
From nobody Mon Jul  6 04:02:31 2015
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From: Ron Even <ron.even.tlv@gmail.com>
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Hi,
Here is the proposed agenda, any comments?
If we will not have more presentations we may have the payload session
joined with another of the AVT WGs
See you in Prague
Roni Even

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From nobody Mon Jul  6 06:09:58 2015
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Hi,
Here is the updated agenda, any comments?
See you in Prague
Roni Even

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From nobody Mon Jul  6 14:43:15 2015
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Subject: [payload] I-D Action: draft-ietf-payload-vp9-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Audio/Video Transport Payloads Working Group of the IETF.

        Title           : RTP Payload Format for VP9 Video
        Authors         : Justin Uberti
                          Stefan Holmer
                          Magnus Flodman
                          Jonathan Lennox
                          Danny Hong
	Filename        : draft-ietf-payload-vp9-00.txt
	Pages           : 20
	Date            : 2015-07-06

Abstract:
   This memo describes an RTP payload format for the VP9 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.  It includes provisions for temporal and spatial
   scalability.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-vp9/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-payload-vp9-00


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

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


From nobody Mon Jul  6 15:03:04 2015
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Subject: Re: [payload] I-D Action: draft-ietf-payload-vp9-00.txt
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This version is largely unchanged from the individual submission, other tha=
n chasing some corresponding changes in the VP8 payload.

The RTP payload formats are probably stable at this point.  SDP negotiation=
 still needs to be worked out.

Please read and comment.

> On Jul 6, 2015, at 5:43 PM, Internet-Drafts@ietf.org wrote:
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
> This draft is a work item of the Audio/Video Transport Payloads Working G=
roup of the IETF.
>=20
>        Title           : RTP Payload Format for VP9 Video
>        Authors         : Justin Uberti
>                          Stefan Holmer
>                          Magnus Flodman
>                          Jonathan Lennox
>                          Danny Hong
> 	Filename        : draft-ietf-payload-vp9-00.txt
> 	Pages           : 20
> 	Date            : 2015-07-06
>=20
> Abstract:
>   This memo describes an RTP payload format for the VP9 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.  It includes provisions for temporal and spatial
>   scalability.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-payload-vp9/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-payload-vp9-00
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20


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From nobody Mon Jul 13 13:57:29 2015
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Cc: draft-ietf-payload-vp8.all@ietf.org, IETF Discussion List <ietf@ietf.org>, General area reviewing team <gen-art@ietf.org>, payload-chairs@ietf.org, payload@ietf.org
Subject: Re: [payload] Gen-art 2nd LC review of draft-ietf-payload-vp8-16
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(+payload and Harald)

Elwyn: Thanks for the detailed review!

Authors: Please address Elwyn's comments at your earliest convenience. 
There's quite a bit here, and I assume we will need a new revision 
before taking this to the IESG telechat, but it would be helpful to 
discuss any proposed changes via email before submitting.

Thanks!

Ben.


On 13 Jul 2015, at 6:58, Elwyn Davies wrote:

> I am the assigned Gen-ART reviewer for this draft. For background on
> Gen-ART, please see the FAQ at
>
> <http://wiki.tools.ietf.org/area/gen/trac/wiki/GenArtfaq>.
>
> Please resolve these comments along with any other Last Call comments
> you may receive.
>
> Document: draft-ietf-payload-vp8-16.txt
> Reviewer: Elwyn Davies
> Review Date: 2015/07/10
> IETF LC End Date: 2015/07/13
> IESG Telechat date: (if known) -
>
> Summary: Gosh, it has been a long time since the last review!
>
> Major issues:
>
> Minor issues:
> s4.2: Various of the frame indices either SHOULD or, in the case of
> KEYIDX, MAY start at random values.  The rationale(s) for these
> requirements are not explained - nor is the reason why it it is only a
> MAY for KEYIDX whereas it is SHOULD for PictureID and what might 
> happen
> were the MAY/SHOULD to be ignored.
>
> Is the rationale something that should be mentioned in the security
> considerations?
>
> s10.1:  The downref to RFC 6386 identified by idnits was duly noted in 
> the last call
> announcement.  I don't have a problem with the downref.
>
> Nits/editorial comments:
>
> General: Consistent usage of "prediction and mode" vs 
> "prediction/mode" would be desirable.
>
> s1, para 2: Needs to introduce the 'macroblock' jargon defined in RFC 
> 6386. Suggest:
> OLD:
> VP8 is based on decomposition of frames into square sub-blocks of
> pixels, prediction of such sub-blocks using previously constructed
> blocks, and...
> NEW:
> VP8 is based on decomposition of frames into square sub-blocks of
> pixels known as "macroblocks" (see Section 2 of [RFC6386]).  
> Prediction
> of such sub-blocks using previously constructed blocks, and...
> END
>
> s2: There are a number of technical terms brought over from RFC 6386.
> These should be listed in s2.  At least the following would be in the 
> list:
> key frame, interframe, golden frame, altref frame, macroblock.
> Also the terms picture selection index (RPSI) and slice loss 
> indication (SLI) from RFC 4585.
>
> s3, para 2: Providing a reference to explain what temporal scalability
> and the hierarchy selection is all about would be useful.  One 
> possibility
> would be:
>
> Lim, J. Yang, and B. Jeon,”Fast Coding Mode Decision for Scalable 
> Video Coding”,
> 10th Int’l Conf. on Advanced Communication Technology, vol. 3, pp. 
> 1897-1900, 2008.
>
> It would also help to add a pointer to the TL0PICIDX description in 
> s4.2, thus:
> ADD:
> Support for temporal scalability is provided by the optional TL0PICIDX 
> and TID/Y/KEYIDX
> fields described in Section 4.2.
> END
>
> ss3 and 4: The discussion of how the frame payload might or might not 
> be distributed across
> multiple packets is not very clear.   The draft has in mind a 
> 'preferred use case' (para 1
> of s4.4 says:
>
>> In the preferred use case, the S bit and PID fields
>> described in Section 4.2 should be used to indicate what the packet
>> contains.
>
> I think it would be helpful to be a bit more positive about this in 
> s3, para 3:
> OLD:
> This memo
> allows the partitions to be sent separately or in the same RTP
> packet.  It may be beneficial for decoder error-concealment to send
> the partitions in different packets, even though it is not mandatory
> according to this specification.
> NEW:
> Accordingly, this document RECOMMENDS that the frame is packetized by 
> the sender
> with each data partition in a separate packet or packets.  This may be 
> beneficial
> for decoder error concealment and the payload format described in 
> Section 4 provides
> fields that allow the partitions to be identified even if the first 
> partition is
> not available.  The sender can, alternatively, aggregate the data 
> partitions into
> a single data stream and, optionally, split it into several packets 
> without
> consideration of the partition boundaries.  The receiver can use the 
> length
> information in the first partition to identify the partitions during 
> decoding.
> END
>
> s4/Figure 1: s/bytes/octets/ (2 places)
>
> Figure 1, caption: s/sequel/Sections 4.2 and 4.3/
>
> Figure 1: The format shown is correct only for the first packet in a 
> frame.  Subsequent
> packets will not contain the payload header and will have later octets 
> of the frame payload.
> This should be mentioned in drawing and/or in the caption.
>
> s4.1, last two paras:
>> Sequence number:  The sequence numbers are monotonically increasing
>>    and set as packets are sent.
>>
>>    The remaining RTP header fields are used as specified in
>>    [RFC3550].
> Redefining the Sequence Number field seems gratuitous and potentially 
> dangerous,
> given that it is *very carefully* defined in RFC 3550 and the overall 
> intent does
> not seem any different from that in RFC 3550.  OTOH a note about the 
> (non?-)use of the X bit
> and the Payload Type field (PT) would be useful.  I suggest replacing 
> the paras above with:
> NEW:
> X bit: MUST be 0.  The VP8 RTP profile does not use the RTP Header 
> Extension capability.
>
> PT (Payload Type):  In line with the policy in Section 3 of [RFC3551], 
> applications
>    using the VP8 RTP payload profile MUST assign a dynamic payload 
> type number to
>    be used in each RTP session and provide a mechanism to indicate the 
> mapping.
>    See Section 6.2 for the mechanism to be used with the Session 
> Description Protocol (SDP)
>    [RFC4566].
>
>    The remaining RTP Fixed Header Fields (V, P, CC, sequence number, 
> SSRC and CSRC
>    identfiers) are used as specified in Section 5.1 of [RFC5330].
> END
> Note that this may be considered to make the reference to RFC 3551 
> normative.
>
> s4.2, X bit:  There is potential for confusion between the X bit in 
> the fixed header
> and the X bit in the VP8 Payload Descriptor.  Perhaps changing this to 
> C for control bits
> would avoid the problem.
>
> s4.2, M bit: Similarly, it might be better to use B (for BIG) in place 
> of M to avoid confusion.
>
> s4.2, para 7 after Fig 2: For consistency:
> OLD:
> When the X bit is set to 1 in the first octet, the extension bit
> field octet MUST be provided as the second octet.  If the X bit is 0,
> the extension bit field octet MUST NOT be present, and no extensions
> (I, L, T, or K) are permitted.
> NEW:
> When the X [or C] bit is set to 1 in the first octet, the Extended 
> Control Bits
> field octet MUST be provided as the second octet.  If the X [or C] bit 
> is 0,
> the extension bit field octet MUST NOT be present, and no extensions
> (I, L, T, or K) are permitted.
> END
>
> s4.2: The issue of using either 'wrap' or 'restart' but not both for 
> restarting number sequences has been raised in various comments by 
> IESG members.
>
> s4.2, TL0PICIDX: I think, given the statements in s3 about not 
> defining the hierarchy, that more explanation is needed of what the 
> 'lowest or base layer' actually means.
>
> s4.2, TL0PICIDX:
>> If TID is larger than 0, TL0PICIDX
>>       indicates which base layer frame the current image depends on.
>>       TL0PICIDX MUST be incremented when TID is 0.
> What happens if L=1 but both T=0 and K=0 so that there is no TID value 
> present? Or indeed if T=0 but K=1 so that the TID field is there but 
> 'MUST be ignored by the receiver'  (definition of TID field)?
>
> s4.3: It would be useful to include the section number (9.1) where the 
> uncompressed data chunk specification is found.  I was somewhat 
> surprised that the ordering is reversed in the protocol spec.
>
> s4.3, VER: The receiver should validate this field - currently only 
> values 0-3 are valid.
>
> s4.3, SizeN:  Make it clear these are integer fields: s/in Size0,/in 
> integer fields Size0,/
>
> s4.3 and s4.4:  It would be helpful to implementers to explain what 
> the offsets in the  first partition payload are for the additional 
> partition count and additional partition sizes.  Having rummaged 
> around in RFC  6386, I have to say that I am unclear whether the 
> values are placed after the full chunk of data indicated by 
> 1stPartitionSize or are part of this data. And if the latter is the 
> case, how to work out where the partition sizes are.  I guess that 
> they ought to be at offset (1stPartitionSize+10 - key frame - or 
> 1stPartitionSize+3 - interframe) in the VP8 Payload field as it is 
> difficult to work out where they are without decoding the partition 
> otherwise.  Also exactly how many partition sizes to expect - I think 
> I read in RFC 6386 that the last partition size is implicit. Help!
>
> In the course of clearing this up, it might be appropriate to move the 
> last sentence of the first para of s4.4 and the second para of s4.4 
> into s4.3 as part of the explanation.  This is the relevant text:
>> Note that the length of the first partition can
>> always be obtained from the first partition size parameter in the VP8
>> payload header.
>>
>> The VP8 bitstream format [RFC6386] specifies that if multiple DCT/WHT
>> partitions are produced, the location of each partition start is
>> found at the end of the first (prediction/mode) partition.  In this
>> RTP payload specification, the location offsets are considered to be
>> part of the first partition.
>
>
> s4.4:  I found this section very difficult to parse.  It is a bit 
> 'stream of consciousness' and would benefit from a reordering and 
> rewrite.  Suggestion (assuming the second para gets moved to s4.3):
> NEW SECTION 4.4:
> An encoded VP8 frame can be divided into two or more partitions, as
> described in Section 1.  It is OPTIONAL for a packetizer implementing
> this RTP specification
> to pay attention to the partition boundaries within an encoded frame.
> If packetization of a frame is done without considering the partition
> boundaries, the PID field MAY be set to zero for all packets, and the
> S bit MUST NOT be set to one for any other packet than the first.
>
> If the preferred usage suggested in Section 3 is followed, with each 
> packet
> carrying data from exactly one partition, the S bit and PID fields
> described in Section 4.2 SHOULD be used to indicate what the packet
> contains.  The PID field should indicate which partition the first
> octet of the payload belongs to, and the S bit indicates that the
> packet starts on a new partition.
>
> If the packetizer does not pay attention to the partition boundaries, 
> one
> packet can contain a fragment of a  partition, a complete partition, 
> or an
> aggregate of fragments and partitions.  There is no explicit signaling 
> of
> partition boundaries in the payload and the partition lengths at the 
> end of
> the first partition have to be used to identify the boundaries. 
> Partitions
> MUST be aggregated in decoding order.  Two fragments from different
> partitions MAY be aggregated into the same packet along with one or 
> more
> complete partitions.
>
> In all cases, the payload of a packet MUST contain data from only one 
> video
> frame.  Consequently the set of packets carrying the data from a 
> particular
> frame will contain exactly one VP8 Payload Header (see Section 4.3) 
> carried
> in the first packet of the frame.  The last, or only, packet carrying 
> data for the
> frame MUST have the M bit set in the RTP header.
>
> s4.5.1:  Th algorithm only applies for the RECOMMENDED use case with 
> partitions in separate packets.
> This should be noted.
>
> s5.2, last para: s/only parts of the frame is corrupted./only part of 
> the frame is corrupted./
>
> s6: s/This payload format has two required parameters./This payload 
> format has two optional parameters./
> [I think this is probably a hangover from the previous version.]
>
> s7: s/Cryptographic system may also allow/The cryptographic system may 
> also allow/
>
> s7:
>> the usage of SRTP [RFC3711] is recommended.
> RECOMMENDED?
>
> s10.2: I suspect that RFC 3551 is normative.


From nobody Tue Jul 14 02:48:04 2015
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Cc: General area reviewing team <gen-art@ietf.org>, draft-ietf-payload-vp8.all@tools.ietf.org, payload@ietf.org
Subject: Re: [payload] Gen-art 2nd LC review of draft-ietf-payload-vp8-16 (with summary)
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[cc to payload wg, rather than ietf-discuss]

Some comments inline.=20
Colin


On 13 Jul 2015, at 13:01, Elwyn Davies <elwynd@dial.pipex.com> wrote:
> I am the assigned Gen-ART reviewer for this draft. For background on =
Gen-ART, please see the FAQ at
>=20
> <http://wiki.tools.ietf.org/area/gen/trac/wiki/GenArtfaq>.
>=20
> Please resolve these comments along with any other Last Call comments
> you may receive.
>=20
> Document: draft-ietf-payload-vp8-16.txt
> Reviewer: Elwyn Davies
> Review Date: 2015/07/10
> IETF LC End Date: 2015/07/13
> IESG Telechat date: (if known) -
>=20
> Summary: Gosh, it has been a long time since the last review! Almost =
ready.  Still needs various minor nits fixed and some clarifications.
...
> Nits/editorial comments:
...
> s4.1, last two paras:
>>    Sequence number:  The sequence numbers are monotonically =
increasing
>>       and set as packets are sent.
>>=20
>>       The remaining RTP header fields are used as specified in
>>       [RFC3550].
> Redefining the Sequence Number field seems gratuitous and potentially =
dangerous,
> given that it is *very carefully* defined in RFC 3550 and the overall =
intent does
> not seem any different from that in RFC 3550.  OTOH a note about the =
(non?-)use of the X bit
> and the Payload Type field (PT) would be useful.  I suggest replacing =
the paras above with:
> NEW:
>   X bit: MUST be 0.  The VP8 RTP profile does not use the RTP Header =
Extension capability.

This is not correct. The draft is an RTP Payload Format, not an RTP =
Profile. RTP Payload Formats cannot prohibit the use of RTP header =
extensions.=20

>   PT (Payload Type):  In line with the policy in Section 3 of =
[RFC3551], applications
>      using the VP8 RTP payload profile MUST assign a dynamic payload =
type number to

This cannot be a MUST strength requirement, since the RTP/AVP profile =
[RFC3551] cannot be mandated (other, incompatible, RTP Profile could =
exist).

>      be used in each RTP session and provide a mechanism to indicate =
the mapping.
>      See Section 6.2 for the mechanism to be used with the Session =
Description Protocol (SDP)
>      [RFC4566].
>=20
>      The remaining RTP Fixed Header Fields (V, P, CC, sequence number, =
SSRC and CSRC
>      identfiers) are used as specified in Section 5.1 of [RFC5330].
> END
> Note that this may be considered to make the reference to RFC 3551 =
normative.
>=20
> s4.2, X bit:  There is potential for confusion between the X bit in =
the fixed header
> and the X bit in the VP8 Payload Descriptor.  Perhaps changing this to =
C for control bits
> would avoid the problem.
>=20
> s4.2, M bit: Similarly, it might be better to use B (for BIG) in place =
of M to avoid confusion.

Agree with these two.

...
> s7: s/Cryptographic system may also allow/The cryptographic system may =
also allow/
>=20
> s7:
>> the usage of SRTP [RFC3711] is recommended.
> RECOMMENDED?

No - see RFC 7202. Section A.13 of draft-ietf-payload-rtp-howto-14 has =
suggested text for this.

> s10.2: I suspect that RFC 3551 is normative.

It shouldn't need to be.

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





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Den 29. juni 2015 23:50, skrev Ben Campbell:
> Hi,
> 
> I'm doing a quick re-look at draft-ietf-payload-vp8-16 before starting a
> repeat IETF Last Call. I have a few comments. I don't think any of these
> need delay the last call, so please address them along with any last
> call comments.
> 
> Please bear in mind that I am reconstructing a lot of history here, so I
> apologize in advance if I bring up things that have already been discussed.
> 
> Thanks!
> 
> Ben.
> 
> ------------
> 
> General:
> 
> Ted Lemon and Pete Resnick made some AD comments when they were still
> ADs. Even though they are no longer on the IESG, their comments look
> reasonable. I don't see evidence that they have been addressed--please
> consider addressing them.
> 
> It looks like you have addressed some, but not all, of Stephen Farrell's
> comments. Has there been discussion on why the ones not addressed were
> not addressed?

I'm afraid we missed some :-(

We'll consider these for the next update round - as far as I remember,
they were all of an editorial nature ("this can be stated more clearly"
or "this needs some more justification").

It wasn't immediately clear to us where responses to those comments
should go either, given that they were never posted to the mailing list,
and we can't respond in the tool that was used to keep them.

> (Note that in both of the paragraphs above, I don't mean by "addressed"
> that the draft necessarily needs to change--just that the authors
> respond to the input--even if only to say why they don't agree with a
> comment.)
> 
> -- 4.5 and 4.5.1: It would be helpful to explain why these are listed as
> examples. I assume it's one of those cases where this algorithm gives
> the right results, but others could also do so. If so, some words to
> that effect would be helpful. It might also be helpful to include a few
> words indicating why 4.5.1 is a child of 4.5 rather than a peer. (I
> assume because partition reconstruction is a subset of frame
> reconstruction?)

A tidier format would be to have

4.5 Reconstruction algorithms
4.5.1 Frame reconstruction
4.5.2 Partition reconstruction

The sections were added in version -09 (July 2013), in response to
Richard Barnes' AD review (where he pointed out that it was "mostly
clear", but that it would be nice to make it explicit).

I guess nobody looked hard at the table of content for the next 2 years.


> 
> -- 7:
> 
> I don't see a response to the secdir review ( Secdir review of
> draft-ietf-payload-vp8-08 ). For the SRTP question, please consider
> using the boilerplate in the appendix of draft-ietf-payload-rtp-howto-14
> (section A.13).
> 

The secdir review was another one that we spent considerable time
finding, and then did not address explicitly...

The text about pathological data refers to the fact that some codecs
have algorithms that allow very high complexity operations to be
triggered by a suitably crafted data stream (10x or more beyond the
decoding of an ordinary stream); VP8 does not contain such algorithms,
so people shouldn't be too worried about this threat. It is in fact a
direct quote from draft-ietf-payload-rtp-howto, so it seems a bit
strange that the reviewer hadn't seen that one before.

The security section seems mostly to be a direct copy from
draft-ietf-payload-rtp-howto-00, including the lowercase "recommended".
I like the text in -14 better, so I'll suggest replacing the -00
boilerplate with the -14 boilerplate for the next version. (Any objections?)

We'll get there sooner or later...





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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

all good.

Thanks all

joel



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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

Please consider updating the security considerations to use the
boilerplate from the first paragraph of A.13 of
draft-ietf-payload-rtp-howto-14.



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Hi,

Please send the chairs the presentations for the Payload session by Monday
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Thanks

Roni Even


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vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>Please send the =
chairs the presentations for the Payload session by Monday =
morning<o:p></o:p></p><p class=3DMsoNormal>Thanks<o:p></o:p></p><p =
class=3DMsoNormal>Roni Even<o:p></o:p></p></div></body></html>
------=_NextPart_000_00C4_01D0C2E5.3E0EEFD0--


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [payload] Erratas on RFC 4867 (The AMR RTP Payload Format)
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--------------090403010507040300030401
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AVTCORE and PAYLOAD,

There has been three erratas submitted toward RFC 4867 the AMR RTP 
payload format:
http://www.rfc-editor.org/errata_search.php?rfc=4867

The numbers for the Erratas are 4347, 4348, and 4349.

I am one of the co-authors and I want to resolve these erratas. I am 
sending this to both AVTCORE and PAYLOAD WG due to that the erratas 
arriving on AVTCORE's mailing list, but is more in PAYLOAD's scope. So 
please keep both lists in the reply. Any judgment of the consensus will 
be done by Roni Even.

They are all related to text regarding the usage of the Codec Mode 
Request field. Unclarities and contradictions in the RFC has caused 
interoperability problems. These have been discussed in 3GPP SA4 and 
they have discussed and agreed on this proposal for addressing these 
issues.

The proposed changes can be seen in the errata as well as the 
motivations. I have produced a side-by-side diff for the text changes 
which is attached to make it easier to see what is proposed to be 
changed in the text.

 From my perspective these erratas should be VERIFIED. The reason is 
that the changes proposed appear to be enforcing what I can remember 
being the intention of the document, but being more easier to understand 
and without the conflict. However, there has been substantial amount of 
time since this was specified and people may have other views. Thus, I 
really like to see if someone disagree with that the erratas encode the 
intention and therefore should not be VERIFIED.

It would be good if people could provide any feedback no later than the 
22nd of August.

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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<body >=20
  <table border=3D"0" cellpadding=3D"0" cellspacing=3D"0">=20
  <tr bgcolor=3D"orange"><th></th><th>&nbsp;amr-orig.txt&nbsp;</th><th> </t=
h><th>&nbsp;amr-changed.txt&nbsp;</th><th></th></tr>=20
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">   CM=
R (4 bits): Indicates a codec mode request sent to the speech</td><td> </td=
><td class=3D"right">   CMR (4 bits): Indicates a codec mode request sent t=
o the speech</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
 encoder at the site of the receiver of this payload.  The value of</td><td=
> </td><td class=3D"right">      encoder at the site of the receiver of thi=
s payload.  The value of</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
 the CMR field is set to the frame type index of the corresponding</td><td>=
 </td><td class=3D"right">      the CMR field is set to the frame type inde=
x of the corresponding</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
 speech mode being requested.  The frame type index may be 0-7 for</td><td>=
 </td><td class=3D"right">      speech mode being requested.  The frame typ=
e index may be 0-7 for</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
 AMR, as defined in Table 1a in [2], or 0-8 for AMR-WB, as defined</td><td>=
 </td><td class=3D"right">      AMR, as defined in Table 1a in [2], or 0-8 =
for AMR-WB, as defined</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td><a name=3D"diff0001" /></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
   in Table 1a in [4].  CMR value 15 indicates that no mode <span class=3D"=
delete">request</span></td><td> </td><td class=3D"rblock">      in Table 1a=
 in [4].  CMR value 15 indicates that <span class=3D"insert">the receiver h=
as</span></td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"><sp=
an class=3D"delete">      is present,</span> and other values are for futur=
e use.</td><td> </td><td class=3D"rblock">      no <span class=3D"insert">p=
reference in which</span> mode <span class=3D"insert">within the negotiated=
 mode set to</span></td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"></t=
d><td> </td><td class=3D"rblock"><span class=3D"insert">      receive,</spa=
n> and other values are for future use.</td><td class=3D"lineno" valign=3D"=
top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left"></td>=
<td> </td><td class=3D"right"></td><td class=3D"lineno" valign=3D"top"></td=
></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">   Th=
e codec mode request received in the CMR field is valid until the</td><td> =
</td><td class=3D"right">   The codec mode request received in the CMR fiel=
d is valid until the</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">   ne=
xt codec mode request is received, i.e., a newly received CMR value</td><td=
> </td><td class=3D"right">   next codec mode request is received, i.e., a =
newly received CMR value</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td><a name=3D"diff0002" /></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
corresponding to a speech mode, or <span class=3D"delete">NO_DATA</span> ov=
errides the previously</td><td> </td><td class=3D"rblock">   corresponding =
to a speech mode, or <span class=3D"insert">CMR=3D15</span> overrides the p=
reviously</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
received CMR value corresponding to a speech mode or <span class=3D"delete"=
>NO_DATA.</span></td><td> </td><td class=3D"rblock">   received CMR value c=
orresponding to a speech mode or <span class=3D"insert">CMR=3D15.</span></t=
d><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
Therefore, if a terminal continuously wishes to receive frames <span class=
=3D"delete">in the</span></td><td> </td><td class=3D"rblock">   Therefore, =
if a terminal continuously wishes to receive frames <span class=3D"insert">=
not</span></td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"><sp=
an class=3D"delete">   same</span> mode X, it needs to set CMR=3DX for all =
its outbound payloads, and</td><td> </td><td class=3D"rblock"><span class=
=3D"insert">   higher than</span> mode X, it needs to set CMR=3DX for all i=
ts outbound</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
if a terminal has no preference in which mode to receive, it SHOULD</td><td=
> </td><td class=3D"rblock">   payloads, and if a terminal has no preferenc=
e in which mode <span class=3D"insert">within</span></td><td class=3D"linen=
o" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
set CMR=3D15 in all its outbound payloads.</td><td> </td><td class=3D"rbloc=
k"><span class=3D"insert">   the negotiated mode set</span> to receive, it =
SHOULD set CMR=3D15 in all its</td><td class=3D"lineno" valign=3D"top"></td=
></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"></t=
d><td> </td><td class=3D"rblock">   outbound payloads.</td><td class=3D"lin=
eno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left"></td>=
<td> </td><td class=3D"right"></td><td class=3D"lineno" valign=3D"top"></td=
></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">   If=
 receiving a payload with a CMR value that is not a speech mode or</td><td>=
 </td><td class=3D"right">   If receiving a payload with a CMR value that i=
s not a speech mode or</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td><a name=3D"diff0003" /></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
<span class=3D"delete">NO_DATA</span>, the CMR MUST be ignored by the recei=
ver.</td><td> </td><td class=3D"rblock">   <span class=3D"insert">CMR=3D15<=
/span>, the CMR MUST be ignored by the receiver.</td><td class=3D"lineno" v=
align=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left"></td>=
<td> </td><td class=3D"right"></td><td class=3D"lineno" valign=3D"top"></td=
></tr>
      <tr><td><a name=3D"diff0004" /></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
The encoder <span class=3D"delete">SHOULD</span> follow a received codec mo=
de <span class=3D"delete">request,</span> but MAY</td><td> </td><td class=
=3D"rblock">   The encoder <span class=3D"insert">MUST</span> follow a rece=
ived codec mode <span class=3D"insert">request as soon as</span></td><td cl=
ass=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
change to a lower-numbered mode if it so chooses, for example, to</td><td> =
</td><td class=3D"rblock"><span class=3D"insert">   possible. It SHOULD use=
 the requested mode,</span> but MAY change to a</td><td class=3D"lineno" va=
lign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
control congestion.</td><td> </td><td class=3D"rblock">   lower-numbered mo=
de if it so chooses, for example, to</td><td class=3D"lineno" valign=3D"top=
"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"></t=
d><td> </td><td class=3D"rblock">   control congestion. <span class=3D"inse=
rt">However, the encoder MUST NOT use a</span></td><td class=3D"lineno" val=
ign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"></t=
d><td> </td><td class=3D"rblock"><span class=3D"insert">   higher-numbered =
mode than the received codec mode request.</span></td><td class=3D"lineno" =
valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left"></td>=
<td> </td><td class=3D"right"></td><td class=3D"lineno" valign=3D"top"></td=
></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
 mode-set: Restricts the active codec mode set to a subset of all</td><td> =
</td><td class=3D"right">      mode-set: Restricts the active codec mode se=
t to a subset of all</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          modes, for example, to be able to support transport</td><td> </td=
><td class=3D"right">               modes, for example, to be able to suppo=
rt transport</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          channels such as GSM networks in gateway use cases.</td><td> </td=
><td class=3D"right">               channels such as GSM networks in gatewa=
y use cases.</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          Possible values are a comma separated list of modes from</td><td>=
 </td><td class=3D"right">               Possible values are a comma separa=
ted list of modes from</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          the set: 0,...,7 (see Table 1a [2]).  The SID frame type</td><td>=
 </td><td class=3D"right">               the set: 0,...,7 (see Table 1a [2]=
).  The SID frame type</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          8 and NO_DATA (frame type 15) are never included in the</td><td> =
</td><td class=3D"right">               8 and NO_DATA (frame type 15) are n=
ever included in the</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          mode set, but can always be used.  If mode-set is</td><td> </td><=
td class=3D"right">               mode set, but can always be used.  If mod=
e-set is</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td><a name=3D"diff0005" /></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
            specified, it MUST be abided, <span class=3D"delete">and</span>=
 frames encoded with</td><td> </td><td class=3D"rblock">               spec=
ified, it MUST be abided, <span class=3D"insert">i.e.</span> frames encoded=
 with</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"left">     =
          modes outside of the subset MUST NOT be sent in any RTP</td><td> =
</td><td class=3D"right">               modes outside of the subset MUST NO=
T be sent in any RTP</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td><a name=3D"diff0006" /></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
            payload <span class=3D"delete">or used in</span> codec mode <sp=
an class=3D"delete">requests.</span>  If not present,</td><td> </td><td cla=
ss=3D"rblock">               payload <span class=3D"insert">and</span> code=
c mode <span class=3D"insert">requests MUST only use modes</span></td><td c=
lass=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock">   =
            all codec modes are allowed for the payload type.</td><td> </td=
><td class=3D"rblock"><span class=3D"insert">               within the mode=
-set or CMR=3D15.</span> If <span class=3D"insert">the mode-set parameter</=
span></td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"></t=
d><td> </td><td class=3D"rblock"><span class=3D"insert">               is</=
span> not present, <span class=3D"insert">then</span> all codec modes are a=
llowed for the</td><td class=3D"lineno" valign=3D"top"></td></tr>
      <tr><td class=3D"lineno" valign=3D"top"></td><td class=3D"lblock"></t=
d><td> </td><td class=3D"rblock">               payload type.</td><td class=
=3D"lineno" valign=3D"top"></td></tr>

     <tr><td></td><td class=3D"left"></td><td> </td><td class=3D"right"></t=
d><td></td></tr>
     <tr bgcolor=3D"gray"><th colspan=3D"5" align=3D"center"><a name=3D"end=
">&nbsp;End of changes. 6 change blocks.&nbsp;</a></th></tr>
     <tr class=3D"stats"><td></td><th><i>13 lines changed or deleted</i></t=
h><th><i> </i></th><th><i>17 lines changed or added</i></th><td></td></tr>
     <tr><td colspan=3D"5" align=3D"center" class=3D"small"><br/>This html =
diff was produced by rfcdiff 1.42. The latest version is available from <a =
href=3D"http://www.tools.ietf.org/tools/rfcdiff/" >http://tools.ietf.org/to=
ols/rfcdiff/</a> </td></tr>
   </table>
   </body>
   </html>
X-Generator: pyht 0.35

<!-- args: {'--oldcolour': 'red', '--width': '', 'difftype': '--html', 'fil=
ename2': '   CMR (4 bits): Indicates a codec mode request sent to the speec=
h\r\n      encoder at the site of the receiver of this payload.  The value =
of\r\n      the CMR field is set to the frame type index of the correspondi=
ng\r\n      speech mode being requested.  The frame type index may be 0-7 f=
or\r\n      AMR, as defined in Table 1a in [2], or 0-8 for AMR-WB, as defin=
ed\r\n      in Table 1a in [4].  CMR value 15 indicates that the receiver h=
as\r\n      no preference in which mode within the negotiated mode set to\r=
\n      receive, and other values are for future use.\r\n\r\n   The codec m=
ode request received in the CMR field is valid until the\r\n   next codec m=
ode request is received, i.e., a newly received CMR value\r\n   correspondi=
ng to a speech mode, or CMR=3D15 overrides the previously\r\n   received CM=
R value corresponding to a speech mode or CMR=3D15.\r\n   Therefore, if a t=
erminal continuously wishes to receive frames not \r\n   higher than mode X=
, it needs to set CMR=3DX for all its outbound\r\n   payloads, and if a ter=
minal has no preference in which mode within \r\n   the negotiated mode set=
 to receive, it SHOULD set CMR=3D15 in all its\r\n   outbound payloads.\r\n=
\r\n   If receiving a payload with a CMR value that is not a speech mode or=
\r\n   CMR=3D15, the CMR MUST be ignored by the receiver.\r\n\r\n   The enc=
oder MUST follow a received codec mode request as soon as\r\n   possible. I=
t SHOULD use the requested mode, but MAY change to a\r\n   lower-numbered m=
ode if it so chooses, for example, to\r\n   control congestion. However, th=
e encoder MUST NOT use a\r\n   higher-numbered mode than the received codec=
 mode request.\r\n   \r\n      mode-set: Restricts the active codec mode se=
t to a subset of all\r\n               modes, for example, to be able to su=
pport transport\r\n               channels such as GSM networks in gateway =
use cases.\r\n               Possible values are a comma separated list of =
modes from\r\n               the set: 0,...,7 (see Table 1a [2]).  The SID =
frame type\r\n               8 and NO_DATA (frame type 15) are never includ=
ed in the\r\n               mode set, but can always be used.  If mode-set =
is\r\n               specified, it MUST be abided, i.e. frames encoded with=
\r\n               modes outside of the subset MUST NOT be sent in any RTP\=
r\n               payload and codec mode requests MUST only use modes\r\n  =
             within the mode-set or CMR=3D15. If the mode-set parameter\r\n=
               is not present, then all codec modes are allowed for the\r\n=
               payload type.\r\n\r\n', 'filename1': '   CMR (4 bits): Indic=
ates a codec mode request sent to the speech\r\n      encoder at the site o=
f the receiver of this payload.  The value of\r\n      the CMR field is set=
 to the frame type index of the corresponding\r\n      speech mode being re=
quested.  The frame type index may be 0-7 for\r\n      AMR, as defined in T=
able 1a in [2], or 0-8 for AMR-WB, as defined\r\n      in Table 1a in [4]. =
 CMR value 15 indicates that no mode request\r\n      is present, and other=
 values are for future use.\r\n\r\n   The codec mode request received in th=
e CMR field is valid until the\r\n   next codec mode request is received, i=
.e., a newly received CMR value\r\n   corresponding to a speech mode, or NO=
_DATA overrides the previously\r\n   received CMR value corresponding to a =
speech mode or NO_DATA.\r\n   Therefore, if a terminal continuously wishes =
to receive frames in the\r\n   same mode X, it needs to set CMR=3DX for all=
 its outbound payloads, and\r\n   if a terminal has no preference in which =
mode to receive, it SHOULD\r\n   set CMR=3D15 in all its outbound payloads.=
\r\n\r\n   If receiving a payload with a CMR value that is not a speech mod=
e or\r\n   NO_DATA, the CMR MUST be ignored by the receiver.\r\n\r\n   \r\n=
   The encoder SHOULD follow a received codec mode request, but MAY\r\n   c=
hange to a lower-numbered mode if it so chooses, for example, to\r\n   cont=
rol congestion.\r\n   \r\n      mode-set: Restricts the active codec mode s=
et to a subset of all\r\n               modes, for example, to be able to s=
upport transport\r\n               channels such as GSM networks in gateway=
 use cases.\r\n               Possible values are a comma separated list of=
 modes from\r\n               the set: 0,...,7 (see Table 1a [2]).  The SID=
 frame type\r\n               8 and NO_DATA (frame type 15) are never inclu=
ded in the\r\n               mode set, but can always be used.  If mode-set=
 is\r\n               specified, it MUST be abided, and frames encoded with=
\r\n               modes outside of the subset MUST NOT be sent in any RTP\=
r\n               payload or used in codec mode requests.  If not present,\=
r\n               all codec modes are allowed for the payload type.\r\n', '=
url1': '', 'submit': 'Generate diff', 'url2': '', '--newcolour': 'green'} -=
->=

--------------090403010507040300030401--


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Subject: Re: [payload] Erratas on RFC 4867 (The AMR RTP Payload Format)
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Wed, 22 Jul 2015 09:43:43 +0200
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Subject: Re: [payload] Payload WG notes - Please read
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Below is the outcome of the side meeting: provide corrections if I the
summary is inaccurate.

It was decided that
1. We will need to revisit how FEC operates with PERC-enabled middleboxes.
This is currently a non-goal.
2. an identifier will be in the FEC payload. For now this is SSRC. However,
if a needed this could be replaced by another identifier (e.g., RSID). .

3. keep the encapsulated length as 16 bits. Add the 4 bits of SSRC Counter
elsewhere, this is still TBD.

To summarise the version 3/3 in the slides was accepted.

Cheers,

Varun

On Tue, Jul 21, 2015 at 6:05 PM, Ali C. Begen (abegen) <abegen@cisco.com>
wrote:
>
> Please have a look and send any corrections to the chairs by July 28th.
Presenters, please fill in the gaps denoted by a question mark.
>
> Thanks.
> -acbegen
>
> IETF 93 - Payload Notes
> Chairs: Roni Even, Ali Begen
> Note takers: Stephan Wenger, Mo Zanaty
> Jabber scribe: Jonathan Lennox
>
>
> Payload Status Update Chairs
> https://www.ietf.org/proceedings/93/slides/slides-93-payload-0.pdf
>
> H265: SDP directorate review by Flemming Andreasen noted no overall
> grammar and no formal grammar for fmtp parameters. No interoperability
issues,
> so no change needed, according to Stephan, Magnus Westerlund, others.
Magnus:
> recommendation in how-to: when you have complex parameters, then you
should
> include ABNF. Otherwise not.
> Stephan: there is an ABNF in the draft for one complex parameter.
>
>
> VP9 RTP Payload Format (Magnus Flodman)
> draft-ietf-payload-vp9-00 <
http://tools.ietf.org/id/draft-ietf-payload-vp9-00>
> https://www.ietf.org/proceedings/93/slides/slides-93-payload-3.pdf
>
> Flexible and Non-Flexible GOF (Group of Frames) structure: Mo thinks
> this may be useful for other codecs, authors have no issues with reusing
this.
> Stephan noted that other codecs (e.g. H.264/5) have other means to convey
this
> within the payload itself not in payload headers.
>
>
> Flexible FEC RTP Payload Format (Varun Singh)
> draft-ietf-payload-flexible-fec-scheme-00 <
http://tools.ietf.org/id/draft-ietf-payload-flexible-fec-scheme-00>
> https://www.ietf.org/proceedings/93/slides/slides-93-payload-1.pdf
>
> Bernard Aboba: How can a sender signal specific FEC patterns, like only
> protect the base layer of SRST RTP streams? Authors clarified that no
static
> signaling (e.g. SDP) is necessary since the sender is free to pick dynami=
c
> patterns and all receivers must handle this.
>
> Slide 8: open issue:
> Jonathan: putting SSRC into RTP payload is scary (layer violation)
> Mo: we are already in layer violation due to SN; could go Fecframe model
> Jonathan: ??? something about Perc,
> fec over encrypted payload
>
> Magnus: Dislikes no support for jumbo packets > 4KB. Better to
> support 64KB packets and expand the FEC header. Stephan agrees with
expansion.
>
> Different opinions expressed on whether source flow SSRC(s) should be
> included in the FEC header, or other identifiers like MID or the newly
proposed
> ESID/RSID. Also different views on signaling multiple source flows.
> Result: Need a side meeting to decide what source flow identifier is
> used to associate with a FEC repair flow and how to handle multiple sourc=
e
> flows. Varun will lead and copy the list.
>
>
> MELPe Codec (Roni Even for Victor Demjanenko)
> draft-demjanenko-payload-melpe-03 <
http://tools.ietf.org/id/draft-demjanenko-payload-melpe-03>
> https://www.ietf.org/proceedings/93/slides/slides-93-payload-4.pdf
>
> Stephan has read it and thinks it is a well written RTP payload format
> draft. No other reviewers, so chairs will ask the list to review.
>
>
> Interleaved RTP Payload Format (Rachel Huang)
> draft-huang-payload-rtp-interleave-00 <
http://tools.ietf.org/id/draft-huang-payload-rtp-interleave-00>
> https://www.ietf.org/proceedings/93/slides/slides-93-payload-2.pdf
>
> Jonathan: Note that PT must be the same.
> Rachel: yes
> Jonathan: assumption aggregation never changes number of packets.
> Rachel: yes
> Jonathan: with more flexible sequence number mechanism, you could do
better
> Rachel: yes
> Imed: Interleave an IDR, what=E2=80=99s the gain?
> Rachel: ???
>
>
>
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload

--=20
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<div dir=3D"ltr"><p dir=3D"ltr">Below is the outcome of the side meeting: p=
rovide corrections if I the summary is inaccurate.</p>
<p dir=3D"ltr">It was decided that<br>
1. We will need to revisit how FEC operates with PERC-enabled middleboxes. =
This is currently a non-goal.<br>
2. an identifier will be in the FEC payload. For now this is SSRC. However,=
 if a needed this could be replaced by another identifier (e.g., RSID). .=
=C2=A0</p><p dir=3D"ltr">
3. keep the encapsulated length as 16 bits. Add the 4 bits of SSRC Counter =
elsewhere, this is still TBD. </p>
<p dir=3D"ltr">To summarise the version 3/3 in the slides was accepted.</p>=
<p>Cheers,</p><p>Varun</p>
<p dir=3D"ltr">On Tue, Jul 21, 2015 at 6:05 PM, Ali C. Begen (abegen) &lt;<=
a href=3D"mailto:abegen@cisco.com" target=3D"_blank">abegen@cisco.com</a>&g=
t; wrote:<br>
&gt;<br>
&gt; Please have a look and send any corrections to the chairs by July 28th=
. Presenters, please fill in the gaps denoted by a question mark.<br>
&gt;<br>
&gt; Thanks.<br>
&gt; -acbegen<br>
&gt;<br>
&gt; IETF 93 - Payload Notes<br>
&gt; Chairs: Roni Even, Ali Begen<br>
&gt; Note takers: Stephan Wenger, Mo Zanaty<br>
&gt; Jabber scribe: Jonathan Lennox<br>
&gt;<br>
&gt;<br>
&gt; Payload Status Update Chairs<br>
&gt; <a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-payloa=
d-0.pdf" target=3D"_blank">https://www.ietf.org/proceedings/93/slides/slide=
s-93-payload-0.pdf</a><br>
&gt;<br>
&gt; H265: SDP directorate review by Flemming Andreasen noted no overall<br=
>
&gt; grammar and no formal grammar for fmtp parameters. No interoperability=
 issues,<br>
&gt; so no change needed, according to Stephan, Magnus Westerlund, others. =
Magnus:<br>
&gt; recommendation in how-to: when you have complex parameters, then you s=
hould<br>
&gt; include ABNF. Otherwise not.<br>
&gt; Stephan: there is an ABNF in the draft for one complex parameter.<br>
&gt;<br>
&gt;<br>
&gt; VP9 RTP Payload Format (Magnus Flodman)<br>
&gt; draft-ietf-payload-vp9-00 &lt;<a href=3D"http://tools.ietf.org/id/draf=
t-ietf-payload-vp9-00" target=3D"_blank">http://tools.ietf.org/id/draft-iet=
f-payload-vp9-00</a>&gt;<br>
&gt; <a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-payloa=
d-3.pdf" target=3D"_blank">https://www.ietf.org/proceedings/93/slides/slide=
s-93-payload-3.pdf</a><br>
&gt;<br>
&gt; Flexible and Non-Flexible GOF (Group of Frames) structure: Mo thinks<b=
r>
&gt; this may be useful for other codecs, authors have no issues with reusi=
ng this.<br>
&gt; Stephan noted that other codecs (e.g. H.264/5) have other means to con=
vey this<br>
&gt; within the payload itself not in payload headers.<br>
&gt;<br>
&gt;<br>
&gt; Flexible FEC RTP Payload Format (Varun Singh)<br>
&gt; draft-ietf-payload-flexible-fec-scheme-00 &lt;<a href=3D"http://tools.=
ietf.org/id/draft-ietf-payload-flexible-fec-scheme-00" target=3D"_blank">ht=
tp://tools.ietf.org/id/draft-ietf-payload-flexible-fec-scheme-00</a>&gt;<br=
>
&gt; <a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-payloa=
d-1.pdf" target=3D"_blank">https://www.ietf.org/proceedings/93/slides/slide=
s-93-payload-1.pdf</a><br>
&gt;<br>
&gt; Bernard Aboba: How can a sender signal specific FEC patterns, like onl=
y<br>
&gt; protect the base layer of SRST RTP streams? Authors clarified that no =
static<br>
&gt; signaling (e.g. SDP) is necessary since the sender is free to pick dyn=
amic<br>
&gt; patterns and all receivers must handle this.<br>
&gt;<br>
&gt; Slide 8: open issue:<br>
&gt; Jonathan: putting SSRC into RTP payload is scary (layer violation)<br>
&gt; Mo: we are already in layer violation due to SN; could go Fecframe mod=
el<br>
&gt; Jonathan: ??? something about Perc,<br>
&gt; fec over encrypted payload<br>
&gt;<br>
&gt; Magnus: Dislikes no support for jumbo packets &gt; 4KB. Better to<br>
&gt; support 64KB packets and expand the FEC header. Stephan agrees with ex=
pansion.<br>
&gt;<br>
&gt; Different opinions expressed on whether source flow SSRC(s) should be<=
br>
&gt; included in the FEC header, or other identifiers like MID or the newly=
 proposed<br>
&gt; ESID/RSID. Also different views on signaling multiple source flows.<br=
>
&gt; Result: Need a side meeting to decide what source flow identifier is<b=
r>
&gt; used to associate with a FEC repair flow and how to handle multiple so=
urce<br>
&gt; flows. Varun will lead and copy the list.<br>
&gt;<br>
&gt;<br>
&gt; MELPe Codec (Roni Even for Victor Demjanenko)<br>
&gt; draft-demjanenko-payload-melpe-03 &lt;<a href=3D"http://tools.ietf.org=
/id/draft-demjanenko-payload-melpe-03" target=3D"_blank">http://tools.ietf.=
org/id/draft-demjanenko-payload-melpe-03</a>&gt;<br>
&gt; <a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-payloa=
d-4.pdf" target=3D"_blank">https://www.ietf.org/proceedings/93/slides/slide=
s-93-payload-4.pdf</a><br>
&gt;<br>
&gt; Stephan has read it and thinks it is a well written RTP payload format=
<br>
&gt; draft. No other reviewers, so chairs will ask the list to review.<br>
&gt;<br>
&gt;<br>
&gt; Interleaved RTP Payload Format (Rachel Huang)<br>
&gt; draft-huang-payload-rtp-interleave-00 &lt;<a href=3D"http://tools.ietf=
.org/id/draft-huang-payload-rtp-interleave-00" target=3D"_blank">http://too=
ls.ietf.org/id/draft-huang-payload-rtp-interleave-00</a>&gt;<br>
&gt; <a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-payloa=
d-2.pdf" target=3D"_blank">https://www.ietf.org/proceedings/93/slides/slide=
s-93-payload-2.pdf</a><br>
&gt;<br>
&gt; Jonathan: Note that PT must be the same.<br>
&gt; Rachel: yes<br>
&gt; Jonathan: assumption aggregation never changes number of packets.<br>
&gt; Rachel: yes<br>
&gt; Jonathan: with more flexible sequence number mechanism, you could do b=
etter<br>
&gt; Rachel: yes<br>
&gt; Imed: Interleave an IDR, what=E2=80=99s the gain?<br>
&gt; Rachel: ???<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; payload mailing list<br>
&gt; <a href=3D"mailto:payload@ietf.org" target=3D"_blank">payload@ietf.org=
</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/payload" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/payload</a><br><br></p>
<p dir=3D"ltr">-- <br>
<a href=3D"http://www.callstats.io" target=3D"_blank">http://www.callstats.=
io</a></p>
</div>

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--_000_D0F06B8077E041678930176886AEAF06vidyocom_--


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <payload@ietf.org>
References: <E144AE8E-8DE7-44A7-88B1-00340DEC549D@cisco.com>
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Date: Wed, 22 Jul 2015 12:02:36 +0300
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Subject: Re: [payload] Payload WG notes - Please read
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Hi Ali,
On Melpe, we will ask the list to adopt this work as a WG item since it =
is within the scope of the working group and the current version has all =
the required information based on RTP howto, this was what both Stephan =
and I (as individual) said at the microphone

H.265 does not need any change and can progress

On VP9 - need to decide on payload header or payload - go to the list?

Roni

-----Original Message-----
From: payload [mailto:payload-bounces@ietf.org] On Behalf Of Ali C. =
Begen (abegen)
Sent: Tuesday, July 21, 2015 7:06 PM
To: payload@ietf.org
Subject: [payload] Payload WG notes - Please read

=20
Please have a look and send any corrections to the chairs by July 28th. =
Presenters, please fill in the gaps denoted by a question mark.

Thanks.
-acbegen

IETF 93 - Payload Notes
Chairs: Roni Even, Ali Begen
Note takers: Stephan Wenger, Mo Zanaty
Jabber scribe: Jonathan Lennox
=20

Payload Status Update Chairs
https://www.ietf.org/proceedings/93/slides/slides-93-payload-0.pdf
=20
H265: SDP directorate review by Flemming Andreasen noted no overall =
grammar and no formal grammar for fmtp parameters. No interoperability =
issues, so no change needed, according to Stephan, Magnus Westerlund, =
others. Magnus:
recommendation in how-to: when you have complex parameters, then you =
should include ABNF.  Otherwise not.
Stephan: there is an ABNF in the draft for one complex parameter.
=20

VP9 RTP Payload Format (Magnus Flodman)
draft-ietf-payload-vp9-00 =
<http://tools.ietf.org/id/draft-ietf-payload-vp9-00>
https://www.ietf.org/proceedings/93/slides/slides-93-payload-3.pdf
=20
Flexible and Non-Flexible GOF (Group of Frames) structure: Mo thinks =
this may be useful for other codecs, authors have no issues with reusing =
this.
Stephan noted that other codecs (e.g. H.264/5) have other means to =
convey this within the payload itself not in payload headers.
=20

Flexible FEC RTP Payload Format (Varun Singh)
draft-ietf-payload-flexible-fec-scheme-00 =
<http://tools.ietf.org/id/draft-ietf-payload-flexible-fec-scheme-00>
https://www.ietf.org/proceedings/93/slides/slides-93-payload-1.pdf
=20
Bernard Aboba: How can a sender signal specific FEC patterns, like only =
protect the base layer of SRST RTP streams? Authors clarified that no =
static signaling (e.g. SDP) is necessary since the sender is free to =
pick dynamic patterns and all receivers must handle this.
=20
Slide 8: open issue:=20
Jonathan: putting SSRC into RTP payload is scary (layer violation)
Mo: we are already in layer violation due to SN; could go Fecframe model
Jonathan: ??? something about Perc,
fec over encrypted payload
=20
Magnus: Dislikes no support for jumbo packets > 4KB. Better to support =
64KB packets and expand the FEC header. Stephan agrees with expansion.
=20
Different opinions expressed on whether source flow SSRC(s) should be =
included in the FEC header, or other identifiers like MID or the newly =
proposed ESID/RSID. Also different views on signaling multiple source =
flows.
Result: Need a side meeting to decide what source flow identifier is =
used to associate with a FEC repair flow and how to handle multiple =
source flows. Varun will lead and copy the list.
=20

MELPe Codec (Roni Even for Victor Demjanenko)
draft-demjanenko-payload-melpe-03 =
<http://tools.ietf.org/id/draft-demjanenko-payload-melpe-03>
https://www.ietf.org/proceedings/93/slides/slides-93-payload-4.pdf
=20
Stephan has read it and thinks it is a well written RTP payload format =
draft. No other reviewers, so chairs will ask the list to review.
=20

Interleaved RTP Payload Format (Rachel Huang)
draft-huang-payload-rtp-interleave-00 =
<http://tools.ietf.org/id/draft-huang-payload-rtp-interleave-00>
https://www.ietf.org/proceedings/93/slides/slides-93-payload-2.pdf
=20
Jonathan: Note that PT must be the same.
Rachel: yes
Jonathan: assumption aggregation never changes number of packets. =20
Rachel: yes
Jonathan: with more flexible sequence number mechanism, you could do =
better
Rachel: yes
Imed: Interleave an IDR, what=E2=80=99s the gain?
Rachel: ???



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


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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Wed, 22 Jul 2015 13:13:30 +0200
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On Wed, Jul 22, 2015 at 10:52 AM, Jonathan Lennox <jonathan@vidyo.com> wrot=
e:
>
> On Jul 22, 2015, at 9:43 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:
>
> Below is the outcome of the side meeting: provide corrections if I the
> summary is inaccurate.
>
> It was decided that
> 1. We will need to revisit how FEC operates with PERC-enabled middleboxes=
.
> This is currently a non-goal.
> 2. an identifier will be in the FEC payload. For now this is SSRC. Howeve=
r,
> if a needed this could be replaced by another identifier (e.g., RSID). .
>
> 3. keep the encapsulated length as 16 bits. Add the 4 bits of SSRC Counte=
r
> elsewhere, this is still TBD.
>
> To summarise the version 3/3 in the slides was accepted.
>
> On point 2 =E2=80=94 my opinion in the side meeting was that even if we d=
efine
> identifiers like RSID in the future, the identifier in FlexFEC should
> nonetheless remain an SSRC.  The reason is that the FEC receiver needs to
> know, after an SSRC change, whether the packets being FEC=E2=80=99d come =
from the
> old SSRC or the new SSRC.

If there is agreement on SSRC, I will not put any editor's note or
placeholder for replacing it.

>
> It was also pointed out that the FEC format, as a degenerate case, can al=
so
> perform retransmission (by creating a FEC packet that covers only a singl=
e
> source packet).  This may be able to resolve some of the annoyances of th=
e
> RFC 4588 RTX format, notably in a BUNDLE context =E2=80=94 it avoids payl=
oad type
> explosion (because the source PT is in the packet), and gives you stream
> association automatically.

True, I will add some text related to this. However, if we want to use
this in rtcweb, rtp-usage may want to express an opinion on using this
in addition to RFC4588?


--=20
http://www.callstats.io


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Varun Singh <vsingh.ietf@gmail.com>, Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [payload] Payload WG notes - Please read
Thread-Index: AQHQw88dRc8Bj1w5zkONpq1MAxj7sZ3ncHeAgAATTICAACdRAIAAI9yA
Date: Wed, 22 Jul 2015 11:21:49 +0000
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References: <E144AE8E-8DE7-44A7-88B1-00340DEC549D@cisco.com> <CAEbPqrywC5vPQrcsfDiHtx8avDceKa7A2_KUO-VqQ6V51Kk_rQ@mail.gmail.com> <D0F06B80-77E0-4167-8930-176886AEAF06@vidyo.com> <CAEbPqrxoMbipLjZbeHBU0_0MsXEPM4Db6Zj26dsRF=aTktHAuQ@mail.gmail.com>
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Subject: Re: [payload] Payload WG notes - Please read
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From nobody Wed Jul 22 08:12:04 2015
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From: Meetecho Team <ietf@meetecho.com>
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Subject: [payload] Meetecho recordings of PAYLOAD WG session
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------=_Part_8_1470806841.1437577917629
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Dear all,

the full recording (synchronized video, audio, slides and jabber room) of the 
PAYLOAD WG session at IETF 93 is available at the following URL:
http://ietf93.conf.meetecho.com/index.php/Recorded_Sessions#PAYLOAD

In case of problems with the playout, just drop an e-mail to ietf-support@meetecho.com.

For the chair(s): please feel free to put the link to the recording in the minutes,
if you think this might be useful.

Cheers,
the Meetecho Team



------=_Part_8_1470806841.1437577917629--


From nobody Mon Jul 27 23:59:59 2015
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [payload] RE: Payload WG notes - Please read
Thread-Index: AQHQw88dRc8Bj1w5zkONpq1MAxj7sZ3wPKRg
Date: Tue, 28 Jul 2015 06:58:53 +0000
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Subject: [payload]  RE: Payload WG notes - Please read
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: MMTP vs RTP
Thread-Index: AQHQyU7OlCaxo9Jv/0SckNCtcBKu0w==
Date: Tue, 28 Jul 2015 16:02:29 +0000
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Subject: [payload] MMTP vs RTP
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--_000_D1DD1FD35292Amzanatyciscocom_
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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dbig5">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 12px; font-fami=
ly: Arial, sans-serif;">
<div>MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS=
 for media streaming applications, both real-time and non-real-time. It int=
egrates FEC, buffering, congestion control and other functions. It was pres=
ented in TSVAREA in IETF 93. See
 below for the slides and draft.</div>
<div><a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-tsvare=
a-1.pdf">https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf=
</a></div>
<div><a href=3D"https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp">http=
s://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp</a></div>
<div><br>
</div>
<div>I found slides 5 and 15 particularly relevant for AVT folks, so inlini=
ng them.</div>
<div><br>
</div>
<div>
<div>Why not RTP? (slide 5)</div>
<div>- Lack of &nbsp;Multiplexing</div>
<div>&nbsp; - One media session per component and without RTP multiplexing,=
 2 ports per session&nbsp;</div>
<div>- Server Maintenance</div>
<div>&nbsp; - RTP Payload Format for every new media codec&nbsp;</div>
<div>&nbsp; - Support needs to be added to the media server&nbsp;</div>
<div>?- Coupling of &nbsp;Presentation and Delivery</div>
<div>&nbsp; - RTP carries presentation and synchronization information at t=
he transport level</div>
<div>- Limited support for Non-Real Time Media</div>
<div>&nbsp; - Presentations consist of &nbsp;timed and non-timed media</div=
>
<div>&nbsp; - Need other protocol or countless number of &nbsp;payload form=
ats to support NRT&nbsp;</div>
</div>
<div><br>
</div>
<div>
<div>Why are we here? (slide 15)</div>
<div>- We want to develop MMTP further in the IETF</div>
<div>- We want to address the Internet (unicast and Multicast)</div>
<div>- We want to reuse existing components such as congestion control and =
security</div>
<div>- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...</div>
<div>?- Can we revive rmt?</div>
<div>?- Can we start a BoF or a new ad-hoc group?&nbsp;</div>
<div>?- Or can we do an informational RFC?</div>
</div>
<div><br>
</div>
<div>I think there should be some dialogue on RTP evolution with the MMTP f=
olks. Some interesting points are raised in this work, such as generic pack=
etization vs. specific RTP payload formats. Perhaps a generic payload draft=
 can address this generic packetization
 (i.e. fragmentation and perhaps aggregation) in the absence of a specific =
RTP payload format for the elementary media stream.</div>
<div><br>
</div>
<div>Thanks to Gorry for bringing this to my attention.</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
</body>
</html>

--_000_D1DD1FD35292Amzanatyciscocom_--


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, "avt@ietf.org" <avt@ietf.org>,  "payload@ietf.org" <payload@ietf.org>
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, "avt@ietf.org" <avt@ietf.org>,  "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] MMTP vs RTP
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Date: Tue, 28 Jul 2015 16:15:32 +0000
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Subject: Re: [payload] [AVTCORE] MMTP vs RTP
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I assumed TSVAREA was already aware of this since it was presented there
and will be in their notes. I wanted to bring this to the attention of RTP
folks that may be interested but probably missed this. If there are any
replies here that may be useful for TSV, I will forward. I try to avoid
cross-posting to lists with significantly different topics and subscribers.

Mo

On 7/28/15, 12:05 PM, Ali C. Begen (abegen) <abegen@cisco.com> wrote:
Is not this thread supposed to cc the transport area, too?

-----Original Message-----
From: avt on behalf of "Mo Zanaty (mzanaty)"
Date: Tuesday, July 28, 2015 at 7:02 PM
To: "avt@ietf.org", "payload@ietf.org"
Cc: "gorry@erg.abdn.ac.uk"
Subject: [AVTCORE] MMTP vs RTP

>MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS
>for media streaming applications, both real-time and non-real-time. It
>integrates FEC, buffering, congestion control and other functions. It was
>presented in TSVAREA in IETF 93. See
> below for the slides and draft.
>https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf
>https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp
>
>I found slides 5 and 15 particularly relevant for AVT folks, so inlining
>them.
>
>Why not RTP? (slide 5)
>- Lack of  Multiplexing
>  - One media session per component and without RTP multiplexing, 2 ports
>per session=20
>- Server Maintenance
>  - RTP Payload Format for every new media codec
>  - Support needs to be added to the media server
>- Coupling of  Presentation and Delivery
>  - RTP carries presentation and synchronization information at the
>transport level
>- Limited support for Non-Real Time Media
>  - Presentations consist of  timed and non-timed media
>  - Need other protocol or countless number of  payload formats to
>support NRT
>
>Why are we here? (slide 15)
>- We want to develop MMTP further in the IETF
>- We want to address the Internet (unicast and Multicast)
>- We want to reuse existing components such as congestion control and
>security
>- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
>- Can we revive rmt?
>- Can we start a BoF or a new ad-hoc group?
>- Or can we do an informational RFC?
>
>I think there should be some dialogue on RTP evolution with the MMTP
>folks. Some interesting points are raised in this work, such as generic
>packetization vs. specific RTP payload formats. Perhaps a generic payload
>draft can address this generic packetization
> (i.e. fragmentation and perhaps aggregation) in the absence of a
>specific RTP payload format for the elementary media stream.
>
>Thanks to Gorry for bringing this to my attention.
>
>Mo


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Speer <michael.speer@pluribusnetworks.com>
Thread-Topic: [AVTCORE] MMTP vs RTP
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Subject: Re: [payload] [AVTCORE] MMTP vs RTP
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This is not my work. I just became aware of it in the last meeting. This wo=
rk would replace RTP itself (RFC 3550) and all RTP payload formats (many sp=
ecs, e.g. RFC 6184 for H.264). The MMT payload format would be the elementa=
ry stream as defined by the codec spec for storage in ISOBMFF containers, a=
ugmented with MMT specs and features such as generic packetization / fragme=
ntation.

Mo


On 7/28/15, 2:05 PM, Michael Speer <michael.speer@pluribusnetworks.com<mail=
to:michael.speer@pluribusnetworks.com>> wrote:

Mo,

Hi, can you post the RFCs you trying to replace?  In particular, what RTP p=
ayload
specification are you trying to replace?

Cheers,
Michael


On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzanaty) <mzanaty@cisco.com<mai=
lto:mzanaty@cisco.com>> wrote:
I assumed TSVAREA was already aware of this since it was presented there
and will be in their notes. I wanted to bring this to the attention of RTP
folks that may be interested but probably missed this. If there are any
replies here that may be useful for TSV, I will forward. I try to avoid
cross-posting to lists with significantly different topics and subscribers.

Mo

On 7/28/15, 12:05 PM, Ali C. Begen (abegen) <abegen@cisco.com<mailto:abegen=
@cisco.com>> wrote:
Is not this thread supposed to cc the transport area, too?

-----Original Message-----
From: avt on behalf of "Mo Zanaty (mzanaty)"
Date: Tuesday, July 28, 2015 at 7:02 PM
To: "avt@ietf.org<mailto:avt@ietf.org>", "payload@ietf.org<mailto:payload@i=
etf.org>"
Cc: "gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>"
Subject: [AVTCORE] MMTP vs RTP

>MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS
>for media streaming applications, both real-time and non-real-time. It
>integrates FEC, buffering, congestion control and other functions. It was
>presented in TSVAREA in IETF 93. See
> below for the slides and draft.
>https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf
>https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp
>
>I found slides 5 and 15 particularly relevant for AVT folks, so inlining
>them.
>
>Why not RTP? (slide 5)
>- Lack of  Multiplexing
>  - One media session per component and without RTP multiplexing, 2 ports
>per session
>- Server Maintenance
>  - RTP Payload Format for every new media codec
>  - Support needs to be added to the media server
>- Coupling of  Presentation and Delivery
>  - RTP carries presentation and synchronization information at the
>transport level
>- Limited support for Non-Real Time Media
>  - Presentations consist of  timed and non-timed media
>  - Need other protocol or countless number of  payload formats to
>support NRT
>
>Why are we here? (slide 15)
>- We want to develop MMTP further in the IETF
>- We want to address the Internet (unicast and Multicast)
>- We want to reuse existing components such as congestion control and
>security
>- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
>- Can we revive rmt?
>- Can we start a BoF or a new ad-hoc group?
>- Or can we do an informational RFC?
>
>I think there should be some dialogue on RTP evolution with the MMTP
>folks. Some interesting points are raised in this work, such as generic
>packetization vs. specific RTP payload formats. Perhaps a generic payload
>draft can address this generic packetization
> (i.e. fragmentation and perhaps aggregation) in the absence of a
>specific RTP payload format for the elementary media stream.
>
>Thanks to Gorry for bringing this to my attention.
>
>Mo

_______________________________________________
Audio/Video Transport Core Maintenance
avt@ietf.org<mailto:avt@ietf.org>
https://www.ietf.org/mailman/listinfo/avt


--_000_D1DD3EDB5295Bmzanatyciscocom_
Content-Type: text/html; charset="us-ascii"
Content-ID: <5FA6CA2AA36D00498CC4E909C43ECF87@emea.cisco.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 12px; font-fami=
ly: Arial, sans-serif;">
<div>This is not my work. I just became aware of it in the last meeting. Th=
is work would replace RTP itself (RFC 3550) and all RTP payload formats (ma=
ny specs, e.g. RFC 6184 for H.264). The MMT payload format would be the ele=
mentary stream as defined by the
 codec spec for storage in ISOBMFF containers, augmented with MMT specs and=
 features such as generic packetization / fragmentation.</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>On 7/28/15, 2:05 PM, Michael Speer &lt;<a href=3D"mailto:michael.speer=
@pluribusnetworks.com">michael.speer@pluribusnetworks.com</a>&gt; wrote:</d=
iv>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">
<div>
<div>
<div>
<div>Mo,<br>
<br>
</div>
Hi, can you post the RFCs you trying to replace?&nbsp; In particular, what =
RTP payload<br>
</div>
specification are you trying to replace?<br>
<br>
</div>
Cheers,<br>
</div>
Michael<br>
<br>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzan=
aty) <span dir=3D"ltr">
&lt;<a href=3D"mailto:mzanaty@cisco.com" target=3D"_blank">mzanaty@cisco.co=
m</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I assumed TSVAREA was already aware of this since it was presented there<br=
>
and will be in their notes. I wanted to bring this to the attention of RTP<=
br>
folks that may be interested but probably missed this. If there are any<br>
replies here that may be useful for TSV, I will forward. I try to avoid<br>
cross-posting to lists with significantly different topics and subscribers.=
<br>
<br>
Mo<br>
<span class=3D""><br>
On 7/28/15, 12:05 PM, Ali C. Begen (abegen) &lt;<a href=3D"mailto:abegen@ci=
sco.com">abegen@cisco.com</a>&gt; wrote:<br>
Is not this thread supposed to cc the transport area, too?<br>
<br>
-----Original Message-----<br>
From: avt on behalf of &quot;Mo Zanaty (mzanaty)&quot;<br>
Date: Tuesday, July 28, 2015 at 7:02 PM<br>
To: &quot;<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>&quot;, &quot;<a =
href=3D"mailto:payload@ietf.org">payload@ietf.org</a>&quot;<br>
Cc: &quot;<a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&=
quot;<br>
Subject: [AVTCORE] MMTP vs RTP<br>
<br>
&gt;MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS<=
br>
&gt;for media streaming applications, both real-time and non-real-time. It<=
br>
&gt;integrates FEC, buffering, congestion control and other functions. It w=
as<br>
&gt;presented in TSVAREA in IETF 93. See<br>
&gt; below for the slides and draft.<br>
&gt;<a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea=
-1.pdf" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/proceedin=
gs/93/slides/slides-93-tsvarea-1.pdf</a><br>
&gt;<a href=3D"https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp" rel=
=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/draft-bouaziz=
i-tsvwg-mmtp</a><br>
&gt;<br>
&gt;I found slides 5 and 15 particularly relevant for AVT folks, so inlinin=
g<br>
&gt;them.<br>
&gt;<br>
&gt;Why not RTP? (slide 5)<br>
&gt;- Lack of&nbsp; Multiplexing<br>
&gt;&nbsp; - One media session per component and without RTP multiplexing, =
2 ports<br>
&gt;per session<br>
&gt;- Server Maintenance<br>
&gt;&nbsp; - RTP Payload Format for every new media codec<br>
&gt;&nbsp; - Support needs to be added to the media server<br>
</span><span class=3D"">&gt;- Coupling of&nbsp; Presentation and Delivery<b=
r>
&gt;&nbsp; - RTP carries presentation and synchronization information at th=
e<br>
&gt;transport level<br>
&gt;- Limited support for Non-Real Time Media<br>
&gt;&nbsp; - Presentations consist of&nbsp; timed and non-timed media<br>
&gt;&nbsp; - Need other protocol or countless number of&nbsp; payload forma=
ts to<br>
&gt;support NRT<br>
&gt;<br>
&gt;Why are we here? (slide 15)<br>
&gt;- We want to develop MMTP further in the IETF<br>
&gt;- We want to address the Internet (unicast and Multicast)<br>
&gt;- We want to reuse existing components such as congestion control and<b=
r>
&gt;security<br>
&gt;- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...<br>
</span>&gt;- Can we revive rmt?<br>
<span class=3D"im HOEnZb">&gt;- Can we start a BoF or a new ad-hoc group?<b=
r>
</span>
<div class=3D"HOEnZb">
<div class=3D"h5">&gt;- Or can we do an informational RFC?<br>
&gt;<br>
&gt;I think there should be some dialogue on RTP evolution with the MMTP<br=
>
&gt;folks. Some interesting points are raised in this work, such as generic=
<br>
&gt;packetization vs. specific RTP payload formats. Perhaps a generic paylo=
ad<br>
&gt;draft can address this generic packetization<br>
&gt; (i.e. fragmentation and perhaps aggregation) in the absence of a<br>
&gt;specific RTP payload format for the elementary media stream.<br>
&gt;<br>
&gt;Thanks to Gorry for bringing this to my attention.<br>
&gt;<br>
&gt;Mo<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" rel=3D"noreferrer" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/avt</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D1DD3EDB5295Bmzanatyciscocom_--


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From: Thomas Edwards <Thomas.Edwards@fox.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Michael Speer <michael.speer@pluribusnetworks.com>
Thread-Topic: [payload] [AVTCORE] MMTP vs RTP
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Subject: Re: [payload] [AVTCORE] MMTP vs RTP
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I will add that MMT is under study by the ATSC TG3/S33 "Specialist Group on=
 Management and Protocols" to be a transport in the ATSC 3.0 digital broadc=
ast television standard.

It is not yet clear to me if MMT is a good candidate to replace all uses of=
 RTP, but it certainly appears to be a good candidate to replace the use of=
 MPEG Transport Streams for delivery of completed linear content over IP sy=
stems.

-Thomas

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


From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com<mailto:mzanaty@cisco.com>>
Date: Tuesday, July 28, 2015 at 11:22 AM
To: Michael Speer <michael.speer@pluribusnetworks.com<mailto:michael.speer@=
pluribusnetworks.com>>
Cc: "gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>" <gorry@erg.abdn.ac.=
uk<mailto:gorry@erg.abdn.ac.uk>>, "avt@ietf.org<mailto:avt@ietf.org>" <avt@=
ietf.org<mailto:avt@ietf.org>>, "payload@ietf.org<mailto:payload@ietf.org>"=
 <payload@ietf.org<mailto:payload@ietf.org>>
Subject: Re: [payload] [AVTCORE] MMTP vs RTP

This is not my work. I just became aware of it in the last meeting. This wo=
rk would replace RTP itself (RFC 3550) and all RTP payload formats (many sp=
ecs, e.g. RFC 6184 for H.264). The MMT payload format would be the elementa=
ry stream as defined by the codec spec for storage in ISOBMFF containers, a=
ugmented with MMT specs and features such as generic packetization / fragme=
ntation.

Mo


On 7/28/15, 2:05 PM, Michael Speer <michael.speer@pluribusnetworks.com<mail=
to:michael.speer@pluribusnetworks.com>> wrote:

Mo,

Hi, can you post the RFCs you trying to replace?  In particular, what RTP p=
ayload
specification are you trying to replace?

Cheers,
Michael


On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzanaty) <mzanaty@cisco.com<mai=
lto:mzanaty@cisco.com>> wrote:
I assumed TSVAREA was already aware of this since it was presented there
and will be in their notes. I wanted to bring this to the attention of RTP
folks that may be interested but probably missed this. If there are any
replies here that may be useful for TSV, I will forward. I try to avoid
cross-posting to lists with significantly different topics and subscribers.

Mo

On 7/28/15, 12:05 PM, Ali C. Begen (abegen) <abegen@cisco.com<mailto:abegen=
@cisco.com>> wrote:
Is not this thread supposed to cc the transport area, too?

-----Original Message-----
From: avt on behalf of "Mo Zanaty (mzanaty)"
Date: Tuesday, July 28, 2015 at 7:02 PM
To: "avt@ietf.org<mailto:avt@ietf.org>", "payload@ietf.org<mailto:payload@i=
etf.org>"
Cc: "gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>"
Subject: [AVTCORE] MMTP vs RTP

>MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS
>for media streaming applications, both real-time and non-real-time. It
>integrates FEC, buffering, congestion control and other functions. It was
>presented in TSVAREA in IETF 93. See
> below for the slides and draft.
>https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf<https:/=
/urldefense.proofpoint.com/v2/url?u=3Dhttps-3A__www.ietf.org_proceedings_93=
_slides_slides-2D93-2Dtsvarea-2D1.pdf&d=3DAwMFAg&c=3Duw6TLu4hwhHdiGJOgwcWD4=
AjKQx6zvFcGEsbfiY9-EI&r=3DlekNOOM5noV61zrPH3rwPyhtNnLLWoLEHgd0quQxly8&m=3Dt=
BjpXMFtLuZk1jwHWIGqof6K2uO7Wx-KbcsnpO6Pvps&s=3DNxulWK2WFT1-IulMBMEZG0IfB3xG=
ui12ThP4r70yv_Y&e=3D>
>https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp<https://urldefense.p=
roofpoint.com/v2/url?u=3Dhttps-3A__tools.ietf.org_html_draft-2Dbouazizi-2Dt=
svwg-2Dmmtp&d=3DAwMFAg&c=3Duw6TLu4hwhHdiGJOgwcWD4AjKQx6zvFcGEsbfiY9-EI&r=3D=
lekNOOM5noV61zrPH3rwPyhtNnLLWoLEHgd0quQxly8&m=3DtBjpXMFtLuZk1jwHWIGqof6K2uO=
7Wx-KbcsnpO6Pvps&s=3DaOIhGhYRg-c1eGOuhU0NEoUhK_oXj5JGAbgopI1nIRk&e=3D>
>
>I found slides 5 and 15 particularly relevant for AVT folks, so inlining
>them.
>
>Why not RTP? (slide 5)
>- Lack of  Multiplexing
>  - One media session per component and without RTP multiplexing, 2 ports
>per session
>- Server Maintenance
>  - RTP Payload Format for every new media codec
>  - Support needs to be added to the media server
>- Coupling of  Presentation and Delivery
>  - RTP carries presentation and synchronization information at the
>transport level
>- Limited support for Non-Real Time Media
>  - Presentations consist of  timed and non-timed media
>  - Need other protocol or countless number of  payload formats to
>support NRT
>
>Why are we here? (slide 15)
>- We want to develop MMTP further in the IETF
>- We want to address the Internet (unicast and Multicast)
>- We want to reuse existing components such as congestion control and
>security
>- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
>- Can we revive rmt?
>- Can we start a BoF or a new ad-hoc group?
>- Or can we do an informational RFC?
>
>I think there should be some dialogue on RTP evolution with the MMTP
>folks. Some interesting points are raised in this work, such as generic
>packetization vs. specific RTP payload formats. Perhaps a generic payload
>draft can address this generic packetization
> (i.e. fragmentation and perhaps aggregation) in the absence of a
>specific RTP payload format for the elementary media stream.
>
>Thanks to Gorry for bringing this to my attention.
>
>Mo

_______________________________________________
Audio/Video Transport Core Maintenance
avt@ietf.org<mailto:avt@ietf.org>
https://www.ietf.org/mailman/listinfo/avt<https://urldefense.proofpoint.com=
/v2/url?u=3Dhttps-3A__www.ietf.org_mailman_listinfo_avt&d=3DAwMFAg&c=3Duw6T=
Lu4hwhHdiGJOgwcWD4AjKQx6zvFcGEsbfiY9-EI&r=3DlekNOOM5noV61zrPH3rwPyhtNnLLWoL=
EHgd0quQxly8&m=3DtBjpXMFtLuZk1jwHWIGqof6K2uO7Wx-KbcsnpO6Pvps&s=3DYCsKHXdmo4=
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>
<div>I will add that MMT is under study by the ATSC TG3/S33 &quot;Specialis=
t Group on Management and Protocols&quot; to be a transport in the ATSC 3.0=
 digital broadcast television standard.</div>
<div><br>
</div>
<div>It is not yet clear to me if MMT is a good candidate to replace all us=
es of RTP, but it certainly appears to be a good candidate to replace the u=
se of MPEG Transport Streams for delivery of completed linear content over =
IP systems.</div>
<div><br>
</div>
<div>-Thomas</div>
<div><br>
</div>
<div>
<div>--&nbsp;</div>
<div>Thomas Edwards<span class=3D"Apple-tab-span" style=3D"white-space:pre"=
> </span></div>
<div>VP Engineering &amp; Development</div>
<div>FOX Networks Engineering and Operations</div>
<div>thomas.edwards@fox.com</div>
<div>Phone: &#43;1.310.369.6696</div>
<div>10201 West Pico Blvd.</div>
<div>Los Angeles, CA 90035</div>
<div><br>
</div>
</div>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>&quot;Mo Zanaty (mzanaty)&quo=
t; &lt;<a href=3D"mailto:mzanaty@cisco.com">mzanaty@cisco.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Tuesday, July 28, 2015 at 11:=
22 AM<br>
<span style=3D"font-weight:bold">To: </span>Michael Speer &lt;<a href=3D"ma=
ilto:michael.speer@pluribusnetworks.com">michael.speer@pluribusnetworks.com=
</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a href=3D"mailto:gorry@e=
rg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&quot; &lt;<a href=3D"mailto:gorry@e=
rg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&gt;, &quot;<a href=3D"mailto:avt@ie=
tf.org">avt@ietf.org</a>&quot; &lt;<a href=3D"mailto:avt@ietf.org">avt@ietf=
.org</a>&gt;,
 &quot;<a href=3D"mailto:payload@ietf.org">payload@ietf.org</a>&quot; &lt;<=
a href=3D"mailto:payload@ietf.org">payload@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [payload] [AVTCORE] MM=
TP vs RTP<br>
</div>
<div><br>
</div>
<div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; color: rgb(0, 0, 0); font-size: 12px; font-famil=
y: Arial, sans-serif;">
<div>This is not my work. I just became aware of it in the last meeting. Th=
is work would replace RTP itself (RFC 3550) and all RTP payload formats (ma=
ny specs, e.g. RFC 6184 for H.264). The MMT payload format would be the ele=
mentary stream as defined by the
 codec spec for storage in ISOBMFF containers, augmented with MMT specs and=
 features such as generic packetization / fragmentation.</div>
<div><br>
</div>
<div>Mo</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>On 7/28/15, 2:05 PM, Michael Speer &lt;<a href=3D"mailto:michael.speer=
@pluribusnetworks.com">michael.speer@pluribusnetworks.com</a>&gt; wrote:</d=
iv>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">
<div>
<div>
<div>
<div>Mo,<br>
<br>
</div>
Hi, can you post the RFCs you trying to replace?&nbsp; In particular, what =
RTP payload<br>
</div>
specification are you trying to replace?<br>
<br>
</div>
Cheers,<br>
</div>
Michael<br>
<br>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzan=
aty) <span dir=3D"ltr">
&lt;<a href=3D"mailto:mzanaty@cisco.com" target=3D"_blank">mzanaty@cisco.co=
m</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I assumed TSVAREA was already aware of this since it was presented there<br=
>
and will be in their notes. I wanted to bring this to the attention of RTP<=
br>
folks that may be interested but probably missed this. If there are any<br>
replies here that may be useful for TSV, I will forward. I try to avoid<br>
cross-posting to lists with significantly different topics and subscribers.=
<br>
<br>
Mo<br>
<span class=3D""><br>
On 7/28/15, 12:05 PM, Ali C. Begen (abegen) &lt;<a href=3D"mailto:abegen@ci=
sco.com">abegen@cisco.com</a>&gt; wrote:<br>
Is not this thread supposed to cc the transport area, too?<br>
<br>
-----Original Message-----<br>
From: avt on behalf of &quot;Mo Zanaty (mzanaty)&quot;<br>
Date: Tuesday, July 28, 2015 at 7:02 PM<br>
To: &quot;<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>&quot;, &quot;<a =
href=3D"mailto:payload@ietf.org">payload@ietf.org</a>&quot;<br>
Cc: &quot;<a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&=
quot;<br>
Subject: [AVTCORE] MMTP vs RTP<br>
<br>
&gt;MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS<=
br>
&gt;for media streaming applications, both real-time and non-real-time. It<=
br>
&gt;integrates FEC, buffering, congestion control and other functions. It w=
as<br>
&gt;presented in TSVAREA in IETF 93. See<br>
&gt; below for the slides and draft.<br>
&gt;<a href=3D"https://urldefense.proofpoint.com/v2/url?u=3Dhttps-3A__www.i=
etf.org_proceedings_93_slides_slides-2D93-2Dtsvarea-2D1.pdf&amp;d=3DAwMFAg&=
amp;c=3Duw6TLu4hwhHdiGJOgwcWD4AjKQx6zvFcGEsbfiY9-EI&amp;r=3DlekNOOM5noV61zr=
PH3rwPyhtNnLLWoLEHgd0quQxly8&amp;m=3DtBjpXMFtLuZk1jwHWIGqof6K2uO7Wx-KbcsnpO=
6Pvps&amp;s=3DNxulWK2WFT1-IulMBMEZG0IfB3xGui12ThP4r70yv_Y&amp;e=3D" rel=3D"=
noreferrer" target=3D"_blank">https://www.ietf.org/proceedings/93/slides/sl=
ides-93-tsvarea-1.pdf</a><br>
&gt;<a href=3D"https://urldefense.proofpoint.com/v2/url?u=3Dhttps-3A__tools=
.ietf.org_html_draft-2Dbouazizi-2Dtsvwg-2Dmmtp&amp;d=3DAwMFAg&amp;c=3Duw6TL=
u4hwhHdiGJOgwcWD4AjKQx6zvFcGEsbfiY9-EI&amp;r=3DlekNOOM5noV61zrPH3rwPyhtNnLL=
WoLEHgd0quQxly8&amp;m=3DtBjpXMFtLuZk1jwHWIGqof6K2uO7Wx-KbcsnpO6Pvps&amp;s=
=3DaOIhGhYRg-c1eGOuhU0NEoUhK_oXj5JGAbgopI1nIRk&amp;e=3D" rel=3D"noreferrer"=
 target=3D"_blank">https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp</a=
><br>
&gt;<br>
&gt;I found slides 5 and 15 particularly relevant for AVT folks, so inlinin=
g<br>
&gt;them.<br>
&gt;<br>
&gt;Why not RTP? (slide 5)<br>
&gt;- Lack of&nbsp; Multiplexing<br>
&gt;&nbsp; - One media session per component and without RTP multiplexing, =
2 ports<br>
&gt;per session<br>
&gt;- Server Maintenance<br>
&gt;&nbsp; - RTP Payload Format for every new media codec<br>
&gt;&nbsp; - Support needs to be added to the media server<br>
</span><span class=3D"">&gt;- Coupling of&nbsp; Presentation and Delivery<b=
r>
&gt;&nbsp; - RTP carries presentation and synchronization information at th=
e<br>
&gt;transport level<br>
&gt;- Limited support for Non-Real Time Media<br>
&gt;&nbsp; - Presentations consist of&nbsp; timed and non-timed media<br>
&gt;&nbsp; - Need other protocol or countless number of&nbsp; payload forma=
ts to<br>
&gt;support NRT<br>
&gt;<br>
&gt;Why are we here? (slide 15)<br>
&gt;- We want to develop MMTP further in the IETF<br>
&gt;- We want to address the Internet (unicast and Multicast)<br>
&gt;- We want to reuse existing components such as congestion control and<b=
r>
&gt;security<br>
&gt;- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...<br>
</span>&gt;- Can we revive rmt?<br>
<span class=3D"im HOEnZb">&gt;- Can we start a BoF or a new ad-hoc group?<b=
r>
</span>
<div class=3D"HOEnZb">
<div class=3D"h5">&gt;- Or can we do an informational RFC?<br>
&gt;<br>
&gt;I think there should be some dialogue on RTP evolution with the MMTP<br=
>
&gt;folks. Some interesting points are raised in this work, such as generic=
<br>
&gt;packetization vs. specific RTP payload formats. Perhaps a generic paylo=
ad<br>
&gt;draft can address this generic packetization<br>
&gt; (i.e. fragmentation and perhaps aggregation) in the absence of a<br>
&gt;specific RTP payload format for the elementary media stream.<br>
&gt;<br>
&gt;Thanks to Gorry for bringing this to my attention.<br>
&gt;<br>
&gt;Mo<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
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org_mailman_listinfo_avt&amp;d=3DAwMFAg&amp;c=3Duw6TLu4hwhHdiGJOgwcWD4AjKQx=
6zvFcGEsbfiY9-EI&amp;r=3DlekNOOM5noV61zrPH3rwPyhtNnLLWoLEHgd0quQxly8&amp;m=
=3DtBjpXMFtLuZk1jwHWIGqof6K2uO7Wx-KbcsnpO6Pvps&amp;s=3DYCsKHXdmo4T7QaGfbqI2=
eF0rgv_ZQ_jk9LFwhTVQwJ0&amp;e=3D" rel=3D"noreferrer" target=3D"_blank">http=
s://www.ietf.org/mailman/listinfo/avt</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</span></div>
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Mo,

Hi, can you post the RFCs you trying to replace?  In particular, what RTP
payload
specification are you trying to replace?

Cheers,
Michael


On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzanaty) <mzanaty@cisco.com>
wrote:

> I assumed TSVAREA was already aware of this since it was presented there
> and will be in their notes. I wanted to bring this to the attention of RTP
> folks that may be interested but probably missed this. If there are any
> replies here that may be useful for TSV, I will forward. I try to avoid
> cross-posting to lists with significantly different topics and subscribers.
>
> Mo
>
> On 7/28/15, 12:05 PM, Ali C. Begen (abegen) <abegen@cisco.com> wrote:
> Is not this thread supposed to cc the transport area, too?
>
> -----Original Message-----
> From: avt on behalf of "Mo Zanaty (mzanaty)"
> Date: Tuesday, July 28, 2015 at 7:02 PM
> To: "avt@ietf.org", "payload@ietf.org"
> Cc: "gorry@erg.abdn.ac.uk"
> Subject: [AVTCORE] MMTP vs RTP
>
> >MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS
> >for media streaming applications, both real-time and non-real-time. It
> >integrates FEC, buffering, congestion control and other functions. It was
> >presented in TSVAREA in IETF 93. See
> > below for the slides and draft.
> >https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf
> >https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp
> >
> >I found slides 5 and 15 particularly relevant for AVT folks, so inlining
> >them.
> >
> >Why not RTP? (slide 5)
> >- Lack of  Multiplexing
> >  - One media session per component and without RTP multiplexing, 2 ports
> >per session
> >- Server Maintenance
> >  - RTP Payload Format for every new media codec
> >  - Support needs to be added to the media server
> >- Coupling of  Presentation and Delivery
> >  - RTP carries presentation and synchronization information at the
> >transport level
> >- Limited support for Non-Real Time Media
> >  - Presentations consist of  timed and non-timed media
> >  - Need other protocol or countless number of  payload formats to
> >support NRT
> >
> >Why are we here? (slide 15)
> >- We want to develop MMTP further in the IETF
> >- We want to address the Internet (unicast and Multicast)
> >- We want to reuse existing components such as congestion control and
> >security
> >- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
> >- Can we revive rmt?
> >- Can we start a BoF or a new ad-hoc group?
> >- Or can we do an informational RFC?
> >
> >I think there should be some dialogue on RTP evolution with the MMTP
> >folks. Some interesting points are raised in this work, such as generic
> >packetization vs. specific RTP payload formats. Perhaps a generic payload
> >draft can address this generic packetization
> > (i.e. fragmentation and perhaps aggregation) in the absence of a
> >specific RTP payload format for the elementary media stream.
> >
> >Thanks to Gorry for bringing this to my attention.
> >
> >Mo
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--001a113ede9243d66e051bf35064
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div>Mo,<br><br></div>Hi, can you post the =
RFCs you trying to replace?=C2=A0 In particular, what RTP payload<br></div>=
specification are you trying to replace?<br><br></div>Cheers,<br></div>Mich=
ael<br><br></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzanaty) <span dir=3D"ltr">&lt;=
<a href=3D"mailto:mzanaty@cisco.com" target=3D"_blank">mzanaty@cisco.com</a=
>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I assumed TSVAREA was=
 already aware of this since it was presented there<br>
and will be in their notes. I wanted to bring this to the attention of RTP<=
br>
folks that may be interested but probably missed this. If there are any<br>
replies here that may be useful for TSV, I will forward. I try to avoid<br>
cross-posting to lists with significantly different topics and subscribers.=
<br>
<br>
Mo<br>
<span class=3D""><br>
On 7/28/15, 12:05 PM, Ali C. Begen (abegen) &lt;<a href=3D"mailto:abegen@ci=
sco.com">abegen@cisco.com</a>&gt; wrote:<br>
Is not this thread supposed to cc the transport area, too?<br>
<br>
-----Original Message-----<br>
From: avt on behalf of &quot;Mo Zanaty (mzanaty)&quot;<br>
Date: Tuesday, July 28, 2015 at 7:02 PM<br>
To: &quot;<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>&quot;, &quot;<a =
href=3D"mailto:payload@ietf.org">payload@ietf.org</a>&quot;<br>
Cc: &quot;<a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&=
quot;<br>
Subject: [AVTCORE] MMTP vs RTP<br>
<br>
&gt;MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS<=
br>
&gt;for media streaming applications, both real-time and non-real-time. It<=
br>
&gt;integrates FEC, buffering, congestion control and other functions. It w=
as<br>
&gt;presented in TSVAREA in IETF 93. See<br>
&gt; below for the slides and draft.<br>
&gt;<a href=3D"https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea=
-1.pdf" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/proceedin=
gs/93/slides/slides-93-tsvarea-1.pdf</a><br>
&gt;<a href=3D"https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp" rel=
=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/draft-bouaziz=
i-tsvwg-mmtp</a><br>
&gt;<br>
&gt;I found slides 5 and 15 particularly relevant for AVT folks, so inlinin=
g<br>
&gt;them.<br>
&gt;<br>
&gt;Why not RTP? (slide 5)<br>
&gt;- Lack of=C2=A0 Multiplexing<br>
&gt;=C2=A0 - One media session per component and without RTP multiplexing, =
2 ports<br>
&gt;per session<br>
&gt;- Server Maintenance<br>
&gt;=C2=A0 - RTP Payload Format for every new media codec<br>
&gt;=C2=A0 - Support needs to be added to the media server<br>
</span><span class=3D"">&gt;- Coupling of=C2=A0 Presentation and Delivery<b=
r>
&gt;=C2=A0 - RTP carries presentation and synchronization information at th=
e<br>
&gt;transport level<br>
&gt;- Limited support for Non-Real Time Media<br>
&gt;=C2=A0 - Presentations consist of=C2=A0 timed and non-timed media<br>
&gt;=C2=A0 - Need other protocol or countless number of=C2=A0 payload forma=
ts to<br>
&gt;support NRT<br>
&gt;<br>
&gt;Why are we here? (slide 15)<br>
&gt;- We want to develop MMTP further in the IETF<br>
&gt;- We want to address the Internet (unicast and Multicast)<br>
&gt;- We want to reuse existing components such as congestion control and<b=
r>
&gt;security<br>
&gt;- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...<br>
</span>&gt;- Can we revive rmt?<br>
<span class=3D"im HOEnZb">&gt;- Can we start a BoF or a new ad-hoc group?<b=
r>
</span><div class=3D"HOEnZb"><div class=3D"h5">&gt;- Or can we do an inform=
ational RFC?<br>
&gt;<br>
&gt;I think there should be some dialogue on RTP evolution with the MMTP<br=
>
&gt;folks. Some interesting points are raised in this work, such as generic=
<br>
&gt;packetization vs. specific RTP payload formats. Perhaps a generic paylo=
ad<br>
&gt;draft can address this generic packetization<br>
&gt; (i.e. fragmentation and perhaps aggregation) in the absence of a<br>
&gt;specific RTP payload format for the elementary media stream.<br>
&gt;<br>
&gt;Thanks to Gorry for bringing this to my attention.<br>
&gt;<br>
&gt;Mo<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" rel=3D"noreferrer" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/avt</a><br>
</div></div></blockquote></div><br></div>

--001a113ede9243d66e051bf35064--


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From: Michael Speer <michael.speer@pluribusnetworks.com>
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Subject: Re: [payload] [AVTCORE] MMTP vs RTP
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All,

I agree with you that it=E2=80=99s not clear if MMT is a good candidate =
to replace all uses of RTP. When we worked on MPEG-TS Payload format
years ago, there where a number of challenges including the fact MPEG-TS =
was not really suited to packet networks such as IP.   My
main question is whether MMT overcomes many of packetization issues we =
dealt with at the time.  RTP does quite well for a variety
of applications as we all know, so let=E2=80=99s be careful about use of =
the words replace.   For MPEG streams, it may well be the case, this
improves the situation for some applications.

The relevant combination of RFCs are to provide context are RFC 3360, =
RFC 2250, RFC 5691 and RFC 3640.  The last
three RFCs provide MPEG packetization history.

Cheers,
Michael


> On Jul 28, 2015, at 12:00 PM, Thomas Edwards <Thomas.Edwards@fox.com> =
wrote:
>=20
> I will add that MMT is under study by the ATSC TG3/S33 "Specialist =
Group on Management and Protocols" to be a transport in the ATSC 3.0 =
digital broadcast television standard.
>=20
> It is not yet clear to me if MMT is a good candidate to replace all =
uses of RTP, but it certainly appears to be a good candidate to replace =
the use of MPEG Transport Streams for delivery of completed linear =
content over IP systems.
>=20
> -Thomas
>=20
> --=20
> Thomas Edwards=20
> VP Engineering & Development
> FOX Networks Engineering and Operations
> thomas.edwards@fox.com
> Phone: +1.310.369.6696
> 10201 West Pico Blvd.
> Los Angeles, CA 90035
>=20
>=20
> From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
> Date: Tuesday, July 28, 2015 at 11:22 AM
> To: Michael Speer <michael.speer@pluribusnetworks.com>
> Cc: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "avt@ietf.org" =
<avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
> Subject: Re: [payload] [AVTCORE] MMTP vs RTP
>=20
> This is not my work. I just became aware of it in the last meeting. =
This work would replace RTP itself (RFC 3550) and all RTP payload =
formats (many specs, e.g. RFC 6184 for H.264). The MMT payload format =
would be the elementary stream as defined by the codec spec for storage =
in ISOBMFF containers, augmented with MMT specs and features such as =
generic packetization / fragmentation.
>=20
> Mo
>=20
>=20
> On 7/28/15, 2:05 PM, Michael Speer =
<michael.speer@pluribusnetworks.com> wrote:
>=20
> Mo,
>=20
> Hi, can you post the RFCs you trying to replace?  In particular, what =
RTP payload
> specification are you trying to replace?
>=20
> Cheers,
> Michael
>=20
>=20
> On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzanaty) =
<mzanaty@cisco.com> wrote:
>> I assumed TSVAREA was already aware of this since it was presented =
there
>> and will be in their notes. I wanted to bring this to the attention =
of RTP
>> folks that may be interested but probably missed this. If there are =
any
>> replies here that may be useful for TSV, I will forward. I try to =
avoid
>> cross-posting to lists with significantly different topics and =
subscribers.
>>=20
>> Mo
>>=20
>> On 7/28/15, 12:05 PM, Ali C. Begen (abegen) <abegen@cisco.com> wrote:
>> Is not this thread supposed to cc the transport area, too?
>>=20
>> -----Original Message-----
>> From: avt on behalf of "Mo Zanaty (mzanaty)"
>> Date: Tuesday, July 28, 2015 at 7:02 PM
>> To: "avt@ietf.org", "payload@ietf.org"
>> Cc: "gorry@erg.abdn.ac.uk"
>> Subject: [AVTCORE] MMTP vs RTP
>>=20
>> >MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 =
TS
>> >for media streaming applications, both real-time and non-real-time. =
It
>> >integrates FEC, buffering, congestion control and other functions. =
It was
>> >presented in TSVAREA in IETF 93. See
>> > below for the slides and draft.
>> >https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf
>> >https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp
>> >
>> >I found slides 5 and 15 particularly relevant for AVT folks, so =
inlining
>> >them.
>> >
>> >Why not RTP? (slide 5)
>> >- Lack of  Multiplexing
>> >  - One media session per component and without RTP multiplexing, 2 =
ports
>> >per session
>> >- Server Maintenance
>> >  - RTP Payload Format for every new media codec
>> >  - Support needs to be added to the media server
>> >- Coupling of  Presentation and Delivery
>> >  - RTP carries presentation and synchronization information at the
>> >transport level
>> >- Limited support for Non-Real Time Media
>> >  - Presentations consist of  timed and non-timed media
>> >  - Need other protocol or countless number of  payload formats to
>> >support NRT
>> >
>> >Why are we here? (slide 15)
>> >- We want to develop MMTP further in the IETF
>> >- We want to address the Internet (unicast and Multicast)
>> >- We want to reuse existing components such as congestion control =
and
>> >security
>> >- A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
>> >- Can we revive rmt?
>> >- Can we start a BoF or a new ad-hoc group?
>> >- Or can we do an informational RFC?
>> >
>> >I think there should be some dialogue on RTP evolution with the MMTP
>> >folks. Some interesting points are raised in this work, such as =
generic
>> >packetization vs. specific RTP payload formats. Perhaps a generic =
payload
>> >draft can address this generic packetization
>> > (i.e. fragmentation and perhaps aggregation) in the absence of a
>> >specific RTP payload format for the elementary media stream.
>> >
>> >Thanks to Gorry for bringing this to my attention.
>> >
>> >Mo
>>=20
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20


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DLP-Filter: Pass
X-CFilter-Loop: Reflected
Archived-At: <http://mailarchive.ietf.org/arch/msg/payload/0OQeePaHgryMizbp7xzsuCdz6iY>
Cc: "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [payload] [AVTCORE] MMTP vs RTP
X-BeenThere: payload@ietf.org
X-Mailman-Version: 2.1.15
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Reply-To: kyunghun.jung@samsung.com
List-Id: Audio/Video Transport Payloads working group discussion list <payload.ietf.org>
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X-List-Received-Date: Tue, 28 Jul 2015 23:54:08 -0000

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From nobody Wed Jul 29 08:04:15 2015
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Date: Wed, 29 Jul 2015 10:02:29 +0100
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [payload] [AVTCORE] MMTP vs RTP
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I can offer a little context here (but not the presenter):

A proposal was received to the TSV Area that asked if the IETF would be 
able to work on transport methods for the new MMT specification. This 
IETF meeting we had a very short "heads up" presentation in the TSV-Area 
meeting in Prague to open this discussion (see proceedings).

I udnerstand the first version of the MMT specs had been published, 
together with considerations about how to transmit this directly over 
broadcast TV multiplexes. Thought had also been given on how to go 
beyond the MPEG-2 Transport Stream concepts to provide much more 
flexible packetisation and multiplexing, and to target IP networks - 
they also want to work with hybird networks that combine multiple 
delivery methods.

The presentation I saw at the IETF suggested this work was open to IETF 
inputs - expecting methods for real-time delivery and a carousel-based 
reliable transport framework. Although there are Specs, and RTP was 
already evaluated and found to not be directly suitable, I heard that 
there is a hope to define the new methods within the IETF. I can't speak 
for all involved, I'm currently only watching what goes on, but I think 
this may be a case of not bringing a transport spec to the IETF, but 
more of bringing a multimedia architecture and asking how to transport 
it. It's early stages.

Gorry
(TSVWG Co-Chair)

On 28/07/2015 23:40, Michael Speer wrote:
> All,
>
> I agree with you that it’s not clear if MMT is a good candidate to replace all uses of RTP. When we worked on MPEG-TS Payload format
> years ago, there where a number of challenges including the fact MPEG-TS was not really suited to packet networks such as IP.   My
> main question is whether MMT overcomes many of packetization issues we dealt with at the time.  RTP does quite well for a variety
> of applications as we all know, so let’s be careful about use of the words replace.   For MPEG streams, it may well be the case, this
> improves the situation for some applications.
>
> The relevant combination of RFCs are to provide context are RFC 3360, RFC 2250, RFC 5691 and RFC 3640.  The last
> three RFCs provide MPEG packetization history.
>
> Cheers,
> Michael
>
>
>> On Jul 28, 2015, at 12:00 PM, Thomas Edwards<Thomas.Edwards@fox.com>  wrote:
>>
>> I will add that MMT is under study by the ATSC TG3/S33 "Specialist Group on Management and Protocols" to be a transport in the ATSC 3.0 digital broadcast television standard.
>>
>> It is not yet clear to me if MMT is a good candidate to replace all uses of RTP, but it certainly appears to be a good candidate to replace the use of MPEG Transport Streams for delivery of completed linear content over IP systems.
>>
>> -Thomas
>>
>> -- 
>> Thomas Edwards
>> VP Engineering&  Development
>> FOX Networks Engineering and Operations
>> thomas.edwards@fox.com
>> Phone: +1.310.369.6696
>> 10201 West Pico Blvd.
>> Los Angeles, CA 90035
>>
>>
>> From: "Mo Zanaty (mzanaty)"<mzanaty@cisco.com>
>> Date: Tuesday, July 28, 2015 at 11:22 AM
>> To: Michael Speer<michael.speer@pluribusnetworks.com>
>> Cc: "gorry@erg.abdn.ac.uk"<gorry@erg.abdn.ac.uk>, "avt@ietf.org"<avt@ietf.org>, "payload@ietf.org"<payload@ietf.org>
>> Subject: Re: [payload] [AVTCORE] MMTP vs RTP
>>
>> This is not my work. I just became aware of it in the last meeting. This work would replace RTP itself (RFC 3550) and all RTP payload formats (many specs, e.g. RFC 6184 for H.264). The MMT payload format would be the elementary stream as defined by the codec spec for storage in ISOBMFF containers, augmented with MMT specs and features such as generic packetization / fragmentation.
>>
>> Mo
>>
>>
>> On 7/28/15, 2:05 PM, Michael Speer<michael.speer@pluribusnetworks.com>  wrote:
>>
>> Mo,
>>
>> Hi, can you post the RFCs you trying to replace?  In particular, what RTP payload
>> specification are you trying to replace?
>>
>> Cheers,
>> Michael
>>
>>
>> On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty (mzanaty)<mzanaty@cisco.com>  wrote:
>>> I assumed TSVAREA was already aware of this since it was presented there
>>> and will be in their notes. I wanted to bring this to the attention of RTP
>>> folks that may be interested but probably missed this. If there are any
>>> replies here that may be useful for TSV, I will forward. I try to avoid
>>> cross-posting to lists with significantly different topics and subscribers.
>>>
>>> Mo
>>>
>>> On 7/28/15, 12:05 PM, Ali C. Begen (abegen)<abegen@cisco.com>  wrote:
>>> Is not this thread supposed to cc the transport area, too?
>>>
>>> -----Original Message-----
>>> From: avt on behalf of "Mo Zanaty (mzanaty)"
>>> Date: Tuesday, July 28, 2015 at 7:02 PM
>>> To: "avt@ietf.org", "payload@ietf.org"
>>> Cc: "gorry@erg.abdn.ac.uk"
>>> Subject: [AVTCORE] MMTP vs RTP
>>>
>>>> MMTP (MPEG Media Transport Protocol) aims to replace RTP and MPEG-2 TS
>>>> for media streaming applications, both real-time and non-real-time. It
>>>> integrates FEC, buffering, congestion control and other functions. It was
>>>> presented in TSVAREA in IETF 93. See
>>>> below for the slides and draft.
>>>> https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf
>>>> https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp
>>>>
>>>> I found slides 5 and 15 particularly relevant for AVT folks, so inlining
>>>> them.
>>>>
>>>> Why not RTP? (slide 5)
>>>> - Lack of  Multiplexing
>>>>   - One media session per component and without RTP multiplexing, 2 ports
>>>> per session
>>>> - Server Maintenance
>>>>   - RTP Payload Format for every new media codec
>>>>   - Support needs to be added to the media server
>>>> - Coupling of  Presentation and Delivery
>>>>   - RTP carries presentation and synchronization information at the
>>>> transport level
>>>> - Limited support for Non-Real Time Media
>>>>   - Presentations consist of  timed and non-timed media
>>>>   - Need other protocol or countless number of  payload formats to
>>>> support NRT
>>>>
>>>> Why are we here? (slide 15)
>>>> - We want to develop MMTP further in the IETF
>>>> - We want to address the Internet (unicast and Multicast)
>>>> - We want to reuse existing components such as congestion control and
>>>> security
>>>> - A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
>>>> - Can we revive rmt?
>>>> - Can we start a BoF or a new ad-hoc group?
>>>> - Or can we do an informational RFC?
>>>>
>>>> I think there should be some dialogue on RTP evolution with the MMTP
>>>> folks. Some interesting points are raised in this work, such as generic
>>>> packetization vs. specific RTP payload formats. Perhaps a generic payload
>>>> draft can address this generic packetization
>>>> (i.e. fragmentation and perhaps aggregation) in the absence of a
>>>> specific RTP payload format for the elementary media stream.
>>>>
>>>> Thanks to Gorry for bringing this to my attention.
>>>>
>>>> Mo
>>> _______________________________________________
>>> Audio/Video Transport Core Maintenance
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt


From nobody Wed Jul 29 08:04:16 2015
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To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, Michael Speer <michael.speer@pluribusnetworks.com>
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From nobody Wed Jul 29 08:18:30 2015
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From: Michael Speer <michael.speer@pluribusnetworks.com>
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Subject: Re: [payload] [AVTCORE]   MMTP vs RTP
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All,

Thank you for the bit of background.   I agree that convergence to a =
non-prorprietary solution that does the job is a
very good thing! I just wanting to provide the background of how we got =
to where we are now to ensure that whatever
new transport that is created overcomes the shortcomings of the past =
with respect to transport of MPEG based media.
If MPEG-DASH provides relief from many of the issues we had in the past =
and the broadcasters and multicasters
are embracing an all IP delivery model, then this is huge progress since =
the time we developed RTP and the companion
payload specifications.

Cheers,
Michael

> On Jul 29, 2015, at 6:52 AM, Bill Ver Steeg (versteb) =
<versteb@cisco.com> wrote:
>=20
> I have seen lots of activity in the MPEG DASH space in this context. =
As many you know, the world is converging on delivery of adaptive =
bitrate video using HTTP Adaptive Streaming. There are lots of =
proprietary methods, but there seems to be a general consensus that MPEG =
DASH is the eventual convergence point. This, IMHO, is a very good =
thing.
>=20
> So, the video broadcast world is noticing this, and trying to skate to =
where the puck is going. They are all moving to get rid of their naked =
MPEG2TS video flows and move to an "all IP" delivery model. The various =
broadcasters/multicasters (SCTE, ATSC, CableLabs, DVB, etc) are =
embracing the delivery of chunked MPEG DASH video over their deliver =
networks. If there is enough commonality between the various delivery =
methods, a lot of very interesting functionality is enabled. The most =
obvious revolve around several devices watching an event and getting =
there streams from various paths. You can also envision hybrid error =
recovery, the use of more than one network to deliver a single stream, =
various DVR/CDN schemes, etc.
>=20
> By having a convergence point at the "video chunk" layer, you get a =
lot of what you need. Once this is in place, it would be really nice to =
have a common way to deliver these chunks. In the unicast world, that is =
HTTP. It works great, and we see the uptake. As the broadcasters embrace =
this technology, they recognize the need for a multicast/broadcast =
friendly transport that would provide the baseline interoperability =
between the new "all IP" video delivery model and the HTTP based =
methods.
>=20
> Obviously, the way to do the design is to have a general purpose way =
to deliver objects over several types of networks, and consider MPEG =
DASH chunks as one type of object. We do not want to design an MPEG-DASH =
specific solution, but would design a general purpose system that was =
suitable for delivering objects that look a lot like DASH chunks.
>=20
> This is where the IETF comes in. IMHO, the expertise to tackle this =
problem lives at the IETF. The other SDOs are currently going down their =
own paths here, and are starting to recognize the merits of a common =
transport layer. Without discussing the relative merits of =
RTP/MMPT/FLUTE/MMT/ROUTE/whatever, I think it is important to have the =
IETF take this work up and "do the right thing". Once we get the right =
folks engaged with the right charter, I think we can come up with a =
design that satisfies the requirements and allows the convergence that =
is required to make the most compelling use cases practical.
>=20
> Bill VerSteeg=20
>=20
>=20
> -----Original Message-----
> From: avt [mailto:avt-bounces@ietf.org] On Behalf Of Gorry Fairhurst
> Sent: Wednesday, July 29, 2015 5:02 AM
> To: Michael Speer
> Cc: payload@ietf.org; Thomas Edwards; avt@ietf.org
> Subject: Re: [AVTCORE] [payload] MMTP vs RTP
>=20
> I can offer a little context here (but not the presenter):
>=20
> A proposal was received to the TSV Area that asked if the IETF would =
be able to work on transport methods for the new MMT specification. This =
IETF meeting we had a very short "heads up" presentation in the TSV-Area =
meeting in Prague to open this discussion (see proceedings).
>=20
> I udnerstand the first version of the MMT specs had been published, =
together with considerations about how to transmit this directly over =
broadcast TV multiplexes. Thought had also been given on how to go =
beyond the MPEG-2 Transport Stream concepts to provide much more =
flexible packetisation and multiplexing, and to target IP networks - =
they also want to work with hybird networks that combine multiple =
delivery methods.
>=20
> The presentation I saw at the IETF suggested this work was open to =
IETF inputs - expecting methods for real-time delivery and a =
carousel-based reliable transport framework. Although there are Specs, =
and RTP was already evaluated and found to not be directly suitable, I =
heard that there is a hope to define the new methods within the IETF. I =
can't speak for all involved, I'm currently only watching what goes on, =
but I think this may be a case of not bringing a transport spec to the =
IETF, but more of bringing a multimedia architecture and asking how to =
transport it. It's early stages.
>=20
> Gorry
> (TSVWG Co-Chair)
>=20
> On 28/07/2015 23:40, Michael Speer wrote:
>> All,
>>=20
>> I agree with you that it=E2=80=99s not clear if MMT is a good =
candidate to replace all uses of RTP. When we worked on MPEG-TS Payload =
format
>> years ago, there where a number of challenges including the fact =
MPEG-TS was not really suited to packet networks such as IP.   My
>> main question is whether MMT overcomes many of packetization issues =
we dealt with at the time.  RTP does quite well for a variety
>> of applications as we all know, so let=E2=80=99s be careful about use =
of the words replace.   For MPEG streams, it may well be the case, this
>> improves the situation for some applications.
>>=20
>> The relevant combination of RFCs are to provide context are RFC 3360,=20=

>> RFC 2250, RFC 5691 and RFC 3640.  The last three RFCs provide MPEG =
packetization history.
>>=20
>> Cheers,
>> Michael
>>=20
>>=20
>>> On Jul 28, 2015, at 12:00 PM, Thomas Edwards<Thomas.Edwards@fox.com> =
 wrote:
>>>=20
>>> I will add that MMT is under study by the ATSC TG3/S33 "Specialist =
Group on Management and Protocols" to be a transport in the ATSC 3.0 =
digital broadcast television standard.
>>>=20
>>> It is not yet clear to me if MMT is a good candidate to replace all =
uses of RTP, but it certainly appears to be a good candidate to replace =
the use of MPEG Transport Streams for delivery of completed linear =
content over IP systems.
>>>=20
>>> -Thomas
>>>=20
>>> --
>>> Thomas Edwards
>>> VP Engineering&  Development
>>> FOX Networks Engineering and Operations thomas.edwards@fox.com
>>> Phone: +1.310.369.6696
>>> 10201 West Pico Blvd.
>>> Los Angeles, CA 90035
>>>=20
>>>=20
>>> From: "Mo Zanaty (mzanaty)"<mzanaty@cisco.com>
>>> Date: Tuesday, July 28, 2015 at 11:22 AM
>>> To: Michael Speer<michael.speer@pluribusnetworks.com>
>>> Cc: "gorry@erg.abdn.ac.uk"<gorry@erg.abdn.ac.uk>,=20
>>> "avt@ietf.org"<avt@ietf.org>, "payload@ietf.org"<payload@ietf.org>
>>> Subject: Re: [payload] [AVTCORE] MMTP vs RTP
>>>=20
>>> This is not my work. I just became aware of it in the last meeting. =
This work would replace RTP itself (RFC 3550) and all RTP payload =
formats (many specs, e.g. RFC 6184 for H.264). The MMT payload format =
would be the elementary stream as defined by the codec spec for storage =
in ISOBMFF containers, augmented with MMT specs and features such as =
generic packetization / fragmentation.
>>>=20
>>> Mo
>>>=20
>>>=20
>>> On 7/28/15, 2:05 PM, Michael =
Speer<michael.speer@pluribusnetworks.com>  wrote:
>>>=20
>>> Mo,
>>>=20
>>> Hi, can you post the RFCs you trying to replace?  In particular, =
what=20
>>> RTP payload specification are you trying to replace?
>>>=20
>>> Cheers,
>>> Michael
>>>=20
>>>=20
>>> On Tue, Jul 28, 2015 at 9:15 AM, Mo Zanaty =
(mzanaty)<mzanaty@cisco.com>  wrote:
>>>> I assumed TSVAREA was already aware of this since it was presented=20=

>>>> there and will be in their notes. I wanted to bring this to the=20
>>>> attention of RTP folks that may be interested but probably missed=20=

>>>> this. If there are any replies here that may be useful for TSV, I=20=

>>>> will forward. I try to avoid cross-posting to lists with =
significantly different topics and subscribers.
>>>>=20
>>>> Mo
>>>>=20
>>>> On 7/28/15, 12:05 PM, Ali C. Begen (abegen)<abegen@cisco.com>  =
wrote:
>>>> Is not this thread supposed to cc the transport area, too?
>>>>=20
>>>> -----Original Message-----
>>>> From: avt on behalf of "Mo Zanaty (mzanaty)"
>>>> Date: Tuesday, July 28, 2015 at 7:02 PM
>>>> To: "avt@ietf.org", "payload@ietf.org"
>>>> Cc: "gorry@erg.abdn.ac.uk"
>>>> Subject: [AVTCORE] MMTP vs RTP
>>>>=20
>>>>> MMTP (MPEG Media Transport Protocol) aims to replace RTP and =
MPEG-2=20
>>>>> TS for media streaming applications, both real-time and=20
>>>>> non-real-time. It integrates FEC, buffering, congestion control =
and=20
>>>>> other functions. It was presented in TSVAREA in IETF 93. See below=20=

>>>>> for the slides and draft.
>>>>> https://www.ietf.org/proceedings/93/slides/slides-93-tsvarea-1.pdf
>>>>> https://tools.ietf.org/html/draft-bouazizi-tsvwg-mmtp
>>>>>=20
>>>>> I found slides 5 and 15 particularly relevant for AVT folks, so=20
>>>>> inlining them.
>>>>>=20
>>>>> Why not RTP? (slide 5)
>>>>> - Lack of  Multiplexing
>>>>>  - One media session per component and without RTP multiplexing, 2=20=

>>>>> ports per session
>>>>> - Server Maintenance
>>>>>  - RTP Payload Format for every new media codec
>>>>>  - Support needs to be added to the media server
>>>>> - Coupling of  Presentation and Delivery
>>>>>  - RTP carries presentation and synchronization information at the=20=

>>>>> transport level
>>>>> - Limited support for Non-Real Time Media
>>>>>  - Presentations consist of  timed and non-timed media
>>>>>  - Need other protocol or countless number of  payload formats to=20=

>>>>> support NRT
>>>>>=20
>>>>> Why are we here? (slide 15)
>>>>> - We want to develop MMTP further in the IETF
>>>>> - We want to address the Internet (unicast and Multicast)
>>>>> - We want to reuse existing components such as congestion control=20=

>>>>> and security
>>>>> - A protocol is needed by many SDOs: MPEG, ATSC, 3GPP, DVB, ...
>>>>> - Can we revive rmt?
>>>>> - Can we start a BoF or a new ad-hoc group?
>>>>> - Or can we do an informational RFC?
>>>>>=20
>>>>> I think there should be some dialogue on RTP evolution with the=20
>>>>> MMTP folks. Some interesting points are raised in this work, such=20=

>>>>> as generic packetization vs. specific RTP payload formats. Perhaps=20=

>>>>> a generic payload draft can address this generic packetization=20
>>>>> (i.e. fragmentation and perhaps aggregation) in the absence of a=20=

>>>>> specific RTP payload format for the elementary media stream.
>>>>>=20
>>>>> Thanks to Gorry for bringing this to my attention.
>>>>>=20
>>>>> Mo
>>>> _______________________________________________
>>>> Audio/Video Transport Core Maintenance avt@ietf.org=20
>>>> https://www.ietf.org/mailman/listinfo/avt
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From nobody Wed Jul 29 14:59:01 2015
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From: "Ben Campbell" <ben@nostrum.com>
To: "Elwyn Davies" <elwynd@dial.pipex.com>, draft-ietf-payload-vp8.all@ietf.org, payload-chairs@ietf.org, "Harald Alvestrand" <harald@alvestrand.no>
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Subject: Re: [payload] Gen-art 2nd LC review of draft-ietf-payload-vp8-16
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Hi VP8 Authors:

Has there been a response to Elwyn's Gen-ART review from the authors? If 
so, I haven't seen it. (I did see Colin's response).

Based Elwyn's review, and on Harald's response to my earlier questions, 
I assume that the authors will wish to update the draft before it goes 
back to an IESG telechat. If that's not the case, please let me know. 
Please be advised that tomorrow (July 30) is the nominal deadline for 
getting things on the August 6 telechat.

Thanks!

Ben.


On 13 Jul 2015, at 15:57, Ben Campbell wrote:

> (+payload and Harald)
>
> Elwyn: Thanks for the detailed review!
>
> Authors: Please address Elwyn's comments at your earliest convenience. 
> There's quite a bit here, and I assume we will need a new revision 
> before taking this to the IESG telechat, but it would be helpful to 
> discuss any proposed changes via email before submitting.
>
> Thanks!
>
> Ben.
>
>
> On 13 Jul 2015, at 6:58, Elwyn Davies wrote:
>
>> I am the assigned Gen-ART reviewer for this draft. For background on
>> Gen-ART, please see the FAQ at
>>
>> <http://wiki.tools.ietf.org/area/gen/trac/wiki/GenArtfaq>.
>>
>> Please resolve these comments along with any other Last Call comments
>> you may receive.
>>
>> Document: draft-ietf-payload-vp8-16.txt
>> Reviewer: Elwyn Davies
>> Review Date: 2015/07/10
>> IETF LC End Date: 2015/07/13
>> IESG Telechat date: (if known) -
>>
>> Summary: Gosh, it has been a long time since the last review!
>>
>> Major issues:
>>
>> Minor issues:
>> s4.2: Various of the frame indices either SHOULD or, in the case of
>> KEYIDX, MAY start at random values.  The rationale(s) for these
>> requirements are not explained - nor is the reason why it it is only 
>> a
>> MAY for KEYIDX whereas it is SHOULD for PictureID and what might 
>> happen
>> were the MAY/SHOULD to be ignored.
>>
>> Is the rationale something that should be mentioned in the security
>> considerations?
>>
>> s10.1:  The downref to RFC 6386 identified by idnits was duly noted 
>> in the last call
>> announcement.  I don't have a problem with the downref.
>>
>> Nits/editorial comments:
>>
>> General: Consistent usage of "prediction and mode" vs 
>> "prediction/mode" would be desirable.
>>
>> s1, para 2: Needs to introduce the 'macroblock' jargon defined in RFC 
>> 6386. Suggest:
>> OLD:
>> VP8 is based on decomposition of frames into square sub-blocks of
>> pixels, prediction of such sub-blocks using previously constructed
>> blocks, and...
>> NEW:
>> VP8 is based on decomposition of frames into square sub-blocks of
>> pixels known as "macroblocks" (see Section 2 of [RFC6386]).  
>> Prediction
>> of such sub-blocks using previously constructed blocks, and...
>> END
>>
>> s2: There are a number of technical terms brought over from RFC 6386.
>> These should be listed in s2.  At least the following would be in the 
>> list:
>> key frame, interframe, golden frame, altref frame, macroblock.
>> Also the terms picture selection index (RPSI) and slice loss 
>> indication (SLI) from RFC 4585.
>>
>> s3, para 2: Providing a reference to explain what temporal 
>> scalability
>> and the hierarchy selection is all about would be useful.  One 
>> possibility
>> would be:
>>
>> Lim, J. Yang, and B. Jeon,”Fast Coding Mode Decision for Scalable 
>> Video Coding”,
>> 10th Int’l Conf. on Advanced Communication Technology, vol. 3, pp. 
>> 1897-1900, 2008.
>>
>> It would also help to add a pointer to the TL0PICIDX description in 
>> s4.2, thus:
>> ADD:
>> Support for temporal scalability is provided by the optional 
>> TL0PICIDX and TID/Y/KEYIDX
>> fields described in Section 4.2.
>> END
>>
>> ss3 and 4: The discussion of how the frame payload might or might not 
>> be distributed across
>> multiple packets is not very clear.   The draft has in mind a 
>> 'preferred use case' (para 1
>> of s4.4 says:
>>
>>> In the preferred use case, the S bit and PID fields
>>> described in Section 4.2 should be used to indicate what the packet
>>> contains.
>>
>> I think it would be helpful to be a bit more positive about this in 
>> s3, para 3:
>> OLD:
>> This memo
>> allows the partitions to be sent separately or in the same RTP
>> packet.  It may be beneficial for decoder error-concealment to send
>> the partitions in different packets, even though it is not mandatory
>> according to this specification.
>> NEW:
>> Accordingly, this document RECOMMENDS that the frame is packetized by 
>> the sender
>> with each data partition in a separate packet or packets.  This may 
>> be beneficial
>> for decoder error concealment and the payload format described in 
>> Section 4 provides
>> fields that allow the partitions to be identified even if the first 
>> partition is
>> not available.  The sender can, alternatively, aggregate the data 
>> partitions into
>> a single data stream and, optionally, split it into several packets 
>> without
>> consideration of the partition boundaries.  The receiver can use the 
>> length
>> information in the first partition to identify the partitions during 
>> decoding.
>> END
>>
>> s4/Figure 1: s/bytes/octets/ (2 places)
>>
>> Figure 1, caption: s/sequel/Sections 4.2 and 4.3/
>>
>> Figure 1: The format shown is correct only for the first packet in a 
>> frame.  Subsequent
>> packets will not contain the payload header and will have later 
>> octets of the frame payload.
>> This should be mentioned in drawing and/or in the caption.
>>
>> s4.1, last two paras:
>>> Sequence number:  The sequence numbers are monotonically increasing
>>> and set as packets are sent.
>>>
>>> The remaining RTP header fields are used as specified in
>>> [RFC3550].
>> Redefining the Sequence Number field seems gratuitous and potentially 
>> dangerous,
>> given that it is *very carefully* defined in RFC 3550 and the overall 
>> intent does
>> not seem any different from that in RFC 3550.  OTOH a note about the 
>> (non?-)use of the X bit
>> and the Payload Type field (PT) would be useful.  I suggest replacing 
>> the paras above with:
>> NEW:
>> X bit: MUST be 0.  The VP8 RTP profile does not use the RTP Header 
>> Extension capability.
>>
>> PT (Payload Type):  In line with the policy in Section 3 of 
>> [RFC3551], applications
>> using the VP8 RTP payload profile MUST assign a dynamic payload type 
>> number to
>> be used in each RTP session and provide a mechanism to indicate the 
>> mapping.
>> See Section 6.2 for the mechanism to be used with the Session 
>> Description Protocol (SDP)
>> [RFC4566].
>>
>> The remaining RTP Fixed Header Fields (V, P, CC, sequence number, 
>> SSRC and CSRC
>> identfiers) are used as specified in Section 5.1 of [RFC5330].
>> END
>> Note that this may be considered to make the reference to RFC 3551 
>> normative.
>>
>> s4.2, X bit:  There is potential for confusion between the X bit in 
>> the fixed header
>> and the X bit in the VP8 Payload Descriptor.  Perhaps changing this 
>> to C for control bits
>> would avoid the problem.
>>
>> s4.2, M bit: Similarly, it might be better to use B (for BIG) in 
>> place of M to avoid confusion.
>>
>> s4.2, para 7 after Fig 2: For consistency:
>> OLD:
>> When the X bit is set to 1 in the first octet, the extension bit
>> field octet MUST be provided as the second octet.  If the X bit is 0,
>> the extension bit field octet MUST NOT be present, and no extensions
>> (I, L, T, or K) are permitted.
>> NEW:
>> When the X [or C] bit is set to 1 in the first octet, the Extended 
>> Control Bits
>> field octet MUST be provided as the second octet.  If the X [or C] 
>> bit is 0,
>> the extension bit field octet MUST NOT be present, and no extensions
>> (I, L, T, or K) are permitted.
>> END
>>
>> s4.2: The issue of using either 'wrap' or 'restart' but not both for 
>> restarting number sequences has been raised in various comments by 
>> IESG members.
>>
>> s4.2, TL0PICIDX: I think, given the statements in s3 about not 
>> defining the hierarchy, that more explanation is needed of what the 
>> 'lowest or base layer' actually means.
>>
>> s4.2, TL0PICIDX:
>>> If TID is larger than 0, TL0PICIDX
>>>   indicates which base layer frame the current image depends on.
>>>   TL0PICIDX MUST be incremented when TID is 0.
>> What happens if L=1 but both T=0 and K=0 so that there is no TID 
>> value present? Or indeed if T=0 but K=1 so that the TID field is 
>> there but 'MUST be ignored by the receiver'  (definition of TID 
>> field)?
>>
>> s4.3: It would be useful to include the section number (9.1) where 
>> the uncompressed data chunk specification is found.  I was somewhat 
>> surprised that the ordering is reversed in the protocol spec.
>>
>> s4.3, VER: The receiver should validate this field - currently only 
>> values 0-3 are valid.
>>
>> s4.3, SizeN:  Make it clear these are integer fields: s/in Size0,/in 
>> integer fields Size0,/
>>
>> s4.3 and s4.4:  It would be helpful to implementers to explain what 
>> the offsets in the  first partition payload are for the additional 
>> partition count and additional partition sizes.  Having rummaged 
>> around in RFC  6386, I have to say that I am unclear whether the 
>> values are placed after the full chunk of data indicated by 
>> 1stPartitionSize or are part of this data. And if the latter is the 
>> case, how to work out where the partition sizes are.  I guess that 
>> they ought to be at offset (1stPartitionSize+10 - key frame - or 
>> 1stPartitionSize+3 - interframe) in the VP8 Payload field as it is 
>> difficult to work out where they are without decoding the partition 
>> otherwise.  Also exactly how many partition sizes to expect - I think 
>> I read in RFC 6386 that the last partition size is implicit. Help!
>>
>> In the course of clearing this up, it might be appropriate to move 
>> the last sentence of the first para of s4.4 and the second para of 
>> s4.4 into s4.3 as part of the explanation.  This is the relevant 
>> text:
>>> Note that the length of the first partition can
>>> always be obtained from the first partition size parameter in the 
>>> VP8
>>> payload header.
>>>
>>> The VP8 bitstream format [RFC6386] specifies that if multiple 
>>> DCT/WHT
>>> partitions are produced, the location of each partition start is
>>> found at the end of the first (prediction/mode) partition.  In this
>>> RTP payload specification, the location offsets are considered to be
>>> part of the first partition.
>>
>>
>> s4.4:  I found this section very difficult to parse.  It is a bit 
>> 'stream of consciousness' and would benefit from a reordering and 
>> rewrite.  Suggestion (assuming the second para gets moved to s4.3):
>> NEW SECTION 4.4:
>> An encoded VP8 frame can be divided into two or more partitions, as
>> described in Section 1.  It is OPTIONAL for a packetizer implementing
>> this RTP specification
>> to pay attention to the partition boundaries within an encoded frame.
>> If packetization of a frame is done without considering the partition
>> boundaries, the PID field MAY be set to zero for all packets, and the
>> S bit MUST NOT be set to one for any other packet than the first.
>>
>> If the preferred usage suggested in Section 3 is followed, with each 
>> packet
>> carrying data from exactly one partition, the S bit and PID fields
>> described in Section 4.2 SHOULD be used to indicate what the packet
>> contains.  The PID field should indicate which partition the first
>> octet of the payload belongs to, and the S bit indicates that the
>> packet starts on a new partition.
>>
>> If the packetizer does not pay attention to the partition boundaries, 
>> one
>> packet can contain a fragment of a  partition, a complete partition, 
>> or an
>> aggregate of fragments and partitions.  There is no explicit 
>> signaling of
>> partition boundaries in the payload and the partition lengths at the 
>> end of
>> the first partition have to be used to identify the boundaries. 
>> Partitions
>> MUST be aggregated in decoding order.  Two fragments from different
>> partitions MAY be aggregated into the same packet along with one or 
>> more
>> complete partitions.
>>
>> In all cases, the payload of a packet MUST contain data from only one 
>> video
>> frame.  Consequently the set of packets carrying the data from a 
>> particular
>> frame will contain exactly one VP8 Payload Header (see Section 4.3) 
>> carried
>> in the first packet of the frame.  The last, or only, packet carrying 
>> data for the
>> frame MUST have the M bit set in the RTP header.
>>
>> s4.5.1:  Th algorithm only applies for the RECOMMENDED use case with 
>> partitions in separate packets.
>> This should be noted.
>>
>> s5.2, last para: s/only parts of the frame is corrupted./only part of 
>> the frame is corrupted./
>>
>> s6: s/This payload format has two required parameters./This payload 
>> format has two optional parameters./
>> [I think this is probably a hangover from the previous version.]
>>
>> s7: s/Cryptographic system may also allow/The cryptographic system 
>> may also allow/
>>
>> s7:
>>> the usage of SRTP [RFC3711] is recommended.
>> RECOMMENDED?
>>
>> s10.2: I suspect that RFC 3551 is normative.


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Date: Thu, 30 Jul 2015 10:14:06 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: Ben Campbell <ben@nostrum.com>, Elwyn Davies <elwynd@dial.pipex.com>,  draft-ietf-payload-vp8.all@ietf.org, payload-chairs@ietf.org
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Subject: Re: [payload] Gen-art 2nd LC review of draft-ietf-payload-vp8-16
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Unfortunately this is vacation time in Scandinavia - won't do anything
with this until mid-August, I'm afraid. (Henrik might.)

So it'll have to wait another 2 telechats or so.....


On 07/29/2015 11:58 PM, Ben Campbell wrote:
> Hi VP8 Authors:
>
> Has there been a response to Elwyn's Gen-ART review from the authors?
> If so, I haven't seen it. (I did see Colin's response).
>
> Based Elwyn's review, and on Harald's response to my earlier
> questions, I assume that the authors will wish to update the draft
> before it goes back to an IESG telechat. If that's not the case,
> please let me know. Please be advised that tomorrow (July 30) is the
> nominal deadline for getting things on the August 6 telechat.
>
> Thanks!
>
> Ben.
>
>
> On 13 Jul 2015, at 15:57, Ben Campbell wrote:
>
>> (+payload and Harald)
>>
>> Elwyn: Thanks for the detailed review!
>>
>> Authors: Please address Elwyn's comments at your earliest
>> convenience. There's quite a bit here, and I assume we will need a
>> new revision before taking this to the IESG telechat, but it would be
>> helpful to discuss any proposed changes via email before submitting.
>>
>> Thanks!
>>
>> Ben.
>>
>>
>> On 13 Jul 2015, at 6:58, Elwyn Davies wrote:
>>
>>> I am the assigned Gen-ART reviewer for this draft. For background on
>>> Gen-ART, please see the FAQ at
>>>
>>> <http://wiki.tools.ietf.org/area/gen/trac/wiki/GenArtfaq>.
>>>
>>> Please resolve these comments along with any other Last Call comments
>>> you may receive.
>>>
>>> Document: draft-ietf-payload-vp8-16.txt
>>> Reviewer: Elwyn Davies
>>> Review Date: 2015/07/10
>>> IETF LC End Date: 2015/07/13
>>> IESG Telechat date: (if known) -
>>>
>>> Summary: Gosh, it has been a long time since the last review!
>>>
>>> Major issues:
>>>
>>> Minor issues:
>>> s4.2: Various of the frame indices either SHOULD or, in the case of
>>> KEYIDX, MAY start at random values.  The rationale(s) for these
>>> requirements are not explained - nor is the reason why it it is only a
>>> MAY for KEYIDX whereas it is SHOULD for PictureID and what might happen
>>> were the MAY/SHOULD to be ignored.
>>>
>>> Is the rationale something that should be mentioned in the security
>>> considerations?
>>>
>>> s10.1:  The downref to RFC 6386 identified by idnits was duly noted
>>> in the last call
>>> announcement.  I don't have a problem with the downref.
>>>
>>> Nits/editorial comments:
>>>
>>> General: Consistent usage of "prediction and mode" vs
>>> "prediction/mode" would be desirable.
>>>
>>> s1, para 2: Needs to introduce the 'macroblock' jargon defined in
>>> RFC 6386. Suggest:
>>> OLD:
>>> VP8 is based on decomposition of frames into square sub-blocks of
>>> pixels, prediction of such sub-blocks using previously constructed
>>> blocks, and...
>>> NEW:
>>> VP8 is based on decomposition of frames into square sub-blocks of
>>> pixels known as "macroblocks" (see Section 2 of [RFC6386]).  Prediction
>>> of such sub-blocks using previously constructed blocks, and...
>>> END
>>>
>>> s2: There are a number of technical terms brought over from RFC 6386.
>>> These should be listed in s2.  At least the following would be in
>>> the list:
>>> key frame, interframe, golden frame, altref frame, macroblock.
>>> Also the terms picture selection index (RPSI) and slice loss
>>> indication (SLI) from RFC 4585.
>>>
>>> s3, para 2: Providing a reference to explain what temporal scalability
>>> and the hierarchy selection is all about would be useful.  One
>>> possibility
>>> would be:
>>>
>>> Lim, J. Yang, and B. Jeon,”Fast Coding Mode Decision for Scalable
>>> Video Coding”,
>>> 10th Int’l Conf. on Advanced Communication Technology, vol. 3, pp.
>>> 1897-1900, 2008.
>>>
>>> It would also help to add a pointer to the TL0PICIDX description in
>>> s4.2, thus:
>>> ADD:
>>> Support for temporal scalability is provided by the optional
>>> TL0PICIDX and TID/Y/KEYIDX
>>> fields described in Section 4.2.
>>> END
>>>
>>> ss3 and 4: The discussion of how the frame payload might or might
>>> not be distributed across
>>> multiple packets is not very clear.   The draft has in mind a
>>> 'preferred use case' (para 1
>>> of s4.4 says:
>>>
>>>> In the preferred use case, the S bit and PID fields
>>>> described in Section 4.2 should be used to indicate what the packet
>>>> contains.
>>>
>>> I think it would be helpful to be a bit more positive about this in
>>> s3, para 3:
>>> OLD:
>>> This memo
>>> allows the partitions to be sent separately or in the same RTP
>>> packet.  It may be beneficial for decoder error-concealment to send
>>> the partitions in different packets, even though it is not mandatory
>>> according to this specification.
>>> NEW:
>>> Accordingly, this document RECOMMENDS that the frame is packetized
>>> by the sender
>>> with each data partition in a separate packet or packets.  This may
>>> be beneficial
>>> for decoder error concealment and the payload format described in
>>> Section 4 provides
>>> fields that allow the partitions to be identified even if the first
>>> partition is
>>> not available.  The sender can, alternatively, aggregate the data
>>> partitions into
>>> a single data stream and, optionally, split it into several packets
>>> without
>>> consideration of the partition boundaries.  The receiver can use the
>>> length
>>> information in the first partition to identify the partitions during
>>> decoding.
>>> END
>>>
>>> s4/Figure 1: s/bytes/octets/ (2 places)
>>>
>>> Figure 1, caption: s/sequel/Sections 4.2 and 4.3/
>>>
>>> Figure 1: The format shown is correct only for the first packet in a
>>> frame.  Subsequent
>>> packets will not contain the payload header and will have later
>>> octets of the frame payload.
>>> This should be mentioned in drawing and/or in the caption.
>>>
>>> s4.1, last two paras:
>>>> Sequence number:  The sequence numbers are monotonically increasing
>>>> and set as packets are sent.
>>>>
>>>> The remaining RTP header fields are used as specified in
>>>> [RFC3550].
>>> Redefining the Sequence Number field seems gratuitous and
>>> potentially dangerous,
>>> given that it is *very carefully* defined in RFC 3550 and the
>>> overall intent does
>>> not seem any different from that in RFC 3550.  OTOH a note about the
>>> (non?-)use of the X bit
>>> and the Payload Type field (PT) would be useful.  I suggest
>>> replacing the paras above with:
>>> NEW:
>>> X bit: MUST be 0.  The VP8 RTP profile does not use the RTP Header
>>> Extension capability.
>>>
>>> PT (Payload Type):  In line with the policy in Section 3 of
>>> [RFC3551], applications
>>> using the VP8 RTP payload profile MUST assign a dynamic payload type
>>> number to
>>> be used in each RTP session and provide a mechanism to indicate the
>>> mapping.
>>> See Section 6.2 for the mechanism to be used with the Session
>>> Description Protocol (SDP)
>>> [RFC4566].
>>>
>>> The remaining RTP Fixed Header Fields (V, P, CC, sequence number,
>>> SSRC and CSRC
>>> identfiers) are used as specified in Section 5.1 of [RFC5330].
>>> END
>>> Note that this may be considered to make the reference to RFC 3551
>>> normative.
>>>
>>> s4.2, X bit:  There is potential for confusion between the X bit in
>>> the fixed header
>>> and the X bit in the VP8 Payload Descriptor.  Perhaps changing this
>>> to C for control bits
>>> would avoid the problem.
>>>
>>> s4.2, M bit: Similarly, it might be better to use B (for BIG) in
>>> place of M to avoid confusion.
>>>
>>> s4.2, para 7 after Fig 2: For consistency:
>>> OLD:
>>> When the X bit is set to 1 in the first octet, the extension bit
>>> field octet MUST be provided as the second octet.  If the X bit is 0,
>>> the extension bit field octet MUST NOT be present, and no extensions
>>> (I, L, T, or K) are permitted.
>>> NEW:
>>> When the X [or C] bit is set to 1 in the first octet, the Extended
>>> Control Bits
>>> field octet MUST be provided as the second octet.  If the X [or C]
>>> bit is 0,
>>> the extension bit field octet MUST NOT be present, and no extensions
>>> (I, L, T, or K) are permitted.
>>> END
>>>
>>> s4.2: The issue of using either 'wrap' or 'restart' but not both for
>>> restarting number sequences has been raised in various comments by
>>> IESG members.
>>>
>>> s4.2, TL0PICIDX: I think, given the statements in s3 about not
>>> defining the hierarchy, that more explanation is needed of what the
>>> 'lowest or base layer' actually means.
>>>
>>> s4.2, TL0PICIDX:
>>>> If TID is larger than 0, TL0PICIDX
>>>>   indicates which base layer frame the current image depends on.
>>>>   TL0PICIDX MUST be incremented when TID is 0.
>>> What happens if L=1 but both T=0 and K=0 so that there is no TID
>>> value present? Or indeed if T=0 but K=1 so that the TID field is
>>> there but 'MUST be ignored by the receiver'  (definition of TID field)?
>>>
>>> s4.3: It would be useful to include the section number (9.1) where
>>> the uncompressed data chunk specification is found.  I was somewhat
>>> surprised that the ordering is reversed in the protocol spec.
>>>
>>> s4.3, VER: The receiver should validate this field - currently only
>>> values 0-3 are valid.
>>>
>>> s4.3, SizeN:  Make it clear these are integer fields: s/in Size0,/in
>>> integer fields Size0,/
>>>
>>> s4.3 and s4.4:  It would be helpful to implementers to explain what
>>> the offsets in the  first partition payload are for the additional
>>> partition count and additional partition sizes.  Having rummaged
>>> around in RFC  6386, I have to say that I am unclear whether the
>>> values are placed after the full chunk of data indicated by
>>> 1stPartitionSize or are part of this data. And if the latter is the
>>> case, how to work out where the partition sizes are.  I guess that
>>> they ought to be at offset (1stPartitionSize+10 - key frame - or
>>> 1stPartitionSize+3 - interframe) in the VP8 Payload field as it is
>>> difficult to work out where they are without decoding the partition
>>> otherwise.  Also exactly how many partition sizes to expect - I
>>> think I read in RFC 6386 that the last partition size is implicit.
>>> Help!
>>>
>>> In the course of clearing this up, it might be appropriate to move
>>> the last sentence of the first para of s4.4 and the second para of
>>> s4.4 into s4.3 as part of the explanation.  This is the relevant text:
>>>> Note that the length of the first partition can
>>>> always be obtained from the first partition size parameter in the VP8
>>>> payload header.
>>>>
>>>> The VP8 bitstream format [RFC6386] specifies that if multiple DCT/WHT
>>>> partitions are produced, the location of each partition start is
>>>> found at the end of the first (prediction/mode) partition.  In this
>>>> RTP payload specification, the location offsets are considered to be
>>>> part of the first partition.
>>>
>>>
>>> s4.4:  I found this section very difficult to parse.  It is a bit
>>> 'stream of consciousness' and would benefit from a reordering and
>>> rewrite.  Suggestion (assuming the second para gets moved to s4.3):
>>> NEW SECTION 4.4:
>>> An encoded VP8 frame can be divided into two or more partitions, as
>>> described in Section 1.  It is OPTIONAL for a packetizer implementing
>>> this RTP specification
>>> to pay attention to the partition boundaries within an encoded frame.
>>> If packetization of a frame is done without considering the partition
>>> boundaries, the PID field MAY be set to zero for all packets, and the
>>> S bit MUST NOT be set to one for any other packet than the first.
>>>
>>> If the preferred usage suggested in Section 3 is followed, with each
>>> packet
>>> carrying data from exactly one partition, the S bit and PID fields
>>> described in Section 4.2 SHOULD be used to indicate what the packet
>>> contains.  The PID field should indicate which partition the first
>>> octet of the payload belongs to, and the S bit indicates that the
>>> packet starts on a new partition.
>>>
>>> If the packetizer does not pay attention to the partition
>>> boundaries, one
>>> packet can contain a fragment of a  partition, a complete partition,
>>> or an
>>> aggregate of fragments and partitions.  There is no explicit
>>> signaling of
>>> partition boundaries in the payload and the partition lengths at the
>>> end of
>>> the first partition have to be used to identify the boundaries.
>>> Partitions
>>> MUST be aggregated in decoding order.  Two fragments from different
>>> partitions MAY be aggregated into the same packet along with one or
>>> more
>>> complete partitions.
>>>
>>> In all cases, the payload of a packet MUST contain data from only
>>> one video
>>> frame.  Consequently the set of packets carrying the data from a
>>> particular
>>> frame will contain exactly one VP8 Payload Header (see Section 4.3)
>>> carried
>>> in the first packet of the frame.  The last, or only, packet
>>> carrying data for the
>>> frame MUST have the M bit set in the RTP header.
>>>
>>> s4.5.1:  Th algorithm only applies for the RECOMMENDED use case with
>>> partitions in separate packets.
>>> This should be noted.
>>>
>>> s5.2, last para: s/only parts of the frame is corrupted./only part
>>> of the frame is corrupted./
>>>
>>> s6: s/This payload format has two required parameters./This payload
>>> format has two optional parameters./
>>> [I think this is probably a hangover from the previous version.]
>>>
>>> s7: s/Cryptographic system may also allow/The cryptographic system
>>> may also allow/
>>>
>>> s7:
>>>> the usage of SRTP [RFC3711] is recommended.
>>> RECOMMENDED?
>>>
>>> s10.2: I suspect that RFC 3551 is normative.


-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [payload] Gen-art 2nd LC review of draft-ietf-payload-vp8-16
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On 30 Jul 2015, at 3:14, Harald Alvestrand wrote:

> Unfortunately this is vacation time in Scandinavia - won't do anything
> with this until mid-August, I'm afraid. (Henrik might.)
>
> So it'll have to wait another 2 telechats or so.....

Understood, thanks!


From nobody Thu Jul 30 09:01:18 2015
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Subject: [payload] Protocol Action: 'RTP Payload Format for G.711.0' to Proposed Standard (draft-ietf-payload-g7110-06.txt)
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The IESG has approved the following document:
- 'RTP Payload Format for G.711.0'
  (draft-ietf-payload-g7110-06.txt) as Proposed Standard

This document is the product of the Audio/Video Transport Payloads
Working Group.

The IESG contact persons are Barry Leiba, Ben Campbell and Alissa Cooper.

A URL of this Internet Draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-g7110/




Technical Summary

The document specifies the Real-Time Transport Protocol (RTP) payload format for ITU-T Recommendation G.711.0.  ITU-T Rec. G.711.0 defines a lossless and stateless compression for G.711 packet payloads typically used in IP networks.  This document also defines a storage mode format for G.711.0 and a media type registration for the  G.711.0 RTP payload format.

Working Group Summary

The initial individual draft had a section about G.711.0 "in the middle" this was removed before the document became a WG document. There were no other concerns or objections to the document.

Document Quality

There is an existing implementation. The request for a media type review was posted on March 4th, 2014.

Personnel

Document Shepherd is Roni Even and the responsible AD is Richard Barnes.

RFC Editor Note

  Please change the first two paragraphs in section 10 to match the boilerplate in draft-ietf-payload-rtp-howto-14:

OLD:

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550], and in any appropriate RTP profile (for
   example RFC 3551 [RFC3551] or [RFC4585]).  This implies that
   confidentiality of the media streams is achieved by encryption; for
   example, through the application of SRTP [RFC3711].  Because the data
   compression used with this payload format is applied end-to-end, any
   encryption needs to be performed after compression.

   Note that the appropriate mechanism to ensure confidentiality and
   integrity of RTP packets and their payloads is very dependent on the
   application and on the transport and signaling protocols employed.
   Thus, although SRTP is given as an example above, other possible
   choices exist.

NEW:

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550] , and in any applicable RTP profile such as
   RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711] or RTP/
   SAVPF [RFC5124].  However, as "Securing the RTP Protocol Framework:
   Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202]
   discusses, it is not an RTP payload format's responsibility to
   discuss or mandate what solutions are used to meet the basic security
   goals like confidentiality, integrity and source authenticity for RTP
   in general.  This responsibility lays on anyone using RTP in an
   application.  They can find guidance on available security mechanisms
   and important considerations in Options for Securing RTP Sessions
   [RFC7201].  Applications SHOULD use one or more appropriate strong
   security mechanisms.  The rest of this security consideration section
   discusses the security impacting properties of the payload format
   itself.

   Because the data compression used with this payload format is applied end-to-end, any
   encryption needs to be performed after compression.



From nobody Fri Jul 31 15:15:26 2015
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Subject: [payload] Last Call: <draft-ietf-payload-rtp-h265-13.txt> (RTP Payload Format for High Efficiency Video Coding) to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Payloads
WG (payload) to consider the following document:
- 'RTP Payload Format for High Efficiency Video Coding'
  <draft-ietf-payload-rtp-h265-13.txt> as Proposed Standard

Please note that this draft has a normative reference to ITU-T Recommendation 
H.265, "High efficiency video  coding", April 2013.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2015-08-14. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   This memo describes an RTP payload format for the video coding
   standard ITU-T Recommendation H.265 and ISO/IEC International
   Standard 23008-2, both also known as High Efficiency Video Coding
   (HEVC) and developed by the Joint Collaborative Team on Video
   Coding (JCT-VC).  The RTP payload format allows for packetization
   of one or more Network Abstraction Layer (NAL) units in each RTP
   packet payload, as well as fragmentation of a NAL unit into
   multiple RTP packets.  Furthermore, it supports transmission of
   an HEVC bitstream over a single as well as multiple RTP streams.
   When multiple RTP streams are used, a single or multiple
   transports may be utilized.  The payload format has wide
   applicability in videoconferencing, Internet video streaming, and
   high bit-rate entertainment-quality video, among others.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-h265/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-h265/ballot/


The following IPR Declarations may be related to this I-D:

   https://datatracker.ietf.org/ipr/2376/
   https://datatracker.ietf.org/ipr/2321/
   https://datatracker.ietf.org/ipr/2508/
   https://datatracker.ietf.org/ipr/2190/


