where should the concatenation scheme be
> specified?

Does the Opus framing allow packets to be split and/or aggregated by a midd=
lebox along frame boundaries, without requiring transcoding?  (The use case=
 for splitting is to forward traffic to a box with a lower maxptime than th=
e stream you're receiving, and the use case for aggregation is to reduce th=
e RTP header overhead at the cost of some additional delay.)

I tried to read the Opus spec to answer my question, and I think I came up =
with the answer, but I wanted to confirm this with the Opus experts.

Based on my reading of the Opus spec, it looks like splitting should be pos=
sible, relatively trivially.  Aggregation, however, is only possible if the=
 packets to be aggregated have the same value for the Opus TOC byte.

I get the impression that the TOC byte is expected to change only relativel=
y infrequently in most encoding modes, in which case this would be fine -- =
there may be transient packets which couldn't be aggregated, resulting in o=
ccasional "short" packets, but most packets could be aggregated.  However, =
if an encoder might change its TOC frequently, this could be more problemat=
ic.

If the (Payload) WG is okay with this limitation (and the limitation of a m=
aximum of 120 ms packets), I think the internal-framing-only is okay; if no=
t, we might need external framing as well. (I appreciate that you don't wan=
t to change the Opus bitstream proper, at this point.)

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [payload] Frames to Packets in draft-ietf-codec-opus-07?
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On 07/27/2011 03:10 PM, Jonathan Lennox wrote:
> Based on my reading of the Opus spec, it looks like splitting should be=
 possible, relatively trivially.  Aggregation, however, is only possible =
if the packets to be aggregated have the same value for the Opus TOC byte=
=2E

That is a correct reading of the Opus spec as of Opus-02 and later.  Prio=
r
to Opus-02, the spec also described a packet mode that could contain
frames of different types, allowing arbitrary aggregation.  I believe it
was removed because the authors expected that it would be rarely used, an=
d
removing it allowed them to improve the efficiency of more common cases.

--Ben


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
Date: Wed, 27 Jul 2011 15:52:54 -0400
Thread-Topic: [payload] draft-ramalho-payload-g7110-00 to be discussedinAVTEXT
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On Jul 27, 2011, at 2:36 PM, Michael Ramalho (mramalho) wrote:

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
>=20
> It doesn't matter whether a particular codec or payload transform is
> lossless or not, from the perspective of SDP and RTP.  SRTP
> encrypt/decrypt and authentication is also completely "lossless".  You
> could, in theory, do SRTP between two middleboxes.
>=20
> MAR: Not even in theory .. but also in practice!

Yes, exactly - they do it in *practice*, but SBCs don't do it by "not telli=
ng the endpoints" or hiding stuff, because the SBCs *are* the endpoints fro=
m the perspective of SDP and IP/UDP/RTP.  The "real" endpoints (namely your=
 phones) don't know SRTP happened in the middle, and they don't know transc=
oding G.711<->G.729 happened either, etc., but the phones not the real endp=
oints from the perspective of IP, UDP, RTP, nor SDP - only from a human per=
spective.

-hadriel



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Date: Fri, 29 Jul 2011 01:46:15 +0300
From: Roni Even <Even.roni@huawei.com>
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Hi,

As was stated in the payload WG status presentation the WG chairs intend to
ask for  a new milestone to the charter Submit RTP Payload Format for IETF
OPUS codec for Proposed Standard. 
 
If there are no objection we will also accept draft-spittka-payload-rtp-opus
as the initial document to address this new work item
 
Thanks
Roni Even

 

 


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class=3DMsoNormal>Hi,<o:p></o:p></p><pre><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>As was =
stated in the payload WG status presentation the WG chairs intend to ask =
for &nbsp;a new milestone to the charter Submit RTP Payload Format for =
IETF OPUS codec for Proposed Standard. =
<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"'>If there =
are no objection we will also accept draft-spittka-payload-rtp-opus as =
the initial document to address this new work =
item<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"'>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><p class=3DMsoNormal><span =
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From prvs=1839367b2=tterribe@xiph.org  Thu Jul 28 16:53:26 2011
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Subject: Re: [payload] [codec] Frames to Packets in draft-ietf-codec-opus-07?
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> Would it be possible to add an appendix or create an additional document
> that describes how a middlebox could detect mode changes and avoid
> concatenating in these cases?

This is basically as simple as examining the first byte of the packet. 
But if you have some example text I'd be happy to make an appendix for 
it (and possibly other related issues).

From giles@thaumas.net  Tue Jul 26 11:01:50 2011
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From: Ralph Giles <giles@thaumas.net>
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Subject: Re: [payload] [codec] Frames to Packets in draft-ietf-codec-opus-07?
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2011/7/25 Christian Hoene <hoene@uni-tuebingen.de>:

> I was wondering why the draft-ietf-codec-opus-07 draft has feature to
> concatenate multiple codec frames to packets. At the same time,
> draft-spittka-payload-rtp-opus-00 does not support to place multiple codec
> entities in on RTP packet.

I believe the codec authors have described some of their reasoning for
the internal framing. However, it sounds like your concern is the
presence of framing at the codec level, rather than at the RTP payload
level.

There is precedent for this in previous specifications.
http://tools.ietf.org/html/rfc3551#section-4.4 "Guidelines for
Frame-Based Audio Encodings" says:

  "Frame-based encodings encode a fixed-length block of audio into
   another block of compressed data, typically also of fixed length.
   For frame-based encodings, the sender MAY choose to combine several
   such frames into a single RTP packet.  The receiver can tell the
   number of frames contained in an RTP packet, if all the frames have
   the same length, by dividing the RTP payload length by the audio
   frame size which is defined as part of the encoding.  This does not
   work when carrying frames of different sizes unless the frame sizes
   are relatively prime.  If not, the frames MUST indicate their size.

   [...]

   All frame-oriented audio codecs SHOULD be able to encode and decode
   several consecutive frames within a single packet. [...]"

Opus can use variable length encoded frames, so the clause that
_frames_ must indicate their size, so that the codec can "decode
several consecutive frames within a single packet" applies. The
referent here is frames and codecs, not the payload format, so I don't
think we're breaking new ground here by having some framing at the
codec level.

I would also point out that the opus codec draft is nearly finished.
Continuing to specify framing there, rather than adding complication
to the newer RTP draft, is the quicker path to standardization of both
levels.

FWIW,
 -r

From matthew@matthew.at  Thu Jul 28 16:39:26 2011
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Subject: Re: [payload] [codec] Frames to Packets in draft-ietf-codec-opus-07?
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In response to comments at the mic at today's meeting:

Would it be possible to add an appendix or create an additional document =
that describes how a middlebox could detect mode changes and avoid =
concatenating in these cases?

Matthew Kaufman


From jmvalin@jmvalin.ca  Sun Jul 31 18:27:15 2011
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Date: Sun, 31 Jul 2011 21:26:47 -0400
From: Jean-Marc Valin <jmvalin@jmvalin.ca>
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Subject: Re: [payload] [codec] Frames to Packets in draft-ietf-codec-opus-07?
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Hi Jonathan, (sorry, just received your post now)

On 11-07-27 03:10 PM, Jonathan Lennox wrote:
> Based on my reading of the Opus spec, it looks like splitting should
> be possible, relatively trivially.  Aggregation, however, is only
> possible if the packets to be aggregated have the same value for the
> Opus TOC byte.

Correct.

> I get the impression that the TOC byte is expected to change only
> relatively infrequently in most encoding modes, in which case this
> would be fine -- there may be transient packets which couldn't be
> aggregated, resulting in occasional "short" packets, but most packets
> could be aggregated.  However, if an encoder might change its TOC
> frequently, this could be more problematic.

I don't really see the TOC byte changing frequently. I would expect that
for many/most uses, the TOC would never change during an RTP session.
For other uses, I'd be surprised to see it changed more than once every
5-10 minutes. The only potentially problematic case (when it comes to
aggregation) would be contents that has music and speech that alternates
frequently.

> If the (Payload) WG is okay with this limitation (and the limitation
> of a maximum of 120 ms packets), I think the internal-framing-only is
> okay; if not, we might need external framing as well. (I appreciate
> that you don't want to change the Opus bitstream proper, at this
> point.)

I don't see these limitations as being a problem, but if they really are
(in practice, not in some theoretical scenario), then I'd rather change
this in the bit-stream than have RTP implementations mess it up on a
regular basis (and having non-RTP implementations do it differently).

Cheers,

	Jean-Marc
