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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] Fwd: I-D Action: draft-burman-rtcweb-mmusic-media-structure-00.txt
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MMUSIC,

This draft is intended as supporting material for next weeks interim
meeting. Its main focus is on commonalities that exist between WebRTC
and CLUE when it comes to identify and handle concepts in the multimedia
transport. It also looks explicit at some by the authors perceived
shortcomings and issues. There is also some discussion on the potential
impact of the SDP evolution that is currently ongoing.

I do hope you have time to review it.

Cheers

Magnus Westerlund

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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title           : Multi-Media Concepts and Relations
	Author(s)       : Bo Burman
                          Magnus Westerlund
	Filename        : draft-burman-rtcweb-mmusic-media-structure-00.txt
	Pages           : 22
	Date            : 2013-02-01

Abstract:
   There are currently significant efforts ongoing in IETF regarding
   more advanced multi-media functionalities, such as the work related
   to RTCWEB and CLUE.  This work includes use cases for both multi-
   party communication and multiple media streams from an individual
   end-point.  The usage of scalable encoding or simulcast encoding as
   well as different types of transport mechanisms have created
   additional needs to correctly identify different types of resources
   and describe their relations to achieve intended functionalities.

   The different usages have both commonalities and differences in needs
   and behavior.  This document attempts to review some usages and
   identify commonalities and needs.  It then continues to highlight
   important aspects that need to be considered in the definition of
   these usages.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-burman-rtcweb-mmusic-media-structure

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-burman-rtcweb-mmusic-media-structure-00


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

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From worley@shell01.TheWorld.com  Fri Feb  1 15:37:21 2013
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Subject: [MMUSIC] Comments on draft-alvestrand-one-rtp-01
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general

I think the concept is that the various media streams are
demultiplexed out of the common RTP scheme by the fact that each
payload type (codec type number) appears in at most one m= line of the
bundle.  But that isn't stated explicitly anywhere.  In any case, the
method for demultiplexing the RTP should be stated explicitly.

section 4

   If the answerer wishes to refuse an entire m= section, while
   accepting the RTP session otherwise, he MAY indicate this by setting
   the port number of the relevant section to zero.

... and remove that m= line's mid value from the group attribute.
RFC 5888 section 9.2

section 7

In regard to RTCP, it appears that RTCP reports on each SSRC
separately.  I would expect that the bandwidth for the RTCP reports on
a stream would be limited to 5% of the bandwidth for that stream.
This keeps the total RTCP bandwidth below 5% of the total stream
bandwidth; indeed it is what you would get if you simply merged the
RTCP streams for the various media streams considered as separate RTP
streams.  In this configuration, the RTCP for an audio stream would
never exceed the RTP for the audio stream.  Although it does result in
the frequency of reports differing between the various streams.

Dale

From worley@shell01.TheWorld.com  Fri Feb  1 15:38:03 2013
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Subject: [MMUSIC] Comments on draft-holmberg-mmusic-sdp-mmt-negotiation-00
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generally

Conceptually, this draft looks on the task as being:  we either
establish a set of media streams defined by a set of m= lines, or we
establish a RTP stream that contains multiple media types.  I think
this is slightly incorrect, the MMT RTP stream should be seen as a
container that encapsulates several media streams -- the anymedia m=
line describes the container and its properties, and the constituent
m= lines describe the constituent media streams.  (And it is possible
that two constituent media streams have the same media type -- one can
easily have more than one audio or video stream in a session.)

Within that change, an answer that accepts the MMT m= line should be
allowed to also accept/reject the constituent m= lines, to indicate
whether or not it wants to process those particular media streams
within the bundle.  (The port numbers for the constituent m= lines in
the answer would be dummies, only significant as to whether they are
zero or not, just as the port numbers for the constituent m= lines in
the offer become dummies if the MMT is accepted.)

abstract

"5-tuple" should be defined; it is not commonly used in RFCs.

section 4

Is there any reason to restrict MMT from being used in non-O/A
situations?  Of course, there would have to be out-of-band knowledge
that the interpreter of the SDP understands MMT, but out-of-band
knowledge (e.g., of the media types and codecs supported) is required
in non-O/A situations already.

section 5.2.1

   A "MMT" group MUST contain a "anymedia" SDP Media Description, and at
------------------------------^
should be "exactly one"
   least one SDP Media Description for a specific SDP Media Type (audio,
   video, etc).

   From a "MMT" group the SDP
   Answerer [RFC3264] will accept either the SDP Media Descriptions for
   the specific SDP Media Types, or the "anymedia" SDP Media
   Description.

This doesn't allow the answerer to reject all of the media in the
group.  A more complete phrasing:

   From a "MMT" group, the SDP
   Answerer [RFC3264] will either (1) accept the "anymedia" SDP Media
   Description and reject the specific SDP media types, (2) reject the
   "anymedia" SDP Media Description and accept some or all of the
   specific SDP media types, or (3) reject both the "anymedia" SDP Media
   Description and the specific SDP media types (to reject all the
   media streams in any form).

However, that would change if the answerer was allowed to
accept/reject specific media streams even when they were bundled into
the anymedia stream.

section 5.2.2.1

   1) Include a "anymedia" SDP Media Description; and
--------------^

should be "exactly one"

section 5.2.2.2

   2) Reject the "anymedia" SDP Media Description, and accept some or
   all of the other SDP Media Descriptions associated with the MMT
   group.

But the answerer must be allowed to reject *both* the anymedia m= line
and all of the contained m= lines (if it doesn't like any of the media
streams involved).  So it would be easier to specify:

      When an SDP Answerer generates an SDP Answer, for each "MMT" group
      in the associated SDP Offer, if it accepts the "anymedia" SDP Media
      Description, it MUST reject all of the other SDP Media Descriptions
      associated with the MMT group.

Within this dichotomy, it would be useful to spell out the
consequences for the a=group:MMT line, viz.:

      1) [...]  Consequently, the mid values of the rejected lines are
      removed from the a=group:MMT line in the answer. Or,

      2) [...]  Consequently, the a=group:MMT line is removed from the
      answer.

However, that would change if the answerer was allowed to
accept/reject specific media streams even when they were bundled into
the anymedia stream.

   NOTE: As described in [RFC3264], an SDP Media Description can be
-------------------------------------------------------------^^^^^^
This probably should be "is", as there is no other way to reject a
media description.
   rejected by setting the port value of the associated m- line to zero
   in the SDP Answer.

... and remove that m= line's mid value from the group attribute.
RFC 5888 section 9.2

section 6.1

   It
   allows, if both endpoints support the mechanism, multiple media typex
   to be multiplexed on a single 5-tuple.

More exactly stated, it allows multiple media *streams* to be
multiplexed on a single 5-tuple.  (In some cases, all of the streams
may have the same media *type*, but we still want to be able to
distinguish them.)

   [...] the SDP mmtype
   attribute will be used to map each PT to a specific media type (e.g.
   audio, video, etc).

ditto

   In addition, as it might not always be possible to
   retreive the media type from the "rtpmap" SDP Attribute value, [...]
---^^^^^^^^

I think "determine" is better here.

section 6.3

   NOTE: Additional extensions are needed in order to specify SDP
   Attribute values for individual media types, or individual media
   sources, associated with the "anymedia" SDP Media Description.

Perhaps the attributes given on the constituent m= lines would specify
the attributes for the constituent media streams.

Alternatively, we could define:

      a=mmtattr:<mid-value>:<attribute>:<value>

to provide "a=<attribute>:<value>" for the constituent media stream
identified by "a=mid:<mid-value>".

section 7

If mmtype is to properly demultiplex received RTP, it has to specify
which of the constituent m= lines that particular PT is mapped to --
just knowing the media type and codec name is not necessarily
sufficient, since more than one constituent m= line use a single
codec.

Also, how is RTCP to be demultiplexed?  (Probably throught SSRCs.)

section 7.2

   a=mmtype: format media

Presumably the space after ":" is a typo.  (Given the syntax rules of
SDP, a space appearing in that location is significant.)

A better syntax would be:

      a=mmtype:<format> <mid> <media>

That would specify the mid value of the constituent media stream.  But
of course, once we know the mid value, we can determine the media type
from the corresponding m= line.  Which would suggest a syntax like:

      a=mmtype:<format> <mid>

Or all the formats for a single mid could be specified together:

      a=mmtype:<mid> <format> <format> ...

It would be useful to mention here that the format number in the
anymedia m= line does not have to be the same as the format number in
the constituent m= line for the same codec, but that the codec does
have to appear in the constituent m= line.  (Or do we wish to require
that the format numbers be the same, for sanity's sake?)

section 9

   SDP Answer (multiplexed media types accepted)
       [...]
       a=group:MMT foo bar

This is backwards; it must be "a=group:MMT zoe" based on RFC 5888
section 9.2.  (Though I'm proposing to change what this attribute
value would be; see below.)

   SDP Answer (multiplexed media types not accepted)
       [...]
       a=group:MMT foo bar

This attribute is incorrect, as the group does not contain one m= line
for anymedia.  This attribute should be omitted.

Incorporating the above suggestions into the examples (changes
indicated by ">"), the offer becomes:

       v=0
       o=alice 2890844526 2890844526 IN IP4 host.atlanta.com
       s=
       c=IN IP4 host.atlanta.com
       t=0 0
       a=group:MMT foo bar zoe
       m=audio 10000 RTP/AVP 0 8 97
       a=mid:foo
       b=AS:200
       a=rtpmap:0 PCMU/8000
       a=rtpmap:8 PCMA/8000
       a=rtpmap:97 iLBC/8000
       m=video 20000 RTP/AVP 31 32
       a=mid:bar
       b=AS:1000
       a=rtpmap:31 H261/90000
       a=rtpmap:32 MPV/90000
       m=anymedia 30000 RTP/AVP 0 8 97 31 32
       a=mid:zoe
       a=rtpmap:0 PCMU/8000
       a=rtpmap:8 PCMA/8000
       a=rtpmap:97 iLBC/8000
       a=rtpmap:31 H261/90000
       a=rtpmap:32 MPV/90000
>      a=mmtype:foo 0 8 97
>      a=mmtype:bar 31 32

The answer accepting the MMT group (and both constituent streams)
becomes:

       v=0
       o=bob 2808844564 2808844564 IN IP4 host.biloxi.com
       s=
       c=IN IP4 host.biloxi.com
       t=0 0
>      a=group:MMT foo bar zoe
>      m=audio 1 RTP/AVP 0
       a=mid:foo
       a=rtpmap:0 PCMU/8000
>      m=video 1 RTP/AVP 32
       a=mid:bar
       a=rtpmap:32 MPV/90000
       m=anymedia 60000 RTP/AVP 0 32
       a=mid:zoe
       a=rtpmap:0 PCMU/8000
       a=rtpmap:32 MPV/90000
>      a=mmtype:foo 0
>      a=mmtype:bar 32

The answer rejecting the MMT group (but accepting both constituent
streams independently) becomes:

       v=0
       o=bob 2808844564 2808844564 IN IP4 host.biloxi.com
       s=
       c=IN IP4 host.biloxi.com
       t=0 0
>      [a=group:MMT omitted]
       m=audio 40000 RTP/AVP 0
       a=mid:foo
       a=rtpmap:0 PCMU/8000
       m=video 50000 RTP/AVP 32
       a=mid:bar
       a=rtpmap:32 MPV/90000
       m=anymedia 0 RTP/AVP 0 32
       a=mid:zoe
       [The a=mid attributes may be missing if the answerer does not
       understand SDP grouping.]

Dale

From worley@shell01.TheWorld.com  Fri Feb  1 15:39:06 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-sdp-bundle-negotiation-00
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general

I think the concept is that the various media streams are
demultiplexed out of the common RTP scheme by the fact that each
payload type (codec number) appears in at most one m= line of the
bundle.  (If so, the payload type values on each m= line must be
distinct, not just the *dynamic* payload type values.  section 7.2)
But that isn't stated explicitly anywhere.  In any case, the method
for demultiplexing the bundle should be stated explicitly.

Also, how the RTCP reports are to be demultiplexed needs to be made
explicit.  (Though I suspect that is via SSRCs.)

If one or more of the m= lines specifies encryption, how is encryption
coordinated between the various media streams?

What is the fallback if the offerer uses BUNDLE but the answerer does
not understand it?  RFC 4566 says that the use of multiple m= lines
with the same address/port is undefined, which means that a
non-understanding the answerer is allowed to reject the offer as
malformed.

section 1

"5-tuple" should be defined

   the group will share a single port number value

The group will also share a single address, so it would be clearer to
say "share a single address and port value".
Or you could phrase it as simply "share a single port".  But saying "a
single port number value" suggests that the addresses might be
different even though the port numbers are the same.

   there is no grouping defined by such means

I assume this means that sharing the port number does not imply
grouping.  Better to phrase this "sharing a port number does not imply
grouping".

section 6.1

   assign SDP media-
   level 'mid' attribute values to each "m=" line associated with the
   "BUNDLE" group.

'mid' attributes are required for all the m= lines.  RFC 5888 section 6

   NOTE: If an entity wants to disable specific media ("m=" line)
   associated with a "BUNDLE" group, it will use a zero port number
   value for the "m=" line associated with the media.

... and remove that m= line's mid value from the group attribute.
RFC 5888 section 9.2

section 6.2

   If the SDP Answererer 

should be "Answerer"

   If the SDP Answererer indicates that it will not use bundled media,
   the SDP Offerer will still use the single port number value for each
   "m= line" associated with the offered "BUNDLE" group, and it will
-------------------------------------------------------------^^
   normally be able to separate each individual media.

should probably be specified as "the Offerer"

   The default
   mechanism for separating media received on a single IP address and
   port doing this is by using a 5-tuple based mapping for each
   individual media.

This would be clearer if it was phrased "The default mechanism for
separating media received on a single IP address and port is by using
the sending address/port to identify each media stream."  Although the
possibility of this requires symmetric media, which is very common,
but not required by RFC 326[1-5].  What do we do if symmetric media isn't
in use?

   If an SDP Offer, offering a "BUNDLE" group, and the SDP Offerer has
----------------------------------------------^
   reasons to believe that the rejection is due to the usage of a single

I think you intended to insert "is rejected" here.

section 6.2 says:

   If the SDP Offerer is aware of the Synchronization
   Source (SSRC) [RFC3550] values that the SDP Answerer will use in the
   media it sends, and the SSRC values will be unique for each media,
   the SDP Offerer can separate media based on the SSRC values.

but section 7.2 says:

   When a single RTP Session is used, media associated with all "m="
   lines part of a bundle group share a single SSRC [RFC3550] numbering
   space.

Is multiple SSRC supported or not?

Or is the intention that the SSRC values have to be different from
each other?  Sometimes that is the implication of "sharing a space".
But by their very nature (RFC 3550 section 8), all SSRCs are from a
single numbering space of unsigned 32-bit integers.

section 7.2

   If transmission of multiple media types are done
   with time overlap RTP and RTCP fails to function.

I don't understand this sentence.  I suspect some words are missing.

Dale

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After reading
    draft-alvestrand-one-rtp-01
    draft-holmberg-mmusic-sdp-mmt-negotiation-00
    draft-ietf-mmusic-sdp-bundle-negotiation-00
    draft-nandakumar-rtcweb-sdp-00
here are my thoughts on bundling:

1. Concept

Overall, we need to fix the concept that a bundle is a container for a
group of media streams, media streams as they are specified by m=
lines.  We need to be able to bundle multiple media streams in a way
that allows them to be demultiplexed accurately, even if different
streams share the same media types, codecs, etc.

In this regard, the transport protocol, transport ports, encryption
specifications, ICE specifications, and (less obviously) payload type
numbers are considered to be part of the transport mechanism and not
part of the media stream properties.

The m= line for a bundle should have all the "transport" attribues
needed for the bundle, while the m= lines for the constituent media
streams would have the attributes that describe the media stream
itself.

(This distinction gets undermined if we use payload types to
demultiplex the media streams, as the bundle m= line has to list the
payload types, and thus probably the codec names and parameters, which
are really constituent-level information.  One solution would be to
require that if a bundle payload type is demultiplexed to a particular
m= line, then that m= line must have the same payload type with the
same codec.  Indeed, there's no particular reason for the bundle m=
line to have a=fmtp attributes at all, though the m= line should
probably list all the payload types the bundle RTP is going to use.)

2. Configuration of m= lines

There needs to be a fallback mechanism if the answerer does not
understand the bundling mechanism.
draft-ietf-mmusic-sdp-bundle-negotiation (BUNDLE) specifies that a
bundle of m= lines all specify the same port number.  That is OK if
the answerer understands BUNDLE, but if the answerer does not
understand BUNDLE, the SDP is malformed (per RFC 4566 section 5.14),
and the answerer may reject it as such.  Because of this, we need to
adopt the mechanism of draft-alvestrand-one-rtp (TOGETHER), where each
m= line in the offer has a distinct port number.

There also needs to be a way for the answerer to individually accept
or reject media streams within the bundle, that is, the m= lines
within the bundle.  TOGETHER does not accomplish that, because the
first m= line in the bundle is the one used to send/receive media (if
the bundle is accepted), and so it cannot be rejected.  As
draft-alvestrand-one-rtp says

   The answerer cannot refuse the first m= section and establish the
   call.  [[NOTE IN DRAFT: This could be relaxed by allowing it when ICE
   is in use, or saying that the first non-zero port number is used.
   Both lead to nonobvious edge cases.  Is the extra complexity worth
   it?]]

draft-holmberg-mmusic-sdp-mmt-negotiation (MMT) avoids this problem by
associating with each bundle a "master" m= line with media type
"anymedia".  If the bundle is accepted, this m= line provides the
transport information.  If the bundle is rejected, then the answerer
accepts or rejects each constituent m= line separately, and that m=
lines attributes determine the transport mechanism.

However, MMT needs to be updated to match the concept that the
anymedia RTP stream multiplexes what would be the RTP streams of one
or more constituent media streams.  Currently, it assumes that
identifying the media type of an incoming RTP packet is sufficient
demultiplexing.  As a consequence of this, it needs to be modified so
that if the answerer accepts the MTT bundle (and thus, the anymedia m=
line), it still must accept or reject each constituent m= line
individually.

3. Demultiplexing RTP

MMT envisions that the media data is demultiplexed based on the
payload types:  The m=anymedia description contains a set of a=mmttype
attributes that (I think should) provide the mapping from the payload
types to the mid values of the constituent m= lines.  (The codec for
each payload type is provided by a=fmtp attributes of the m=anymedia
description as normal.)

Note that this does not allow recovery of the payload types that would
have been used for that codec and m= line (if multiple payload types
for that m= line have the same codec).  But we assume that payload
types are a transport matter and are of no interest to the higher
layers of media processing.

But perhaps instead, we should require that there be a matching
payload type for the constituent m= line and use its a=fmtp
attribute.  That allows us to not duplicate any codec specifications
from the constituent m= lines into the m=anymedia line, and does allow
recovering the "original" payload types.

4. RTCP

My knowledge of RTCP is very thin.  But it's clear that incoming RTCP
over the bundle transport must be demultiplexable to the point that we
can separate which RTCP information is applicable to which constituent
media stream.  My understanding is that RTCP information is identified
by the SSRC of the media source it is reporting on, and that it is
paired with the right media stream by examining the SSRC value in the
RTP packet headers.  If so, demultiplexing RTCP requires no new
protocol facilities.

In regard to bandwidth consumption, I assume that the bundle's RTCP
stream is essentially the union of the RTCP streams (or rather, the
RCTP packets thereof, which are packed into the transport-level RTCP
packets) that would be generated for each constituent m= line.  In
particular, the frequency of the RTCP packets for a particular media
stream is scaled to the needs of that media stream, rather than there
being RTCP reports at same frequency for all media streams.  Thus, as
long as the RTCP reports on each media stream are less than 5% of the
RTP bandwidth for that media stream, the aggregate RTCP bandwidth is
less then 5% of the aggregate RTP bandwidth.

5. Examples

Incorporating these suggestions into
draft-holmberg-mmusic-sdp-mmt-negotiation, I get this example:

   SDP Offer (MMT offered)

       v=0
       o=alice 2890844526 2890844526 IN IP4 host.atlanta.com
       s=
       c=IN IP4 host.atlanta.com
       t=0 0
       a=group:MMT foo bar zoe
       m=audio 10000 RTP/AVP 0 8 97
       a=mid:foo
       b=AS:200
       a=rtpmap:0 PCMU/8000
       a=rtpmap:8 PCMA/8000
       a=rtpmap:97 iLBC/8000
       m=video 20000 RTP/AVP 31 32
       a=mid:bar
       b=AS:1000
       a=rtpmap:31 H261/90000
       a=rtpmap:32 MPV/90000
       m=anymedia 30000 RTP/AVP 0 8 97 31 32
       a=mid:zoe
       a=mmtype:foo 0 8 97
       a=mmtype:bar 31 32

   SDP Answer (MMT accepted, including both constituent streams)

       v=0
       o=bob 2808844564 2808844564 IN IP4 host.biloxi.com
       s=
       c=IN IP4 host.biloxi.com
       t=0 0
       a=group:MMT foo bar zoe
       m=audio 1 RTP/AVP 0
       a=mid:foo
       a=rtpmap:0 PCMU/8000
       m=video 1 RTP/AVP 32
       a=mid:bar
       a=rtpmap:32 MPV/90000
       m=anymedia 60000 RTP/AVP 0 32
       a=mid:zoe
       a=mmtype:foo 0
       a=mmtype:bar 32

   SDP Answer (MMT not accepted, but both constituent streams accepted)

       v=0
       o=bob 2808844564 2808844564 IN IP4 host.biloxi.com
       s=
       c=IN IP4 host.biloxi.com
       t=0 0
       m=audio 40000 RTP/AVP 0
       a=mid:foo
       a=rtpmap:0 PCMU/8000
       m=video 50000 RTP/AVP 32
       a=mid:bar
       a=rtpmap:32 MPV/90000
       m=anymedia 0 RTP/AVP 0
       a=mid:zoe
       [The a=mid attributes may be missing if the answerer does not
       understand SDP grouping.]

   SDP Offer with ICE (multiplexed media types offered)

       v=0
       o=alice 2890844526 2890844526 IN IP4 host.atlanta.com
       s=
       c=IN IP4 host.atlanta.com
       t=0 0
       a=group:MMT foo bar zoe
       m=audio 10000 RTP/AVP 0 8 97
       a=mid:foo
       b=AS:200
       a=rtpmap:0 PCMU/8000
       a=rtpmap:8 PCMA/8000
       a=rtpmap:97 iLBC/8000
       a=candidate:1 1 UDP 1694498815 host.atlanta.com 10000 typ host
       m=video 20000 RTP/AVP 31 32
       a=mid:bar
       b=AS:1000
       a=rtpmap:31 H261/90000
       a=rtpmap:32 MPV/90000
       a=candidate:1 1 UDP 1694498815 host.atlanta.com 20000 typ host
       m=anymedia 30000 RTP/AVP 0 8 97 31 32
       a=mid:zoe
       a=mmtype:foo 0 8 97
       a=mmtype:bar 31 32
       a=candidate:1 1 UDP 1694498815 host.atlanta.com 30000 typ host

Dale

From harald@alvestrand.no  Sat Feb  2 08:54:18 2013
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Subject: Re: [MMUSIC] Thoughts on bundling mechanisms
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On 02/02/2013 12:38 AM, Dale R. Worley wrote:
> After reading
>      draft-alvestrand-one-rtp-01
>      draft-holmberg-mmusic-sdp-mmt-negotiation-00
>      draft-ietf-mmusic-sdp-bundle-negotiation-00
>      draft-nandakumar-rtcweb-sdp-00
> here are my thoughts on bundling:
>
> 1. Concept
>
> Overall, we need to fix the concept that a bundle is a container for a
> group of media streams, media streams as they are specified by m=
> lines.  We need to be able to bundle multiple media streams in a way
> that allows them to be demultiplexed accurately, even if different
> streams share the same media types, codecs, etc.
>
> In this regard, the transport protocol, transport ports, encryption
> specifications, ICE specifications, and (less obviously) payload type
> numbers are considered to be part of the transport mechanism and not
> part of the media stream properties.
>
> The m= line for a bundle should have all the "transport" attribues
> needed for the bundle, while the m= lines for the constituent media
> streams would have the attributes that describe the media stream
> itself.


This is an interesting concept - if I understand you correctly, you say 
that instead of having an m-line describe one or more media streams + a 
set of transport attributes, you have two kinds of m-lines, one of which 
describes transport attributes and the other one of which specifies 
media information.

This is a new concept - I think it would be an interesting evolution of 
SDP - but I have problems seeing how to make any kind of one-exchange 
compatibility mode function with this model. I may be misunderstanding 
your suggestion.



>
> (This distinction gets undermined if we use payload types to
> demultiplex the media streams, as the bundle m= line has to list the
> payload types, and thus probably the codec names and parameters, which
> are really constituent-level information.  One solution would be to
> require that if a bundle payload type is demultiplexed to a particular
> m= line, then that m= line must have the same payload type with the
> same codec.  Indeed, there's no particular reason for the bundle m=
> line to have a=fmtp attributes at all, though the m= line should
> probably list all the payload types the bundle RTP is going to use.)

I think we all agree that multiplexing different media streams using 
payload type is a horrible idea, and nobody should do that.

>
> 2. Configuration of m= lines
>
> There needs to be a fallback mechanism if the answerer does not
> understand the bundling mechanism.
> draft-ietf-mmusic-sdp-bundle-negotiation (BUNDLE) specifies that a
> bundle of m= lines all specify the same port number.  That is OK if
> the answerer understands BUNDLE, but if the answerer does not
> understand BUNDLE, the SDP is malformed (per RFC 4566 section 5.14),
> and the answerer may reject it as such.  Because of this, we need to
> adopt the mechanism of draft-alvestrand-one-rtp (TOGETHER), where each
> m= line in the offer has a distinct port number.
>
> There also needs to be a way for the answerer to individually accept
> or reject media streams within the bundle, that is, the m= lines
> within the bundle.  TOGETHER does not accomplish that, because the
> first m= line in the bundle is the one used to send/receive media (if
> the bundle is accepted), and so it cannot be rejected.  As
> draft-alvestrand-one-rtp says
>
>     The answerer cannot refuse the first m= section and establish the
>     call.  [[NOTE IN DRAFT: This could be relaxed by allowing it when ICE
>     is in use, or saying that the first non-zero port number is used.
>     Both lead to nonobvious edge cases.  Is the extra complexity worth
>     it?]]
>
> draft-holmberg-mmusic-sdp-mmt-negotiation (MMT) avoids this problem by
> associating with each bundle a "master" m= line with media type
> "anymedia".  If the bundle is accepted, this m= line provides the
> transport information.  If the bundle is rejected, then the answerer
> accepts or rejects each constituent m= line separately, and that m=
> lines attributes determine the transport mechanism.
>
> However, MMT needs to be updated to match the concept that the
> anymedia RTP stream multiplexes what would be the RTP streams of one
> or more constituent media streams.  Currently, it assumes that
> identifying the media type of an incoming RTP packet is sufficient
> demultiplexing.  As a consequence of this, it needs to be modified so
> that if the answerer accepts the MTT bundle (and thus, the anymedia m=
> line), it still must accept or reject each constituent m= line
> individually.
>
> 3. Demultiplexing RTP
>
> MMT envisions that the media data is demultiplexed based on the
> payload types:  The m=anymedia description contains a set of a=mmttype
> attributes that (I think should) provide the mapping from the payload
> types to the mid values of the constituent m= lines.  (The codec for
> each payload type is provided by a=fmtp attributes of the m=anymedia
> description as normal.)

I disagree with that characterization. All of MMT, BUNDLE and TOGETHER 
agree that demultiplexing is based on SSRC. The payload type tells you 
what type the payload is.

>
> Note that this does not allow recovery of the payload types that would
> have been used for that codec and m= line (if multiple payload types
> for that m= line have the same codec).  But we assume that payload
> types are a transport matter and are of no interest to the higher
> layers of media processing.

I don't understand this sentence.
The payload type is "recovered" by looking at the 7-bit field called 
"payload type" in the RTP header of each RTP packet. It's hardly a 
transport matter at all.

>
> But perhaps instead, we should require that there be a matching
> payload type for the constituent m= line and use its a=fmtp
> attribute.  That allows us to not duplicate any codec specifications
> from the constituent m= lines into the m=anymedia line, and does allow
> recovering the "original" payload types.
>
> 4. RTCP
>
> My knowledge of RTCP is very thin.  But it's clear that incoming RTCP
> over the bundle transport must be demultiplexable to the point that we
> can separate which RTCP information is applicable to which constituent
> media stream.  My understanding is that RTCP information is identified
> by the SSRC of the media source it is reporting on, and that it is
> paired with the right media stream by examining the SSRC value in the
> RTP packet headers.  If so, demultiplexing RTCP requires no new
> protocol facilities.
>
> In regard to bandwidth consumption, I assume that the bundle's RTCP
> stream is essentially the union of the RTCP streams (or rather, the
> RCTP packets thereof, which are packed into the transport-level RTCP
> packets) that would be generated for each constituent m= line.  In
> particular, the frequency of the RTCP packets for a particular media
> stream is scaled to the needs of that media stream, rather than there
> being RTCP reports at same frequency for all media streams.  Thus, as
> long as the RTCP reports on each media stream are less than 5% of the
> RTP bandwidth for that media stream, the aggregate RTCP bandwidth is
> less then 5% of the aggregate RTP bandwidth.
>
> 5. Examples
>
> Incorporating these suggestions into
> draft-holmberg-mmusic-sdp-mmt-negotiation, I get this example:
>
>     SDP Offer (MMT offered)
>
>         v=0
>         o=alice 2890844526 2890844526 IN IP4 host.atlanta.com
>         s=
>         c=IN IP4 host.atlanta.com
>         t=0 0
>         a=group:MMT foo bar zoe
>         m=audio 10000 RTP/AVP 0 8 97
>         a=mid:foo
>         b=AS:200
>         a=rtpmap:0 PCMU/8000
>         a=rtpmap:8 PCMA/8000
>         a=rtpmap:97 iLBC/8000
>         m=video 20000 RTP/AVP 31 32
>         a=mid:bar
>         b=AS:1000
>         a=rtpmap:31 H261/90000
>         a=rtpmap:32 MPV/90000
>         m=anymedia 30000 RTP/AVP 0 8 97 31 32
>         a=mid:zoe
>         a=mmtype:foo 0 8 97
>         a=mmtype:bar 31 32

This seems to indicate that zoe "incorporates by reference" the foo and 
bar m-lines.
A better name in this case might be "m=transport" rather than "m=anymedia".

>
>     SDP Answer (MMT accepted, including both constituent streams)
>
>         v=0
>         o=bob 2808844564 2808844564 IN IP4 host.biloxi.com
>         s=
>         c=IN IP4 host.biloxi.com
>         t=0 0
>         a=group:MMT foo bar zoe
>         m=audio 1 RTP/AVP 0
>         a=mid:foo
>         a=rtpmap:0 PCMU/8000
>         m=video 1 RTP/AVP 32
>         a=mid:bar
>         a=rtpmap:32 MPV/90000
>         m=anymedia 60000 RTP/AVP 0 32
>         a=mid:zoe
>         a=mmtype:foo 0
>         a=mmtype:bar 32

Query: Are you making the port number 1 in the m=audio and m=video lines 
carry special meaning?

>
>     SDP Answer (MMT not accepted, but both constituent streams accepted)
>
>         v=0
>         o=bob 2808844564 2808844564 IN IP4 host.biloxi.com
>         s=
>         c=IN IP4 host.biloxi.com
>         t=0 0
>         m=audio 40000 RTP/AVP 0
>         a=mid:foo
>         a=rtpmap:0 PCMU/8000
>         m=video 50000 RTP/AVP 32
>         a=mid:bar
>         a=rtpmap:32 MPV/90000
>         m=anymedia 0 RTP/AVP 0
>         a=mid:zoe
>         [The a=mid attributes may be missing if the answerer does not
>         understand SDP grouping.]
>
>     SDP Offer with ICE (multiplexed media types offered)
>
>         v=0
>         o=alice 2890844526 2890844526 IN IP4 host.atlanta.com
>         s=
>         c=IN IP4 host.atlanta.com
>         t=0 0
>         a=group:MMT foo bar zoe
>         m=audio 10000 RTP/AVP 0 8 97
>         a=mid:foo
>         b=AS:200
>         a=rtpmap:0 PCMU/8000
>         a=rtpmap:8 PCMA/8000
>         a=rtpmap:97 iLBC/8000
>         a=candidate:1 1 UDP 1694498815 host.atlanta.com 10000 typ host
>         m=video 20000 RTP/AVP 31 32
>         a=mid:bar
>         b=AS:1000
>         a=rtpmap:31 H261/90000
>         a=rtpmap:32 MPV/90000
>         a=candidate:1 1 UDP 1694498815 host.atlanta.com 20000 typ host
>         m=anymedia 30000 RTP/AVP 0 8 97 31 32
>         a=mid:zoe
>         a=mmtype:foo 0 8 97
>         a=mmtype:bar 31 32
>         a=candidate:1 1 UDP 1694498815 host.atlanta.com 30000 typ host

Based on this offer, I don't quite see what this offer buys you above 
the MMT proposal.


From harald@alvestrand.no  Sat Feb  2 10:38:58 2013
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Subject: Re: [MMUSIC] Comments on draft-alvestrand-one-rtp-01
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On 02/02/2013 12:36 AM, Dale R. Worley wrote:
> general
>
> I think the concept is that the various media streams are
> demultiplexed out of the common RTP scheme by the fact that each
> payload type (codec type number) appears in at most one m= line of the
> bundle.  But that isn't stated explicitly anywhere.  In any case, the
> method for demultiplexing the RTP should be stated explicitly.

This is incorrect. Media streams are demultiplexed by SSRC.
Payload types are specified in at most one m= line because if the same 
payload type value appeared in two different m= lines, it would be 
impossible to tell from the incoming RTP packet which type it was 
intended to be.

>
> section 4
>
>     If the answerer wishes to refuse an entire m= section, while
>     accepting the RTP session otherwise, he MAY indicate this by setting
>     the port number of the relevant section to zero.
>
> ... and remove that m= line's mid value from the group attribute.
> RFC 5888 section 9.2

Good catch.

>
> section 7
>
> In regard to RTCP, it appears that RTCP reports on each SSRC
> separately.  I would expect that the bandwidth for the RTCP reports on
> a stream would be limited to 5% of the bandwidth for that stream.
> This keeps the total RTCP bandwidth below 5% of the total stream
> bandwidth; indeed it is what you would get if you simply merged the
> RTCP streams for the various media streams considered as separate RTP
> streams.  In this configuration, the RTCP for an audio stream would
> never exceed the RTP for the audio stream.  Although it does result in
> the frequency of reports differing between the various streams.

This is incorrect (or I'm serioiusly misunderstanding something). An 
RTCP sender or receiver report (RFC 3550 section 6.4) is supposed to 
report on all SSRCs, and the transmission interval is set for the RTP 
session, not separately for each media stream.


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


From harald@alvestrand.no  Sat Feb  2 15:13:14 2013
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Subject: [MMUSIC] Multicast and one-m-line-per-stream (Re: Thoughts on m= lines and media streams)
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Adam,

I hate to play the multicast card, because I'm fundamentally 
uninterested in multicast at the moment, but....

in your scheme 2, how would you represent multicast and transport-mixer 
based sessions?


On 02/01/2013 01:17 AM, Adam Roach wrote:
> Based on the conversations we've had so far, I've put together the 
> following summary of my thoughts on the topic of whether we want to 
> put multiple media streams in a single m= line.
>
> http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt
>
> Sorry for the relatively short lead time before next week's meeting -- 
> I had meant to put this kind of summary together a week or two ago, 
> but have been far busier than I anticipated.
>
> /a
> _______________________________________________
> mmusic mailing list
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From: Justin Uberti <juberti@google.com>
Date: Sat, 2 Feb 2013 22:04:47 -0800
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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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The prior commenters captured some of my concerns already, but here's a
quick summary of issues I noticed while reading over this draft:

1) "1a" and "2", as described, have fatal problems if the receiver rejects
the m= line that contains all the attributes (e.g. a=ice-ufrag, a=crypto,
etc)

2) In "1b", it's not clear what the purpose of the "a=group:MMT foo bar
zoe" line is.

3) Proposal "2" does not include the needed a=ssrc lines for demultiplexing
the received streams. IOW, proposal 2 requires RFC 5576 just like 1a and 1b.

4) The codec selection argument in Section 2.3.1 feels like a "Red Herring"
to me. Practically speaking, I think there are few, if any, real-world
settings where one might want to use a codec prioritization mechanism to
ask for a low-bitrate codec for stream A and a high-bitrate codec for
stream B (where the codecs used are different). A more realistic example
would be where one might want a lower bitrate  for a specific multi-rate
codec (e.g. Opus) to be used for stream A, and higher bitrate for stream B.
This of course can be handled with the per-ssrc fmtp attributes that are
specified in RFC 5576. However, I think in almost all cases, the receiver
is in a much worse position to make these decisions that the sender; the
sender, aware of the estimated bandwidth to the receiver, and the relative
value/complexity of its streams, can make more informed decisions about
bandwidth allocation, and perform them without any signaling delay.

5) The directional attribute discussion in Section 4 omits one important
consequence of the fact that a=ssrc indications are unidirectional, namely
that streams could be added or removed by both endpoints simultaneously
without causing a glare condition. m= lines, on the other hand, since they
are identified by their index, do not have this property; if both sides
decide to add a m= line at the same time, bad things happen.

6) The legacy interop argument in Section 4 feels a bit hand-wavy. Aside
from the one case of having video + presentation streams, I have not yet
heard of widely deployed equipment that uses multiple audio or video m=
lines, and certainly any such equipment will not understand BUNDLE or MMT.
Therefore a gateway is required, and any gateway that was performing BUNDLE
or MMT could certainly do the minor additional work to package the SDP as 1
m= line (remember, it already needs to add the a=ssrc attributes). Ergo, I
don't agree that the general case of N m= lines needs to be handled by all
RTCWEB implementations.

In summary, I see option "2" as a "1b" variant that comes with a more
verbose and redundant syntax, several shortcomings (#1 and #5 above), and
questionable upside.

-j-




On Thu, Jan 31, 2013 at 11:48 PM, Stach, Thomas <
thomas.stach@siemens-enterprise.com> wrote:

>
> Adam,
>
> thanks for analysing the various proposals and putting the draft together.
> From a quick read I have some issues with backward compatibility of your
> proposal to include
> "common" attributes only in the first m-section.
> Assuming that the examples in the draft are offers, an answerer that
> didn't implement any of the mechanisms would probably show the
> following behavior.
>
> Alternativ 1a and 2 have ICE candidates only for the first m-section.
> An answerer that doesn't understand the grouping semantics could -
> dependent on the network topology - end up having audio, but no video
> connectivity.
> With having other attributes only in the first m-section we could get
> other issues.
>
>
> Alternative 1b doesn't have this problem, but the offer would most likely
> be rejected resulting in no media at all.
> I preferred the original mmt proposal as it had additional m-sections,
> that could be accepted by a legacy answerer without the need for a GW that
> performs complex SDP operations.
>
>
> Regards
> Thomas
>
>
> Adam Roach wrote:
> > Based on the conversations we've had so far, I've put together the
> > following summary of my thoughts on the topic of whether we
> > want to put
> > multiple media streams in a single m= line.
> >
> > http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt
> >
> > Sorry for the relatively short lead time before next week's
> > meeting -- I
> > had meant to put this kind of summary together a week or two ago, but
> > have been far busier than I anticipated.
> >
> > /a
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

--0016369f9b5422ca4004d4cbc248
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<div dir=3D"ltr">The prior commenters captured some of my concerns already,=
 but here&#39;s a quick summary of issues I noticed while reading over this=
 draft:<div style><br></div><div style>1) &quot;1a&quot; and &quot;2&quot;,=
 as described, have fatal problems if the receiver rejects the m=3D line th=
at contains all the attributes (e.g. a=3Dice-ufrag, a=3Dcrypto, etc)</div>

<div style><br></div><div style>2) In &quot;1b&quot;, it&#39;s not clear wh=
at the purpose of the &quot;<span style=3D"color:rgb(0,0,0);white-space:pre=
-wrap">a=3Dgroup:MMT foo bar zoe&quot; line is. </span><br></div><div style=
>

<span style=3D"color:rgb(0,0,0);white-space:pre-wrap"><br></span></div><div=
 style>3) Proposal &quot;2&quot; does not include the needed a=3Dssrc lines=
 for demultiplexing the received streams. IOW, proposal 2 requires RFC 5576=
 just like 1a and 1b.</div>

<div style><br></div><div style>4) The codec selection argument in Section =
2.3.1 feels like a &quot;Red Herring&quot; to me. Practically speaking, I t=
hink there are few, if any, real-world settings where one might want to use=
 a codec prioritization mechanism to ask for a low-bitrate codec for stream=
 A and a high-bitrate codec for stream B (where the codecs used are differe=
nt). A more realistic example would be where one might want a lower bitrate=
 =C2=A0for a specific multi-rate codec (e.g. Opus) to be used for stream A,=
 and higher bitrate for stream B. This of course can be handled with the pe=
r-ssrc fmtp attributes that are specified in RFC 5576. However, I think in =
almost all cases, the receiver is in a much worse position to make these de=
cisions that the sender; the sender, aware of the estimated bandwidth to th=
e receiver, and the relative value/complexity of its streams, can make more=
 informed decisions about bandwidth allocation, and perform them without an=
y signaling delay.</div>

<div style><br></div><div style>5) The directional attribute discussion in =
Section 4 omits one important consequence of the fact that a=3Dssrc indicat=
ions are unidirectional, namely that streams could be added or removed by b=
oth endpoints simultaneously without causing a glare condition. m=3D lines,=
 on the other hand, since they are identified by their index, do not have t=
his property; if both sides decide to add a m=3D line at the same time, bad=
 things happen.</div>

<div style><br></div><div style>6) The legacy interop argument in Section 4=
 feels a bit hand-wavy. Aside from the one case of having video + presentat=
ion streams, I have not yet heard of widely deployed equipment that uses mu=
ltiple audio or video m=3D lines, and certainly any such equipment will not=
 understand BUNDLE or MMT. Therefore a gateway is required, and any gateway=
 that was performing BUNDLE or MMT could certainly do the minor additional =
work to package the SDP as 1 m=3D line (remember, it already needs to add t=
he a=3Dssrc attributes). Ergo, I don&#39;t agree that the general case of N=
 m=3D lines needs to be handled by all RTCWEB implementations.</div>

<div style><br></div><div style>In summary, I see option &quot;2&quot; as a=
 &quot;1b&quot; variant that comes with a more verbose and redundant syntax=
, several shortcomings (#1 and #5 above), and questionable upside.</div>

<div style><br></div><div style>-j-</div><div style><br></div><div style><b=
r></div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote"=
>On Thu, Jan 31, 2013 at 11:48 PM, Stach, Thomas <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:thomas.stach@siemens-enterprise.com" target=3D"_blank">thomas=
.stach@siemens-enterprise.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
Adam,<br>
<br>
thanks for analysing the various proposals and putting the draft together.<=
br>
>From a quick read I have some issues with backward compatibility of your pr=
oposal to include<br>
&quot;common&quot; attributes only in the first m-section.<br>
Assuming that the examples in the draft are offers, an answerer that<br>
didn&#39;t implement any of the mechanisms would probably show the<br>
following behavior.<br>
<br>
Alternativ 1a and 2 have ICE candidates only for the first m-section.<br>
An answerer that doesn&#39;t understand the grouping semantics could -<br>
dependent on the network topology - end up having audio, but no video<br>
connectivity.<br>
With having other attributes only in the first m-section we could get other=
 issues.<br>
<br>
<br>
Alternative 1b doesn&#39;t have this problem, but the offer would most like=
ly<br>
be rejected resulting in no media at all.<br>
I preferred the original mmt proposal as it had additional m-sections,<br>
that could be accepted by a legacy answerer without the need for a GW that =
performs complex SDP operations.<br>
<br>
<br>
Regards<br>
<span class=3D"HOEnZb"><font color=3D"#888888">Thomas<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
Adam Roach wrote:<br>
&gt; Based on the conversations we&#39;ve had so far, I&#39;ve put together=
 the<br>
&gt; following summary of my thoughts on the topic of whether we<br>
&gt; want to put<br>
&gt; multiple media streams in a single m=3D line.<br>
&gt;<br>
&gt; <a href=3D"http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt" ta=
rget=3D"_blank">http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt</a>=
<br>
&gt;<br>
&gt; Sorry for the relatively short lead time before next week&#39;s<br>
&gt; meeting -- I<br>
&gt; had meant to put this kind of summary together a week or two ago, but<=
br>
&gt; have been far busier than I anticipated.<br>
&gt;<br>
&gt; /a<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</div></div></blockquote></div><br></div>

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Subject: [MMUSIC] More thoughts on multiple media streams in a single m= line
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Sorry for the short lead time, but here's a quick read for the plane.

Peter Thatcher and I have put together a writeup of our thoughts on the
advantages of putting multiple media streams in a single m= line.

http://www.ietf.org/id/draft-pthatcher-mmusic-many-sources-00.txt

We look forward to discussing in more detail at the interim.

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<div dir=3D"ltr">Sorry for the short lead time, but here&#39;s a quick read=
 for the plane.<div><br></div><div>Peter Thatcher and I have put together a=
 writeup of our thoughts on the advantages of putting multiple media stream=
s in a single m=3D line.</div>

<div><br></div><div><a href=3D"http://www.ietf.org/id/draft-pthatcher-mmusi=
c-many-sources-00.txt">http://www.ietf.org/id/draft-pthatcher-mmusic-many-s=
ources-00.txt</a><br></div><div><br></div><div style>We look forward to dis=
cussing in more detail at the interim.</div>

</div>

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <mmusic@ietf.org>
Date: Mon, 4 Feb 2013 16:36:10 +0200
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Subject: [MMUSIC] Some more about SDP support for multiple streams
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Hi,
Sorry for the last minute but here are more thoughts on the subject.

I will not be able to attend the interim meeting and am not sure if can join
this topic since it may be too late at night for me.

Roni Even

________________________________________
From: internet-drafts@ietf.org [internet-drafts@ietf.org]
Sent: Monday, February 04, 2013 4:30 PM
To: Roni Even
Cc: jonathan@vidyo.com; bill.wu@huawei.com
Subject: New Version Notification for
draft-even-mmusic-multiple-streams-00.txt

A new version of I-D, draft-even-mmusic-multiple-streams-00.txt
has been successfully submitted by Roni Even and posted to the IETF
repository.

Filename:        draft-even-mmusic-multiple-streams
Revision:        00
Title:           Describing multiple RTP media streams in SDP
Creation date:   2013-02-04
WG ID:           Individual Submission
Number of pages: 10
URL:
http://www.ietf.org/internet-drafts/draft-even-mmusic-multiple-streams-00.tx
t
Status:
http://datatracker.ietf.org/doc/draft-even-mmusic-multiple-streams
Htmlized:
http://tools.ietf.org/html/draft-even-mmusic-multiple-streams-00


Abstract:
   This document describes issues when describing multiple RTP streams
   in a single RTP session using SDP.  The document looks at current
   solutions and provides directions to address the issues.




The IETF Secretariat=


From pkyzivat@alum.mit.edu  Mon Feb  4 07:18:22 2013
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Subject: Re: [MMUSIC] February Interim registration and logistics
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Is there a detailed agenda, with actual start times?
I've looked but can't find one.

	Thanks,
	Paul

On 1/18/13 2:33 PM, Hadriel Kaplan wrote:
> Howdy,
> We're pleased to host the upcoming interim meetings of IETF RTCWeb, IETF MMUSIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, 2013, at our Corporate Galactic Headquarters in Bedford, Massachusetts.
>
> Please register at the following page if you plan to attend:
> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c
>
> The logistics/venue info page is here:
> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics
>
> You should NOT register if you're only going to attend remotely.  Details for remote participation will be sent out later on the relevant mailing lists.
>
> We hope to see you here!
> And bring warm clothing - it's slightly chilly. ;)
>
> -hadriel
>
> p.s. We've tested the registration page on several browsers and it works generally, except for on Lynx, Links, or Elinks. (Lynx and Links because they don't support Javascript, but why it doesn't work on Elinks w/SpiderMonkey I haven't looked into)  The logistics page seems to work on all browsers.
> :)
>
>
>


From ron.even.tlv@gmail.com  Mon Feb  4 08:56:32 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <rtcweb-chairs@tools.ietf.org>, <mmusic-chairs@tools.ietf.org>
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Subject: Re: [MMUSIC] [rtcweb] Boston meeting agenda
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Hi,
Is there a final agenda with time schedule. There is no relation between the
agenda here and i=the one for mmusic and rtcweb in the reference.
I would like to attend from remote and since there is time difference it
will be helpful to plan ahead.
Thanks
Roni Even who will try to attend from remote being 7 hours ahead of Boston

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Stefan H?kansson LK
Sent: 30 January, 2013 11:51 AM
To: mmusic@ietf.org; rtcweb@ietf.org; public-webrtc@w3.org;
public-media-capture@w3.org
Subject: [rtcweb] Boston meeting agenda

All,

below is a merged agenda covering both W3C and IETF parts of the meeting.
Details are likely to change - the main message is that W3C meetings will be
before lunch, and IETF after. Before lunch could mean
8:30 - 12:30, after lunch 13:30 - 17:30; but the exact times are TBC. A hard
stop is planned for 17:00 on Feb 7th as many have planes to catch.

(The sources used to assemble the agenda below are the W3C meeting wiki's,
[1][2], and the mail sent to mmusic and rtcweb Jan 14th [3]).

--Stefan (for all involved chairs)

Feb 5th
=======
Morning session: W3C Media Capture TF
-------------------------------------
* device enumeration
* error handling
* "immediate stream" gUM
* Resource reservation

Afternoon session: IETF mmusic and rtcweb WGs
---------------------------------------------
* Multiple Flow SDP representation
* SDP for Data channels

Feb 6th
=======
Morning session: W3C Media Capture TF + WebRTC WG
-------------------------------------------------
* Identity aspects of MediaStreams (MediaCap)
* Settings/constraints (MediaCap)
* Recording (MediaCap)
* Call flow (WebRTC)

Afternoon session: IETF mmusic and rtcweb WGs
---------------------------------------------
* SDP Grouping (Bundle/MMT/one-rtp)
* Mapping stream/track label concepts to SDP identifiers

Feb 7th
=======
Morning session: W3C WebRTC WG
------------------------------
* MediaStream and MediaStreamTrack - SDP interface
* Error handling
* Data channel
* Stats

Afternoon session: IETF mmusic and rtcweb WGs
---------------------------------------------
* Trickle ICE
* JSEP

[1] http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013
[2] http://www.w3.org/2011/04/webrtc/wiki/February_6_-_February_7_2013
[3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg06069.html




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


From btv1==7476154b17d==HKaplan@acmepacket.com  Mon Feb  4 10:25:35 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] February Interim registration and logistics
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Hi Paul,
An email confirmation and with some logistics info was just sent out.
The breakfast starts at 8am each day, with the meeting starting at 8:30-ish=
.
But I'm not the one handling detailed meeting agenda with presentations and=
 stuff - I assumed that was up to the Chairs.
The last one I saw was this one:
http://www.ietf.org/mail-archive/web/mmusic/current/msg10168.html

-hadriel


On Feb 4, 2013, at 10:18 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Is there a detailed agenda, with actual start times?
> I've looked but can't find one.
>=20
> 	Thanks,
> 	Paul
>=20
> On 1/18/13 2:33 PM, Hadriel Kaplan wrote:
>> Howdy,
>> We're pleased to host the upcoming interim meetings of IETF RTCWeb, IETF=
 MMUSIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, 2013, at=
 our Corporate Galactic Headquarters in Bedford, Massachusetts.
>>=20
>> Please register at the following page if you plan to attend:
>> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c
>>=20
>> The logistics/venue info page is here:
>> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics
>>=20
>> You should NOT register if you're only going to attend remotely.  Detail=
s for remote participation will be sent out later on the relevant mailing l=
ists.
>>=20
>> We hope to see you here!
>> And bring warm clothing - it's slightly chilly. ;)
>>=20
>> -hadriel
>>=20
>> p.s. We've tested the registration page on several browsers and it works=
 generally, except for on Lynx, Links, or Elinks. (Lynx and Links because t=
hey don't support Javascript, but why it doesn't work on Elinks w/SpiderMon=
key I haven't looked into)  The logistics page seems to work on all browser=
s.
>> :)
>>=20
>>=20
>>=20
>=20


From roni.even@mail01.huawei.com  Mon Feb  4 06:34:31 2013
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From: Roni Even <roni.even@mail01.huawei.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Some more about SDP support for multiple streams
Thread-Index: AQHOAuS4JTNGZUN1QkOqimA9NpZubg==
Date: Mon, 4 Feb 2013 14:34:22 +0000
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Subject: [MMUSIC] Some more about SDP support for multiple streams
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Hi,
Sorry for the last minute but here are more thoughts on the subject.

I will not be able to attend the interim meeting and am not sure if can joi=
n this topic since it may be too late at night for me.

Roni Even

________________________________________
From: internet-drafts@ietf.org [internet-drafts@ietf.org]
Sent: Monday, February 04, 2013 4:30 PM
To: Roni Even
Cc: jonathan@vidyo.com; bill.wu@huawei.com
Subject: New Version Notification for draft-even-mmusic-multiple-streams-00=
.txt

A new version of I-D, draft-even-mmusic-multiple-streams-00.txt
has been successfully submitted by Roni Even and posted to the
IETF repository.

Filename:        draft-even-mmusic-multiple-streams
Revision:        00
Title:           Describing multiple RTP media streams in SDP
Creation date:   2013-02-04
WG ID:           Individual Submission
Number of pages: 10
URL:             http://www.ietf.org/internet-drafts/draft-even-mmusic-mult=
iple-streams-00.txt
Status:          http://datatracker.ietf.org/doc/draft-even-mmusic-multiple=
-streams
Htmlized:        http://tools.ietf.org/html/draft-even-mmusic-multiple-stre=
ams-00


Abstract:
   This document describes issues when describing multiple RTP streams
   in a single RTP session using SDP.  The document looks at current
   solutions and provides directions to address the issues.




The IETF Secretariat=

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From: "Vojislav Vucetic (vvucetic)" <vvucetic@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Hadriel Kaplan <HKaplan@acmepacket.com>
Thread-Topic: [rtcweb] February Interim registration and logistics
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Subject: Re: [MMUSIC] [rtcweb] February Interim registration and logistics
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What about the remote access to the sessions?

Thank you,

vvv

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Paul Kyzivat
Sent: Monday, February 04, 2013 10:18 AM
To: Hadriel Kaplan
Cc: rtcweb@ietf.org; mmusic@ietf.org (E-mail)
Subject: Re: [rtcweb] February Interim registration and logistics

Is there a detailed agenda, with actual start times?
I've looked but can't find one.

	Thanks,
	Paul

On 1/18/13 2:33 PM, Hadriel Kaplan wrote:
> Howdy,
> We're pleased to host the upcoming interim meetings of IETF RTCWeb, IETF =
MMUSIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, 2013, at =
our Corporate Galactic Headquarters in Bedford, Massachusetts.
>
> Please register at the following page if you plan to attend:
> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c
>
> The logistics/venue info page is here:
> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics
>
> You should NOT register if you're only going to attend remotely.  Details=
 for remote participation will be sent out later on the relevant mailing li=
sts.
>
> We hope to see you here!
> And bring warm clothing - it's slightly chilly. ;)
>
> -hadriel
>
> p.s. We've tested the registration page on several browsers and it works =
generally, except for on Lynx, Links, or Elinks. (Lynx and Links because th=
ey don't support Javascript, but why it doesn't work on Elinks w/SpiderMonk=
ey I haven't looked into)  The logistics page seems to work on all browsers=
.
> :)
>
>
>

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

From spromano@unina.it  Mon Feb  4 10:40:13 2013
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To: Vojislav Vucetic (vvucetic) <vvucetic@cisco.com>
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Subject: Re: [MMUSIC] [rtcweb] February Interim registration and logistics
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+1 for info about remote participation.

Thanks,

Simon

Il giorno 04/feb/2013, alle ore 16:34, Vojislav Vucetic (vvucetic) ha =
scritto:

> What about the remote access to the sessions?
>=20
> Thank you,
>=20
> vvv
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Paul Kyzivat
> Sent: Monday, February 04, 2013 10:18 AM
> To: Hadriel Kaplan
> Cc: rtcweb@ietf.org; mmusic@ietf.org (E-mail)
> Subject: Re: [rtcweb] February Interim registration and logistics
>=20
> Is there a detailed agenda, with actual start times?
> I've looked but can't find one.
>=20
> 	Thanks,
> 	Paul
>=20
> On 1/18/13 2:33 PM, Hadriel Kaplan wrote:
>> Howdy,
>> We're pleased to host the upcoming interim meetings of IETF RTCWeb, =
IETF MMUSIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, =
2013, at our Corporate Galactic Headquarters in Bedford, Massachusetts.
>>=20
>> Please register at the following page if you plan to attend:
>> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c
>>=20
>> The logistics/venue info page is here:
>> http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics
>>=20
>> You should NOT register if you're only going to attend remotely.  =
Details for remote participation will be sent out later on the relevant =
mailing lists.
>>=20
>> We hope to see you here!
>> And bring warm clothing - it's slightly chilly. ;)
>>=20
>> -hadriel
>>=20
>> p.s. We've tested the registration page on several browsers and it =
works generally, except for on Lynx, Links, or Elinks. (Lynx and Links =
because they don't support Javascript, but why it doesn't work on Elinks =
w/SpiderMonkey I haven't looked into)  The logistics page seems to work =
on all browsers.
>> :)
>>=20
>>=20
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

                     					       _\\|//_
                           				      ( O-O )
   ~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~
                    				Simon Pietro Romano
             				 Universita' di Napoli Federico =
II
                		     Computer Engineering Department=20
	             Phone: +39 081 7683823 -- Fax: +39 081 7683816
                                           e-mail: spromano@unina.it

		    <<Molti mi dicono che lo scoraggiamento =CB l'alibi =
degli=20
		    idioti. Ci rifletto un istante; e mi scoraggio>>. =
Magritte.
               			                     oooO
  ~~~~~~~~~~~~~~~~~~~~~~~(   )~~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
					                 \ (            =
(   )
			                                  \_)          ) =
/
                                                                       =
(_/






--Apple-Mail=_4BAF32B8-13CD-4832-8889-B9E53A744B33
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	charset=iso-8859-1

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">+1 =
for info about remote =
participation.<div><br></div><div>Thanks,</div><div><br></div><div>Simon</=
div><div><br><div><div>Il giorno 04/feb/2013, alle ore 16:34, Vojislav =
Vucetic (vvucetic) ha scritto:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>What =
about the remote access to the sessions?<br><br>Thank =
you,<br><br>vvv<br><br>-----Original Message-----<br>From: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat<br>Sent: =
Monday, February 04, 2013 10:18 AM<br>To: Hadriel Kaplan<br>Cc: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a =
href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a> (E-mail)<br>Subject: =
Re: [rtcweb] February Interim registration and logistics<br><br>Is there =
a detailed agenda, with actual start times?<br>I've looked but can't =
find one.<br><br><span class=3D"Apple-tab-span" style=3D"white-space:pre">=
	</span>Thanks,<br><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Paul<br><br>On 1/18/13 2:33 PM, =
Hadriel Kaplan wrote:<br><blockquote =
type=3D"cite">Howdy,<br></blockquote><blockquote type=3D"cite">We're =
pleased to host the upcoming interim meetings of IETF RTCWeb, IETF =
MMUSIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, 2013, =
at our Corporate Galactic Headquarters in Bedford, =
Massachusetts.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Please register =
at the following page if you plan to attend:<br></blockquote><blockquote =
type=3D"cite"><a =
href=3D"http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c">http:=
//info.acmepacket.com/acme-packet-hosting-ietf-and-w3c</a><br></blockquote=
><blockquote type=3D"cite"><br></blockquote><blockquote type=3D"cite">The =
logistics/venue info page is here:<br></blockquote><blockquote =
type=3D"cite"><a =
href=3D"http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logist=
ics">http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics=
</a><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">You should NOT =
register if you're only going to attend remotely. &nbsp;Details for =
remote participation will be sent out later on the relevant mailing =
lists.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">We hope to see =
you here!<br></blockquote><blockquote type=3D"cite">And bring warm =
clothing - it's slightly chilly. ;)<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite">-hadriel<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">p.s. We've =
tested the registration page on several browsers and it works generally, =
except for on Lynx, Links, or Elinks. (Lynx and Links because they don't =
support Javascript, but why it doesn't work on Elinks w/SpiderMonkey I =
haven't looked into) &nbsp;The logistics page seems to work on all =
browsers.<br></blockquote><blockquote =
type=3D"cite">:)<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><br>_______________________________________=
________<br>rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br>_____________________________________________=
__<br>rtcweb mailing =
list<br>rtcweb@ietf.org<br>https://www.ietf.org/mailman/listinfo/rtcweb<br=
><br></div></blockquote></div><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: -webkit-auto; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><div>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
	</span><span class=3D"Apple-converted-space">&nbsp;</span>&nbsp; =
&nbsp; &nbsp; _\\|//_</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
</span>&nbsp; &nbsp; &nbsp;&nbsp;( O-O )</div><div>&nbsp; =
&nbsp;~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~</div><di=
v>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
</span>Simon Pietro Romano</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;<span class=3D"Apple-tab-span" style=3D"white-space: pre; =
">				</span><span =
class=3D"Apple-converted-space">&nbsp;</span>Universita' di Napoli =
Federico II</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;<span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
	</span>&nbsp; &nbsp; &nbsp;Computer Engineering =
Department&nbsp;</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">	</span>&nbsp; &nbsp; &nbsp;&nbsp; &nbsp; =
&nbsp; &nbsp; Phone: +39 081 7683823 -- Fax: +39 081 =
7683816</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;e-mail: <a =
href=3D"mailto:spromano@unina.it">spromano@unina.it</a></div><div><br></di=
v><div><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">		=
</span>&nbsp; &nbsp; &lt;&lt;Molti mi dicono che lo scoraggiamento =CB =
l'alibi degli&nbsp;</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">		</span>&nbsp;&nbsp; =
&nbsp;idioti. Ci rifletto un istante; e mi scoraggio&gt;&gt;. =
Magritte.</div><div>&nbsp; &nbsp; &nbsp; &nbsp;&nbsp; &nbsp; &nbsp; =
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">			=
</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;oooO</div><div>&nbsp; ~~~~~~~~~~~~~~~~~~~~~~~( &nbsp; =
)~~~&nbsp;Oooo~~~~~~~~~~~~~~~~~~~~~~~~~</div><div><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
	</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;\ ( &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;( &nbsp; =
)</div><div><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
		</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
\_) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;) /</div><div>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp;(_/</div></div><div><br></div></div></span><br =
class=3D"Apple-interchange-newline"></div></span><br =
class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></body></html>=

--Apple-Mail=_4BAF32B8-13CD-4832-8889-B9E53A744B33--

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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: mmusic@ietf.org
Subject: [MMUSIC] Comments on draft-ietf-mmusic-sctp-sdp-00
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A couple of nits about draft-ietf-mmusic-sctp-sdp-00:

section 7

Examples contain "*" in m= lines instead of payload types.  The
resader will probably assume that "*" stands for a sequence of payload
types, but it's best to have examples be valid.  Replacing all the
"*"s with "0 8" would suffice to make the examples valid:

              m=image 54111 SCTP 0 8
              c=IN IP4 192.0.2.2
              a=setup:actpass
              a=connection:new

section 7.2

The text suggests that the intention is to reuse the existing SCTP
connection, but the SDP offer and answer both contain
"a=connection:new".  Shouldn't these use "a=connection:existing"?
Compare with RFC 4145 section 7.3.

Dale

From btv1==7476154b17d==HKaplan@acmepacket.com  Mon Feb  4 13:11:29 2013
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Vojislav Vucetic (vvucetic)" <vvucetic@CISCO.COM>
Thread-Topic: [MMUSIC] [rtcweb] February Interim registration and logistics
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Cc: "fluffy@cisco.com \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] February Interim registration and logistics
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It's a WebEx session, graciously offered/hosted by Cisco.  I haven't seen w=
hat the URL/dial-in info is for it yet.

-hadriel


On Feb 4, 2013, at 10:34 AM, Vojislav Vucetic (vvucetic) <vvucetic@CISCO.CO=
M> wrote:

> What about the remote access to the sessions?=20
>=20
> Thank you,=20
>=20
> vvv=20
>=20
> -----Original Message-----=20
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Paul Kyzivat=20
> Sent: Monday, February 04, 2013 10:18 AM=20
> To: Hadriel Kaplan=20
> Cc: rtcweb@ietf.org; mmusic@ietf.org (E-mail)=20
> Subject: Re: [rtcweb] February Interim registration and logistics=20
>=20
> Is there a detailed agenda, with actual start times?=20
> I've looked but can't find one.=20
>=20
> Thanks,=20
> Paul=20
>=20
> On 1/18/13 2:33 PM, Hadriel Kaplan wrote:=20
> > Howdy,=20
> > We're pleased to host the upcoming interim meetings of IETF RTCWeb, IET=
F MMUSIC, W3C WebRTC and W3C Media Capture groups, on February 5-7, 2013, a=
t our Corporate Galactic Headquarters in Bedford, Massachusetts.=20
> >=20
> > Please register at the following page if you plan to attend:=20
> > http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c=20
> >=20
> > The logistics/venue info page is here:=20
> > http://info.acmepacket.com/acme-packet-hosting-ietf-and-w3c/logistics=20
> >=20
> > You should NOT register if you're only going to attend remotely. Detail=
s for remote participation will be sent out later on the relevant mailing l=
ists.=20
> >=20
> > We hope to see you here!=20
> > And bring warm clothing - it's slightly chilly. ;)=20
> >=20
> > -hadriel=20
> >=20
> > p.s. We've tested the registration page on several browsers and it work=
s generally, except for on Lynx, Links, or Elinks. (Lynx and Links because =
they don't support Javascript, but why it doesn't work on Elinks w/SpiderMo=
nkey I haven't looked into) The logistics page seems to work on all browser=
s.=20
> > :)=20
> >=20
> >=20
> >=20
>=20
> _______________________________________________=20
> rtcweb mailing list=20
> rtcweb@ietf.org=20
> https://www.ietf.org/mailman/listinfo/rtcweb=20
> _______________________________________________=20
> mmusic mailing list=20
> mmusic@ietf.org=20
> https://www.ietf.org/mailman/listinfo/mmusic=20


From eckelcu@cisco.com  Mon Feb  4 18:28:10 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "mmusic (E-mail)" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Draft on Multimedia concepts and relations for Intermim
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Hi Magnus and Bo,

Thanks for putting this together, and more importantly, for attaching a cop=
y in the e-mail so I was able to access it on the plane without internet.
I agree with most of the reasoning in the various sections; however, there =
are a couple related concepts for which I'd like more clarification.

1) MediaStream
Is a MediaStream restricted to a single source? I did not think it was, but=
 the description on page 8 seems to imply this restriction. For example, co=
uld I have a single MediaStream with the following 4 MediaStreamTracks, whe=
re source 1 and source 2 are two video cameras from the same endpoint, or a=
 mic and an camera from the same endpoint:

Source 1 - encoding 1
Source 1 - encoding 2
Source 2 - encoding 1
Source 2 - encoding 2

2) PeerConnection
Later, on page 18, a related restriction of a single encoding per PC is sug=
gested. In the case of a layered encoding, this seems to imply each layer b=
eing in its own PC. Is that the recommendation being suggested?

Thanks,
Charles=20



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Magnus Westerlund
> Sent: Thursday, January 31, 2013 2:46 AM
> To: mmusic (E-mail); rtcweb@ietf.org
> Subject: [rtcweb] Draft on Multimedia concepts and relations for Intermim
>=20
> Hi,
>=20
> Attached is a draft on multimedia concepts and relations that we believe
> relevant for the upcoming interim. This includes commonalties between
> WebRTC and CLUE and other RTP usages.
>=20
> Sorry for the late submission, i do hope you have time to review it. At
> will also be properly submitted to IETF as soon as we figure out why the
> submission tools refuses to process it.
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------

From worley@shell01.TheWorld.com  Mon Feb  4 19:28:06 2013
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Date: Mon, 4 Feb 2013 22:27:06 -0500 (EST)
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From: worley@ariadne.com (Dale R. Worley)
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In-reply-to: <510D5D3C.1060103@alvestrand.no> (harald@alvestrand.no)
References: <201302012336.r11NaMnt1017265@shell01.TheWorld.com> <510D5D3C.1060103@alvestrand.no>
Subject: Re: [MMUSIC] Comments on draft-alvestrand-one-rtp-01
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> From: Harald Alvestrand <harald@alvestrand.no>
> 
> > I think the concept is that the various media streams are
> > demultiplexed out of the common RTP scheme by the fact that each
> > payload type (codec type number) appears in at most one m= line of the
> > bundle.  But that isn't stated explicitly anywhere.  In any case, the
> > method for demultiplexing the RTP should be stated explicitly.
> 
> This is incorrect. Media streams are demultiplexed by SSRC.

This is more complex than it first appears, including ambiguity in the
term "media stream".  I'm considering the problem of attributing each
incoming RTP packet to a particular m= line in the SDP.  But that is
based on an assumption that identifying the correct m= line is
important to the application layer above.  More about that later...

> > In regard to RTCP, it appears that RTCP reports on each SSRC
> > separately.  I would expect that the bandwidth for the RTCP reports on
> > a stream would be limited to 5% of the bandwidth for that stream.
> > This keeps the total RTCP bandwidth below 5% of the total stream
> > bandwidth; indeed it is what you would get if you simply merged the
> > RTCP streams for the various media streams considered as separate RTP
> > streams.  In this configuration, the RTCP for an audio stream would
> > never exceed the RTP for the audio stream.  Although it does result in
> > the frequency of reports differing between the various streams.
> 
> This is incorrect (or I'm serioiusly misunderstanding something). An 
> RTCP sender or receiver report (RFC 3550 section 6.4) is supposed to 
> report on all SSRCs, and the transmission interval is set for the RTP 
> session, not separately for each media stream.

Your understanding is much more likely to be correct than mine, and
RFC 3550 section 6.4 supports your statement.  But as somebody noted,
if there is both an audio and video stream, RTCP reports should be
sent sufficiently frequently for the video stream that the information
contained in them for the audio stream approaches the bandwidth of the
audio stream itself.

We may not need to care, given the RTCP is still a small fraction of
the overall bandwidth demand, but it may desirable to modify the rules
in section 6.4 to allow less frequent reports for sources that are
producing packets less frequently.

Dale

From fluffy@cisco.com  Mon Feb  4 19:35:56 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>, mmusic WG <mmusic@ietf.org>
Thread-Topic: Webex for  WebRTC / RTCWeb / MMusic interim meeting
Thread-Index: AQHOA1Hnb0mXRjeshEmbqUcNz+ZEDw==
Date: Tue, 5 Feb 2013 03:35:05 +0000
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Subject: [MMUSIC] Webex for  WebRTC / RTCWeb / MMusic interim meeting
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--_000_C5E08FE080ACFD4DAE31E4BDBF944EB1133ABA13xmbalnx02ciscoc_
Content-Type: text/plain; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable


Here is the dial in information for the interim meeting for WebRTC, RTCWeb,=
 and MMUSIC over the next few day. It is the same conference for all the me=
etings on all days to avoid the need to switch back and forth.

I find the easiest way to use this is join the web page, but then have it d=
ial you back to phone a PSTN phone line.

Please mute yourself when not speaking.

(Note the bridge is starting up at 8:00 but the meeting does not start unti=
l 8:30 )




Hello ,

Cullen Jennings invites you to attend this online meeting.

Topic: WebRTC / RTCWeb / MMusic
Date: Every 1 day, from Tuesday, February 5, 2013 to Thursday, February 7, =
2013
Time: 8:00 am, Eastern Standard Time (New York, GMT-05:00)
Meeting Number: 200 160 998
Meeting Password: acme


-------------------------------------------------------
To join the online meeting (Now from mobile devices!)
-------------------------------------------------------
1. Go to https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&UID=3D0&PW=
=3DNN2I3YzUxYjJk&RT=3DMiMxMQ%3D%3D
2. Enter your name and email address.
3. Enter the meeting password: acme
4. Click "Join Now".

To view in other time zones or languages, please click the link:
https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&UID=3D0&PW=3DNN2I3Y=
zUxYjJk&ORT=3DMiMxMQ%3D%3D

----------------------------------------------------------------
ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
----------------------------------------------------------------

The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpita=
s area and (866) 349-3520 for the RTP area.

Please dial the local access number for your area from the list below:
- San Jose/Milpitas (408) area: 525-6800
- RTP (919) area: 392-3330

-------------------------------------------------------
To join the teleconference only
-------------------------------------------------------
1. Dial into Cisco WebEx (view all Global Access Numbers at
http://cisco.com/en/US/about/doing_business/conferencing/index.html
2. Follow the prompts to enter the Meeting Number (listed above) or Access =
Code followed by the # sign.

San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330

US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117

India: +91.80.4350.1111 Germany: +49.619.6773.9002

Japan: +81.3.5763.9394 China: +86.10.8515.5666

-------------------------------------------------------
For assistance
-------------------------------------------------------
1. Go to https://cisco.webex.com/ciscosales/mc
2. On the left navigation bar, click "Support".

You can contact me at:
fluffy@cisco.com<mailto:fluffy@cisco.com>


To add this meeting to your calendar program (for example Microsoft Outlook=
), click this link:
https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&UID=3D0&ICS=3DMI&LD=
=3D1&RD=3D2&ST=3D1&SHA2=3DAAAAAkgeSvE1NzcP1FAQFi0Kzp6vvYe1NCey7X109lhs/gVs&=
RT=3DMiMxMQ%3D%3D


--_000_C5E08FE080ACFD4DAE31E4BDBF944EB1133ABA13xmbalnx02ciscoc_
Content-Type: text/html; charset="us-ascii"
Content-ID: <2B4D0BA1CF5D5C4D8114892C36FE77EA@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; ">
<div><br>
</div>
<div>Here is the dial in information for the interim meeting for WebRTC, RT=
CWeb, and MMUSIC over the next few day. It is the same conference for all t=
he meetings on all days to avoid the need to switch back and forth.&nbsp;</=
div>
<div><br>
</div>
<div>I find the easiest way to use this is join the web page, but then have=
 it dial you back to phone a PSTN phone line.&nbsp;</div>
<div><br>
</div>
<div>Please mute yourself when not speaking.&nbsp;</div>
<div><br>
</div>
<div>(Note the bridge is starting up at 8:00 but the meeting does not start=
 until 8:30 )</div>
<div><br>
</div>
<div><br>
<div>
<blockquote type=3D"cite"><font face=3D"Tahoma, Arial, sans-serif, Helvetic=
a, Geneva" size=3D"2"><br>
<br>
Hello , <br>
<br>
Cullen Jennings invites you to attend this online meeting. <br>
<br>
Topic: WebRTC / RTCWeb / MMusic <br>
Date: Every 1 day, from Tuesday, February 5, 2013 to Thursday, February 7, =
2013 <br>
Time: 8:00 am, Eastern Standard Time (New York, GMT-05:00) <br>
Meeting Number: 200 160 998 <br>
Meeting Password: acme <br>
<br>
<br>
------------------------------------------------------- <br>
To join the online meeting (Now from mobile devices!) <br>
------------------------------------------------------- <br>
1. Go to <a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D216374677=
&amp;UID=3D0&amp;PW=3DNN2I3YzUxYjJk&amp;RT=3DMiMxMQ%3D%3D" target=3D"_blank=
">
https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&amp;UID=3D0&amp;PW=
=3DNN2I3YzUxYjJk&amp;RT=3DMiMxMQ%3D%3D</a>
<br>
2. Enter your name and email address. <br>
3. Enter the meeting password: acme <br>
4. Click &quot;Join Now&quot;. <br>
<br>
To view in other time zones or languages, please click the link: <br>
<a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&amp;UID=
=3D0&amp;PW=3DNN2I3YzUxYjJk&amp;ORT=3DMiMxMQ%3D%3D" target=3D"_blank">https=
://cisco.webex.com/ciscosales/j.php?ED=3D216374677&amp;UID=3D0&amp;PW=3DNN2=
I3YzUxYjJk&amp;ORT=3DMiMxMQ%3D%3D</a>
<br>
<br>
---------------------------------------------------------------- <br>
ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes <br>
---------------------------------------------------------------- <br>
<br>
The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpita=
s area and (866) 349-3520 for the RTP area.
<br>
<br>
Please dial the local access number for your area from the list below: <br>
- San Jose/Milpitas (408) area: 525-6800 <br>
- RTP (919) area: 392-3330 <br>
<br>
------------------------------------------------------- <br>
To join the teleconference only <br>
------------------------------------------------------- <br>
1. Dial into Cisco WebEx (view all Global Access Numbers at <br>
<a href=3D"http://cisco.com/en/US/about/doing_business/conferencing/index.h=
tml" target=3D"_blank">http://cisco.com/en/US/about/doing_business/conferen=
cing/index.html</a>
<br>
2. Follow the prompts to enter the Meeting Number (listed above) or Access =
Code followed by the # sign.
<br>
<br>
San Jose, CA: &#43;1.408.525.6800 RTP: &#43;1.919.392.3330 <br>
<br>
US/Canada: &#43;1.866.432.9903 United Kingdom: &#43;44.20.8824.0117 <br>
<br>
India: &#43;91.80.4350.1111 Germany: &#43;49.619.6773.9002 <br>
<br>
Japan: &#43;81.3.5763.9394 China: &#43;86.10.8515.5666 <br>
<br>
------------------------------------------------------- <br>
For assistance <br>
------------------------------------------------------- <br>
1. Go to <a href=3D"https://cisco.webex.com/ciscosales/mc" target=3D"_blank=
">https://cisco.webex.com/ciscosales/mc</a>
<br>
2. On the left navigation bar, click &quot;Support&quot;. <br>
<br>
You can contact me at: <br>
<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a> <br>
<br>
<br>
To add this meeting to your calendar program (for example Microsoft Outlook=
), click this link:
<br>
<a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&amp;UID=
=3D0&amp;ICS=3DMI&amp;LD=3D1&amp;RD=3D2&amp;ST=3D1&amp;SHA2=3DAAAAAkgeSvE1N=
zcP1FAQFi0Kzp6vvYe1NCey7X109lhs/gVs&amp;RT=3DMiMxMQ%3D%3D" target=3D"_blank=
">https://cisco.webex.com/ciscosales/j.php?ED=3D216374677&amp;UID=3D0&amp;I=
CS=3DMI&amp;LD=3D1&amp;RD=3D2&amp;ST=3D1&amp;SHA2=3DAAAAAkgeSvE1NzcP1FAQFi0=
Kzp6vvYe1NCey7X109lhs/gVs&amp;RT=3DMiMxMQ%3D%3D</a>
<br>
<br>
</font></blockquote>
</div>
</div>
</body>
</html>

--_000_C5E08FE080ACFD4DAE31E4BDBF944EB1133ABA13xmbalnx02ciscoc_--

From worley@shell01.TheWorld.com  Mon Feb  4 19:28:04 2013
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From: worley@alum.mit.edu (Dale R. Worley)
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Subject: [MMUSIC] A conceptual model of SDP bundling and related desiderata
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Here is the next edition of my thoughts about SDP bundling.  (See
y'all tomorrow!)

* Terminology

The important RFCs in this area use inconsistent terminology.  Here,
we use:

- "transport association" and "5-tuple"

A 5-tuple is the description of a particular transport association,
such as a TCP connection.  The components of the 5-tuple are the
identity of the protocol being used and the addresses and transport
ports of the two endpoints.

- "media"

We use this term for (1) media content, considered in an abstract way,
that is, without consideration of its particular encoding or the
framing information around it, and (2) the particular bits and bytes
used to encode and transmit the abstract media content.

- "multimedia session"

We use this term for the totality of the media that is
transmitted/received as described by a particular session description
(i.e., SDP instance).  This is taken from RFC 4566 section 2.

- "RTP session"

We use this term for the totality of the media that is
transmitted/received as described by a particular media description
(i.e., m= line) in a particular session description.  Since each media
description specifies one 5-tuple, RTP sessions correspond to
transport associations.  This is taken from RFC 3550 section 2.2.  (In
SIP usage (RFC 3264), this is called a "media stream", which term is
used in RTP usage to refer to the RTP with a single SSRC.)

It is understood that the RTP session can be dissected into "media
streams" that have separate SSRCs, but that is not relevant in this
analysis.  E.g., in SIP telephony, various activities of the far
endpoint can cause the ultimate source of the audio (and hence the
SSRC) to change dynamically without warning, but each new source
stands in for the previous one seamlessly in the user interface.

(However, there is a way (RFC 4566 section 5.14) for an m= line to
specify a set of ports and thus a set of related RTP sessions.  We do
not address that.)

In general, further agreed terminology is needed to describe the
aggregate of media possessing a particular SSRC, etc., but that is not
needed for this discussion.

* Conceptual structure

The central idea of bundling is to multiplex the media that would be
several RTP sessions into one RTP session, with particular emphasis on
allowing one transport association to carry media that are presented
to the higher, application layer, as multiple RTP sessions.

At the interface between the SDP-configured layer and the lower,
transport layer, we want the media organized into a single RTP
session.  The transport-related properties of the RTP session (e.g.,
transport 5-tuple, encryption, ICE) will be described by the
transport-related attributes of a single media description.

At the interface between the SDP-configured layer and the higher,
application layer, we want to have the media organized into several
RTP sessions.  The application-related properties of the RTP session
(e.g., media type and label) will be described by the
application-related attributes of separate media descriptions.

(There are some attributes (e.g., bandwidth limitation) that can apply
separately to both the bundled RTP session and the constituent RTP
sessions.)

However, we do not include the payload type numbers as information
available to the application; only the codec identity and its
parameters are accessible to the application.  This gives the bundling
mechanism freedom to place constraints on the use of payload types.

The application may distinguish between the media of each constituent
RTP session in ways that are not determinable from the properties of
the media themselves.  (E.g., in telepresence systems or in systems
where multiple video feeds are to be assembled into one large image,
multiple video streams have non-trivial relationships that are
specified by additional application-level SDP attributes.  Or there
may be several RTP sessions whose media types are the same but whose
logical roles in the multimedia session are not interchangeable.)

The application considers all media within a particular RTP session to
be a single logical media stream, regardless of changes in codec
(payload type) and SSRC (which often happen because the ultimate
source of the media changes).  E.g., a SIP telephone sends/receives
SDP with one audio media description, and all RTP sent/received within
that RTP session is connected to/from the telephone's user interface,
despite that far-end call transfer events or audio conference
processing may cause the payload type and SSRC of received RTP to
change without warning.

* Desiderata for a bundling mechanism

(I use the term "desiderata" -- "things that are desired" -- rather
than "requirements", because we may discover that we can't optimally
satisfy all of these criteria.)

    DES 1:  For each bundle, there is a group of media descriptions which
    describe the application-level RTP sessions.

    DES 2:  For each bundle, there is a media description that describes
    the transport-level RTP session.

These requirements do not specify whether the transport-level media
description may or may not also be one of the application-level media
descriptions.

    DES 3:  Bundles may contain other bundles as constituents.

Of course, no bundle may directly or indirectly contain itself.

I don't expect any current implementation to implement bundles within
bundles, but we should design the mechanism to allow this, as some day
we will likely need it.

    DES 4:  A bundle may contain zero constituents.

A bundle with no constituents serves no purpose for the transport of
media, but we are likely to someday need to describe such a bundle.
(Compare that an SDP m= line is syntactically constrained to specify
at least one payload type.  When SDP was used only to specify
multicast sessions, this constraint was common sense.  But once SDP
offer/answer was invented, when a media description was rejected, the
natural construction would be an m= line with a zero port and no
payload types.  But a payload type was syntactically required, so we
now have to provide at least one useless payload type for rejected m=
lines.)

    DES 5:  If an answerer that does not understand the bundle
    mechanism processes an offer that contains a bundle, it must
    interpret it as the collection of application-level media
    descriptions, and the offerer must be able to properly interpret
    the answer that it produces.

    DES 6:  If an answerer that does understand the bundle mechanism
    processes an offer that contains a bundle, it must be able to (1)
    accept the bundle and selectively accept/reject each constituent
    RTP session within it, (2) reject the bundle as a whole, or (3)
    reject the bundling and selectively accept/reject each constituent
    RTP session as separate RTP sessions.

Presumably answer (3) is like that which would be produced by an
answerer that does not understand the bundling mechanism.

    DES 7:  There must be a reliable way to demultiplex incoming RTP
    into the separate application-level RTP sessions.  Similarly,
    there must be a reliable way to demultiplex the associated RTCP
    information.

I know little about RTCP, but I believe that the RTCP information is
tagged with the SSRC about which it reports, and the SSRC is used to
correlate the RTCP reports with the RTP sessions containing media with
the same SSRC.  If so, RTCP demultiplexing requires no additional
specification, as RTCP is understood in the same way as in non-bundled
RTP.

    DES 8:  The specification must specify any needed additional
    procedures for handling SSRC collisions between media sources
    within different application-level RTP sessions, as those can now
    collide.

Dale

From ron.even.tlv@gmail.com  Mon Feb  4 23:25:49 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <mmusic@ietf.org>
Date: Tue, 5 Feb 2013 09:22:49 +0200
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Subject: [MMUSIC] New Version Notification for draft-even-mmusic-multiple-streams-01.txt
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Hi,
Resubmitted with small editorial changes and some clarifying text
Roni


________________________________________
From: internet-drafts@ietf.org [internet-drafts@ietf.org]
Sent: Tuesday, February 05, 2013 9:22 AM
To: Roni Even
Cc: jonathan@vidyo.com; bill.wu@huawei.com
Subject: New Version Notification for
draft-even-mmusic-multiple-streams-01.txt

A new version of I-D, draft-even-mmusic-multiple-streams-01.txt
has been successfully submitted by Roni Even and posted to the IETF
repository.

Filename:        draft-even-mmusic-multiple-streams
Revision:        01
Title:           Describing multiple RTP media streams in SDP
Creation date:   2013-02-05
WG ID:           Individual Submission
Number of pages: 10
URL:
http://www.ietf.org/internet-drafts/draft-even-mmusic-multiple-streams-01.tx
t
Status:
http://datatracker.ietf.org/doc/draft-even-mmusic-multiple-streams
Htmlized:
http://tools.ietf.org/html/draft-even-mmusic-multiple-streams-01
Diff:
http://www.ietf.org/rfcdiff?url2=draft-even-mmusic-multiple-streams-01

Abstract:
   This document describes issues when describing multiple RTP streams
   in a single RTP session using SDP.  The document looks at current
   solutions and provides paths toward addressing the issues.




The IETF Secretariat=


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The * is valid.  The part of the m= line that you know to contain
payload types is defined as being specific to the type.  Thus,
  m=rabbit 4995 BUNNIES *
is valid SDP, assuming of course that someone has defined a top-level
MIME type of 'rabbit' and so forth.

On 04/02/2013, Dale R. Worley <worley@ariadne.com> wrote:
> A couple of nits about draft-ietf-mmusic-sctp-sdp-00:
>
> section 7
>
> Examples contain "*" in m= lines instead of payload types.  The
> resader will probably assume that "*" stands for a sequence of payload
> types, but it's best to have examples be valid.  Replacing all the
> "*"s with "0 8" would suffice to make the examples valid:
>
>               m=image 54111 SCTP 0 8
>               c=IN IP4 192.0.2.2
>               a=setup:actpass
>               a=connection:new
>
> section 7.2
>
> The text suggests that the intention is to reuse the existing SCTP
> connection, but the SDP offer and answer both contain
> "a=connection:new".  Shouldn't these use "a=connection:existing"?
> Compare with RFC 4145 section 7.3.
>
> Dale
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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On 2/3/13 01:04, Justin Uberti wrote:
> 1) "1a" and "2", as described, have fatal problems if the receiver 
> rejects the m= line that contains all the attributes (e.g. 
> a=ice-ufrag, a=crypto, etc)

Yeah, this part of the solution was put together kind of off-the-cuff. 
I'd considered three approaches, and this one was the least verbose. My 
second choice, which has better backwards compatibility properties 
anyway, is mandating that the transport-level attributes are in all the 
m= sections to which they apply. You end up with redundant information, 
but it does get around issues with rejecting some sections.

> 2) In "1b", it's not clear what the purpose of the "a=group:MMT foo 
> bar zoe" line is.

Unfortunately, I really didn't have the time I wanted to put this draft 
together, and there's more than a little carelessness in the SDP 
examples. That shouldn't be there.

> 3) Proposal "2" does not include the needed a=ssrc lines for 
> demultiplexing the received streams. IOW, proposal 2 requires RFC 5576 
> just like 1a and 1b.

And this is the sloppiness I referred to above. The different sections 
should have non-overlapping RTP payload types.

> 4) The codec selection argument in Section 2.3.1 feels like a "Red 
> Herring" to me. Practically speaking, I think there are few, if any, 
> real-world settings where one might want to use a codec prioritization 
> mechanism to ask for a low-bitrate codec for stream A and a 
> high-bitrate codec for stream B (where the codecs used are different). 
> A more realistic example would be where one might want a lower bitrate 
>  for a specific multi-rate codec (e.g. Opus) to be used for stream A, 
> and higher bitrate for stream B. This of course can be handled with 
> the per-ssrc fmtp attributes that are specified in RFC 5576. However, 
> I think in almost all cases, the receiver is in a much worse position 
> to make these decisions that the sender; the sender, aware of the 
> estimated bandwidth to the receiver, and the relative value/complexity 
> of its streams, can make more informed decisions about bandwidth 
> allocation, and perform them without any signaling delay.

You're asserting that the person sending the media (to whom the quality 
has little or no impact) is a better party to select the quality of a 
stream than the person receiving it?

I have nothing to say to that. It's sufficiently false on its face that 
I really have nowhere to start grappling with it.

> 5) The directional attribute discussion in Section 4 omits one 
> important consequence of the fact that a=ssrc indications are 
> unidirectional, namely that streams could be added or removed by both 
> endpoints simultaneously without causing a glare condition. m= lines, 
> on the other hand, since they are identified by their index, do not 
> have this property; if both sides decide to add a m= line at the same 
> time, bad things happen.

None of the three approaches avoids the necessity to handle glare. With 
a stateful offer/answer approach, there's simply no way around it. We 
would have to go back to the drawing board if we wanted to come up with 
a system that avoided glare (and even then, I don't think success is 
very likely).

> 6) The legacy interop argument in Section 4 feels a bit hand-wavy. 
> Aside from the one case of having video + presentation streams, I have 
> not yet heard of widely deployed equipment that uses multiple audio or 
> video m= lines, and certainly any such equipment will not understand 
> BUNDLE or MMT. Therefore a gateway is required, and any gateway that 
> was performing BUNDLE or MMT could certainly do the minor additional 
> work to package the SDP as 1 m= line (remember, it already needs to 
> add the a=ssrc attributes). Ergo, I don't agree that the general case 
> of N m= lines needs to be handled by all RTCWEB implementations.

Truth be told, even though I called this "legacy interop," it's more 
accurate to think of it as "interop with non-RTCWEB implementations that 
use SDP," of which legacy is an important subset.

In any case, if we're jumping to needing a gateway to do substantial SDP 
rewrites, then I question the use of SDP at all in RTCWEB. The selection 
of SDP, to my understanding, was to provide simpler interop with 
non-RTCWEB implementations. Once we assume that you're necessarily 
rewriting it for such interop to work, we may as well abandon it altogether.

/a

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<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 2/3/13 01:04, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-22bJkEPOE=7kUQAV8UiEh3QT=DBOzccXPvOZDKbog1hw@mail.gmail.com"
      type="cite">
      <div dir="ltr">1) "1a" and "2", as described, have fatal problems
        if the receiver rejects the m= line that contains all the
        attributes (e.g. a=ice-ufrag, a=crypto, etc)</div>
    </blockquote>
    <br>
    Yeah, this part of the solution was put together kind of
    off-the-cuff. I'd considered three approaches, and this one was the
    least verbose. My second choice, which has better backwards
    compatibility properties anyway, is mandating that the
    transport-level attributes are in all the m= sections to which they
    apply. You end up with redundant information, but it does get around
    issues with rejecting some sections.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-22bJkEPOE=7kUQAV8UiEh3QT=DBOzccXPvOZDKbog1hw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div style="">2) In "1b", it's not clear what the purpose of the
          "<span style="color:rgb(0,0,0);white-space:pre-wrap">a=group:MMT
            foo bar zoe" line is. </span><br>
        </div>
      </div>
    </blockquote>
    <br>
    Unfortunately, I really didn't have the time I wanted to put this
    draft together, and there's more than a little carelessness in the
    SDP examples. That shouldn't be there.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-22bJkEPOE=7kUQAV8UiEh3QT=DBOzccXPvOZDKbog1hw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div style="">
          <span style="color:rgb(0,0,0);white-space:pre-wrap"></span>3)
          Proposal "2" does not include the needed a=ssrc lines for
          demultiplexing the received streams. IOW, proposal 2 requires
          RFC 5576 just like 1a and 1b.</div>
      </div>
    </blockquote>
    <br>
    And this is the sloppiness I referred to above. The different
    sections should have non-overlapping RTP payload types.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-22bJkEPOE=7kUQAV8UiEh3QT=DBOzccXPvOZDKbog1hw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div style="">4) The codec selection argument in Section 2.3.1
          feels like a "Red Herring" to me. Practically speaking, I
          think there are few, if any, real-world settings where one
          might want to use a codec prioritization mechanism to ask for
          a low-bitrate codec for stream A and a high-bitrate codec for
          stream B (where the codecs used are different). A more
          realistic example would be where one might want a lower
          bitrate Â for a specific multi-rate codec (e.g. Opus) to be
          used for stream A, and higher bitrate for stream B. This of
          course can be handled with the per-ssrc fmtp attributes that
          are specified in RFC 5576. However, I think in almost all
          cases, the receiver is in a much worse position to make these
          decisions that the sender; the sender, aware of the estimated
          bandwidth to the receiver, and the relative value/complexity
          of its streams, can make more informed decisions about
          bandwidth allocation, and perform them without any signaling
          delay.</div>
      </div>
    </blockquote>
    <br>
    You're asserting that the person sending the media (to whom the
    quality has little or no impact) is a better party to select the
    quality of a stream than the person receiving it?<br>
    <br>
    I have nothing to say to that. It's sufficiently false on its face
    that I really have nowhere to start grappling with it.<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-22bJkEPOE=7kUQAV8UiEh3QT=DBOzccXPvOZDKbog1hw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div style="">5) The directional attribute discussion in Section
          4 omits one important consequence of the fact that a=ssrc
          indications are unidirectional, namely that streams could be
          added or removed by both endpoints simultaneously without
          causing a glare condition. m= lines, on the other hand, since
          they are identified by their index, do not have this property;
          if both sides decide to add a m= line at the same time, bad
          things happen.</div>
      </div>
    </blockquote>
    <br>
    None of the three approaches avoids the necessity to handle glare.
    With a stateful offer/answer approach, there's simply no way around
    it. We would have to go back to the drawing board if we wanted to
    come up with a system that avoided glare (and even then, I don't
    think success is very likely).<br>
    <br>
    <blockquote
cite="mid:CAOJ7v-22bJkEPOE=7kUQAV8UiEh3QT=DBOzccXPvOZDKbog1hw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div style="">6) The legacy interop argument in Section 4 feels
          a bit hand-wavy. Aside from the one case of having video +
          presentation streams, I have not yet heard of widely deployed
          equipment that uses multiple audio or video m= lines, and
          certainly any such equipment will not understand BUNDLE or
          MMT. Therefore a gateway is required, and any gateway that was
          performing BUNDLE or MMT could certainly do the minor
          additional work to package the SDP as 1 m= line (remember, it
          already needs to add the a=ssrc attributes). Ergo, I don't
          agree that the general case of N m= lines needs to be handled
          by all RTCWEB implementations.</div>
      </div>
    </blockquote>
    <br>
    Truth be told, even though I called this "legacy interop," it's more
    accurate to think of it as "interop with non-RTCWEB implementations
    that use SDP," of which legacy is an important subset.<br>
    <br>
    In any case, if we're jumping to needing a gateway to do substantial
    SDP rewrites, then I question the use of SDP at all in RTCWEB. The
    selection of SDP, to my understanding, was to provide simpler
    interop with non-RTCWEB implementations. Once we assume that you're
    necessarily rewriting it for such interop to work, we may as well
    abandon it altogether.<br>
    <br>
    /a<br>
  </body>
</html>

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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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On 2/4/13 02:45, Justin Uberti wrote:
> Sorry for the short lead time, but here's a quick read for the plane.

I addressed several of the assertions made in this draft in my previous 
response (e.g. the need for glare resolution regardless of which 
solution we select). Some additional thoughts:

>     o  Adding or removing a source requires sending a full list of all
>        sources, which doesn't scale well to 10s or 100s of sources,
>        especially when they are added or removed frequently.

Realistically, this isn't how multiparty conferences are handled. It's 
unrealistic to think that my DSL connection is going to be sucking down 
hundreds of video streams just because I'm in a video conference with 
hundreds of people.

In practice, the way existing conferencing systems handle this is to 
send a small, manageable number of streams to each participant, changing 
them out as necessary to have a reasonable user experience (either under 
the control of the user, its own algorithms, or a combination of the two).

Which is to say: even with hundreds of source streams being fed into a 
conference system, you're really looking at a realistic upper bound on 
m= sections of a dozen or two, simply because you can't reasonably 
expect recipients to receive more than that many streams simultaneously.

And really, if you're imagining hundreds of media streams flowing 
towards an endpoint, any arguments about the bits consumed by SDP go 
well beyond disingenuous.

>     o  Because this solution requires BUNDLE, which is still not widely
>        supported, it can be deployed in fewer situations

This is completely out of left field: the alternative is MMT, which is 
even less well developed than BUNDLE.

>     4.  The RFC5576-based approach has already been in use for many years
>         in a major video conferencing system (Google+ Hangouts), and has
>         worked very well.  We feel that we have searched the "unknown"
>         space and solved all of the major issues.

I think this actually does a good job of illustrating your blind spot: 
you're working within RTCWEB, but developing for a single, very precise 
use case. For example, this kind of experience doesn't translate well 
into the challenges that might arise from systems that are dealing with 
radically different user experiences. Google-hangout-type uses are 
certainly important, but they're not the only thing we're designing for. 
You need to think about systems with pre-encoded (stored) media; systems 
with radical imbalances in the processing power and device capabilities 
between the endpoints; systems with different stream control models, 
etc, etc.

>         On the other hand, the multi m=
>         section approach has never been tried by anyone, that we are
>         aware of

Do a web search for "Marconi ViPr."

/a

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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
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    <div class="moz-cite-prefix">On 2/4/13 02:45, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-3Ndb7vz9gchQ0Y308uH4nFUrKv8AkmeNrdf-3F03MLQg@mail.gmail.com"
      type="cite">
      <div dir="ltr">Sorry for the short lead time, but here's a quick
        read for the plane.</div>
    </blockquote>
    <br>
    I addressed several of the assertions made in this draft in my
    previous response (e.g. the need for glare resolution regardless of
    which solution we select). Some additional thoughts:<br>
    <br>
    <blockquote type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre>   o  Adding or removing a source requires sending a full list of all
      sources, which doesn't scale well to 10s or 100s of sources,
      especially when they are added or removed frequently.</pre>
    </blockquote>
    <br>
    Realistically, this isn't how multiparty conferences are handled.
    It's unrealistic to think that my DSL connection is going to be
    sucking down hundreds of video streams just because I'm in a video
    conference with hundreds of people.<br>
    <br>
    In practice, the way existing conferencing systems handle this is to
    send a small, manageable number of streams to each participant,
    changing them out as necessary to have a reasonable user experience
    (either under the control of the user, its own algorithms, or a
    combination of the two).<br>
    <br>
    Which is to say: even with hundreds of source streams being fed into
    a conference system, you're really looking at a realistic upper
    bound on m= sections of a dozen or two, simply because you can't
    reasonably expect recipients to receive more than that many streams
    simultaneously.<br>
    <br>
    And really, if you're imagining hundreds of media streams flowing
    towards an endpoint, any arguments about the bits consumed by SDP go
    well beyond disingenuous.<br>
    <br>
    <blockquote type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre>   o  Because this solution requires BUNDLE, which is still not widely
      supported, it can be deployed in fewer situations</pre>
    </blockquote>
    <br>
    This is completely out of left field: the alternative is MMT, which
    is even less well developed than BUNDLE.<br>
    <br>
    <blockquote type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre>   4.  The RFC5576-based approach has already been in use for many years
       in a major video conferencing system (Google+ Hangouts), and has
       worked very well.  We feel that we have searched the "unknown"
       space and solved all of the major issues.</pre>
    </blockquote>
    <br>
    I think this actually does a good job of illustrating your blind
    spot: you're working within RTCWEB, but developing for a single,
    very precise use case. For example, this kind of experience doesn't
    translate well into the challenges that might arise from systems
    that are dealing with radically different user experiences.
    Google-hangout-type uses are certainly important, but they're not
    the only thing we're designing for. You need to think about systems
    with pre-encoded (stored) media; systems with radical imbalances in
    the processing power and device capabilities between the endpoints;
    systems with different stream control models, etc, etc.<br>
    <br>
    <blockquote type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre>       On the other hand, the multi m=
       section approach has never been tried by anyone, that we are
       aware of
</pre>
    </blockquote>
    <br>
    Do a web search for "Marconi ViPr."<br>
    <br>
    /a<br>
  </body>
</html>

--------------030801030007050804020702--

From adam@nostrum.com  Tue Feb  5 07:52:35 2013
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Subject: Re: [MMUSIC] Multicast and one-m-line-per-stream (Re: Thoughts on m= lines and media streams)
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Before I respond, I do want to make sure that this is the same Harald 
Alvestrand who was proposing that we dismiss the vast majority of 
existing SDP attributes on the premise that they are of no interest to 
WEBRTC.

/a

On 2/2/13 18:13, Harald Alvestrand wrote:
> Adam,
>
> I hate to play the multicast card, because I'm fundamentally 
> uninterested in multicast at the moment, but....
>
> in your scheme 2, how would you represent multicast and 
> transport-mixer based sessions?
>
>
> On 02/01/2013 01:17 AM, Adam Roach wrote:
>> Based on the conversations we've had so far, I've put together the 
>> following summary of my thoughts on the topic of whether we want to 
>> put multiple media streams in a single m= line.
>>
>> http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt
>>
>> Sorry for the relatively short lead time before next week's meeting 
>> -- I had meant to put this kind of summary together a week or two 
>> ago, but have been far busier than I anticipated.
>>
>> /a
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, Adam Roach <adam@nostrum.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [MMUSIC] Thoughts on m= lines and media streams
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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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Hi,

>>       1) "1a" and "2", as described, have fatal problems if the receiver
>> rejects the m=3D line that contains all the attributes (e.g. a=3Dice-ufr=
ag,
>> a=3Dcrypto, etc)
>>
>>
>> Yeah, this part of the solution was put together kind of off-the-cuff. I=
'd
>> considered three approaches, and this one was the least verbose. My
>> second choice, which has better backwards compatibility properties
>> anyway, is mandating that the transport-level attributes are in all the =
m=3D
>> sections to which they apply. You end up with redundant information, but=
 it
>> does get around issues with rejecting some sections.
>
> Why not specify such  transport attributes at the session level. For thos=
e attributes that are currently media level only, extend them to define how=
 they work at the session level.
>
> For example, b=3D at the session level is already defined to be the maxim=
um bandwidth across the entire session whereas b=3D at the m=3Dsection leve=
l is for that m=3Dsection only.
> For media direction/mode, a=3Dsendonly/sendrecv/inactive/recvonly can be =
specified at the session level, in which case it applies to all m=3Dsection=
s unless they specify their own value.
> Similarly, ICE candidates could be specified at the session level and hav=
e those apply at all media levels, unless the media level specifies its own=
 value.

That has been considered, but how would it work if the remote endpoint does=
n't support the mechanism?

Regards,

Christer




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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Adam Roach'" <adam@nostrum.com>, <mmusic@ietf.org>
References: <510B09A0.5040904@nostrum.com>
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Date: Tue, 5 Feb 2013 18:14:27 +0200
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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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Hi,
One requirement that may be important for your analysis is if there must be
one offer answer exchange or more for setting the call. 
Some of the current solutions are doing a second one in order to reflect the
current state and prevent codec switching or new content without explicit
signaling. This is because they are not capable of doing it. This can
address the SBC issue that expect to have the correct SDP.
BTW: this is also an issue in CLUE since the initial offer may have a
minimal set and the actual SDP can only be specified after the CLUE
negotitation.

Roni Even

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Adam Roach
Sent: 01 February, 2013 2:18 AM
To: mmusic@ietf.org
Subject: [MMUSIC] Thoughts on m= lines and media streams

Based on the conversations we've had so far, I've put together the following
summary of my thoughts on the topic of whether we want to put multiple media
streams in a single m= line.

http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt

Sorry for the relatively short lead time before next week's meeting -- I had
meant to put this kind of summary together a week or two ago, but have been
far busier than I anticipated.

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


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Adam Roach <adam@nostrum.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [MMUSIC] More thoughts on multiple media streams in a single m= line
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I think we are all in agreement that for a large conference (e.g. 100+ audi=
o/video streams), we cannot require that each media stream have its own m=
=3Dsection, nor can we require that each media stream's SSRC is signaled in=
 SDP. For the sake of example, let's assume a centralized conference server=
 is able to send up to 10 video streams, where each video stream may be any=
 of the 100 video streams at various points in time (e.g. active speakers s=
witched in/out, specific sources requested, etc.), I think there are three =
realistic alternatives:

1) Use 10 m=3Dsections to define the ten video streams, no multiplexing req=
uired
2) Use 1 m=3Dsection to define  a video stream, and indicate that up to 10 =
video streams are multiplexed within it
3) Use 'n' m=3Dsections to define the video streams, where 1<=3Dn<=3D10

We already have all we need to do (1). There is a desire to improve on (1).
If we define some additional syntax/semantics that enable (2), people can c=
hoose to use (2) or (3) depending on works best for their scenario. There h=
ave already been some drafts submitted along these lines, I suggest to take=
 a closer look at those and improve on them. Sounds reasonable?

Charles

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Adam Roach
> Sent: Tuesday, February 05, 2013 10:26 AM
> To: Justin Uberti
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single
> m=3D line
>=20
> On 2/4/13 02:45, Justin Uberti wrote:
>=20
>=20
> 	Sorry for the short lead time, but here's a quick read for the plane.
>=20
>=20
> I addressed several of the assertions made in this draft in my previous
> response (e.g. the need for glare resolution regardless of which solution=
 we
> select). Some additional thoughts:
>=20
>=20
>=20
> 	   o  Adding or removing a source requires sending a full list of all
> 	      sources, which doesn't scale well to 10s or 100s of sources,
> 	      especially when they are added or removed frequently.
>=20
>=20
> Realistically, this isn't how multiparty conferences are handled. It's
> unrealistic to think that my DSL connection is going to be sucking down
> hundreds of video streams just because I'm in a video conference with
> hundreds of people.
>=20
> In practice, the way existing conferencing systems handle this is to send=
 a
> small, manageable number of streams to each participant, changing them
> out as necessary to have a reasonable user experience (either under the
> control of the user, its own algorithms, or a combination of the two).
>=20
> Which is to say: even with hundreds of source streams being fed into a
> conference system, you're really looking at a realistic upper bound on m=
=3D
> sections of a dozen or two, simply because you can't reasonably expect
> recipients to receive more than that many streams simultaneously.
>=20
> And really, if you're imagining hundreds of media streams flowing towards
> an endpoint, any arguments about the bits consumed by SDP go well
> beyond disingenuous.
>=20
>=20
>=20
> 	   o  Because this solution requires BUNDLE, which is still not widely
> 	      supported, it can be deployed in fewer situations
>=20
>=20
> This is completely out of left field: the alternative is MMT, which is ev=
en
> less well developed than BUNDLE.
>=20
>=20
>=20
> 	   4.  The RFC5576-based approach has already been in use for many
> years
> 	       in a major video conferencing system (Google+ Hangouts), and
> has
> 	       worked very well.  We feel that we have searched the
> "unknown"
> 	       space and solved all of the major issues.
>=20
>=20
> I think this actually does a good job of illustrating your blind spot: yo=
u're
> working within RTCWEB, but developing for a single, very precise use case=
.
> For example, this kind of experience doesn't translate well into the
> challenges that might arise from systems that are dealing with radically
> different user experiences. Google-hangout-type uses are certainly
> important, but they're not the only thing we're designing for. You need t=
o
> think about systems with pre-encoded (stored) media; systems with radical
> imbalances in the processing power and device capabilities between the
> endpoints; systems with different stream control models, etc, etc.
>=20
>=20
>=20
> 	       On the other hand, the multi m=3D
> 	       section approach has never been tried by anyone, that we are
> 	       aware of
>=20
>=20
> Do a web search for "Marconi ViPr."
>=20
> /a


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On Jan 28, 2013, at 12:12 PM, Flemming Andreasen <fandreas@cisco.com> wrote=
:

> Thanks for taking a look and providing your comments Hadriel. As per prev=
ious discussions, we wanted to first agree on the scope of the document and=
 whether there was still value in pursuing such a document. As per Gonzalo'=
s e-mail starting this thread, do you have any comments/concerns on that pa=
rt (if so, please let the authors know what those are) ?


Sorry didn't respond to the above yet=85 I assumed that the WG already deci=
ded to continue moving towards publishing it.

Back in March last year I asked what the point/purpose of the draft was, he=
re:
http://www.ietf.org/mail-archive/web/mmusic/current/msg09278.html

As chair you called for a WG Poll, here:
http://www.ietf.org/mail-archive/web/mmusic/current/msg09368.html

And provided your rationale for publishing it here:
http://www.ietf.org/mail-archive/web/mmusic/current/msg09369.html

The response to that was positive (5 people agreeing, none against).

So I was just moving on to discussing what's in the draft, vs. why bother p=
ublishing it.

-hadriel


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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: draft-ietf-mmusic-media-path-middleboxes@tools.ietf.org, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] I-D	Action: draft-ietf-mmusic-media-path-middleboxes-06.txt
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On 2/5/13 11:41 AM, Hadriel Kaplan wrote:
> On Jan 28, 2013, at 12:12 PM, Flemming Andreasen <fandreas@cisco.com> wrote:
>
>> Thanks for taking a look and providing your comments Hadriel. As per previous discussions, we wanted to first agree on the scope of the document and whether there was still value in pursuing such a document. As per Gonzalo's e-mail starting this thread, do you have any comments/concerns on that part (if so, please let the authors know what those are) ?
>
> Sorry didn't respond to the above yet… I assumed that the WG already decided to continue moving towards publishing it.
>
> Back in March last year I asked what the point/purpose of the draft was, here:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg09278.html
>
> As chair you called for a WG Poll, here:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg09368.html
>
> And provided your rationale for publishing it here:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg09369.html
>
> The response to that was positive (5 people agreeing, none against).
>
> So I was just moving on to discussing what's in the draft, vs. why bother publishing it.
Thanks Hadriel

Based on that, let's continue with the technical issues you've raised.

Authors: Please chime in and work on resolving the issues being raised.

Thanks

-- Flemming


> -hadriel
>
> .
>


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@nostrum.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [MMUSIC] Thoughts on m= lines and media streams
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Date: Tue, 5 Feb 2013 17:05:25 +0000
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Peter Thatcher <pthatcher@google.com>
Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Tuesday, February 05, 2013 11:09 AM
> To: Charles Eckel (eckelcu); Adam Roach; Justin Uberti
> Cc: mmusic@ietf.org; Peter Thatcher
> Subject: Re: [MMUSIC] Thoughts on m=3D lines and media streams
>=20
>=20
> Hi,
>=20
> >> 1) "1a" and "2", as described, have fatal problems if the receiver
> >> rejects the m=3D line that contains all the attributes (e.g. a=3Dice-u=
frag,
> >> a=3Dcrypto, etc)
> >>
> >>
> >> Yeah, this part of the solution was put together kind of off-the-cuff.=
 I'd
> >> considered three approaches, and this one was the least verbose. My
> >> second choice, which has better backwards compatibility properties
> >> anyway, is mandating that the transport-level attributes are in all th=
e m=3D
> >> sections to which they apply. You end up with redundant information,
> but it
> >> does get around issues with rejecting some sections.
> >
> > Why not specify such transport attributes at the session level. For tho=
se
> attributes that are currently media level only, extend them to define how
> they work at the session level.
> >
> > For example, b=3D at the session level is already defined to be the max=
imum
> bandwidth across the entire session whereas b=3D at the m=3Dsection level=
 is for
> that m=3Dsection only.
> > For media direction/mode, a=3Dsendonly/sendrecv/inactive/recvonly can b=
e
> specified at the session level, in which case it applies to all m=3Dsecti=
ons
> unless they specify their own value.
> > Similarly, ICE candidates could be specified at the session level and h=
ave
> those apply at all media levels, unless the media level specifies its own
> value.
>=20
> That has been considered, but how would it work if the remote endpoint
> doesn't support the mechanism?

Crap. That said, I think you have that same concern with any approach. Usin=
g ICE candidates as an example, how do you know if the remote endpoint supp=
orts the same candidate existing in multiple m=3Dsections and will use them=
 as intended.

Cheers,
Charles =20
=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From juberti@google.com  Tue Feb  5 09:18:14 2013
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From: Justin Uberti <juberti@google.com>
Date: Tue, 5 Feb 2013 09:17:49 -0800
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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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On Tue, Feb 5, 2013 at 6:32 AM, Adam Roach <adam@nostrum.com> wrote:

>  On 2/3/13 01:04, Justin Uberti wrote:
>
> 1) "1a" and "2", as described, have fatal problems if the receiver rejects
> the m= line that contains all the attributes (e.g. a=ice-ufrag, a=crypto,
> etc)
>
>
> Yeah, this part of the solution was put together kind of off-the-cuff. I'd
> considered three approaches, and this one was the least verbose. My second
> choice, which has better backwards compatibility properties anyway, is
> mandating that the transport-level attributes are in all the m= sections to
> which they apply. You end up with redundant information, but it does get
> around issues with rejecting some sections.
>
>
>  2) In "1b", it's not clear what the purpose of the "a=group:MMT foo bar
> zoe" line is.
>
>
> Unfortunately, I really didn't have the time I wanted to put this draft
> together, and there's more than a little carelessness in the SDP examples.
> That shouldn't be there.
>
>
>   3) Proposal "2" does not include the needed a=ssrc lines for
> demultiplexing the received streams. IOW, proposal 2 requires RFC 5576 just
> like 1a and 1b.
>
>
> And this is the sloppiness I referred to above. The different sections
> should have non-overlapping RTP payload types.
>

Alas, RTP payload types are not sufficient for demux, since RTCP packets
don't include RTP PTs.

>
>
>  4) The codec selection argument in Section 2.3.1 feels like a "Red
> Herring" to me. Practically speaking, I think there are few, if any,
> real-world settings where one might want to use a codec prioritization
> mechanism to ask for a low-bitrate codec for stream A and a high-bitrate
> codec for stream B (where the codecs used are different). A more realistic
> example would be where one might want a lower bitrate  for a specific
> multi-rate codec (e.g. Opus) to be used for stream A, and higher bitrate
> for stream B. This of course can be handled with the per-ssrc fmtp
> attributes that are specified in RFC 5576. However, I think in almost all
> cases, the receiver is in a much worse position to make these decisions
> that the sender; the sender, aware of the estimated bandwidth to the
> receiver, and the relative value/complexity of its streams, can make more
> informed decisions about bandwidth allocation, and perform them without any
> signaling delay.
>
>
> You're asserting that the person sending the media (to whom the quality
> has little or no impact) is a better party to select the quality of a
> stream than the person receiving it?
>
> I have nothing to say to that. It's sufficiently false on its face that I
> really have nowhere to start grappling with it.
>

I should have been more clear. Yes, the receiver is the one who needs to
choose the relative priorities of streams, but the sender is in a better
place to make actual decisions about how those preferences should be
carried out (for example, what bitrate should be sent, or which layers of a
layered codec should be used).

Even if the receiver retains full control, the receiver could still ask for
different bitrates given using the mechanisms I described above.


>
>
>  5) The directional attribute discussion in Section 4 omits one important
> consequence of the fact that a=ssrc indications are unidirectional, namely
> that streams could be added or removed by both endpoints simultaneously
> without causing a glare condition. m= lines, on the other hand, since they
> are identified by their index, do not have this property; if both sides
> decide to add a m= line at the same time, bad things happen.
>
>
> None of the three approaches avoids the necessity to handle glare. With a
> stateful offer/answer approach, there's simply no way around it. We would
> have to go back to the drawing board if we wanted to come up with a system
> that avoided glare (and even then, I don't think success is very likely).
>

Since WebRTC leaves the signaling up to the application, if your signaling
system allows for partial updates, you can have "stream-add" and
"stream-remove" operations that are glareless.

>
>
>  6) The legacy interop argument in Section 4 feels a bit hand-wavy. Aside
> from the one case of having video + presentation streams, I have not yet
> heard of widely deployed equipment that uses multiple audio or video m=
> lines, and certainly any such equipment will not understand BUNDLE or MMT.
> Therefore a gateway is required, and any gateway that was performing BUNDLE
> or MMT could certainly do the minor additional work to package the SDP as 1
> m= line (remember, it already needs to add the a=ssrc attributes). Ergo, I
> don't agree that the general case of N m= lines needs to be handled by all
> RTCWEB implementations.
>
>
> Truth be told, even though I called this "legacy interop," it's more
> accurate to think of it as "interop with non-RTCWEB implementations that
> use SDP," of which legacy is an important subset.
>
> In any case, if we're jumping to needing a gateway to do substantial SDP
> rewrites, then I question the use of SDP at all in RTCWEB. The selection of
> SDP, to my understanding, was to provide simpler interop with non-RTCWEB
> implementations. Once we assume that you're necessarily rewriting it for
> such interop to work, we may as well abandon it altogether.
>
> /a
>

--20cf306f720cc786c604d4fd64cd
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Feb 5, 2013 at 6:32 AM, Adam Roach <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt=
;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im">
    <div>On 2/3/13 01:04, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">1) &quot;1a&quot; and &quot;2&quot;, as described, h=
ave fatal problems
        if the receiver rejects the m=3D line that contains all the
        attributes (e.g. a=3Dice-ufrag, a=3Dcrypto, etc)</div>
    </blockquote>
    <br></div>
    Yeah, this part of the solution was put together kind of
    off-the-cuff. I&#39;d considered three approaches, and this one was the
    least verbose. My second choice, which has better backwards
    compatibility properties anyway, is mandating that the
    transport-level attributes are in all the m=3D sections to which they
    apply. You end up with redundant information, but it does get around
    issues with rejecting some sections.<div class=3D"im"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>2) In &quot;1b&quot;, it&#39;s not clear what the purpose of t=
he
          &quot;<span style=3D"white-space:pre-wrap">a=3Dgroup:MMT
            foo bar zoe&quot; line is. </span><br>
        </div>
      </div>
    </blockquote>
    <br></div>
    Unfortunately, I really didn&#39;t have the time I wanted to put this
    draft together, and there&#39;s more than a little carelessness in the
    SDP examples. That shouldn&#39;t be there.<div class=3D"im"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>
          <span style=3D"white-space:pre-wrap"></span>3)
          Proposal &quot;2&quot; does not include the needed a=3Dssrc lines=
 for
          demultiplexing the received streams. IOW, proposal 2 requires
          RFC 5576 just like 1a and 1b.</div>
      </div>
    </blockquote>
    <br></div>
    And this is the sloppiness I referred to above. The different
    sections should have non-overlapping RTP payload types.</div></blockquo=
te><div><br></div><div style>Alas, RTP payload types are not sufficient for=
 demux, since RTCP packets don&#39;t include RTP PTs.=C2=A0</div><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">

<div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>4) The codec selection argument in Section 2.3.1
          feels like a &quot;Red Herring&quot; to me. Practically speaking,=
 I
          think there are few, if any, real-world settings where one
          might want to use a codec prioritization mechanism to ask for
          a low-bitrate codec for stream A and a high-bitrate codec for
          stream B (where the codecs used are different). A more
          realistic example would be where one might want a lower
          bitrate =C2=A0for a specific multi-rate codec (e.g. Opus) to be
          used for stream A, and higher bitrate for stream B. This of
          course can be handled with the per-ssrc fmtp attributes that
          are specified in RFC 5576. However, I think in almost all
          cases, the receiver is in a much worse position to make these
          decisions that the sender; the sender, aware of the estimated
          bandwidth to the receiver, and the relative value/complexity
          of its streams, can make more informed decisions about
          bandwidth allocation, and perform them without any signaling
          delay.</div>
      </div>
    </blockquote>
    <br></div>
    You&#39;re asserting that the person sending the media (to whom the
    quality has little or no impact) is a better party to select the
    quality of a stream than the person receiving it?<br>
    <br>
    I have nothing to say to that. It&#39;s sufficiently false on its face
    that I really have nowhere to start grappling with it.</div></blockquot=
e><div><br></div><div style>I should have been more clear. Yes, the receive=
r is the one who needs to choose the relative priorities of streams, but th=
e sender is in a better place to make actual decisions about how those pref=
erences should be carried out (for example, what bitrate should be sent, or=
 which layers of a layered codec should be used).</div>

<div style><br></div><div style>Even if the receiver retains full control, =
the receiver could still ask for different bitrates given using the mechani=
sms I described above.</div><div style>=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">

<div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>5) The directional attribute discussion in Section
          4 omits one important consequence of the fact that a=3Dssrc
          indications are unidirectional, namely that streams could be
          added or removed by both endpoints simultaneously without
          causing a glare condition. m=3D lines, on the other hand, since
          they are identified by their index, do not have this property;
          if both sides decide to add a m=3D line at the same time, bad
          things happen.</div>
      </div>
    </blockquote>
    <br></div>
    None of the three approaches avoids the necessity to handle glare.
    With a stateful offer/answer approach, there&#39;s simply no way around
    it. We would have to go back to the drawing board if we wanted to
    come up with a system that avoided glare (and even then, I don&#39;t
    think success is very likely).</div></blockquote><div><br></div><div st=
yle>Since WebRTC leaves the signaling up to the application, if your signal=
ing system allows for partial updates, you can have &quot;stream-add&quot; =
and &quot;stream-remove&quot; operations that are glareless.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000"><d=
iv class=3D"im"><br>
    <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>6) The legacy interop argument in Section 4 feels
          a bit hand-wavy. Aside from the one case of having video +
          presentation streams, I have not yet heard of widely deployed
          equipment that uses multiple audio or video m=3D lines, and
          certainly any such equipment will not understand BUNDLE or
          MMT. Therefore a gateway is required, and any gateway that was
          performing BUNDLE or MMT could certainly do the minor
          additional work to package the SDP as 1 m=3D line (remember, it
          already needs to add the a=3Dssrc attributes). Ergo, I don&#39;t
          agree that the general case of N m=3D lines needs to be handled
          by all RTCWEB implementations.</div>
      </div>
    </blockquote>
    <br></div>
    Truth be told, even though I called this &quot;legacy interop,&quot; it=
&#39;s more
    accurate to think of it as &quot;interop with non-RTCWEB implementation=
s
    that use SDP,&quot; of which legacy is an important subset.<br>
    <br>
    In any case, if we&#39;re jumping to needing a gateway to do substantia=
l
    SDP rewrites, then I question the use of SDP at all in RTCWEB. The
    selection of SDP, to my understanding, was to provide simpler
    interop with non-RTCWEB implementations. Once we assume that you&#39;re
    necessarily rewriting it for such interop to work, we may as well
    abandon it altogether.<span class=3D"HOEnZb"><font color=3D"#888888"><b=
r>
    <br>
    /a<br>
  </font></span></div>

</blockquote></div><br></div></div>

--20cf306f720cc786c604d4fd64cd--

From adam@nostrum.com  Tue Feb  5 09:30:36 2013
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Subject: Re: [MMUSIC] Thoughts on m= lines and media streams
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On 2/5/13 12:17, Justin Uberti wrote:
> Even if the receiver retains full control, the receiver could still 
> ask for different bitrates given using the mechanisms I described above.

Ah, yes, I forgot to speak to those directly. You're making assumptions 
here that all the codecs we want to use will be one-size-fits-all (with 
all adjustments to all uses possible by parameters on those codecs) from 
now until this technology is obsolete. I think that's short-sighted.

/a

From ron.even.tlv@gmail.com  Tue Feb  5 10:00:41 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Charles Eckel \(eckelcu\)'" <eckelcu@cisco.com>, "'Adam Roach'" <adam@nostrum.com>, "'Justin Uberti'" <juberti@google.com>
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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Charles,
Even for 1 you need maxssrc, you assume that one m-line means one RTP
stream. 
Roni

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Charles Eckel (eckelcu)
Sent: 05 February, 2013 6:20 PM
To: Adam Roach; Justin Uberti
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m=
line

I think we are all in agreement that for a large conference (e.g. 100+
audio/video streams), we cannot require that each media stream have its own
m=section, nor can we require that each media stream's SSRC is signaled in
SDP. For the sake of example, let's assume a centralized conference server
is able to send up to 10 video streams, where each video stream may be any
of the 100 video streams at various points in time (e.g. active speakers
switched in/out, specific sources requested, etc.), I think there are three
realistic alternatives:

1) Use 10 m=sections to define the ten video streams, no multiplexing
required
2) Use 1 m=section to define  a video stream, and indicate that up to 10
video streams are multiplexed within it
3) Use 'n' m=sections to define the video streams, where 1<=n<=10

We already have all we need to do (1). There is a desire to improve on (1).
If we define some additional syntax/semantics that enable (2), people can
choose to use (2) or (3) depending on works best for their scenario. There
have already been some drafts submitted along these lines, I suggest to take
a closer look at those and improve on them. Sounds reasonable?

Charles

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On 
> Behalf Of Adam Roach
> Sent: Tuesday, February 05, 2013 10:26 AM
> To: Justin Uberti
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] More thoughts on multiple media streams in a 
> single m= line
> 
> On 2/4/13 02:45, Justin Uberti wrote:
> 
> 
> 	Sorry for the short lead time, but here's a quick read for the
plane.
> 
> 
> I addressed several of the assertions made in this draft in my 
> previous response (e.g. the need for glare resolution regardless of 
> which solution we select). Some additional thoughts:
> 
> 
> 
> 	   o  Adding or removing a source requires sending a full list of
all
> 	      sources, which doesn't scale well to 10s or 100s of sources,
> 	      especially when they are added or removed frequently.
> 
> 
> Realistically, this isn't how multiparty conferences are handled. It's 
> unrealistic to think that my DSL connection is going to be sucking 
> down hundreds of video streams just because I'm in a video conference 
> with hundreds of people.
> 
> In practice, the way existing conferencing systems handle this is to 
> send a small, manageable number of streams to each participant, 
> changing them out as necessary to have a reasonable user experience 
> (either under the control of the user, its own algorithms, or a
combination of the two).
> 
> Which is to say: even with hundreds of source streams being fed into a 
> conference system, you're really looking at a realistic upper bound on 
> m= sections of a dozen or two, simply because you can't reasonably 
> expect recipients to receive more than that many streams simultaneously.
> 
> And really, if you're imagining hundreds of media streams flowing 
> towards an endpoint, any arguments about the bits consumed by SDP go 
> well beyond disingenuous.
> 
> 
> 
> 	   o  Because this solution requires BUNDLE, which is still not
widely
> 	      supported, it can be deployed in fewer situations
> 
> 
> This is completely out of left field: the alternative is MMT, which is 
> even less well developed than BUNDLE.
> 
> 
> 
> 	   4.  The RFC5576-based approach has already been in use for many 
> years
> 	       in a major video conferencing system (Google+ Hangouts), and 
> has
> 	       worked very well.  We feel that we have searched the
"unknown"
> 	       space and solved all of the major issues.
> 
> 
> I think this actually does a good job of illustrating your blind spot: 
> you're working within RTCWEB, but developing for a single, very precise
use case.
> For example, this kind of experience doesn't translate well into the 
> challenges that might arise from systems that are dealing with 
> radically different user experiences. Google-hangout-type uses are 
> certainly important, but they're not the only thing we're designing 
> for. You need to think about systems with pre-encoded (stored) media; 
> systems with radical imbalances in the processing power and device 
> capabilities between the endpoints; systems with different stream control
models, etc, etc.
> 
> 
> 
> 	       On the other hand, the multi m=
> 	       section approach has never been tried by anyone, that we are
> 	       aware of
> 
> 
> Do a web search for "Marconi ViPr."
> 
> /a

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Subject: Re: [MMUSIC] Multicast and one-m-line-per-stream (Re: Thoughts on m= lines and media streams)
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On 02/05/2013 04:52 PM, Adam Roach wrote:
> Before I respond, I do want to make sure that this is the same Harald
> Alvestrand who was proposing that we dismiss the vast majority of
> existing SDP attributes on the premise that they are of no interest to
> WEBRTC.

Told you I hated to do this :-)

>
> /a
>
> On 2/2/13 18:13, Harald Alvestrand wrote:
> > Adam,
> >
> > I hate to play the multicast card, because I'm fundamentally
> > uninterested in multicast at the moment, but....
> >
> > in your scheme 2, how would you represent multicast and
> > transport-mixer based sessions?
> >
> >
> > On 02/01/2013 01:17 AM, Adam Roach wrote:
> >> Based on the conversations we've had so far, I've put together the
> >> following summary of my thoughts on the topic of whether we want to
> >> put multiple media streams in a single m= line.
> >>
> >> http://www.ietf.org/id/draft-roach-mmusic-mlines-00.txt
> >>
> >> Sorry for the relatively short lead time before next week's meeting
> >> -- I had meant to put this kind of summary together a week or two
> >> ago, but have been far busier than I anticipated.
> >>
> >> /a
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From eckelcu@cisco.com  Tue Feb  5 11:44:00 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Roni Even <ron.even.tlv@gmail.com>, "'Adam Roach'" <adam@nostrum.com>, "'Justin Uberti'" <juberti@google.com>
Thread-Topic: [MMUSIC] More thoughts on multiple media streams in a single m= line
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Date: Tue, 5 Feb 2013 19:43:57 +0000
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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True, max-ssrc or something that provides similar functionality, would be u=
seful in all three.

Cheers,
Charles

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]
> Sent: Tuesday, February 05, 2013 12:58 PM
> To: Charles Eckel (eckelcu); 'Adam Roach'; 'Justin Uberti'
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] More thoughts on multiple media streams in a single
> m=3D line
>=20
> Charles,
> Even for 1 you need maxssrc, you assume that one m-line means one RTP
> stream.
> Roni
>=20
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of
> Charles Eckel (eckelcu)
> Sent: 05 February, 2013 6:20 PM
> To: Adam Roach; Justin Uberti
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single
> m=3D
> line
>=20
> I think we are all in agreement that for a large conference (e.g. 100+
> audio/video streams), we cannot require that each media stream have its
> own
> m=3Dsection, nor can we require that each media stream's SSRC is signaled=
 in
> SDP. For the sake of example, let's assume a centralized conference serve=
r
> is able to send up to 10 video streams, where each video stream may be an=
y
> of the 100 video streams at various points in time (e.g. active speakers
> switched in/out, specific sources requested, etc.), I think there are thr=
ee
> realistic alternatives:
>=20
> 1) Use 10 m=3Dsections to define the ten video streams, no multiplexing
> required
> 2) Use 1 m=3Dsection to define  a video stream, and indicate that up to 1=
0
> video streams are multiplexed within it
> 3) Use 'n' m=3Dsections to define the video streams, where 1<=3Dn<=3D10
>=20
> We already have all we need to do (1). There is a desire to improve on (1=
).
> If we define some additional syntax/semantics that enable (2), people can
> choose to use (2) or (3) depending on works best for their scenario. Ther=
e
> have already been some drafts submitted along these lines, I suggest to
> take
> a closer look at those and improve on them. Sounds reasonable?
>=20
> Charles
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Adam Roach
> > Sent: Tuesday, February 05, 2013 10:26 AM
> > To: Justin Uberti
> > Cc: mmusic@ietf.org
> > Subject: Re: [MMUSIC] More thoughts on multiple media streams in a
> > single m=3D line
> >
> > On 2/4/13 02:45, Justin Uberti wrote:
> >
> >
> > 	Sorry for the short lead time, but here's a quick read for the
> plane.
> >
> >
> > I addressed several of the assertions made in this draft in my
> > previous response (e.g. the need for glare resolution regardless of
> > which solution we select). Some additional thoughts:
> >
> >
> >
> > 	   o  Adding or removing a source requires sending a full list of
> all
> > 	      sources, which doesn't scale well to 10s or 100s of sources,
> > 	      especially when they are added or removed frequently.
> >
> >
> > Realistically, this isn't how multiparty conferences are handled. It's
> > unrealistic to think that my DSL connection is going to be sucking
> > down hundreds of video streams just because I'm in a video conference
> > with hundreds of people.
> >
> > In practice, the way existing conferencing systems handle this is to
> > send a small, manageable number of streams to each participant,
> > changing them out as necessary to have a reasonable user experience
> > (either under the control of the user, its own algorithms, or a
> combination of the two).
> >
> > Which is to say: even with hundreds of source streams being fed into a
> > conference system, you're really looking at a realistic upper bound on
> > m=3D sections of a dozen or two, simply because you can't reasonably
> > expect recipients to receive more than that many streams simultaneously=
.
> >
> > And really, if you're imagining hundreds of media streams flowing
> > towards an endpoint, any arguments about the bits consumed by SDP go
> > well beyond disingenuous.
> >
> >
> >
> > 	   o  Because this solution requires BUNDLE, which is still not
> widely
> > 	      supported, it can be deployed in fewer situations
> >
> >
> > This is completely out of left field: the alternative is MMT, which is
> > even less well developed than BUNDLE.
> >
> >
> >
> > 	   4.  The RFC5576-based approach has already been in use for many
> > years
> > 	       in a major video conferencing system (Google+ Hangouts), and
> > has
> > 	       worked very well.  We feel that we have searched the
> "unknown"
> > 	       space and solved all of the major issues.
> >
> >
> > I think this actually does a good job of illustrating your blind spot:
> > you're working within RTCWEB, but developing for a single, very precise
> use case.
> > For example, this kind of experience doesn't translate well into the
> > challenges that might arise from systems that are dealing with
> > radically different user experiences. Google-hangout-type uses are
> > certainly important, but they're not the only thing we're designing
> > for. You need to think about systems with pre-encoded (stored) media;
> > systems with radical imbalances in the processing power and device
> > capabilities between the endpoints; systems with different stream contr=
ol
> models, etc, etc.
> >
> >
> >
> > 	       On the other hand, the multi m=3D
> > 	       section approach has never been tried by anyone, that we are
> > 	       aware of
> >
> >
> > Do a web search for "Marconi ViPr."
> >
> > /a
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From harald@alvestrand.no  Tue Feb  5 20:28:36 2013
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Subject: [MMUSIC] The 171-attribute spreadsheet
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This is a multi-part message in MIME format.
--------------030404010108030502030206
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Just in case anyone else lost the link, the spreadsheet with all defined 
SDP attributes that an m-line can have is here:

https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=0

Everyone with the link should be able to edit.

             Harald


--------------030404010108030502030206
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Just in case anyone else lost the link, the spreadsheet with all
    defined SDP attributes that an m-line can have is here:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
href="https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=0">https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=0</a><br>
    <br>
    Everyone with the link should be able to edit.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
  </body>
</html>

--------------030404010108030502030206--

From ari.keranen@ericsson.com  Wed Feb  6 08:58:37 2013
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Subject: Re: [MMUSIC] [rtcweb] Boston meeting agenda
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Hi,

The agenda is (roughly) still as shown below. The afternoon sessions 
(RTCWEB/MMUSIC parts) start at 1:30 PM EST. The meeting material 
(including chair slides with updated agenda) is available here:
http://www.ietf.org/proceedings/interim/2013/02/05/rtcweb/proceedings.html


Cheers,
Ari

On 2/4/13 11:53 AM, Roni Even wrote:
> Hi,
> Is there a final agenda with time schedule. There is no relation between the
> agenda here and i=the one for mmusic and rtcweb in the reference.
> I would like to attend from remote and since there is time difference it
> will be helpful to plan ahead.
> Thanks
> Roni Even who will try to attend from remote being 7 hours ahead of Boston
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Stefan H?kansson LK
> Sent: 30 January, 2013 11:51 AM
> To: mmusic@ietf.org; rtcweb@ietf.org; public-webrtc@w3.org;
> public-media-capture@w3.org
> Subject: [rtcweb] Boston meeting agenda
>
> All,
>
> below is a merged agenda covering both W3C and IETF parts of the meeting.
> Details are likely to change - the main message is that W3C meetings will be
> before lunch, and IETF after. Before lunch could mean
> 8:30 - 12:30, after lunch 13:30 - 17:30; but the exact times are TBC. A hard
> stop is planned for 17:00 on Feb 7th as many have planes to catch.
>
> (The sources used to assemble the agenda below are the W3C meeting wiki's,
> [1][2], and the mail sent to mmusic and rtcweb Jan 14th [3]).
>
> --Stefan (for all involved chairs)
>
> Feb 5th
> =======
> Morning session: W3C Media Capture TF
> -------------------------------------
> * device enumeration
> * error handling
> * "immediate stream" gUM
> * Resource reservation
>
> Afternoon session: IETF mmusic and rtcweb WGs
> ---------------------------------------------
> * Multiple Flow SDP representation
> * SDP for Data channels
>
> Feb 6th
> =======
> Morning session: W3C Media Capture TF + WebRTC WG
> -------------------------------------------------
> * Identity aspects of MediaStreams (MediaCap)
> * Settings/constraints (MediaCap)
> * Recording (MediaCap)
> * Call flow (WebRTC)
>
> Afternoon session: IETF mmusic and rtcweb WGs
> ---------------------------------------------
> * SDP Grouping (Bundle/MMT/one-rtp)
> * Mapping stream/track label concepts to SDP identifiers
>
> Feb 7th
> =======
> Morning session: W3C WebRTC WG
> ------------------------------
> * MediaStream and MediaStreamTrack - SDP interface
> * Error handling
> * Data channel
> * Stats
>
> Afternoon session: IETF mmusic and rtcweb WGs
> ---------------------------------------------
> * Trickle ICE
> * JSEP
>
> [1] http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013
> [2] http://www.w3.org/2011/04/webrtc/wiki/February_6_-_February_7_2013
> [3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg06069.html
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From keith.drage@alcatel-lucent.com  Wed Feb  6 13:52:24 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, mmusic <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 6 Feb 2013 22:51:57 +0100
Thread-Topic: [rtcweb] The 171-attribute spreadsheet
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This is still missing any criteria for deciding whether something is in or =
out. I assume importance to RTCWEB has only two values from what I can see =
(1 =3D important, 0 =3D not important). I believe I could make a use case f=
or all of these being supported, I'm sure all of them can be found in SDP w=
hen interworking - the question is whether any of these cases are useful to=
 applications or not, and whether an otherwise successful communication wou=
ld fail if an attribute was not understood, and therefore dropped.

I can solve the problem of g.3gpp.cat and g.3gpp.crs for you. These are mis=
registrations and are in fact values of the content attribute (which is mis=
sing from your list because it has only just been inserted as a correction,=
 and probably is important to RTCWEB). It is in the process of being sorted=
 but requires some changes to the registration data in the references speci=
fication to make things clear.

Keith



________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: 06 February 2013 04:29
To: mmusic; rtcweb@ietf.org
Subject: [rtcweb] The 171-attribute spreadsheet

Just in case anyone else lost the link, the spreadsheet with all defined SD=
P attributes that an m-line can have is here:

https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpRV2pkbXVG=
NV9lUFI3cHlIRFE#gid=3D0

Everyone with the link should be able to edit.

            Harald

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>This is still missing any criteria for
deciding whether something is in or out. I assume importance to RTCWEB has =
only
two values from what I can see (1 =3D important, 0 =3D not important). I be=
lieve I
could make a use case for all of these being supported, I&#8217;m sure all =
of
them can be found in SDP when interworking &#8211; the question is whether =
any
of these cases are useful to applications or not, and whether an otherwise
successful communication would fail if an attribute was not understood, and
therefore dropped.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I can solve the problem of g.3gpp.cat =
and
g.3gpp.crs for you. These are misregistrations and are in fact values of th=
e
content attribute (which is missing from your list because it has only just
been inserted as a correction, and probably is important to RTCWEB). It is =
in
the process of being sorted but requires some changes to the registration d=
ata
in the references specification to make things clear.<o:p></o:p></span></fo=
nt></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Keith<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-siz=
e:12.0pt;
color:windowtext'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 color=3Dblack face=3DTahoma><span la=
ng=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:b=
old'>From:</span></font></b><font
size=3D2 color=3Dblack face=3DTahoma><span lang=3DEN-US style=3D'font-size:=
10.0pt;
font-family:Tahoma;color:windowtext'> rtcweb-bounces@ietf.org
[mailto:rtcweb-bounces@ietf.org] <b><span style=3D'font-weight:bold'>On Beh=
alf Of
</span></b>Harald Alvestrand<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 06 February 2013 04:29=
<br>
<b><span style=3D'font-weight:bold'>To:</span></b> mmusic; rtcweb@ietf.org<=
br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [rtcweb] The
171-attribute spreadsheet</span></font><font color=3Dblack><span lang=3DEN-=
US
style=3D'color:windowtext'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Just in case anyo=
ne else
lost the link, the spreadsheet with all defined SDP attributes that an m-li=
ne
can have is here:<br>
<br>
<a
href=3D"https://docs.google.com/spreadsheet/ccc?key=3D0Av-aF69sk5C6dDhPblpR=
V2pkbXVGNV9lUFI3cHlIRFE#gid=3D0">https://docs.google.com/spreadsheet/ccc?ke=
y=3D0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=3D0</a><br>
<br>
Everyone with the link should be able to edit.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<o=
:p></o:p></span></font></p>

</div>

</div>

</body>

</html>

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BUNDLE, ONE-RTP, MMT:

* Note - If we proceed with MMT, we should use m=application instead of 
m=anymedia

* Decision - Are you generally okay with any of the approaches that 
reduce the number of long-lived flows to this level OR is there some 
other feature in one of the proposals you must have before this meets 
your needs?

6 for the former, 17 for the latter


Cullen's plan to dig us out of this hole:

* Proposed Superset of Requirement 2 - Should be able to interop with 
legacy without the JS doing anything special.

* Proposed Relaxation of Requirement 2 - Should be able to interop with 
legacy, but not necessarily in one O/A exchange within a single dialog 
(without relying on failure cases).

* Requirement 3 - Change video flows to media flows

* Proposed Requirement - When we have a large conference with 1000's of 
people, we don't have to send new signaling to every participant of the 
conference.

* Proposed Requirement - Must be able to configure a set of similar 
flows such that once established, the endpoints can add or remove flows 
without having to do an O/A cycle (i.e., without having to exchange SDP).

* Proposed Requirement - Have this process terminate in some finite 
period of time.

* Point of Discussion - Plan A: same port on each m-line or different 
candidate sets?

* Action - Plan B: Write the drafts to extend SDP to explicitly describe 
multiple cameras with multiple streams from each camera under a single 
m-line.

* Decision - Should we come up with a set of requirements and proceed 
with Plan A and Plan B with the timelines outlined here?

24 for, 5 against

* Note - Mary Barnes refuses to go back to work until she gets her mini 
candy bars.

DataChannel SDP

* Action - Justin to draft API for SDP-only channel open

* Decision (rtcweb hats) -
   2a) In-band only - 6
   2b) SDP & in-band - 10
   2c) SDP-only - 15
   Work on both will continue until a decision is reached.

* Action - Martin believes he can send a proposal to the list which will 
change some minds.

From harald@alvestrand.no  Wed Feb  6 19:16:27 2013
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Subject: Re: [MMUSIC] [rtcweb] The 171-attribute spreadsheet
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On 02/06/2013 10:51 PM, DRAGE, Keith (Keith) wrote:
>
> This is still missing any criteria for deciding whether something is 
> in or out. I assume importance to RTCWEB has only two values from what 
> I can see (1 = important, 0 = not important). I believe I could make a 
> use case for all of these being supported, I'm sure all of them can be 
> found in SDP when interworking -- the question is whether any of these 
> cases are useful to applications or not, and whether an otherwise 
> successful communication would fail if an attribute was not 
> understood, and therefore dropped.
>
I don't think "in" or "out" is what we're looking for.
It's either Transport, Flow or Media - which gives us a basis for seeing 
how bigthe problem is:

- under BUNDLE, there must be new rules for Transport ("pick first", 
"sum values", or "other processing")
- under "two M-lines" and MMT, theFlow attributes need to be redefined 
to be per-SSRC, or we have to decide to ignore them
- I think the Media ones are OK with either proposal, unless some usage 
of them depends on there being only one flow in an M-line, in which case 
they're really Flow, even though they're marking Media

I put in only 1 and 0 for RTCWEB because I started in on the easy cases- 
if you feel a more graded scale is useful, please suggest values!

> I can solve the problem of g.3gpp.cat and g.3gpp.crs for you. These 
> are misregistrations and are in fact values of the content attribute 
> (which is missing from your list because it has only just been 
> inserted as a correction, and probably is important to RTCWEB). It is 
> in the process of being sorted but requires some changes to the 
> registration data in the references specification to make things clear.
>
values of a=content? That's good to know about. Hope you can put notes 
on those in the "notes" columns!

> Keith
>
> ------------------------------------------------------------------------
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On 
> Behalf Of *Harald Alvestrand
> *Sent:* 06 February 2013 04:29
> *To:* mmusic; rtcweb@ietf.org
> *Subject:* [rtcweb] The 171-attribute spreadsheet
>
> Just in case anyone else lost the link, the spreadsheet with all 
> defined SDP attributes that an m-line can have is here:
>
> https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=0
>
> Everyone with the link should be able to edit.
>
>             Harald
>


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    <div class="moz-cite-prefix">On 02/06/2013 10:51 PM, DRAGE, Keith
      (Keith) wrote:<br>
    </div>
    <blockquote
cite="mid:EDC0A1AE77C57744B664A310A0B23AE20D97134B8B@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com"
      type="cite">
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      <div class="Section1">
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy">This is still missing
              any criteria for
              deciding whether something is in or out. I assume
              importance to RTCWEB has only
              two values from what I can see (1 = important, 0 = not
              important). I believe I
              could make a use case for all of these being supported,
              I&#8217;m sure all of
              them can be found in SDP when interworking &#8211; the question
              is whether any
              of these cases are useful to applications or not, and
              whether an otherwise
              successful communication would fail if an attribute was
              not understood, and
              therefore dropped.</span></font></p>
      </div>
    </blockquote>
    <font color="navy"><font size="2"><font face="Arial">I don't think <font
            size="2">"in" or "out" is <font size="2">what we're looking
              for.<br>
              <font size="2">It's either Transport, Flow or Media -
                which gives us a basis for seeing how big<font size="2">
                  <font size="2">the problem is:<br>
                    <br>
                    <font size="2">- under <font size="2">BUNDLE, <font
                          size="2">there must <font size="2">be <font
                              size="2">ne<font size="2">w rule<font
                                  size="2">s <font size="2">for
                                    Transport ("pick first", "sum
                                    values", or "other proces<font
                                      size="2">sing")<br>
                                      <font size="2">- under "<font
                                          size="2">two M-lines" and MMT,
                                          the<font size="2"> <font
                                              size="2">Flow attributes
                                              need to be redefined to be
                                              per-SSRC, or we have to de<font
                                                size="2">cide to ign<font
                                                  size="2">ore them<br>
                                                  <font size="2">- I
                                                    think the <font
                                                      size="2">Media
                                                      ones are OK with
                                                      either proposal<font
                                                        size="2">,
                                                        unless some
                                                        usage of them
                                                        depends on there
                                                        being only one <font
                                                          size="2">flow
                                                          in an M-line<font
                                                          size="2">, in
                                                          which c<font
                                                          size="2">ase
                                                          they're <font
                                                          size="2">really
                                                          Flow, even
                                                          though they're
                                                          mark<font
                                                          size="2">ing
                                                          Media<br>
                                                          <br>
                                                          <font size="2">I
                                                          put in only 1
                                                          and 0 for RTCW<font
                                                          size="2">EB
                                                          because <font
                                                          size="2">I
                                                          started in on
                                                          the easy cases<font
                                                          size="2"> - if
                                                          you feel a
                                                          more g<font
                                                          size="2">raded
                                                          scale <font
                                                          size="2">is
                                                          useful, pl<font
                                                          size="2">ease
                                                          suggest
                                                          values!<br>
                                                          <br>
                                                          </font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font>
    <blockquote
cite="mid:EDC0A1AE77C57744B664A310A0B23AE20D97134B8B@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com"
      type="cite">
      <div class="Section1">
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p></o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy">I can solve the
              problem of g.3gpp.cat and
              g.3gpp.crs for you. These are misregistrations and are in
              fact values of the
              content attribute (which is missing from your list because
              it has only just
              been inserted as a correction, and probably is important
              to RTCWEB). It is in
              the process of being sorted but requires some changes to
              the registration data
              in the references specification to make things clear.</span></font></p>
      </div>
    </blockquote>
    <font color="navy"><font size="2"><font face="Arial"><font size="2">values
            of a=content? That's good to know about. <font size="2">Hope
              you can put notes on th<font size="2">ose in the "notes"
                columns!<br>
                <br>
              </font></font></font></font></font></font>
    <blockquote
cite="mid:EDC0A1AE77C57744B664A310A0B23AE20D97134B8B@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com"
      type="cite">
      <div class="Section1">
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p></o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy">Keith<o:p></o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <div>
            <div class="MsoNormal" style="text-align:center"
              align="center"><font color="black" face="Times New Roman"
                size="3"><span style="font-size:12.0pt;
                  color:windowtext" lang="EN-US">
                  <hr tabindex="-1" align="center" size="2" width="100%">
                </span></font></div>
            <p class="MsoNormal"><b><font color="black" face="Tahoma"
                  size="2"><span
style="font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:bold"
                    lang="EN-US">From:</span></font></b><font
                color="black" face="Tahoma" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Tahoma;color:windowtext" lang="EN-US">
                  <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                  [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] <b><span
                      style="font-weight:bold">On Behalf Of
                    </span></b>Harald Alvestrand<br>
                  <b><span style="font-weight:bold">Sent:</span></b> 06
                  February 2013 04:29<br>
                  <b><span style="font-weight:bold">To:</span></b>
                  mmusic; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b><span style="font-weight:bold">Subject:</span></b>
                  [rtcweb] The
                  171-attribute spreadsheet</span></font><font
                color="black"><span style="color:windowtext"
                  lang="EN-US"><o:p></o:p></span></font></p>
          </div>
          <p class="MsoNormal"><font color="black" face="Times New
              Roman" size="3"><span style="font-size:12.0pt"><o:p>&nbsp;</o:p></span></font></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><font
              color="black" face="Times New Roman" size="3"><span
                style="font-size:12.0pt">Just in case anyone else
                lost the link, the spreadsheet with all defined SDP
                attributes that an m-line
                can have is here:<br>
                <br>
                <a moz-do-not-send="true"
href="https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=0">https://docs.google.com/spreadsheet/ccc?key=0Av-aF69sk5C6dDhPblpRV2pkbXVGNV9lUFI3cHlIRFE#gid=0</a><br>
                <br>
                Everyone with the link should be able to edit.<br>
                <br>
                &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<o:p></o:p></span></font></p>
        </div>
      </div>
    </blockquote>
    <br>
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, Roni Even <ron.even.tlv@gmail.com>
Thread-Topic: [MMUSIC] [rtcweb] Boston meeting agenda
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Date: Thu, 7 Feb 2013 06:22:26 +0000
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Cc: "mmusic-chairs@tools.ietf.org" <mmusic-chairs@tools.ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "rtcweb-chairs@tools.ietf.org" <rtcweb-chairs@tools.ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Boston meeting agenda
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Are we recording these sessions? Will someone be sharing the URLs for the r=
ecording?

Muthu

|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf O=
f Ari Ker=E4nen
|Sent: Wednesday, February 06, 2013 10:29 PM
|To: Roni Even
|Cc: mmusic-chairs@tools.ietf.org; rtcweb@ietf.org; rtcweb-chairs@tools.iet=
f.org; mmusic@ietf.org
|Subject: Re: [MMUSIC] [rtcweb] Boston meeting agenda
|
|Hi,
|
|The agenda is (roughly) still as shown below. The afternoon sessions
|(RTCWEB/MMUSIC parts) start at 1:30 PM EST. The meeting material
|(including chair slides with updated agenda) is available here:
|http://www.ietf.org/proceedings/interim/2013/02/05/rtcweb/proceedings.html
|
|
|Cheers,
|Ari
|
|On 2/4/13 11:53 AM, Roni Even wrote:
|> Hi,
|> Is there a final agenda with time schedule. There is no relation between=
 the
|> agenda here and i=3Dthe one for mmusic and rtcweb in the reference.
|> I would like to attend from remote and since there is time difference it
|> will be helpful to plan ahead.
|> Thanks
|> Roni Even who will try to attend from remote being 7 hours ahead of Bost=
on
|>
|> -----Original Message-----
|> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf=
 Of
|> Stefan H?kansson LK
|> Sent: 30 January, 2013 11:51 AM
|> To: mmusic@ietf.org; rtcweb@ietf.org; public-webrtc@w3.org;
|> public-media-capture@w3.org
|> Subject: [rtcweb] Boston meeting agenda
|>
|> All,
|>
|> below is a merged agenda covering both W3C and IETF parts of the meeting=
.
|> Details are likely to change - the main message is that W3C meetings wil=
l be
|> before lunch, and IETF after. Before lunch could mean
|> 8:30 - 12:30, after lunch 13:30 - 17:30; but the exact times are TBC. A =
hard
|> stop is planned for 17:00 on Feb 7th as many have planes to catch.
|>
|> (The sources used to assemble the agenda below are the W3C meeting wiki'=
s,
|> [1][2], and the mail sent to mmusic and rtcweb Jan 14th [3]).
|>
|> --Stefan (for all involved chairs)
|>
|> Feb 5th
|> =3D=3D=3D=3D=3D=3D=3D
|> Morning session: W3C Media Capture TF
|> -------------------------------------
|> * device enumeration
|> * error handling
|> * "immediate stream" gUM
|> * Resource reservation
|>
|> Afternoon session: IETF mmusic and rtcweb WGs
|> ---------------------------------------------
|> * Multiple Flow SDP representation
|> * SDP for Data channels
|>
|> Feb 6th
|> =3D=3D=3D=3D=3D=3D=3D
|> Morning session: W3C Media Capture TF + WebRTC WG
|> -------------------------------------------------
|> * Identity aspects of MediaStreams (MediaCap)
|> * Settings/constraints (MediaCap)
|> * Recording (MediaCap)
|> * Call flow (WebRTC)
|>
|> Afternoon session: IETF mmusic and rtcweb WGs
|> ---------------------------------------------
|> * SDP Grouping (Bundle/MMT/one-rtp)
|> * Mapping stream/track label concepts to SDP identifiers
|>
|> Feb 7th
|> =3D=3D=3D=3D=3D=3D=3D
|> Morning session: W3C WebRTC WG
|> ------------------------------
|> * MediaStream and MediaStreamTrack - SDP interface
|> * Error handling
|> * Data channel
|> * Stats
|>
|> Afternoon session: IETF mmusic and rtcweb WGs
|> ---------------------------------------------
|> * Trickle ICE
|> * JSEP
|>
|> [1] http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013
|> [2] http://www.w3.org/2011/04/webrtc/wiki/February_6_-_February_7_2013
|> [3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg06069.html
|>
|>
|>
|>
|> _______________________________________________
|> rtcweb mailing list
|> rtcweb@ietf.org
|> https://www.ietf.org/mailman/listinfo/rtcweb
|>
|
|_______________________________________________
|mmusic mailing list
|mmusic@ietf.org
|https://www.ietf.org/mailman/listinfo/mmusic

From ron.even.tlv@gmail.com  Thu Feb  7 04:44:52 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Timothy B. Terriberry'" <tterriberry@mozilla.com>, <rtcweb@ietf.org>, <mmusic@ietf.org>
References: <5112DC51.30604@mozilla.com>
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Date: Thu, 7 Feb 2013 14:41:43 +0200
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Subject: Re: [MMUSIC] Minutes from 6 Feb joint interim session
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Hi,
On Action - Plan B: Write the drafts to extend SDP to explicitly describe
multiple cameras with multiple streams from each camera under a single
m-line.

We did it in CLUE look at
http://tools.ietf.org/html/draft-even-clue-rtp-mapping-04  examples is
section 6.1 and 6.2. (6.2 was taken out in the 05 version since we agreed in
CLUE that simulcast is out of scope for CLUE)

Roni Even


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Timothy B. Terriberry
Sent: 07 February, 2013 12:42 AM
To: rtcweb@ietf.org; mmusic@ietf.org
Subject: [MMUSIC] Minutes from 6 Feb joint interim session

BUNDLE, ONE-RTP, MMT:

* Note - If we proceed with MMT, we should use m=application instead of
m=anymedia

* Decision - Are you generally okay with any of the approaches that reduce
the number of long-lived flows to this level OR is there some other feature
in one of the proposals you must have before this meets your needs?

6 for the former, 17 for the latter


Cullen's plan to dig us out of this hole:

* Proposed Superset of Requirement 2 - Should be able to interop with legacy
without the JS doing anything special.

* Proposed Relaxation of Requirement 2 - Should be able to interop with
legacy, but not necessarily in one O/A exchange within a single dialog
(without relying on failure cases).

* Requirement 3 - Change video flows to media flows

* Proposed Requirement - When we have a large conference with 1000's of
people, we don't have to send new signaling to every participant of the
conference.

* Proposed Requirement - Must be able to configure a set of similar flows
such that once established, the endpoints can add or remove flows without
having to do an O/A cycle (i.e., without having to exchange SDP).

* Proposed Requirement - Have this process terminate in some finite period
of time.

* Point of Discussion - Plan A: same port on each m-line or different
candidate sets?

* Action - Plan B: Write the drafts to extend SDP to explicitly describe
multiple cameras with multiple streams from each camera under a single
m-line.

* Decision - Should we come up with a set of requirements and proceed with
Plan A and Plan B with the timelines outlined here?

24 for, 5 against

* Note - Mary Barnes refuses to go back to work until she gets her mini
candy bars.

DataChannel SDP

* Action - Justin to draft API for SDP-only channel open

* Decision (rtcweb hats) -
   2a) In-band only - 6
   2b) SDP & in-band - 10
   2c) SDP-only - 15
   Work on both will continue until a decision is reached.

* Action - Martin believes he can send a proposal to the list which will
change some minds.
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic


From mary.ietf.barnes@gmail.com  Thu Feb  7 06:01:51 2013
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Cc: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, "rtcweb-chairs@tools.ietf.org" <rtcweb-chairs@tools.ietf.org>, "mmusic-chairs@tools.ietf.org" <mmusic-chairs@tools.ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb]  Boston meeting agenda
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The intent was to record the Webex sessions, thus a link to the
recordings should be posted in the meeting minutes.  However, the
recording for the Feb. 6th session yesterday wasn't started until
partway into the session so there will be a gap.  Hopefully, someone
will remember to start the recording at the beginning of the session
tomorrow.

Mary.

On Thu, Feb 7, 2013 at 12:22 AM, Muthu Arul Mozhi Perumal (mperumal)
<mperumal@cisco.com> wrote:
> Are we recording these sessions? Will someone be sharing the URLs for the
> recording?
>
> Muthu
>
> |-----Original Message-----
> |From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf=
 Of
> Ari Ker=E4nen
> |Sent: Wednesday, February 06, 2013 10:29 PM
> |To: Roni Even
> |Cc: mmusic-chairs@tools.ietf.org; rtcweb@ietf.org;
> rtcweb-chairs@tools.ietf.org; mmusic@ietf.org
> |Subject: Re: [MMUSIC] [rtcweb] Boston meeting agenda
> |
> |Hi,
> |
> |The agenda is (roughly) still as shown below. The afternoon sessions
> |(RTCWEB/MMUSIC parts) start at 1:30 PM EST. The meeting material
> |(including chair slides with updated agenda) is available here:
> |http://www.ietf.org/proceedings/interim/2013/02/05/rtcweb/proceedings.ht=
ml
> |
> |
> |Cheers,
> |Ari
> |
> |On 2/4/13 11:53 AM, Roni Even wrote:
> |> Hi,
> |> Is there a final agenda with time schedule. There is no relation betwe=
en
> the
> |> agenda here and i=3Dthe one for mmusic and rtcweb in the reference.
> |> I would like to attend from remote and since there is time difference =
it
> |> will be helpful to plan ahead.
> |> Thanks
> |> Roni Even who will try to attend from remote being 7 hours ahead of
> Boston
> |>
> |> -----Original Message-----
> |> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Beha=
lf
> Of
> |> Stefan H?kansson LK
> |> Sent: 30 January, 2013 11:51 AM
> |> To: mmusic@ietf.org; rtcweb@ietf.org; public-webrtc@w3.org;
> |> public-media-capture@w3.org
> |> Subject: [rtcweb] Boston meeting agenda
> |>
> |> All,
> |>
> |> below is a merged agenda covering both W3C and IETF parts of the meeti=
ng.
> |> Details are likely to change - the main message is that W3C meetings w=
ill
> be
> |> before lunch, and IETF after. Before lunch could mean
> |> 8:30 - 12:30, after lunch 13:30 - 17:30; but the exact times are TBC. =
A
> hard
> |> stop is planned for 17:00 on Feb 7th as many have planes to catch.
> |>
> |> (The sources used to assemble the agenda below are the W3C meeting
> wiki's,
> |> [1][2], and the mail sent to mmusic and rtcweb Jan 14th [3]).
> |>
> |> --Stefan (for all involved chairs)
> |>
> |> Feb 5th
> |> =3D=3D=3D=3D=3D=3D=3D
> |> Morning session: W3C Media Capture TF
> |> -------------------------------------
> |> * device enumeration
> |> * error handling
> |> * "immediate stream" gUM
> |> * Resource reservation
> |>
> |> Afternoon session: IETF mmusic and rtcweb WGs
> |> ---------------------------------------------
> |> * Multiple Flow SDP representation
> |> * SDP for Data channels
> |>
> |> Feb 6th
> |> =3D=3D=3D=3D=3D=3D=3D
> |> Morning session: W3C Media Capture TF + WebRTC WG
> |> -------------------------------------------------
> |> * Identity aspects of MediaStreams (MediaCap)
> |> * Settings/constraints (MediaCap)
> |> * Recording (MediaCap)
> |> * Call flow (WebRTC)
> |>
> |> Afternoon session: IETF mmusic and rtcweb WGs
> |> ---------------------------------------------
> |> * SDP Grouping (Bundle/MMT/one-rtp)
> |> * Mapping stream/track label concepts to SDP identifiers
> |>
> |> Feb 7th
> |> =3D=3D=3D=3D=3D=3D=3D
> |> Morning session: W3C WebRTC WG
> |> ------------------------------
> |> * MediaStream and MediaStreamTrack - SDP interface
> |> * Error handling
> |> * Data channel
> |> * Stats
> |>
> |> Afternoon session: IETF mmusic and rtcweb WGs
> |> ---------------------------------------------
> |> * Trickle ICE
> |> * JSEP
> |>
> |> [1] http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013
> |> [2] http://www.w3.org/2011/04/webrtc/wiki/February_6_-_February_7_2013
> |> [3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg06069.html
> |>
> |>
> |>
> |>
> |> _______________________________________________
> |> rtcweb mailing list
> |> rtcweb@ietf.org
> |> https://www.ietf.org/mailman/listinfo/rtcweb
> |>
> |
> |_______________________________________________
> |mmusic mailing list
> |mmusic@ietf.org
> |https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From ted.ietf@gmail.com  Thu Feb  7 06:06:03 2013
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Date: Thu, 7 Feb 2013 09:06:02 -0500
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From: Ted Hardie <ted.ietf@gmail.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: Re: [MMUSIC] [rtcweb]  Boston meeting agenda
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On Thu, Feb 7, 2013 at 9:01 AM, Mary Barnes <mary.ietf.barnes@gmail.com> wr=
ote:
> The intent was to record the Webex sessions, thus a link to the
> recordings should be posted in the meeting minutes.  However, the
> recording for the Feb. 6th session yesterday wasn't started until
> partway into the session so there will be a gap.  Hopefully, someone
> will remember to start the recording at the beginning of the session
> tomorrow.

Apologies for missing the recording yesterday; we will try to do
better today.  Nagging from those in the room would be welcome.


Ted

>
> Mary.
>
> On Thu, Feb 7, 2013 at 12:22 AM, Muthu Arul Mozhi Perumal (mperumal)
> <mperumal@cisco.com> wrote:
>> Are we recording these sessions? Will someone be sharing the URLs for th=
e
>> recording?
>>
>> Muthu
>>
>> |-----Original Message-----
>> |From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behal=
f Of
>> Ari Ker=E4nen
>> |Sent: Wednesday, February 06, 2013 10:29 PM
>> |To: Roni Even
>> |Cc: mmusic-chairs@tools.ietf.org; rtcweb@ietf.org;
>> rtcweb-chairs@tools.ietf.org; mmusic@ietf.org
>> |Subject: Re: [MMUSIC] [rtcweb] Boston meeting agenda
>> |
>> |Hi,
>> |
>> |The agenda is (roughly) still as shown below. The afternoon sessions
>> |(RTCWEB/MMUSIC parts) start at 1:30 PM EST. The meeting material
>> |(including chair slides with updated agenda) is available here:
>> |http://www.ietf.org/proceedings/interim/2013/02/05/rtcweb/proceedings.h=
tml
>> |
>> |
>> |Cheers,
>> |Ari
>> |
>> |On 2/4/13 11:53 AM, Roni Even wrote:
>> |> Hi,
>> |> Is there a final agenda with time schedule. There is no relation betw=
een
>> the
>> |> agenda here and i=3Dthe one for mmusic and rtcweb in the reference.
>> |> I would like to attend from remote and since there is time difference=
 it
>> |> will be helpful to plan ahead.
>> |> Thanks
>> |> Roni Even who will try to attend from remote being 7 hours ahead of
>> Boston
>> |>
>> |> -----Original Message-----
>> |> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Beh=
alf
>> Of
>> |> Stefan H?kansson LK
>> |> Sent: 30 January, 2013 11:51 AM
>> |> To: mmusic@ietf.org; rtcweb@ietf.org; public-webrtc@w3.org;
>> |> public-media-capture@w3.org
>> |> Subject: [rtcweb] Boston meeting agenda
>> |>
>> |> All,
>> |>
>> |> below is a merged agenda covering both W3C and IETF parts of the meet=
ing.
>> |> Details are likely to change - the main message is that W3C meetings =
will
>> be
>> |> before lunch, and IETF after. Before lunch could mean
>> |> 8:30 - 12:30, after lunch 13:30 - 17:30; but the exact times are TBC.=
 A
>> hard
>> |> stop is planned for 17:00 on Feb 7th as many have planes to catch.
>> |>
>> |> (The sources used to assemble the agenda below are the W3C meeting
>> wiki's,
>> |> [1][2], and the mail sent to mmusic and rtcweb Jan 14th [3]).
>> |>
>> |> --Stefan (for all involved chairs)
>> |>
>> |> Feb 5th
>> |> =3D=3D=3D=3D=3D=3D=3D
>> |> Morning session: W3C Media Capture TF
>> |> -------------------------------------
>> |> * device enumeration
>> |> * error handling
>> |> * "immediate stream" gUM
>> |> * Resource reservation
>> |>
>> |> Afternoon session: IETF mmusic and rtcweb WGs
>> |> ---------------------------------------------
>> |> * Multiple Flow SDP representation
>> |> * SDP for Data channels
>> |>
>> |> Feb 6th
>> |> =3D=3D=3D=3D=3D=3D=3D
>> |> Morning session: W3C Media Capture TF + WebRTC WG
>> |> -------------------------------------------------
>> |> * Identity aspects of MediaStreams (MediaCap)
>> |> * Settings/constraints (MediaCap)
>> |> * Recording (MediaCap)
>> |> * Call flow (WebRTC)
>> |>
>> |> Afternoon session: IETF mmusic and rtcweb WGs
>> |> ---------------------------------------------
>> |> * SDP Grouping (Bundle/MMT/one-rtp)
>> |> * Mapping stream/track label concepts to SDP identifiers
>> |>
>> |> Feb 7th
>> |> =3D=3D=3D=3D=3D=3D=3D
>> |> Morning session: W3C WebRTC WG
>> |> ------------------------------
>> |> * MediaStream and MediaStreamTrack - SDP interface
>> |> * Error handling
>> |> * Data channel
>> |> * Stats
>> |>
>> |> Afternoon session: IETF mmusic and rtcweb WGs
>> |> ---------------------------------------------
>> |> * Trickle ICE
>> |> * JSEP
>> |>
>> |> [1] http://www.w3.org/wiki/Media_Capture/Feb_5-6_2013
>> |> [2] http://www.w3.org/2011/04/webrtc/wiki/February_6_-_February_7_201=
3
>> |> [3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg06069.html
>> |>
>> |>
>> |>
>> |>
>> |> _______________________________________________
>> |> rtcweb mailing list
>> |> rtcweb@ietf.org
>> |> https://www.ietf.org/mailman/listinfo/rtcweb
>> |>
>> |
>> |_______________________________________________
>> |mmusic mailing list
>> |mmusic@ietf.org
>> |https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>

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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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Hi

I took a look the trickle-ice draft and have a couple of questions/comments:

1) Section 1 says
<quote>
    This specification does not define usage of trickle ICE with any
    specific signalling protocol, contrary to [RFC5245] which contains a
    usage for ICE wht SIP.  Such usages would have to be specified in
    separate documents.
</quote>
however you do define it specifically using SDP and Offer/Answer (I 
believe). It would be worthwhile calling that out.

2) Section 5.1 says
<quote>
    As mentioned earlier in this section, Offers and Answers can contain
    any set of candidates, which means that a trickle ICE session
    description MAY contain no candidates at all.  In such cases the
    agent would still need to place an address in the "c=" line(s). If
    the use of a host address there is undesirable (e.g. for privacy
    reasons), the agent MAY set the connection address to 0.0.0.0. In
    this case it MUST also set the port number to 1.  There is no need to
    include a fictitious 0.0.0.0 candidate when doing so.
<quote>
I had to read this several times before I (think) I understood that when 
you say "the agent MAY set the connection address to 0.0.0.0", you are 
referring to the connection address in the "a=candidate" attribute, not 
the "c=" line. Can you clarify and rephrase accordingly.

3) Section 9.1
I'm confused about this whole section and it's premise that 
"a=candidate" attributes and media descriptions are not clearly linked. 
The "a=candidate" attribute is defined as a media-level attribute only 
and describes as such in RFC 5245, so I must be missing something here.


4) Section 9.2. Rather than talking about sending "end of candidates" 
via "the signalling channel" (which is not defined), could we change 
this to say and "end of candidates" need to be sent to the peer, and the 
only mechanism defined to do so in this document is through an O/A 
exchange (new exchange or answer to an offer) ?

5) Section 14 (Example)
The exchange of the "additional candidates" does not use an O/A 
exchange. As for comment 4), this assumes some other un-defined 
mechanism somewhere - would it be useful to show it as an O/A exchange 
as well, but note that other mechanisms may be defined ?

6) Section A.1 (MID/Stream Indicates in SDP)
I'm still unclear on this (per comment 3).


Thanks

-- Flemming (as Individual)



On 1/30/13 11:38 AM, Emil Ivov wrote:
> Hey all,
>
> Justin, Eric and I have just submitted an update to the trickle ICE
> draft, accommodating comments from Atlanta:
>
> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>
> More details about the changes here:
> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B
>
> Diff2:
> http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&difftype=--html&submit=Go!&url2=draft-ivov-mmusic-trickle-ice-00
>
> During the MMUSIC presentation and the mailing list discussions it was
> also pointed out that it is important for this work to happen in
> parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
> therefore submitted the following document:
>
> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00
>
> As always, comments are welcome!
>
> Cheers,
> Emil
>
> -------- Original Message --------
> Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
> Date: Mon, 28 Jan 2013 11:39:21 -0800
> From: internet-drafts@ietf.org
> To: emcho@jitsi.org
> CC: justin@uberti.name, ekr@rtfm.com
>
>
> A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
> has been successfully submitted by Emil Ivov and posted to the
> IETF repository.
>
> Filename:	 draft-ivov-mmusic-trickle-ice
> Revision:	 00
> Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
> Interactive Connectivity Establishment (ICE) Protocol
> Creation date:	 2013-01-28
> WG ID:		 Individual Submission
> Number of pages: 20
> URL:
> http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt
> Status:
> http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
> Htmlized:        http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>
>
> Abstract:
>     This document describes an extension to the Interactive Connectivity
>     Establishment (ICE) protocol that allows ICE agents to send and
>     receive candidates incrementally rather than exchanging complete
>     lists.  With such incremental provisioning, ICE agents can begin
>     connectivity checks while they are still gathering candidates and
>     considerably shorten the time necessary for ICE processing to
>     complete.
>
>     The above mechanism is also referred to as "trickle ICE".
>
>
>
>
>
> The IETF Secretariat
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


--------------060104060608000709020806
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Hi<br>
    <br>
    I took a look the trickle-ice draft and have a couple of
    questions/comments:<br>
    <br>
    1) Section 1 says<br>
    &lt;quote&gt;<br>
    <meta charset="utf-8">
    &nbsp;&nbsp; This specification does not define usage of trickle ICE with any<br>
    &nbsp;&nbsp; specific signalling protocol, contrary to [RFC5245] which
    contains a<br>
    &nbsp;&nbsp; usage for ICE wht SIP.&nbsp; Such usages would have to be specified in<br>
    &nbsp;&nbsp; separate documents.<br>
    &lt;/quote&gt;<br>
    however you do define it specifically using SDP and Offer/Answer (I
    believe). It would be worthwhile calling that out. <br>
    <br>
    2) Section 5.1 says<br>
    &lt;quote&gt;<br>
    &nbsp;&nbsp; As mentioned earlier in this section, Offers and Answers can
    contain<br>
    &nbsp;&nbsp; any set of candidates, which means that a trickle ICE session<br>
    &nbsp;&nbsp; description MAY contain no candidates at all.&nbsp; In such cases the<br>
    &nbsp;&nbsp; agent would still need to place an address in the "c=" line(s).&nbsp;
    If<br>
    &nbsp;&nbsp; the use of a host address there is undesirable (e.g. for privacy<br>
    &nbsp;&nbsp; reasons), the agent MAY set the connection address to 0.0.0.0.&nbsp;
    In<br>
    &nbsp;&nbsp; this case it MUST also set the port number to 1.&nbsp; There is no
    need to<br>
    &nbsp;&nbsp; include a fictitious 0.0.0.0 candidate when doing so.<br>
    &lt;quote&gt;<br>
    I had to read this several times before I (think) I understood that
    when you say "the agent MAY set the connection address to 0.0.0.0",
    you are referring to the connection address in the "a=candidate"
    attribute, not the "c=" line. Can you clarify and rephrase
    accordingly. <br>
    <br>
    3) Section 9.1<br>
    I'm confused about this whole section and it's premise that
    "a=candidate" attributes and media descriptions are not clearly
    linked. The "a=candidate" attribute is defined as a media-level
    attribute only and describes as such in RFC 5245, so I must be
    missing something here. <br>
    <br>
    <br>
    4) Section 9.2. Rather than talking about sending "end of
    candidates" via "the signalling channel" (which is not defined),
    could we change this to say and "end of candidates" need to be sent
    to the peer, and the only mechanism defined to do so in this
    document is through an O/A exchange (new exchange or answer to an
    offer) ? <br>
    <br>
    5) Section 14 (Example)<br>
    The exchange of the "additional candidates" does not use an O/A
    exchange. As for comment 4), this assumes some other un-defined
    mechanism somewhere - would it be useful to show it as an O/A
    exchange as well, but note that other mechanisms may be defined ? <br>
    <br>
    6) Section A.1 (MID/Stream Indicates in SDP)<br>
    I'm still unclear on this (per comment 3). <br>
    <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming (as Individual)<br>
    <br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 1/30/13 11:38 AM, Emil Ivov wrote:<br>
    </div>
    <blockquote cite="mid:51094C89.7020302@jitsi.org" type="cite">
      <pre wrap="">Hey all,

Justin, Eric and I have just submitted an update to the trickle ICE
draft, accommodating comments from Atlanta:

<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00</a>

More details about the changes here:
<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B</a>

Diff2:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&amp;difftype=--html&amp;submit=Go!&amp;url2=draft-ivov-mmusic-trickle-ice-00">http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&amp;difftype=--html&amp;submit=Go!&amp;url2=draft-ivov-mmusic-trickle-ice-00</a>

During the MMUSIC presentation and the mailing list discussions it was
also pointed out that it is important for this work to happen in
parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
therefore submitted the following document:

<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00</a>

As always, comments are welcome!

Cheers,
Emil

-------- Original Message --------
Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
Date: Mon, 28 Jan 2013 11:39:21 -0800
From: <a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>
To: <a class="moz-txt-link-abbreviated" href="mailto:emcho@jitsi.org">emcho@jitsi.org</a>
CC: <a class="moz-txt-link-abbreviated" href="mailto:justin@uberti.name">justin@uberti.name</a>, <a class="moz-txt-link-abbreviated" href="mailto:ekr@rtfm.com">ekr@rtfm.com</a>


A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
has been successfully submitted by Emil Ivov and posted to the
IETF repository.

Filename:	 draft-ivov-mmusic-trickle-ice
Revision:	 00
Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) Protocol
Creation date:	 2013-01-28
WG ID:		 Individual Submission
Number of pages: 20
URL:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt">http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt</a>
Status:
<a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice">http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice</a>
Htmlized:        <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00</a>


Abstract:
   This document describes an extension to the Interactive Connectivity
   Establishment (ICE) protocol that allows ICE agents to send and
   receive candidates incrementally rather than exchanging complete
   lists.  With such incremental provisioning, ICE agents can begin
   connectivity checks while they are still gathering candidates and
   considerably shorten the time necessary for ICE processing to
   complete.

   The above mechanism is also referred to as "trickle ICE".





The IETF Secretariat




_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
.

</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------060104060608000709020806--

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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On 2/7/13 10:00 AM, Flemming Andreasen wrote:
> Hi
>
> I took a look the trickle-ice draft and have a couple of 
> questions/comments:
>
> 1) Section 1 says
> <quote>
>    This specification does not define usage of trickle ICE with any
>    specific signalling protocol, contrary to [RFC5245] which contains a
>    usage for ICE wht SIP.  Such usages would have to be specified in
>    separate documents.
> </quote>
> however you do define it specifically using SDP and Offer/Answer (I 
> believe). It would be worthwhile calling that out.
>
> 2) Section 5.1 says
> <quote>
>    As mentioned earlier in this section, Offers and Answers can contain
>    any set of candidates, which means that a trickle ICE session
>    description MAY contain no candidates at all.  In such cases the
>    agent would still need to place an address in the "c=" line(s).  If
>    the use of a host address there is undesirable (e.g. for privacy
>    reasons), the agent MAY set the connection address to 0.0.0.0. In
>    this case it MUST also set the port number to 1.  There is no need to
>    include a fictitious 0.0.0.0 candidate when doing so.
> <quote>
> I had to read this several times before I (think) I understood that 
> when you say "the agent MAY set the connection address to 0.0.0.0", 
> you are referring to the connection address in the "a=candidate" 
> attribute, not the "c=" line. Can you clarify and rephrase accordingly.
>
> 3) Section 9.1
> I'm confused about this whole section and it's premise that 
> "a=candidate" attributes and media descriptions are not clearly 
> linked. The "a=candidate" attribute is defined as a media-level 
> attribute only and describes as such in RFC 5245, so I must be missing 
> something here.
>
>
> 4) Section 9.2. Rather than talking about sending "end of candidates" 
> via "the signalling channel" (which is not defined), could we change 
> this to say and "end of candidates" need to be sent to the peer, and 
> the only mechanism defined to do so in this document is through an O/A 
> exchange (new exchange or answer to an offer) ?
>
> 5) Section 14 (Example)
> The exchange of the "additional candidates" does not use an O/A 
> exchange. As for comment 4), this assumes some other un-defined 
> mechanism somewhere - would it be useful to show it as an O/A exchange 
> as well, but note that other mechanisms may be defined ?
>
> 6) Section A.1 (MID/Stream Indicates in SDP)
> I'm still unclear on this (per comment 3).
>
7) It seems the inclusion of the "trickle" ice-options is what will 
trigger the use of tricke-ICE. I think this is a slight overloading of 
the ice-options semantics inasmuch as it is intended to indicate support 
for a given extension, which does not necessarily imply you actually 
want to use that extension. I think it would be useful to have a 
separate way of indicating you also want to use trickle-ICE.

Thanks

-- Flemming


>
> Thanks
>
> -- Flemming (as Individual)
>
>
>
> On 1/30/13 11:38 AM, Emil Ivov wrote:
>> Hey all,
>>
>> Justin, Eric and I have just submitted an update to the trickle ICE
>> draft, accommodating comments from Atlanta:
>>
>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>
>> More details about the changes here:
>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B
>>
>> Diff2:
>> http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&difftype=--html&submit=Go!&url2=draft-ivov-mmusic-trickle-ice-00
>>
>> During the MMUSIC presentation and the mailing list discussions it was
>> also pointed out that it is important for this work to happen in
>> parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
>> therefore submitted the following document:
>>
>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00
>>
>> As always, comments are welcome!
>>
>> Cheers,
>> Emil
>>
>> -------- Original Message --------
>> Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
>> Date: Mon, 28 Jan 2013 11:39:21 -0800
>> From:internet-drafts@ietf.org
>> To:emcho@jitsi.org
>> CC:justin@uberti.name,ekr@rtfm.com
>>
>>
>> A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
>> has been successfully submitted by Emil Ivov and posted to the
>> IETF repository.
>>
>> Filename:	 draft-ivov-mmusic-trickle-ice
>> Revision:	 00
>> Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
>> Interactive Connectivity Establishment (ICE) Protocol
>> Creation date:	 2013-01-28
>> WG ID:		 Individual Submission
>> Number of pages: 20
>> URL:
>> http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt
>> Status:
>> http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
>> Htmlized:http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>
>>
>> Abstract:
>>     This document describes an extension to the Interactive Connectivity
>>     Establishment (ICE) protocol that allows ICE agents to send and
>>     receive candidates incrementally rather than exchanging complete
>>     lists.  With such incremental provisioning, ICE agents can begin
>>     connectivity checks while they are still gathering candidates and
>>     considerably shorten the time necessary for ICE processing to
>>     complete.
>>
>>     The above mechanism is also referred to as "trickle ICE".
>>
>>
>>
>>
>>
>> The IETF Secretariat
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> .
>>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <br>
    <div class="moz-cite-prefix">On 2/7/13 10:00 AM, Flemming Andreasen
      wrote:<br>
    </div>
    <blockquote cite="mid:5113C177.3000507@cisco.com" type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      Hi<br>
      <br>
      I took a look the trickle-ice draft and have a couple of
      questions/comments:<br>
      <br>
      1) Section 1 says<br>
      &lt;quote&gt;<br>
      &nbsp;&nbsp; This specification does not define usage of trickle ICE with
      any<br>
      &nbsp;&nbsp; specific signalling protocol, contrary to [RFC5245] which
      contains a<br>
      &nbsp;&nbsp; usage for ICE wht SIP.&nbsp; Such usages would have to be specified
      in<br>
      &nbsp;&nbsp; separate documents.<br>
      &lt;/quote&gt;<br>
      however you do define it specifically using SDP and Offer/Answer
      (I believe). It would be worthwhile calling that out. <br>
      <br>
      2) Section 5.1 says<br>
      &lt;quote&gt;<br>
      &nbsp;&nbsp; As mentioned earlier in this section, Offers and Answers can
      contain<br>
      &nbsp;&nbsp; any set of candidates, which means that a trickle ICE session<br>
      &nbsp;&nbsp; description MAY contain no candidates at all.&nbsp; In such cases
      the<br>
      &nbsp;&nbsp; agent would still need to place an address in the "c="
      line(s).&nbsp; If<br>
      &nbsp;&nbsp; the use of a host address there is undesirable (e.g. for
      privacy<br>
      &nbsp;&nbsp; reasons), the agent MAY set the connection address to 0.0.0.0.&nbsp;
      In<br>
      &nbsp;&nbsp; this case it MUST also set the port number to 1.&nbsp; There is no
      need to<br>
      &nbsp;&nbsp; include a fictitious 0.0.0.0 candidate when doing so.<br>
      &lt;quote&gt;<br>
      I had to read this several times before I (think) I understood
      that when you say "the agent MAY set the connection address to
      0.0.0.0", you are referring to the connection address in the
      "a=candidate" attribute, not the "c=" line. Can you clarify and
      rephrase accordingly. <br>
      <br>
      3) Section 9.1<br>
      I'm confused about this whole section and it's premise that
      "a=candidate" attributes and media descriptions are not clearly
      linked. The "a=candidate" attribute is defined as a media-level
      attribute only and describes as such in RFC 5245, so I must be
      missing something here. <br>
      <br>
      <br>
      4) Section 9.2. Rather than talking about sending "end of
      candidates" via "the signalling channel" (which is not defined),
      could we change this to say and "end of candidates" need to be
      sent to the peer, and the only mechanism defined to do so in this
      document is through an O/A exchange (new exchange or answer to an
      offer) ? <br>
      <br>
      5) Section 14 (Example)<br>
      The exchange of the "additional candidates" does not use an O/A
      exchange. As for comment 4), this assumes some other un-defined
      mechanism somewhere - would it be useful to show it as an O/A
      exchange as well, but note that other mechanisms may be defined ?
      <br>
      <br>
      6) Section A.1 (MID/Stream Indicates in SDP)<br>
      I'm still unclear on this (per comment 3). <br>
      <br>
    </blockquote>
    7) It seems the inclusion of the "trickle" ice-options is what will
    trigger the use of tricke-ICE. I think this is a slight overloading
    of the ice-options semantics inasmuch as it is intended to indicate
    support for a given extension, which does not necessarily imply you
    actually want to use that extension. I think it would be useful to
    have a separate way of indicating you also want to use trickle-ICE.
    <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <br>
    <br>
    <blockquote cite="mid:5113C177.3000507@cisco.com" type="cite"> <br>
      Thanks <br>
      <br>
      -- Flemming (as Individual)<br>
      <br>
      <br>
      <br>
      <div class="moz-cite-prefix">On 1/30/13 11:38 AM, Emil Ivov wrote:<br>
      </div>
      <blockquote cite="mid:51094C89.7020302@jitsi.org" type="cite">
        <pre wrap="">Hey all,

Justin, Eric and I have just submitted an update to the trickle ICE
draft, accommodating comments from Atlanta:

<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00</a>

More details about the changes here:
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B</a>

Diff2:
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&amp;difftype=--html&amp;submit=Go%21&amp;url2=draft-ivov-mmusic-trickle-ice-00">http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&amp;difftype=--html&amp;submit=Go!&amp;url2=draft-ivov-mmusic-trickle-ice-00</a>

During the MMUSIC presentation and the mailing list discussions it was
also pointed out that it is important for this work to happen in
parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
therefore submitted the following document:

<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00</a>

As always, comments are welcome!

Cheers,
Emil

-------- Original Message --------
Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
Date: Mon, 28 Jan 2013 11:39:21 -0800
From: <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>
To: <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:emcho@jitsi.org">emcho@jitsi.org</a>
CC: <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:justin@uberti.name">justin@uberti.name</a>, <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:ekr@rtfm.com">ekr@rtfm.com</a>


A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
has been successfully submitted by Emil Ivov and posted to the
IETF repository.

Filename:	 draft-ivov-mmusic-trickle-ice
Revision:	 00
Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) Protocol
Creation date:	 2013-01-28
WG ID:		 Individual Submission
Number of pages: 20
URL:
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt">http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt</a>
Status:
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice">http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice</a>
Htmlized:        <a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00">http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00</a>


Abstract:
   This document describes an extension to the Interactive Connectivity
   Establishment (ICE) protocol that allows ICE agents to send and
   receive candidates incrementally rather than exchanging complete
   lists.  With such incremental provisioning, ICE agents can begin
   connectivity checks while they are still gathering candidates and
   considerably shorten the time necessary for ICE processing to
   complete.

   The above mechanism is also referred to as "trickle ICE".





The IETF Secretariat




_______________________________________________
mmusic mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
.

</pre>
      </blockquote>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On Thu, Feb 7, 2013 at 10:10 AM, Flemming Andreasen <fandreas@cisco.com> wrote:
>

> 7) It seems the inclusion of the "trickle" ice-options is what will trigger
> the use of tricke-ICE. I think this is a slight overloading of the
> ice-options semantics inasmuch as it is intended to indicate support for a
> given extension, which does not necessarily imply you actually want to use
> that extension. I think it would be useful to have a separate way of
> indicating you also want to use trickle-ICE.
>

Why wouldn't just starting trickle-ICE be sufficient indication, given
that it has been indicated as supported?  I think the half-trickle
option covers the case where you don't know that it is supported, but
it may be I am misunderstanding the issue.

regards,

Ted

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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On 2/7/13 10:19 AM, Ted Hardie wrote:
> On Thu, Feb 7, 2013 at 10:10 AM, Flemming Andreasen <fandreas@cisco.com> wrote:
>> 7) It seems the inclusion of the "trickle" ice-options is what will trigger
>> the use of tricke-ICE. I think this is a slight overloading of the
>> ice-options semantics inasmuch as it is intended to indicate support for a
>> given extension, which does not necessarily imply you actually want to use
>> that extension. I think it would be useful to have a separate way of
>> indicating you also want to use trickle-ICE.
>>
> Why wouldn't just starting trickle-ICE be sufficient indication,
How would you know you started trickle-ICE or vanilla ICE ?

-- Flemming
> given
> that it has been indicated as supported?  I think the half-trickle
> option covers the case where you don't know that it is supported, but
> it may be I am misunderstanding the issue.
>
> regards,
>
> Ted
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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From: Ted Hardie <ted.ietf@gmail.com>
To: Flemming Andreasen <fandreas@cisco.com>
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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On Thu, Feb 7, 2013 at 10:27 AM, Flemming Andreasen <fandreas@cisco.com> wrote:
>
> On 2/7/13 10:19 AM, Ted Hardie wrote:
>>
>> On Thu, Feb 7, 2013 at 10:10 AM, Flemming Andreasen <fandreas@cisco.com>
>> wrote:
>>>
>>> 7) It seems the inclusion of the "trickle" ice-options is what will
>>> trigger
>>> the use of tricke-ICE. I think this is a slight overloading of the
>>> ice-options semantics inasmuch as it is intended to indicate support for
>>> a
>>> given extension, which does not necessarily imply you actually want to
>>> use
>>> that extension. I think it would be useful to have a separate way of
>>> indicating you also want to use trickle-ICE.
>>>
>> Why wouldn't just starting trickle-ICE be sufficient indication,
>
> How would you know you started trickle-ICE or vanilla ICE ?
>

Well, you'd have a very strong indication if the candidate list were
empty, and a reasonable indication if it contained only host
candidates.   If that weren't the case, you'd have to keep open the
possibility that more candidates might come in.

But I think I've been thinking of this the other way around--if both
sides are known to support trickle-ICE, the client sends
a=end-of-candidates whenever it's done *even if that is on the first
candidate set sent*.  Perhaps that is "vanilla ICE with sprinkles",
but it seems to solve the problem without another indicator/round of
negotiation.  As you note, that may overload the the ice-option
semantics, but it seems likely to work reasonable well.

regards,

Ted


> -- Flemming
>>
>> given
>> that it has been indicated as supported?  I think the half-trickle
>> option covers the case where you don't know that it is supported, but
>> it may be I am misunderstanding the issue.
>>
>> regards,
>>
>> Ted
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>

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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On 2/7/13 11:02 AM, Ted Hardie wrote:
> On Thu, Feb 7, 2013 at 10:27 AM, Flemming Andreasen <fandreas@cisco.com> wrote:
>> On 2/7/13 10:19 AM, Ted Hardie wrote:
>>> On Thu, Feb 7, 2013 at 10:10 AM, Flemming Andreasen <fandreas@cisco.com>
>>> wrote:
>>>> 7) It seems the inclusion of the "trickle" ice-options is what will
>>>> trigger
>>>> the use of tricke-ICE. I think this is a slight overloading of the
>>>> ice-options semantics inasmuch as it is intended to indicate support for
>>>> a
>>>> given extension, which does not necessarily imply you actually want to
>>>> use
>>>> that extension. I think it would be useful to have a separate way of
>>>> indicating you also want to use trickle-ICE.
>>>>
>>> Why wouldn't just starting trickle-ICE be sufficient indication,
>> How would you know you started trickle-ICE or vanilla ICE ?
>>
> Well, you'd have a very strong indication if the candidate list were
> empty, and a reasonable indication if it contained only host
> candidates.   If that weren't the case, you'd have to keep open the
> possibility that more candidates might come in.
The above would seem to favor an explicit indication.

> But I think I've been thinking of this the other way around--if both
> sides are known to support trickle-ICE, the client sends
> a=end-of-candidates whenever it's done *even if that is on the first
> candidate set sent*.
Not sure I follow - this would terminate trickle-ICE prematurely, no ?

> Perhaps that is "vanilla ICE with sprinkles",
> but it seems to solve the problem without another indicator/round of
> negotiation.  As you note, that may overload the the ice-option
> semantics, but it seems likely to work reasonable well.
I don't think it does since it would seem to terminate trickle-ICE 
prematurely, but maybe we should wait for the authors to clarify.

Thanks

-- Flemming


> regards,
>
> Ted
>
>
>> -- Flemming
>>> given
>>> that it has been indicated as supported?  I think the half-trickle
>>> option covers the case where you don't know that it is supported, but
>>> it may be I am misunderstanding the issue.
>>>
>>> regards,
>>>
>>> Ted
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


From ted.ietf@gmail.com  Thu Feb  7 08:18:41 2013
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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On Thu, Feb 7, 2013 at 11:09 AM, Flemming Andreasen <fandreas@cisco.com> wrote:
>  If that weren't the case, you'd have to keep open the
>> possibility that more candidates might come in.
>
> The above would seem to favor an explicit indication.
>
>
>> But I think I've been thinking of this the other way around--if both
>> sides are known to support trickle-ICE, the client sends
>> a=end-of-candidates whenever it's done *even if that is on the first
>> candidate set sent*.
>
> Not sure I follow - this would terminate trickle-ICE prematurely, no ?
>
It would terminate it, but not prematurely.  The client sends
a=end-of-candidates whenever it's done sending candidates; it just
allows you to send that indicator if you already had the full list on
the first use.  So if it did the blocking version of candidate
gathering, it sends that as the indicator that it is done.

>
>> Perhaps that is "vanilla ICE with sprinkles",
>> but it seems to solve the problem without another indicator/round of
>> negotiation.  As you note, that may overload the the ice-option
>> semantics, but it seems likely to work reasonable well.
>
> I don't think it does since it would seem to terminate trickle-ICE
> prematurely, but maybe we should wait for the authors to clarify.
>

Well, I'd certainly be happy to have the authors or others chime in.
But I am personally okay with the presumption that you always use
Trickle-ICE if Trickle-ICE is available at both ends, because it is
easy to get Vanilla ICE semantics using Trickle-ICE syntax.  I agree,
though, that this overloads the ice-option.

regards,

Ted Hardie

From matthew@matthew.at  Thu Feb  7 08:54:02 2013
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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On 2/7/2013 8:18 AM, Ted Hardie wrote:
>
> It would terminate it, but not prematurely.  The client sends
> a=end-of-candidates whenever it's done sending candidates; it just
> allows you to send that indicator if you already had the full list on
> the first use.  So if it did the blocking version of candidate
> gathering, it sends that as the indicator that it is done.

Isn't part of the point of trickle-ICE that you potentially are never 
done sending candidates? (Interface and connectivity changes and the 
like, avoiding a complete session restart by testing and switching to 
new candidates on the fly)

Matthew Kaufman


From ted.ietf@gmail.com  Thu Feb  7 08:58:36 2013
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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On Thu, Feb 7, 2013 at 11:53 AM, Matthew Kaufman <matthew@matthew.at> wrote:
> On 2/7/2013 8:18 AM, Ted Hardie wrote:
>>
>>
>> It would terminate it, but not prematurely.  The client sends
>> a=end-of-candidates whenever it's done sending candidates; it just
>> allows you to send that indicator if you already had the full list on
>> the first use.  So if it did the blocking version of candidate
>> gathering, it sends that as the indicator that it is done.
>
>
> Isn't part of the point of trickle-ICE that you potentially are never done
> sending candidates? (Interface and connectivity changes and the like,
> avoiding a complete session restart by testing and switching to new
> candidates on the fly)

You could use it that way, but it supports modes that allow you to
claim you are done.  The "send host candidates first, then trickle
server reflexive and call it done" is one called out in the draft.
Since you can claim you are done, getting the semantics of vanilla ICE
seems easy enough.

Just my two cents,

Ted

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

From ted.ietf@gmail.com  Thu Feb  7 09:09:29 2013
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To: rtcweb@ietf.org, mmusic@ietf.org
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Subject: [MMUSIC] IETF Interim starting fifteen minutes early at 1:15 p.m. Boston time
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In order to give folks a little extra time to get to the airport,
we're going to start at 1:15 and aim to finish on time at 5:00.


Ted

From emil@sip-communicator.org  Thu Feb  7 09:28:03 2013
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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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Hey all,

On 07.02.13, 11:58, Ted Hardie wrote:
> On Thu, Feb 7, 2013 at 11:53 AM, Matthew Kaufman <matthew@matthew.at> wrote:
>> On 2/7/2013 8:18 AM, Ted Hardie wrote:
>>>
>>>
>>> It would terminate it, but not prematurely.  The client sends
>>> a=end-of-candidates whenever it's done sending candidates; it just
>>> allows you to send that indicator if you already had the full list on
>>> the first use.  So if it did the blocking version of candidate
>>> gathering, it sends that as the indicator that it is done.
>>
>>
>> Isn't part of the point of trickle-ICE that you potentially are never done
>> sending candidates? (Interface and connectivity changes and the like,
>> avoiding a complete session restart by testing and switching to new
>> candidates on the fly)
> 
> You could use it that way, but it supports modes that allow you to
> claim you are done.  The "send host candidates first, then trickle
> server reflexive and call it done" is one called out in the draft.
> Since you can claim you are done, getting the semantics of vanilla ICE
> seems easy enough.

Yup that's the idea. The whole point of having end-of-candidates
indications is to basically say: I am not done till I say I am done (and
I really am when I say I am).

Emil
> 
> Just my two cents,
> 
> Ted


-- 
https://jitsi.org

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Hey Flemming, Ted, all,

On 07.02.13, 10:10, Flemming Andreasen wrote:
> 7) It seems the inclusion of the "trickle" ice-options is what will
> trigger the use of tricke-ICE. I think this is a slight overloading of
> the ice-options semantics inasmuch as it is intended to indicate support
> for a given extension, which does not necessarily imply you actually
> want to use that extension. I think it would be useful to have a
> separate way of indicating you also want to use trickle-ICE.

The intention of the current trickle ICE text is to say the following:

1. When your (the offerer's) intention is to do half trickle you include
the "trickle" ice-options token and you are prepared to have the
answerer trickling. You are also prepared to have them do vanilla ICE.
In either case you can either include end-of-candidates in your offer or
you can choose to not include it depending on how you intend to behave
if the answerer turns out to support trickle.

1.1. If the answer does indicate support trickle ICE by the answerer,
then you are both abiding by the trickle-ice spec. You can both still
choose to behave like vanilla ICE agents and send complete sets of
candidates from the start but you know that you have to handle trickle
semantics and you know that you have to send end-of-candidates when you
are done.

1.2 If the answerer does not support trickle, then there's no "trickle"
ice-options token in the answer and the offerer knows that trickling is
no longer possible (regardless of whether or not there was
end-of-candidates in the offer).

2. If the offerer's intention is to generate a vanilla ICE offer then it
does not include the "trickle" ice-options token and everyone just uses
vanilla ICE semantics.

I believe that this is the part in 5245 that you had a problem with and
I think that this is the text you are referring to:

    First, ICE provides the a=ice-options SDP attribute.  Each extension
    or change to ICE is associated with a token.  When an agent
    supporting such an extension or change generates an offer or an
    answer, it MUST include the token for that extension in this
    attribute.  *This allows each side to know what the other side is
    doing*.  This attribute MUST NOT be present if the agent doesn't
    support any ICE extensions or changes.

I am not sure exactly what the intention of this text was (maybe we
could clarify this in 5245bis) but that highlighted part makes me think
the point was to make sure everyone knows what behaviour to expect
rather than what features they have in their ICE stack even if they are
not planning on using them.

In that sense, if I send what I intend to be a vanilla offer then you
can count on me not to use any trickle semantics and you would make me
really unhappy if you did it yourself (because I might be gateway-ing
your stuff to a vanilla agent). In that sense I'd really like you to
behave as if I were a vanilla ICE agent.

In other words, I believe it would be legitimate for me not to include
the "trickle" token in ice-options.

Does this make sense?

Emil

> 
> Thanks
> 
> -- Flemming
> 
> 
>>
>> Thanks
>>
>> -- Flemming (as Individual)
>>
>>
>>
>> On 1/30/13 11:38 AM, Emil Ivov wrote:
>>> Hey all,
>>>
>>> Justin, Eric and I have just submitted an update to the trickle ICE
>>> draft, accommodating comments from Atlanta:
>>>
>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>
>>> More details about the changes here:
>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B
>>>
>>> Diff2:
>>> http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&difftype=--html&submit=Go!&url2=draft-ivov-mmusic-trickle-ice-00
>>>
>>> During the MMUSIC presentation and the mailing list discussions it was
>>> also pointed out that it is important for this work to happen in
>>> parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
>>> therefore submitted the following document:
>>>
>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00
>>>
>>> As always, comments are welcome!
>>>
>>> Cheers,
>>> Emil
>>>
>>> -------- Original Message --------
>>> Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
>>> Date: Mon, 28 Jan 2013 11:39:21 -0800
>>> From: internet-drafts@ietf.org
>>> To: emcho@jitsi.org
>>> CC: justin@uberti.name, ekr@rtfm.com
>>>
>>>
>>> A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
>>> has been successfully submitted by Emil Ivov and posted to the
>>> IETF repository.
>>>
>>> Filename:	 draft-ivov-mmusic-trickle-ice
>>> Revision:	 00
>>> Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
>>> Interactive Connectivity Establishment (ICE) Protocol
>>> Creation date:	 2013-01-28
>>> WG ID:		 Individual Submission
>>> Number of pages: 20
>>> URL:
>>> http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt
>>> Status:
>>> http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
>>> Htmlized:        http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>
>>>
>>> Abstract:
>>>    This document describes an extension to the Interactive Connectivity
>>>    Establishment (ICE) protocol that allows ICE agents to send and
>>>    receive candidates incrementally rather than exchanging complete
>>>    lists.  With such incremental provisioning, ICE agents can begin
>>>    connectivity checks while they are still gathering candidates and
>>>    considerably shorten the time necessary for ICE processing to
>>>    complete.
>>>
>>>    The above mechanism is also referred to as "trickle ICE".
>>>
>>>
>>>
>>>
>>>
>>> The IETF Secretariat
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> .
>>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> 

-- 
https://jitsi.org

From harald@alvestrand.no  Fri Feb  8 12:56:49 2013
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References: <CAOJ7v-3Ndb7vz9gchQ0Y308uH4nFUrKv8AkmeNrdf-3F03MLQg@mail.gmail.com> <5111247B.8060904@nostrum.com>
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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This is a multi-part message in MIME format.
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On 02/05/2013 04:25 PM, Adam Roach wrote:
> On 2/4/13 02:45, Justin Uberti wrote:
>> Sorry for the short lead time, but here's a quick read for the plane.
>
> I addressed several of the assertions made in this draft in my 
> previous response (e.g. the need for glare resolution regardless of 
> which solution we select). Some additional thoughts:
>
>>     o  Adding or removing a source requires sending a full list of all
>>        sources, which doesn't scale well to 10s or 100s of sources,
>>        especially when they are added or removed frequently.
>
> Realistically, this isn't how multiparty conferences are handled. It's 
> unrealistic to think that my DSL connection is going to be sucking 
> down hundreds of video streams just because I'm in a video conference 
> with hundreds of people.

(challenging assumptions)
why is it realistic to assume that you're on a DSL line?
In Kansas City, a hundred video streams each doing 100 Kbits/s or so may 
be a completely realistic way of doing conference-participation 
thumbnails; it's only 10 Mbits, which isn't that much on a gigabit link.

(with today's codecs, CPU may be a more limiting factor. It's the DSL 
part I reacted to.)



--------------050508080600050302040005
Content-Type: text/html; charset=ISO-8859-1
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 02/05/2013 04:25 PM, Adam Roach
      wrote:<br>
    </div>
    <blockquote cite="mid:5111247B.8060904@nostrum.com" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 2/4/13 02:45, Justin Uberti wrote:<br>
      </div>
      <blockquote
cite="mid:CAOJ7v-3Ndb7vz9gchQ0Y308uH4nFUrKv8AkmeNrdf-3F03MLQg@mail.gmail.com"
        type="cite">
        <div dir="ltr">Sorry for the short lead time, but here's a quick
          read for the plane.</div>
      </blockquote>
      <br>
      I addressed several of the assertions made in this draft in my
      previous response (e.g. the need for glare resolution regardless
      of which solution we select). Some additional thoughts:<br>
      <br>
      <blockquote type="cite">
        <meta http-equiv="content-type" content="text/html;
          charset=ISO-8859-1">
        <pre>   o  Adding or removing a source requires sending a full list of all
      sources, which doesn't scale well to 10s or 100s of sources,
      especially when they are added or removed frequently.</pre>
      </blockquote>
      <br>
      Realistically, this isn't how multiparty conferences are handled.
      It's unrealistic to think that my DSL connection is going to be
      sucking down hundreds of video streams just because I'm in a video
      conference with hundreds of people.<br>
    </blockquote>
    <br>
    (challenging assumptions)<br>
    why is it realistic to assume that you're on a DSL line?<br>
    In Kansas City, a hundred video streams each doing 100 Kbits/s or so
    may be a completely realistic way of doing conference-participation
    thumbnails; it's only 10 Mbits, which isn't that much on a gigabit
    link.<br>
    <br>
    (with today's codecs, CPU may be a more limiting factor. It's the
    DSL part I reacted to.)<br>
    <br>
    <br>
  </body>
</html>

--------------050508080600050302040005--

From andrew.hutton@siemens-enterprise.com  Fri Feb  8 19:01:33 2013
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
Thread-Topic: Requirements for Bundle, Fumble etc.
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Subject: [MMUSIC] Requirements for Bundle, Fumble etc.
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Hi,

I suggest that the requirements discussed at the interim (http://www.ietf.o=
rg/proceedings/interim/2013/02/05/rtcweb/slides/slides-interim-2013-rtcweb-=
1-10.pdf) regarding bundling on to a single 5-tuple need the following addi=
tional requirements.

1. Must work for a dual stack client when ICE candidates are used for addre=
ss family negotiation. I think this is interesting because in this scenario=
 there will of course be different candidates for each address family and w=
e will probably need some rules to make sure this works.

2. Must work when a TURN server is used. This is interesting because althou=
gh a single TURN allocation will be needed for the bundled candidates it wi=
ll I assume be necessary to do additional TURN allocations just in case the=
 peer does not support bundle.

Regards
Andy=20

From emil@sip-communicator.org  Sat Feb  9 10:18:30 2013
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Subject: Re: [MMUSIC] Requirements for Bundle, Fumble etc.
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Hey all,

On 08.02.13, 22:01, Hutton, Andrew wrote:
> 2. Must work when a TURN server is used. This is interesting because
> although a single TURN allocation will be needed for the bundled
> candidates it will I assume be necessary to do additional TURN
> allocations just in case the peer does not support bundle.

This can actually be stated in a more generic way. I already mentioned
it to Cullen but it's probably worth putting it on the list too:

Some of the solutions that we discussed in Boston had the property of
requiring multiple port allocations which would translate into multiple
sets of ICE candidates. Note that, given how the whole point here is to
talk to non-webrtc offer/answer devices, offerers may often be choosing
not to use full trickle ICE and only resort to half trickle (or even
vanilla ICE).

Also note that, contrary to dedicated devices, WebRTC applications would
often have no indication that a call is about to occur, until a "Call
Us" button on a web page is clicked.

In other words, using solutions like one-rtp or MMT would mean that
clients would have to go through all the (paced) steps of harvesting
candidates in a blocking manner, while the user is waiting and without
even the option of using the vanilla ICE optimization of pre-harvesting
them in anticipation for a call.

I am not sure if this qualifies as a requirement but it would probably
be good to keep it in mind.

Cheers,
Emil

-- 
https://jitsi.org

From worley@shell01.TheWorld.com  Sat Feb  9 19:47:40 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [MMUSIC] Requirements for Bundle, Fumble etc.
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> From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
> 
> 1. Must work for a dual stack client when ICE candidates are used
> for address family negotiation. I think this is interesting because
> in this scenario there will of course be different candidates for
> each address family and we will probably need some rules to make
> sure this works.

Doesn't ICE already have rules for negotiating both IPv4 and IPv6
addresses simultaneously?  (I'm not very familiar with ICE, but
skimming RFC 5245 suggests that SDP can provide candidates with both
IPv4 and IPv6 addresses, and that the ICE negotiation handles them
both.)

Dale

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To: "Dale R. Worley" <worley@ariadne.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Requirements for Bundle, Fumble etc.
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On 09 February 2013 22:47 Dale R. Worley wrote:


>=20
> Doesn't ICE already have rules for negotiating both IPv4 and IPv6
> addresses simultaneously?  (I'm not very familiar with ICE, but
> skimming RFC 5245 suggests that SDP can provide candidates with both
> IPv4 and IPv6 addresses, and that the ICE negotiation handles them
> both.)
>=20

Absolutely ICE does have these rules we do however need to make sure that t=
hey work when bundle is being negotiated at the same time.

Andy

From internet-drafts@ietf.org  Sun Feb 10 22:26:07 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-msid-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Cross Session Stream Identification in the Session Descr=
iption Protocol
	Author(s)       : Harald Alvestrand
	Filename        : draft-ietf-mmusic-msid-00.txt
	Pages           : 13
	Date            : 2013-02-10

Abstract:
   This document specifies a grouping mechanism for RTP media streams
   that can be used to specify relations between media streams within
   different RTP sessions as well as within a single RTP session, and
   independently of whether these media streams are described within one
   SDP m-line or in multiple m-lines.

   This mechanism is used to signal the association between the RTP
   concept of SSRC and the WebRTC concept of "MediaStream" /
   "MediaStreamTrack" using SDP signaling.

   This document is a work item of the MMUSIC WG, whose discussion list
   is mmusic@ietf.org.



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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-msid-00


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


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Cc: Ari Keranen <ari.keranen@ericsson.com>, MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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Hi Emil

As noted at the Interim meeting, I think it's cleaner to have 
ice-options used to strictly indicate features supported, and then have 
a separate way of indicating their use.

However, in taking another look at RFC 5245, it's somewhat unclear what 
semantics were intended. Section 15.5 for example suggests "supported" 
semantics, however Section 21.1.7 suggests "used" semantics. There are a 
few more examples of each throughout the doc so I agree with you that 
5245bis should clarify this (and depending on that clarification, we 
should use it the same way here).

Thanks

-- Flemming


On 2/7/13 1:00 PM, Emil Ivov wrote:
> Hey Flemming, Ted, all,
>
> On 07.02.13, 10:10, Flemming Andreasen wrote:
>> 7) It seems the inclusion of the "trickle" ice-options is what will
>> trigger the use of tricke-ICE. I think this is a slight overloading of
>> the ice-options semantics inasmuch as it is intended to indicate support
>> for a given extension, which does not necessarily imply you actually
>> want to use that extension. I think it would be useful to have a
>> separate way of indicating you also want to use trickle-ICE.
> The intention of the current trickle ICE text is to say the following:
>
> 1. When your (the offerer's) intention is to do half trickle you include
> the "trickle" ice-options token and you are prepared to have the
> answerer trickling. You are also prepared to have them do vanilla ICE.
> In either case you can either include end-of-candidates in your offer or
> you can choose to not include it depending on how you intend to behave
> if the answerer turns out to support trickle.
>
> 1.1. If the answer does indicate support trickle ICE by the answerer,
> then you are both abiding by the trickle-ice spec. You can both still
> choose to behave like vanilla ICE agents and send complete sets of
> candidates from the start but you know that you have to handle trickle
> semantics and you know that you have to send end-of-candidates when you
> are done.
>
> 1.2 If the answerer does not support trickle, then there's no "trickle"
> ice-options token in the answer and the offerer knows that trickling is
> no longer possible (regardless of whether or not there was
> end-of-candidates in the offer).
>
> 2. If the offerer's intention is to generate a vanilla ICE offer then it
> does not include the "trickle" ice-options token and everyone just uses
> vanilla ICE semantics.
>
> I believe that this is the part in 5245 that you had a problem with and
> I think that this is the text you are referring to:
>
>      First, ICE provides the a=ice-options SDP attribute.  Each extension
>      or change to ICE is associated with a token.  When an agent
>      supporting such an extension or change generates an offer or an
>      answer, it MUST include the token for that extension in this
>      attribute.  *This allows each side to know what the other side is
>      doing*.  This attribute MUST NOT be present if the agent doesn't
>      support any ICE extensions or changes.
>
> I am not sure exactly what the intention of this text was (maybe we
> could clarify this in 5245bis) but that highlighted part makes me think
> the point was to make sure everyone knows what behaviour to expect
> rather than what features they have in their ICE stack even if they are
> not planning on using them.
>
> In that sense, if I send what I intend to be a vanilla offer then you
> can count on me not to use any trickle semantics and you would make me
> really unhappy if you did it yourself (because I might be gateway-ing
> your stuff to a vanilla agent). In that sense I'd really like you to
> behave as if I were a vanilla ICE agent.
>
> In other words, I believe it would be legitimate for me not to include
> the "trickle" token in ice-options.
>
> Does this make sense?
>
> Emil
>
>> Thanks
>>
>> -- Flemming
>>
>>
>>> Thanks
>>>
>>> -- Flemming (as Individual)
>>>
>>>
>>>
>>> On 1/30/13 11:38 AM, Emil Ivov wrote:
>>>> Hey all,
>>>>
>>>> Justin, Eric and I have just submitted an update to the trickle ICE
>>>> draft, accommodating comments from Atlanta:
>>>>
>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>>
>>>> More details about the changes here:
>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendix-B
>>>>
>>>> Diff2:
>>>> http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-01&difftype=--html&submit=Go!&url2=draft-ivov-mmusic-trickle-ice-00
>>>>
>>>> During the MMUSIC presentation and the mailing list discussions it was
>>>> also pointed out that it is important for this work to happen in
>>>> parallel with a SIP usage for trickle ICE. Enrico, Christer and I have
>>>> therefore submitted the following document:
>>>>
>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00
>>>>
>>>> As always, comments are welcome!
>>>>
>>>> Cheers,
>>>> Emil
>>>>
>>>> -------- Original Message --------
>>>> Subject: New Version Notification for draft-ivov-mmusic-trickle-ice-00.txt
>>>> Date: Mon, 28 Jan 2013 11:39:21 -0800
>>>> From: internet-drafts@ietf.org
>>>> To: emcho@jitsi.org
>>>> CC: justin@uberti.name, ekr@rtfm.com
>>>>
>>>>
>>>> A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
>>>> has been successfully submitted by Emil Ivov and posted to the
>>>> IETF repository.
>>>>
>>>> Filename:	 draft-ivov-mmusic-trickle-ice
>>>> Revision:	 00
>>>> Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
>>>> Interactive Connectivity Establishment (ICE) Protocol
>>>> Creation date:	 2013-01-28
>>>> WG ID:		 Individual Submission
>>>> Number of pages: 20
>>>> URL:
>>>> http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-00.txt
>>>> Status:
>>>> http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
>>>> Htmlized:        http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>>
>>>>
>>>> Abstract:
>>>>     This document describes an extension to the Interactive Connectivity
>>>>     Establishment (ICE) protocol that allows ICE agents to send and
>>>>     receive candidates incrementally rather than exchanging complete
>>>>     lists.  With such incremental provisioning, ICE agents can begin
>>>>     connectivity checks while they are still gathering candidates and
>>>>     considerably shorten the time necessary for ICE processing to
>>>>     complete.
>>>>
>>>>     The above mechanism is also referred to as "trickle ICE".
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> The IETF Secretariat
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>> .
>>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic


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Date: Mon, 11 Feb 2013 21:07:23 -0500
From: Flemming Andreasen <fandreas@cisco.com>
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Subject: [MMUSIC] MMUSIC agenda requests for IETF 86
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MMUSIC is currently scheduled to meet for a total of 4 hours at IETF 86 
in Orlando, Florida.

If you want to request a time slot to discuss a draft, please send an 
e-mail to the chairs.

As usual, we want to ensure we take advantage of our meeting time in a 
productive manner that leads to forward progress. This means that we 
want presenters to come prepared to discuss open issues that have 
already been raised on the mailing list and ideally received some 
feedback already. Priority will be given drafts that gather attention 
and discussion in the mailing list prior to the meeting.

Thanks

-- Ari and Flemming

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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-bundle-negotiation-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Multiplexing Negotiation Using Session Description Proto=
col (SDP) Port Numbers
	Author(s)       : Christer Holmberg
                          Harald Tveit Alvestrand
	Filename        : draft-ietf-mmusic-sdp-bundle-negotiation-02.txt
	Pages           : 11
	Date            : 2013-02-12

Abstract:
   This specification defines a new SDP Grouping Framework SDP grouping
   framework extension, "BUNDLE", that can be used with the Session
   Description Protocol (SDP) Offer/Answer mechanism to negotiate the
   usage of bundled media, which refers to the usage of a single 5-tuple
   for media associated with multiple SDP media descriptions ("m=3D"
   lines).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-bundle-negotiation

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-bundle-negotiation=
-02


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


From christer.holmberg@ericsson.com  Tue Feb 12 02:13:03 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Draft new version: bundle-02
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Subject: [MMUSIC] Draft new version: bundle-02
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Hi,

Due to approaching expiration of the previous version, I've submitted a new=
 version (-02) of draft-bundle.

There are no changes since the previous version.

Regards,

Christer

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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-micr=
osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Due to approaching expiration of the previous versio=
n, I&#8217;ve submitted a new version (-02) of draft-bundle.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">There are no changes since the previous version.<o:p=
></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
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--_000_7594FB04B1934943A5C02806D1A2204B0D7367ESESSMB209ericsso_--

From gonzalo.camarillo@ericsson.com  Tue Feb 12 03:45:07 2013
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Date: Tue, 12 Feb 2013 13:45:04 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [MMUSIC] Interim meeting and way forward
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Folks,

first of all, I hope everybody has already managed to get back home
after the storm. Also, I want to thank Hadriel and Acme for being great
hosts. The information flow, the meeting facilities, the catering, the
pullovers, and everything else was just perfect.

With respect to the technical work that got done in Boston and, more
importantly, that remains to be done, I want to share a few thoughts
with the group.

A fact that is obvious but some of us sometimes forget about is that
RTCWeb, as any other technology, has a window of opportunity. The
usefulness and relevance of any standards we develop in this area will
be extremely high within that window, but will decrease dramatically if
the window closes (e.g., if too many new applications out there are
based on non-standard mechanisms).

While we have everything in the group to meet the relevant market's
deadlines, we should keep them in mind nevertheless. In particular, we
are going to be facing cases where *any* decision is better than *no*
decision, or where the *perfect* will be the enemy of the *good*.

In practical terms, the group should think twice before opening
already-made decisions and open them only for very good reasons. Also,
the group is going to have to make decisions based on *rough* consensus,
as opposed to *wide* consensus or unanimity. This may sound like
motherhood and apple pie but I want to stress these points anyway since
they are important and need to be kept in mind.

For Orlando, I have allocated a *significant* amount of time to RTCWeb
and MMUSIC. Those sessions will allow us to take advantage of the
momentum we currently have after the interim and to make more progress
in all the relevant specifications.

Cheers,

Gonzalo
AD for the RTCWeb and MMUSIC WGs


From christer.holmberg@ericsson.com  Tue Feb 12 04:04:35 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Flemming Andreasen <fandreas@cisco.com>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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References: <20130128193921.20420.53308.idtracker@ietfa.amsl.com> <51094C89.7020302@jitsi.org> <5113C177.3000507@cisco.com> <5113C3FB.3000600@cisco.com> <5113EBCD.7050901@jitsi.org> <51199BD6.8070606@cisco.com>
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Cc: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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Hi,

When a SIP dialog is established, entities normally don't indicate support =
for features that they don't want to use within that dialog.

However, if the semantics of ice-options is unclear, another alternative is=
 to define a SIP option-tag, which provides more freedom to specify the sem=
antics for individual option-tags.

Regards,

Christer

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Flemming Andreasen
Sent: 12. helmikuuta 2013 3:33
To: Emil Ivov
Cc: Ari Ker=E4nen; MMUSIC IETF WG
Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a ne=
w SIP usage draft.]

Hi Emil

As noted at the Interim meeting, I think it's cleaner to have ice-options u=
sed to strictly indicate features supported, and then have a separate way o=
f indicating their use.

However, in taking another look at RFC 5245, it's somewhat unclear what sem=
antics were intended. Section 15.5 for example suggests "supported"=20
semantics, however Section 21.1.7 suggests "used" semantics. There are a fe=
w more examples of each throughout the doc so I agree with you that 5245bis=
 should clarify this (and depending on that clarification, we should use it=
 the same way here).

Thanks

-- Flemming


On 2/7/13 1:00 PM, Emil Ivov wrote:
> Hey Flemming, Ted, all,
>
> On 07.02.13, 10:10, Flemming Andreasen wrote:
>> 7) It seems the inclusion of the "trickle" ice-options is what will=20
>> trigger the use of tricke-ICE. I think this is a slight overloading=20
>> of the ice-options semantics inasmuch as it is intended to indicate=20
>> support for a given extension, which does not necessarily imply you=20
>> actually want to use that extension. I think it would be useful to=20
>> have a separate way of indicating you also want to use trickle-ICE.
> The intention of the current trickle ICE text is to say the following:
>
> 1. When your (the offerer's) intention is to do half trickle you=20
> include the "trickle" ice-options token and you are prepared to have=20
> the answerer trickling. You are also prepared to have them do vanilla ICE=
.
> In either case you can either include end-of-candidates in your offer=20
> or you can choose to not include it depending on how you intend to=20
> behave if the answerer turns out to support trickle.
>
> 1.1. If the answer does indicate support trickle ICE by the answerer,=20
> then you are both abiding by the trickle-ice spec. You can both still=20
> choose to behave like vanilla ICE agents and send complete sets of=20
> candidates from the start but you know that you have to handle trickle=20
> semantics and you know that you have to send end-of-candidates when=20
> you are done.
>
> 1.2 If the answerer does not support trickle, then there's no "trickle"
> ice-options token in the answer and the offerer knows that trickling=20
> is no longer possible (regardless of whether or not there was=20
> end-of-candidates in the offer).
>
> 2. If the offerer's intention is to generate a vanilla ICE offer then=20
> it does not include the "trickle" ice-options token and everyone just=20
> uses vanilla ICE semantics.
>
> I believe that this is the part in 5245 that you had a problem with=20
> and I think that this is the text you are referring to:
>
>      First, ICE provides the a=3Dice-options SDP attribute.  Each extensi=
on
>      or change to ICE is associated with a token.  When an agent
>      supporting such an extension or change generates an offer or an
>      answer, it MUST include the token for that extension in this
>      attribute.  *This allows each side to know what the other side is
>      doing*.  This attribute MUST NOT be present if the agent doesn't
>      support any ICE extensions or changes.
>
> I am not sure exactly what the intention of this text was (maybe we=20
> could clarify this in 5245bis) but that highlighted part makes me=20
> think the point was to make sure everyone knows what behaviour to=20
> expect rather than what features they have in their ICE stack even if=20
> they are not planning on using them.
>
> In that sense, if I send what I intend to be a vanilla offer then you=20
> can count on me not to use any trickle semantics and you would make me=20
> really unhappy if you did it yourself (because I might be gateway-ing=20
> your stuff to a vanilla agent). In that sense I'd really like you to=20
> behave as if I were a vanilla ICE agent.
>
> In other words, I believe it would be legitimate for me not to include=20
> the "trickle" token in ice-options.
>
> Does this make sense?
>
> Emil
>
>> Thanks
>>
>> -- Flemming
>>
>>
>>> Thanks
>>>
>>> -- Flemming (as Individual)
>>>
>>>
>>>
>>> On 1/30/13 11:38 AM, Emil Ivov wrote:
>>>> Hey all,
>>>>
>>>> Justin, Eric and I have just submitted an update to the trickle ICE=20
>>>> draft, accommodating comments from Atlanta:
>>>>
>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>>
>>>> More details about the changes here:
>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendi
>>>> x-B
>>>>
>>>> Diff2:
>>>> http://www.ietf.org/rfcdiff?url1=3Ddraft-rescorla-mmusic-ice-trickle-
>>>> 01&difftype=3D--html&submit=3DGo!&url2=3Ddraft-ivov-mmusic-trickle-ice=
-00
>>>>
>>>> During the MMUSIC presentation and the mailing list discussions it=20
>>>> was also pointed out that it is important for this work to happen=20
>>>> in parallel with a SIP usage for trickle ICE. Enrico, Christer and=20
>>>> I have therefore submitted the following document:
>>>>
>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00
>>>>
>>>> As always, comments are welcome!
>>>>
>>>> Cheers,
>>>> Emil
>>>>
>>>> -------- Original Message --------
>>>> Subject: New Version Notification for=20
>>>> draft-ivov-mmusic-trickle-ice-00.txt
>>>> Date: Mon, 28 Jan 2013 11:39:21 -0800
>>>> From: internet-drafts@ietf.org
>>>> To: emcho@jitsi.org
>>>> CC: justin@uberti.name, ekr@rtfm.com
>>>>
>>>>
>>>> A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
>>>> has been successfully submitted by Emil Ivov and posted to the IETF=20
>>>> repository.
>>>>
>>>> Filename:	 draft-ivov-mmusic-trickle-ice
>>>> Revision:	 00
>>>> Title:		 Trickle ICE: Incremental Provisioning of Candidates for the
>>>> Interactive Connectivity Establishment (ICE) Protocol
>>>> Creation date:	 2013-01-28
>>>> WG ID:		 Individual Submission
>>>> Number of pages: 20
>>>> URL:
>>>> http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-0
>>>> 0.txt
>>>> Status:
>>>> http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
>>>> Htmlized:        http://tools.ietf.org/html/draft-ivov-mmusic-trickle-=
ice-00
>>>>
>>>>
>>>> Abstract:
>>>>     This document describes an extension to the Interactive Connectivi=
ty
>>>>     Establishment (ICE) protocol that allows ICE agents to send and
>>>>     receive candidates incrementally rather than exchanging complete
>>>>     lists.  With such incremental provisioning, ICE agents can begin
>>>>     connectivity checks while they are still gathering candidates and
>>>>     considerably shorten the time necessary for ICE processing to
>>>>     complete.
>>>>
>>>>     The above mechanism is also referred to as "trickle ICE".
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> The IETF Secretariat
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>> .
>>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic

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

From partha@parthasarathi.co.in  Tue Feb 12 08:12:49 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: <mmusic@ietf.org>
Date: Tue, 12 Feb 2013 21:42:40 +0530
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Subject: [MMUSIC] Offer/Answer Considerations for G.723 Annex A and G.729 Annex B draft
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Hi all,

Offer/Answer Considerations for G.723 Annex A and G.729 Annex B draft is
written as an update for RFC 4856 offer/answer considerations. The draft was
adopted in MMUSIC after a good amount of discussion in Payload WG and MMUSIC
WG. Now, All the review comments are incorporated and updated as
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-g723-g729/. The
authors hope that the draft work is complete and ready for WGLC soon. Please
let us know in case you have any comments/feedback to the draft.

Thanks
Partha/Muthu 


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e !important; float: none; ">&nbsp;</span><br style=3D"font-family: Tahoma,=
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; font-variant: normal; font-weight: normal; letter-spacing: normal; line-h=
eight: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text=
-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webki=
t-text-size-adjust: auto; -webkit-text-stroke-width: 0px; ">
<br style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; fon=
t-size: small; font-style: normal; font-variant: normal; font-weight: norma=
l; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -we=
bkit-auto; text-indent: 0px; text-transform: none; white-space: normal; wid=
ows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-str=
oke-width: 0px; ">
<span style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; f=
ont-size: small; font-style: normal; font-variant: normal; font-weight: nor=
mal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -=
webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; w=
idows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-s=
troke-width: 0px; display: inline !important; float: none; ">WebRTC
 / RTCWeb / MMusic-20130206 1953-1<span class=3D"Apple-converted-space">&nb=
sp;</span></span><br style=3D"font-family: Tahoma, Arial, sans-serif, Helve=
tica, Geneva; font-size: small; font-style: normal; font-variant: normal; f=
ont-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2=
; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-s=
pace: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto;=
 -webkit-text-stroke-width: 0px; ">
<span style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; f=
ont-size: small; font-style: normal; font-variant: normal; font-weight: nor=
mal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -=
webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; w=
idows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-s=
troke-width: 0px; display: inline !important; float: none; ">Wednesday,
 February 6, 2013 11:53 am San Francisco Time<span class=3D"Apple-converted=
-space">&nbsp;</span></span><br style=3D"font-family: Tahoma, Arial, sans-s=
erif, Helvetica, Geneva; font-size: small; font-style: normal; font-variant=
: normal; font-weight: normal; letter-spacing: normal; line-height: normal;=
 orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: no=
ne; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-ad=
just: auto; -webkit-text-stroke-width: 0px; ">
<span style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; f=
ont-size: small; font-style: normal; font-variant: normal; font-weight: nor=
mal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -=
webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; w=
idows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-s=
troke-width: 0px; display: inline !important; float: none; ">2
 Hours 52 Minutes<span class=3D"Apple-converted-space">&nbsp;</span></span>=
<br style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; fon=
t-size: small; font-style: normal; font-variant: normal; font-weight: norma=
l; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -we=
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ows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-str=
oke-width: 0px; ">
</blockquote>
</div>
<br>
</body>
</html>

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To: "mmusic@ietf.org WG" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Recording of Feb 7 interim meeting 
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Your recording is now available on the WebEx service site. Click the link b=
elow to play it:

https://cisco.webex.com/ciscosales/lsr.php?AT=3Dpb&SP=3DMC&rID=3D65887687&r=
Key=3D8c217cc13c9e0ea7

WebRTC / RTCWeb / MMusic-20130207 1348-1
Thursday, February 7, 2013 5:48 am San Francisco Time
7 Hours


--_000_C5E08FE080ACFD4DAE31E4BDBF944EB1133DB254xmbalnx02ciscoc_
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ows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-str=
oke-width: 0px; ">
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pacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px=
; ">https://cisco.webex.com/ciscosales/lsr.php?AT=3Dpb&amp;SP=3DMC&amp;rID=
=3D65887687&amp;rKey=3D8c217cc13c9e0ea7</a><span style=3D"font-family: Taho=
ma, Arial, sans-serif, Helvetica, Geneva; font-size: small; font-style: nor=
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bkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; display: inlin=
e !important; float: none; ">&nbsp;</span><br style=3D"font-family: Tahoma,=
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<br style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; fon=
t-size: small; font-style: normal; font-variant: normal; font-weight: norma=
l; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -we=
bkit-auto; text-indent: 0px; text-transform: none; white-space: normal; wid=
ows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-str=
oke-width: 0px; ">
<span style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; f=
ont-size: small; font-style: normal; font-variant: normal; font-weight: nor=
mal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -=
webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; w=
idows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-s=
troke-width: 0px; display: inline !important; float: none; ">WebRTC
 / RTCWeb / MMusic-20130207 1348-1<span class=3D"Apple-converted-space">&nb=
sp;</span></span><br style=3D"font-family: Tahoma, Arial, sans-serif, Helve=
tica, Geneva; font-size: small; font-style: normal; font-variant: normal; f=
ont-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2=
; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-s=
pace: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto;=
 -webkit-text-stroke-width: 0px; ">
<span style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; f=
ont-size: small; font-style: normal; font-variant: normal; font-weight: nor=
mal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -=
webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; w=
idows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-s=
troke-width: 0px; display: inline !important; float: none; ">Thursday,
 February 7, 2013 5:48 am San Francisco Time<span class=3D"Apple-converted-=
space">&nbsp;</span></span><br style=3D"font-family: Tahoma, Arial, sans-se=
rif, Helvetica, Geneva; font-size: small; font-style: normal; font-variant:=
 normal; font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: non=
e; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adj=
ust: auto; -webkit-text-stroke-width: 0px; ">
<span style=3D"font-family: Tahoma, Arial, sans-serif, Helvetica, Geneva; f=
ont-size: small; font-style: normal; font-variant: normal; font-weight: nor=
mal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -=
webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; w=
idows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-s=
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ord-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width=
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--_000_C5E08FE080ACFD4DAE31E4BDBF944EB1133DB254xmbalnx02ciscoc_--

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Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE update and a new SIP usage draft.]
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Hi,

This clearly needs to be clarified in ICE-bis.

Regarding which way ("just support" or "support and will use") it should 
be clarified, we have one precedent. RTP+ECN RFC says 
(http://tools.ietf.org/html/rfc6679#section-6.4):

>    The answering party includes this same attribute at the session level
>    in the SDP answer if it also has the capability and removes the
>    attribute if it does not wish to use ECN or doesn't have the
>    capability to use ECN.

This seems to imply that it should be "support and will use" to be 
consistent with current registration(s). And this would be consistent 
with the current Trickle ICE too.

Then, should we have also "ice-supported" attribute, or SIP option-tag, 
or nothing? I wonder in what case would one use only the "supported"? 
Would the semantics be "I support but will not use unless you indicate 
that you want to use", or something else?


Cheers,
Ari
(as individual)

On 2/12/13 2:04 PM, Christer Holmberg wrote:
> Hi,
>
> When a SIP dialog is established, entities normally don't indicate
> support for features that they don't want to use within that dialog.
>
> However, if the semantics of ice-options is unclear, another
> alternative is to define a SIP option-tag, which provides more
> freedom to specify the semantics for individual option-tags.
>
> Regards,
>
> Christer
>
> -----Original Message----- From: mmusic-bounces@ietf.org
> [mailto:mmusic-bounces@ietf.org] On Behalf Of Flemming Andreasen
> Sent: 12. helmikuuta 2013 3:33 To: Emil Ivov Cc: Ari Keränen; MMUSIC
> IETF WG Subject: Re: [MMUSIC] Trickle ICE comments [Re: Trickle ICE
> update and a new SIP usage draft.]
>
> Hi Emil
>
> As noted at the Interim meeting, I think it's cleaner to have
> ice-options used to strictly indicate features supported, and then
> have a separate way of indicating their use.
>
> However, in taking another look at RFC 5245, it's somewhat unclear
> what semantics were intended. Section 15.5 for example suggests
> "supported" semantics, however Section 21.1.7 suggests "used"
> semantics. There are a few more examples of each throughout the doc
> so I agree with you that 5245bis should clarify this (and depending
> on that clarification, we should use it the same way here).
>
> Thanks
>
> -- Flemming
>
>
> On 2/7/13 1:00 PM, Emil Ivov wrote:
>> Hey Flemming, Ted, all,
>>
>> On 07.02.13, 10:10, Flemming Andreasen wrote:
>>> 7) It seems the inclusion of the "trickle" ice-options is what
>>> will trigger the use of tricke-ICE. I think this is a slight
>>> overloading of the ice-options semantics inasmuch as it is
>>> intended to indicate support for a given extension, which does
>>> not necessarily imply you actually want to use that extension. I
>>> think it would be useful to have a separate way of indicating you
>>> also want to use trickle-ICE.
>> The intention of the current trickle ICE text is to say the
>> following:
>>
>> 1. When your (the offerer's) intention is to do half trickle you
>> include the "trickle" ice-options token and you are prepared to
>> have the answerer trickling. You are also prepared to have them do
>> vanilla ICE. In either case you can either include
>> end-of-candidates in your offer or you can choose to not include it
>> depending on how you intend to behave if the answerer turns out to
>> support trickle.
>>
>> 1.1. If the answer does indicate support trickle ICE by the
>> answerer, then you are both abiding by the trickle-ice spec. You
>> can both still choose to behave like vanilla ICE agents and send
>> complete sets of candidates from the start but you know that you
>> have to handle trickle semantics and you know that you have to send
>> end-of-candidates when you are done.
>>
>> 1.2 If the answerer does not support trickle, then there's no
>> "trickle" ice-options token in the answer and the offerer knows
>> that trickling is no longer possible (regardless of whether or not
>> there was end-of-candidates in the offer).
>>
>> 2. If the offerer's intention is to generate a vanilla ICE offer
>> then it does not include the "trickle" ice-options token and
>> everyone just uses vanilla ICE semantics.
>>
>> I believe that this is the part in 5245 that you had a problem
>> with and I think that this is the text you are referring to:
>>
>> First, ICE provides the a=ice-options SDP attribute.  Each
>> extension or change to ICE is associated with a token.  When an
>> agent supporting such an extension or change generates an offer or
>> an answer, it MUST include the token for that extension in this
>> attribute.  *This allows each side to know what the other side is
>> doing*.  This attribute MUST NOT be present if the agent doesn't
>> support any ICE extensions or changes.
>>
>> I am not sure exactly what the intention of this text was (maybe
>> we could clarify this in 5245bis) but that highlighted part makes
>> me think the point was to make sure everyone knows what behaviour
>> to expect rather than what features they have in their ICE stack
>> even if they are not planning on using them.
>>
>> In that sense, if I send what I intend to be a vanilla offer then
>> you can count on me not to use any trickle semantics and you would
>> make me really unhappy if you did it yourself (because I might be
>> gateway-ing your stuff to a vanilla agent). In that sense I'd
>> really like you to behave as if I were a vanilla ICE agent.
>>
>> In other words, I believe it would be legitimate for me not to
>> include the "trickle" token in ice-options.
>>
>> Does this make sense?
>>
>> Emil
>>
>>> Thanks
>>>
>>> -- Flemming
>>>
>>>
>>>> Thanks
>>>>
>>>> -- Flemming (as Individual)
>>>>
>>>>
>>>>
>>>> On 1/30/13 11:38 AM, Emil Ivov wrote:
>>>>> Hey all,
>>>>>
>>>>> Justin, Eric and I have just submitted an update to the
>>>>> trickle ICE draft, accommodating comments from Atlanta:
>>>>>
>>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>>>
>>>>> More details about the changes here:
>>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#appendi
>>>>>
>>>>>
x-B
>>>>>
>>>>> Diff2:
>>>>> http://www.ietf.org/rfcdiff?url1=draft-rescorla-mmusic-ice-trickle-
>>>>>
>>>>>
01&difftype=--html&submit=Go!&url2=draft-ivov-mmusic-trickle-ice-00
>>>>>
>>>>> During the MMUSIC presentation and the mailing list
>>>>> discussions it was also pointed out that it is important for
>>>>> this work to happen in parallel with a SIP usage for trickle
>>>>> ICE. Enrico, Christer and I have therefore submitted the
>>>>> following document:
>>>>>
>>>>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-00
>>>>>
>>>>>
>>>>>
As always, comments are welcome!
>>>>>
>>>>> Cheers, Emil
>>>>>
>>>>> -------- Original Message -------- Subject: New Version
>>>>> Notification for draft-ivov-mmusic-trickle-ice-00.txt Date:
>>>>> Mon, 28 Jan 2013 11:39:21 -0800 From:
>>>>> internet-drafts@ietf.org To: emcho@jitsi.org CC:
>>>>> justin@uberti.name, ekr@rtfm.com
>>>>>
>>>>>
>>>>> A new version of I-D, draft-ivov-mmusic-trickle-ice-00.txt
>>>>> has been successfully submitted by Emil Ivov and posted to
>>>>> the IETF repository.
>>>>>
>>>>> Filename:	 draft-ivov-mmusic-trickle-ice Revision:	 00 Title:
>>>>> Trickle ICE: Incremental Provisioning of Candidates for the
>>>>> Interactive Connectivity Establishment (ICE) Protocol
>>>>> Creation date:	 2013-01-28 WG ID:		 Individual Submission
>>>>> Number of pages: 20 URL:
>>>>> http://www.ietf.org/internet-drafts/draft-ivov-mmusic-trickle-ice-0
>>>>>
>>>>>
0.txt
>>>>> Status:
>>>>> http://datatracker.ietf.org/doc/draft-ivov-mmusic-trickle-ice
>>>>>
>>>>>
Htmlized:        http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00
>>>>>
>>>>>
>>>>> Abstract: This document describes an extension to the
>>>>> Interactive Connectivity Establishment (ICE) protocol that
>>>>> allows ICE agents to send and receive candidates
>>>>> incrementally rather than exchanging complete lists.  With
>>>>> such incremental provisioning, ICE agents can begin
>>>>> connectivity checks while they are still gathering candidates
>>>>> and considerably shorten the time necessary for ICE
>>>>> processing to complete.
>>>>>
>>>>> The above mechanism is also referred to as "trickle ICE".
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> The IETF Secretariat
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________ mmusic
>>>>> mailing list mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic .
>>>>>
>>>>
>>>>
>>>> _______________________________________________ mmusic mailing
>>>> list mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________ mmusic mailing list
> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>


From ron.even.tlv@gmail.com  Wed Feb 13 06:25:28 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>, <rtcweb@ietf.org>, "'mmusic'" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] [rtcweb] Interim meeting and way forward
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Hi Gonzalo,
Is there a meeting summary from the interim meeting that list the
already-made decisions from the interim or are you talking about ones from
IETF85?
Thanks
Roni Even

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Gonzalo Camarillo
Sent: 12 February, 2013 1:45 PM
To: rtcweb@ietf.org; mmusic
Subject: [rtcweb] Interim meeting and way forward

Folks,

first of all, I hope everybody has already managed to get back home after
the storm. Also, I want to thank Hadriel and Acme for being great hosts. The
information flow, the meeting facilities, the catering, the pullovers, and
everything else was just perfect.

With respect to the technical work that got done in Boston and, more
importantly, that remains to be done, I want to share a few thoughts with
the group.

A fact that is obvious but some of us sometimes forget about is that RTCWeb,
as any other technology, has a window of opportunity. The usefulness and
relevance of any standards we develop in this area will be extremely high
within that window, but will decrease dramatically if the window closes
(e.g., if too many new applications out there are based on non-standard
mechanisms).

While we have everything in the group to meet the relevant market's
deadlines, we should keep them in mind nevertheless. In particular, we are
going to be facing cases where *any* decision is better than *no* decision,
or where the *perfect* will be the enemy of the *good*.

In practical terms, the group should think twice before opening already-made
decisions and open them only for very good reasons. Also, the group is going
to have to make decisions based on *rough* consensus, as opposed to *wide*
consensus or unanimity. This may sound like motherhood and apple pie but I
want to stress these points anyway since they are important and need to be
kept in mind.

For Orlando, I have allocated a *significant* amount of time to RTCWeb and
MMUSIC. Those sessions will allow us to take advantage of the momentum we
currently have after the interim and to make more progress in all the
relevant specifications.

Cheers,

Gonzalo
AD for the RTCWeb and MMUSIC WGs

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


From gonzalo.camarillo@ericsson.com  Wed Feb 13 07:00:24 2013
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Cc: rtcweb@ietf.org, 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Interim meeting and way forward
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Hi Roni,

at this point, I am talking in general and strategically, not about any
decision in particular. Which particular topics would benefit from the
suggestions below and which wouldn't should be kind-of obvious to anyone
who is somewhat involved in the technical work. Also, as usual, the
technical discussions on individual topics will be moderated and
facilitated by the chairs.

Thanks,

Gonzalo

On 13/02/2013 4:22 PM, Roni Even wrote:
> Hi Gonzalo,
> Is there a meeting summary from the interim meeting that list the
> already-made decisions from the interim or are you talking about ones from
> IETF85?
> Thanks
> Roni Even
> 
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Gonzalo Camarillo
> Sent: 12 February, 2013 1:45 PM
> To: rtcweb@ietf.org; mmusic
> Subject: [rtcweb] Interim meeting and way forward
> 
> Folks,
> 
> first of all, I hope everybody has already managed to get back home after
> the storm. Also, I want to thank Hadriel and Acme for being great hosts. The
> information flow, the meeting facilities, the catering, the pullovers, and
> everything else was just perfect.
> 
> With respect to the technical work that got done in Boston and, more
> importantly, that remains to be done, I want to share a few thoughts with
> the group.
> 
> A fact that is obvious but some of us sometimes forget about is that RTCWeb,
> as any other technology, has a window of opportunity. The usefulness and
> relevance of any standards we develop in this area will be extremely high
> within that window, but will decrease dramatically if the window closes
> (e.g., if too many new applications out there are based on non-standard
> mechanisms).
> 
> While we have everything in the group to meet the relevant market's
> deadlines, we should keep them in mind nevertheless. In particular, we are
> going to be facing cases where *any* decision is better than *no* decision,
> or where the *perfect* will be the enemy of the *good*.
> 
> In practical terms, the group should think twice before opening already-made
> decisions and open them only for very good reasons. Also, the group is going
> to have to make decisions based on *rough* consensus, as opposed to *wide*
> consensus or unanimity. This may sound like motherhood and apple pie but I
> want to stress these points anyway since they are important and need to be
> kept in mind.
> 
> For Orlando, I have allocated a *significant* amount of time to RTCWeb and
> MMUSIC. Those sessions will allow us to take advantage of the momentum we
> currently have after the interim and to make more progress in all the
> relevant specifications.
> 
> Cheers,
> 
> Gonzalo
> AD for the RTCWeb and MMUSIC WGs
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From magnus.westerlund@ericsson.com  Wed Feb 13 07:07:25 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Draft on Multimedia concepts and relations for Intermim
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Hi Charles,

Lets respond to your questions.

On 2013-02-05 03:28, Charles Eckel (eckelcu) wrote:
> Hi Magnus and Bo,
> 
> Thanks for putting this together, and more importantly, for attaching
> a copy in the e-mail so I was able to access it on the plane without
> internet. I agree with most of the reasoning in the various sections;
> however, there are a couple related concepts for which I'd like more
> clarification.
> 
> 1) MediaStream Is a MediaStream restricted to a single source? I did
> not think it was, but the description on page 8 seems to imply this
> restriction. For example, could I have a single MediaStream with the
> following 4 MediaStreamTracks, where source 1 and source 2 are two
> video cameras from the same endpoint, or a mic and an camera from the
> same endpoint:
> 
> 
> Source 1 - encoding 1 
> Source 1 - encoding 2 
> Source 2 - encoding 1 
> Source 2 - encoding 2

No, MediaStream is definitely not restricted to a single Media source.
In fact a common MediaStream will contain one MediaStreamTrack with an
audio source, and another MediaStreamTrack with Video. Thus two
different media sources.

Your example above gets us into I think land where discussion is ongoing
and agreement needs to be established. From my perspective there need to
be a way of dealing with control of encoding and and a plan for
different encodings of the same media source. If that is achieved by
having multiple MediaStreamTracks of the same media source, having
multiple MediaStreams with the same set of MediaStreamTracks but
different encoding settings, or explicit encoding settings within a
MediaStram and MediaStreamTrack or having them tied to the
PeerConnection is a choice that needs to happen. They have different
impact on differnt parts, but we really need to make a choice here.

> 
> 2) PeerConnection Later, on page 18, a related restriction of a
> single encoding per PC is suggested. In the case of a layered
> encoding, this seems to imply each layer being in its own PC. Is that
> the recommendation being suggested?

For layered encoding this proposal is far from optimal. It works for
simulcast where having different RTP streams each carrying a different
encoding is fine. Thus we proposed it as a solution which appeared to
require minimal impact with some simple restrictions regarding usage of
CNAME, i.e. same CNAME for both SSRC, even if they are in different RTP
sessions due to the different PeerConnections. I assume for layering
that one like to be able to send the layers in a single RTP stream.

When it comes to layering and simulcast I think the first step is
actually to agree on requirements level. Are we supporting it from
start, are we at least having it on the future road-map to avoid
solutions which makes it difficult to add in the future?

I have gotten the impression that a number of different people have
interest in it. Maybe not to implement it immediately, but at least have
a plan for how it should be included.

Cheers

Magnus Westerlund

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


From christer.holmberg@ericsson.com  Wed Feb 13 11:48:28 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: BUNDLE: RTP only?
Thread-Index: Ac4KIxU5eB8ChmgfTeKF683MxdSffg==
Date: Wed, 13 Feb 2013 19:48:23 +0000
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Subject: [MMUSIC] BUNDLE: RTP only?
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(Sorry for sending this to both RTCWEB and MMUSIC, but it is related to bot=
h the need of RTCWEB, and the solution produced by MMUSIC)

Hi,

At the interim, during my BUNDLE presentation, I indicated that we need to =
make a decision whether the solution (whatever it will be...) shall support=
 bundling of non-RTP media. Not saying it can't be changed, but as currentl=
y documented all alternatives only support bundling of RTP media.

There might have been some comments from the mic, but as far as I know ther=
e was no decision (if there was, please point me to it).

So, my question is: do we keep the scope to RTP media?

Regards,

Christer=

From pkyzivat@alum.mit.edu  Wed Feb 13 12:06:03 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/13/13 2:48 PM, Christer Holmberg wrote:
>
> (Sorry for sending this to both RTCWEB and MMUSIC, but it is related to both the need of RTCWEB, and the solution produced by MMUSIC)
>
> Hi,
>
> At the interim, during my BUNDLE presentation, I indicated that we need to make a decision whether the solution (whatever it will be...) shall support bundling of non-RTP media. Not saying it can't be changed, but as currently documented all alternatives only support bundling of RTP media.
>
> There might have been some comments from the mic, but as far as I know there was no decision (if there was, please point me to it).
>
> So, my question is: do we keep the scope to RTP media?

*If* there is a goal to share a single 5-tuple between RTP and SCTP then 
we need a solution allows bundling them.

The key to what you can bundle together is not whether it is RTP or not. 
It is whether you can specify the mechanism for demuxing the 5-tuple and 
associated the parts with the appropriate m-line in the bundle.

	Thanks,
	Paul


From christer.holmberg@ericsson.com  Wed Feb 13 12:08:13 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE: RTP only?
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Hi,

>> (Sorry for sending this to both RTCWEB and MMUSIC, but it is related to =
both the need of RTCWEB, and the solution produced by MMUSIC)
>>
>> Hi,
>>
>> At the interim, during my BUNDLE presentation, I indicated that we need =
to make a decision whether the solution (whatever it will be...) shall supp=
ort bundling of non-RTP media. Not saying it can't be changed, but as curre=
ntly documented all alternatives only support bundling of RTP media.
>>
>> There might have been some comments from the mic, but as far as I know t=
here was no decision (if there was, please point me to it).
>>
>> So, my question is: do we keep the scope to RTP media?
>
> *If* there is a goal to share a single 5-tuple between RTP and SCTP then
> we need a solution allows bundling them.

That's another way of presenting my question, yes :)

> The key to what you can bundle together is not whether it is RTP or not.
> It is whether you can specify the mechanism for demuxing the 5-tuple and
> associated the parts with the appropriate m-line in the bundle.

Yes, and the question is whether we need to perform the exercise to look in=
to that, or if we focus on RTP.

Regards,

Christer


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [MMUSIC] BUNDLE: RTP only?
Thread-Index: AQHOCin4yriVbi3BTkiuqlV2yqiiBw==
Date: Wed, 13 Feb 2013 20:37:42 +0000
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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I think EKR expressed that he felt it was important for data channel to be =
included in the multiplexing.=20


On Feb 13, 2013, at 12:48 PM, Christer Holmberg <christer.holmberg@ericsson=
.com> wrote:

>=20
> (Sorry for sending this to both RTCWEB and MMUSIC, but it is related to b=
oth the need of RTCWEB, and the solution produced by MMUSIC)
>=20
> Hi,
>=20
> At the interim, during my BUNDLE presentation, I indicated that we need t=
o make a decision whether the solution (whatever it will be...) shall suppo=
rt bundling of non-RTP media. Not saying it can't be changed, but as curren=
tly documented all alternatives only support bundling of RTP media.
>=20
> There might have been some comments from the mic, but as far as I know th=
ere was no decision (if there was, please point me to it).
>=20
> So, my question is: do we keep the scope to RTP media?
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] [rtcweb]  BUNDLE: RTP only?
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On 2/13/2013 3:37 PM, Cullen Jennings (fluffy) wrote:
> I think EKR expressed that he felt it was important for data channel to be included in the multiplexing.

And I agree(d), strongly.  We want 1 port to minimize NAT issues and 
startup times.

>
> At the interim, during my BUNDLE presentation, I indicated that we need to make a decision whether the solution (whatever it will be...) shall support bundling of non-RTP media. Not saying it can't be changed, but as currently documented all alternatives only support bundling of RTP media.
>
> There might have been some comments from the mic, but as far as I know there was no decision (if there was, please point me to it).
>
> So, my question is: do we keep the scope to RTP media?
>

-- 
Randell Jesup
randell-ietf@jesup.org


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
Thread-Topic: [MMUSIC] Requirements for Bundle, Fumble etc.
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On Feb 8, 2013, at 8:01 PM, "Hutton, Andrew" <andrew.hutton@siemens-enterpr=
ise.com> wrote:

> Hi,
>=20
> I suggest that the requirements discussed at the interim (http://www.ietf=
.org/proceedings/interim/2013/02/05/rtcweb/slides/slides-interim-2013-rtcwe=
b-1-10.pdf) regarding bundling on to a single 5-tuple need the following ad=
ditional requirements.
>=20
> 1. Must work for a dual stack client when ICE candidates are used for add=
ress family negotiation. I think this is interesting because in this scenar=
io there will of course be different candidates for each address family and=
 we will probably need some rules to make sure this works.
>=20
> 2. Must work when a TURN server is used. This is interesting because alth=
ough a single TURN allocation will be needed for the bundled candidates it =
will I assume be necessary to do additional TURN allocations just in case t=
he peer does not support bundle.
>=20
> Regards
> Andy=20

+1 makes sense to me. Now if only I had read this before up


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From: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [MMUSIC] [rtcweb]  BUNDLE: RTP only?
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I would phrase this somewhat less strongly: less ports is better than more
ports,
so all things being equal, a multiplexing mechanism that keeps you down to
one port is better than not. That said, if we decide that plan A is dead and
we are going to plan B, I can live with having data on a separate port.

-Ekr

On Wed, Feb 13, 2013 at 12:42 PM, Randell Jesup <randell-ietf@jesup.org>wrote:

> On 2/13/2013 3:37 PM, Cullen Jennings (fluffy) wrote:
>
>> I think EKR expressed that he felt it was important for data channel to
>> be included in the multiplexing.
>>
>
> And I agree(d), strongly.  We want 1 port to minimize NAT issues and
> startup times.
>
>
>
>> At the interim, during my BUNDLE presentation, I indicated that we need
>> to make a decision whether the solution (whatever it will be...) shall
>> support bundling of non-RTP media. Not saying it can't be changed, but as
>> currently documented all alternatives only support bundling of RTP media.
>>
>> There might have been some comments from the mic, but as far as I know
>> there was no decision (if there was, please point me to it).
>>
>> So, my question is: do we keep the scope to RTP media?
>>
>>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--20cf3068447dbb90b404d5a21c41
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I would phrase this somewhat less strongly: less ports is better than more =
ports,<div>so all things being equal, a multiplexing mechanism that keeps y=
ou down to</div><div>one port is better than not. That said, if we decide t=
hat plan A is dead and</div>


<div>we are going to plan B, I can live with having data on a separate port=
.</div><div><br></div><div>-Ekr</div><div><br><div><div><div class=3D"gmail=
_quote">On Wed, Feb 13, 2013 at 12:42 PM, Randell Jesup <span dir=3D"ltr">&=
lt;<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf=
@jesup.org</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"><div>On 2/13/2013 3:37 PM, Cullen Jennings (=
fluffy) wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think EKR expressed that he felt it was important for data channel to be =
included in the multiplexing.<br>
</blockquote>
<br></div>
And I agree(d), strongly. =A0We want 1 port to minimize NAT issues and star=
tup times.<div><div><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
At the interim, during my BUNDLE presentation, I indicated that we need to =
make a decision whether the solution (whatever it will be...) shall support=
 bundling of non-RTP media. Not saying it can&#39;t be changed, but as curr=
ently documented all alternatives only support bundling of RTP media.<br>



<br>
There might have been some comments from the mic, but as far as I know ther=
e was no decision (if there was, please point me to it).<br>
<br>
So, my question is: do we keep the scope to RTP media?<br>
<br>
</blockquote>
<br></div></div><span><font color=3D"#888888">
-- <br>
Randell Jesup<br>
<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@je=
sup.org</a><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div></div></div>

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To: Eric Rescorla <ekr@rtfm.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb]  BUNDLE: RTP only?
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A question:=20

Since we have a clear way of de-multiplexing RTP and DTLS, is it necessary t=
o entangle that (solved) problem with the A/V mux issue? I realize that SDP h=
as the tendency to "deeply intermingle" just about everything, but at the RT=
P/DTLS level, we're in much better shape.

Bernard "Do we need to boil the entire ocean, or just a lake or two?" Aboba

On Feb 13, 2013, at 13:49, "Eric Rescorla" <ekr@rtfm.com> wrote:

> I would phrase this somewhat less strongly: less ports is better than more=
 ports,
> so all things being equal, a multiplexing mechanism that keeps you down to=

> one port is better than not. That said, if we decide that plan A is dead a=
nd
> we are going to plan B, I can live with having data on a separate port.
>=20
> -Ekr
>=20
> On Wed, Feb 13, 2013 at 12:42 PM, Randell Jesup <randell-ietf@jesup.org> w=
rote:
>> On 2/13/2013 3:37 PM, Cullen Jennings (fluffy) wrote:
>>> I think EKR expressed that he felt it was important for data channel to b=
e included in the multiplexing.
>>=20
>> And I agree(d), strongly.  We want 1 port to minimize NAT issues and star=
tup times.
>>=20
>>=20
>>>=20
>>> At the interim, during my BUNDLE presentation, I indicated that we need t=
o make a decision whether the solution (whatever it will be...) shall suppor=
t bundling of non-RTP media. Not saying it can't be changed, but as currentl=
y documented all alternatives only support bundling of RTP media.
>>>=20
>>> There might have been some comments from the mic, but as far as I know t=
here was no decision (if there was, please point me to it).
>>>=20
>>> So, my question is: do we keep the scope to RTP media?
>>=20
>> --=20
>> Randell Jesup
>> randell-ietf@jesup.org
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>A question:&nbsp;</div><div><br></div><div>Since we have a clear way of de-multiplexing RTP and DTLS, is it necessary to entangle that (solved) problem with the A/V mux issue? I realize that SDP has the tendency to "deeply intermingle" just about everything, but at the RTP/DTLS level, we're in much better shape.</div><div><br></div><div>Bernard "Do we need to boil the entire ocean, or just a lake or two?" Aboba</div><div><br>On Feb 13, 2013, at 13:49, "Eric Rescorla" &lt;<a href="mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br><br></div><blockquote type="cite"><div>I would phrase this somewhat less strongly: less ports is better than more ports,<div>so all things being equal, a multiplexing mechanism that keeps you down to</div><div>one port is better than not. That said, if we decide that plan A is dead and</div>


<div>we are going to plan B, I can live with having data on a separate port.</div><div><br></div><div>-Ekr</div><div><br><div><div><div class="gmail_quote">On Wed, Feb 13, 2013 at 12:42 PM, Randell Jesup <span dir="ltr">&lt;<a href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>On 2/13/2013 3:37 PM, Cullen Jennings (fluffy) wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I think EKR expressed that he felt it was important for data channel to be included in the multiplexing.<br>
</blockquote>
<br></div>
And I agree(d), strongly. &nbsp;We want 1 port to minimize NAT issues and startup times.<div><div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
At the interim, during my BUNDLE presentation, I indicated that we need to make a decision whether the solution (whatever it will be...) shall support bundling of non-RTP media. Not saying it can't be changed, but as currently documented all alternatives only support bundling of RTP media.<br>



<br>
There might have been some comments from the mic, but as far as I know there was no decision (if there was, please point me to it).<br>
<br>
So, my question is: do we keep the scope to RTP media?<br>
<br>
</blockquote>
<br></div></div><span><font color="#888888">
-- <br>
Randell Jesup<br>
<a href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a><br>
<a href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div></div></div>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>rtcweb mailing list</span><br><span><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>
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_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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From pkyzivat@alum.mit.edu  Wed Feb 13 19:15:26 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/13/13 3:08 PM, Christer Holmberg wrote:
>
> Hi,
>
>>> (Sorry for sending this to both RTCWEB and MMUSIC, but it is related to both the need of RTCWEB, and the solution produced by MMUSIC)
>>>
>>> Hi,
>>>
>>> At the interim, during my BUNDLE presentation, I indicated that we need to make a decision whether the solution (whatever it will be...) shall support bundling of non-RTP media. Not saying it can't be changed, but as currently documented all alternatives only support bundling of RTP media.
>>>
>>> There might have been some comments from the mic, but as far as I know there was no decision (if there was, please point me to it).
>>>
>>> So, my question is: do we keep the scope to RTP media?
>>
>> *If* there is a goal to share a single 5-tuple between RTP and SCTP then
>> we need a solution allows bundling them.
>
> That's another way of presenting my question, yes :)
>
>> The key to what you can bundle together is not whether it is RTP or not.
>> It is whether you can specify the mechanism for demuxing the 5-tuple and
>> associated the parts with the appropriate m-line in the bundle.
>
> Yes, and the question is whether we need to perform the exercise to look into that, or if we focus on RTP.

We know how to do the demux between DTLS payloads, ICE, and RTP/SRTP 
payloads. And if we restrict to RTP/SRTP and SCTP over DTLS, then we 
know how to demux RTP and SCTP.

There are various schemes that can be used to demux multiple RTP streams 
and associate them with individual m-lines in the bundle.

What troubles me about it is that the rules for what you can bundle 
together are dependent on the demux strategy. Describing the rules for 
this as part of the specification of the bundling mechanism is going to 
be pretty ugly. I don't see any way to make it clean and elegant.

	Thanks,
	Paul


From bernard_aboba@hotmail.com  Wed Feb 13 20:27:21 2013
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To: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic <mmusic@ietf.org>
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Subject: Re: [MMUSIC] [rtcweb] Interim meeting and way forward
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Good thoughts=2C Gonzalo.  I share your concern about the rate of progress.=
  However=2C I do not believe that the solution is to allocate more time.  =
If that were true then we would all be feeling a lot better after three day=
s of work at the interim.  IMHO=2C there is no amount of effort that can pr=
oduce movement on this set of problems without getting better clarity on th=
e context of the discussions and the process by which we get to consensus. =
=20

One observation from the interim (and other previous meetings) is that ther=
e is not a consistent understanding of the *context* for making decisions=
=2C and what the definition of "consensus" is.  IMHO=2C there is a world of=
 difference between asking for consensus on the solution to a set of proble=
ms in the context of RTCWEB=2C versus setting the context to be within the =
scope of MMUSIC {e.g. applicable to RTCWEB=2C CLUE and possible other WGs t=
oo}.   If the context is RTCWEB=2C we have a use case document to help us u=
nderstand the requirements the WebRTC protocols and APIs=3B  if it is MMUSI=
C=2C then the set of requirements is not obvious (at least by me) and we sh=
ould not be surprised if the group cannot come to consensus.=20

With respect to understanding of the nature of "consensus" in the IETF=2C P=
ete Resnick has recently published an I-D which is well worth reading:=20
http://tools.ietf.org/html/draft-resnick-on-consensus

> Date: Tue=2C 12 Feb 2013 13:45:04 +0200
> From: Gonzalo.Camarillo@ericsson.com
> To: rtcweb@ietf.org=3B mmusic@ietf.org
> Subject: [rtcweb] Interim meeting and way forward
>=20
> Folks=2C
>=20
> first of all=2C I hope everybody has already managed to get back home
> after the storm. Also=2C I want to thank Hadriel and Acme for being great
> hosts. The information flow=2C the meeting facilities=2C the catering=2C =
the
> pullovers=2C and everything else was just perfect.
>=20
> With respect to the technical work that got done in Boston and=2C more
> importantly=2C that remains to be done=2C I want to share a few thoughts
> with the group.
>=20
> A fact that is obvious but some of us sometimes forget about is that
> RTCWeb=2C as any other technology=2C has a window of opportunity. The
> usefulness and relevance of any standards we develop in this area will
> be extremely high within that window=2C but will decrease dramatically if
> the window closes (e.g.=2C if too many new applications out there are
> based on non-standard mechanisms).
>=20
> While we have everything in the group to meet the relevant market's
> deadlines=2C we should keep them in mind nevertheless. In particular=2C w=
e
> are going to be facing cases where *any* decision is better than *no*
> decision=2C or where the *perfect* will be the enemy of the *good*.
>=20
> In practical terms=2C the group should think twice before opening
> already-made decisions and open them only for very good reasons. Also=2C
> the group is going to have to make decisions based on *rough* consensus=
=2C
> as opposed to *wide* consensus or unanimity. This may sound like
> motherhood and apple pie but I want to stress these points anyway since
> they are important and need to be kept in mind.
>=20
> For Orlando=2C I have allocated a *significant* amount of time to RTCWeb
> and MMUSIC. Those sessions will allow us to take advantage of the
> momentum we currently have after the interim and to make more progress
> in all the relevant specifications.
>=20
> Cheers=2C
>=20
> Gonzalo
> AD for the RTCWeb and MMUSIC WGs
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Good thoughts=2C Gonzalo.&nbsp=
=3B I share your concern about the rate of progress.&nbsp=3B However=2C I d=
o not believe that the solution is to allocate more time.&nbsp=3B If that w=
ere true then we would all be feeling a lot better after three days of work=
 at the interim.&nbsp=3B IMHO=2C there is no amount of effort that can prod=
uce movement on this set of problems without getting better clarity on the =
context of the discussions and the process by which we get to consensus.&nb=
sp=3B <br><br>One observation from the interim (and other previous meetings=
) is that there is not a consistent understanding of the *context* for maki=
ng decisions=2C and what the definition of "consensus" is.&nbsp=3B IMHO=2C =
there is a world of difference between asking for consensus on the solution=
 to a set of problems in the context of RTCWEB=2C versus setting the contex=
t to be within the scope of MMUSIC {e.g. applicable to RTCWEB=2C CLUE and p=
ossible other WGs too}.&nbsp=3B&nbsp=3B If the context is RTCWEB=2C we have=
 a use case document to help us understand the requirements the WebRTC prot=
ocols and APIs=3B&nbsp=3B if it is MMUSIC=2C then the set of requirements i=
s not obvious (at least by me) and we should not be surprised if the group =
cannot come to consensus. <br><br>With respect to understanding of the natu=
re of "consensus" in the IETF=2C Pete Resnick has recently published an I-D=
 which is well worth reading: <br>http://tools.ietf.org/html/draft-resnick-=
on-consensus<br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B Date:=
 Tue=2C 12 Feb 2013 13:45:04 +0200<br>&gt=3B From: Gonzalo.Camarillo@ericss=
on.com<br>&gt=3B To: rtcweb@ietf.org=3B mmusic@ietf.org<br>&gt=3B Subject: =
[rtcweb] Interim meeting and way forward<br>&gt=3B <br>&gt=3B Folks=2C<br>&=
gt=3B <br>&gt=3B first of all=2C I hope everybody has already managed to ge=
t back home<br>&gt=3B after the storm. Also=2C I want to thank Hadriel and =
Acme for being great<br>&gt=3B hosts. The information flow=2C the meeting f=
acilities=2C the catering=2C the<br>&gt=3B pullovers=2C and everything else=
 was just perfect.<br>&gt=3B <br>&gt=3B With respect to the technical work =
that got done in Boston and=2C more<br>&gt=3B importantly=2C that remains t=
o be done=2C I want to share a few thoughts<br>&gt=3B with the group.<br>&g=
t=3B <br>&gt=3B A fact that is obvious but some of us sometimes forget abou=
t is that<br>&gt=3B RTCWeb=2C as any other technology=2C has a window of op=
portunity. The<br>&gt=3B usefulness and relevance of any standards we devel=
op in this area will<br>&gt=3B be extremely high within that window=2C but =
will decrease dramatically if<br>&gt=3B the window closes (e.g.=2C if too m=
any new applications out there are<br>&gt=3B based on non-standard mechanis=
ms).<br>&gt=3B <br>&gt=3B While we have everything in the group to meet the=
 relevant market's<br>&gt=3B deadlines=2C we should keep them in mind never=
theless. In particular=2C we<br>&gt=3B are going to be facing cases where *=
any* decision is better than *no*<br>&gt=3B decision=2C or where the *perfe=
ct* will be the enemy of the *good*.<br>&gt=3B <br>&gt=3B In practical term=
s=2C the group should think twice before opening<br>&gt=3B already-made dec=
isions and open them only for very good reasons. Also=2C<br>&gt=3B the grou=
p is going to have to make decisions based on *rough* consensus=2C<br>&gt=
=3B as opposed to *wide* consensus or unanimity. This may sound like<br>&gt=
=3B motherhood and apple pie but I want to stress these points anyway since=
<br>&gt=3B they are important and need to be kept in mind.<br>&gt=3B <br>&g=
t=3B For Orlando=2C I have allocated a *significant* amount of time to RTCW=
eb<br>&gt=3B and MMUSIC. Those sessions will allow us to take advantage of =
the<br>&gt=3B momentum we currently have after the interim and to make more=
 progress<br>&gt=3B in all the relevant specifications.<br>&gt=3B <br>&gt=
=3B Cheers=2C<br>&gt=3B <br>&gt=3B Gonzalo<br>&gt=3B AD for the RTCWeb and =
MMUSIC WGs<br>&gt=3B <br>&gt=3B ___________________________________________=
____<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.org<br>&gt=3B http=
s://www.ietf.org/mailman/listinfo/rtcweb<br></div> 		 	   		  </div></body>
</html>=

--_1a405a2f-3e15-4def-b782-a5651947f2be_--

From harald@alvestrand.no  Thu Feb 14 00:24:43 2013
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References: <7594FB04B1934943A5C02806D1A2204B0D8131@ESESSMB209.ericsson.se>, <511BF229.1090702@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B0D81BD@ESESSMB209.ericsson.se> <511C56CB.1080609@alum.mit.edu>
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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RTCWEB has a requirement (or rather: it has assumed that there was a 
solution) for signalling over SDP that SCTP/DTLS/UDP is sent over the 
same port pair as SRTP.

If BUNDLE is not going to be that solution, we have to start designing 
another.

On 02/14/2013 04:15 AM, Paul Kyzivat wrote:
> On 2/13/13 3:08 PM, Christer Holmberg wrote:
>>
>> Hi,
>>
>>>> (Sorry for sending this to both RTCWEB and MMUSIC, but it is 
>>>> related to both the need of RTCWEB, and the solution produced by 
>>>> MMUSIC)
>>>>
>>>> Hi,
>>>>
>>>> At the interim, during my BUNDLE presentation, I indicated that we 
>>>> need to make a decision whether the solution (whatever it will 
>>>> be...) shall support bundling of non-RTP media. Not saying it can't 
>>>> be changed, but as currently documented all alternatives only 
>>>> support bundling of RTP media.
>>>>
>>>> There might have been some comments from the mic, but as far as I 
>>>> know there was no decision (if there was, please point me to it).
>>>>
>>>> So, my question is: do we keep the scope to RTP media?
>>>
>>> *If* there is a goal to share a single 5-tuple between RTP and SCTP 
>>> then
>>> we need a solution allows bundling them.
>>
>> That's another way of presenting my question, yes :)
>>
>>> The key to what you can bundle together is not whether it is RTP or 
>>> not.
>>> It is whether you can specify the mechanism for demuxing the 5-tuple 
>>> and
>>> associated the parts with the appropriate m-line in the bundle.
>>
>> Yes, and the question is whether we need to perform the exercise to 
>> look into that, or if we focus on RTP.
>
> We know how to do the demux between DTLS payloads, ICE, and RTP/SRTP 
> payloads. And if we restrict to RTP/SRTP and SCTP over DTLS, then we 
> know how to demux RTP and SCTP.
>
> There are various schemes that can be used to demux multiple RTP 
> streams and associate them with individual m-lines in the bundle.
>
> What troubles me about it is that the rules for what you can bundle 
> together are dependent on the demux strategy. Describing the rules for 
> this as part of the specification of the bundling mechanism is going 
> to be pretty ugly. I don't see any way to make it clean and elegant.
>
>     Thanks,
>     Paul
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From ari.keranen@ericsson.com  Thu Feb 14 05:49:28 2013
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From: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
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Hi all,

Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for 
background, see [1] and [2]), as discussed at the Atlanta meeting, will 
be one of the important changes in the new version of ICE.

Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be 
configurable, SHOULD have a default of 500 ms, and MUST NOT be 
configurable to be less than 500 ms."

This results in very slow pace of starting connectivity checks if ICE is 
used with non-RTP traffic. With RTP, the Ta is calculated based on the 
expected bandwidth used by the RTP stream and can be down to 20 ms. 
Therefore, many current implementations apparently ignore the "MUST NOT" 
above and set Ta to something lower.

The quite conservative value of 500 ms was originally chosen because of 
the fear of congesting the links with the connectivity check messages. 
This is obviously still a valid concern, but would suggesting something 
lower still make sense? How low would it be reasonably safe to go? What 
is currently used in practice? Can we come up with a formula similar to 
RTP (perhaps based on expected bandwidth usage)?


Cheers,
Ari

[1] http://tools.ietf.org/html/rfc5245#appendix-B.1
[2] http://tools.ietf.org/html/rfc5245#section-16.2

From magnus.westerlund@ericsson.com  Thu Feb 14 06:00:35 2013
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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On 2013-02-05 16:25, Adam Roach wrote:
> On 2/4/13 02:45, Justin Uberti wrote:
>> Sorry for the short lead time, but here's a quick read for the plane.
> 
> I addressed several of the assertions made in this draft in my previous
> response (e.g. the need for glare resolution regardless of which
> solution we select). Some additional thoughts:
> 
>>    o  Adding or removing a source requires sending a full list of all
>>       sources, which doesn't scale well to 10s or 100s of sources,
>>       especially when they are added or removed frequently.
> 
> Realistically, this isn't how multiparty conferences are handled. It's
> unrealistic to think that my DSL connection is going to be sucking down
> hundreds of video streams just because I'm in a video conference with
> hundreds of people.
> 
> In practice, the way existing conferencing systems handle this is to
> send a small, manageable number of streams to each participant, changing
> them out as necessary to have a reasonable user experience (either under
> the control of the user, its own algorithms, or a combination of the two).
> 
> Which is to say: even with hundreds of source streams being fed into a
> conference system, you're really looking at a realistic upper bound on
> m= sections of a dozen or two, simply because you can't reasonably
> expect recipients to receive more than that many streams simultaneously.
> 

Hi,

I definitely want to comment on this from an RTP usage perspective that
might be very relevant to remember. Even with a centralized conference
solution there is two very distinct solutions in RTP how to transmit a
session where you have total of 100 participants and you receive 5 video
streams.

A) Central point with conceptual sources
In this case the central entity creates conceptual RTP streams that
carries all the media intended for a specific usage, for example main
video, thumbnail one, thumbnail two, thumbnail three, thumbnail four.

B) Central point doing source projection
In this case there exist 100 potential RTP streams and SSRC from the
central point down to the receiver. However, at any point no more than
five will actually have RTP packets. But as activity changes and which
is most relevant for the receiver the SSRCs in use will switch.

For a solution where each SSRC requires its own m= line A) is definitely
preferable as it will not require renegotiating the properties for these
five conceptual streams. However, B will require a large number of m=
lines and dynamic updates as participants comes and go.

IF one had a single m= line per RTP session and media type, the amount
of updates/renegotiation depends on the amount of meta data about
streams that you have put in the SDP. They solutions A) and B) can be
identical from an SDP perspective.

The above two cases I am quite certain we did discuss in the CLUE
interim meeting in Stockholm when discussing topologies and also the
relation ship between RTP streams (SSRC) and their origin, i.e. media
source/capture.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
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From dwing@cisco.com  Thu Feb 14 07:21:00 2013
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From: "Dan Wing" <dwing@cisco.com>
To: "=?iso-8859-1?Q?'Ari_Ker=E4nen'?=" <ari.keranen@ericsson.com>, "'mmusic'" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On =
Behalf
> Of Ari Ker=E4nen
> Sent: Thursday, February 14, 2013 5:49 AM
> To: mmusic
> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
>=20
> Hi all,
>=20
> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for
> background, see [1] and [2]), as discussed at the Atlanta meeting, =
will
> be one of the important changes in the new version of ICE.
>=20
> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
> configurable, SHOULD have a default of 500 ms, and MUST NOT be
> configurable to be less than 500 ms."
>=20
> This results in very slow pace of starting connectivity checks if ICE =
is
> used with non-RTP traffic. With RTP, the Ta is calculated based on the
> expected bandwidth used by the RTP stream and can be down to 20 ms.
> Therefore, many current implementations apparently ignore the "MUST =
NOT"
> above and set Ta to something lower.

That would explain failures claimed to justify the desire for running
over one port. =20

The 20ms threshold for the initial connectivity checks is because some=20
NATs could not create new mappings quickly.  This problems appears to
persist today, considering that both Chrome and Windows 8 limit their=20
number of outstanding DNS queries in an apparent attempt to work around
that very same NAT UDP mapping behavior.

If trying to create NAT mappings with 0ms pacing, it will certainly=20
cause NAT mapping failures and force retries.

> The quite conservative value of 500 ms was originally chosen because =
of
> the fear of congesting the links with the connectivity check messages.
> This is obviously still a valid concern, but would suggesting =
something
> lower still make sense? How low would it be reasonably safe to go? =
What
> is currently used in practice? Can we come up with a formula similar =
to
> RTP (perhaps based on expected bandwidth usage)?

A good question for the transport area, which is the origination of the
suggestion to send ICE connectivity checks paced to not exceed the=20
bandwidth of the subsequent media.  Perhaps the arguments that TCP=20
IW=3D10 is not harmful could be brought to the party.

-d


>=20
> Cheers,
> Ari
>=20
> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
> [2] http://tools.ietf.org/html/rfc5245#section-16.2
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From ari.keranen@ericsson.com  Thu Feb 14 07:32:58 2013
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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On 2/14/13 5:20 PM, Dan Wing wrote:
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>> Of Ari Keränen
>> Sent: Thursday, February 14, 2013 5:49 AM
>> To: mmusic
>> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
>>
>> Hi all,
>>
>> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for
>> background, see [1] and [2]), as discussed at the Atlanta meeting, will
>> be one of the important changes in the new version of ICE.
>>
>> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
>> configurable, SHOULD have a default of 500 ms, and MUST NOT be
>> configurable to be less than 500 ms."
>>
>> This results in very slow pace of starting connectivity checks if ICE is
>> used with non-RTP traffic. With RTP, the Ta is calculated based on the
>> expected bandwidth used by the RTP stream and can be down to 20 ms.
>> Therefore, many current implementations apparently ignore the "MUST NOT"
>> above and set Ta to something lower.
>
> That would explain failures claimed to justify the desire for running
> over one port.

Interesting point. Does anyone know what kind of pacing was actually 
used when these failures were observed?

> The 20ms threshold for the initial connectivity checks is because some
> NATs could not create new mappings quickly.  This problems appears to
> persist today, considering that both Chrome and Windows 8 limit their
> number of outstanding DNS queries in an apparent attempt to work around
> that very same NAT UDP mapping behavior.
>
> If trying to create NAT mappings with 0ms pacing, it will certainly
> cause NAT mapping failures and force retries.

Yes, definitely. So we shouldn't go below 20ms with non-RTP either. 
Perhaps need to add some more justification/warning text about this too.

>> The quite conservative value of 500 ms was originally chosen because of
>> the fear of congesting the links with the connectivity check messages.
>> This is obviously still a valid concern, but would suggesting something
>> lower still make sense? How low would it be reasonably safe to go? What
>> is currently used in practice? Can we come up with a formula similar to
>> RTP (perhaps based on expected bandwidth usage)?
>
> A good question for the transport area, which is the origination of the
> suggestion to send ICE connectivity checks paced to not exceed the
> bandwidth of the subsequent media.  Perhaps the arguments that TCP
> IW=10 is not harmful could be brought to the party.

FYI, I forwarded the original mail to the Transport ADs too.


Cheers,
Ari

>>
>> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
>> [2] http://tools.ietf.org/html/rfc5245#section-16.2
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>


From emil@sip-communicator.org  Thu Feb 14 07:37:03 2013
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Date: Thu, 14 Feb 2013 16:36:56 +0100
From: Emil Ivov <emcho@jitsi.org>
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Cc: 'mmusic' <mmusic@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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Any idea why there had to be different pacing for RTP and non-RTP flows
in the first place?

Emil

On 14.02.13, 16:32, Ari Ker=E4nen wrote:
> On 2/14/13 5:20 PM, Dan Wing wrote:
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Beh=
alf
>>> Of Ari Ker=E4nen
>>> Sent: Thursday, February 14, 2013 5:49 AM
>>> To: mmusic
>>> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
>>>
>>> Hi all,
>>>
>>> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for
>>> background, see [1] and [2]), as discussed at the Atlanta meeting, wi=
ll
>>> be one of the important changes in the new version of ICE.
>>>
>>> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
>>> configurable, SHOULD have a default of 500 ms, and MUST NOT be
>>> configurable to be less than 500 ms."
>>>
>>> This results in very slow pace of starting connectivity checks if ICE=
 is
>>> used with non-RTP traffic. With RTP, the Ta is calculated based on th=
e
>>> expected bandwidth used by the RTP stream and can be down to 20 ms.
>>> Therefore, many current implementations apparently ignore the "MUST N=
OT"
>>> above and set Ta to something lower.
>>
>> That would explain failures claimed to justify the desire for running
>> over one port.
>=20
> Interesting point. Does anyone know what kind of pacing was actually=20
> used when these failures were observed?
>=20
>> The 20ms threshold for the initial connectivity checks is because some=

>> NATs could not create new mappings quickly.  This problems appears to
>> persist today, considering that both Chrome and Windows 8 limit their
>> number of outstanding DNS queries in an apparent attempt to work aroun=
d
>> that very same NAT UDP mapping behavior.
>>
>> If trying to create NAT mappings with 0ms pacing, it will certainly
>> cause NAT mapping failures and force retries.
>=20
> Yes, definitely. So we shouldn't go below 20ms with non-RTP either.=20
> Perhaps need to add some more justification/warning text about this too=
=2E
>=20
>>> The quite conservative value of 500 ms was originally chosen because =
of
>>> the fear of congesting the links with the connectivity check messages=
=2E
>>> This is obviously still a valid concern, but would suggesting somethi=
ng
>>> lower still make sense? How low would it be reasonably safe to go? Wh=
at
>>> is currently used in practice? Can we come up with a formula similar =
to
>>> RTP (perhaps based on expected bandwidth usage)?
>>
>> A good question for the transport area, which is the origination of th=
e
>> suggestion to send ICE connectivity checks paced to not exceed the
>> bandwidth of the subsequent media.  Perhaps the arguments that TCP
>> IW=3D10 is not harmful could be brought to the party.
>=20
> FYI, I forwarded the original mail to the Transport ADs too.
>=20
>=20
> Cheers,
> Ari
>=20
>>>
>>> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
>>> [2] http://tools.ietf.org/html/rfc5245#section-16.2
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20

--=20
https://jitsi.org


From ari.keranen@ericsson.com  Thu Feb 14 07:44:35 2013
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Cc: 'mmusic' <mmusic@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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On 2/14/13 5:36 PM, Emil Ivov wrote:
> Any idea why there had to be different pacing for RTP and non-RTP flows
> in the first place?

Because with RTP you know the to-be-used bandwidth and can match checks 
with that (http://tools.ietf.org/html/rfc5245#section-16.1):
>    These formulas are aimed at causing STUN transactions to be paced at
>    the same rate as media.  This ensures that ICE will work properly
>    under the same network conditions needed to support the media as
>    well.


Cheers,
Ari

From magnus.westerlund@ericsson.com  Thu Feb 14 07:51:20 2013
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To: Emil Ivov <emcho@jitsi.org>
References: <511CEB64.6070306@ericsson.com> <05a401ce0ac6$db547230$91fd5690$@cisco.com> <511D03A7.1060008@ericsson.com> <511D0498.3030909@jitsi.org>
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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On 2013-02-14 16:36, Emil Ivov wrote:
> Any idea why there had to be different pacing for RTP and non-RTP flows
> in the first place?

I think our arguments where that in the context of a non-congestion
controlled voice application it would send with 20 ms interval anyway.
For other transports and protocols it was not obvious for what intended
environment it would be used. For example a low power network
application that sends beyond 30 kbps could be problematic. Thus we took
one of our conservative values from the transport books, i.e. 500 ms.

I would point out that this is a difficult transport issue. Compared to
TCP IW10, that burst occurs first after the initial handshake when one
has actually established that there is a responder.

With ICE there is a clear issue with an attacker bloating a offer/answer
with candidates that recreates the voice hammer attack using ICE's STUN
checks towards a target address or domain.

Also the STUN checks will have different source address pairs which in
it self could be an issue for attack prevention firewalls etc.

Thus I would be careful with speeding up Ta for RTP to much, and I think
someone needs to fine empiric data that it would work.

For the the 500 ms that can likely be significantly lowered without any
issues. But some considerations around the wider usage needs to be taken
into account.

Cheers

Magnus
One of the ADs responsible for the pacing in ICE.

> 
> Emil
> 
> On 14.02.13, 16:32, Ari Keränen wrote:
>> On 2/14/13 5:20 PM, Dan Wing wrote:
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>>>> Of Ari Keränen
>>>> Sent: Thursday, February 14, 2013 5:49 AM
>>>> To: mmusic
>>>> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
>>>>
>>>> Hi all,
>>>>
>>>> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for
>>>> background, see [1] and [2]), as discussed at the Atlanta meeting, will
>>>> be one of the important changes in the new version of ICE.
>>>>
>>>> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
>>>> configurable, SHOULD have a default of 500 ms, and MUST NOT be
>>>> configurable to be less than 500 ms."
>>>>
>>>> This results in very slow pace of starting connectivity checks if ICE is
>>>> used with non-RTP traffic. With RTP, the Ta is calculated based on the
>>>> expected bandwidth used by the RTP stream and can be down to 20 ms.
>>>> Therefore, many current implementations apparently ignore the "MUST NOT"
>>>> above and set Ta to something lower.
>>>
>>> That would explain failures claimed to justify the desire for running
>>> over one port.
>>
>> Interesting point. Does anyone know what kind of pacing was actually 
>> used when these failures were observed?
>>
>>> The 20ms threshold for the initial connectivity checks is because some
>>> NATs could not create new mappings quickly.  This problems appears to
>>> persist today, considering that both Chrome and Windows 8 limit their
>>> number of outstanding DNS queries in an apparent attempt to work around
>>> that very same NAT UDP mapping behavior.
>>>
>>> If trying to create NAT mappings with 0ms pacing, it will certainly
>>> cause NAT mapping failures and force retries.
>>
>> Yes, definitely. So we shouldn't go below 20ms with non-RTP either. 
>> Perhaps need to add some more justification/warning text about this too.
>>
>>>> The quite conservative value of 500 ms was originally chosen because of
>>>> the fear of congesting the links with the connectivity check messages.
>>>> This is obviously still a valid concern, but would suggesting something
>>>> lower still make sense? How low would it be reasonably safe to go? What
>>>> is currently used in practice? Can we come up with a formula similar to
>>>> RTP (perhaps based on expected bandwidth usage)?
>>>
>>> A good question for the transport area, which is the origination of the
>>> suggestion to send ICE connectivity checks paced to not exceed the
>>> bandwidth of the subsequent media.  Perhaps the arguments that TCP
>>> IW=10 is not harmful could be brought to the party.
>>
>> FYI, I forwarded the original mail to the Transport ADs too.
>>
>>
>> Cheers,
>> Ari
>>
>>>>
>>>> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
>>>> [2] http://tools.ietf.org/html/rfc5245#section-16.2
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
> 


-- 

Magnus Westerlund

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


From ari.keranen@ericsson.com  Thu Feb 14 08:12:40 2013
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
References: <511CEB64.6070306@ericsson.com> <05a401ce0ac6$db547230$91fd5690$@cisco.com> <511D03A7.1060008@ericsson.com> <511D0498.3030909@jitsi.org> <511D07F1.3070808@ericsson.com>
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Cc: 'mmusic' <mmusic@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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Hi Magnus,

Thanks for clarifying the background on this! Some comments and 
questions inline.

On 2/14/13 5:51 PM, Magnus Westerlund wrote:
> On 2013-02-14 16:36, Emil Ivov wrote:
>> Any idea why there had to be different pacing for RTP and non-RTP flows
>> in the first place?
>
> I think our arguments where that in the context of a non-congestion
> controlled voice application it would send with 20 ms interval anyway.
> For other transports and protocols it was not obvious for what intended
> environment it would be used. For example a low power network
> application that sends beyond 30 kbps could be problematic. Thus we took
> one of our conservative values from the transport books, i.e. 500 ms.
>
> I would point out that this is a difficult transport issue. Compared to
> TCP IW10, that burst occurs first after the initial handshake when one
> has actually established that there is a responder.
>
> With ICE there is a clear issue with an attacker bloating a offer/answer
> with candidates that recreates the voice hammer attack using ICE's STUN
> checks towards a target address or domain.
>
> Also the STUN checks will have different source address pairs which in
> it self could be an issue for attack prevention firewalls etc.
>
> Thus I would be careful with speeding up Ta for RTP to much, and I think
> someone needs to fine empiric data that it would work.

I'd guess the 20 ms interval is quite OK for the RTP since, as Dan 
noted, you can't go much faster without overwhelming the NATs.

> For the the 500 ms that can likely be significantly lowered without any
> issues. But some considerations around the wider usage needs to be taken
> into account.

Would you have any recommendation for a good, less conservative value? 
If the bandwidth of the non-RTP stream is known, I guess that could be 
used in a similar way as with RTP to calculate Ta (with lower bound of 
20ms). For cases where the bw is not known, we could perhaps 
RFC2119-recommend some fairly-conservative value (say, around 100ms?) 
and "MUST NOT" for below 20ms?


Cheers,
Ari
(as individual)

> Cheers
>
> Magnus
> One of the ADs responsible for the pacing in ICE.
>
>>
>> Emil
>>
>> On 14.02.13, 16:32, Ari Keränen wrote:
>>> On 2/14/13 5:20 PM, Dan Wing wrote:
>>>>> -----Original Message-----
>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>>>>> Of Ari Keränen
>>>>> Sent: Thursday, February 14, 2013 5:49 AM
>>>>> To: mmusic
>>>>> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
>>>>>
>>>>> Hi all,
>>>>>
>>>>> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for
>>>>> background, see [1] and [2]), as discussed at the Atlanta meeting, will
>>>>> be one of the important changes in the new version of ICE.
>>>>>
>>>>> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
>>>>> configurable, SHOULD have a default of 500 ms, and MUST NOT be
>>>>> configurable to be less than 500 ms."
>>>>>
>>>>> This results in very slow pace of starting connectivity checks if ICE is
>>>>> used with non-RTP traffic. With RTP, the Ta is calculated based on the
>>>>> expected bandwidth used by the RTP stream and can be down to 20 ms.
>>>>> Therefore, many current implementations apparently ignore the "MUST NOT"
>>>>> above and set Ta to something lower.
>>>>
>>>> That would explain failures claimed to justify the desire for running
>>>> over one port.
>>>
>>> Interesting point. Does anyone know what kind of pacing was actually
>>> used when these failures were observed?
>>>
>>>> The 20ms threshold for the initial connectivity checks is because some
>>>> NATs could not create new mappings quickly.  This problems appears to
>>>> persist today, considering that both Chrome and Windows 8 limit their
>>>> number of outstanding DNS queries in an apparent attempt to work around
>>>> that very same NAT UDP mapping behavior.
>>>>
>>>> If trying to create NAT mappings with 0ms pacing, it will certainly
>>>> cause NAT mapping failures and force retries.
>>>
>>> Yes, definitely. So we shouldn't go below 20ms with non-RTP either.
>>> Perhaps need to add some more justification/warning text about this too.
>>>
>>>>> The quite conservative value of 500 ms was originally chosen because of
>>>>> the fear of congesting the links with the connectivity check messages.
>>>>> This is obviously still a valid concern, but would suggesting something
>>>>> lower still make sense? How low would it be reasonably safe to go? What
>>>>> is currently used in practice? Can we come up with a formula similar to
>>>>> RTP (perhaps based on expected bandwidth usage)?
>>>>
>>>> A good question for the transport area, which is the origination of the
>>>> suggestion to send ICE connectivity checks paced to not exceed the
>>>> bandwidth of the subsequent media.  Perhaps the arguments that TCP
>>>> IW=10 is not harmful could be brought to the party.
>>>
>>> FYI, I forwarded the original mail to the Transport ADs too.
>>>
>>>
>>> Cheers,
>>> Ari
>>>
>>>>>
>>>>> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
>>>>> [2] http://tools.ietf.org/html/rfc5245#section-16.2
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>
>


From emil@sip-communicator.org  Thu Feb 14 08:13:04 2013
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Date: Thu, 14 Feb 2013 17:12:48 +0100
From: Emil Ivov <emcho@jitsi.org>
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, 'mmusic' <mmusic@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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Thanks for explaining Magnus,

some more inline

On 14.02.13, 16:51, Magnus Westerlund wrote:
> On 2013-02-14 16:36, Emil Ivov wrote:
>> Any idea why there had to be different pacing for RTP and non-RTP flow=
s
>> in the first place?
>=20
> I think our arguments where that in the context of a non-congestion
> controlled voice application it would send with 20 ms interval anyway.
> For other transports and protocols it was not obvious for what intended=

> environment it would be used. For example a low power network
> application that sends beyond 30 kbps could be problematic. Thus we too=
k
> one of our conservative values from the transport books, i.e. 500 ms.
>=20
> I would point out that this is a difficult transport issue. Compared to=

> TCP IW10, that burst occurs first after the initial handshake when one
> has actually established that there is a responder.
>=20
> With ICE there is a clear issue with an attacker bloating a offer/answe=
r
> with candidates that recreates the voice hammer attack using ICE's STUN=

> checks towards a target address or domain.

OK, but we already have 5.7.3 and the 100 check limitation, right?

I suppose the thing that is really problematic here is the strength of
the recommendation. Other sections in the spec seem to often refer to
often leave the choice to the application advising use of a rate that
matches the traffic that's about to be exchanged.

It seems that a similar recommendation would make sense here. WDYT?

Emil



>=20
> Also the STUN checks will have different source address pairs which in
> it self could be an issue for attack prevention firewalls etc.
>=20
> Thus I would be careful with speeding up Ta for RTP to much, and I thin=
k
> someone needs to fine empiric data that it would work.
>=20
> For the the 500 ms that can likely be significantly lowered without any=

> issues. But some considerations around the wider usage needs to be take=
n
> into account.
>=20
> Cheers
>=20
> Magnus
> One of the ADs responsible for the pacing in ICE.
>=20
>>
>> Emil
>>
>> On 14.02.13, 16:32, Ari Ker=E4nen wrote:
>>> On 2/14/13 5:20 PM, Dan Wing wrote:
>>>>> -----Original Message-----
>>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On B=
ehalf
>>>>> Of Ari Ker=E4nen
>>>>> Sent: Thursday, February 14, 2013 5:49 AM
>>>>> To: mmusic
>>>>> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
>>>>>
>>>>> Hi all,
>>>>>
>>>>> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for
>>>>> background, see [1] and [2]), as discussed at the Atlanta meeting, =
will
>>>>> be one of the important changes in the new version of ICE.
>>>>>
>>>>> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
>>>>> configurable, SHOULD have a default of 500 ms, and MUST NOT be
>>>>> configurable to be less than 500 ms."
>>>>>
>>>>> This results in very slow pace of starting connectivity checks if I=
CE is
>>>>> used with non-RTP traffic. With RTP, the Ta is calculated based on =
the
>>>>> expected bandwidth used by the RTP stream and can be down to 20 ms.=

>>>>> Therefore, many current implementations apparently ignore the "MUST=
 NOT"
>>>>> above and set Ta to something lower.
>>>>
>>>> That would explain failures claimed to justify the desire for runnin=
g
>>>> over one port.
>>>
>>> Interesting point. Does anyone know what kind of pacing was actually =

>>> used when these failures were observed?
>>>
>>>> The 20ms threshold for the initial connectivity checks is because so=
me
>>>> NATs could not create new mappings quickly.  This problems appears t=
o
>>>> persist today, considering that both Chrome and Windows 8 limit thei=
r
>>>> number of outstanding DNS queries in an apparent attempt to work aro=
und
>>>> that very same NAT UDP mapping behavior.
>>>>
>>>> If trying to create NAT mappings with 0ms pacing, it will certainly
>>>> cause NAT mapping failures and force retries.
>>>
>>> Yes, definitely. So we shouldn't go below 20ms with non-RTP either.=20
>>> Perhaps need to add some more justification/warning text about this t=
oo.
>>>
>>>>> The quite conservative value of 500 ms was originally chosen becaus=
e of
>>>>> the fear of congesting the links with the connectivity check messag=
es.
>>>>> This is obviously still a valid concern, but would suggesting somet=
hing
>>>>> lower still make sense? How low would it be reasonably safe to go? =
What
>>>>> is currently used in practice? Can we come up with a formula simila=
r to
>>>>> RTP (perhaps based on expected bandwidth usage)?
>>>>
>>>> A good question for the transport area, which is the origination of =
the
>>>> suggestion to send ICE connectivity checks paced to not exceed the
>>>> bandwidth of the subsequent media.  Perhaps the arguments that TCP
>>>> IW=3D10 is not harmful could be brought to the party.
>>>
>>> FYI, I forwarded the original mail to the Transport ADs too.
>>>
>>>
>>> Cheers,
>>> Ari
>>>
>>>>>
>>>>> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
>>>>> [2] http://tools.ietf.org/html/rfc5245#section-16.2
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>=20
>=20

--=20
https://jitsi.org


From pkyzivat@alum.mit.edu  Thu Feb 14 10:14:34 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Harald,

On 2/14/13 3:24 AM, Harald Alvestrand wrote:
> RTCWEB has a requirement (or rather: it has assumed that there was a
> solution) for signalling over SDP that SCTP/DTLS/UDP is sent over the
> same port pair as SRTP.
>
> If BUNDLE is not going to be that solution, we have to start designing
> another.

Yes, I know. But nobody has made a proposal that actually encompasses 
that desire. I think we have turned over a rock and found a large can of 
worms.

There is a deep connection here between the mechanisms used to demux 
packets into separate logical "flows" and the SDP notations used to 
describe those flows.

SDP was designed to describe flows that are demuxed by 3-tuple - 
incoming transport type, address, and port. SDP O/A extended that to 
describe symmetric pairs of such flows. Restriction to symmetry of 
send/recv address/port potentially extends that to demux by 5-tuple. But 
we are already on fuzzy ground of SDP expressiveness by that time.

Now we want to extend that expressiveness to further levels and criteria 
for demuxing. The bundle *syntax* lets us describe the things we want to 
demux, but without any explicit indication of what demux algorithm to 
apply. The existing bundle proposal applies some restrictions on what 
can be bundled - presumably to limit it what can be demuxed - but 
doesn't explicitly say how to demux. (There are restrictions on use of 
payload types across m-lines that suggest an expectation that demux is 
to be done based on payload type.) The restrictions it imposes don't 
permit bundling RTP and SCTP.

Unless we want to devolve into a Do What I Mean situation, we need to 
find a way to explicitly say what we mean.

A good start would be to first be *specific* (and exhaustive) about what 
we would like to multiplex (without regard to how we will notate that), 
and how we propose to mux/demux it.

There is info out there on how to mux/demux UDP/TLS/RTP/SAVP and 
DTLS/SCTP over the same DTLS connection. (Though it isn't immediately 
obvious where that is specified.)

Presumably the goal for RTCWEB is to demux distinct MediaStreamTracks 
from a single UDP/TLS/RTP/SAVP session. I doubt we want to do that by 
payload type. Based on discussion last week, this may sometimes be done 
by SSRC, but I think we had cases when multiple SSRCs might map to the 
same track, and the mappings may not always be known in advance. CLUE 
has further needs, where an SSRC may move about from one 
"capture-encoding" to another. (A clue capture-encoding is, I think, 
more or less equivalent to an RTCWEB track.)

So I think we need a serious discussion of mux/demux 
criteria/algorithms. This seems to be a prerequisite to a solution to 
the bundling problem in MMUSIC. But MMUSIC may not be the proper wg for 
this discussion. I don't know which one is.

	Thanks,
	Paul

> On 02/14/2013 04:15 AM, Paul Kyzivat wrote:
>> On 2/13/13 3:08 PM, Christer Holmberg wrote:
>>>
>>> Hi,
>>>
>>>>> (Sorry for sending this to both RTCWEB and MMUSIC, but it is
>>>>> related to both the need of RTCWEB, and the solution produced by
>>>>> MMUSIC)
>>>>>
>>>>> Hi,
>>>>>
>>>>> At the interim, during my BUNDLE presentation, I indicated that we
>>>>> need to make a decision whether the solution (whatever it will
>>>>> be...) shall support bundling of non-RTP media. Not saying it can't
>>>>> be changed, but as currently documented all alternatives only
>>>>> support bundling of RTP media.
>>>>>
>>>>> There might have been some comments from the mic, but as far as I
>>>>> know there was no decision (if there was, please point me to it).
>>>>>
>>>>> So, my question is: do we keep the scope to RTP media?
>>>>
>>>> *If* there is a goal to share a single 5-tuple between RTP and SCTP
>>>> then
>>>> we need a solution allows bundling them.
>>>
>>> That's another way of presenting my question, yes :)
>>>
>>>> The key to what you can bundle together is not whether it is RTP or
>>>> not.
>>>> It is whether you can specify the mechanism for demuxing the 5-tuple
>>>> and
>>>> associated the parts with the appropriate m-line in the bundle.
>>>
>>> Yes, and the question is whether we need to perform the exercise to
>>> look into that, or if we focus on RTP.
>>
>> We know how to do the demux between DTLS payloads, ICE, and RTP/SRTP
>> payloads. And if we restrict to RTP/SRTP and SCTP over DTLS, then we
>> know how to demux RTP and SCTP.
>>
>> There are various schemes that can be used to demux multiple RTP
>> streams and associate them with individual m-lines in the bundle.
>>
>> What troubles me about it is that the rules for what you can bundle
>> together are dependent on the demux strategy. Describing the rules for
>> this as part of the specification of the bundling mechanism is going
>> to be pretty ugly. I don't see any way to make it clean and elegant.
>>
>>     Thanks,
>>     Paul
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From martin.thomson@gmail.com  Thu Feb 14 10:34:10 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 13 February 2013 19:15, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> What troubles me about it is that the rules for what you can bundle together
> are dependent on the demux strategy. Describing the rules for this as part
> of the specification of the bundling mechanism is going to be pretty ugly. I
> don't see any way to make it clean and elegant.


I didn't see this as a deal-breaker.  You could think of it as: "if
you can demux these things, then it is valid to multiplex them".
Don't bother saying anything more than that.

As Bernard observes, it's the RTP demux that is hard.  Solve that and
let the rest sort themselves out.  Sure, you might not be able to
demux protocol MEAT and protocol CHEESE, but that's an easy solve -
don't try to multiplex them.

From pkyzivat@alum.mit.edu  Thu Feb 14 10:55:30 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 1:34 PM, Martin Thomson wrote:
> On 13 February 2013 19:15, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> What troubles me about it is that the rules for what you can bundle together
>> are dependent on the demux strategy. Describing the rules for this as part
>> of the specification of the bundling mechanism is going to be pretty ugly. I
>> don't see any way to make it clean and elegant.
>
>
> I didn't see this as a deal-breaker.  You could think of it as: "if
> you can demux these things, then it is valid to multiplex them".
> Don't bother saying anything more than that.
>
> As Bernard observes, it's the RTP demux that is hard.  Solve that and
> let the rest sort themselves out.  Sure, you might not be able to
> demux protocol MEAT and protocol CHEESE, but that's an easy solve -
> don't try to multiplex them.

I agree we don't need an algorithm to demux anything from anything.

But we do need a specified way to look at the SDP and know what the 
mux/demux algorithm is. The rule for bundling should make that a 
prerequisite for any given bundle.

	Thanks,
	Paul


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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>
> *If* there is a goal to share a single 5-tuple between RTP and SCTP then 
> we need a solution allows bundling them.

Do any of the current proposed bundling techniques include both RTP
and SCTP?

Dale

From christer.holmberg@ericsson.com  Thu Feb 14 12:13:44 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE: RTP only?
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Hi Dale,

I see no reason why BUNDLE couldn't be used also for non-RTP. We would need=
 to consider it when we study the study the SDP attribute impact, though, b=
ut that exercise we have to do for RTP anyway.

For MMT it is more tricky. As it only uses a single m=3D line for the whole=
 bundle, we would need to define a new "MMT" proto value for that m=3D line=
 (currently the draft uses RTP). Then we would need to, inside the m=3D lin=
e, define a way to specify the proto value for each individual media.

Regards,

Christer
________________________________________
From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] on behalf of Dale R=
. Worley [worley@ariadne.com]
Sent: Thursday, 14 February 2013 9:53 PM
To: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE: RTP only?

> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>
> *If* there is a goal to share a single 5-tuple between RTP and SCTP then
> we need a solution allows bundling them.

Do any of the current proposed bundling techniques include both RTP
and SCTP?

Dale
_______________________________________________
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https://www.ietf.org/mailman/listinfo/mmusic

From matthew@matthew.at  Thu Feb 14 13:10:56 2013
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On Feb 14, 2013, at 10:14 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>=20
>=20
> Yes, I know. But nobody has made a proposal that actually encompasses that=
 desire. I think we have turned over a rock and found a large can of worms.
>=20
> There is a deep connection here between the mechanisms used to demux packe=
ts into separate logical "flows" and the SDP notations used to describe thos=
e flows.
>=20
> SDP was designed to describe flows that are demuxed by 3-tuple - incoming t=
ransport type, address, and port. SDP O/A extended that to describe symmetri=
c pairs of such flows. Restriction to symmetry of send/recv address/port pot=
entially extends that to demux by 5-tuple. But we are already on fuzzy groun=
d of SDP expressiveness by that time.
>=20
> Now we want to extend that expressiveness to further levels and criteria f=
or demuxing. The bundle *syntax* lets us describe the things we want to demu=
x, but without any explicit indication of what demux algorithm to apply. The=
 existing bundle proposal applies some restrictions on what can be bundled -=
 presumably to limit it what can be demuxed - but doesn't explicitly say how=
 to demux. (There are restrictions on use of payload types across m-lines th=
at suggest an expectation that demux is to be done based on payload type.) T=
he restrictions it imposes don't permit bundling RTP and SCTP.
>=20
> Unless we want to devolve into a Do What I Mean situation, we need to find=
 a way to explicitly say what we mean.
>=20

Yes, and *only* because RTCWEB turned over the rock *and* continues to insis=
t that SDP be the way for the developers to "explicitly say what we mean" ar=
e we here in MMUSIC with a big can of worms and some urgency around coming u=
p with any solution, no matter how ugly.

There are better ways to construct an API with which developers could tell t=
he browser what they want it to do, and then MMUSIC could work on carefully s=
tandardizing only what is needed for interworking cases.


Matthew Kaufman

(Sent from my iPhone)=

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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 14 February 2013 10:55, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> I agree we don't need an algorithm to demux anything from anything.
>
> But we do need a specified way to look at the SDP and know what the
> mux/demux algorithm is. The rule for bundling should make that a
> prerequisite for any given bundle.

That's only necessary if there is a choice of algorithm.  As long as
you know that you are doing protocol DUCK and protocol GOOSE somehow -
and it matters not how, for now - then the DUCK/GOOSE multiplexing
rules apply.

Where the multiplexed things appear on multiple m= lines, seeing the
protocol labels should suffice.  As Christer notes, the same applies
if you go with the one m= line for all, you still need a way to say
that you want to do SCTP over DTLS somewhere, presumably using the
same sort of label you would use for multiple m= lines.

From dwing@cisco.com  Thu Feb 14 13:56:25 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Paul Kyzivat
> Sent: Thursday, February 14, 2013 10:14 AM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE: RTP only?
> 
> Harald,
> 
> On 2/14/13 3:24 AM, Harald Alvestrand wrote:
> > RTCWEB has a requirement (or rather: it has assumed that there was a
> > solution) for signalling over SDP that SCTP/DTLS/UDP is sent over the
> > same port pair as SRTP.
> >
> > If BUNDLE is not going to be that solution, we have to start designing
> > another.
> 
> Yes, I know. But nobody has made a proposal that actually encompasses
> that desire. I think we have turned over a rock and found a large can of
> worms.
> 
> There is a deep connection here between the mechanisms used to demux
> packets into separate logical "flows" and the SDP notations used to
> describe those flows.
> 
> SDP was designed to describe flows that are demuxed by 3-tuple -
> incoming transport type, address, and port. SDP O/A extended that to
> describe symmetric pairs of such flows. Restriction to symmetry of
> send/recv address/port potentially extends that to demux by 5-tuple. But
> we are already on fuzzy ground of SDP expressiveness by that time.
> 
> Now we want to extend that expressiveness to further levels and criteria
> for demuxing. The bundle *syntax* lets us describe the things we want to
> demux, but without any explicit indication of what demux algorithm to
> apply. The existing bundle proposal applies some restrictions on what
> can be bundled - presumably to limit it what can be demuxed - but
> doesn't explicitly say how to demux. (There are restrictions on use of
> payload types across m-lines that suggest an expectation that demux is
> to be done based on payload type.) The restrictions it imposes don't
> permit bundling RTP and SCTP.
> 
> Unless we want to devolve into a Do What I Mean situation, we need to
> find a way to explicitly say what we mean.
> 
> A good start would be to first be *specific* (and exhaustive) about what
> we would like to multiplex (without regard to how we will notate that),
> and how we propose to mux/demux it.
> 
> There is info out there on how to mux/demux UDP/TLS/RTP/SAVP and
> DTLS/SCTP over the same DTLS connection. (Though it isn't immediately
> obvious where that is specified.)

http://tools.ietf.org/html/rfc5764 explains how to demux RTP (and SRTP),
DTLS, and STUN:

                   +----------------+
                   | 127 < B < 192 -+--> forward to RTP
                   |                |
       packet -->  |  19 < B < 64  -+--> forward to DTLS
                   |                |
                   |       B < 2   -+--> forward to STUN
                   +----------------+

To that, I believe we want to consider SCTP-over-UDP, which I believe
was going to be protected with DTLS.  So, that looks like this, where
DTLS demuxes and processes DTLS handshakes itself and decrypts encrypted
payloads and sends them to SCTP.


            +----------------+
            | 127 < B < 192 -+--> forward to RTP     
            |                |                      /SCTP
packet -->  |  19 < B < 64  -+--> forward to DTLS -<
            |                |                      \DTLS handshake
            |       B < 2   -+--> forward to STUN
            +----------------+

-d


> Presumably the goal for RTCWEB is to demux distinct MediaStreamTracks
> from a single UDP/TLS/RTP/SAVP session. I doubt we want to do that by
> payload type. Based on discussion last week, this may sometimes be done
> by SSRC, but I think we had cases when multiple SSRCs might map to the
> same track, and the mappings may not always be known in advance. CLUE
> has further needs, where an SSRC may move about from one "capture-
> encoding" to another. (A clue capture-encoding is, I think, more or less
> equivalent to an RTCWEB track.)
> 
> So I think we need a serious discussion of mux/demux
> criteria/algorithms. This seems to be a prerequisite to a solution to
> the bundling problem in MMUSIC. But MMUSIC may not be the proper wg for
> this discussion. I don't know which one is.
> 
> 	Thanks,
> 	Paul
> 
> > On 02/14/2013 04:15 AM, Paul Kyzivat wrote:
> >> On 2/13/13 3:08 PM, Christer Holmberg wrote:
> >>>
> >>> Hi,
> >>>
> >>>>> (Sorry for sending this to both RTCWEB and MMUSIC, but it is
> >>>>> related to both the need of RTCWEB, and the solution produced by
> >>>>> MMUSIC)
> >>>>>
> >>>>> Hi,
> >>>>>
> >>>>> At the interim, during my BUNDLE presentation, I indicated that we
> >>>>> need to make a decision whether the solution (whatever it will
> >>>>> be...) shall support bundling of non-RTP media. Not saying it
> >>>>> can't be changed, but as currently documented all alternatives
> >>>>> only support bundling of RTP media.
> >>>>>
> >>>>> There might have been some comments from the mic, but as far as I
> >>>>> know there was no decision (if there was, please point me to it).
> >>>>>
> >>>>> So, my question is: do we keep the scope to RTP media?
> >>>>
> >>>> *If* there is a goal to share a single 5-tuple between RTP and SCTP
> >>>> then we need a solution allows bundling them.
> >>>
> >>> That's another way of presenting my question, yes :)
> >>>
> >>>> The key to what you can bundle together is not whether it is RTP or
> >>>> not.
> >>>> It is whether you can specify the mechanism for demuxing the
> >>>> 5-tuple and associated the parts with the appropriate m-line in the
> >>>> bundle.
> >>>
> >>> Yes, and the question is whether we need to perform the exercise to
> >>> look into that, or if we focus on RTP.
> >>
> >> We know how to do the demux between DTLS payloads, ICE, and RTP/SRTP
> >> payloads. And if we restrict to RTP/SRTP and SCTP over DTLS, then we
> >> know how to demux RTP and SCTP.
> >>
> >> There are various schemes that can be used to demux multiple RTP
> >> streams and associate them with individual m-lines in the bundle.
> >>
> >> What troubles me about it is that the rules for what you can bundle
> >> together are dependent on the demux strategy. Describing the rules
> >> for this as part of the specification of the bundling mechanism is
> >> going to be pretty ugly. I don't see any way to make it clean and
> elegant.
> >>
> >>     Thanks,
> >>     Paul
> >>
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From john@jlc.net  Thu Feb 14 14:25:11 2013
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From: John Leslie <john@jlc.net>
To: Matthew Kaufman <matthew@matthew.at>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Matthew Kaufman <matthew@matthew.at> wrote:
> On Feb 14, 2013, at 10:14 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> On 2/14/13 3:24 AM, Harald Alvestrand wrote:
>> 
>>> RTCWEB has a requirement (or rather: it has assumed that there was a
>>> solution) for signalling over SDP that SCTP/DTLS/UDP is sent over the
>>> same port pair as SRTP.

   (This sentence confuses me, but that's not the point...)

>>> If BUNDLE is not going to be that solution, we have to start designing
>>> another.

   Harald posted to MMUSIC and not RTCWEB. I'm not sure I agree that
MMUSIC has to start designing that...

>> Yes, I know. But nobody has made a proposal that actually encompasses
>> that desire. I think we have turned over a rock and found a large can
>> of worms.

   In my career, I have encountered many cans of worms. I take them as a
signal that my assumptions may have been faulty.

>> There is a deep connection here between the mechanisms used to demux
>> packets into separate logical "flows" and the SDP notations used to
>> describe those flows.

   (Thus, trying to change either threatens to reveal worms...)

>> SDP was designed to describe flows that are demuxed by 3-tuple -
>> incoming transport type, address, and port. SDP O/A extended that to
>> describe symmetric pairs of such flows. Restriction to symmetry of
>> send/recv address/port potentially extends that to demux by 5-tuple.
>> But we are already on fuzzy ground of SDP expressiveness by that time.

   Is it possibly time to stop digging?

>> Now we want to extend that expressiveness to further levels and
>> criteria for demuxing. The bundle *syntax* lets us describe the
>> things we want to demux, but without any explicit indication of
>> what demux algorithm to apply.

   Am I imagining it, or is the ground starting to shake?

>> The existing bundle proposal applies some restrictions on what can
>> be bundled - presumably to limit it what can be demuxed - but
>> doesn't explicitly say how to demux. (There are restrictions on
>> use of payload types across m-lines that suggest an expectation
>> that demux is to be done based on payload type.) The restrictions
>> it imposes don't permit bundling RTP and SCTP.

   ISTM that the existing "bundle" isn't a good way to relate RTP
and SCTP...

>> Unless we want to devolve into a Do What I Mean situation,

   PLEASE NO!!!

>> we need to find a way to explicitly say what we mean.

   Again, Paul posted to MMUSIC and not RTCWEB. Again, I'm not sure
I agree that MMUSIC needs to design that.

> Yes, and *only* because RTCWEB turned over the rock *and* continues
> to insist that SDP be the way for the developers to "explicitly say
> what we mean" are we here in MMUSIC with a big can of worms and some
> urgency around coming up with any solution, no matter how ugly.
> 
> There are better ways to construct an API with which developers
> could tell the browser what they want it to do, and then MMUSIC
> could work on carefully standardizing only what is needed for
> interworking cases.

   Perhaps MMUSIC needs to learn how to say "No" to RTCWEB...

   In an ideal world, MMUSIC would have already designed what RTCWEB
wants. But revisiting basic principles tends to be dangerous and
lead to delays. RTCWEB has good folks who could design a way to do
what they want without needing MMUSIC to revisit its basic design.

   IMHO... YMMV...

--
John Leslie <john@jlc.net>

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Subject: [MMUSIC] draft-thornburgh-adobe-rtmfp notice and feedback request
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hi MMUSIC folks.

i recently uploaded an Internet-Draft describing a vendor network transport=
 protocol, "Secure Real-Time Media Flow Protocol (RTMFP)" from Adobe System=
s Incorporated.  this protocol is the foundation for the real-time and peer=
-to-peer communication capabilities in Flash Player and related Adobe produ=
cts, which are widely deployed on the Internet today.  we believe this tran=
sport protocol has applicability beyond Flash and will be of interest to th=
e Internet community.  this is the protocol presented by Matthew Kaufman to=
 the TSV Area at IETF-77.

we are looking toward this specification being published as an Informationa=
l RFC.  to that end i am soliciting feedback, comments, and reviews of the =
I-D to gauge community interest and to help us improve its quality.  i real=
ize that this kind of protocol isn't in MMUSIC's charter; however, it was r=
ecommended to me that members of this working group should be informed of t=
he existence of this I-D.

there is IPR associated with this specification, which i have disclosed alo=
ng with a royalty-free license declaration.  that disclosure/declaration is=
 linked from the HTML-ized I-D page.

---
Filename:        draft-thornburgh-adobe-rtmfp
Revision:        04
Title:           Adobe's Secure Real-Time Media Flow Protocol
Creation date:   2013-02-14
Group:           Individual Submission
Number of pages: 98
URL:             http://www.ietf.org/internet-drafts/draft-thornburgh-adobe=
-rtmfp-04.txt
Status:          http://datatracker.ietf.org/doc/draft-thornburgh-adobe-rtm=
fp
Htmlized:        http://tools.ietf.org/html/draft-thornburgh-adobe-rtmfp-04
Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-thornburgh-adobe-=
rtmfp-04

Abstract:
   This memo describes the Secure Real-Time Media Flow Protocol (RTMFP),
   an endpoint-to-endpoint communication protocol designed to securely
   transport parallel flows of real-time video, audio, and data
   messages, as well as bulk data, over IP networks.  RTMFP has features
   making it effective for peer-to-peer (P2P) as well as client-server
   communications, even when Network Address Translators (NATs) are
   used.
---

thank you.

-michael thornburgh

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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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I had always assumed that bundle/fumble, etc. (I.e., anything under Cullen's
Plan A) would potentially mux all m-lines together. I assumed that Plan B
(multiple flows per m-line) would not.

-Ekr


On Thu, Feb 14, 2013 at 11:53 AM, Dale R. Worley <worley@ariadne.com> wrote:

> > From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> >
> > *If* there is a goal to share a single 5-tuple between RTP and SCTP then
> > we need a solution allows bundling them.
>
> Do any of the current proposed bundling techniques include both RTP
> and SCTP?
>
> Dale
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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I had always assumed that bundle/fumble, etc. (I.e., anything under Cullen&#39;s<div>Plan A) would potentially mux all m-lines together. I assumed that Plan B</div><div>(multiple flows per m-line) would not.</div><div><br>

</div><div>-Ekr</div><div><br></div><div><div><br><div class="gmail_quote">On Thu, Feb 14, 2013 at 11:53 AM, Dale R. Worley <span dir="ltr">&lt;<a href="mailto:worley@ariadne.com" target="_blank">worley@ariadne.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">&gt; From: Paul Kyzivat &lt;<a href="mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;<br>
<div class="im">&gt;<br>
&gt; *If* there is a goal to share a single 5-tuple between RTP and SCTP then<br>
&gt; we need a solution allows bundling them.<br>
<br>
</div>Do any of the current proposed bundling techniques include both RTP<br>
and SCTP?<br>
<span class="HOEnZb"><font color="#888888"><br>
Dale<br>
</font></span><div class="HOEnZb"><div class="h5">_______________________________________________<br>
mmusic mailing list<br>
<a href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href="https://www.ietf.org/mailman/listinfo/mmusic" target="_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</div></div></blockquote></div><br></div></div>

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From pkyzivat@alum.mit.edu  Thu Feb 14 14:51:38 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 3:13 PM, Christer Holmberg wrote:
> Hi Dale,
>
> I see no reason why BUNDLE couldn't be used also for non-RTP. We would need to consider it when we study the study the SDP attribute impact, though, but that exercise we have to do for RTP anyway.

Again this is all to do with how you do the demux.
E.g. if you bundled multiple m-lines with a UDP proto then it isn't 
clear how you could demux them.

	Thanks,
	Paul

> For MMT it is more tricky. As it only uses a single m= line for the whole bundle, we would need to define a new "MMT" proto value for that m= line (currently the draft uses RTP). Then we would need to, inside the m= line, define a way to specify the proto value for each individual media.
>
> Regards,
>
> Christer
> ________________________________________
> From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] on behalf of Dale R. Worley [worley@ariadne.com]
> Sent: Thursday, 14 February 2013 9:53 PM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE: RTP only?
>
>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>>
>> *If* there is a goal to share a single 5-tuple between RTP and SCTP then
>> we need a solution allows bundling them.
>
> Do any of the current proposed bundling techniques include both RTP
> and SCTP?
>
> Dale
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Ekr said: 

 

"I had always assumed that bundle/fumble, etc. (I.e., anything under
Cullen's

Plan A) would potentially mux all m-lines together. I assumed that Plan B

(multiple flows per m-line) would not.'

 

[BA] Plan B multiplexes multiple SSRCs of the same media type (single m
line)  on the same transport address.   Plan A additionally allows for
multiplexing of different media types (multiple m lines).  So I don't see
them as really competing (e.g. it's possible to want to multiplex multiple
SSRCs from multiple media types on the same transport address). 

 

 


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class=3DMsoNormal>(multiple flows per m-line) would not.<span =
style=3D'color:#1F497D'>&#8217;</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>[BA] Plan B multiplexes multiple SSRCs of the same media type (single =
m line) &nbsp;on the same transport address.&nbsp; &nbsp;Plan A =
additionally allows for multiplexing of different media types (multiple =
m lines). &nbsp;So I don&#8217;t see them as really competing (e.g. =
it&#8217;s possible to want to multiplex multiple SSRCs from multiple =
media types on the same transport address). =
<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
------=_NextPart_000_0103_01CE0AC2.E44FEEE0--

From pkyzivat@alum.mit.edu  Thu Feb 14 15:13:22 2013
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References: <7594FB04B1934943A5C02806D1A2204B0D8131@ESESSMB209.ericsson.se> <511BF229.1090702@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B0D81BD@ESESSMB209.ericsson.se> <511C56CB.1080609@alum.mit.edu> <CABkgnnXVBbeRvy783dTMMdGkwOEqgRFsGWDwERZkpuiNogijHg@mail.gmail.com> <511D3320.2090001@alum.mit.edu> <CABkgnnVQQV_FNRD1hrEOY42cSySbmSyL8MTs3u6GwnRo9JKcYw@mail.gmail.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 4:40 PM, Martin Thomson wrote:
> On 14 February 2013 10:55, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> I agree we don't need an algorithm to demux anything from anything.
>>
>> But we do need a specified way to look at the SDP and know what the
>> mux/demux algorithm is. The rule for bundling should make that a
>> prerequisite for any given bundle.
>
> That's only necessary if there is a choice of algorithm.  As long as
> you know that you are doing protocol DUCK and protocol GOOSE somehow -
> and it matters not how, for now - then the DUCK/GOOSE multiplexing
> rules apply.

If DUCK and GOOSE are identified in the SDP, and its not necessary to 
have any other context than that SDP to discover the DUCK/GOOSE 
multiplexing rules, then I agree with you.

(I realize that RTCWEB has another channel that can be used for 
anything. But depending on that to resolve this would render it 
inapplicable environments like SIP where the SDP is the only likely 
place to convey that information. I gather you are not suggesting going 
that way.)

> Where the multiplexed things appear on multiple m= lines, seeing the
> protocol labels should suffice.  As Christer notes, the same applies
> if you go with the one m= line for all, you still need a way to say
> that you want to do SCTP over DTLS somewhere, presumably using the
> same sort of label you would use for multiple m= lines.

The protocol labels should suffice *if* there is some sort of standard 
or registry that provides a mapping from "bundled protos" to 
"multiplexing rules".

	Thanks,
	Paul


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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 4:56 PM, Dan Wing wrote:
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>> Of Paul Kyzivat
>> Sent: Thursday, February 14, 2013 10:14 AM
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE: RTP only?
>>
>> Harald,
>>
>> On 2/14/13 3:24 AM, Harald Alvestrand wrote:
>>> RTCWEB has a requirement (or rather: it has assumed that there was a
>>> solution) for signalling over SDP that SCTP/DTLS/UDP is sent over the
>>> same port pair as SRTP.
>>>
>>> If BUNDLE is not going to be that solution, we have to start designing
>>> another.
>>
>> Yes, I know. But nobody has made a proposal that actually encompasses
>> that desire. I think we have turned over a rock and found a large can of
>> worms.
>>
>> There is a deep connection here between the mechanisms used to demux
>> packets into separate logical "flows" and the SDP notations used to
>> describe those flows.
>>
>> SDP was designed to describe flows that are demuxed by 3-tuple -
>> incoming transport type, address, and port. SDP O/A extended that to
>> describe symmetric pairs of such flows. Restriction to symmetry of
>> send/recv address/port potentially extends that to demux by 5-tuple. But
>> we are already on fuzzy ground of SDP expressiveness by that time.
>>
>> Now we want to extend that expressiveness to further levels and criteria
>> for demuxing. The bundle *syntax* lets us describe the things we want to
>> demux, but without any explicit indication of what demux algorithm to
>> apply. The existing bundle proposal applies some restrictions on what
>> can be bundled - presumably to limit it what can be demuxed - but
>> doesn't explicitly say how to demux. (There are restrictions on use of
>> payload types across m-lines that suggest an expectation that demux is
>> to be done based on payload type.) The restrictions it imposes don't
>> permit bundling RTP and SCTP.
>>
>> Unless we want to devolve into a Do What I Mean situation, we need to
>> find a way to explicitly say what we mean.
>>
>> A good start would be to first be *specific* (and exhaustive) about what
>> we would like to multiplex (without regard to how we will notate that),
>> and how we propose to mux/demux it.
>>
>> There is info out there on how to mux/demux UDP/TLS/RTP/SAVP and
>> DTLS/SCTP over the same DTLS connection. (Though it isn't immediately
>> obvious where that is specified.)
>
> http://tools.ietf.org/html/rfc5764 explains how to demux RTP (and SRTP),
> DTLS, and STUN:
>
>                     +----------------+
>                     | 127 < B < 192 -+--> forward to RTP
>                     |                |
>         packet -->  |  19 < B < 64  -+--> forward to DTLS
>                     |                |
>                     |       B < 2   -+--> forward to STUN
>                     +----------------+
>
> To that, I believe we want to consider SCTP-over-UDP, which I believe
> was going to be protected with DTLS.  So, that looks like this, where
> DTLS demuxes and processes DTLS handshakes itself and decrypts encrypted
> payloads and sends them to SCTP.
>
>
>              +----------------+
>              | 127 < B < 192 -+--> forward to RTP
>              |                |                      /SCTP
> packet -->  |  19 < B < 64  -+--> forward to DTLS -<
>              |                |                      \DTLS handshake
>              |       B < 2   -+--> forward to STUN
>              +----------------+
>
> -d

Thanks for the reference.

That still leaves unspecified how to demux the RTP relative to the 
bundled RTP m-lines.

ISTM we are going to need a draft that discusses this. I don't know what 
WG that belongs in.

	Thanks,
	Paul


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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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--00504502b883fc7e3e04d5b7a64b
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On Thu, Feb 14, 2013 at 2:52 PM, Bernard Aboba <bernard_aboba@hotmail.com>w=
rote:

> Ekr said: ****
>
> ** **
>
> =93I had always assumed that bundle/fumble, etc. (I.e., anything under
> Cullen's****
>
> Plan A) would potentially mux all m-lines together. I assumed that Plan B=
*
> ***
>
> (multiple flows per m-line) would not.=92****
>
> ** **
>
> [BA] Plan B multiplexes multiple SSRCs of the same media type (single m
> line)  on the same transport address.   Plan A additionally allows for
> multiplexing of different media types (multiple m lines).  So I don=92t s=
ee
> them as really competing (e.g. it=92s possible to want to multiplex multi=
ple
> SSRCs from multiple media types on the same transport address).
>

My sense from what I heard at the interim meeting was that both plan
A and plan B would have sufficient levels of multiplexing that people
would be willing to forego the other. (though of course the more
the better). Perhaps I misunderstood.

-Ekr


>
> **
>

--00504502b883fc7e3e04d5b7a64b
Content-Type: text/html; charset=windows-1252
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<br><br><div class=3D"gmail_quote">On Thu, Feb 14, 2013 at 2:52 PM, Bernard=
 Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><p class=3D"MsoNormal"><=
span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-s=
erif&quot;;color:#1f497d">Ekr said: <u></u><u></u></span></p><p class=3D"Ms=
oNormal">

<u></u>=A0<u></u></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=
=93</span>I had always assumed that bundle/fumble, etc. (I.e., anything und=
er Cullen&#39;s<u></u><u></u></p><div class=3D"im"><div><p class=3D"MsoNorm=
al">Plan A) would potentially mux all m-lines together. I assumed that Plan=
 B<u></u><u></u></p>

</div></div><div><p class=3D"MsoNormal">(multiple flows per m-line) would n=
ot.<span style=3D"color:#1f497d">=92</span><u></u><u></u></p><p class=3D"Ms=
oNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">[BA] Plan B multiplexes m=
ultiple SSRCs of the same media type (single m line) =A0on the same transpo=
rt address.=A0 =A0Plan A additionally allows for multiplexing of different =
media types (multiple m lines). =A0So I don=92t see them as really competin=
g (e.g. it=92s possible to want to multiplex multiple SSRCs from multiple m=
edia types on the same transport address).</span></p>

</div></div></blockquote><div><br></div><div>My sense from what I heard at =
the interim meeting was that both plan</div><div>A and plan B would have su=
fficient levels of multiplexing that people</div><div>would be willing to f=
orego the other. (though of course the more</div>

<div>the better). Perhaps I misunderstood.</div><div><br></div><div>-Ekr</d=
iv><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=
=3D"blue" vlink=3D"purple">

<div><p class=3D"MsoNormal">=A0</p></div><div><p class=3D"MsoNormal"><u></u=
></p></div></div></blockquote></div><br>

--00504502b883fc7e3e04d5b7a64b--

From pkyzivat@alum.mit.edu  Thu Feb 14 15:31:33 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 5:52 PM, Bernard Aboba wrote:
> Ekr said:
>
> “I had always assumed that bundle/fumble, etc. (I.e., anything under
> Cullen's
>
> Plan A) would potentially mux all m-lines together. I assumed that Plan B
>
> (multiple flows per m-line) would not.’
>
> [BA] Plan B multiplexes multiple SSRCs of the same media type (single m
> line)  on the same transport address.   Plan A additionally allows for
> multiplexing of different media types (multiple m lines).  So I don’t
> see them as really competing (e.g. it’s possible to want to multiplex
> multiple SSRCs from multiple media types on the same transport address).

IIUC, Plan A expects a single "flow" (track) per m-line, and uses stuff 
in the SDP to identify exactly which track it is. So it requires demux 
of each "flow" to a particular m-line. I'm not sure how that was 
expected to work.

	Thanks,
	Paul


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From: "Dan Wing" <dwing@cisco.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Emil Ivov'" <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Thursday, February 14, 2013 7:51 AM
> To: Emil Ivov
> Cc: Ari Ker=E4nen; 'mmusic'; Dan Wing
> Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
>=20
> On 2013-02-14 16:36, Emil Ivov wrote:
> > Any idea why there had to be different pacing for RTP and non-RTP
> > flows in the first place?
>=20
> I think our arguments where that in the context of a non-congestion
> controlled voice application it would send with 20 ms interval anyway.
> For other transports and protocols it was not obvious for what =
intended
> environment it would be used. For example a low power network
> application that sends beyond 30 kbps could be problematic. Thus we =
took
> one of our conservative values from the transport books, i.e. 500 ms.
>=20
> I would point out that this is a difficult transport issue. Compared =
to
> TCP IW10, that burst occurs first after the initial handshake when one
> has actually established that there is a responder.
>=20
> With ICE there is a clear issue with an attacker bloating a =
offer/answer
> with candidates that recreates the voice hammer attack using ICE's =
STUN
> checks towards a target address or domain.
>=20
> Also the STUN checks will have different source address pairs which in
> it self could be an issue for attack prevention firewalls etc.
>=20
> Thus I would be careful with speeding up Ta for RTP to much, and I =
think
> someone needs to fine empiric data that it would work.
>=20
> For the the 500 ms that can likely be significantly lowered without =
any
> issues. But some considerations around the wider usage needs to be =
taken
> into account.

Most of those concerns are for the ICE connectivity checks (to the =
remote
peer).  There are different concerns for the initial STUN (and TURN)=20
communications to the STUN/TURN servers, which are the communications
that create the NAT (or firewall) mappings.  I noticed the two different
concerns are nicely written in
http://tools.ietf.org/html/rfc5245#appendix-B.1

Question, playing devil's advocate:  is the voice hammer attack (or a
STUN connectivity check hammer attack) worse than a web page full of
HTML IMG tags pointing at a victim's web server, which I believe has
no protection or pacing?  Has such an HTML IMG attack been a real
problem, or do we consider it will be a problem?

-d

> Cheers
>=20
> Magnus
> One of the ADs responsible for the pacing in ICE.
>=20
> >
> > Emil
> >
> > On 14.02.13, 16:32, Ari Ker=E4nen wrote:
> >> On 2/14/13 5:20 PM, Dan Wing wrote:
> >>>> -----Original Message-----
> >>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>>> Behalf Of Ari Ker=E4nen
> >>>> Sent: Thursday, February 14, 2013 5:49 AM
> >>>> To: mmusic
> >>>> Subject: [MMUSIC] Updating the pacing of ICE connectivity checks
> >>>>
> >>>> Hi all,
> >>>>
> >>>> Updating the pacing parameter, Ta, of ICE for non-RTP traffic =
(for
> >>>> background, see [1] and [2]), as discussed at the Atlanta =
meeting,
> >>>> will be one of the important changes in the new version of ICE.
> >>>>
> >>>> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be
> >>>> configurable, SHOULD have a default of 500 ms, and MUST NOT be
> >>>> configurable to be less than 500 ms."
> >>>>
> >>>> This results in very slow pace of starting connectivity checks if
> >>>> ICE is used with non-RTP traffic. With RTP, the Ta is calculated
> >>>> based on the expected bandwidth used by the RTP stream and can be
> down to 20 ms.
> >>>> Therefore, many current implementations apparently ignore the =
"MUST
> NOT"
> >>>> above and set Ta to something lower.
> >>>
> >>> That would explain failures claimed to justify the desire for
> >>> running over one port.
> >>
> >> Interesting point. Does anyone know what kind of pacing was =
actually
> >> used when these failures were observed?
> >>
> >>> The 20ms threshold for the initial connectivity checks is because
> >>> some NATs could not create new mappings quickly.  This problems
> >>> appears to persist today, considering that both Chrome and Windows =
8
> >>> limit their number of outstanding DNS queries in an apparent =
attempt
> >>> to work around that very same NAT UDP mapping behavior.
> >>>
> >>> If trying to create NAT mappings with 0ms pacing, it will =
certainly
> >>> cause NAT mapping failures and force retries.
> >>
> >> Yes, definitely. So we shouldn't go below 20ms with non-RTP either.
> >> Perhaps need to add some more justification/warning text about this
> too.
> >>
> >>>> The quite conservative value of 500 ms was originally chosen
> >>>> because of the fear of congesting the links with the connectivity
> check messages.
> >>>> This is obviously still a valid concern, but would suggesting
> >>>> something lower still make sense? How low would it be reasonably
> >>>> safe to go? What is currently used in practice? Can we come up =
with
> >>>> a formula similar to RTP (perhaps based on expected bandwidth
> usage)?
> >>>
> >>> A good question for the transport area, which is the origination =
of
> >>> the suggestion to send ICE connectivity checks paced to not exceed
> >>> the bandwidth of the subsequent media.  Perhaps the arguments that
> >>> TCP
> >>> IW=3D10 is not harmful could be brought to the party.
> >>
> >> FYI, I forwarded the original mail to the Transport ADs too.
> >>
> >>
> >> Cheers,
> >> Ari
> >>
> >>>>
> >>>> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
> >>>> [2] http://tools.ietf.org/html/rfc5245#section-16.2
> >>>> _______________________________________________
> >>>> mmusic mailing list
> >>>> mmusic@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/mmusic
> >>>
> >>
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> >>
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


From matthew@matthew.at  Thu Feb 14 15:46:39 2013
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Date: Thu, 14 Feb 2013 15:46:40 -0800
To: Dan Wing <dwing@cisco.com>
Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, =?utf-8?Q?Ari_Ker=C3=A4nen?= <ari.keranen@ericsson.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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On Feb 14, 2013, at 3:43 PM, "Dan Wing" <dwing@cisco.com> wrote:

>=20
> Question, playing devil's advocate:  is the voice hammer attack (or a
> STUN connectivity check hammer attack) worse than a web page full of
> HTML IMG tags pointing at a victim's web server, which I believe has
> no protection or pacing?  Has such an HTML IMG attack been a real
> problem, or do we consider it will be a problem?

Most browsers do have a limit for the number of simultaneous IMG loads, some=
 even more so if they all appear to be pointed at the same domain.

Matthew Kaufman=

From dwing@cisco.com  Thu Feb 14 15:55:05 2013
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From: "Dan Wing" <dwing@cisco.com>
To: "'Matthew Kaufman'" <matthew@matthew.at>
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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> -----Original Message-----
> From: Matthew Kaufman [mailto:matthew@matthew.at]
> Sent: Thursday, February 14, 2013 3:47 PM
> To: Dan Wing
> Cc: Magnus Westerlund; Emil Ivov; Ari Ker=E4nen; mmusic
> Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
>=20
>=20
>=20
> On Feb 14, 2013, at 3:43 PM, "Dan Wing" <dwing@cisco.com> wrote:
>=20
> >
> > Question, playing devil's advocate:  is the voice hammer attack (or =
a
> > STUN connectivity check hammer attack) worse than a web page full of
> > HTML IMG tags pointing at a victim's web server, which I believe has
> > no protection or pacing?  Has such an HTML IMG attack been a real
> > problem, or do we consider it will be a problem?
>=20
> Most browsers do have a limit for the number of simultaneous IMG =
loads,
> some even more so if they all appear to be pointed at the same domain.
>=20
> Matthew Kaufman=3D

I guess you are referring to the per-server HTTP connection limit, but=20
I am guessing on your reference.  If so, perhaps modeling the ICEbis=20
limits (for ICE  connectivity checks) on that design is a straw man=20
we should consider.  If not, perhaps you could clarify what Most
Browsers do regarding such a crafted web page (which of course need
not be IMG, but any HTML tag that pulls content from another host).

-d



From pkyzivat@alum.mit.edu  Thu Feb 14 16:25:13 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 4:10 PM, Matthew Kaufman wrote:
>
>
> On Feb 14, 2013, at 10:14 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>>
>>
>> Yes, I know. But nobody has made a proposal that actually encompasses that desire. I think we have turned over a rock and found a large can of worms.
>>
>> There is a deep connection here between the mechanisms used to demux packets into separate logical "flows" and the SDP notations used to describe those flows.
>>
>> SDP was designed to describe flows that are demuxed by 3-tuple - incoming transport type, address, and port. SDP O/A extended that to describe symmetric pairs of such flows. Restriction to symmetry of send/recv address/port potentially extends that to demux by 5-tuple. But we are already on fuzzy ground of SDP expressiveness by that time.
>>
>> Now we want to extend that expressiveness to further levels and criteria for demuxing. The bundle *syntax* lets us describe the things we want to demux, but without any explicit indication of what demux algorithm to apply. The existing bundle proposal applies some restrictions on what can be bundled - presumably to limit it what can be demuxed - but doesn't explicitly say how to demux. (There are restrictions on use of payload types across m-lines that suggest an expectation that demux is to be done based on payload type.) The restrictions it imposes don't permit bundling RTP and SCTP.
>>
>> Unless we want to devolve into a Do What I Mean situation, we need to find a way to explicitly say what we mean.
>>
>
> Yes, and *only* because RTCWEB turned over the rock *and* continues to insist that SDP be the way for the developers to "explicitly say what we mean" are we here in MMUSIC with a big can of worms and some urgency around coming up with any solution, no matter how ugly.
>
> There are better ways to construct an API with which developers could tell the browser what they want it to do, and then MMUSIC could work on carefully standardizing only what is needed for interworking cases.

I realize that for RTCWEB the browser is always in the middle,
but this is really about telling the other endpoint what you want to do,
since these media streams are e2e.

If this isn't done in SDP then it won't be available to SIP endpoints.

	Thanks,
	Paul


From matthew@matthew.at  Thu Feb 14 16:55:34 2013
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From: Matthew Kaufman <matthew@matthew.at>
Date: Thu, 14 Feb 2013 16:55:37 -0800
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On Feb 14, 2013, at 4:25 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 2/14/13 4:10 PM, Matthew Kaufman wrote:
>>=20
>>=20
>> On Feb 14, 2013, at 10:14 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:=

>>>=20
>>> Yes, I know. But nobody has made a proposal that actually encompasses th=
at desire. I think we have turned over a rock and found a large can of worms=
.
>>>=20
>>> There is a deep connection here between the mechanisms used to demux pac=
kets into separate logical "flows" and the SDP notations used to describe th=
ose flows.
>>>=20
>>> SDP was designed to describe flows that are demuxed by 3-tuple - incomin=
g transport type, address, and port. SDP O/A extended that to describe symme=
tric pairs of such flows. Restriction to symmetry of send/recv address/port p=
otentially extends that to demux by 5-tuple. But we are already on fuzzy gro=
und of SDP expressiveness by that time.
>>>=20
>>> Now we want to extend that expressiveness to further levels and criteria=
 for demuxing. The bundle *syntax* lets us describe the things we want to de=
mux, but without any explicit indication of what demux algorithm to apply. T=
he existing bundle proposal applies some restrictions on what can be bundled=
 - presumably to limit it what can be demuxed - but doesn't explicitly say h=
ow to demux. (There are restrictions on use of payload types across m-lines t=
hat suggest an expectation that demux is to be done based on payload type.) T=
he restrictions it imposes don't permit bundling RTP and SCTP.
>>>=20
>>> Unless we want to devolve into a Do What I Mean situation, we need to fi=
nd a way to explicitly say what we mean.
>>=20
>> Yes, and *only* because RTCWEB turned over the rock *and* continues to in=
sist that SDP be the way for the developers to "explicitly say what we mean"=
 are we here in MMUSIC with a big can of worms and some urgency around comin=
g up with any solution, no matter how ugly.
>>=20
>> There are better ways to construct an API with which developers could tel=
l the browser what they want it to do, and then MMUSIC could work on careful=
ly standardizing only what is needed for interworking cases.
>=20
> I realize that for RTCWEB the browser is always in the middle,
> but this is really about telling the other endpoint what you want to do,
> since these media streams are e2e.

Browser to browser e2e streams can be signaled through whatever means the de=
veloper of the end to end application comes up with...

>=20
> If this isn't done in SDP then it won't be available to SIP endpoints.

Right. But maybe not every possible combination is needed when working with S=
IP endpoints, and maybe, just maybe, the artificial urgency created by RTCWE=
B's need to have this fully specified yesterday wouldn't be putting us at se=
rious risk of a bad solution to what is likely much more complicated than an=
yone has yet imagined.

Matthew Kaufman=

From christer.holmberg@ericsson.com  Thu Feb 14 23:35:31 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE: RTP only?
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Hi,

>> I see no reason why BUNDLE couldn't be used also for non-RTP. We would n=
eed to consider it when we study the study the SDP attribute impact, though=
, but that exercise we have to do for RTP anyway.
>
> Again this is all to do with how you do the demux.
> E.g. if you bundled multiple m-lines with a UDP proto then it isn't clear=
 how you could demux them.

Sure. If it is not possible to demux a protocol, then you obviously can't d=
o it.

But, the scope of BUNDLE is not really to solve those kind of problems. The=
 scope is to define how the demuxing, when it can be done, is negotiated on=
 a signaling level, and what restrictions etc there are associated with tha=
t..

Regards,

Christer




> For MMT it is more tricky. As it only uses a single m=3D line for the who=
le bundle, we would need to define a new "MMT" proto value for that m=3D li=
ne (currently the draft uses RTP). Then we would need to, inside the m=3D l=
ine, define a way to specify the proto value for each individual media.
>
> Regards,
>
> Christer
> ________________________________________
> From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] on behalf of=20
> Dale R. Worley [worley@ariadne.com]
> Sent: Thursday, 14 February 2013 9:53 PM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE: RTP only?
>
>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>>
>> *If* there is a goal to share a single 5-tuple between RTP and SCTP=20
>> then we need a solution allows bundling them.
>
> Do any of the current proposed bundling techniques include both RTP=20
> and SCTP?
>
> Dale
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
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From pkyzivat@alum.mit.edu  Fri Feb 15 08:10:26 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/15/13 2:35 AM, Christer Holmberg wrote:
> Hi,
>
>>> I see no reason why BUNDLE couldn't be used also for non-RTP. We would need to consider it when we study the study the SDP attribute impact, though, but that exercise we have to do for RTP anyway.
>>
>> Again this is all to do with how you do the demux.
>> E.g. if you bundled multiple m-lines with a UDP proto then it isn't clear how you could demux them.
>
> Sure. If it is not possible to demux a protocol, then you obviously can't do it.
>
> But, the scope of BUNDLE is not really to solve those kind of problems. The scope is to define how the demuxing, when it can be done, is negotiated on a signaling level, and what restrictions etc there are associated with that..

No, BUNDLE doesn't need to solve the problem. But it does require that 
there be a solution to the problem. And IMO it requires that the 
mux/demux mechanism be determinable from the SDP.

ISTM that a *start* is to ensure that bundle combinations we care about 
have mux/demux algorithms documented someplace that can be referenced. 
Then we reference those in the bundle draft, or create a registry and 
reference that from the bundle draft.

	Thanks,
	Paul

> Regards,
>
> Christer
>
>
>
>
>> For MMT it is more tricky. As it only uses a single m= line for the whole bundle, we would need to define a new "MMT" proto value for that m= line (currently the draft uses RTP). Then we would need to, inside the m= line, define a way to specify the proto value for each individual media.
>>
>> Regards,
>>
>> Christer
>> ________________________________________
>> From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] on behalf of
>> Dale R. Worley [worley@ariadne.com]
>> Sent: Thursday, 14 February 2013 9:53 PM
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE: RTP only?
>>
>>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>>>
>>> *If* there is a goal to share a single 5-tuple between RTP and SCTP
>>> then we need a solution allows bundling them.
>>
>> Do any of the current proposed bundling techniques include both RTP
>> and SCTP?
>>
>> Dale
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
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>


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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/14/13 7:55 PM, Matthew Kaufman wrote:
>
>
> On Feb 14, 2013, at 4:25 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> On 2/14/13 4:10 PM, Matthew Kaufman wrote:
>>>
>>>
>>> On Feb 14, 2013, at 10:14 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>>>
>>>> Yes, I know. But nobody has made a proposal that actually encompasses that desire. I think we have turned over a rock and found a large can of worms.
>>>>
>>>> There is a deep connection here between the mechanisms used to demux packets into separate logical "flows" and the SDP notations used to describe those flows.
>>>>
>>>> SDP was designed to describe flows that are demuxed by 3-tuple - incoming transport type, address, and port. SDP O/A extended that to describe symmetric pairs of such flows. Restriction to symmetry of send/recv address/port potentially extends that to demux by 5-tuple. But we are already on fuzzy ground of SDP expressiveness by that time.
>>>>
>>>> Now we want to extend that expressiveness to further levels and criteria for demuxing. The bundle *syntax* lets us describe the things we want to demux, but without any explicit indication of what demux algorithm to apply. The existing bundle proposal applies some restrictions on what can be bundled - presumably to limit it what can be demuxed - but doesn't explicitly say how to demux. (There are restrictions on use of payload types across m-lines that suggest an expectation that demux is to be done based on payload type.) The restrictions it imposes don't permit bundling RTP and SCTP.
>>>>
>>>> Unless we want to devolve into a Do What I Mean situation, we need to find a way to explicitly say what we mean.
>>>
>>> Yes, and *only* because RTCWEB turned over the rock *and* continues to insist that SDP be the way for the developers to "explicitly say what we mean" are we here in MMUSIC with a big can of worms and some urgency around coming up with any solution, no matter how ugly.
>>>
>>> There are better ways to construct an API with which developers could tell the browser what they want it to do, and then MMUSIC could work on carefully standardizing only what is needed for interworking cases.
>>
>> I realize that for RTCWEB the browser is always in the middle,
>> but this is really about telling the other endpoint what you want to do,
>> since these media streams are e2e.
>
> Browser to browser e2e streams can be signaled through whatever means the developer of the end to end application comes up with...

What about browser to non-browser app?

E.g. CLUE in a browser to clue in a server.

>> If this isn't done in SDP then it won't be available to SIP endpoints.
>
> Right. But maybe not every possible combination is needed when working with SIP endpoints, and maybe, just maybe, the artificial urgency created by RTCWEB's need to have this fully specified yesterday wouldn't be putting us at serious risk of a bad solution to what is likely much more complicated than anyone has yet imagined.

There will be at least *some* combinations that are needed when working 
with SIP endpoints. Maybe we at least figure out a solution that works 
for those.

	Thanks,
	Paul


From worley@shell01.TheWorld.com  Fri Feb 15 08:35:32 2013
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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> From: Matthew Kaufman <matthew@matthew.at>
> 
> Most browsers do have a limit for the number of simultaneous IMG
> loads, some even more so if they all appear to be pointed at the
> same domain.

IIRC, browsers aren't supposed to have more than two TCP connections
to any one server name.  Though I suppose you can get around that by
creating a zillion A records in your private domain that all point to
the same IP address.

   RFC 2616 (HTTP/1.1) section 8.1.4

   Clients that use persistent connections SHOULD limit the number of
   simultaneous connections that they maintain to a given server. A
   single-user client SHOULD NOT maintain more than 2 connections with
   any server or proxy.

Dale

From fluffy@cisco.com  Fri Feb 15 08:36:22 2013
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Hi All,

As discussed at the interim, Richard and I are planning to have a call to d=
iscuss issues around the same port or different ports in bundle. This is no=
t an official WG call or anything but anyone is welcome to join us on the c=
all. Bridge info below.

Cullen




Cullen Jennings invites you to attend this online meeting.

Topic: Bundle Ports
Date: Thursday, February 21, 2013
Time: 7:00 am, Pacific Standard Time (San Francisco, GMT-08:00)
Meeting Number: 207 772 340
Meeting Password: ietf


-------------------------------------------------------
To join the online meeting (Now from mobile devices!)
-------------------------------------------------------
1. Go to https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&UID=
=3D0&PW=3DNMjYwN2JjNzYz&RT=3DMiM0
2. Enter your name and email address.
3. Enter the meeting password: ietf
4. Click "Join Now".

To view in other time zones or languages, please click the link:
https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&UID=3D0&PW=3DN=
MjYwN2JjNzYz&ORT=3DMiM0

----------------------------------------------------------------
ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
----------------------------------------------------------------

The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpita=
s area and (866) 349-3520 for the RTP area.

Please dial the local access number for your area from the list below:
- San Jose/Milpitas (408) area: 525-6800
- RTP (919) area: 392-3330

-------------------------------------------------------
To join the teleconference only
-------------------------------------------------------
1. Dial into Cisco WebEx (view all Global Access Numbers at
http://cisco.com/en/US/about/doing_business/conferencing/index.html
2. Follow the prompts to enter the Meeting Number (listed above) or Access =
Code followed by the # sign.

San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330

US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117

India: +91.80.4350.1111 Germany: +49.619.6773.9002

Japan: +81.3.5763.9394 China: +86.10.8515.5666

-------------------------------------------------------


To add this meeting to your calendar program (for example Microsoft Outlook=
), click this link:
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<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div><br>
</div>
Hi All,
<div><br>
</div>
<div>As discussed at the interim, Richard and I are planning to have a call=
 to discuss issues around the same port or different ports in bundle. This =
is not an official WG call or anything but anyone is welcome to join us on =
the call. Bridge info below.&nbsp;</div>
<div><br>
</div>
<div>Cullen</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>
<blockquote type=3D"cite"><font face=3D"Tahoma, Arial, sans-serif, Helvetic=
a, Geneva" size=3D"2"><br>
Cullen Jennings invites you to attend this online meeting. <br>
<br>
Topic: Bundle Ports <br>
Date: Thursday, February 21, 2013 <br>
Time: 7:00 am, Pacific Standard Time (San Francisco, GMT-08:00) <br>
Meeting Number: 207 772 340 <br>
Meeting Password: ietf <br>
<br>
<br>
------------------------------------------------------- <br>
To join the online meeting (Now from mobile devices!) <br>
------------------------------------------------------- <br>
1. Go to <a href=3D"https://ciscosales.webex.com/ciscosales/j.php?ED=3D2175=
53672&amp;UID=3D0&amp;PW=3DNMjYwN2JjNzYz&amp;RT=3DMiM0" target=3D"_blank">
https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&amp;UID=3D0&am=
p;PW=3DNMjYwN2JjNzYz&amp;RT=3DMiM0</a>
<br>
2. Enter your name and email address. <br>
3. Enter the meeting password: ietf <br>
4. Click &quot;Join Now&quot;. <br>
<br>
To view in other time zones or languages, please click the link: <br>
<a href=3D"https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&amp=
;UID=3D0&amp;PW=3DNMjYwN2JjNzYz&amp;ORT=3DMiM0" target=3D"_blank">https://c=
iscosales.webex.com/ciscosales/j.php?ED=3D217553672&amp;UID=3D0&amp;PW=3DNM=
jYwN2JjNzYz&amp;ORT=3DMiM0</a>
<br>
<br>
---------------------------------------------------------------- <br>
ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes <br>
---------------------------------------------------------------- <br>
<br>
The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpita=
s area and (866) 349-3520 for the RTP area.
<br>
<br>
Please dial the local access number for your area from the list below: <br>
- San Jose/Milpitas (408) area: 525-6800 <br>
- RTP (919) area: 392-3330 <br>
<br>
------------------------------------------------------- <br>
To join the teleconference only <br>
------------------------------------------------------- <br>
1. Dial into Cisco WebEx (view all Global Access Numbers at <br>
<a href=3D"http://cisco.com/en/US/about/doing_business/conferencing/index.h=
tml" target=3D"_blank">http://cisco.com/en/US/about/doing_business/conferen=
cing/index.html</a>
<br>
2. Follow the prompts to enter the Meeting Number (listed above) or Access =
Code followed by the # sign.
<br>
<br>
San Jose, CA: &#43;1.408.525.6800 RTP: &#43;1.919.392.3330 <br>
<br>
US/Canada: &#43;1.866.432.9903 United Kingdom: &#43;44.20.8824.0117 <br>
<br>
India: &#43;91.80.4350.1111 Germany: &#43;49.619.6773.9002 <br>
<br>
Japan: &#43;81.3.5763.9394 China: &#43;86.10.8515.5666 <br>
<br>
------------------------------------------------------- <br>
<br>
<br>
To add this meeting to your calendar program (for example Microsoft Outlook=
), click this link:
<br>
<a href=3D"https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&amp=
;UID=3D0&amp;ICS=3DMI&amp;LD=3D1&amp;RD=3D2&amp;ST=3D1&amp;SHA2=3DAAAAAkvkB=
wPvL1IZY/a2pGU6xcxB7i5WW2-fXSdiMwm3e2Gw&amp;RT=3DMiM0" target=3D"_blank">ht=
tps://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&amp;UID=3D0&amp;=
ICS=3DMI&amp;LD=3D1&amp;RD=3D2&amp;ST=3D1&amp;SHA2=3DAAAAAkvkBwPvL1IZY/a2pG=
U6xcxB7i5WW2-fXSdiMwm3e2Gw&amp;RT=3DMiM0</a>
<br>
<br>
</font></blockquote>
</div>
</div>
</body>
</html>

--_000_C5E08FE080ACFD4DAE31E4BDBF944EB1133E51C1xmbalnx02ciscoc_--

From gonzalo.camarillo@ericsson.com  Fri Feb 15 08:47:21 2013
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Date: Fri, 15 Feb 2013 18:47:18 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
References: <511CEB64.6070306@ericsson.com> <05a401ce0ac6$db547230$91fd5690$@cisco.com> <511D03A7.1060008@ericsson.com> <511D0498.3030909@jitsi.org> <511D07F1.3070808@ericsson.com>
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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Hi,

to add to what Magnus said, it is important that we understand what
implementations have been doing so far (regardless of what the RFC says)
and that we come up with recommendations implementers will actually
follow. Keep in mind we are here to write specifications, not fiction ;-)

Thanks,

Gonzalo


On 14/02/2013 5:51 PM, Magnus Westerlund wrote:
> Thus I would be careful with speeding up Ta for RTP to much, and I think
> someone needs to fine empiric data that it would work.
> 
> For the the 500 ms that can likely be significantly lowered without any
> issues. But some considerations around the wider usage needs to be taken
> into account.


From emil@sip-communicator.org  Fri Feb 15 09:16:40 2013
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Date: Fri, 15 Feb 2013 18:16:34 +0100
From: Emil Ivov <emcho@jitsi.org>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
References: <511CEB64.6070306@ericsson.com> <05a401ce0ac6$db547230$91fd5690$@cisco.com> <511D03A7.1060008@ericsson.com> <511D0498.3030909@jitsi.org> <511D07F1.3070808@ericsson.com> <511E6696.6090609@ericsson.com>
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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Well for one, we hadn't even noticed that there was different pacing for
non-RTP traffic, so when we added support for Pseudo TCP to ice4j.org,
we just let it use the default timers. It really didn't cross our
collective minds that the same stack in the same application would have
the right of going down to 20ms if it used the magic word (and RTP
header) but then go back up to 500 ms if it didn't. It's almost as if it
were punished for having bad non-RTP thoughts ;)

Anyways, I think that, had we known about the "MUST NOT drop below 500
ms" limitation, we would have simply ignored it: these are user
perceptible delays after all, and no one would like to wayt 10+ seconds
for a file to start transferring.

Cheers,
Emil

On 15.02.13, 17:47, Gonzalo Camarillo wrote:
> Hi,
> 
> to add to what Magnus said, it is important that we understand what
> implementations have been doing so far (regardless of what the RFC says)
> and that we come up with recommendations implementers will actually
> follow. Keep in mind we are here to write specifications, not fiction ;-)
> 
> Thanks,
> 
> Gonzalo
> 
> 
> On 14/02/2013 5:51 PM, Magnus Westerlund wrote:
>> Thus I would be careful with speeding up Ta for RTP to much, and I think
>> someone needs to fine empiric data that it would work.
>>
>> For the the 500 ms that can likely be significantly lowered without any
>> issues. But some considerations around the wider usage needs to be taken
>> into account.
> 
> 

-- 
https://jitsi.org

From rlb@ipv.sx  Fri Feb 15 09:50:58 2013
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Date: Fri, 15 Feb 2013 12:50:55 -0500
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From: Richard Barnes <rlb@ipv.sx>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On Friday, February 15, 2013, Christer Holmberg wrote:

> Hi,
>
> >> I see no reason why BUNDLE couldn't be used also for non-RTP. We would
> need to consider it when we study the study the SDP attribute impact,
> though, but that exercise we have to do for RTP anyway.
> >
> > Again this is all to do with how you do the demux.
> > E.g. if you bundled multiple m-lines with a UDP proto then it isn't
> clear how you could demux them.
>
> Sure. If it is not possible to demux a protocol, then you obviously can't
> do it.
>
> But, the scope of BUNDLE is not really to solve those kind of problems.
> The scope is to define how the demuxing, when it can be done, is negotiated
> on a signaling level, and what restrictions etc there are associated with
> that..
>
> Regards,
>
> Christer
>
>
This seems right to me.  The SDP just needs to allow the offeror to declare
"I am going to send packets of types DUCK and GOOSE and AVOCADO over this
channel." The mux part is trivial.

The offerer shouldn't offer a combination he can't demux. The answerer
shouldn't accept a combination he can't demux.  In other words, this is a
local matter, like codec support.

That's not to say that demux recommendations shouldn't be written -- they
should -- but the SDP does not need to account for demux algorithms.

Just my $.02,
--Richard



>
>
>
> > For MMT it is more tricky. As it only uses a single m= line for the
> whole bundle, we would need to define a new "MMT" proto value for that m=
> line (currently the draft uses RTP). Then we would need to, inside the m=
> line, define a way to specify the proto value for each individual media.
> >
> > Regards,
> >
> > Christer
> > ________________________________________
> > From: mmusic-bounces@ietf.org <javascript:;> [mmusic-bounces@ietf.org<javascript:;>]
> on behalf of
> > Dale R. Worley [worley@ariadne.com <javascript:;>]
> > Sent: Thursday, 14 February 2013 9:53 PM
> > To: mmusic@ietf.org <javascript:;>
> > Subject: Re: [MMUSIC] BUNDLE: RTP only?
> >
> >> From: Paul Kyzivat <pkyzivat@alum.mit.edu <javascript:;>>
> >>
> >> *If* there is a goal to share a single 5-tuple between RTP and SCTP
> >> then we need a solution allows bundling them.
> >
> > Do any of the current proposed bundling techniques include both RTP
> > and SCTP?
> >
> > Dale
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org <javascript:;>
> > https://www.ietf.org/mailman/listinfo/mmusic
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org <javascript:;>
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org <javascript:;>
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org <javascript:;>
> https://www.ietf.org/mailman/listinfo/mmusic
>

--bcaec54ee608666edb04d5c70428
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On Friday, February 15, 2013, Christer Holmberg  wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">Hi,<br>
<br>
&gt;&gt; I see no reason why BUNDLE couldn&#39;t be used also for non-RTP. =
We would need to consider it when we study the study the SDP attribute impa=
ct, though, but that exercise we have to do for RTP anyway.<br>
&gt;<br>
&gt; Again this is all to do with how you do the demux.<br>
&gt; E.g. if you bundled multiple m-lines with a UDP proto then it isn&#39;=
t clear how you could demux them.<br>
<br>
Sure. If it is not possible to demux a protocol, then you obviously can&#39=
;t do it.<br>
<br>
But, the scope of BUNDLE is not really to solve those kind of problems. The=
 scope is to define how the demuxing, when it can be done, is negotiated on=
 a signaling level, and what restrictions etc there are associated with tha=
t..<br>

<br>
Regards,<br>
<br>
Christer<br>
<br></blockquote><div><br></div><div>This seems right to me. =A0The SDP jus=
t=A0needs to allow the offeror to=A0declare &quot;I am going to send packet=
s of types DUCK and GOOSE and AVOCADO over this channel.&quot; The mux part=
 is trivial.=A0</div>
<div><br></div><div>The offerer shouldn&#39;t offer a combination he can&#3=
9;t demux. The answerer shouldn&#39;t accept a combination he can&#39;t dem=
ux. =A0In other words, this is a local matter, like codec support.=A0</div>
<div><br></div><div>That&#39;s not to say that demux recommendations should=
n&#39;t be written -- they should -- but the SDP does not need to account f=
or demux algorithms.=A0</div><div><br></div><div>Just my $.02,<span></span>=
</div>
<div>--Richard</div><div><br></div><div>=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">
<br>
<br>
<br>
&gt; For MMT it is more tricky. As it only uses a single m=3D line for the =
whole bundle, we would need to define a new &quot;MMT&quot; proto value for=
 that m=3D line (currently the draft uses RTP). Then we would need to, insi=
de the m=3D line, define a way to specify the proto value for each individu=
al media.<br>

&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt; ________________________________________<br>
&gt; From: <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#=
39;mmusic-bounces@ietf.org&#39;)">mmusic-bounces@ietf.org</a> [<a href=3D"j=
avascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmusic-bounces@ietf.=
org&#39;)">mmusic-bounces@ietf.org</a>] on behalf of<br>

&gt; Dale R. Worley [<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvm=
l&#39;, &#39;worley@ariadne.com&#39;)">worley@ariadne.com</a>]<br>
&gt; Sent: Thursday, 14 February 2013 9:53 PM<br>
&gt; To: <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39=
;mmusic@ietf.org&#39;)">mmusic@ietf.org</a><br>
&gt; Subject: Re: [MMUSIC] BUNDLE: RTP only?<br>
&gt;<br>
&gt;&gt; From: Paul Kyzivat &lt;<a href=3D"javascript:;" onclick=3D"_e(even=
t, &#39;cvml&#39;, &#39;pkyzivat@alum.mit.edu&#39;)">pkyzivat@alum.mit.edu<=
/a>&gt;<br>
&gt;&gt;<br>
&gt;&gt; *If* there is a goal to share a single 5-tuple between RTP and SCT=
P<br>
&gt;&gt; then we need a solution allows bundling them.<br>
&gt;<br>
&gt; Do any of the current proposed bundling techniques include both RTP<br=
>
&gt; and SCTP?<br>
&gt;<br>
&gt; Dale<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmu=
sic@ietf.org&#39;)">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmu=
sic@ietf.org&#39;)">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmusic@i=
etf.org&#39;)">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmusic@i=
etf.org&#39;)">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote>

--bcaec54ee608666edb04d5c70428--

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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Updating the pacing of ICE connectivity checks
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, "Dan Wing \(dwing\)" <dwing@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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On Feb 15, 2013, at 18:16 PM, Emil Ivov <emcho@jitsi.org>
 wrote:

> Well for one, we hadn't even noticed that there was different pacing for
> non-RTP traffic, so when we added support for Pseudo TCP to ice4j.org,
> we just let it use the default timers. It really didn't cross our
> collective minds that the same stack in the same application would have
> the right of going down to 20ms if it used the magic word (and RTP
> header) but then go back up to 500 ms if it didn't. It's almost as if it
> were punished for having bad non-RTP thoughts ;)
>=20
Ha,ha. For us it was opposite. Never noticed the non-RTP stream limitation.=
..

> Anyways, I think that, had we known about the "MUST NOT drop below 500
> ms" limitation, we would have simply ignored it: these are user
> perceptible delays after all, and no one would like to wayt 10+ seconds
> for a file to start transferring.
>=20

On a side note I think simpler is better. Choosing a 20ms timer (or any val=
ue) is better than calculating something from what bandwidth you are using.=
 It is hard to say what bandwidth a H.264 encoder is going to use, depends =
on how much movement in the picture and so on.

.-.
P=E5l-Erik

> Cheers,
> Emil
>=20
> On 15.02.13, 17:47, Gonzalo Camarillo wrote:
>> Hi,
>>=20
>> to add to what Magnus said, it is important that we understand what
>> implementations have been doing so far (regardless of what the RFC says)
>> and that we come up with recommendations implementers will actually
>> follow. Keep in mind we are here to write specifications, not fiction ;-=
)
>>=20
>> Thanks,
>>=20
>> Gonzalo
>>=20
>>=20
>> On 14/02/2013 5:51 PM, Magnus Westerlund wrote:
>>> Thus I would be careful with speeding up Ta for RTP to much, and I thin=
k
>>> someone needs to fine empiric data that it would work.
>>>=20
>>> For the the 500 ms that can likely be significantly lowered without any
>>> issues. But some considerations around the wider usage needs to be take=
n
>>> into account.
>>=20
>>=20
>=20
> --=20
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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On 2/15/13 12:50 PM, Richard Barnes wrote:
> On Friday, February 15, 2013, Christer Holmberg wrote:
>
>     Hi,
>
>      >> I see no reason why BUNDLE couldn't be used also for non-RTP. We
>     would need to consider it when we study the study the SDP attribute
>     impact, though, but that exercise we have to do for RTP anyway.
>      >
>      > Again this is all to do with how you do the demux.
>      > E.g. if you bundled multiple m-lines with a UDP proto then it
>     isn't clear how you could demux them.
>
>     Sure. If it is not possible to demux a protocol, then you obviously
>     can't do it.
>
>     But, the scope of BUNDLE is not really to solve those kind of
>     problems. The scope is to define how the demuxing, when it can be
>     done, is negotiated on a signaling level, and what restrictions etc
>     there are associated with that..
>
>     Regards,
>
>     Christer
>
>
> This seems right to me.  The SDP just needs to allow the offeror
> to declare "I am going to send packets of types DUCK and GOOSE and
> AVOCADO over this channel." The mux part is trivial.
>
> The offerer shouldn't offer a combination he can't demux. The answerer
> shouldn't accept a combination he can't demux.  In other words, this is
> a local matter, like codec support.

This doesn't work unless the offerer and the answerer *agree* on the 
mux/demux algorithm.

E.g. It would be bad if the offerer uses a unique payload type for each 
"flow"/track and switches among SSRCs within it, while the answerer 
demuxes into flows/tracks based on SSRC.

> That's not to say that demux recommendations shouldn't be written --
> they should -- but the SDP does not need to account for demux algorithms.

You used an analogy to codec support. SDP doesn't define particular 
codecs, but it does provide the mechanism for the offerer and answerer 
to agree on which codecs will be used.

The same would be true here *if* the SDP allowed the mux/demux algorithm 
to be declared or negotiated.

	Thanks,
	Paul

> Just my $.02,
> --Richard
>
>
>
>
>      > For MMT it is more tricky. As it only uses a single m= line for
>     the whole bundle, we would need to define a new "MMT" proto value
>     for that m= line (currently the draft uses RTP). Then we would need
>     to, inside the m= line, define a way to specify the proto value for
>     each individual media.
>      >
>      > Regards,
>      >
>      > Christer
>      > ________________________________________
>      > From: mmusic-bounces@ietf.org <javascript:;>
>     [mmusic-bounces@ietf.org <javascript:;>] on behalf of
>      > Dale R. Worley [worley@ariadne.com <javascript:;>]
>      > Sent: Thursday, 14 February 2013 9:53 PM
>      > To: mmusic@ietf.org <javascript:;>
>      > Subject: Re: [MMUSIC] BUNDLE: RTP only?
>      >
>      >> From: Paul Kyzivat <pkyzivat@alum.mit.edu <javascript:;>>
>      >>
>      >> *If* there is a goal to share a single 5-tuple between RTP and SCTP
>      >> then we need a solution allows bundling them.
>      >
>      > Do any of the current proposed bundling techniques include both RTP
>      > and SCTP?
>      >
>      > Dale
>      > _______________________________________________
>      > mmusic mailing list
>      > mmusic@ietf.org <javascript:;>
>      > https://www.ietf.org/mailman/listinfo/mmusic
>      > _______________________________________________
>      > mmusic mailing list
>      > mmusic@ietf.org <javascript:;>
>      > https://www.ietf.org/mailman/listinfo/mmusic
>      >
>
>     _______________________________________________
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>
>
>
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>


From dwing@cisco.com  Fri Feb 15 10:57:42 2013
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From: "Dan Wing" <dwing@cisco.com>
To: "'Emil Ivov'" <emcho@jitsi.org>, "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
References: <511CEB64.6070306@ericsson.com> <05a401ce0ac6$db547230$91fd5690$@cisco.com> <511D03A7.1060008@ericsson.com> <511D0498.3030909@jitsi.org> <511D07F1.3070808@ericsson.com> <511E6696.6090609@ericsson.com> <511E6D72.7070206@jitsi.org>
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Date: Fri, 15 Feb 2013 10:57:36 -0800
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Cc: 'Magnus Westerlund' <magnus.westerlund@ericsson.com>, =?iso-8859-1?Q?'Ari_Ker=E4nen'?= <ari.keranen@ericsson.com>, 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: Friday, February 15, 2013 9:17 AM
> To: Gonzalo Camarillo
> Cc: Magnus Westerlund; Ari Ker=E4nen; 'mmusic'; Dan Wing
> Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
>=20
> Well for one, we hadn't even noticed that there was different pacing =
for
> non-RTP traffic, so when we added support for Pseudo TCP to ice4j.org,
> we just let it use the default timers. It really didn't cross our
> collective minds that the same stack in the same application would =
have
> the right of going down to 20ms if it used the magic word (and RTP
> header) but then go back up to 500 ms if it didn't. It's almost as if =
it
> were punished for having bad non-RTP thoughts ;)
>=20
> Anyways, I think that, had we known about the "MUST NOT drop below 500
> ms" limitation, we would have simply ignored it: these are user
> perceptible delays after all, and no one would like to wayt 10+ =
seconds
> for a file to start transferring.

A quick a analysis of what browsers have been doing in this area:  The=20
TCP retransmit timer on most OSs is 3 seconds.  But Firefox and Chrome=20
have backup connection timers that are more aggressive (250ms and 300ms,
if I recall correctly).  IE does not have a backup connection timer.  So
they are not as aggressive as 20ms (which would be too aggressive on a =
lot
of networks) but they also do not strictly follow the IETF TCP transport
guidelines to retry a TCP connection after a long 3 seconds.

-d


> Cheers,
> Emil
>=20
> On 15.02.13, 17:47, Gonzalo Camarillo wrote:
> > Hi,
> >
> > to add to what Magnus said, it is important that we understand what
> > implementations have been doing so far (regardless of what the RFC
> > says) and that we come up with recommendations implementers will
> > actually follow. Keep in mind we are here to write specifications, =
not
> > fiction ;-)
> >
> > Thanks,
> >
> > Gonzalo
> >
> >
> > On 14/02/2013 5:51 PM, Magnus Westerlund wrote:
> >> Thus I would be careful with speeding up Ta for RTP to much, and I
> >> think someone needs to fine empiric data that it would work.
> >>
> >> For the the 500 ms that can likely be significantly lowered without
> >> any issues. But some considerations around the wider usage needs to
> >> be taken into account.
> >
> >
>=20
> --
> https://jitsi.org


From martin.thomson@gmail.com  Fri Feb 15 11:41:36 2013
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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On 15 February 2013 10:57, Dan Wing <dwing@cisco.com> wrote:
> A quick a analysis of what browsers have been doing in this area:  The
> TCP retransmit timer on most OSs is 3 seconds.  But Firefox and Chrome
> have backup connection timers that are more aggressive (250ms and 300ms,
> if I recall correctly).  IE does not have a backup connection timer.  So
> they are not as aggressive as 20ms (which would be too aggressive on a lot
> of networks) but they also do not strictly follow the IETF TCP transport
> guidelines to retry a TCP connection after a long 3 seconds.

That's good info.  I could only find that for Firefox easily.  Given
that they also ignore the limit Dale pointed out and create 6
connections to any given server, this leads to an aggregate rate of
SYN packets of one every 50ms, though it's probably more bursty than
the even pacing required by RFC 5245.

From adam@nostrum.com  Fri Feb 15 13:10:59 2013
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Call about Bundle Ports Feb 21
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During SIPit?

/a

On 2/15/13 10:36, Cullen Jennings (fluffy) wrote:
>
> Hi All,
>
> As discussed at the interim, Richard and I are planning to have a call 
> to discuss issues around the same port or different ports in bundle. 
> This is not an official WG call or anything but anyone is welcome to 
> join us on the call. Bridge info below.
>
> Cullen
>
>
>
>>
>> Cullen Jennings invites you to attend this online meeting.
>>
>> Topic: Bundle Ports
>> Date: Thursday, February 21, 2013
>> Time: 7:00 am, Pacific Standard Time (San Francisco, GMT-08:00)
>> Meeting Number: 207 772 340
>> Meeting Password: ietf
>>
>>
>> -------------------------------------------------------
>> To join the online meeting (Now from mobile devices!)
>> -------------------------------------------------------
>> 1. Go to 
>> https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&UID=0&PW=NMjYwN2JjNzYz&RT=MiM0 
>>
>> 2. Enter your name and email address.
>> 3. Enter the meeting password: ietf
>> 4. Click "Join Now".
>>
>> To view in other time zones or languages, please click the link:
>> https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&UID=0&PW=NMjYwN2JjNzYz&ORT=MiM0 
>>
>>
>> ----------------------------------------------------------------
>> ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
>> ----------------------------------------------------------------
>>
>> The affected toll free numbers are: (866) 432-9903 for the San 
>> Jose/Milpitas area and (866) 349-3520 for the RTP area.
>>
>> Please dial the local access number for your area from the list below:
>> - San Jose/Milpitas (408) area: 525-6800
>> - RTP (919) area: 392-3330
>>
>> -------------------------------------------------------
>> To join the teleconference only
>> -------------------------------------------------------
>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
>> 2. Follow the prompts to enter the Meeting Number (listed above) or 
>> Access Code followed by the # sign.
>>
>> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
>>
>> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
>>
>> India: +91.80.4350.1111 Germany: +49.619.6773.9002
>>
>> Japan: +81.3.5763.9394 China: +86.10.8515.5666
>>
>> -------------------------------------------------------
>>
>>
>> To add this meeting to your calendar program (for example Microsoft 
>> Outlook), click this link:
>> https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&UID=0&ICS=MI&LD=1&RD=2&ST=1&SHA2=AAAAAkvkBwPvL1IZY/a2pGU6xcxB7i5WW2-fXSdiMwm3e2Gw&RT=MiM0 
>>
>>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">During SIPit?<br>
      <br>
      /a<br>
      <br>
      On 2/15/13 10:36, Cullen Jennings (fluffy) wrote:<br>
    </div>
    <blockquote
cite="mid:C5E08FE080ACFD4DAE31E4BDBF944EB1133E51C1@xmb-aln-x02.cisco.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div><br>
      </div>
      Hi All,
      <div><br>
      </div>
      <div>As discussed at the interim, Richard and I are planning to
        have a call to discuss issues around the same port or different
        ports in bundle. This is not an official WG call or anything but
        anyone is welcome to join us on the call. Bridge info below.&nbsp;</div>
      <div><br>
      </div>
      <div>Cullen</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div>
        <div>
          <blockquote type="cite"><font face="Tahoma, Arial, sans-serif,
              Helvetica, Geneva" size="2"><br>
              Cullen Jennings invites you to attend this online meeting.
              <br>
              <br>
              Topic: Bundle Ports <br>
              Date: Thursday, February 21, 2013 <br>
              Time: 7:00 am, Pacific Standard Time (San Francisco,
              GMT-08:00) <br>
              Meeting Number: 207 772 340 <br>
              Meeting Password: ietf <br>
              <br>
              <br>
              ------------------------------------------------------- <br>
              To join the online meeting (Now from mobile devices!) <br>
              ------------------------------------------------------- <br>
              1. Go to <a moz-do-not-send="true"
href="https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&amp;UID=0&amp;PW=NMjYwN2JjNzYz&amp;RT=MiM0"
                target="_blank">
https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&amp;UID=0&amp;PW=NMjYwN2JjNzYz&amp;RT=MiM0</a>
              <br>
              2. Enter your name and email address. <br>
              3. Enter the meeting password: ietf <br>
              4. Click "Join Now". <br>
              <br>
              To view in other time zones or languages, please click the
              link: <br>
              <a moz-do-not-send="true"
href="https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&amp;UID=0&amp;PW=NMjYwN2JjNzYz&amp;ORT=MiM0"
                target="_blank">https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&amp;UID=0&amp;PW=NMjYwN2JjNzYz&amp;ORT=MiM0</a>
              <br>
              <br>
              ----------------------------------------------------------------
              <br>
              ALERT:Toll-Free Dial Restrictions for (408) and (919) Area
              Codes <br>
              ----------------------------------------------------------------
              <br>
              <br>
              The affected toll free numbers are: (866) 432-9903 for the
              San Jose/Milpitas area and (866) 349-3520 for the RTP
              area.
              <br>
              <br>
              Please dial the local access number for your area from the
              list below: <br>
              - San Jose/Milpitas (408) area: 525-6800 <br>
              - RTP (919) area: 392-3330 <br>
              <br>
              ------------------------------------------------------- <br>
              To join the teleconference only <br>
              ------------------------------------------------------- <br>
              1. Dial into Cisco WebEx (view all Global Access Numbers
              at <br>
              <a moz-do-not-send="true"
href="http://cisco.com/en/US/about/doing_business/conferencing/index.html"
                target="_blank">http://cisco.com/en/US/about/doing_business/conferencing/index.html</a>
              <br>
              2. Follow the prompts to enter the Meeting Number (listed
              above) or Access Code followed by the # sign.
              <br>
              <br>
              San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330 <br>
              <br>
              US/Canada: +1.866.432.9903 United Kingdom:
              +44.20.8824.0117 <br>
              <br>
              India: +91.80.4350.1111 Germany: +49.619.6773.9002 <br>
              <br>
              Japan: +81.3.5763.9394 China: +86.10.8515.5666 <br>
              <br>
              ------------------------------------------------------- <br>
              <br>
              <br>
              To add this meeting to your calendar program (for example
              Microsoft Outlook), click this link:
              <br>
              <a moz-do-not-send="true"
href="https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&amp;UID=0&amp;ICS=MI&amp;LD=1&amp;RD=2&amp;ST=1&amp;SHA2=AAAAAkvkBwPvL1IZY/a2pGU6xcxB7i5WW2-fXSdiMwm3e2Gw&amp;RT=MiM0"
                target="_blank">https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&amp;UID=0&amp;ICS=MI&amp;LD=1&amp;RD=2&amp;ST=1&amp;SHA2=AAAAAkvkBwPvL1IZY/a2pGU6xcxB7i5WW2-fXSdiMwm3e2Gw&amp;RT=MiM0</a>
              <br>
              <br>
            </font></blockquote>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------070702040402030806020106--

From worley@shell01.TheWorld.com  Fri Feb 15 13:28:02 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: mmusic@ietf.org
Subject: [MMUSIC] Some comments on draft-ietf-mmusic-sctp-sdp-03
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* section 4.3

The following example is given:

           m=application 54111 DTLS/SCTP 5000 5001 5002
           c=IN IP4 79.97.215.79
           a=sctpmap:5000 webrtc-DataChannel 2
           a=sctpmap:5001 bfcp 1
           a=sctpmap:5002 t38 1
           a=webrtc-DataChannel:5000 stream=1;label="channel 1";subprotocol="chat";
           a=webrtc-DataChannel:5000 stream=2;label="channel 2";subprotocol="file transfer";max_retr=3

This example is missing the a=fingerprint attribute, which is
mandatory according to section 3.

This specifies that SCTP-over-DTLS-over-DCCP will be used, with the
near-end DCCP port being 54111.  However, the numbers "5000", "5001",
and "5002" seem to have no function, in that they appear nowhere on
the wire.  At least one "format" number is syntactically required in
the m= line, but there seems to be no reason to provide multiple
format numbers.

A more natural organization of the attributes for DTLS/SCTP use seems
to be this, which identifies the component media by their SCTP stream
identifiers (which is how they will be demultiplexed):

           m=application 54111 DTLS/SCTP 0
           c=IN IP4 79.97.215.79
           a=sctpmap:webrtc-DataChannel 2
           a=sctpmap:bfcp 1
           a=sctpmap:t38 1
           a=webrtc-DataChannel:5000 stream=1;label="channel 1";subprotocol="chat";
           a=webrtc-DataChannel:5000 stream=2;label="channel 2";subprotocol="file transfer";max_retr=3

Or perhaps use the SCTP stream identifiers could be used as the
primary identifiers in the a=webrtc-DataChannel attributes:

           a=webrtc-DataChannel:1;label="channel 1";subprotocol="chat";
           a=webrtc-DataChannel:2;label="channel 2";subprotocol="file transfer";max_retr=3

It's also not clear why the BFCP and T.38 media are assigned the same
SCCP stream identifier (i.e., 1).  That seems to be a mistake.

* section 5

There are a number of attributes with the generic format
"a=<name>=<value>".  I believe this should be "a=<name>:<value>".

         stream-attr       =  "a=stream=" streamnumber
         label-attr       =  "a=label=" labelstring
         subprotocol-attr       =  "a=subprotocol=" labelstring
         maxretr-attr       =  "a=maxretr=" maxretrvalue
         maxtime-attr       =  "a=maxtime=" maxtimevalue

* section 5.3

This section defines an apparently new "a=label" attribute:

         label-attr       =  "a=label=" labelstring
         labelstring      =  text
         text             =  byte-string

But the label syntax is incompatible with the existing label
attribute:

    RFC 4574 section 4

      label-attribute    = "a=label:" pointer
      pointer            = token
      token              = 1*(token-char)
      token-char         = %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                           / %x41-5A / %x5E-7E

And it's not clear why this "new" attribute is defined at all, since
there is a label carried in the a=webrtc-DataChannel attribute
already.

Dale

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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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> From: Harald Alvestrand <harald@alvestrand.no>
> 
> RTCWEB has a requirement (or rather: it has assumed that there was a 
> solution) for signalling over SDP that SCTP/DTLS/UDP is sent over the 
> same port pair as SRTP.

More exactly, RTCWEB has an assumption that there is a strategy for
sending SCTP/DTLS/UDP over the same port as SRTP, and that we will
figure out how to signal it.  (It appears that sending RTP and RTCP
(or SRTP and SRTP) over the same port is already solved.)

The question is whether this is a *requirement*, and if so, to put it
in the requirements document -- I don't see it in
draft-ietf-rtcweb-use-cases-and-requirements-10.

However what I do see in that draft is:

   F29             The browser MUST be able to send streams and
                   data to a peer in the presence of NATs that
                   block UDP traffic.

and we seem to be assuming that the carrier protocol for multiplexing
is UDP...

Dale

From worley@shell01.TheWorld.com  Fri Feb 15 14:24:08 2013
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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The problem is divisible into two parts.  One part is just clarifying
what needs to be supported.  The second, harder part is specifying all
the mechanisms.

When multiplexing, we have to establish:

1. What is the nature of the constituent data streams that can be
   multiplexed?

2. What method is used to multiplex the constituents?  Does there need
   to be an encapsulation protocol?

3. What method is used to demultiplex the incoming PDUs of the
   multiplexed data?

4. What carrier protocol is used to carry the multiplexed data?

5. How will we signal whatever information is needed to control the
   demultiplexing process?  (This is affected by any need to
   interoperate with existing systems.)

6. Where in the stack of protocols do we apply encryption, and what is
   its nature?

Items 1 and 3 are the ones that are most constrained by the
requirements of the situation.

As far as I can tell from what I've seen, the nature of the
constituent data streams is:

- the RTP or SRTP traffic arising from zero or more SDP media
  descriptions (Here we assume that it is important when
  demultiplexing RTP that it is important to identify the correct
  constituent media description for each incoming RTP packet; that RTP
  cannot be interpreted except in the context of the correct media
  description.)

- the corresponding RTCP or Secure RTCP traffic (assuming we want to
  have RTP/RTCP multiplexing)

- the data carried over a single SCTP connection (which includes a
  number of SCTP streams), or possibly more than one SCTP connection
  if that is to be supported (It is not clear if the multiplexing
  process has access to the data in the individual SCTP streams, or
  whether the constituent data stream is the packets generated by
  SCTP's multiplexing process, or whether it must handle the packets
  generated by DTLS encrypting the SCTP packets.)

The carrier protocol seems to be assumed to be UDP supplemented by the
STUN and ICE traffic needed to support a UDP association with the
far-end.  However there may be value in using SCTP-over-DTLS-over-UDP
as the carrier protocol.

In regard to encryption, it would be more efficient to apply
encryption only to the multiplexed data.  That suggests that the
constituent data is unencrypted: RTP instead of SRTP, SCTP instead of
DTLS/SCTP.

There tends to be an assumption that the demultiplexing process is
trivial, that all incoming PDUs can be identified and routed to their
correct destinations without any additional encapsulation
information.  We might be lucky enough that that is true, but it is
not the common case.

Dale

From ron.even.tlv@gmail.com  Fri Feb 15 22:45:18 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Adam Roach'" <adam@nostrum.com>
References: <CAOJ7v-3Ndb7vz9gchQ0Y308uH4nFUrKv8AkmeNrdf-3F03MLQg@mail.gmail.com>	<5111247B.8060904@nostrum.com> <511CEDFF.7060001@ericsson.com>
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Date: Sat, 16 Feb 2013 08:41:54 +0200
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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Hi,
I fully agree with Magnus cases and I think they should be addressed.=20
At CLUE when working on RTP mapping we addressed this used cases by =
having
multiple RTP streams in a single m-line
Roni Even

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of
Magnus Westerlund
Sent: 14 February, 2013 4:01 PM
To: Adam Roach
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] More thoughts on multiple media streams in a =
single m=3D
line

On 2013-02-05 16:25, Adam Roach wrote:
> On 2/4/13 02:45, Justin Uberti wrote:
>> Sorry for the short lead time, but here's a quick read for the plane.
>=20
> I addressed several of the assertions made in this draft in my=20
> previous response (e.g. the need for glare resolution regardless of=20
> which solution we select). Some additional thoughts:
>=20
>>    o  Adding or removing a source requires sending a full list of all
>>       sources, which doesn't scale well to 10s or 100s of sources,
>>       especially when they are added or removed frequently.
>=20
> Realistically, this isn't how multiparty conferences are handled. It's =

> unrealistic to think that my DSL connection is going to be sucking=20
> down hundreds of video streams just because I'm in a video conference=20
> with hundreds of people.
>=20
> In practice, the way existing conferencing systems handle this is to=20
> send a small, manageable number of streams to each participant,=20
> changing them out as necessary to have a reasonable user experience=20
> (either under the control of the user, its own algorithms, or a
combination of the two).
>=20
> Which is to say: even with hundreds of source streams being fed into a =

> conference system, you're really looking at a realistic upper bound on =

> m=3D sections of a dozen or two, simply because you can't reasonably=20
> expect recipients to receive more than that many streams =
simultaneously.
>=20

Hi,

I definitely want to comment on this from an RTP usage perspective that
might be very relevant to remember. Even with a centralized conference
solution there is two very distinct solutions in RTP how to transmit a
session where you have total of 100 participants and you receive 5 video
streams.

A) Central point with conceptual sources In this case the central entity
creates conceptual RTP streams that carries all the media intended for a
specific usage, for example main video, thumbnail one, thumbnail two,
thumbnail three, thumbnail four.

B) Central point doing source projection In this case there exist 100
potential RTP streams and SSRC from the central point down to the =
receiver.
However, at any point no more than five will actually have RTP packets. =
But
as activity changes and which is most relevant for the receiver the =
SSRCs in
use will switch.

For a solution where each SSRC requires its own m=3D line A) is =
definitely
preferable as it will not require renegotiating the properties for these
five conceptual streams. However, B will require a large number of m=3D =
lines
and dynamic updates as participants comes and go.

IF one had a single m=3D line per RTP session and media type, the amount =
of
updates/renegotiation depends on the amount of meta data about streams =
that
you have put in the SDP. They solutions A) and B) can be identical from =
an
SDP perspective.

The above two cases I am quite certain we did discuss in the CLUE =
interim
meeting in Stockholm when discussing topologies and also the relation =
ship
between RTP streams (SSRC) and their origin, i.e. media source/capture.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Adam Roach'" <adam@nostrum.com>
Date: Sat, 16 Feb 2013 14:03:43 +0200
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line - requirements for "fumble"!!!
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Hi,
The second topology brings the following requirement that I assume also
relate to what is called "fumble"

"it should be possible to support receipt of multiple RTP sources =
without
explicit per-source signaling or negotiation."

Roni Even


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of
Magnus Westerlund
Sent: 14 February, 2013 4:01 PM
To: Adam Roach
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] More thoughts on multiple media streams in a =
single m=3D
line

On 2013-02-05 16:25, Adam Roach wrote:
> On 2/4/13 02:45, Justin Uberti wrote:
>> Sorry for the short lead time, but here's a quick read for the plane.
>=20
> I addressed several of the assertions made in this draft in my=20
> previous response (e.g. the need for glare resolution regardless of=20
> which solution we select). Some additional thoughts:
>=20
>>    o  Adding or removing a source requires sending a full list of all
>>       sources, which doesn't scale well to 10s or 100s of sources,
>>       especially when they are added or removed frequently.
>=20
> Realistically, this isn't how multiparty conferences are handled. It's =

> unrealistic to think that my DSL connection is going to be sucking=20
> down hundreds of video streams just because I'm in a video conference=20
> with hundreds of people.
>=20
> In practice, the way existing conferencing systems handle this is to=20
> send a small, manageable number of streams to each participant,=20
> changing them out as necessary to have a reasonable user experience=20
> (either under the control of the user, its own algorithms, or a
combination of the two).
>=20
> Which is to say: even with hundreds of source streams being fed into a =

> conference system, you're really looking at a realistic upper bound on =

> m=3D sections of a dozen or two, simply because you can't reasonably=20
> expect recipients to receive more than that many streams =
simultaneously.
>=20

Hi,

I definitely want to comment on this from an RTP usage perspective that
might be very relevant to remember. Even with a centralized conference
solution there is two very distinct solutions in RTP how to transmit a
session where you have total of 100 participants and you receive 5 video
streams.

A) Central point with conceptual sources In this case the central entity
creates conceptual RTP streams that carries all the media intended for a
specific usage, for example main video, thumbnail one, thumbnail two,
thumbnail three, thumbnail four.

B) Central point doing source projection In this case there exist 100
potential RTP streams and SSRC from the central point down to the =
receiver.
However, at any point no more than five will actually have RTP packets. =
But
as activity changes and which is most relevant for the receiver the =
SSRCs in
use will switch.

For a solution where each SSRC requires its own m=3D line A) is =
definitely
preferable as it will not require renegotiating the properties for these
five conceptual streams. However, B will require a large number of m=3D =
lines
and dynamic updates as participants comes and go.

IF one had a single m=3D line per RTP session and media type, the amount =
of
updates/renegotiation depends on the amount of meta data about streams =
that
you have put in the SDP. They solutions A) and B) can be identical from =
an
SDP perspective.

The above two cases I am quite certain we did discuss in the CLUE =
interim
meeting in Stockholm when discussing topologies and also the relation =
ship
between RTP streams (SSRC) and their origin, i.e. media source/capture.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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


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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [MMUSIC] Call about Bundle Ports Feb 21
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At this point, it's going to be difficult for several of the CLUE WG
participants to attend at this point.  A doodle would be nice to
ensure you have a reasonable representation from the groups involved.

Thanks,
Mary.

On Fri, Feb 15, 2013 at 10:36 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com> wrote:
>
> Hi All,
>
> As discussed at the interim, Richard and I are planning to have a call to
> discuss issues around the same port or different ports in bundle. This is
> not an official WG call or anything but anyone is welcome to join us on the
> call. Bridge info below.
>
> Cullen
>
>
>
>
> Cullen Jennings invites you to attend this online meeting.
>
> Topic: Bundle Ports
> Date: Thursday, February 21, 2013
> Time: 7:00 am, Pacific Standard Time (San Francisco, GMT-08:00)
> Meeting Number: 207 772 340
> Meeting Password: ietf
>
>
> -------------------------------------------------------
> To join the online meeting (Now from mobile devices!)
> -------------------------------------------------------
> 1. Go to
> https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&UID=0&PW=NMjYwN2JjNzYz&RT=MiM0
> 2. Enter your name and email address.
> 3. Enter the meeting password: ietf
> 4. Click "Join Now".
>
> To view in other time zones or languages, please click the link:
> https://ciscosales.webex.com/ciscosales/j.php?ED=217553672&UID=0&PW=NMjYwN2JjNzYz&ORT=MiM0
>
> ----------------------------------------------------------------
> ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
> ----------------------------------------------------------------
>
> The affected toll free numbers are: (866) 432-9903 for the San Jose/Milpitas
> area and (866) 349-3520 for the RTP area.
>
> Please dial the local access number for your area from the list below:
> - San Jose/Milpitas (408) area: 525-6800
> - RTP (919) area: 392-3330
>
> -------------------------------------------------------
> To join the teleconference only
> -------------------------------------------------------
> 1. Dial into Cisco WebEx (view all Global Access Numbers at
> http://cisco.com/en/US/about/doing_business/conferencing/index.html
> 2. Follow the prompts to enter the Meeting Number (listed above) or Access
> Code followed by the # sign.
>
> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
>
> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
>
> India: +91.80.4350.1111 Germany: +49.619.6773.9002
>
> Japan: +81.3.5763.9394 China: +86.10.8515.5666
>
> -------------------------------------------------------
>
>
> To add this meeting to your calendar program (for example Microsoft
> Outlook), click this link:
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>
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line
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Comments below.=20

> I definitely want to comment on this from an RTP usage perspective that
> might be very relevant to remember. Even with a centralized conference
> solution there is two very distinct solutions in RTP how to transmit a
> session where you have total of 100 participants and you receive 5 video
> streams.
>=20
> A) Central point with conceptual sources
> In this case the central entity creates conceptual RTP streams that
> carries all the media intended for a specific usage=2C for example main
> video=2C thumbnail one=2C thumbnail two=2C thumbnail three=2C thumbnail f=
our.
>=20
> B) Central point doing source projection
> In this case there exist 100 potential RTP streams and SSRC from the
> central point down to the receiver. However=2C at any point no more than
> five will actually have RTP packets. But as activity changes and which
> is most relevant for the receiver the SSRCs in use will switch.
>=20
> For a solution where each SSRC requires its own m=3D line A) is definitel=
y
> preferable as it will not require renegotiating the properties for these
> five conceptual streams. However=2C B will require a large number of m=3D
> lines and dynamic updates as participants comes and go.

[BA] I think there are advantages to A even if multiple SSRCs are declared =
within a m-line.  If the "main" video and "thumbnails" change SSRCs=2C even=
 with multiple SSRCs on an m-line=2C you'd still need to renegotiate unless=
 SVC is being used=2C right?=20
=20
> If one had a single m=3D line per RTP session and media type=2C the amoun=
t
> of updates/renegotiation depends on the amount of meta data about
> streams that you have put in the SDP. The solutions A) and B) can be
> identical from an SDP perspective.

[BA] That's not obvious to me. Even if you have a single m=3Dline per RTP s=
ession and media type=2C=20
you can still have individual a=3Dssrc lines within a m=3D line.  In case B=
=2C as participants come and go=2C
aren't dynamic updates still required?  Also=2C instead of many m=3Dlines=
=2C you'll have many a=3Dssrc
lines.=20

> The above two cases I am quite certain we did discuss in the CLUE
> interim meeting in Stockholm when discussing topologies and also the
> relationship between RTP streams (SSRC) and their origin=2C i.e. media
> source/capture.

[BA] Were there any specific requirements that arose in the CLUE discussion=
?  In general=2C given some of the context issues=2C it would be helpful to=
 understand whether we're engineering a solution for RTCWEB use cases=2C CL=
UE use cases=2C or both.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Comments below. <br><br><div>&gt=
=3B I definitely want to comment on this from an RTP usage perspective that=
<br>&gt=3B might be very relevant to remember. Even with a centralized conf=
erence<br>&gt=3B solution there is two very distinct solutions in RTP how t=
o transmit a<br>&gt=3B session where you have total of 100 participants and=
 you receive 5 video<br>&gt=3B streams.<br>&gt=3B <br>&gt=3B A) Central poi=
nt with conceptual sources<br>&gt=3B In this case the central entity create=
s conceptual RTP streams that<br>&gt=3B carries all the media intended for =
a specific usage=2C for example main<br>&gt=3B video=2C thumbnail one=2C th=
umbnail two=2C thumbnail three=2C thumbnail four.<br>&gt=3B <br>&gt=3B B) C=
entral point doing source projection<br>&gt=3B In this case there exist 100=
 potential RTP streams and SSRC from the<br>&gt=3B central point down to th=
e receiver. However=2C at any point no more than<br>&gt=3B five will actual=
ly have RTP packets. But as activity changes and which<br>&gt=3B is most re=
levant for the receiver the SSRCs in use will switch.<br>&gt=3B <br>&gt=3B =
For a solution where each SSRC requires its own m=3D line A) is definitely<=
br>&gt=3B preferable as it will not require renegotiating the properties fo=
r these<br>&gt=3B five conceptual streams. However=2C B will require a larg=
e number of m=3D<br>&gt=3B lines and dynamic updates as participants comes =
and go.<br><br>[BA] I think there are advantages to A even if multiple SSRC=
s are declared within a m-line.&nbsp=3B If the "main" video and "thumbnails=
" change SSRCs=2C even with multiple SSRCs on an m-line=2C you'd still need=
 to renegotiate unless SVC is being used=2C right? <br>&nbsp=3B<br>&gt=3B I=
f one had a single m=3D line per RTP session and media type=2C the amount<b=
r>&gt=3B of updates/renegotiation depends on the amount of meta data about<=
br>&gt=3B streams that you have put in the SDP. The solutions A) and B) can=
 be<br>&gt=3B identical from an SDP perspective.<br><br>[BA] That's not obv=
ious to me. Even if you have a single m=3Dline per RTP session and media ty=
pe=2C <br>you can still have individual a=3Dssrc lines within a m=3D line.&=
nbsp=3B In case B=2C as participants come and go=2C<br>aren't dynamic updat=
es still required?&nbsp=3B Also=2C instead of many m=3Dlines=2C you'll have=
 many a=3Dssrc<br>lines. <br><br>&gt=3B The above two cases I am quite cert=
ain we did discuss in the CLUE<br>&gt=3B interim meeting in Stockholm when =
discussing topologies and also the<br>&gt=3B relationship between RTP strea=
ms (SSRC) and their origin=2C i.e. media<br>&gt=3B source/capture.<br><br>[=
BA] Were there any specific requirements that arose in the CLUE discussion?=
&nbsp=3B In general=2C given some of the context issues=2C it would be help=
ful to understand whether we're engineering a solution for RTCWEB use cases=
=2C CLUE use cases=2C or both. <br></div> 		 	   		  </div></body>
</html>=

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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line - requirements for "fumble"!!!
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[BA]  If A/V multiplexing is negotiated=2C then the only RTP de-multiplexin=
g points you have left are the SSRC and the Payload type.  If you rule out =
"explicit per-source signaling"=2C that rules out SSRC de-multiplexing=2C w=
hich leaves you only with Payload Type.  Am I missing something?=20

I do agree that without A/V multiplexing=2C it is possible to avoid explici=
t per-source signaling=2C since we can use the RTP session to de-multiplex=
=2C so that we're still ok.  But then we're not talking about Bundle/Grumbl=
e/Fumble/Humble=2C right?=20

> The second topology brings the following requirement that I assume also
> relate to what is called "fumble"
>=20
> "it should be possible to support receipt of multiple RTP sources without
> explicit per-source signaling or negotiation."
>=20
> Roni Even

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>[BA]&nbsp=3B If A/V multiplexing=
 is negotiated=2C then the only RTP de-multiplexing points you have left ar=
e the SSRC and the Payload type.&nbsp=3B If you rule out "explicit per-sour=
ce signaling"=2C that rules out SSRC de-multiplexing=2C which leaves you on=
ly with Payload Type.&nbsp=3B Am I missing something? <br><br>I do agree th=
at without A/V multiplexing=2C it is possible to avoid explicit per-source =
signaling=2C since we can use the RTP session to de-multiplex=2C so that we=
're still ok.&nbsp=3B But then we're not talking about Bundle/Grumble/Fumbl=
e/Humble=2C right? <br><br><div>&gt=3B The second topology brings the follo=
wing requirement that I assume also<br>&gt=3B relate to what is called "fum=
ble"<br>&gt=3B <br>&gt=3B "it should be possible to support receipt of mult=
iple RTP sources without<br>&gt=3B explicit per-source signaling or negotia=
tion."<br>&gt=3B <br>&gt=3B Roni Even<br><br></div> 		 	   		  </div></body=
>
</html>=

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [MMUSIC] BUNDLE: RTP only?
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Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Sort of - I think the right question is can SCTP over DTLS over UDP demux f=
rom SRTP over UDP and from STUN over UDP and the answer to that is yes.=20


On Feb 14, 2013, at 12:53 PM, Dale R. Worley <worley@ariadne.com> wrote:

>> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
>>=20
>> *If* there is a goal to share a single 5-tuple between RTP and SCTP then=
=20
>> we need a solution allows bundling them.
>=20
> Do any of the current proposed bundling techniques include both RTP
> and SCTP?
>=20
> Dale
> _______________________________________________
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> mmusic@ietf.org
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sat, 16 Feb 2013 16:07:28 -0800
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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Dale,

I had assumed that the layering diagram looked as follows:

     SCTP
     DTLS    SRT(C)P       STUN
               UDP


This means that the demux problem becomes something like

- If packet is STUN, forward to ICE stack.
- If packet is SRT(C)P:
  + Primary demux on payload type
  + Secondary demux on SSRC
- If packet is DTLS
  + Demux SCTP associations by port number
    . SCTP streams are demuxed by stream ID

1. The first level of demux is described in RFC 5764 S 5.1.2.
2. The SRT(C)P demux appears to be specified in:
   http://tools.ietf.org/html/draft-ietf-avtcore-multi-media-rtp-session-01
3. SCTP over DTLS encapsulation (and the elevant muxing) appears to
   be specified in:
   http://tools.ietf.org/id/draft-tuexen-tsvwg-sctp-dtls-encaps-01.txt
   and since the port is part of SCTP, this seems straightforward
   if we wish to allow it.


Obviously, one could, as you suggest, tunnel everything over SCTP,
but this seems like it would be less efficient in terms of bandwidth
(since you would add a bunch of headers to the RTP) and would also
be less compatible with existing SRTP endpoints.

Best,
-Ekr










On Fri, Feb 15, 2013 at 2:22 PM, Dale R. Worley <worley@ariadne.com> wrote:

> The problem is divisible into two parts.  One part is just clarifying
> what needs to be supported.  The second, harder part is specifying all
> the mechanisms.
>
> When multiplexing, we have to establish:
>
> 1. What is the nature of the constituent data streams that can be
>    multiplexed?
>
> 2. What method is used to multiplex the constituents?  Does there need
>    to be an encapsulation protocol?
>
> 3. What method is used to demultiplex the incoming PDUs of the
>    multiplexed data?
>
> 4. What carrier protocol is used to carry the multiplexed data?
>
> 5. How will we signal whatever information is needed to control the
>    demultiplexing process?  (This is affected by any need to
>    interoperate with existing systems.)
>
> 6. Where in the stack of protocols do we apply encryption, and what is
>    its nature?
>
> Items 1 and 3 are the ones that are most constrained by the
> requirements of the situation.
>
> As far as I can tell from what I've seen, the nature of the
> constituent data streams is:
>
> - the RTP or SRTP traffic arising from zero or more SDP media
>   descriptions (Here we assume that it is important when
>   demultiplexing RTP that it is important to identify the correct
>   constituent media description for each incoming RTP packet; that RTP
>   cannot be interpreted except in the context of the correct media
>   description.)
>
> - the corresponding RTCP or Secure RTCP traffic (assuming we want to
>   have RTP/RTCP multiplexing)
>
> - the data carried over a single SCTP connection (which includes a
>   number of SCTP streams), or possibly more than one SCTP connection
>   if that is to be supported (It is not clear if the multiplexing
>   process has access to the data in the individual SCTP streams, or
>   whether the constituent data stream is the packets generated by
>   SCTP's multiplexing process, or whether it must handle the packets
>   generated by DTLS encrypting the SCTP packets.)
>
> The carrier protocol seems to be assumed to be UDP supplemented by the
> STUN and ICE traffic needed to support a UDP association with the
> far-end.  However there may be value in using SCTP-over-DTLS-over-UDP
> as the carrier protocol.
>
> In regard to encryption, it would be more efficient to apply
> encryption only to the multiplexed data.  That suggests that the
> constituent data is unencrypted: RTP instead of SRTP, SCTP instead of
> DTLS/SCTP.
>
> There tends to be an assumption that the demultiplexing process is
> trivial, that all incoming PDUs can be identified and routed to their
> correct destinations without any additional encapsulation
> information.  We might be lucky enough that that is true, but it is
> not the common case.
>
> Dale
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

--047d7b6dc0bc40f6d204d5e067d1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Dale,</div><div><br></div><div>I had assumed that the layering diagram=
 looked as follows:</div><div><br></div><div>=A0 =A0 =A0SCTP</div><div>=A0 =
=A0 =A0DTLS =A0 =A0SRT(C)P =A0 =A0 =A0 STUN</div><div>=A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0UDP</div><div><br></div><div>

<br></div><div>This means that the demux problem becomes something like</di=
v><div><br></div><div>- If packet is STUN, forward to ICE stack.</div><div>=
- If packet is SRT(C)P:</div><div>=A0 + Primary demux on payload type</div>

<div>=A0 + Secondary demux on SSRC</div><div>- If packet is DTLS</div><div>=
=A0 + Demux SCTP associations by port number</div><div>=A0 =A0 . SCTP strea=
ms are demuxed by stream ID</div><div><br></div><div>1. The first level of =
demux is described in RFC 5764 S 5.1.2.</div>

<div>2. The SRT(C)P demux appears to be specified in:</div><div>=A0 =A0<a h=
ref=3D"http://tools.ietf.org/html/draft-ietf-avtcore-multi-media-rtp-sessio=
n-01">http://tools.ietf.org/html/draft-ietf-avtcore-multi-media-rtp-session=
-01</a></div>

<div>3. SCTP over DTLS encapsulation (and the elevant muxing) appears to</d=
iv><div>=A0 =A0be specified in:</div><div>=A0 =A0<a href=3D"http://tools.ie=
tf.org/id/draft-tuexen-tsvwg-sctp-dtls-encaps-01.txt">http://tools.ietf.org=
/id/draft-tuexen-tsvwg-sctp-dtls-encaps-01.txt</a></div>

<div>=A0 =A0and since the port is part of SCTP, this seems straightforward<=
/div><div>=A0 =A0if we wish to allow it.</div><div><br></div><div><br></div=
><div>Obviously, one could, as you suggest, tunnel everything over SCTP,</d=
iv>
<div>
but this seems like it would be less efficient in terms of bandwidth</div><=
div>(since you would add a bunch of headers to the RTP) and would also</div=
><div>be less compatible with existing SRTP endpoints.</div><div><br></div>

<div>Best,</div><div>-Ekr</div><div>=A0 =A0 =A0 =A0</div><div><br></div><di=
v><br></div><div><br></div><div><br></div><div><br></div><div>=A0 =A0</div>=
<div>=A0 =A0</div><div><br></div><br><div class=3D"gmail_quote">On Fri, Feb=
 15, 2013 at 2:22 PM, Dale R. Worley <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:worley@ariadne.com" target=3D"_blank">worley@ariadne.com</a>&gt;</span> w=
rote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">The problem is divisible into two parts. =A0=
One part is just clarifying<br>
what needs to be supported. =A0The second, harder part is specifying all<br=
>
the mechanisms.<br>
<br>
When multiplexing, we have to establish:<br>
<br>
1. What is the nature of the constituent data streams that can be<br>
=A0 =A0multiplexed?<br>
<br>
2. What method is used to multiplex the constituents? =A0Does there need<br=
>
=A0 =A0to be an encapsulation protocol?<br>
<br>
3. What method is used to demultiplex the incoming PDUs of the<br>
=A0 =A0multiplexed data?<br>
<br>
4. What carrier protocol is used to carry the multiplexed data?<br>
<br>
5. How will we signal whatever information is needed to control the<br>
=A0 =A0demultiplexing process? =A0(This is affected by any need to<br>
=A0 =A0interoperate with existing systems.)<br>
<br>
6. Where in the stack of protocols do we apply encryption, and what is<br>
=A0 =A0its nature?<br>
<br>
Items 1 and 3 are the ones that are most constrained by the<br>
requirements of the situation.<br>
<br>
As far as I can tell from what I&#39;ve seen, the nature of the<br>
constituent data streams is:<br>
<br>
- the RTP or SRTP traffic arising from zero or more SDP media<br>
=A0 descriptions (Here we assume that it is important when<br>
=A0 demultiplexing RTP that it is important to identify the correct<br>
=A0 constituent media description for each incoming RTP packet; that RTP<br=
>
=A0 cannot be interpreted except in the context of the correct media<br>
=A0 description.)<br>
<br>
- the corresponding RTCP or Secure RTCP traffic (assuming we want to<br>
=A0 have RTP/RTCP multiplexing)<br>
<br>
- the data carried over a single SCTP connection (which includes a<br>
=A0 number of SCTP streams), or possibly more than one SCTP connection<br>
=A0 if that is to be supported (It is not clear if the multiplexing<br>
=A0 process has access to the data in the individual SCTP streams, or<br>
=A0 whether the constituent data stream is the packets generated by<br>
=A0 SCTP&#39;s multiplexing process, or whether it must handle the packets<=
br>
=A0 generated by DTLS encrypting the SCTP packets.)<br>
<br>
The carrier protocol seems to be assumed to be UDP supplemented by the<br>
STUN and ICE traffic needed to support a UDP association with the<br>
far-end. =A0However there may be value in using SCTP-over-DTLS-over-UDP<br>
as the carrier protocol.<br>
<br>
In regard to encryption, it would be more efficient to apply<br>
encryption only to the multiplexed data. =A0That suggests that the<br>
constituent data is unencrypted: RTP instead of SRTP, SCTP instead of<br>
DTLS/SCTP.<br>
<br>
There tends to be an assumption that the demultiplexing process is<br>
trivial, that all incoming PDUs can be identified and routed to their<br>
correct destinations without any additional encapsulation<br>
information. =A0We might be lucky enough that that is true, but it is<br>
not the common case.<br>
<div class=3D"im"><br>
Dale<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
</div><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_b=
lank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote></div><br>

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Bernard Aboba'" <bernard_aboba@hotmail.com>
References: <01bf01ce0c3d$ad50dfe0$07f29fa0$@gmail.com> <BLU002-W129664B3416CC1EEE909973930D0@phx.gbl>
In-Reply-To: <BLU002-W129664B3416CC1EEE909973930D0@phx.gbl>
Date: Sun, 17 Feb 2013 10:37:57 +0200
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Subject: Re: [MMUSIC] More thoughts on multiple media streams in a single m= line - requirements for "fumble"!!!
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Hi,

The SSRCs will still be unique in the RTP session, I assume that what you
are asking is how do you know what this SSRC is. It is not only if it is
audio or video but may also apply to what is this audio and video is. For
example, in a video conferencing systems with three monitors and cameras,
how do we know where this SSRC should go. 

In CLUE we use a Capture ID that can be defined for each SSRC but can also
be carried as a RTP header extension to provide the information. So when a
new RTP stream with an SSRC that was not in the SDP it will have an RTP
header extension with the CaptureID allowing the receiver to know what the
stream is for.

 

Roni Even

 

From: Bernard Aboba [mailto:bernard_aboba@hotmail.com] 
Sent: 16 February, 2013 11:07 PM
To: Roni Even
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] More thoughts on multiple media streams in a single m=
line - requirements for "fumble"!!!

 

[BA]  If A/V multiplexing is negotiated, then the only RTP de-multiplexing
points you have left are the SSRC and the Payload type.  If you rule out
"explicit per-source signaling", that rules out SSRC de-multiplexing, which
leaves you only with Payload Type.  Am I missing something? 

I do agree that without A/V multiplexing, it is possible to avoid explicit
per-source signaling, since we can use the RTP session to de-multiplex, so
that we're still ok.  But then we're not talking about
Bundle/Grumble/Fumble/Humble, right? 

> The second topology brings the following requirement that I assume also
> relate to what is called "fumble"
> 
> "it should be possible to support receipt of multiple RTP sources without
> explicit per-source signaling or negotiation."
> 
> Roni Even


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The SSRCs will still be unique in the RTP session, I assume that what =
you are asking is how do you know what this SSRC is. It is not only if =
it is audio or video but may also apply to what is this audio and video =
is. For example, in a video conferencing systems with three monitors and =
cameras, how do we know where this SSRC should go. =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>In CLUE we use a Capture ID that can be defined for each SSRC but can =
also be carried as a RTP header extension to provide the information. So =
when a new RTP stream with an SSRC that was not in the SDP it will have =
an RTP header extension with the CaptureID allowing the receiver to know =
what the stream is for.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni Even<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Bernard Aboba [mailto:bernard_aboba@hotmail.com] <br><b>Sent:</b> 16 =
February, 2013 11:07 PM<br><b>To:</b> Roni Even<br><b>Cc:</b> =
mmusic@ietf.org<br><b>Subject:</b> RE: [MMUSIC] More thoughts on =
multiple media streams in a single m=3D line - requirements for =
&quot;fumble&quot;!!!<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-family:"Calibri","sans-serif"'>[BA]&nbsp; If A/V =
multiplexing is negotiated, then the only RTP de-multiplexing points you =
have left are the SSRC and the Payload type.&nbsp; If you rule out =
&quot;explicit per-source signaling&quot;, that rules out SSRC =
de-multiplexing, which leaves you only with Payload Type.&nbsp; Am I =
missing something? <br><br>I do agree that without A/V multiplexing, it =
is possible to avoid explicit per-source signaling, since we can use the =
RTP session to de-multiplex, so that we're still ok.&nbsp; But then =
we're not talking about Bundle/Grumble/Fumble/Humble, right? =
<o:p></o:p></span></p><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-family:"Calibri","sans-serif"'>&gt; The second topology =
brings the following requirement that I assume also<br>&gt; relate to =
what is called &quot;fumble&quot;<br>&gt; <br>&gt; &quot;it should be =
possible to support receipt of multiple RTP sources without<br>&gt; =
explicit per-source signaling or negotiation.&quot;<br>&gt; <br>&gt; =
Roni Even<o:p></o:p></span></p></div></div></div></body></html>
------=_NextPart_000_0234_01CE0CFA.DBF8DC40--


From fluffy@cisco.com  Sun Feb 17 11:52:57 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Thread-Topic: [MMUSIC] Call about Bundle Ports Feb 21
Thread-Index: AQHODIEYDx8V7J82L02mLqBV7r7eH5h+3FGA
Date: Sun, 17 Feb 2013 19:52:53 +0000
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Call about Bundle Ports Feb 21
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When Richard and I went throughout times that might work for Europe and US =
and our schedules it was pretty limited down to just two slots. I just pick=
ed one. It's an awful time - it's during SIPIT, it's in the middle of the d=
raft deadlines, it's pretty much not going to work for most people. That sa=
id, Richard and I wanted to talk and figured there was no reason not to hav=
e anyone else who was interested join in. It's not a WG meeting - clearly n=
o decisions will be made at it. The problem is, given we want to talk befor=
e the -01 deadline, and we want to talk after the -00 deadline so the new d=
rafts are out, there was really no good time.=20


On Feb 16, 2013, at 1:06 PM, Mary Barnes <mary.ietf.barnes@gmail.com> wrote=
:

> At this point, it's going to be difficult for several of the CLUE WG
> participants to attend at this point.  A doodle would be nice to
> ensure you have a reasonable representation from the groups involved.
>=20
> Thanks,
> Mary.
>=20
> On Fri, Feb 15, 2013 at 10:36 AM, Cullen Jennings (fluffy)
> <fluffy@cisco.com> wrote:
>>=20
>> Hi All,
>>=20
>> As discussed at the interim, Richard and I are planning to have a call t=
o
>> discuss issues around the same port or different ports in bundle. This i=
s
>> not an official WG call or anything but anyone is welcome to join us on =
the
>> call. Bridge info below.
>>=20
>> Cullen
>>=20
>>=20
>>=20
>>=20
>> Cullen Jennings invites you to attend this online meeting.
>>=20
>> Topic: Bundle Ports
>> Date: Thursday, February 21, 2013
>> Time: 7:00 am, Pacific Standard Time (San Francisco, GMT-08:00)
>> Meeting Number: 207 772 340
>> Meeting Password: ietf
>>=20
>>=20
>> -------------------------------------------------------
>> To join the online meeting (Now from mobile devices!)
>> -------------------------------------------------------
>> 1. Go to
>> https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&UID=3D0&PW=
=3DNMjYwN2JjNzYz&RT=3DMiM0
>> 2. Enter your name and email address.
>> 3. Enter the meeting password: ietf
>> 4. Click "Join Now".
>>=20
>> To view in other time zones or languages, please click the link:
>> https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&UID=3D0&PW=
=3DNMjYwN2JjNzYz&ORT=3DMiM0
>>=20
>> ----------------------------------------------------------------
>> ALERT:Toll-Free Dial Restrictions for (408) and (919) Area Codes
>> ----------------------------------------------------------------
>>=20
>> The affected toll free numbers are: (866) 432-9903 for the San Jose/Milp=
itas
>> area and (866) 349-3520 for the RTP area.
>>=20
>> Please dial the local access number for your area from the list below:
>> - San Jose/Milpitas (408) area: 525-6800
>> - RTP (919) area: 392-3330
>>=20
>> -------------------------------------------------------
>> To join the teleconference only
>> -------------------------------------------------------
>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
>> 2. Follow the prompts to enter the Meeting Number (listed above) or Acce=
ss
>> Code followed by the # sign.
>>=20
>> San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330
>>=20
>> US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117
>>=20
>> India: +91.80.4350.1111 Germany: +49.619.6773.9002
>>=20
>> Japan: +81.3.5763.9394 China: +86.10.8515.5666
>>=20
>> -------------------------------------------------------
>>=20
>>=20
>> To add this meeting to your calendar program (for example Microsoft
>> Outlook), click this link:
>> https://ciscosales.webex.com/ciscosales/j.php?ED=3D217553672&UID=3D0&ICS=
=3DMI&LD=3D1&RD=3D2&ST=3D1&SHA2=3DAAAAAkvkBwPvL1IZY/a2pGU6xcxB7i5WW2-fXSdiM=
wm3e2Gw&RT=3DMiM0
>>=20
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20


From worley@shell01.TheWorld.com  Mon Feb 18 07:06:04 2013
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: Eric Rescorla <ekr@rtfm.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE: RTP only?
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> From: Eric Rescorla <ekr@rtfm.com>
> 
> I had assumed that the layering diagram looked as follows:
> 
>      SCTP
>      DTLS    SRT(C)P       STUN
>                UDP

And though you don't state it explicitily, no additional framing info
on the second step.

But it seems to me that it's inefficient in that encryption is done
separately for every input stream.  Or is datagram encryption a lot
more clever than I think it is?

Dale

From internet-drafts@ietf.org  Mon Feb 18 11:49:44 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-bundle-negotiation-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Multiplexing Negotiation Using Session Description Proto=
col (SDP) Port Numbers
	Author(s)       : Christer Holmberg
                          Harald Tveit Alvestrand
                          Cullen Jennings
	Filename        : draft-ietf-mmusic-sdp-bundle-negotiation-03.txt
	Pages           : 19
	Date            : 2013-02-18

Abstract:
   This specification defines a new SDP Grouping Framework extension,
   "BUNDLE", that can be used with the Session Description Protocol
   (SDP) Offer/Answer mechanism to negotiate the usage of bundled media,
   which refers to the usage of a single 5-tuple for media associated
   with multiple SDP media descriptions ("m=3D" lines).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-bundle-negotiation

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-bundle-negotiation=
-03


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


From christer.holmberg@ericsson.com  Mon Feb 18 11:50:28 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03
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Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi,

I've submitted a new version (-03) of BUNDLE.

Based on discussions with Cullen, and exchange of the concerns we had with =
the different port alternatives (different-ports vs identical-port), this v=
ersion now describes a mechanism where both alternatives more or less are c=
ombined. When the first BUNDLE offer is sent, and it is now known whether t=
he answerer supports BUNDLE (or, even identical ports), different ports are=
 used (read: the one-rtp alterntaive). Once BUNDLE answer has been received=
, a new offer, with identical port numbers, is sent.

We think that having to send a second offer is not going to increase comple=
xity and delay, because in many use-cases subsequent offer(s) will be sent =
anyway. For example, RTCWEB entities will do it because of ICE.

Cullen has also been added as co-author.

Now, we do realize that there are still issues that have to be described, a=
nd hopefully we will be able to focus on those from now on, but our wish is=
 that this proposal will unite the different "port camps" :)

Regards,

Christer=

From ari.keranen@ericsson.com  Mon Feb 18 11:59:30 2013
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Subject: [MMUSIC] Fwd: New Version Notification for draft-keranen-mmusic-rfc5245bis-00.txt
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Hi all,

We have submitted a -00 version of the ICE-bis draft. This version is 
almost identical to RFC5245; only editorial changes (typo fixes, 
reference updates, etc.) have been done to give a proper basis for diffing.

A diff between the RFC5245 and the bis-00:
http://tools.ietf.org/rfcdiff?url2=draft-keranen-mmusic-rfc5245bis-00.txt

The -01 version (to be submitted in a week) will have the SDP split from 
the base draft as discussed at Atlanta.


Cheers,
Ari

-------- Original Message --------
A new version of I-D, draft-keranen-mmusic-rfc5245bis-00.txt
has been successfully submitted by Ari Keranen and posted to the
IETF repository.

Filename:	 draft-keranen-mmusic-rfc5245bis
Revision:	 00
Title:		 Interactive Connectivity Establishment (ICE): A Protocol for 
Network Address Translator (NAT) Traversal for Offer/Answer Protocols
Creation date:	 2013-02-18
Group:		 Individual Submission
Number of pages: 118
URL: 
http://www.ietf.org/internet-drafts/draft-keranen-mmusic-rfc5245bis-00.txt
Status: 
http://datatracker.ietf.org/doc/draft-keranen-mmusic-rfc5245bis
Htmlized: 
http://tools.ietf.org/html/draft-keranen-mmusic-rfc5245bis-00


Abstract:
    This document describes a protocol for Network Address Translator
    (NAT) traversal for UDP-based multimedia sessions established with
    the offer/answer model.  This protocol is called Interactive
    Connectivity Establishment (ICE).  ICE makes use of the Session
    Traversal Utilities for NAT (STUN) protocol and its extension,
    Traversal Using Relay NAT (TURN).  ICE can be used by any protocol
    utilizing the offer/answer model, such as the Session Initiation
    Protocol (SIP).

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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From internet-drafts@ietf.org  Mon Feb 18 15:10:41 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-traffic-class-for-sdp-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : The Session Description Protocol (SDP) 'trafficclass' At=
tribute
	Author(s)       : James Polk
                          Subha Dhesikan
                          Paul E. Jones
	Filename        : draft-ietf-mmusic-traffic-class-for-sdp-03.txt
	Pages           : 27
	Date            : 2013-02-18

Abstract:
   This document proposes a new Session Description Protocol (SDP)
   attribute to identify the traffic class a session is requesting
   in its offer/answer exchange.




The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-traffic-class-for-sdp

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-traffic-class-for-sdp-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-traffic-class-for-sdp-=
03


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


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Subject: Re: [MMUSIC] Draft new version:	draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi Christer,

The updated draft looks good as 
1) The different port is passed in the offer which reduces interop issue
with most of the SIP devices. 
2) The mechanism allows to select same port and different port in the
answerer side

My comments & clarification queries are as follows:

1) The details Multiple answer for single bundled offer is not clear in the
draft. The multiple answer may be due to SIP forking or WebRTC FedEx usecase
sort of scenario. 
2) Whether it is allowed in answerer side to generate new offer before
receiving the identical port new offer from offerer? It is allowed from RFC
3264 perspective.
3) Open issue #2 in Sec 6.2: Please explain the side-effect in case bundle
is done in the middle of the session.
4) Nits in Sec 8.2 para 2 Line 2: "generates an SDP" instead of "generates
and SDP"

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Tuesday, February 19, 2013 1:20 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> negotiation-03
> 
> 
> Hi,
> 
> I've submitted a new version (-03) of BUNDLE.
> 
> Based on discussions with Cullen, and exchange of the concerns we had
> with the different port alternatives (different-ports vs identical-
> port), this version now describes a mechanism where both alternatives
> more or less are combined. When the first BUNDLE offer is sent, and it
> is now known whether the answerer supports BUNDLE (or, even identical
> ports), different ports are used (read: the one-rtp alterntaive). Once
> BUNDLE answer has been received, a new offer, with identical port
> numbers, is sent.
> 
> We think that having to send a second offer is not going to increase
> complexity and delay, because in many use-cases subsequent offer(s)
> will be sent anyway. For example, RTCWEB entities will do it because of
> ICE.
> 
> Cullen has also been added as co-author.
> 
> Now, we do realize that there are still issues that have to be
> described, and hopefully we will be able to focus on those from now on,
> but our wish is that this proposal will unite the different "port
> camps" :)
> 
> Regards,
> 
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Tue Feb 19 11:12:03 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new	version: draft-ietf-mmusic-sdp-bundle-negotiation-03
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Subject: Re: [MMUSIC] Draft new	version:	draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi Partha,

>The updated draft looks good as
>1) The different port is passed in the offer which reduces interop issue
>with most of the SIP devices.
>2) The mechanism allows to select same port and different port in the
>answerer side
>
>My comments & clarification queries are as follows:
>
>1) The details Multiple answer for single bundled offer is not clear in th=
e
>draft. The multiple answer may be due to SIP forking or WebRTC FedEx useca=
se
>sort of scenario.

With "Multiple answer", do you mean answers received on different early dia=
logs? If so, each answer will be handled separately, and a new offer has to=
 be sent on each early dialog (assuming the answer indicated support of BUN=
DLE).

>2) Whether it is allowed in answerer side to generate new offer before
>receiving the identical port new offer from offerer? It is allowed from RF=
C
>3264 perspective.

I see no reason to forbid it. However, in order to prevent a race condition=
, maybe we should recommend that the answerer side does not generate an off=
er by itself, until it has received the single-port offer?

>3) Open issue #2 in Sec 6.2: Please explain the side-effect in case bundle
>is done in the middle of the session.

I am not sure I understand what you mean by "side-effect"?

>4) Nits in Sec 8.2 para 2 Line 2: "generates an SDP" instead of "generates=
 and SDP"

Will be fixed :)

Thanks!

Regards,

Christer




> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Tuesday, February 19, 2013 1:20 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> negotiation-03
>
>
> Hi,
>
> I've submitted a new version (-03) of BUNDLE.
>
> Based on discussions with Cullen, and exchange of the concerns we had
> with the different port alternatives (different-ports vs identical-
> port), this version now describes a mechanism where both alternatives
> more or less are combined. When the first BUNDLE offer is sent, and it
> is now known whether the answerer supports BUNDLE (or, even identical
> ports), different ports are used (read: the one-rtp alterntaive). Once
> BUNDLE answer has been received, a new offer, with identical port
> numbers, is sent.
>
> We think that having to send a second offer is not going to increase
> complexity and delay, because in many use-cases subsequent offer(s)
> will be sent anyway. For example, RTCWEB entities will do it because of
> ICE.
>
> Cullen has also been added as co-author.
>
> Now, we do realize that there are still issues that have to be
> described, and hopefully we will be able to focus on those from now on,
> but our wish is that this proposal will unite the different "port
> camps" :)
>
> Regards,
>
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

_______________________________________________
mmusic mailing list
mmusic@ietf.org
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FYI

From: internet-drafts@ietf.org [internet-drafts@ietf.org]
Sent: Sunday, February 17, 2013 10:26 AM
To: Roni Even
Cc: jonathan@vidyo.com; bill.wu@huawei.com
Subject: New Version Notification for
draft-even-mmusic-multiple-streams-02.txt

A new version of I-D, draft-even-mmusic-multiple-streams-02.txt
has been successfully submitted by Roni Even and posted to the IETF
repository.

Filename:        draft-even-mmusic-multiple-streams
Revision:        02
Title:           Describing multiple RTP media streams in SDP
Creation date:   2013-02-17
Group:           Individual Submission
Number of pages: 14
URL:
http://www.ietf.org/internet-drafts/draft-even-mmusic-multiple-streams-02.tx
t
Status:
http://datatracker.ietf.org/doc/draft-even-mmusic-multiple-streams
Htmlized:
http://tools.ietf.org/html/draft-even-mmusic-multiple-streams-02
Diff:
http://www.ietf.org/rfcdiff?url2=draft-even-mmusic-multiple-streams-02

Abstract:
   This document describes issues when describing multiple RTP streams
   in a single RTP session using SDP and considers the different RTP
   topologies that should be supported.  The document looks at current
   solutions and provides paths toward addressing the issues.




The IETF Secretariat=


From christer.holmberg@ericsson.com  Wed Feb 20 04:27:58 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi Richard,

It would have been good with some example messages/flows, but I'll give my =
comments based on my understanding of the text. Also, I'll focus on the "Un=
specified address for subsequent m lines in SDP answer" alternative, as it =
is the one you recommend, but I believe my comments also apply to some of t=
he other alternatives.

In general, you only focus on one single issue, but leave a number of quest=
ions unanswered.

First, as you will now have a single non-zero m- line, you still assume tha=
t entities (including intermediaries) will take information (media types, p=
ayload type mappings etc etc etc - even protocol, if we decide to also allo=
w the data channel in the bundle) from all m- lines. Or, do you intend to p=
ut all that information into a single m- line? If so, in my opinion MMT is =
a much better approach.

Second, what if a user wants the disable, or reject, a specific stream, usi=
ng port zero? How is it done in case the stream is already represented by a=
 zero m- line? How is it done in case the stream is represented by the non-=
zero m- line which contains the port number value used for the whole bundle=
?

Regards,

Christer







-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 19. helmikuuta 2013 0:09
To: mmusic@ietf.org; rtcweb@ietf.org
Subject: [MMUSIC] FW: New Version Notification for draft-ejzak-mmusic-bundl=
e-alternatives-00.txt

Please note that I have also submitted a draft on how to fix BUNDLE.  This =
doc is based on the modified BUNDLE proposal with different port informatio=
n for each m line and compares several different additional modifications t=
o BUNDLE.  The doc ends up recommending the use of the unspecified address =
for subsequent m lines in a bundle group in the SDP answer (not offer).  Pl=
ease see the doc for details.

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
Sent: Monday, February 18, 2013 3:57 PM
To: Ejzak, Richard P (Richard)
Subject: New Version Notification for draft-ejzak-mmusic-bundle-alternative=
s-00.txt


A new version of I-D, draft-ejzak-mmusic-bundle-alternatives-00.txt
has been successfully submitted by Richard Ejzak and posted to the IETF rep=
ository.

Filename:	 draft-ejzak-mmusic-bundle-alternatives
Revision:	 00
Title:		 Alternatives to BUNDLE
Creation date:	 2013-02-18
Group:		 Individual Submission
Number of pages: 7
URL:             http://www.ietf.org/internet-drafts/draft-ejzak-mmusic-bun=
dle-alternatives-00.txt
Status:          http://datatracker.ietf.org/doc/draft-ejzak-mmusic-bundle-=
alternatives
Htmlized:        http://tools.ietf.org/html/draft-ejzak-mmusic-bundle-alter=
natives-00


Abstract:
   This paper discusses some potential modifications to the BUNDLE
   proposal for multiplexing of multiple media types within a single
   5-tuple to address limitations of the current proposal and
   alternatives already considered by MMUSIC.

                                                                           =
      =20


The IETF Secretariat

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

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt
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Hi Christer,

Thanks for your comments!  See below.

Richard

> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>=20
> First, as you will now have a single non-zero m- line, you still assume
> that entities (including intermediaries) will take information (media
> types, payload type mappings etc etc etc - even protocol, if we decide
> to also allow the data channel in the bundle) from all m- lines. Or, do
> you intend to put all that information into a single m- line? If so, in
> my opinion MMT is a much better approach.

The intent is to reuse all of the existing information in the subsequent m =
lines except the connection, candidate and non-zero port information.  That=
 excludes the crypto info as well.

>=20
> Second, what if a user wants the disable, or reject, a specific stream,
> using port zero? How is it done in case the stream is already
> represented by a zero m- line? How is it done in case the stream is
> represented by the non-zero m- line which contains the port number
> value used for the whole bundle?

A zero port value would retain its original semantic of disabling/rejecting=
 an individual m line.  In my proposal all subsequent m lines in the SDP an=
swer that are accepted would have valid and unique non-zero port values tha=
t are different from the port value in the first m line of the bundle group=
.=20

I admit that I did not think through how to disable the first m line in the=
 group, but let me propose the following:

Let's specify that we get the connection and port information for the bundl=
e from the first valid (non-zero port) m line in the bundle group of the SD=
P answer (not offer).  If the answerer wants to disable the first valid m l=
ine(s) of the bundle group in the first SDP offer, both ends use the connec=
tion and port information from the next valid m line in the SDP answer (and=
 its corresponding m line in the SDP offer).  The offerer had valid informa=
tion for each m line in the SDP offer so it doesn't matter which set we use=
 after the first offer/answer transaction.  But we don't want to change the=
 5-tuple or crypto info for the bundle with any subsequent offer/answer.

If the offerer wants to disable the first valid m line(s) of a bundle group=
 in a subsequent offer, it places the connection and port information for t=
he bundle group in the next valid m line of the bundle group in the SDP off=
er (it would actually move this information from the previous first valid m=
 line to the new one to avoid restarting crypto or changing ports).  In res=
ponse to the new offer disabling the initial valid m line(s), the answerer =
similarly places the connection and port information for the bundle group i=
n the corresponding m line of the SDP answer.  The answerer is not allowed =
to disable the first valid m line of a bundle group in the SDP answer it se=
nds in response to a subsequent offer.  If the answerer needs to disable th=
e first valid m line(s), it must instead accept the m line(s) in response t=
o the latest pending offer (if there is one) and immediately send a new SDP=
 offer disabling the line by following the SDP offerer rule above. =20

The only other rule to add is that the SDP offerer will never re-enable (se=
t port to a non-zero value) a disabled m line before the first valid m line=
 in a bundle group but will instead create a new m line and add it to the g=
roup.  This makes it unlikely that the answerer will reject the first valid=
 m line of the bundle group in a subsequent SDP offer since it had previous=
ly accepted it and did not initiate the new transaction due to any changes =
in the configuration of local resources.

This isn't particularly elegant, but it applies the definition consistently=
 and I think it works in all cases.

>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Ejzak, Richard P (Richard)
> Sent: 19. helmikuuta 2013 0:09
> To: mmusic@ietf.org; rtcweb@ietf.org
> Subject: [MMUSIC] FW: New Version Notification for draft-ejzak-mmusic-
> bundle-alternatives-00.txt
>=20
> Please note that I have also submitted a draft on how to fix BUNDLE.
> This doc is based on the modified BUNDLE proposal with different port
> information for each m line and compares several different additional
> modifications to BUNDLE.  The doc ends up recommending the use of the
> unspecified address for subsequent m lines in a bundle group in the SDP
> answer (not offer).  Please see the doc for details.
>=20
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Monday, February 18, 2013 3:57 PM
> To: Ejzak, Richard P (Richard)
> Subject: New Version Notification for draft-ejzak-mmusic-bundle-
> alternatives-00.txt
>=20
>=20
> A new version of I-D, draft-ejzak-mmusic-bundle-alternatives-00.txt
> has been successfully submitted by Richard Ejzak and posted to the IETF
> repository.
>=20
> Filename:	 draft-ejzak-mmusic-bundle-alternatives
> Revision:	 00
> Title:		 Alternatives to BUNDLE
> Creation date:	 2013-02-18
> Group:		 Individual Submission
> Number of pages: 7
> URL:             http://www.ietf.org/internet-drafts/draft-ejzak-
> mmusic-bundle-alternatives-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-ejzak-mmusic-
> bundle-alternatives
> Htmlized:        http://tools.ietf.org/html/draft-ejzak-mmusic-bundle-
> alternatives-00
>=20
>=20
> Abstract:
>    This paper discusses some potential modifications to the BUNDLE
>    proposal for multiplexing of multiple media types within a single
>    5-tuple to address limitations of the current proposal and
>    alternatives already considered by MMUSIC.
>=20
>=20
>=20
>=20
> The IETF Secretariat
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, <mmusic@ietf.org>
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Subject: Re: [MMUSIC] Draft new	version:	draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi Christer,

The main usecase for "Multiple answer" is different answer in early
dialog/dialog (SIP forking). It is possible to see each answer as a separate
dialog (offer-answer) in case the applicability of the bundle draft is RFC
3261(SIP).  As per this draft applicability is currently RFC 3264
(offer-answer), So, is there a need to define/clarify about the higher
aggregation like (early) dialog for handling "multiple answer" within bundle
negotiation?

Please read inline for more comments.

Thanks
Partha


> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, February 20, 2013 12:42 AM
> To: Parthasarathi R; mmusic@ietf.org
> Subject: RE: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> negotiation-03
> 
> 
> Hi Partha,
> 
> >The updated draft looks good as
> >1) The different port is passed in the offer which reduces interop
> issue
> >with most of the SIP devices.
> >2) The mechanism allows to select same port and different port in the
> >answerer side
> >
> >My comments & clarification queries are as follows:
> >
> >1) The details Multiple answer for single bundled offer is not clear
> in the
> >draft. The multiple answer may be due to SIP forking or WebRTC FedEx
> usecase
> >sort of scenario.
> 
> With "Multiple answer", do you mean answers received on different early
> dialogs? If so, each answer will be handled separately, and a new offer
> has to be sent on each early dialog (assuming the answer indicated
> support of BUNDLE).
> 
> >2) Whether it is allowed in answerer side to generate new offer before
> >receiving the identical port new offer from offerer? It is allowed
> from RFC
> >3264 perspective.
> 
> I see no reason to forbid it. However, in order to prevent a race
> condition, maybe we should recommend that the answerer side does not
> generate an offer by itself, until it has received the single-port
> offer?
<Partha> I agree with you. Could you please add the recommendation in the
next revision </Partha>

> 
> >3) Open issue #2 in Sec 6.2: Please explain the side-effect in case
> bundle
> >is done in the middle of the session.
> 
> I am not sure I understand what you mean by "side-effect"?
<Partha> I'm not seeing any issue in modify the bundle negotiation after the
session is established. IMO, the bundle should be allowed in the middle of
the session. I'm asking any known side-effect in case bundle is done in the
middle of the session </Partha>

> 
> >4) Nits in Sec 8.2 para 2 Line 2: "generates an SDP" instead of
> "generates and SDP"
> 
> Will be fixed :)
> 
> Thanks!
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Christer Holmberg
> > Sent: Tuesday, February 19, 2013 1:20 AM
> > To: mmusic@ietf.org
> > Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> > negotiation-03
> >
> >
> > Hi,
> >
> > I've submitted a new version (-03) of BUNDLE.
> >
> > Based on discussions with Cullen, and exchange of the concerns we had
> > with the different port alternatives (different-ports vs identical-
> > port), this version now describes a mechanism where both alternatives
> > more or less are combined. When the first BUNDLE offer is sent, and
> it
> > is now known whether the answerer supports BUNDLE (or, even identical
> > ports), different ports are used (read: the one-rtp alterntaive).
> Once
> > BUNDLE answer has been received, a new offer, with identical port
> > numbers, is sent.
> >
> > We think that having to send a second offer is not going to increase
> > complexity and delay, because in many use-cases subsequent offer(s)
> > will be sent anyway. For example, RTCWEB entities will do it because
> of
> > ICE.
> >
> > Cullen has also been added as co-author.
> >
> > Now, we do realize that there are still issues that have to be
> > described, and hopefully we will be able to focus on those from now
> on,
> > but our wish is that this proposal will unite the different "port
> > camps" :)
> >
> > Regards,
> >
> > Christer
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] Draft new	version:	draft-ietf-mmusic-sdp-bundle-negotiation-03
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On 2/20/13 12:40 PM, Parthasarathi R wrote:

>>> 2) Whether it is allowed in answerer side to generate new offer before
>>> receiving the identical port new offer from offerer? It is allowed
>> from RFC
>>> 3264 perspective.
>>
>> I see no reason to forbid it. However, in order to prevent a race
>> condition, maybe we should recommend that the answerer side does not
>> generate an offer by itself, until it has received the single-port
>> offer?
> <Partha> I agree with you. Could you please add the recommendation in the
> next revision </Partha>

IMO this is a bad idea.

The UAS may not be able to wait, or it may complicate the programming to 
wait.

	Thanks,
	Paul


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new	version: draft-ietf-mmusic-sdp-bundle-negotiation-03
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Subject: Re: [MMUSIC] Draft new	version:	draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi Partha,

>The main usecase for "Multiple answer" is different answer in early
>dialog/dialog (SIP forking). It is possible to see each answer as a separa=
te
>dialog (offer-answer) in case the applicability of the bundle draft is RFC
>3261(SIP).  As per this draft applicability is currently RFC 3264
>(offer-answer), So, is there a need to define/clarify about the higher
>aggregation like (early) dialog for handling "multiple answer" within bund=
le
>negotiation?

I have difficulties to understand what you are asking. There is no such thi=
ng as "multiple answer" in RFC 3264. There is an offer, followed by an answ=
er.=20

>>>2) Whether it is allowed in answerer side to generate new offer before
>>>receiving the identical port new offer from offerer? It is allowed
>>>from RFC 3264 perspective.
>>>
>>I see no reason to forbid it. However, in order to prevent a race
>>condition, maybe we should recommend that the answerer side does not
>>generate an offer by itself, until it has received the single-port
>offer?
><Partha> I agree with you. Could you please add the recommendation in the
>next revision </Partha>

We can discuss such details later (and I noted that Paul has another opinon=
 :), but first I would like to know whether the WG on a high level can agre=
e on moving forward with this :)

>>>3) Open issue #2 in Sec 6.2: Please explain the side-effect in case
>>>bundle is done in the middle of the session.
>>
>> I am not sure I understand what you mean by "side-effect"?
><Partha> I'm not seeing any issue in modify the bundle negotiation after t=
he
>session is established. IMO, the bundle should be allowed in the middle of
>the session. I'm asking any known side-effect in case bundle is done in th=
e
>middle of the session </Partha>

I can't think of any side-effects.

Regards,

Christer



> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Christer Holmberg
> > Sent: Tuesday, February 19, 2013 1:20 AM
> > To: mmusic@ietf.org
> > Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> > negotiation-03
> >
> >
> > Hi,
> >
> > I've submitted a new version (-03) of BUNDLE.
> >
> > Based on discussions with Cullen, and exchange of the concerns we had
> > with the different port alternatives (different-ports vs identical-
> > port), this version now describes a mechanism where both alternatives
> > more or less are combined. When the first BUNDLE offer is sent, and
> it
> > is now known whether the answerer supports BUNDLE (or, even identical
> > ports), different ports are used (read: the one-rtp alterntaive).
> Once
> > BUNDLE answer has been received, a new offer, with identical port
> > numbers, is sent.
> >
> > We think that having to send a second offer is not going to increase
> > complexity and delay, because in many use-cases subsequent offer(s)
> > will be sent anyway. For example, RTCWEB entities will do it because
> of
> > ICE.
> >
> > Cullen has also been added as co-author.
> >
> > Now, we do realize that there are still issues that have to be
> > described, and hopefully we will be able to focus on those from now
> on,
> > but our wish is that this proposal will unite the different "port
> > camps" :)
> >
> > Regards,
> >
> > Christer
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Date: Wed, 20 Feb 2013 16:27:03 -0800
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Subject: [MMUSIC] Proposed Agenda For WG Meetings
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In the tradition of the Interim, I tried to sit down and work out an
agenda that reflects that I think the important concerns are, namely:

- Video codecs (as promised)
- Multiplexing of media flows
- Trickle ICE
- DataChannels

Two notes about cross-WG interaction:

- I'm assuming that most of the MMUSIC time will be used for
  RTCWEB-type stuff, though I left 30 min for other MMUSIC
  business on Tue.

- I don't know how much time AVTEXT needs given the AVTEXT/
  MMUSIC merger on Fri. We can reduce the draft-jennings-rtcweb-plan
  time accordingly based on agenda requests.

Hope this is useful. I'm of course happy to get on a call to discuss
this if that helps.

-Ekr


RTCWEB I: Tuesday 0900-1130
0900 - 0905  Administrivia
0905 - 1100  Video Codec MTI discussion
  - draft-alvestrand-rtcweb-vp8-00 (30 mins)
  -
draft-burman-rtcweb-h264-proposal-00+draft-dbenham-webrtcvideomti-00+draft-marjou-rtcweb-video-codec-00
(30 mins)
  - General discussion 30 min)
  - Call the question of which mandatory to implement video codec to select
(5 min)
  - Next steps (20 min)
 1100 - 1115  draft-ietf-rtcweb-use-cases-and-requirements (LC comments)
 1115 - 1130  draft-ietf-draft-ietf-rtcweb-overview (LC comments)


MMUSIC I: Tuesday 1520-1830
1520 - 1525  Administrivia
1525 - 1605  Trickle ICE (Ivov)
- Under what conditions can you do full trickle (discovery)
- When can checking stop? [UPDATE vs. in-band vs...?]
- SDP/SIP encoding for additional trickle candidates
1605 - 1650  - Bundle
 - draft-ejzak-mmusic-bundle-alternatives (15 min)
 - draft-ietf-mmusic-sdp-bundle-negotiation (15 min)
 - discussion/consensus calls (15 min)
1650 - 1700  [Break]
1700 - 1800  SDP attribute analysis for multiplexing
 - draft-nandakumar-mmusic-sdp-mux-attributes
1800 - 1830  [Other MMUSIC business]


RTCWEB II: Thursday 0900-1130
0900 - 0905  Administrivia
0905 - 1025  Data Channel negotiation
 - draft-jesup-rtcweb-data-protocol (15 min)
 - draft-thomson-rtcweb-data (15 min)
 - draft-marcon-rtcweb-data-channel-management (15 min)
 - Discussion (30 min)
 - Consensus call (5 min)
1025 - 1100 JSEP Update (Uberti)
 - draft-ietf-rtcweb-jsep
1100 - 1115  RTCP-XR (Westerlund)
1115 - 1130  RTP Concepts and relation
- daft-burman-rtcweb-mmusic-media-structure


MMUSIC II: Friday 0900-1100
0900 - 0905  Administrivia
0905 - 0925  Requirements (draft-jennings-mmusic-media-req)
0925 - 1015  PC-track to m-line mapping (draft-jennings-rtcweb-plan)
1015 - 1100  MSID changes to match multiplexing (draft-jennings-rtcweb-plan)

--047d7b6dc0bcb147b304d6312431
Content-Type: text/html; charset=ISO-8859-1
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<div>In the tradition of the Interim, I tried to sit down and work out an</=
div><div>agenda that reflects that I think the important concerns are, name=
ly:</div><div><br></div><div>- Video codecs (as promised)</div><div>- Multi=
plexing of media flows</div>

<div>- Trickle ICE</div><div>- DataChannels</div><div><br></div><div>Two no=
tes about cross-WG interaction:</div><div><br></div><div>- I&#39;m assuming=
 that most of the MMUSIC time will be used for</div><div>=A0 RTCWEB-type st=
uff, though I left 30 min for other MMUSIC</div>

<div>=A0 business on Tue.</div><div><br></div><div>- I don&#39;t know how m=
uch time AVTEXT needs given the AVTEXT/</div><div>=A0 MMUSIC merger on Fri.=
 We can reduce the draft-jennings-rtcweb-plan</div><div>=A0 time accordingl=
y based on agenda requests.</div>

<div><br></div><div>Hope this is useful. I&#39;m of course happy to get on =
a call to discuss</div><div>this if that helps.</div><div><br></div><div>-E=
kr</div><div><br></div><div><br></div><div>RTCWEB I: Tuesday 0900-1130</div=
>

<div>0900 - 0905 =A0Administrivia</div><div>0905 - 1100 =A0Video Codec MTI =
discussion</div><div>=A0 - draft-alvestrand-rtcweb-vp8-00 (30 mins)</div><d=
iv>=A0 - draft-burman-rtcweb-h264-proposal-00+draft-dbenham-webrtcvideomti-=
00+draft-marjou-rtcweb-video-codec-00 (30 mins)</div>

<div>=A0 - General discussion 30 min)</div><div>=A0 - Call the question of =
which mandatory to implement video codec to select (5 min)</div><div>=A0 - =
Next steps (20 min)</div><div>=A01100 - 1115 =A0draft-ietf-rtcweb-use-cases=
-and-requirements (LC comments)</div>

<div>=A01115 - 1130 =A0draft-ietf-draft-ietf-rtcweb-overview (LC comments)<=
/div><div>=A0 =A0=A0</div><div><br></div><div>MMUSIC I: Tuesday 1520-1830</=
div><div>1520 - 1525 =A0Administrivia</div><div>1525 - 1605 =A0Trickle ICE =
(Ivov)</div>

<div>- Under what conditions can you do full trickle (discovery)</div><div>=
- When can checking stop? [UPDATE vs. in-band vs...?]</div><div>- SDP/SIP e=
ncoding for additional trickle candidates</div><div>1605 - 1650 =A0- Bundle=
=A0</div>

<div>=A0- draft-ejzak-mmusic-bundle-alternatives (15 min)</div><div>=A0- dr=
aft-ietf-mmusic-sdp-bundle-negotiation (15 min)</div><div>=A0- discussion/c=
onsensus calls (15 min)</div><div>1650 - 1700 =A0[Break]</div><div>1700 - 1=
800 =A0SDP attribute analysis for multiplexing</div>

<div>=A0- draft-nandakumar-mmusic-sdp-mux-attributes</div><div>1800 - 1830 =
=A0[Other MMUSIC business]</div><div><br></div><div><br></div><div>RTCWEB I=
I: Thursday 0900-1130</div><div>0900 - 0905 =A0Administrivia</div><div>0905=
 - 1025 =A0Data Channel negotiation</div>

<div>=A0- draft-jesup-rtcweb-data-protocol (15 min)</div><div>=A0- draft-th=
omson-rtcweb-data (15 min)</div><div>=A0- draft-marcon-rtcweb-data-channel-=
management (15 min)</div><div>=A0- Discussion (30 min)</div><div>=A0- Conse=
nsus call (5 min)</div>

<div>1025 - 1100 JSEP Update (Uberti)</div><div>=A0- draft-ietf-rtcweb-jsep=
</div><div>1100 - 1115 =A0RTCP-XR (Westerlund)</div><div>1115 - 1130 =A0RTP=
 Concepts and relation</div><div>- daft-burman-rtcweb-mmusic-media-structur=
e</div>

<div><br></div><div><br></div><div>MMUSIC II: Friday 0900-1100</div><div>09=
00 - 0905 =A0Administrivia</div><div>0905 - 0925 =A0Requirements (draft-jen=
nings-mmusic-media-req)</div><div>0925 - 1015 =A0PC-track to m-line mapping=
 (draft-jennings-rtcweb-plan)</div>

<div>1015 - 1100 =A0MSID changes to match multiplexing (draft-jennings-rtcw=
eb-plan)</div><div><br></div>

--047d7b6dc0bcb147b304d6312431--

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03
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Subject: Re: [MMUSIC] Draft new version:	draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi Christer,

I read the draft and I think it is a good step forward. I like the overall =
approach and think it can work. I do have questions about some of the speci=
fics, and also some suggested changes and additions.

1) In section 1, the 4th paragraph ends as follows:

   The port number
   value associated with the first "m=3D" line associated with the
   "BUNDLE" group represents the port value number offered to be used in
   the single 5-tuple.

I think this is no longer necessary per the rules in section 6.1, and it sh=
ould be removed.

2) Section 2 says the following about RTCP:

   Bundled media: Two or more RTP streams using a single 5-tuple.  The
   RTCP streams associated with the RTP streams also use a single
   5-tuple, which might be the same, but can also be different, as the
   one used by the RTP streams.

This is contradicted by section 6.1, which says the SDP offer MUST
   o  5.  Insert an SDP 'rtpc-mux' attribute.

So one of these needs to be changed, and replace 'rtpc-mux' with 'rtcp-mux'=
 :)

3)   OPEN ISSUE #1: If the SDP Answerer supports "BUNDLE", do we mandate
   that "m=3D" lines must be grouped in the same way in the SDP Answer as
   they are grouped in the SDP Offer?

I'd say yes, because otherwise you are not sure if something is wrong (e.g.=
 an intermediary has changed the ports, or something does not really unders=
tand bundle). I think if the ports in the SDP answer are not grouped the sa=
me as in the offer, it should be interpreted as a failure to bundle, and an=
other offer should be sent to use different ports.

4) I do not understand the intent of the following text in section 6.2:

   However, if the "m=3D" lines associated with the "BUNDLE"
   group contain different port number values (i.e. the SDP Offer is
   generated until it is known whether the SDP Answerer supports the
   "BUNDLE" grouping extension) the SDP Offerer MUST NOT set the port
   number value of the first "m=3D" line associated with the "BUNDLE"
   group to zero.

Perhaps there is a typo, but in any case, I do not think the first "m=3D" l=
ines needs to be treated specially any more. This is similar to comment (1)=
.

5)   OPEN ISSUE #2: If the session has been established without BUNDLE, do
   we allow BUNDLE to be enabled later during the session?

Yes, but need to use different ports until it is known that both offerer an=
d answerer support bundle.

6)   OPEN ISSUE #3: If the session has been established with BUNDLE, do we
   allow BUNDLE to be disabled later during the session, meaning that

Yes, for failure recovery.=20

7) Section 6.3 ends as follows:

   Until the SDP Answerer receives the new SDP Offer, which contains an
   identical port number value for each "m=3D" line associated with a
   "BUNDLE" group, it MUST use the port, and related information (ICE
   candidates, SDES keys etc) associated with the first "m=3D" line in the
   "BUNDLE" group.

I think the SDP Answerer needs to wait for the next offer and not assume th=
e first "m=3D" line contains the port to be used for the bundle.

8) Section 10

The SDP Answer shows the same port being used in the "m=3D" lines. I was th=
inking this first answer would use different ports, and only if there is an=
other offer/answer exchange would both the offer and the answer switch to u=
sing the same port.

Cheers,
Charles

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Monday, February 18, 2013 11:50 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> negotiation-03
>=20
>=20
> Hi,
>=20
> I've submitted a new version (-03) of BUNDLE.
>=20
> Based on discussions with Cullen, and exchange of the concerns we had
> with the different port alternatives (different-ports vs identical-port),=
 this
> version now describes a mechanism where both alternatives more or less
> are combined. When the first BUNDLE offer is sent, and it is now known
> whether the answerer supports BUNDLE (or, even identical ports), differen=
t
> ports are used (read: the one-rtp alterntaive). Once BUNDLE answer has
> been received, a new offer, with identical port numbers, is sent.
>=20
> We think that having to send a second offer is not going to increase
> complexity and delay, because in many use-cases subsequent offer(s) will
> be sent anyway. For example, RTCWEB entities will do it because of ICE.
>=20
> Cullen has also been added as co-author.
>=20
> Now, we do realize that there are still issues that have to be described,=
 and
> hopefully we will be able to focus on those from now on, but our wish is
> that this proposal will unite the different "port camps" :)
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From christer.holmberg@ericsson.com  Wed Feb 20 23:10:07 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03 - Charles' comments
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Subject: Re: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03 - Charles' comments
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Hi Charles,

Thanks for your comments! Reply inline.

>I read the draft and I think it is a good step forward. I like the overall=
 approach and think it can work. I do have questions about some of the spec=
ifics, and also some suggested changes and additions.
>
>1) In section 1, the 4th paragraph ends as follows:
>
>   The port number
>   value associated with the first "m=3D" line associated with the
>   "BUNDLE" group represents the port value number offered to be used in
>   the single 5-tuple.
>
>I think this is no longer necessary per the rules in section 6.1, and it s=
hould be removed.

The text talks about the Offer where different port numbers are used, ie wh=
en it is still not known whether the Answerer supports bundle. But, maybe t=
hat needs to be more clear.

--------------------------

> 2) Section 2 says the following about RTCP:
>
>   Bundled media: Two or more RTP streams using a single 5-tuple.  The
>   RTCP streams associated with the RTP streams also use a single
>   5-tuple, which might be the same, but can also be different, as the
>   one used by the RTP streams.
>
>This is contradicted by section 6.1, which says the SDP offer MUST
>   o  5.  Insert an SDP 'rtpc-mux' attribute.
>
> So one of these needs to be changed, and replace 'rtpc-mux' with 'rtcp-mu=
x' :)

Correct. I got the same comment off-line from Harald, but forgot to fix it.

Do people have any preference? Shall du usage of rtcp-mux be a SHOULD, a MU=
ST, or a RECOMMEND? :)

--------------------------

>3)   OPEN ISSUE #1: If the SDP Answerer supports "BUNDLE", do we mandate
>   that "m=3D" lines must be grouped in the same way in the SDP Answer as
>   they are grouped in the SDP Offer?
>
> I'd say yes, because otherwise you are not sure if something is wrong (e.=
g. an intermediary has changed the ports, or something=20
> does not really understand bundle). I think if the ports in the SDP answe=
r are not grouped the same as in the offer, it should be=20
> interpreted as a failure to bundle, and another offer should be sent to u=
se different ports.

I'm still thinking about it, but at least at this point I would agree with =
you :)

--------------------------

>4) I do not understand the intent of the following text in section 6.2:
>
>   However, if the "m=3D" lines associated with the "BUNDLE"
>   group contain different port number values (i.e. the SDP Offer is
>   generated until it is known whether the SDP Answerer supports the
>   "BUNDLE" grouping extension) the SDP Offerer MUST NOT set the port
>   number value of the first "m=3D" line associated with the "BUNDLE"
>   group to zero.
>
> Perhaps there is a typo, but in any case, I do not think the first "m=3D"=
 lines needs to be treated specially any more. This is similar to comment (=
1).

Keep in mind that, in this case, only the port of the first "m=3D" line rep=
resents the bundle port (as different port values are given for the other m=
=3D lines).

--------------------------

>5)   OPEN ISSUE #2: If the session has been established without BUNDLE, do
>   we allow BUNDLE to be enabled later during the session?
>
> Yes, but need to use different ports until it is known that both offerer =
and answerer support bundle.

Sure, that rule would still apply.

--------------------------

>6)   OPEN ISSUE #3: If the session has been established with BUNDLE, do we
>   allow BUNDLE to be disabled later during the session, meaning that
>
> Yes, for failure recovery.=20

I was more thinking of the case where endpoints already have successfully n=
egotiated the usage of bundle, and then at some point during the session ch=
ose (for whatever reason) to disable the multiplexing.

Basically it would mean sending a new offer, without BUNDLE, and with diffe=
rent port number values - ie a "legacy"  offer :)

--------------------------

>7) Section 6.3 ends as follows:
>
>   Until the SDP Answerer receives the new SDP Offer, which contains an
>   identical port number value for each "m=3D" line associated with a
>   "BUNDLE" group, it MUST use the port, and related information (ICE
>   candidates, SDES keys etc) associated with the first "m=3D" line in the
>   "BUNDLE" group.
>
> I think the SDP Answerer needs to wait for the next offer and not assume =
the first "m=3D" line contains the port to be used for the bundle.

Is there a specific reason why you think that we can't mandate the Offerer =
to insert the bundle port in the first "m=3D" line?

--------------------------

>8) Section 10
>
>The SDP Answer shows the same port being used in the "m=3D" lines. I was t=
hinking this first answer would use different ports, and only if=20
>there is another offer/answer exchange would both the offer and the answer=
 switch to using the same port.

My personal opinion is that the answerer already knows that the offerer wan=
ts to use bundle, and is expected to send a new offer. Also, by using the s=
ame ports already in the first answer, we would prevent a case where the an=
swerer does NOT support bundle, but blindly copies the grouping attributes =
from the offer to the answer.

But, opinions are welcome :)

Regards,

Christer

From christer.holmberg@ericsson.com  Thu Feb 21 04:26:00 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,

>> First, as you will now have a single non-zero m- line, you still assume
>> that entities (including intermediaries) will take information (media
>> types, payload type mappings etc etc etc - even protocol, if we decide
>> to also allow the data channel in the bundle) from all m- lines. Or, do
>> you intend to put all that information into a single m- line? If so, in
>> my opinion MMT is a much better approach.
>
> The intent is to reuse all of the existing information in the subsequent =
m lines except the connection, candidate and non-zero port information.  Th=
at excludes the crypto info as well.

And how do you expect intermediaries to understand that, when the ports are=
 zero?

And, even if we don't consider intermediaries, I think it's BAD DESIGN to a=
ssume that we use a stream, and associated data, from m- lines with port ze=
ro. Port zero means that the stream associated with the m- line is DISABLED=
.

>> Second, what if a user wants the disable, or reject, a specific stream,
>> using port zero? How is it done in case the stream is already
>> represented by a zero m- line? How is it done in case the stream is
>> represented by the non-zero m- line which contains the port number
>> value used for the whole bundle?
>
> A zero port value would retain its original semantic of disabling/rejecti=
ng an individual m line.  In my proposal all subsequent m lines in the SDP =
answer that are accepted=20
>would have valid and unique non-zero port values that are different from t=
he port value in the first m line of the bundle group.

>I admit that I did not think through how to disable the first m line in th=
e group, but let me propose the following:
>
>Let's specify that we get the connection and port information for the bund=
le from the first valid (non-zero port) m line=20
>in the bundle group of the SDP answer (not offer).  If the answerer wants =
to disable the first valid m line(s) of the bundle=20
>group in the first SDP offer, both ends use the connection and port inform=
ation from the next valid m line in the SDP=20
>answer (and its corresponding m line in the SDP offer).  The offerer had v=
alid information for each m line in the SDP=20
>offer so it doesn't matter which set we use after the first offer/answer t=
ransaction.  But we don't want to=20
>change the 5-tuple or crypto info for the bundle with any subsequent offer=
/answer.
>
>If the offerer wants to disable the first valid m line(s) of a bundle grou=
p in a subsequent offer, it places the=20
>connection and port information for the bundle group in the next valid m l=
ine of the bundle group in the SDP=20
>offer (it would actually move this information from the previous first val=
id m line to the new one to avoid=20
>restarting crypto or changing ports).
>In response to the new offer disabling the initial valid m line(s), the an=
swerer similarly places the connection and=20
>port information for the bundle group in the corresponding m line of the S=
DP answer. =20

I think that's very hacky, moving information from one m- line to another..=
.

>The answerer is not allowed to disable the first valid m line of a bundle =
group in the SDP answer it sends in response=20
>to a subsequent offer. If the answerer needs to disable the first valid m =
line(s), it must instead accept the m line(s) in=20
>response to the latest pending offer (if there is one) and immediately sen=
d a new SDP offer disabling the line by following=20
>the SDP offerer rule above.

I don't think we should have a solution where the answerer is not allowed t=
o reject an m- line.

>The only other rule to add is that the SDP offerer will never re-enable (s=
et port to a non-zero value) a disabled m line=20
>before the first valid m line in a bundle group but will instead create a =
new m line and add it to the group.  This makes=20
>it unlikely that the answerer will reject the first valid m line of the bu=
ndle group in a subsequent SDP offer since it had=20
>previously accepted it and did not initiate the new transaction due to any=
 changes in the configuration of local resources.

What if you have a session where a stream is enabled/disabled many times. I=
f, every time you re-enable, is going to add a new m- line, you may end up =
with a very large SDP...

>This isn't particularly elegant, but it applies the definition consistentl=
y and I think it works in all cases.

I think it's very hacky, and I don't see how you think that intermediaries =
(which your draft focues on) would be able to handle this...

Regards,

Christer









> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Ejzak, Richard P (Richard)
> Sent: 19. helmikuuta 2013 0:09
> To: mmusic@ietf.org; rtcweb@ietf.org
> Subject: [MMUSIC] FW: New Version Notification for draft-ejzak-mmusic-
> bundle-alternatives-00.txt
>
> Please note that I have also submitted a draft on how to fix BUNDLE.
> This doc is based on the modified BUNDLE proposal with different port
> information for each m line and compares several different additional
> modifications to BUNDLE.  The doc ends up recommending the use of the
> unspecified address for subsequent m lines in a bundle group in the SDP
> answer (not offer).  Please see the doc for details.
>
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Monday, February 18, 2013 3:57 PM
> To: Ejzak, Richard P (Richard)
> Subject: New Version Notification for draft-ejzak-mmusic-bundle-
> alternatives-00.txt
>
>
> A new version of I-D, draft-ejzak-mmusic-bundle-alternatives-00.txt
> has been successfully submitted by Richard Ejzak and posted to the IETF
> repository.
>
> Filename:      draft-ejzak-mmusic-bundle-alternatives
> Revision:      00
> Title:                 Alternatives to BUNDLE
> Creation date:         2013-02-18
> Group:                 Individual Submission
> Number of pages: 7
> URL:             http://www.ietf.org/internet-drafts/draft-ejzak-
> mmusic-bundle-alternatives-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-ejzak-mmusic-
> bundle-alternatives
> Htmlized:        http://tools.ietf.org/html/draft-ejzak-mmusic-bundle-
> alternatives-00
>
>
> Abstract:
>    This paper discusses some potential modifications to the BUNDLE
>    proposal for multiplexing of multiple media types within a single
>    5-tuple to address limitations of the current proposal and
>    alternatives already considered by MMUSIC.
>
>
>
>
> The IETF Secretariat
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From fandreas@cisco.com  Thu Feb 21 05:04:56 2013
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Cc: Flemming Andreasen <fandreas@cisco.com>, Ari Keranen <ari.keranen@ericsson.com>
Subject: [MMUSIC] Reminder [Re: MMUSIC agenda requests for IETF 86]
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Dead-line for agenda requests will be this Sunday, February 24. If you 
have not submitted an agenda request yet but have a draft you would like 
to discuss, please submit your request to the chairs with an indication of
- The draft you would like to discuss
- The major open issues you want to discuss

Again, priority will be given to active drafts that require discussion 
of open issues in the WG. General status updates of drafts can be 
handled by the chairs.

Thanks

-- Ari & Flemming


On 2/11/13 9:07 PM, Flemming Andreasen wrote:
> MMUSIC is currently scheduled to meet for a total of 4 hours at IETF 
> 86 in Orlando, Florida.
>
> If you want to request a time slot to discuss a draft, please send an 
> e-mail to the chairs.
>
> As usual, we want to ensure we take advantage of our meeting time in a 
> productive manner that leads to forward progress. This means that we 
> want presenters to come prepared to discuss open issues that have 
> already been raised on the mailing list and ideally received some 
> feedback already. Priority will be given drafts that gather attention 
> and discussion in the mailing list prior to the meeting.
>
> Thanks
>
> -- Ari and Flemming
> .
>


From richard.ejzak@alcatel-lucent.com  Thu Feb 21 06:06:27 2013
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi Christer,
My proposal never has a zero port on a valid m line, in either offer or ans=
wer.  I only use unspecified address (with valid port number) for subsequen=
t m lines in answers.  I'm not sure how you got this from the description. =
 Let's discuss this on the call today.  I hope you are available.

Richard

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, February 21, 2013 6:26 AM
> To: Ejzak, Richard P (Richard); mmusic@ietf.org; rtcweb@ietf.org
> Subject: RE: [MMUSIC] FW: New Version Notification for draft-ejzak-
> mmusic-bundle-alternatives-00.txt
>=20
>=20
> Hi,
>=20
> >> First, as you will now have a single non-zero m- line, you still
> >> assume that entities (including intermediaries) will take
> information
> >> (media types, payload type mappings etc etc etc - even protocol, if
> >> we decide to also allow the data channel in the bundle) from all m-
> >> lines. Or, do you intend to put all that information into a single
> m-
> >> line? If so, in my opinion MMT is a much better approach.
> >
> > The intent is to reuse all of the existing information in the
> subsequent m lines except the connection, candidate and non-zero port
> information.  That excludes the crypto info as well.
>=20
> And how do you expect intermediaries to understand that, when the ports
> are zero?
>=20
> And, even if we don't consider intermediaries, I think it's BAD DESIGN
> to assume that we use a stream, and associated data, from m- lines with
> port zero. Port zero means that the stream associated with the m- line
> is DISABLED.
>=20
> >> Second, what if a user wants the disable, or reject, a specific
> >> stream, using port zero? How is it done in case the stream is
> already
> >> represented by a zero m- line? How is it done in case the stream is
> >> represented by the non-zero m- line which contains the port number
> >> value used for the whole bundle?
> >
> > A zero port value would retain its original semantic of
> >disabling/rejecting an individual m line.  In my proposal all
> subsequent m lines in the SDP answer that are accepted would have valid
> and unique non-zero port values that are different from the port value
> in the first m line of the bundle group.
>=20
> >I admit that I did not think through how to disable the first m line
> in the group, but let me propose the following:
> >
> >Let's specify that we get the connection and port information for the
> >bundle from the first valid (non-zero port) m line in the bundle group
> >of the SDP answer (not offer).  If the answerer wants to disable the
> >first valid m line(s) of the bundle group in the first SDP offer, both
> >ends use the connection and port information from the next valid m
> line
> >in the SDP answer (and its corresponding m line in the SDP offer).
> The offerer had valid information for each m line in the SDP offer so
> it doesn't matter which set we use after the first offer/answer
> transaction.  But we don't want to change the 5-tuple or crypto info
> for the bundle with any subsequent offer/answer.
> >
> >If the offerer wants to disable the first valid m line(s) of a bundle
> >group in a subsequent offer, it places the connection and port
> >information for the bundle group in the next valid m line of the
> bundle
> >group in the SDP offer (it would actually move this information from
> the previous first valid m line to the new one to avoid restarting
> crypto or changing ports).
> >In response to the new offer disabling the initial valid m line(s),
> the
> >answerer similarly places the connection and port information for the
> bundle group in the corresponding m line of the SDP answer.
>=20
> I think that's very hacky, moving information from one m- line to
> another...
>=20
> >The answerer is not allowed to disable the first valid m line of a
> >bundle group in the SDP answer it sends in response to a subsequent
> >offer. If the answerer needs to disable the first valid m line(s), it
> >must instead accept the m line(s) in response to the latest pending
> offer (if there is one) and immediately send a new SDP offer disabling
> the line by following the SDP offerer rule above.
>=20
> I don't think we should have a solution where the answerer is not
> allowed to reject an m- line.
>=20
> >The only other rule to add is that the SDP offerer will never re-
> enable
> >(set port to a non-zero value) a disabled m line before the first
> valid
> >m line in a bundle group but will instead create a new m line and add
> >it to the group.  This makes it unlikely that the answerer will reject
> the first valid m line of the bundle group in a subsequent SDP offer
> since it had previously accepted it and did not initiate the new
> transaction due to any changes in the configuration of local resources.
>=20
> What if you have a session where a stream is enabled/disabled many
> times. If, every time you re-enable, is going to add a new m- line, you
> may end up with a very large SDP...
>=20
> >This isn't particularly elegant, but it applies the definition
> consistently and I think it works in all cases.
>=20
> I think it's very hacky, and I don't see how you think that
> intermediaries (which your draft focues on) would be able to handle
> this...
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Ejzak, Richard P (Richard)
> > Sent: 19. helmikuuta 2013 0:09
> > To: mmusic@ietf.org; rtcweb@ietf.org
> > Subject: [MMUSIC] FW: New Version Notification for draft-ejzak-
> mmusic-
> > bundle-alternatives-00.txt
> >
> > Please note that I have also submitted a draft on how to fix BUNDLE.
> > This doc is based on the modified BUNDLE proposal with different port
> > information for each m line and compares several different additional
> > modifications to BUNDLE.  The doc ends up recommending the use of the
> > unspecified address for subsequent m lines in a bundle group in the
> > SDP answer (not offer).  Please see the doc for details.
> >
> > -----Original Message-----
> > From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> > Sent: Monday, February 18, 2013 3:57 PM
> > To: Ejzak, Richard P (Richard)
> > Subject: New Version Notification for draft-ejzak-mmusic-bundle-
> > alternatives-00.txt
> >
> >
> > A new version of I-D, draft-ejzak-mmusic-bundle-alternatives-00.txt
> > has been successfully submitted by Richard Ejzak and posted to the
> > IETF repository.
> >
> > Filename:      draft-ejzak-mmusic-bundle-alternatives
> > Revision:      00
> > Title:                 Alternatives to BUNDLE
> > Creation date:         2013-02-18
> > Group:                 Individual Submission
> > Number of pages: 7
> > URL:             http://www.ietf.org/internet-drafts/draft-ejzak-
> > mmusic-bundle-alternatives-00.txt
> > Status:          http://datatracker.ietf.org/doc/draft-ejzak-mmusic-
> > bundle-alternatives
> > Htmlized:        http://tools.ietf.org/html/draft-ejzak-mmusic-
> bundle-
> > alternatives-00
> >
> >
> > Abstract:
> >    This paper discusses some potential modifications to the BUNDLE
> >    proposal for multiplexing of multiple media types within a single
> >    5-tuple to address limitations of the current proposal and
> >    alternatives already considered by MMUSIC.
> >
> >
> >
> >
> > The IETF Secretariat
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic

From christer.holmberg@ericsson.com  Thu Feb 21 06:13:31 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt
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Date: Thu, 21 Feb 2013 14:13:22 +0000
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Subject: Re: [MMUSIC] FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt
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Hi,

>My proposal never has a zero port on a valid m line, in either offer or an=
swer.  I only use unspecified address (with valid port number) for subseque=
nt m lines in answers.  I'm not sure how you got this from the description.=
  Let's discuss=20
>this on the call today.  I hope you are available.

Sorry, I mixed up terminology.

But, still, if you want to disable/reject the m- line which contains the bu=
ndle ADDRESS, you will have to move the address to another m- line.

And, still, you expect intermediaries to retrieve media types, protocols, p=
ayload types etc from the m- lines, which addresses are unspecified - but a=
t the same time you say that no resources will be reserved for the m- lines=
 with unspecified addresses...

Regards,

Christer




> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, February 21, 2013 6:26 AM
> To: Ejzak, Richard P (Richard); mmusic@ietf.org; rtcweb@ietf.org
> Subject: RE: [MMUSIC] FW: New Version Notification for draft-ejzak-=20
> mmusic-bundle-alternatives-00.txt
>=20
>=20
> Hi,
>=20
> >> First, as you will now have a single non-zero m- line, you still=20
> >> assume that entities (including intermediaries) will take
> information
> >> (media types, payload type mappings etc etc etc - even protocol, if=20
> >> we decide to also allow the data channel in the bundle) from all m-=20
> >> lines. Or, do you intend to put all that information into a single
> m-
> >> line? If so, in my opinion MMT is a much better approach.
> >
> > The intent is to reuse all of the existing information in the
> subsequent m lines except the connection, candidate and non-zero port=20
> information.  That excludes the crypto info as well.
>=20
> And how do you expect intermediaries to understand that, when the=20
> ports are zero?
>=20
> And, even if we don't consider intermediaries, I think it's BAD DESIGN=20
> to assume that we use a stream, and associated data, from m- lines=20
> with port zero. Port zero means that the stream associated with the m-=20
> line is DISABLED.
>=20
> >> Second, what if a user wants the disable, or reject, a specific=20
> >> stream, using port zero? How is it done in case the stream is
> already
> >> represented by a zero m- line? How is it done in case the stream is=20
> >> represented by the non-zero m- line which contains the port number=20
> >> value used for the whole bundle?
> >
> > A zero port value would retain its original semantic of=20
> >disabling/rejecting an individual m line.  In my proposal all
> subsequent m lines in the SDP answer that are accepted would have=20
> valid and unique non-zero port values that are different from the port=20
> value in the first m line of the bundle group.
>=20
> >I admit that I did not think through how to disable the first m line
> in the group, but let me propose the following:
> >
> >Let's specify that we get the connection and port information for the=20
> >bundle from the first valid (non-zero port) m line in the bundle=20
> >group of the SDP answer (not offer).  If the answerer wants to=20
> >disable the first valid m line(s) of the bundle group in the first=20
> >SDP offer, both ends use the connection and port information from the=20
> >next valid m
> line
> >in the SDP answer (and its corresponding m line in the SDP offer).
> The offerer had valid information for each m line in the SDP offer so=20
> it doesn't matter which set we use after the first offer/answer=20
> transaction.  But we don't want to change the 5-tuple or crypto info=20
> for the bundle with any subsequent offer/answer.
> >
> >If the offerer wants to disable the first valid m line(s) of a bundle=20
> >group in a subsequent offer, it places the connection and port=20
> >information for the bundle group in the next valid m line of the
> bundle
> >group in the SDP offer (it would actually move this information from
> the previous first valid m line to the new one to avoid restarting=20
> crypto or changing ports).
> >In response to the new offer disabling the initial valid m line(s),
> the
> >answerer similarly places the connection and port information for the
> bundle group in the corresponding m line of the SDP answer.
>=20
> I think that's very hacky, moving information from one m- line to=20
> another...
>=20
> >The answerer is not allowed to disable the first valid m line of a=20
> >bundle group in the SDP answer it sends in response to a subsequent=20
> >offer. If the answerer needs to disable the first valid m line(s), it=20
> >must instead accept the m line(s) in response to the latest pending
> offer (if there is one) and immediately send a new SDP offer disabling=20
> the line by following the SDP offerer rule above.
>=20
> I don't think we should have a solution where the answerer is not=20
> allowed to reject an m- line.
>=20
> >The only other rule to add is that the SDP offerer will never re-
> enable
> >(set port to a non-zero value) a disabled m line before the first
> valid
> >m line in a bundle group but will instead create a new m line and add=20
> >it to the group.  This makes it unlikely that the answerer will=20
> >reject
> the first valid m line of the bundle group in a subsequent SDP offer=20
> since it had previously accepted it and did not initiate the new=20
> transaction due to any changes in the configuration of local resources.
>=20
> What if you have a session where a stream is enabled/disabled many=20
> times. If, every time you re-enable, is going to add a new m- line,=20
> you may end up with a very large SDP...
>=20
> >This isn't particularly elegant, but it applies the definition
> consistently and I think it works in all cases.
>=20
> I think it's very hacky, and I don't see how you think that=20
> intermediaries (which your draft focues on) would be able to handle=20
> this...
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
> > Behalf Of Ejzak, Richard P (Richard)
> > Sent: 19. helmikuuta 2013 0:09
> > To: mmusic@ietf.org; rtcweb@ietf.org
> > Subject: [MMUSIC] FW: New Version Notification for draft-ejzak-
> mmusic-
> > bundle-alternatives-00.txt
> >
> > Please note that I have also submitted a draft on how to fix BUNDLE.
> > This doc is based on the modified BUNDLE proposal with different=20
> > port information for each m line and compares several different=20
> > additional modifications to BUNDLE.  The doc ends up recommending=20
> > the use of the unspecified address for subsequent m lines in a=20
> > bundle group in the SDP answer (not offer).  Please see the doc for det=
ails.
> >
> > -----Original Message-----
> > From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> > Sent: Monday, February 18, 2013 3:57 PM
> > To: Ejzak, Richard P (Richard)
> > Subject: New Version Notification for draft-ejzak-mmusic-bundle-=20
> > alternatives-00.txt
> >
> >
> > A new version of I-D, draft-ejzak-mmusic-bundle-alternatives-00.txt
> > has been successfully submitted by Richard Ejzak and posted to the=20
> > IETF repository.
> >
> > Filename:      draft-ejzak-mmusic-bundle-alternatives
> > Revision:      00
> > Title:                 Alternatives to BUNDLE
> > Creation date:         2013-02-18
> > Group:                 Individual Submission
> > Number of pages: 7
> > URL:             http://www.ietf.org/internet-drafts/draft-ejzak-
> > mmusic-bundle-alternatives-00.txt
> > Status:          http://datatracker.ietf.org/doc/draft-ejzak-mmusic-
> > bundle-alternatives
> > Htmlized:        http://tools.ietf.org/html/draft-ejzak-mmusic-
> bundle-
> > alternatives-00
> >
> >
> > Abstract:
> >    This paper discusses some potential modifications to the BUNDLE
> >    proposal for multiplexing of multiple media types within a single
> >    5-tuple to address limitations of the current proposal and
> >    alternatives already considered by MMUSIC.
> >
> >
> >
> >
> > The IETF Secretariat
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic

From richard.ejzak@alcatel-lucent.com  Thu Feb 21 06:52:46 2013
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt
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Subject: Re: [MMUSIC] FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt
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Hi Christer,
Please see below.

Richard

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]


> >My proposal never has a zero port on a valid m line, in either offer
> or
> >answer.  I only use unspecified address (with valid port number) for
> subsequent m lines in answers.  I'm not sure how you got this from the
> description.  Let's discuss this on the call today.  I hope you are
> available.
>=20
> Sorry, I mixed up terminology.
>=20
> But, still, if you want to disable/reject the m- line which contains
> the bundle ADDRESS, you will have to move the address to another m-
> line.

Yes.  As long as there is a simple rule for doing so I don't see a problem.

>=20
> And, still, you expect intermediaries to retrieve media types,
> protocols, payload types etc from the m- lines, which addresses are
> unspecified - but at the same time you say that no resources will be
> reserved for the m- lines with unspecified addresses...

Intermediaries forwarding BUNDLE information have to understand SOMETHING a=
bout BUNDLE or they have no business forwarding it in the first place.  It'=
s only in this case that the intermediary might need to use m lines with un=
specified address to identity additional bandwidth requirements.  I meant t=
hat -port- resources could be released in this case.  I should have been mo=
re careful in describing this point.

Sending a subsequent offer with the same port in all bundled m lines has it=
s own problems in 3pcc scenarios.  You should always send an offer that can=
 pass through intermediaries that do not understand BUNDLE.  My proposal ha=
s that characteristic but I'm not sure that yours does.

>=20
> Regards,
>=20
> Christer

From harald@alvestrand.no  Thu Feb 21 06:59:58 2013
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Subject: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt)
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On 02/21/2013 03:51 PM, Ejzak, Richard P (Richard) wrote:
> Sending a subsequent offer with the same port in all bundled m lines 
> has its own problems in 3pcc scenarios. You should always send an 
> offer that can pass through intermediaries that do not understand 
> BUNDLE. My proposal has that characteristic but I'm not sure that 
> yours does.
Designing around intermediaries that pass along information that they do 
not understand, and then expect to act independently based on the 
message that they partially understood, seems to me to be a quite 
limiting approach.

We discussed this at the interim (I think Cullen was pretty vocal that 
there's a limit to how much brokenness we can design around, and he 
thought intermediaries passing BUNDLE without understanding BUNDLE were 
clearly beyond that limit).

(Cullen, apologies if I misstate what you said).


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From: Suhas Nandakumar <suhasietf@gmail.com>
To: Eric Rescorla <ekr@rtfm.com>
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Cc: rtcweb@ietf.org, mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] Proposed Agenda For WG Meetings
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+1

Looks Neat !!

./Suhas

On Wed, Feb 20, 2013 at 4:27 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> In the tradition of the Interim, I tried to sit down and work out an
> agenda that reflects that I think the important concerns are, namely:
>
> - Video codecs (as promised)
> - Multiplexing of media flows
> - Trickle ICE
> - DataChannels
>
> Two notes about cross-WG interaction:
>
> - I'm assuming that most of the MMUSIC time will be used for
>   RTCWEB-type stuff, though I left 30 min for other MMUSIC
>   business on Tue.
>
> - I don't know how much time AVTEXT needs given the AVTEXT/
>   MMUSIC merger on Fri. We can reduce the draft-jennings-rtcweb-plan
>   time accordingly based on agenda requests.
>
> Hope this is useful. I'm of course happy to get on a call to discuss
> this if that helps.
>
> -Ekr
>
>
> RTCWEB I: Tuesday 0900-1130
> 0900 - 0905  Administrivia
> 0905 - 1100  Video Codec MTI discussion
>   - draft-alvestrand-rtcweb-vp8-00 (30 mins)
>   -
> draft-burman-rtcweb-h264-proposal-00+draft-dbenham-webrtcvideomti-00+draft-marjou-rtcweb-video-codec-00
> (30 mins)
>   - General discussion 30 min)
>   - Call the question of which mandatory to implement video codec to
> select (5 min)
>   - Next steps (20 min)
>  1100 - 1115  draft-ietf-rtcweb-use-cases-and-requirements (LC comments)
>  1115 - 1130  draft-ietf-draft-ietf-rtcweb-overview (LC comments)
>
>
> MMUSIC I: Tuesday 1520-1830
> 1520 - 1525  Administrivia
> 1525 - 1605  Trickle ICE (Ivov)
> - Under what conditions can you do full trickle (discovery)
> - When can checking stop? [UPDATE vs. in-band vs...?]
> - SDP/SIP encoding for additional trickle candidates
> 1605 - 1650  - Bundle
>  - draft-ejzak-mmusic-bundle-alternatives (15 min)
>  - draft-ietf-mmusic-sdp-bundle-negotiation (15 min)
>  - discussion/consensus calls (15 min)
> 1650 - 1700  [Break]
> 1700 - 1800  SDP attribute analysis for multiplexing
>  - draft-nandakumar-mmusic-sdp-mux-attributes
> 1800 - 1830  [Other MMUSIC business]
>
>
> RTCWEB II: Thursday 0900-1130
> 0900 - 0905  Administrivia
> 0905 - 1025  Data Channel negotiation
>  - draft-jesup-rtcweb-data-protocol (15 min)
>  - draft-thomson-rtcweb-data (15 min)
>  - draft-marcon-rtcweb-data-channel-management (15 min)
>  - Discussion (30 min)
>  - Consensus call (5 min)
> 1025 - 1100 JSEP Update (Uberti)
>  - draft-ietf-rtcweb-jsep
> 1100 - 1115  RTCP-XR (Westerlund)
> 1115 - 1130  RTP Concepts and relation
> - daft-burman-rtcweb-mmusic-media-structure
>
>
> MMUSIC II: Friday 0900-1100
> 0900 - 0905  Administrivia
> 0905 - 0925  Requirements (draft-jennings-mmusic-media-req)
> 0925 - 1015  PC-track to m-line mapping (draft-jennings-rtcweb-plan)
> 1015 - 1100  MSID changes to match multiplexing
> (draft-jennings-rtcweb-plan)
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7b86d68cb0cd2204d631801d
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+1 <br><br>Looks Neat !!<br><br>./Suhas<br><br><div class=3D"gmail_quote">O=
n Wed, Feb 20, 2013 at 4:27 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrot=
e:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>In the tradition of the Interim, I trie=
d to sit down and work out an</div><div>agenda that reflects that I think t=
he important concerns are, namely:</div>
<div><br></div><div>- Video codecs (as promised)</div><div>- Multiplexing o=
f media flows</div>

<div>- Trickle ICE</div><div>- DataChannels</div><div><br></div><div>Two no=
tes about cross-WG interaction:</div><div><br></div><div>- I&#39;m assuming=
 that most of the MMUSIC time will be used for</div><div>=A0 RTCWEB-type st=
uff, though I left 30 min for other MMUSIC</div>


<div>=A0 business on Tue.</div><div><br></div><div>- I don&#39;t know how m=
uch time AVTEXT needs given the AVTEXT/</div><div>=A0 MMUSIC merger on Fri.=
 We can reduce the draft-jennings-rtcweb-plan</div><div>=A0 time accordingl=
y based on agenda requests.</div>


<div><br></div><div>Hope this is useful. I&#39;m of course happy to get on =
a call to discuss</div><div>this if that helps.</div><div><br></div><div>-E=
kr</div><div><br></div><div><br></div><div>RTCWEB I: Tuesday 0900-1130</div=
>


<div>0900 - 0905 =A0Administrivia</div><div>0905 - 1100 =A0Video Codec MTI =
discussion</div><div>=A0 - draft-alvestrand-rtcweb-vp8-00 (30 mins)</div><d=
iv>=A0 - draft-burman-rtcweb-h264-proposal-00+draft-dbenham-webrtcvideomti-=
00+draft-marjou-rtcweb-video-codec-00 (30 mins)</div>


<div>=A0 - General discussion 30 min)</div><div>=A0 - Call the question of =
which mandatory to implement video codec to select (5 min)</div><div>=A0 - =
Next steps (20 min)</div><div>=A01100 - 1115 =A0draft-ietf-rtcweb-use-cases=
-and-requirements (LC comments)</div>


<div>=A01115 - 1130 =A0draft-ietf-draft-ietf-rtcweb-overview (LC comments)<=
/div><div>=A0 =A0=A0</div><div><br></div><div>MMUSIC I: Tuesday 1520-1830</=
div><div>1520 - 1525 =A0Administrivia</div><div>1525 - 1605 =A0Trickle ICE =
(Ivov)</div>


<div>- Under what conditions can you do full trickle (discovery)</div><div>=
- When can checking stop? [UPDATE vs. in-band vs...?]</div><div>- SDP/SIP e=
ncoding for additional trickle candidates</div><div>1605 - 1650 =A0- Bundle=
=A0</div>


<div>=A0- draft-ejzak-mmusic-bundle-alternatives (15 min)</div><div>=A0- dr=
aft-ietf-mmusic-sdp-bundle-negotiation (15 min)</div><div>=A0- discussion/c=
onsensus calls (15 min)</div><div>1650 - 1700 =A0[Break]</div><div>1700 - 1=
800 =A0SDP attribute analysis for multiplexing</div>


<div>=A0- draft-nandakumar-mmusic-sdp-mux-attributes</div><div>1800 - 1830 =
=A0[Other MMUSIC business]</div><div><br></div><div><br></div><div>RTCWEB I=
I: Thursday 0900-1130</div><div>0900 - 0905 =A0Administrivia</div><div>0905=
 - 1025 =A0Data Channel negotiation</div>


<div>=A0- draft-jesup-rtcweb-data-protocol (15 min)</div><div>=A0- draft-th=
omson-rtcweb-data (15 min)</div><div>=A0- draft-marcon-rtcweb-data-channel-=
management (15 min)</div><div>=A0- Discussion (30 min)</div><div>=A0- Conse=
nsus call (5 min)</div>


<div>1025 - 1100 JSEP Update (Uberti)</div><div>=A0- draft-ietf-rtcweb-jsep=
</div><div>1100 - 1115 =A0RTCP-XR (Westerlund)</div><div>1115 - 1130 =A0RTP=
 Concepts and relation</div><div>- daft-burman-rtcweb-mmusic-media-structur=
e</div>


<div><br></div><div><br></div><div>MMUSIC II: Friday 0900-1100</div><div>09=
00 - 0905 =A0Administrivia</div><div>0905 - 0925 =A0Requirements (draft-jen=
nings-mmusic-media-req)</div><div>0925 - 1015 =A0PC-track to m-line mapping=
 (draft-jennings-rtcweb-plan)</div>


<div>1015 - 1100 =A0MSID changes to match multiplexing (draft-jennings-rtcw=
eb-plan)</div><div><br></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

--047d7b86d68cb0cd2204d631801d--

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt)
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Subject: Re: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt)
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Hi Harald,
I did not mean that an intermediary should forward attributes it doesn't un=
derstand but that an intermediary should be able to deal with any SDP it re=
ceives.  If some intermediaries choke on having the same port in multiple m=
 lines then we should avoid doing that, even during subsequent offers since=
 there are multiple 3pcc scenarios where subsequent offers can flow through=
 different intermediaries.

We had a lot of discussion of this and other aspects during the call today.=
  We have more work to do before Orlando.

Richard

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Thursday, February 21, 2013 9:00 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Requiring support for nonconformant intermediate
> systems (Re: FW: New Version Notification for draft-ejzak-mmusic-
> bundle-alternatives-00.txt)
>=20
> On 02/21/2013 03:51 PM, Ejzak, Richard P (Richard) wrote:
> > Sending a subsequent offer with the same port in all bundled m lines
> > has its own problems in 3pcc scenarios. You should always send an
> > offer that can pass through intermediaries that do not understand
> > BUNDLE. My proposal has that characteristic but I'm not sure that
> > yours does.
> Designing around intermediaries that pass along information that they
> do not understand, and then expect to act independently based on the
> message that they partially understood, seems to me to be a quite
> limiting approach.
>=20
> We discussed this at the interim (I think Cullen was pretty vocal that
> there's a limit to how much brokenness we can design around, and he
> thought intermediaries passing BUNDLE without understanding BUNDLE were
> clearly beyond that limit).
>=20
> (Cullen, apologies if I misstate what you said).
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03 - Charles' comments
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Subject: Re: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-negotiation-03 - Charles' comments
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, February 20, 2013 11:10 PM
> To: Charles Eckel (eckelcu); mmusic@ietf.org
> Subject: RE: [MMUSIC] Draft new version: draft-ietf-mmusic-sdp-bundle-
> negotiation-03 - Charles' comments
>=20
> Hi Charles,
>=20
> Thanks for your comments! Reply inline.
>=20
> >I read the draft and I think it is a good step forward. I like the overa=
ll
> approach and think it can work. I do have questions about some of the
> specifics, and also some suggested changes and additions.
> >
> >1) In section 1, the 4th paragraph ends as follows:
> >
> >   The port number
> >   value associated with the first "m=3D" line associated with the
> >   "BUNDLE" group represents the port value number offered to be used in
> >   the single 5-tuple.
> >
> >I think this is no longer necessary per the rules in section 6.1, and it=
 should
> be removed.
>=20
> The text talks about the Offer where different port numbers are used, ie
> when it is still not known whether the Answerer supports bundle. But,
> maybe that needs to be more clear.

I understand that. However, in that case, it seems to me that the port numb=
er of the first "m=3D" line no longer has any special significance because =
each "m=3D" line has its own port number, and if another offer is sent to m=
ake all the ports the same, that port number may be different than any of t=
he port numbers in the initial offer.
=20
> --------------------------
>=20
> > 2) Section 2 says the following about RTCP:
> >
> >   Bundled media: Two or more RTP streams using a single 5-tuple.  The
> >   RTCP streams associated with the RTP streams also use a single
> >   5-tuple, which might be the same, but can also be different, as the
> >   one used by the RTP streams.
> >
> >This is contradicted by section 6.1, which says the SDP offer MUST
> >   o  5.  Insert an SDP 'rtpc-mux' attribute.
> >
> > So one of these needs to be changed, and replace 'rtpc-mux' with 'rtcp-
> mux' :)
>=20
> Correct. I got the same comment off-line from Harald, but forgot to fix i=
t.
>=20
> Do people have any preference? Shall du usage of rtcp-mux be a SHOULD, a
> MUST, or a RECOMMEND? :)

According to draft-ietf-rtcweb-rtp-usage, it is mandatory to support both m=
uxed and non-muxed, so SHOULD seems most appropriate to me. What is importa=
nt is that rtcp-mux is used or not use consistently across all BUNDLED "m=
=3D" lines.

> --------------------------
>=20
> >3)   OPEN ISSUE #1: If the SDP Answerer supports "BUNDLE", do we
> mandate
> >   that "m=3D" lines must be grouped in the same way in the SDP Answer a=
s
> >   they are grouped in the SDP Offer?
> >
> > I'd say yes, because otherwise you are not sure if something is wrong (=
e.g.
> an intermediary has changed the ports, or something
> > does not really understand bundle). I think if the ports in the SDP ans=
wer
> are not grouped the same as in the offer, it should be
> > interpreted as a failure to bundle, and another offer should be sent to=
 use
> different ports.
>=20
> I'm still thinking about it, but at least at this point I would agree wit=
h you :)
>=20
> --------------------------
>=20
> >4) I do not understand the intent of the following text in section 6.2:
> >
> >   However, if the "m=3D" lines associated with the "BUNDLE"
> >   group contain different port number values (i.e. the SDP Offer is
> >   generated until it is known whether the SDP Answerer supports the
> >   "BUNDLE" grouping extension) the SDP Offerer MUST NOT set the port
> >   number value of the first "m=3D" line associated with the "BUNDLE"
> >   group to zero.
> >
> > Perhaps there is a typo, but in any case, I do not think the first "m=
=3D" lines
> needs to be treated specially any more. This is similar to comment (1).
>=20
> Keep in mind that, in this case, only the port of the first "m=3D" line
> represents the bundle port (as different port values are given for the ot=
her
> m=3D lines).

If the "m=3D" lines have different ports, my thinking is that no special me=
aning is tied to any of them. In a subsequent offer/answer exchange where t=
he ports of all the bundled "m=3D" lines are set to the same value, that va=
lue need not be the same as what was in the previous offer that used separa=
te ports for each "m=3D" line.

For example, if the initial offer is:


       v=3D0
       o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.com
       s=3D
       c=3DIN IP4 host.atlanta.com
       t=3D0 0
       a=3Dgroup:BUNDLE foo bar
       m=3Daudio 10000 RTP/AVP 0 8 97
       a=3Dmid:foo
       b=3DAS:200
       a=3Drtpmap:0 PCMU/8000
       a=3Drtpmap:8 PCMA/8000
       a=3Drtpmap:97 iLBC/8000
       m=3Dvideo 10002 RTP/AVP 31 32
       a=3Dmid:bar
       b=3DAS:1000
       a=3Drtpmap:31 H261/90000
       a=3Drtpmap:32 MPV/90000

The subsequent offer may be:


       v=3D0
       o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.com
       s=3D
       c=3DIN IP4 host.atlanta.com
       t=3D0 0
       a=3Dgroup:BUNDLE foo bar
       m=3Daudio 20000 RTP/AVP 0 8 97
       a=3Dmid:foo
       b=3DAS:200
       a=3Drtpmap:0 PCMU/8000
       a=3Drtpmap:8 PCMA/8000
       a=3Drtpmap:97 iLBC/8000
       m=3Dvideo 20000 RTP/AVP 31 32
       a=3Dmid:bar
       b=3DAS:1000
       a=3Drtpmap:31 H261/90000
       a=3Drtpmap:32 MPV/90000
=20
> --------------------------
>=20
> >5)   OPEN ISSUE #2: If the session has been established without BUNDLE, =
do
> >   we allow BUNDLE to be enabled later during the session?
> >
> > Yes, but need to use different ports until it is known that both offere=
r and
> answerer support bundle.
>=20
> Sure, that rule would still apply.
>=20
> --------------------------
>=20
> >6)   OPEN ISSUE #3: If the session has been established with BUNDLE, do =
we
> >   allow BUNDLE to be disabled later during the session, meaning that
> >
> > Yes, for failure recovery.
>=20
> I was more thinking of the case where endpoints already have successfully
> negotiated the usage of bundle, and then at some point during the session
> chose (for whatever reason) to disable the multiplexing.
>=20
> Basically it would mean sending a new offer, without BUNDLE, and with
> different port number values - ie a "legacy"  offer :)

Yes.
=20
> --------------------------
>=20
> >7) Section 6.3 ends as follows:
> >
> >   Until the SDP Answerer receives the new SDP Offer, which contains an
> >   identical port number value for each "m=3D" line associated with a
> >   "BUNDLE" group, it MUST use the port, and related information (ICE
> >   candidates, SDES keys etc) associated with the first "m=3D" line in t=
he
> >   "BUNDLE" group.
> >
> > I think the SDP Answerer needs to wait for the next offer and not assum=
e
> the first "m=3D" line contains the port to be used for the bundle.
>=20
> Is there a specific reason why you think that we can't mandate the Offere=
r
> to insert the bundle port in the first "m=3D" line?

Because it gets confusing if the first bundled "m=3D" line in the SDP offer=
 is rejected in the first SDP answer. I just think it is cleaner to wait fo=
r the next offer. I suppose it is less efficient, so I agree there are trad=
eoffs.
=20
> --------------------------
>=20
> >8) Section 10
> >
> >The SDP Answer shows the same port being used in the "m=3D" lines. I was
> thinking this first answer would use different ports, and only if
> >there is another offer/answer exchange would both the offer and the
> answer switch to using the same port.
>=20
> My personal opinion is that the answerer already knows that the offerer
> wants to use bundle, and is expected to send a new offer. Also, by using =
the
> same ports already in the first answer, we would prevent a case where the
> answerer does NOT support bundle, but blindly copies the grouping
> attributes from the offer to the answer.

If you wait until the second offer/answer exchange to set the ports to be t=
he same, you still get this property. Also by waiting until the second offe=
r/answer exchange to make the ports the same you increase the odds of the i=
nitial offer/answer completing correctly end to end.
=20
> But, opinions are welcome :)

Yes, as I am not all the convinced as to which way is better yet either.

Cheers,
Charles


> Regards,
>=20
> Christer

From fluffy@cisco.com  Thu Feb 21 11:30:41 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
Thread-Topic: Cullen Notes from ports call today.  
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My random notes more for myself than anyone else =85

Christer explained new draft is combination between old and new. If you don=
't each side support bundle, use diff port number in offer, after that you =
use same with=20

Question about 3pcc use cases. Not clear exactly what cases because some of=
 the B2BUA cases, it's not an issues.  Dale  point at music on hold MOH cas=
e, draft-worrley-service-example. To me this looks like a B2BUA that is off=
ering SDP that it does not understand. If an INVITE with replaces was used =
instead of re-INVITE in same dialog seems like this would work fine.=20

Some people thought that every offer/answer re was in depended of previous =
one. People gave example where is just not this way. For example, ICE can n=
ot restarted on updated offer, m line can be offer.=20

Richard pointed out this is lesser of evils, suggestion is initial offer di=
ffers ports, answer if it does not want to bundle, use diff ports. If answe=
r wants to do multiplexing, then 5 tuple used is first valid non zero m-lin=
e is used for the entire bundle. All others are disabled by port=3D0 or con=
nection address is unspecified. Still have bundle group to group these all =
together. For subsequent offer answers, does the offer have candidates for =
all m-lines or just one valid line. Revised idea, the port used by offerer,=
 is first valid m line in offer, the port used in answer, is first valid po=
rt in answer. May not be the same m-line. Middle boxes would need to know a=
bout this to have any hope of working. Problem that the answer can't disabl=
e some m-line's in subsequent o/a.=20

Open issue is does a subsequent offer in a dialog has different or same por=
ts.=20

how to have m-line that are only used if you do bundle?


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Subject: [MMUSIC] Submitted an update to the Traffic Class Label (TCL) attribute
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MMUSIC

I'm letting the WG know that we've updated the trafficclass attribute 
draft to -03.

Abstract:
    This document proposes a new Session Description Protocol (SDP)
    attribute to identify the traffic class a session is requesting
    in its offer/answer exchange.

The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-traffic-class-for-sdp

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-traffic-class-for-sdp-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-traffic-class-for-sdp-03

Here are the changes listed in Appendix 1 from the above URL

A.1  From -02 to -03

    These are the following changes made between the WG -02 version and
    the -03 version:

    - Rearranged a fair amount of text

    - Separated and defined the components into separate subsections.

    - built 5 different tables, one per category, that lists within each
      category - what applications are appropriate as well as what
      adjectives are appropriate for each application within that
      category.

Other stuff we did (not mentioned in the appendix), that we also changed:

- We also rather significantly modified the IANA considerations section,
   including where the bar is to register different types of components.
   I.e., "category" takes a 'PS RFC', the "application" and "adjective" take
   'spec required', but we're thinking of making this only need 'expert
    review'. What does the WG think of this?

- We worked with Paul K on the ABNF, which he likes now.

- created a "partial" admission qualifier for endpoints that send more
   traffic than had capacity admission applied to the session.

- we want to learn about any traffic flows we aren't considering that
   we should be.

- the rearranging made many little things visible that needed fixing,
   adding or deleting.

- what we have *not* addressed is Dan Wing's desire for a label to address
   his sensor traffic, or put another way, should we have a
   periodic/infrequent/sporadic-like non-media 'category'?. This is
   something that makes sense to the authors, but we want to hear from the
   WG on this... we're having problems coming up with a proper category name.

Comments and questions are appreciated.

James/Paul/Subha


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From: Cullen Jennings <fluffy@iii.ca>
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
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Subject: [MMUSIC] Plan draft discussed in boston
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At the Boston interim, I discussed putting together a few drafts.=20

First the requirements around the media stuff at=20
http://tools.ietf.org/html/draft-jennings-mmusic-media-req


The analysis of how bundle multiplexing would work with existing SDP =
attributes=20
http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes

This draft lays out the framework and takes a first pass at all the =
classifications but it some of the attributes, particularly harder =
things like cap neg, need some work. I suspect trying to get the authors =
of the original attribute drafts to review it would work well. Suhas and =
I did a lot of reading to get this to this state but it is a huge amount =
of work to analysis all of these.=20


Finally a proposed plan for a way forward the media related issues at =20=

http://tools.ietf.org/html/draft-jennings-rtcweb-plan

I plan to make some changes to plan before the -01 deadline so let me =
know if you think there are things that should be different.=20

Cullen



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] Signaling trickle support
Thread-Index: AQHOEKoGoTjVL1kZDE2wMFT9eg1eLA==
Date: Fri, 22 Feb 2013 03:09:28 +0000
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org> <509D1200.3070702@cisco.com> <50A20CE5.5000200@alvestrand.no>
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On Nov 13, 2012, at 2:03 AM, Harald Alvestrand <harald@alvestrand.no> wrote=
:

>> Because of the min. 20 ms Ta timer constraints, which apply cross-media =
stream ? Can somebody refresh my mind as to why that is (e.g. are residenti=
al NATs at this point generally not capable of creating NAT bindings faster=
 than at 20 ms intervals) ?
> I have been told that in BEHAVE testing, NATs were found that would behav=
e incoherently at short binding creation intervals, but none were found tha=
t couldn't handle one new binding every 20 ms, so this was set at the minim=
um interval for potentially binding-creation actions.
>=20
>            Harald

Some of the testing was getting the programs working for the results in htt=
p://tools.ietf.org/html/draft-jennings-midcom-stun-results-02 but other stu=
ff done at Cornel and I think MIT came up with similar results.=20



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Flemming Andreasen (fandreas)" <fandreas@cisco.com>
Thread-Topic: [MMUSIC] Signaling trickle support
Thread-Index: AQHOEKpDpXIC0D/b6U6TColU2zjO2Q==
Date: Fri, 22 Feb 2013 03:11:10 +0000
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References: <CABkgnnVR-prV8xgzPx9H-MYRM_vTui-8GGfTD0yy1JPvJJzrrQ@mail.gmail.com> <509AB627.1090507@alum.mit.edu> <509B5F87.1010807@ericsson.com> <7594FB04B1934943A5C02806D1A2204B027800@ESESSMB209.ericsson.se> <509BC947.709@cisco.com> <509BCBFB.2020305@jitsi.org> <CABkgnnVMf53Vk_Rd-0erZN4ZLDKM4ePOvMfLJnfOUNVxdWYLqQ@mail.gmail.com> <509BD563.9080808@cisco.com> <509BDE01.9060705@jitsi.org> <509C02F2.10303@cisco.com> <509C0724.3050300@jitsi.org> <509C3066.6020209@cisco.com> <509C3F33.8060809@jitsi.org> <509D1200.3070702@cisco.com> <50A20CE5.5000200@alvestrand.no> <072901cdc1bd$60d2cd60$22786820$@cisco.com> <50A289F2.1030004@cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, "Dan Wing \(dwing\)" <dwing@cisco.com>
Subject: Re: [MMUSIC] Signaling trickle support
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On Nov 13, 2012, at 10:57 AM, Flemming Andreasen <fandreas@cisco.com> wrote=
:

> I've been told (second hand) that Google Maps routinely opens 50 TCP conn=
ections at a time, but don't have any details as to the speed with which th=
ey are opened. Anybody have any further insights or similar examples (that =
seem to work in practice) ?

Google maps is a bad example as it regularly has this fail in many location=
s - when it fails, there are whole strips of the map that do not load. So i=
t is pretty much an example of something that is on the fairly aggressive l=
evel.=20


From worley@shell01.TheWorld.com  Thu Feb 21 19:19:08 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [MMUSIC] Requirements for bundling
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Here are the requirements (desiderata, really) from the latest version
of my draft, draft-worley-sdp-bundle.  No doubt they're not complete.

3.  Desiderata

   This section lists desiderata for the bundle mechanism in SDP.  (I
   use the term "desiderata" -- "things that are desired" -- rather than
   "requirements", because we may discover that we can't optimally
   satisfy all of these criteria at the same time.)  The first section
   lists desiderata that are arise from considering the ways
   applications may wish to bundling.  The second section lists
   desiderata that arise from compatibility with existing Internet
   infrastructure.

3.1.  Feature Desiderata

   DES F1  For each bundle, there is a group of media descriptions which
      describe the application-level RTP sessions.  This specification
      must allow the same granularity of description as when the media
      flows were not multiplexed.  This description includes identifiers
      which connect the media flows with the application and with each
      other.

   This requirement is taken from draft-jennings-mmusic-media-req-00.

   DES F2  For each bundle, there is a media description that describes
      the transport-level RTP session.

   DES F1 and DES F2 do not specify whether the transport-level media
   description may or may not also be one of the application-level media
   descriptions.

   DES F3  There must be a uniform way to deal with new SDP parameters,
      so that newly defined SDP parameters do not require a specific
      updating of the bundling procedures.

   This desideratum is taken from slides-interim-2013-rtcweb-1-10.pdf.

   DES F4  Multiple separate bundles within one SDP must be supported.

   DES F5  Bundles may contain other bundles as constituents.

   Of course, no bundle may directly or indirectly contain itself.  (I
   don't expect any current implementation to implement bundles within
   bundles, but we should design the mechanism to allow this, as some
   day we will likely need it.)

   DES F6  A bundle may contain zero constituents.

   A bundle with no constituents serves no purpose for the transport of
   media, but we are likely to someday need to describe such a bundle.
   (Compare that an SDP m= line is syntactically constrained to specify
   at least one payload type.  When SDP was used only to specify
   multicast sessions, this constraint was common sense.  But once SDP
   offer/answer was invented, when a media description was rejected, the
   natural representation would be an m= line with a zero port and no
   payload types.  But a payload type was syntactically required, so we
   now have to provide at least one token payload type in rejected m=
   lines.)

   DES F7  If an answerer that does understand the bundle mechanism
      processes an offer that contains a bundle, it must be able to (1)
      accept the bundle and selectively accept or reject each
      constituent RTP session within it, (2) reject the bundle as a
      whole, or (3) reject the bundling and selectively accept or reject
      each constituent RTP session as separate RTP sessions.

   Presumably answer (3) resembles that which would be produced by an
   answerer that does not understand the bundle mechanism.  It is a
   lower priority that the answerer can distinguish between accepting
   the bundle while rejecting all of its constituents, and rejecting the
   bundle as a whole.  But those two conditions differ conceptually
   regarding whether any "framing" actions of the bundle are performed.

   DES F8  There must be a reliable way to demultiplex incoming RTP into
      the separate application-level RTP sessions.  Similarly, there
      must be a reliable way to demultiplex the associated RTCP
      information.

   The RTCP information for each media stream is tagged with the SSRC
   about which it reports, and the SSRC is used to correlate the RTCP
   reports with the RTP sessions containing media with the same SSRC.
   So regarding RTCP, this desideratum appears to be straightforward to
   satisfy.

   DES F9  The specification must specify any needed additional
      procedures for handling SSRC collisions between media sources
      within different application-level RTP sessions, as those can now
      collide.

   In the terminology of RFC 3550, the constituent media descriptions
   are now part of one RTP session.

   DES F10  When an offer is constructed, the offerer must not need to
      pre-allocate TURN relays for constituent media descriptions.  When
      both endpoints support bundling, the mechanism must not require
      the offerer to allocate TURN relays for constituent media
      descriptions.

   This desideratum was suggested by Andrew Hutton.

   DES F11  It must be possible to add and remove one way video flows
      within the bundle without requiring an additional offer/answer
      cycle.

   Presumably this can be accomplished as it is now, with a single media
   description carrying multiple video flows that are distinguished only
   by their SSRCs.  This desideratum is taken from slides-interim-2013-
   rtcweb-1-10.pdf.

   DES F12  Bundling must not interfere with ICE usage, and in
      particular, ICE's ability to negotiate both IPv4 and IPv6
      addresses simultaneously.

   This desideratum was suggested by Andrew Hutton.

3.2.  Compatibility Desiderata

   DES C1  In offer/answer usage, an endpoint using the bundle mechanism
      must interwork correctly with an endpoint that does not understand
      the bundle mechanism.

   DES C2  Interworking must continue when SDP endpoints are replaced
      with other endpoints during a sequence of offer/answer exchanges
      (such as happens in 3PCC or call transfers "behind an SBC"),
      including when a supporting endpoint is replaced by a non-
      supporting endpoint or vice-versa.

   SDP features (e.g., codec set and ICE) are generally designed so that
   an offerer offers every facility it is willing to support in the
   current instance.  Thus, if the answerer is a different endpoint than
   the previous answerer, the new answerer will negotiate a compatible
   set of facilities without needing knowledge of its predecessor's SDP.
   The offerer will smoothly transition to the new facilities.  This
   property is required to support 3PCC situations (e.g., RFC 3725 and
   draft-worley-service-example).  This desideratum was suggested by
   Richard Ejzak.

   DES C3  Avoid using media types in m= lines other than audio and
      video unless required for user media, as some SBCs reject SDP that
      uses other media types.

   This desideratum was suggested by Hadriel Kaplan.

   DES C4  Any additional m= lines prescribed by the bundle mechanism
      should be ordered after the constituent m= lines.

   Many devices that have only one audio or video channel accept the
   first m= line with that media type and reject any further ones

   non-DES C5  SBCs generally pass through attributes that they do not
      understand.  SBCs generally pass through codec specifications that
      they do not understand, even if they are configured to transcode
      certain specific codecs.

   This non-desideratum was suggested by Hadriel Kaplan.

   DES C6  After offer/answer processing is finished, if the exchanged
      SDP is examined by a non-supporting SBC, the set of transport
      associations that it sees being specified for media exchange
      should be the set that are actually used for media transfer.

   This is needed because SBCs monitor the packet traffic on the
   transport associations and if no media is seen on one of the
   associations for a significant period of time, the SBC will tear down
   the call.  This desideratum was suggested by Hadriel Kaplan.

   DES C7  In a session description, no endpoint of a transport
      association may be used multiple times.

   Such duplication is not defined by [sdp].  Some SBCs do not support
   such duplication (ultimately, because it was not supported by [RFC
   2327]), and they reject SDP specifying duplicated transport
   association endpoints.  This desideratum was suggested by Cullen
   Jennings.

   DES C8  Offer/answer processing between supporting processors must be
      completed in one exchange.  When interworking between supporting
      and non-supporting processors, it is less desirable but admissible
      that a second offer/answer exchange may be needed to complete
      configuring the multimedia session.

Dale

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
Thread-Topic: [MMUSIC] Updating the pacing of ICE connectivity checks
Thread-Index: AQHOEKxK/muj19GG2UO6zABx16NwAw==
Date: Fri, 22 Feb 2013 03:25:40 +0000
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Subject: Re: [MMUSIC] Updating the pacing of ICE connectivity checks
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This is a hard problem to figure out but glad it is being tackled as the cu=
rrent draft was fiction created to get it approved more.

I'd be tempted to think of their being a few different things to conserve o=
r constraints we need to not exceed=20

1) nat timing=20

2) total ports in usage on the NAT

3) bandwidth usage that is not congestion controlled=20

For 1) I'm not seeing any reasons to move off the 20 ms lower bound - but i=
f someone has empirical data that would be nice as the empirical data we ar=
e basing this on is old.=20

For 2) some Carrier Grade Nats (CGNs) are restricting a given client to som=
ething like 1000 ports. There may be other limits too. It might be worth se=
tting up ICE so that a given ICE client never used more than 100 ports at a=
 time. I think this is a new type of constraint that is not a big deal toda=
y but may be a bigger deal in the future and we should consider it in an IC=
E bis.=20

For 3) Its very hard to come up with something that is fast enough that the=
 media guys like it and slow enough the transport folks are sure it is safe=
. My strawman proposal would be use more more bandwidth than G.711 with 20 =
ms packets as we have been doing that all over the place - without congesti=
on control  - and the internet has not melted down.=20




On Feb 14, 2013, at 6:49 AM, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote=
:

> Hi all,
>=20
> Updating the pacing parameter, Ta, of ICE for non-RTP traffic (for backgr=
ound, see [1] and [2]), as discussed at the Atlanta meeting, will be one of=
 the important changes in the new version of ICE.
>=20
> Currently the RFC5245 says (for non-RTP traffic) "Ta SHOULD be configurab=
le, SHOULD have a default of 500 ms, and MUST NOT be configurable to be les=
s than 500 ms."
>=20
> This results in very slow pace of starting connectivity checks if ICE is =
used with non-RTP traffic. With RTP, the Ta is calculated based on the expe=
cted bandwidth used by the RTP stream and can be down to 20 ms. Therefore, =
many current implementations apparently ignore the "MUST NOT" above and set=
 Ta to something lower.
>=20
> The quite conservative value of 500 ms was originally chosen because of t=
he fear of congesting the links with the connectivity check messages. This =
is obviously still a valid concern, but would suggesting something lower st=
ill make sense? How low would it be reasonably safe to go? What is currentl=
y used in practice? Can we come up with a formula similar to RTP (perhaps b=
ased on expected bandwidth usage)?
>=20
>=20
> Cheers,
> Ari
>=20
> [1] http://tools.ietf.org/html/rfc5245#appendix-B.1
> [2] http://tools.ietf.org/html/rfc5245#section-16.2
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Fri Feb 22 03:49:10 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "Harald Alvestrand" <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt)
Thread-Index: AQHOEEQhUP7WLYIH+E+EYYvRtN1DsJiEg1+AgAE9WEA=
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Subject: Re: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt)
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Hi,

>I did not mean that an intermediary should forward attributes it doesn't u=
nderstand but that an intermediary should be able to deal with any=20
>SDP it receives.  If some intermediaries choke on having the same port in =
multiple m lines then we should avoid doing that, even during=20
>subsequent offers since there are multiple 3pcc scenarios where subsequent=
 offers can flow through different intermediaries.

The BUNDLE solution expects the intermediary to accept the same port number=
.=20

Your solution expects the intermediary to understand that, eventhough the a=
ddresses for all-but-one m- lines are unspecified, media associated with th=
ose m- lines will still be sent - using the port of another m- line.

I still strongly think that sending media associated with m- lines, for whi=
ch the address is unspecified, is a design error.

(It is true that we need to expect that, or somehow ensure that, an interme=
diary calculates the correct bandwidth for bundled media, but that applies =
to both solutions).

Regards,

Christer



> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
> Behalf Of Harald Alvestrand
> Sent: Thursday, February 21, 2013 9:00 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Requiring support for nonconformant intermediate=20
> systems (Re: FW: New Version Notification for draft-ejzak-mmusic-
> bundle-alternatives-00.txt)
>=20
> On 02/21/2013 03:51 PM, Ejzak, Richard P (Richard) wrote:
> > Sending a subsequent offer with the same port in all bundled m lines=20
> > has its own problems in 3pcc scenarios. You should always send an=20
> > offer that can pass through intermediaries that do not understand=20
> > BUNDLE. My proposal has that characteristic but I'm not sure that=20
> > yours does.
> Designing around intermediaries that pass along information that they=20
> do not understand, and then expect to act independently based on the=20
> message that they partially understood, seems to me to be a quite=20
> limiting approach.
>=20
> We discussed this at the interim (I think Cullen was pretty vocal that=20
> there's a limit to how much brokenness we can design around, and he=20
> thought intermediaries passing BUNDLE without understanding BUNDLE=20
> were clearly beyond that limit).
>=20
> (Cullen, apologies if I misstate what you said).
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
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From christer.holmberg@ericsson.com  Fri Feb 22 03:51:15 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "mmusic@ietf.org (E-mail)" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Cullen Notes from ports call today.
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Subject: Re: [MMUSIC] Cullen Notes from ports call today.
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>how to have m-line that are only used if you do bundle?

There is a solution alternative for that, called MMT :)

Regards,

Christer


From richard.ejzak@alcatel-lucent.com  Fri Feb 22 06:41:46 2013
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt)
Thread-Index: AQHOEEQk5dAw9HDOIE68f+nShlOa4piEkZrAgAGG5AD//9XbgA==
Date: Fri, 22 Feb 2013 14:41:05 +0000
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References: <03FBA798AC24E3498B74F47FD082A92F36EA6762@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B0FC32F@ESESSMB209.ericsson.se>, <03FBA798AC24E3498B74F47FD082A92F36EA6AA6@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B0FEDDE@ESESSMB209.ericsson.se> <03FBA798AC24E3498B74F47FD082A92F36EA6C8F@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B0FF4F0@ESESSMB209.ericsson.se> <03FBA798AC24E3498B74F47FD082A92F36EA6CDB@US70UWXCHMBA04.zam.alcatel-lucent.com> <51263661.3060307@alvestrand.no> <03FBA798AC24E3498B74F47FD082A92F36EA6DB3@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B1049E9@ESESSMB209.ericsson.se>
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Subject: Re: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt)
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Hi Christer,

> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]

> Your solution expects the intermediary to understand that, eventhough
> the addresses for all-but-one m- lines are unspecified, media
> associated with those m- lines will still be sent - using the port of
> another m- line.

But this requirement is only placed on intermediaries that have already ind=
icated support for BUNDLE by forwarding the related attributes.  The unspec=
ified address only needs to be included in the SDP answer, and not the offe=
r, so it is returned within an established dialog, and there is no danger o=
f sending it to an unknown intermediary.

>=20
> I still strongly think that sending media associated with m- lines, for
> which the address is unspecified, is a design error.

I'm not sure how the term "design error" applies to a solution that behaves=
 correctly and consistently under the given assumptions (there are other as=
pects to be fixed but bear with me for sake of argument).  It is true that =
I propose to clarify the meaning of unspecified address in the presence of =
BUNDLE, which is a new and as yet non-standardized concept.  There are many=
 examples of such "clarifications" as we expand the capabilities that can b=
e represented in SDP.  The key is backward compatibility.  If the concern i=
s about intermediaries that forward BUNDLE attributes without understanding=
 them, then there is little that can be done anyway.

Richard

>=20
> (It is true that we need to expect that, or somehow ensure that, an
> intermediary calculates the correct bandwidth for bundled media, but
> that applies to both solutions).
>=20
> Regards,
>=20
> Christer


From wwwrun@rfc-editor.org  Fri Feb 22 12:27:24 2013
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Cc: mmusic@ietf.org, rfc-editor@rfc-editor.org
Subject: [MMUSIC] RFC 6849 on An Extension to the Session Description Protocol (SDP) and Real-time Transport Protocol (RTP) for Media Loopback
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6849

        Title:      An Extension to the Session 
                    Description Protocol (SDP) and Real-time Transport 
                    Protocol (RTP) for Media Loopback 
        Author:     H. Kaplan, Ed.,
                    K. Hedayat, N. Venna,
                    P. Jones, N. Stratton
        Status:     Standards Track
        Stream:     IETF
        Date:       February 2013
        Mailbox:    hkaplan@acmepacket.com, 
                    kh274@cornell.edu, 
                    vnagarjuna@saperix.com,
                    paulej@packetizer.com, 
                    nathan@robotics.net
        Pages:      33
        Characters: 66214
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-mmusic-media-loopback-27.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6849.txt

The wide deployment of Voice over IP (VoIP), real-time text, and Video 
over IP services has introduced new challenges in managing and maintaining
real-time voice/text/video quality, reliability, and overall
performance.  In particular, media delivery is an area that needs
attention.  One method of meeting these challenges is monitoring
the media delivery performance by looping media back to the
transmitter.  This is typically referred to as "active monitoring"
of services.  Media loopback is especially popular in ensuring the
quality of transport to the edge of a given VoIP, real-time text, or
Video over IP service.  Today, in networks that deliver real-time
media, short of running 'ping' and 'traceroute' to the edge,
administrators are left without the necessary tools to actively
monitor, manage, and diagnose quality issues with their service.
The extension defined herein adds new Session Description Protocol
(SDP) media types and attributes that enable establishment of media
sessions where the media is looped back to the transmitter.  Such
media sessions will serve as monitoring and troubleshooting tools
by providing the means for measurement of more advanced VoIP,
real-time text, and Video over IP performance metrics.  

This document is a product of the Multiparty Multimedia Session Control Working Group of the IETF.

This is now a Proposed Standard.

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

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From worley@shell01.TheWorld.com  Fri Feb 22 13:06:03 2013
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: mmusic@ietf.org
Subject: [MMUSIC] Another bundling proposal
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I've written an alternative bundling proposal which I think manages
the problems of bundling better.

A series of tutorial examples are here:
http://tools.ietf.org/html/draft-worley-sdp-bundle-02#section-4
so it's easy to see how the proposal works.

Distinctive features of this proposal are:

- A separate m= line describes the bundled media, allowing unambiguous
  attribution of attributes to either the bundle or a constituent, and
  independent rejection of each constituent.

- Backward compatibility is achieved straightforwardly.

- Multiplexed media flows are attributed to their original constituent
  media descriptions without requiring each to be declared
  individually in the SDP.

- The multiplexed media flow is artificially given the media type
  "audio" to prevent rejection by SBCs.

- An encapsulating RTP format is used so demultiplexing can be done
  without coordinating payload types between media descriptions, and
  to ensure the bundled media description is rejected by
  non-supporting endpoints.

- Zero/non-zero ports and valid/null addresses are controlled to avoid
  duplicate port usage, and to ensure SBCs see exactly the media flows
  specified by the SDP.



Filename:	 draft-worley-sdp-bundle
Revision:	 02
Title:		 Kumquat: A Generic Bundle Mechanism for the Session Description Protocol (SDP)
Creation date:	 2013-02-22
Group:		 Individual Submission
Number of pages: 27
URL:             http://www.ietf.org/internet-drafts/draft-worley-sdp-bundle-02.txt
Status:          http://datatracker.ietf.org/doc/draft-worley-sdp-bundle
Htmlized:        http://tools.ietf.org/html/draft-worley-sdp-bundle-02
Diff:            http://www.ietf.org/rfcdiff?url2=draft-worley-sdp-bundle-02

Abstract:
   This document defines a generic bundle mechanism for the Session
   Description Protocol (SDP) by which the media described by a number
   of media descriptions ("m= lines") are multiplexed and transmitted
   over a single transport association.  The transport association is
   described by an additional media description, allowing SDP attributes
   to be applied to the aggregate, independently of attributes applied
   to the constituents.  In offer/answer usage, the bundle mechanism is
   backward compatible with SDP processors that do not understand the
   mechanism.  The mechanism is designed to be compatible with the
   limitations of the existing Internet infrastructure.

From eckelcu@cisco.com  Fri Feb 22 13:39:26 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Dale R. Worley" <worley@ariadne.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Another bundling proposal
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Subject: Re: [MMUSIC] Another bundling proposal
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Hi Dale,

Your proposal has some very nice properties; however, my concern with the o=
verall approach is that I fear the initial offer will not be handled very w=
ell by typical pre-BUMDLE implementations. Due to the port being '9' and th=
e connection address being 0.0.0.0, I would expect either:
1) a re-INVITE is needed for things to be renegotiated (best case)
2) the call fails (worst case)
Of course other variations between the best and worst case could happen, bu=
t the odds of arriving at the worst case seem unacceptably high to me. That=
 is just my 2 cents without actually testing anything to validate it. I'd l=
ike to hear what others think.

Cheers,
Charles


> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Dale R. Worley
> Sent: Friday, February 22, 2013 1:06 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Another bundling proposal
>=20
> I've written an alternative bundling proposal which I think manages
> the problems of bundling better.
>=20
> A series of tutorial examples are here:
> http://tools.ietf.org/html/draft-worley-sdp-bundle-02#section-4
> so it's easy to see how the proposal works.
>=20
> Distinctive features of this proposal are:
>=20
> - A separate m=3D line describes the bundled media, allowing unambiguous
>   attribution of attributes to either the bundle or a constituent, and
>   independent rejection of each constituent.
>=20
> - Backward compatibility is achieved straightforwardly.
>=20
> - Multiplexed media flows are attributed to their original constituent
>   media descriptions without requiring each to be declared
>   individually in the SDP.
>=20
> - The multiplexed media flow is artificially given the media type
>   "audio" to prevent rejection by SBCs.
>=20
> - An encapsulating RTP format is used so demultiplexing can be done
>   without coordinating payload types between media descriptions, and
>   to ensure the bundled media description is rejected by
>   non-supporting endpoints.
>=20
> - Zero/non-zero ports and valid/null addresses are controlled to avoid
>   duplicate port usage, and to ensure SBCs see exactly the media flows
>   specified by the SDP.
>=20
>=20
>=20
> Filename:	 draft-worley-sdp-bundle
> Revision:	 02
> Title:		 Kumquat: A Generic Bundle Mechanism for the Session
> Description Protocol (SDP)
> Creation date:	 2013-02-22
> Group:		 Individual Submission
> Number of pages: 27
> URL:             http://www.ietf.org/internet-drafts/draft-worley-sdp-bun=
dle-
> 02.txt
> Status:          http://datatracker.ietf.org/doc/draft-worley-sdp-bundle
> Htmlized:        http://tools.ietf.org/html/draft-worley-sdp-bundle-02
> Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-worley-sdp-bund=
le-02
>=20
> Abstract:
>    This document defines a generic bundle mechanism for the Session
>    Description Protocol (SDP) by which the media described by a number
>    of media descriptions ("m=3D lines") are multiplexed and transmitted
>    over a single transport association.  The transport association is
>    described by an additional media description, allowing SDP attributes
>    to be applied to the aggregate, independently of attributes applied
>    to the constituents.  In offer/answer usage, the bundle mechanism is
>    backward compatible with SDP processors that do not understand the
>    mechanism.  The mechanism is designed to be compatible with the
>    limitations of the existing Internet infrastructure.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, "Dale R. Worley" <worley@ariadne.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Date: Fri, 22 Feb 2013 18:37:31 -0800
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Subject: Re: [MMUSIC] Another bundling proposal
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While I do agree that legacy implementations will have issues with using th=
e same port on multiple m=3D lines=2C the proposal does explicitly address =
multiplexing issues via the "kumquat payload format" (an RTP tunnel of sort=
s).  Describing the implications for RTP in the proposal is better rather t=
han implicitly assuming a given de-multiplexing approach or the presence of=
 RTP extensions (e.g. the capture RTP extensions Roni referred to in a prev=
ious post).=20

Since the RTP tunnelling approach does require changes to lots of component=
s in the system
(e.g. MCUs)=2C it is not necessarily the easiest thing to deploy.  However=
=2C it does appear to obviate the need to declare individual SSRCs=2C which=
 a number of the other proposals do not (e.g. if payload types are not requ=
ired to unique among all m-lines=2C then if multiple media types are multip=
lexed then SSRC is the only remaining de-multiplex point=2C so SSRC declara=
tion is required).=20

Also=2C tunnelling proposals can be used to address some of the ICE perform=
ance issues as well.  There is really no intrinsic reason why those kind of=
 proposals can't be used to allow multiple endpoints to share a TURN alloca=
tion=2C for example=2C removing TURN as a performance bottleneck without re=
quiring Trickle ICE.=20

> From: eckelcu@cisco.com
> To: worley@ariadne.com=3B mmusic@ietf.org
> Date: Fri=2C 22 Feb 2013 21:39:23 +0000
> Subject: Re: [MMUSIC] Another bundling proposal
>=20
> Hi Dale=2C
>=20
> Your proposal has some very nice properties=3B however=2C my concern with=
 the overall approach is that I fear the initial offer will not be handled =
very well by typical pre-BUMDLE implementations. Due to the port being '9' =
and the connection address being 0.0.0.0=2C I would expect either:
> 1) a re-INVITE is needed for things to be renegotiated (best case)
> 2) the call fails (worst case)
> Of course other variations between the best and worst case could happen=
=2C but the odds of arriving at the worst case seem unacceptably high to me=
. That is just my 2 cents without actually testing anything to validate it.=
 I'd like to hear what others think.
>=20
> Cheers=2C
> Charles
>=20
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Dale R. Worley
> > Sent: Friday=2C February 22=2C 2013 1:06 PM
> > To: mmusic@ietf.org
> > Subject: [MMUSIC] Another bundling proposal
> >=20
> > I've written an alternative bundling proposal which I think manages
> > the problems of bundling better.
> >=20
> > A series of tutorial examples are here:
> > http://tools.ietf.org/html/draft-worley-sdp-bundle-02#section-4
> > so it's easy to see how the proposal works.
> >=20
> > Distinctive features of this proposal are:
> >=20
> > - A separate m=3D line describes the bundled media=2C allowing unambigu=
ous
> >   attribution of attributes to either the bundle or a constituent=2C an=
d
> >   independent rejection of each constituent.
> >=20
> > - Backward compatibility is achieved straightforwardly.
> >=20
> > - Multiplexed media flows are attributed to their original constituent
> >   media descriptions without requiring each to be declared
> >   individually in the SDP.
> >=20
> > - The multiplexed media flow is artificially given the media type
> >   "audio" to prevent rejection by SBCs.
> >=20
> > - An encapsulating RTP format is used so demultiplexing can be done
> >   without coordinating payload types between media descriptions=2C and
> >   to ensure the bundled media description is rejected by
> >   non-supporting endpoints.
> >=20
> > - Zero/non-zero ports and valid/null addresses are controlled to avoid
> >   duplicate port usage=2C and to ensure SBCs see exactly the media flow=
s
> >   specified by the SDP.
> >=20
> >=20
> >=20
> > Filename:	 draft-worley-sdp-bundle
> > Revision:	 02
> > Title:		 Kumquat: A Generic Bundle Mechanism for the Session
> > Description Protocol (SDP)
> > Creation date:	 2013-02-22
> > Group:		 Individual Submission
> > Number of pages: 27
> > URL:             http://www.ietf.org/internet-drafts/draft-worley-sdp-b=
undle-
> > 02.txt
> > Status:          http://datatracker.ietf.org/doc/draft-worley-sdp-bundl=
e
> > Htmlized:        http://tools.ietf.org/html/draft-worley-sdp-bundle-02
> > Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-worley-sdp-bu=
ndle-02
> >=20
> > Abstract:
> >    This document defines a generic bundle mechanism for the Session
> >    Description Protocol (SDP) by which the media described by a number
> >    of media descriptions ("m=3D lines") are multiplexed and transmitted
> >    over a single transport association.  The transport association is
> >    described by an additional media description=2C allowing SDP attribu=
tes
> >    to be applied to the aggregate=2C independently of attributes applie=
d
> >    to the constituents.  In offer/answer usage=2C the bundle mechanism =
is
> >    backward compatible with SDP processors that do not understand the
> >    mechanism.  The mechanism is designed to be compatible with the
> >    limitations of the existing Internet infrastructure.
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
 		 	   		  =

--_0fe1ee6b-a1b5-48f9-a605-b0a902f0cb45_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
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<body class=3D'hmmessage'><div dir=3D'ltr'>While I do agree that legacy imp=
lementations will have issues with using the same port on multiple m=3D lin=
es=2C the proposal does explicitly address multiplexing issues via the "kum=
quat payload format" (an RTP tunnel of sorts).&nbsp=3B Describing the impli=
cations for RTP in the proposal is better rather than implicitly assuming a=
 given de-multiplexing approach or the presence of RTP extensions (e.g. the=
 capture RTP extensions Roni referred to in a previous post). <br><br>Since=
 the RTP tunnelling approach does require changes to lots of components in =
the system<br>(e.g. MCUs)=2C it is not necessarily the easiest thing to dep=
loy.&nbsp=3B However=2C it does appear to obviate the need to declare indiv=
idual SSRCs=2C which a number of the other proposals do not (e.g. if payloa=
d types are not required to unique among all m-lines=2C then if multiple me=
dia types are multiplexed then SSRC is the only remaining de-multiplex poin=
t=2C so SSRC declaration is required). <br><br>Also=2C tunnelling proposals=
 can be used to address some of the ICE performance issues as well.&nbsp=3B=
 There is really no intrinsic reason why those kind of proposals can't be u=
sed to allow multiple endpoints to share a TURN allocation=2C for example=
=2C removing TURN as a performance bottleneck without requiring Trickle ICE=
. <br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B From: eckelcu@c=
isco.com<br>&gt=3B To: worley@ariadne.com=3B mmusic@ietf.org<br>&gt=3B Date=
: Fri=2C 22 Feb 2013 21:39:23 +0000<br>&gt=3B Subject: Re: [MMUSIC] Another=
 bundling proposal<br>&gt=3B <br>&gt=3B Hi Dale=2C<br>&gt=3B <br>&gt=3B You=
r proposal has some very nice properties=3B however=2C my concern with the =
overall approach is that I fear the initial offer will not be handled very =
well by typical pre-BUMDLE implementations. Due to the port being '9' and t=
he connection address being 0.0.0.0=2C I would expect either:<br>&gt=3B 1) =
a re-INVITE is needed for things to be renegotiated (best case)<br>&gt=3B 2=
) the call fails (worst case)<br>&gt=3B Of course other variations between =
the best and worst case could happen=2C but the odds of arriving at the wor=
st case seem unacceptably high to me. That is just my 2 cents without actua=
lly testing anything to validate it. I'd like to hear what others think.<br=
>&gt=3B <br>&gt=3B Cheers=2C<br>&gt=3B Charles<br>&gt=3B <br>&gt=3B <br>&gt=
=3B &gt=3B -----Original Message-----<br>&gt=3B &gt=3B From: mmusic-bounces=
@ietf.org [mailto:mmusic-bounces@ietf.org] On<br>&gt=3B &gt=3B Behalf Of Da=
le R. Worley<br>&gt=3B &gt=3B Sent: Friday=2C February 22=2C 2013 1:06 PM<b=
r>&gt=3B &gt=3B To: mmusic@ietf.org<br>&gt=3B &gt=3B Subject: [MMUSIC] Anot=
her bundling proposal<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B I've written an al=
ternative bundling proposal which I think manages<br>&gt=3B &gt=3B the prob=
lems of bundling better.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B A series of tut=
orial examples are here:<br>&gt=3B &gt=3B http://tools.ietf.org/html/draft-=
worley-sdp-bundle-02#section-4<br>&gt=3B &gt=3B so it's easy to see how the=
 proposal works.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B Distinctive features of=
 this proposal are:<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B - A separate m=3D li=
ne describes the bundled media=2C allowing unambiguous<br>&gt=3B &gt=3B   a=
ttribution of attributes to either the bundle or a constituent=2C and<br>&g=
t=3B &gt=3B   independent rejection of each constituent.<br>&gt=3B &gt=3B <=
br>&gt=3B &gt=3B - Backward compatibility is achieved straightforwardly.<br=
>&gt=3B &gt=3B <br>&gt=3B &gt=3B - Multiplexed media flows are attributed t=
o their original constituent<br>&gt=3B &gt=3B   media descriptions without =
requiring each to be declared<br>&gt=3B &gt=3B   individually in the SDP.<b=
r>&gt=3B &gt=3B <br>&gt=3B &gt=3B - The multiplexed media flow is artificia=
lly given the media type<br>&gt=3B &gt=3B   "audio" to prevent rejection by=
 SBCs.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B - An encapsulating RTP format is =
used so demultiplexing can be done<br>&gt=3B &gt=3B   without coordinating =
payload types between media descriptions=2C and<br>&gt=3B &gt=3B   to ensur=
e the bundled media description is rejected by<br>&gt=3B &gt=3B   non-suppo=
rting endpoints.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B - Zero/non-zero ports a=
nd valid/null addresses are controlled to avoid<br>&gt=3B &gt=3B   duplicat=
e port usage=2C and to ensure SBCs see exactly the media flows<br>&gt=3B &g=
t=3B   specified by the SDP.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B <br>&gt=3B =
&gt=3B <br>&gt=3B &gt=3B Filename:	 draft-worley-sdp-bundle<br>&gt=3B &gt=
=3B Revision:	 02<br>&gt=3B &gt=3B Title:		 Kumquat: A Generic Bundle Mecha=
nism for the Session<br>&gt=3B &gt=3B Description Protocol (SDP)<br>&gt=3B =
&gt=3B Creation date:	 2013-02-22<br>&gt=3B &gt=3B Group:		 Individual Subm=
ission<br>&gt=3B &gt=3B Number of pages: 27<br>&gt=3B &gt=3B URL:          =
   http://www.ietf.org/internet-drafts/draft-worley-sdp-bundle-<br>&gt=3B &=
gt=3B 02.txt<br>&gt=3B &gt=3B Status:          http://datatracker.ietf.org/=
doc/draft-worley-sdp-bundle<br>&gt=3B &gt=3B Htmlized:        http://tools.=
ietf.org/html/draft-worley-sdp-bundle-02<br>&gt=3B &gt=3B Diff:            =
http://www.ietf.org/rfcdiff?url2=3Ddraft-worley-sdp-bundle-02<br>&gt=3B &gt=
=3B <br>&gt=3B &gt=3B Abstract:<br>&gt=3B &gt=3B    This document defines a=
 generic bundle mechanism for the Session<br>&gt=3B &gt=3B    Description P=
rotocol (SDP) by which the media described by a number<br>&gt=3B &gt=3B    =
of media descriptions ("m=3D lines") are multiplexed and transmitted<br>&gt=
=3B &gt=3B    over a single transport association.  The transport associati=
on is<br>&gt=3B &gt=3B    described by an additional media description=2C a=
llowing SDP attributes<br>&gt=3B &gt=3B    to be applied to the aggregate=
=2C independently of attributes applied<br>&gt=3B &gt=3B    to the constitu=
ents.  In offer/answer usage=2C the bundle mechanism is<br>&gt=3B &gt=3B   =
 backward compatible with SDP processors that do not understand the<br>&gt=
=3B &gt=3B    mechanism.  The mechanism is designed to be compatible with t=
he<br>&gt=3B &gt=3B    limitations of the existing Internet infrastructure.=
<br>&gt=3B &gt=3B _______________________________________________<br>&gt=3B=
 &gt=3B mmusic mailing list<br>&gt=3B &gt=3B mmusic@ietf.org<br>&gt=3B &gt=
=3B https://www.ietf.org/mailman/listinfo/mmusic<br>&gt=3B ________________=
_______________________________<br>&gt=3B mmusic mailing list<br>&gt=3B mmu=
sic@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo/mmusic<br></di=
v> 		 	   		  </div></body>
</html>=

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From pkyzivat@alum.mit.edu  Sat Feb 23 18:50:09 2013
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Subject: Re: [MMUSIC] Another bundling proposal
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This is an interesting approach, though Bernard's comments need to be 
considered.

One observation is that this introduces a new level of multiplexing 
based on a special codec. While it can work, there is already a proposal 
on the table for multiplexing multiple RTP sessions on a single 5-tuple: 
draft-westerlund-avtcore-transport-multiplexing-04. ISTM that this new 
bundling proposal could be adapted to use that multiplexing technique.

Bernard already raised issues about port 9 and address 0.0.0.0. I'll add 
that when this is done for multiple m-lines we stand some chance of 
getting errors due to reusing the same transport address. AFAIK there is 
no particular reason to use port 9 in this case, so perhaps it would be 
safer to use a different non-zero port in each such line.

I had originally found little reason to pursue 
draft-westerlund-avtcore-transport-multiplexing-04, but now I start to 
think that this provides an answer to the problem of demuxing without 
having to specify SSRCs or restrict payload types. I think it could be 
used, rather than an RTP header extension, to specify capture-encodings 
for clue. (But it can't specify mapping to multiple capture-encodings.)

	Thanks,
	Paul

On 2/22/13 4:05 PM, Dale R. Worley wrote:
> I've written an alternative bundling proposal which I think manages
> the problems of bundling better.
>
> A series of tutorial examples are here:
> http://tools.ietf.org/html/draft-worley-sdp-bundle-02#section-4
> so it's easy to see how the proposal works.
>
> Distinctive features of this proposal are:
>
> - A separate m= line describes the bundled media, allowing unambiguous
>    attribution of attributes to either the bundle or a constituent, and
>    independent rejection of each constituent.
>
> - Backward compatibility is achieved straightforwardly.
>
> - Multiplexed media flows are attributed to their original constituent
>    media descriptions without requiring each to be declared
>    individually in the SDP.
>
> - The multiplexed media flow is artificially given the media type
>    "audio" to prevent rejection by SBCs.
>
> - An encapsulating RTP format is used so demultiplexing can be done
>    without coordinating payload types between media descriptions, and
>    to ensure the bundled media description is rejected by
>    non-supporting endpoints.
>
> - Zero/non-zero ports and valid/null addresses are controlled to avoid
>    duplicate port usage, and to ensure SBCs see exactly the media flows
>    specified by the SDP.
>
>
>
> Filename:	 draft-worley-sdp-bundle
> Revision:	 02
> Title:		 Kumquat: A Generic Bundle Mechanism for the Session Description Protocol (SDP)
> Creation date:	 2013-02-22
> Group:		 Individual Submission
> Number of pages: 27
> URL:             http://www.ietf.org/internet-drafts/draft-worley-sdp-bundle-02.txt
> Status:          http://datatracker.ietf.org/doc/draft-worley-sdp-bundle
> Htmlized:        http://tools.ietf.org/html/draft-worley-sdp-bundle-02
> Diff:            http://www.ietf.org/rfcdiff?url2=draft-worley-sdp-bundle-02
>
> Abstract:
>     This document defines a generic bundle mechanism for the Session
>     Description Protocol (SDP) by which the media described by a number
>     of media descriptions ("m= lines") are multiplexed and transmitted
>     over a single transport association.  The transport association is
>     described by an additional media description, allowing SDP attributes
>     to be applied to the aggregate, independently of attributes applied
>     to the constituents.  In offer/answer usage, the bundle mechanism is
>     backward compatible with SDP processors that do not understand the
>     mechanism.  The mechanism is designed to be compatible with the
>     limitations of the existing Internet infrastructure.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From bernard_aboba@hotmail.com  Sat Feb 23 23:14:21 2013
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Date: Sat, 23 Feb 2013 23:14:17 -0800
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Subject: Re: [MMUSIC] Another bundling proposal (and why you can't neglect RTP)
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--_ced18e30-2242-41f0-93e7-d9284ff74c17_
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Paul said:=20

> I had originally found little reason to pursue=20
> draft-westerlund-avtcore-transport-multiplexing-04=2C but now I start to=
=20
> think that this provides an answer to the problem of demuxing without=20
> having to specify SSRCs or restrict payload types. I think it could be=20
> used=2C rather than an RTP header extension=2C to specify capture-encodin=
gs=20
> for clue. (But it can't specify mapping to multiple capture-encodings.)

[BA] Yes=2C I've also begun to wonder if we haven't been giving that (and p=
ossibly)
other transport multiplexing approaches insufficient consideration.  A/V=20
multiplexing isn't just an SDP issue=2C it is also an RTP issue.  Say we do=
n't
specify SSRCs and instead require that the payload type is unique among=20
all multiplexed m=3D lines so we can use PT to de-multiplex incoming RTP.=20
Without specifying SSRCs=2C how do we map the RTCP reports=2C which
only include SSRCs and not payload types to the appropriate media=20
stream tracks?   It seems to me that Cullen's "Plan A" has a fundamental
flaw. =20

 		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
{
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body.hmmessage
{
font-size: 12pt=3B
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<body class=3D'hmmessage'><div dir=3D'ltr'>Paul said: <br><br><div><div id=
=3D"SkyDrivePlaceholder"></div>&gt=3B I had originally found little reason =
to pursue <br>&gt=3B draft-westerlund-avtcore-transport-multiplexing-04=2C =
but now I start to <br>&gt=3B think that this provides an answer to the pro=
blem of demuxing without <br>&gt=3B having to specify SSRCs or restrict pay=
load types. I think it could be <br>&gt=3B used=2C rather than an RTP heade=
r extension=2C to specify capture-encodings <br>&gt=3B for clue. (But it ca=
n't specify mapping to multiple capture-encodings.)<br><br>[BA] Yes=2C I've=
 also begun to wonder if we haven't been giving that (and possibly)<br>othe=
r transport multiplexing approaches insufficient consideration.&nbsp=3B A/V=
 <br>multiplexing isn't just an SDP issue=2C it is also an RTP issue.&nbsp=
=3B Say we don't<br>specify SSRCs and instead require that the payload type=
 is unique among <br>all multiplexed m=3D lines so we can use PT to de-mult=
iplex incoming RTP. <br>Without specifying SSRCs=2C how do we map the RTCP =
reports=2C which<br>only include SSRCs and not payload types to the appropr=
iate media <br>stream tracks?&nbsp=3B&nbsp=3B It seems to me that Cullen's =
"Plan A" has a fundamental<br>flaw.&nbsp=3B <br><br></div> 		 	   		  </div=
></body>
</html>=

--_ced18e30-2242-41f0-93e7-d9284ff74c17_--

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org" <draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org>
Thread-Topic: SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-decodability-07
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Date: Sun, 24 Feb 2013 21:54:54 +0000
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Subject: [MMUSIC] SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-decodability-07
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I am the assigned SDP directorate reviewer for this draft. For background
on the SDP directorate, please see the FAQ at
http://www.ietf.org/iesg/directorate/sdp.html

Please wait for direction from your document shepherd or AD before posting
a new version of the draft.

Summary:
--------
- The RTCP XR extension syntax looks fine
- There is one (small) technical issue
- There are quite a few editorial issues

Technical Issues:
-----------------
1) I am puzzled why you named the new block as "non-psi-ts-decodability".
It does not follow what you have in the title. I also did not get the
"non" part.

Editorial Issues:
-----------------
1) I don't have the ISO document handy but I think the container is called
MPEG2 Transport Stream not MPEG Transport Stream. Please double check.

2) The abstract mentions the use of broadcast of transport streams in IPTV
systems. IPTV uses only unicast and multicast. Cable and satellite systems
use the broadcast of transport streams.

3) Document's short name is "RTCP XR Non PSI Decodability". Why?

4) The title has "metric" but the abstract has "metrics." Please be
consistent.

5) Section 1.1. "application-dependant" -> "application-dependent"

6) Use either "decodability" or "de-codability".

7) Section 1.4. AFAIK, there is no such thing called "MPEG4 TS content."
The content could be encoded with MPEG4 but the transport stream is still
MPEG2 Transport Stream. So, you could say "MPEG4 over MPEG2-TS over RTP."

8) The text seems to be using both "MPEG TS PSI Independent decodability
statistics metrics" and "MPEG-TS PSI Independent Decodability Metrics". I
think you need the former.

9) There are several typos (mostly due to copy/paste errors). Please fix
them. Acknowledgements section also has misspellings and typos.


From fluffy@cisco.com  Sun Feb 24 16:21:57 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: mmusic WG <mmusic@ietf.org>
Thread-Topic: draft-nandakumar-mmusic-sdp-mux-attributes-01 
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Cc: "Suhas Nandakumar \(snandaku\)" <snandaku@cisco.com>
Subject: [MMUSIC] draft-nandakumar-mmusic-sdp-mux-attributes-01
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FYI =85

Very minimal changes from the -00=20

Begin forwarded message:

> From: internet-drafts@ietf.org
> Subject: New Version Notification for draft-nandakumar-mmusic-sdp-mux-att=
ributes-01.txt
> Date: February 24, 2013 5:20:24 PM MST
> To: fluffy@iii.ca
> Cc: snandaku@cisco.com
>=20
>=20
> A new version of I-D, draft-nandakumar-mmusic-sdp-mux-attributes-01.txt
> has been successfully submitted by Cullen Jennings and posted to the
> IETF repository.
>=20
> Filename:	 draft-nandakumar-mmusic-sdp-mux-attributes
> Revision:	 01
> Title:		 A Framework for SDP Attributes when Multiplexing
> Creation date:	 2013-02-24
> Group:		 Individual Submission
> Number of pages: 52
> URL:             http://www.ietf.org/internet-drafts/draft-nandakumar-mmu=
sic-sdp-mux-attributes-01.txt
> Status:          http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-=
sdp-mux-attributes
> Htmlized:        http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-m=
ux-attributes-01
> Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-nandakumar-mmus=
ic-sdp-mux-attributes-01
>=20
> Abstract:
>   Communication sessions in RTCWeb are described and negotiated using
>   Session Description Protocol(SDP)[RFC4566] mechanisms.  SDP uses
>   attributes to describes specific aspects of such a session.  These
>   attributes have different impacts when using varions of multiplexing
>   RTP on a single transport layer flow.  The scope of this
>   specification is to provide a framework for analyzing the multiplex
>   characteristics for SDP attributes and provide multiplexing
>   characteristics of existing attributes.
>=20
>=20
>=20
>=20
> The IETF Secretariat
>=20


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From: worley@ariadne.com (Dale R. Worley)
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To: Harald Alvestrand <harald@alvestrand.no>
In-reply-to: <51263661.3060307@alvestrand.no> (harald@alvestrand.no)
References: <03FBA798AC24E3498B74F47FD082A92F36EA6762@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B0FC32F@ESESSMB209.ericsson.se>, <03FBA798AC24E3498B74F47FD082A92F36EA6AA6@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B0FEDDE@ESESSMB209.ericsson.se> <03FBA798AC24E3498B74F47FD082A92F36EA6C8F@US70UWXCHMBA04.zam.alcatel-lucent.com> <7594FB04B1934943A5C02806D1A2204B0FF4F0@ESESSMB209.ericsson.se> <03FBA798AC24E3498B74F47FD082A92F36EA6CDB@US70UWXCHMBA04.zam.alcatel-lucent.com> <51263661.3060307@alvestrand.no>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt)
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> From: Harald Alvestrand <harald@alvestrand.no>
> 
> Designing around intermediaries that pass along information that they do 
> not understand, and then expect to act independently based on the 
> message that they partially understood, seems to me to be a quite 
> limiting approach.

If we're going to succeed, we've got to be aware of the environment in
which the solution operates.  In principle, intermediate systems
should operate transparently, but they don't.

> We discussed this at the interim (I think Cullen was pretty vocal that 
> there's a limit to how much brokenness we can design around, and he 
> thought intermediaries passing BUNDLE without understanding BUNDLE were 
> clearly beyond that limit).

Maybe Cullen was vocal about that, but he also points out that C***o
NATs *have* to pay attention to the port numbers in media
descriptions, so they enforce the uniqueness of port assignments.

Because of problems like this, four of the "compatibility desiderata"
I included in my proposal document the behavior of middleboxes we have
to tolerate.
(http://tools.ietf.org/html/draft-worley-sdp-bundle-02#section-3.2)

Dale

From worley@shell01.TheWorld.com  Sun Feb 24 17:40:02 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: mmusic@ietf.org
Subject: [MMUSIC] Design feature chart for bundling proposals
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I've submitted a revision of my draft, draft-worley-sdp-bundle-03.
The proposed bundling mechanism is unchanged (although it will need to
be improved, based on the feedback I've received).

The useful part of the new version is a draft of a design feature
comparison chart for the current bundling proposals.  (6 that I've found
so far!)  The chart can be found at
http://tools.ietf.org/html/draft-worley-sdp-bundle-03#section-7
(I'd welcome any corrections or additions to the chart.)  The entire
draft is at
http://tools.ietf.org/html/draft-worley-sdp-bundle-03

Here is a copy of the chart itself (more detailed notes in the draft
itself):

7.  Design Features and Comparison

   Key:

         x = feature present in proposal
         - = feature not present in proposal
         . = feature not discussed in proposal
       N/A = feature is not relevant because of another feature choice

		   worley-sdp-bundle-03 (KUMQUAT)
		   |   ietf-mmusic-sdp-bundle-negotiation-03 (BUNDLE)
		   |   |   holmberg-mmusic-sdp-mmt-negotiation-00 (MMT)
		   |   |   |   alvestrand-one-rtp-02 (TOGETHER)
		   |   |   |   |   ejzak-mmusic-bundle-alternatives-00
		   |   |   |   |   |   Roach alternative 1a
		   |   |   |   |   |   |(roach-mmusic-mlines-00)
		   |   |   |   |   |   |   Roach alternative 1b
		   |   |   |   |   |   |   |   Roach alternative 2
		   |   |   |   |   |   |   |   |   westerlund-avtcore-
		   |   |   |   |   |   |   |   |   |transport-multiplexing-04
		   |   |   |   |   |   |   |   |   |(SHIM)
		   V   V   V   V   V   V   V   V   V
   MD grouping:
       one         -   -   -   -   -   -   x   -   -
       per type    -   -   -   -   -   x   -   -   -
       none        x   x   x   x   x   -   -   x   x

   Separate bundle MD:
       no          -   x   -   x   x   x   x   x   x
       m=anymedia  -   -   x   -   -   -   -   -   -
       m=audio     x   -   -   -   -   -   -   -   -
       m=multipart -   -   -   -   -   -   -   -   -

   Ports in constituent MDs in offer:
       N/A         -   -   -   -   -   x   x   -   -
       same        -   -   -   x   x   -   -   x   x
       different   -   x   x   -   x   -   -   -   -
       null        x   -   -   -   -   -   -   -   -

   Immediate update:
       none        -   -   x   x   x   x   x   x   x
       for support -   x   -   -   -   -   -   -   -
       for compat. x   -   -   -   -   -   -   -   -

   Constituent MD ports after establishment:
       N/A         -   -   -   -   -   x   x   -   -
       same        -   x   -   x   -   -   -   x   x
       different   -   -   -   -   -   -   -   -   -
       null        -   -   -   -   -   -   -   -   -
       rejected    x   -   x   -   x   -   -   -   -

   Bundle MD payload types:
       N/A         -   -   -   -   -   x   x   -   -
       one MD      -   x   -   x   .   -   -   x   x
       all MDs     -   -   x   -   .   -   -   -   -
       encap.      x   -   -   -   .   -   -   -   -

   Constituent MD payload types:
       N/A         -   -   -   -   -   x   x   -   -
       overlapping x   -   -   -   .   -   -   x   x
       distinct    -   x   x   x   .   -   -   -   -

   Demultiplexing based on:
       N/A         -   -   -   -   -   x   x   -   -
       PT          -   x   x   x   .   -   -   -   -
       encap.      x   -   -   -   .   -   -   x   x

   Rejection of bundle MD based on:
       N/A         -   x   -   x   x   x   x   x   x
       media type  -   -   x   -   -   -   -   -   -
       codec       x   -   -   -   -   -   -   -   -

Dale

From rachel.huang@huawei.com  Mon Feb 25 00:28:30 2013
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, "draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org" <draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org>
Thread-Topic: [MMUSIC] SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-decodability-07
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Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-decodability-07
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Dear Ali,

Thank you for the review. Please see inline.

Best Regards!
Rachel


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Ali C. Begen (abegen)
Sent: Monday, February 25, 2013 5:55 AM
To: draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org
Cc: mmusic@ietf.org; gonzalo.camarillo@ericsson.com
Subject: [MMUSIC] SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-dec=
odability-07

I am the assigned SDP directorate reviewer for this draft. For background
on the SDP directorate, please see the FAQ at
http://www.ietf.org/iesg/directorate/sdp.html

Please wait for direction from your document shepherd or AD before posting
a new version of the draft.

Summary:
--------
- The RTCP XR extension syntax looks fine
- There is one (small) technical issue
- There are quite a few editorial issues

Technical Issues:
-----------------
1) I am puzzled why you named the new block as "non-psi-ts-decodability".
It does not follow what you have in the title. I also did not get the
"non" part.

[Rachel]: To get consistent with the title we used, how about we propose to=
 change the block name in SDP as "ts-psi-indep-decodability"?

Editorial Issues:
-----------------
1) I don't have the ISO document handy but I think the container is called
MPEG2 Transport Stream not MPEG Transport Stream. Please double check.

[Rachel]: Okay. Thanks.

2) The abstract mentions the use of broadcast of transport streams in IPTV
systems. IPTV uses only unicast and multicast. Cable and satellite systems
use the broadcast of transport streams.

[Rachel]: Okay. Will fix this.

3) Document's short name is "RTCP XR Non PSI Decodability". Why?

[Rachel]: Will be changed to "RTCP XR PSI Independent Decodability".

4) The title has "metric" but the abstract has "metrics." Please be
consistent.

[Rachel]:Okay.

5) Section 1.1. "application-dependant" -> "application-dependent"

[Rachel]:Okay.

6) Use either "decodability" or "de-codability".

[Rachel]:Okay.

7) Section 1.4. AFAIK, there is no such thing called "MPEG4 TS content."
The content could be encoded with MPEG4 but the transport stream is still
MPEG2 Transport Stream. So, you could say "MPEG4 over MPEG2-TS over RTP."

[Rachel]:Right.

8) The text seems to be using both "MPEG TS PSI Independent decodability
statistics metrics" and "MPEG-TS PSI Independent Decodability Metrics". I
think you need the former.

[Rachel]:Okay.

9) There are several typos (mostly due to copy/paste errors). Please fix
them. Acknowledgements section also has misspellings and typos.

[Rachel]: Will do.

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

From christer.holmberg@ericsson.com  Mon Feb 25 01:10:00 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "Harald Alvestrand" <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-00.txt)
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Subject: Re: [MMUSIC] Requiring support for nonconformant intermediate systems (Re: FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-00.txt)
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Hi Richard,

>> Your solution expects the intermediary to understand that, eventhough=20
>> the addresses for all-but-one m- lines are unspecified, media=20
>> associated with those m- lines will still be sent - using the port of=20
>> another m- line.
>
> But this requirement is only placed on intermediaries that have already i=
ndicated support for BUNDLE by forwarding=20
> the related attributes.  The unspecified address only needs to be include=
d in the SDP answer, and not the offer, so it=20
> is returned within an established dialog, and there is no danger of sendi=
ng it to an unknown intermediary.
>=20
>> I still strongly think that sending media associated with m- lines,=20
>> for which the address is unspecified, is a design error.
>
> I'm not sure how the term "design error" applies to a solution that behav=
es correctly and consistently under the given=20
> assumptions (there are other aspects to be fixed but bear with me for sak=
e of argument).  It is true that I propose to=20
> clarify the meaning of unspecified address in the presence of BUNDLE, whi=
ch is a new and as yet non-standardized=20
> concept.  There are many examples of such "clarifications" as we expand t=
he capabilities that can be represented=20
> in SDP.  The key is backward compatibility.  If the concern is about inte=
rmediaries that forward BUNDLE attributes=20
> without understanding them, then there is little that can be done anyway.

There are intermediaries that do not modify SDP - they simply look at the a=
ddress information in order to open gates etc.

But, my comment is not related to intermediaries: I think it's an general S=
DP design error.

Regards,

Christer

From abegen@cisco.com  Mon Feb 25 01:49:02 2013
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Huangyihong (Rachel)" <rachel.huang@huawei.com>, "draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org" <draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org>
Thread-Topic: [MMUSIC] SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-decodability-07
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Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-xrblock-rtcp-xr-decodability-07
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-----Original Message-----
From: "Huangyihong   (Rachel)" <rachel.huang@huawei.com>
Date: Monday, February 25, 2013 3:28 AM
To: "Ali C. Begen" <abegen@cisco.com>,
"draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org"
<draft-ietf-xrblock-rtcp-xr-decodability@tools.ietf.org>
Cc: "mmusic@ietf.org" <mmusic@ietf.org>, "gonzalo.camarillo@ericsson.com"
<gonzalo.camarillo@ericsson.com>
Subject: RE: [MMUSIC] SDP Directorate: Review of
draft-ietf-xrblock-rtcp-xr-decodability-07

>Technical Issues:
>-----------------
>1) I am puzzled why you named the new block as "non-psi-ts-decodability".
>It does not follow what you have in the title. I also did not get the
>"non" part.
>
>[Rachel]: To get consistent with the title we used, how about we propose
>to change the block name in SDP as "ts-psi-indep-decodability"?

I see that you already submitted a revision. Your current title says:

RTP Control Protocol (RTCP) Extended Report (XR) Block for MPEG2 Program
      Specific Information (PSI) Independent Transport Stream (TS)
                Decodability Statistics Metric reporting


This still does not make sense to me. "Transport Stream" should be before
"Program Specific Information". My other comment on the title was it
should be "=8A Metrics Reporting" not "Metric reporting".

-acbegen


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From christer.holmberg@ericsson.com  Mon Feb 25 05:09:27 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Another bundling proposal - Christer's comments
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Subject: Re: [MMUSIC] Another bundling proposal - Christer's comments
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Hi,

I had previously provided comments to Dale off-line. Some has been addresse=
d in the -03 version of the draft. But, I still have some questions.

First, unless I've missed something, the mechanism is still for RTP ONLY. I=
t is not possible to e.g. put the DataChannel in the bundle. I think we asa=
p need to decide whether that is a requirement or not.

Second, you use the non-bundle m- lines for codec negotiation etc. But, pre=
viously, when MMT was discussed, I believe that people indicated that they =
do not want to "refer" to other m- lines from the MMT m- line. Instead,  th=
e information should be explicitly provided in the MMT m- line (which is th=
e reason why the SDP size gets pretty big with MMT, as most information is =
duplicated).

Regards,

Christer





-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Dale R. Worley
Sent: 22. helmikuuta 2013 23:06
To: mmusic@ietf.org
Subject: [MMUSIC] Another bundling proposal

I've written an alternative bundling proposal which I think manages the pro=
blems of bundling better.

A series of tutorial examples are here:
http://tools.ietf.org/html/draft-worley-sdp-bundle-02#section-4
so it's easy to see how the proposal works.

Distinctive features of this proposal are:

- A separate m=3D line describes the bundled media, allowing unambiguous
  attribution of attributes to either the bundle or a constituent, and
  independent rejection of each constituent.

- Backward compatibility is achieved straightforwardly.

- Multiplexed media flows are attributed to their original constituent
  media descriptions without requiring each to be declared
  individually in the SDP.

- The multiplexed media flow is artificially given the media type
  "audio" to prevent rejection by SBCs.

- An encapsulating RTP format is used so demultiplexing can be done
  without coordinating payload types between media descriptions, and
  to ensure the bundled media description is rejected by
  non-supporting endpoints.

- Zero/non-zero ports and valid/null addresses are controlled to avoid
  duplicate port usage, and to ensure SBCs see exactly the media flows
  specified by the SDP.



Filename:	 draft-worley-sdp-bundle
Revision:	 02
Title:		 Kumquat: A Generic Bundle Mechanism for the Session Description Pr=
otocol (SDP)
Creation date:	 2013-02-22
Group:		 Individual Submission
Number of pages: 27
URL:             http://www.ietf.org/internet-drafts/draft-worley-sdp-bundl=
e-02.txt
Status:          http://datatracker.ietf.org/doc/draft-worley-sdp-bundle
Htmlized:        http://tools.ietf.org/html/draft-worley-sdp-bundle-02
Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-worley-sdp-bundle=
-02

Abstract:
   This document defines a generic bundle mechanism for the Session
   Description Protocol (SDP) by which the media described by a number
   of media descriptions ("m=3D lines") are multiplexed and transmitted
   over a single transport association.  The transport association is
   described by an additional media description, allowing SDP attributes
   to be applied to the aggregate, independently of attributes applied
   to the constituents.  In offer/answer usage, the bundle mechanism is
   backward compatible with SDP processors that do not understand the
   mechanism.  The mechanism is designed to be compatible with the
   limitations of the existing Internet infrastructure.
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: How does an answerer reject the first m line in a BUNDLE?
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Subject: [MMUSIC] How does an answerer reject the first m line in a BUNDLE?
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Hi Christer,
In rereading your latest BUNDLE draft I was confused about what the answere=
r does when it needs to reject the first m line in a bundle group.  The fol=
lowing text in your draft implies to me that the answerer is not allowed to=
 reject the first m line:


"Until the SDP Answerer receives the new SDP Offer, which contains an

   identical port number value for each "m=3D" line associated with a

   "BUNDLE" group, it MUST use the port, and related information (ICE

   candidates, SDES keys etc) associated with the first "m=3D" line in the

   "BUNDLE" group."

What does this text mean if the answerer rejects the first line?  Doesn't t=
his mean that the answerer must wait for the "new" SDP offer with identical=
 ports for each group member before it knows the 5-tuple to use for the bun=
dle, thus delaying call setup?  What does it mean to use SDES keys for a re=
jected m line (since these keys should only apply to this m line).

It may not be common to reject the first line but I think we have to allow =
for it.

BR, Richard

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<p class=3D"MsoNormal">Hi Christer,<o:p></o:p></p>
<p class=3D"MsoNormal">In rereading your latest BUNDLE draft I was confused=
 about what the answerer does when it needs to reject the first m line in a=
 bundle group.&nbsp; The following text in your draft implies to me that th=
e answerer is not allowed to reject the
 first m line:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&#8220;Until the SDP Answerer receives the new SDP Offer, which contains=
 an<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&nbsp;&nbsp; identical port number value for each &quot;m=3D&quot; line =
associated with a<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&nbsp;&nbsp; &quot;BUNDLE&quot; group, it MUST use the port, and related=
 information (ICE<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&nbsp;&nbsp; candidates, SDES keys etc) associated with the first &quot;=
m=3D&quot; line in the<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&nbsp;&nbsp; &quot;BUNDLE&quot; group.&#8221;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">What does this text mean if the answerer rejects the=
 first line?&nbsp; Doesn&#8217;t this mean that the answerer must wait for =
the &#8220;new&#8221; SDP offer with identical ports for each group member =
before it knows the 5-tuple to use for the bundle, thus delaying
 call setup?&nbsp; What does it mean to use SDES keys for a rejected m line=
 (since these keys should only apply to this m line).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">It may not be common to reject the first line but I =
think we have to allow for it.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">BR, Richard<o:p></o:p></p>
</div>
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From ari.keranen@ericsson.com  Mon Feb 25 10:28:14 2013
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Subject: [MMUSIC] Fwd: New Version Notification for draft-keranen-mmusic-rfc5245bis-01.txt
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Hi all,

Here's the -01 version of the ICE-bis draft. This version has the SDP 
and SIP specific parts split to a separate document, as discussed at the 
Atlanta meeting.

Work still remains to be done (E.g., should we use some other term 
instead of offer/answer? Should also all RTP specific text be 
(re)moved?), but in general, for the authors, this seems like the right 
approach.

Give it a quick read (or at least check out the diff against -00), and 
let us know if the approach seems sensible to you too.


Cheers,
Ari
(as individual)


-------- Original Message --------
A new version of I-D, draft-keranen-mmusic-rfc5245bis-01.txt
has been successfully submitted by Ari Keranen and posted to the
IETF repository.

Filename:	 draft-keranen-mmusic-rfc5245bis
Revision:	 01
Title:		 Interactive Connectivity Establishment (ICE): A Protocol for 
Network Address Translator (NAT) Traversal for Offer/Answer Protocols
Creation date:	 2013-02-25
Group:		 Individual Submission
Number of pages: 89
URL: 
http://www.ietf.org/internet-drafts/draft-keranen-mmusic-rfc5245bis-01.txt
Status: 
http://datatracker.ietf.org/doc/draft-keranen-mmusic-rfc5245bis
Htmlized: 
http://tools.ietf.org/html/draft-keranen-mmusic-rfc5245bis-01
Diff: 
http://www.ietf.org/rfcdiff?url2=draft-keranen-mmusic-rfc5245bis-01

Abstract:
    This document describes a protocol for Network Address Translator
    (NAT) traversal for UDP-based multimedia sessions established with
    the offer/answer model.  This protocol is called Interactive
    Connectivity Establishment (ICE).  ICE makes use of the Session
    Traversal Utilities for NAT (STUN) protocol and its extension,
    Traversal Using Relay NAT (TURN).  ICE can be used by any protocol
    utilizing the offer/answer model, such as the Session Initiation
    Protocol (SIP).

    This document obsoletes RFC 5245.


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Subject: [MMUSIC] Fwd: New Version Notification for draft-petithuguenin-mmusic-ice-sip-sdp-01.txt
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Hi all,

And here's the draft on how to use ICE-bis with SIP & SDP.

It too is still very much work in progress (it's currently mostly 
cut-n-paste from ICE) but from this you should get the idea on how the 
split would roughly look like.


Cheers,
Ari
(as individual)


-------- Original Message --------
A new version of I-D, draft-petithuguenin-mmusic-ice-sip-sdp-01.txt
has been successfully submitted by Marc Petit-Huguenin and posted to the
IETF repository.

Filename:	 draft-petithuguenin-mmusic-ice-sip-sdp
Revision:	 01
Title:		 Using Interactive Connectivity Establishment (ICE) with Session 
Description Protocol (SDP) offer/answer and Session Initiation Protocol 
(SIP)
Creation date:	 2013-02-25
Group:		 Individual Submission
Number of pages: 40
URL: 
http://www.ietf.org/internet-drafts/draft-petithuguenin-mmusic-ice-sip-sdp-01.txt
Status: 
http://datatracker.ietf.org/doc/draft-petithuguenin-mmusic-ice-sip-sdp
Htmlized: 
http://tools.ietf.org/html/draft-petithuguenin-mmusic-ice-sip-sdp-01
Diff: 
http://www.ietf.org/rfcdiff?url2=draft-petithuguenin-mmusic-ice-sip-sdp-01

Abstract:
    This document describes how Interactive Connectivity Establishment
    (ICE) is used with Session Description Protocol (SDP) offer/answer
    and Session Initiation Protocol (SIP).

From internet-drafts@ietf.org  Mon Feb 25 12:00:47 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc4566bis-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : SDP: Session Description Protocol
	Author(s)       : Mark Handley
                          Van Jacobson
                          Colin Perkins
                          Ali Begen
	Filename        : draft-ietf-mmusic-rfc4566bis-07.txt
	Pages           : 49
	Date            : 2013-02-25

Abstract:
   This memo defines the Session Description Protocol (SDP).  SDP is
   intended for describing multimedia sessions for the purposes of
   session announcement, session invitation, and other forms of
   multimedia session initiation.  This document obsoletes RFC 4566.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc4566bis

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rfc4566bis-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rfc4566bis-07


Internet-Drafts are also available by anonymous FTP at:
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To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "mmusic@ietf.org" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] How does an answerer reject the first m line in a BUNDLE?
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Hi,


>In rereading your latest BUNDLE draft I was confused about what the answer=
er does when it needs to reject the first m line in a bundle group.  The fo=
llowing text in your draft implies to me that the answerer is not allowed t=
o reject the first m line:
>

>=93Until the SDP Answerer receives the new SDP Offer, which contains an

>   identical port number value for each "m=3D" line associated with a

>   "BUNDLE" group, it MUST use the port, and related information (ICE

>   candidates, SDES keys etc) associated with the first "m=3D" line in the

>   "BUNDLE" group.=94
>
>What does this text mean if the answerer rejects the first line?  Doesn=92=
t this mean that the answerer must wait for the =93new=94 SDP offer with id=
entical ports for each group member before it knows the
>5-tuple to use for the bundle, thus delaying call setup?  What does it mea=
n to use SDES keys for a rejected m line (since these keys should only appl=
y to this m line).

If the answerer rejects the first m- line, I guess it has to wait for the s=
econd offer.

Note that Charles suggested that the answerer shall always wait for the sec=
ond offer anyway, so that's something that needs to be discussed.

>It may not be common to reject the first line but I think we have to allow=
 for it.

Absolutely.

Regards,

Christer


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10pt;">
<p>Hi,</p>
<p>&nbsp;</p>
<div style=3D"FONT-FAMILY: Times New Roman; COLOR: #000000; FONT-SIZE: 16px=
" DanaPrelude=3D"1">
<div>
<div class=3D"WordSection1">
<p class=3D"MsoNormal">&gt;In rereading your latest BUNDLE draft I was conf=
used about what the answerer does when it needs to reject the first m line =
in a bundle group.&nbsp; The following text in your draft implies to me tha=
t the answerer is not allowed to reject the
 first m line:</p>
<p class=3D"MsoNormal">&gt;</p>
<p class=3D"MsoPlainText"><span style=3D"FONT-FAMILY: 'Courier New'">&gt;=
=93Until the SDP Answerer receives the new SDP Offer, which contains an</sp=
an></p>
<p class=3D"MsoPlainText"><span style=3D"FONT-FAMILY: 'Courier New'">&gt;&n=
bsp;&nbsp; identical port number value for each &quot;m=3D&quot; line assoc=
iated with a</span></p>
<p class=3D"MsoPlainText"><span style=3D"FONT-FAMILY: 'Courier New'">&gt;&n=
bsp;&nbsp; &quot;BUNDLE&quot; group, it MUST use the port, and related info=
rmation (ICE</span></p>
<p class=3D"MsoPlainText"><span style=3D"FONT-FAMILY: 'Courier New'">&gt;&n=
bsp;&nbsp; candidates, SDES keys etc) associated with the first &quot;m=3D&=
quot; line in the</span></p>
<p class=3D"MsoPlainText"><span style=3D"FONT-FAMILY: 'Courier New'">&gt;&n=
bsp;&nbsp; &quot;BUNDLE&quot; group.=94</span></p>
<p class=3D"MsoNormal">&gt;</p>
<p class=3D"MsoNormal">&gt;What does this text mean if the answerer rejects=
 the first line?&nbsp; Doesn=92t this mean that the answerer must wait for =
the =93new=94 SDP offer with identical ports for each group member before i=
t knows the
</p>
<p class=3D"MsoNormal">&gt;5-tuple to use for the bundle, thus delaying cal=
l setup?&nbsp; What does it mean to use SDES keys for a rejected m line (si=
nce these keys should only apply to this m line).</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">If the answerer rejects the first m- line, I guess i=
t has to wait for the second offer.</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">Note that Charles suggested that the answerer shall =
always wait for the second offer anyway, so that's something that needs to =
be discussed.</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">&gt;It may not be common to reject the first line bu=
t I think we have to allow for it.</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">Absolutely.</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">Regards,</p>
<p class=3D"MsoNormal">&nbsp;</p>
<p class=3D"MsoNormal">Christer</p>
<p class=3D"MsoNormal">&nbsp;</p>
</div>
</div>
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ed"){ dstb();}</script></div>
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</body>
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From internet-drafts@ietf.org  Mon Feb 25 13:20:21 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc2326bis-31.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Real Time Streaming Protocol 2.0 (RTSP)
	Author(s)       : Henning Schulzrinne
                          Anup Rao
                          Rob Lanphier
                          Magnus Westerlund
                          Martin Stiemerling
	Filename        : draft-ietf-mmusic-rfc2326bis-31.txt
	Pages           : 301
	Date            : 2013-02-25

Abstract:
   This memorandum defines RTSP version 2.0 which obsoletes RTSP version
   1.0 defined in RFC 2326.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for setup and control of the delivery of data with real-time
   properties.  RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video.  Sources of data can include both live data feeds and stored
   clips.  This protocol is intended to control multiple data delivery
   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc2326bis

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-31

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rfc2326bis-31


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


From worley@shell01.TheWorld.com  Mon Feb 25 14:45:04 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [MMUSIC] Summary of demultiplexing techniques
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I've submitted draft-worley-sdp-bundle-04.  That doesn't address any
critiques either, but it has a section that summarizes demultiplexing
techniques.  Essentially, RFC5764 section 5.1.2, at greater length and
including SCTP.  That discussion is at:
http://tools.ietf.org/html/draft-worley-sdp-bundle-04#section-8
Though it may all be old hat to y'all, I had to work through it to get
up to speed, so I figured I'd document it in the process.

Dale

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Cc: mmusic@ietf.org, christer.holmberg@ericsson.com
Subject: Re: [MMUSIC] How does an answerer reject the first m line in a BUNDLE?
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> From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
> 
> In rereading your latest BUNDLE draft I was confused about what the
> answerer does when it needs to reject the first m line in a bundle
> group.  The following text in your draft implies to me that the
> answerer is not allowed to reject the first m line:

My opinion is that handling the "reject the first m= line" problem is
sufficiently messy that it is one of the major reasons why we should
have a separate bundle media description.  A separate bundle media
description allows the constituent media descriptions to be accepted
or rejected independently without complicated logic to relocate the
bundle transport flow information between media descriptions.

Dale

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Subject: Re: [MMUSIC] Another bundling proposal - Christer's comments
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> From: Christer Holmberg <christer.holmberg@ericsson.com>
> 
> I had previously provided comments to Dale off-line. Some has been
> addressed in the -03 version of the draft. But, I still have some
> questions.

I'm now in the position to deal with technical improvements.

> First, unless I've missed something, the mechanism is still for RTP
> ONLY. It is not possible to e.g. put the DataChannel in the
> bundle. I think we asap need to decide whether that is a requirement
> or not.

That's true.  Although none of the other 5 proposals I've seen discuss
SCTP/DataChannel multiplexing either.  Fortunately, adding support for
SCTP isn't difficult, either with encapsulation or without.

> Second, you use the non-bundle m- lines for codec negotiation
> etc. But, previously, when MMT was discussed, I believe that people
> indicated that they do not want to "refer" to other m- lines from
> the MMT m- line. Instead,  the information should be explicitly
> provided in the MMT m- line (which is the reason why the SDP size
> gets pretty big with MMT, as most information is duplicated).

I've heard that mentioned before, but I've never seen the argument in
favor of it.  Can someone tell me what it is, or provide a pointer?

Dale

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Subject: Re: [MMUSIC] Another bundling proposal
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> From: Paul Kyzivat <pkyzivat@alum.mit.edu>

> One observation is that this introduces a new level of multiplexing 
> based on a special codec. While it can work, there is already a proposal 
> on the table for multiplexing multiple RTP sessions on a single 5-tuple: 
> draft-westerlund-avtcore-transport-multiplexing-04. ISTM that this new 
> bundling proposal could be adapted to use that multiplexing technique.

I haven't had the time to compare the SHIM encapsulation with the
KUMQUAT encapsulation, other than to see that SHIM encapsulation
doesn't produce valid-looking RTP, whereas KUMQUAT does.  Also,
KUMQUAT maintains word(4-octet)-alignment in the RTP payload.

But as the demultiplexing analysis shows, you don't *have* to
encapsulate to allow demultiplexing to work, as long as the payload
types of the constituent MDs are distinct, as well as the SCTP port
numbers in the SCTP constituent MDs.

> Bernard already raised issues about port 9 and address 0.0.0.0. I'll add 
> that when this is done for multiple m-lines we stand some chance of 
> getting errors due to reusing the same transport address. AFAIK there is 
> no particular reason to use port 9 in this case, so perhaps it would be 
> safer to use a different non-zero port in each such line.

Using null addresses in offers could be a problem.  But there's a fix
for that.

Dale

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Subject: Re: [MMUSIC] Another bundling proposal (and why you can't neglect RTP)
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> From: Bernard Aboba <bernard_aboba@hotmail.com>
> 
> Yes, I've also begun to wonder if we haven't been giving that (and possibly)
> other transport multiplexing approaches insufficient consideration.  A/V 
> multiplexing isn't just an SDP issue, it is also an RTP issue.  Say we don't
> specify SSRCs and instead require that the payload type is unique among 
> all multiplexed m= lines so we can use PT to de-multiplex incoming RTP. 
> Without specifying SSRCs, how do we map the RTCP reports, which
> only include SSRCs and not payload types to the appropriate media 
> stream tracks?   It seems to me that Cullen's "Plan A" has a fundamental
> flaw.  

My understanding is that RTCP is expected to be demultiplexed by
examining the SSRC in each RTCP sub-packet (called an "RTCP packet" in
RFC 3550) and then finding the RTP media stream with the matching
SSRC.  The RTCP gets sent wherever the matching RTP gets sent.  (This
is described in the demultiplexing essay I've sent to the list.)

Dale

From richard.ejzak@alcatel-lucent.com  Mon Feb 25 15:36:34 2013
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From christer.holmberg@ericsson.com  Tue Feb 26 00:04:00 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [MMUSIC] Another bundling proposal - Christer's comments
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Subject: Re: [MMUSIC] Another bundling proposal - Christer's comments
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Hi,

>> I had previously provided comments to Dale off-line. Some has been=20
>> addressed in the -03 version of the draft. But, I still have some=20
>> questions.
>
>I'm now in the position to deal with technical improvements.
>
>> First, unless I've missed something, the mechanism is still for RTP=20
>> ONLY. It is not possible to e.g. put the DataChannel in the bundle. I=20
>> think we asap need to decide whether that is a requirement or not.
>
> That's true.  Although none of the other 5 proposals I've seen discuss SC=
TP/DataChannel multiplexing either.  Fortunately, adding support for SCTP i=
sn't difficult, either with encapsulation or without.

You would encapsulate SCTP inside the new payload type?

>> Second, you use the non-bundle m- lines for codec negotiation etc.=20
>> But, previously, when MMT was discussed, I believe that people=20
>> indicated that they do not want to "refer" to other m- lines from the=20
>> MMT m- line. Instead,  the information should be explicitly provided=20
>> in the MMT m- line (which is the reason why the SDP size gets pretty=20
>> big with MMT, as most information is duplicated).
>
> I've heard that mentioned before, but I've never seen the argument in fav=
or of it.  Can someone tell me what it is, or provide a pointer?

It was indicated in a meeting (at the moment I can't remember which meeting=
, though), but I never checked whether it was added to the minutes. I guess=
 it will be on audio recording.

But, as far as I remember, there were never much discussion about it. It wa=
s an open issue on one of my slides, and people simply indicated support fo=
r adding explicit information, rather the referencing.

Regards,

Christer


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From: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] Milestone and WG item for Incremental provisioning of candidates for ICE (aka Trickle ICE)
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MMUSIC,

Incremental provisioning of candidates for ICE (also known as Trickle 
ICE) has been actively discussed on the list and presented at the 
previous meeting and Boston interim. Based on the discussions, there 
appears to be need and interest to work on this topic.

Currently we have no milestone or WG document for the work so the chairs 
would like to ask the working group's opinion on this.

Would you support having an MMUSIC milestone for this work?

At the moment there is one document 
(http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00) that would 
be suitable as the WG item for this milestone.

If you support having the new milestone, would you support adopting 
draft-ivov-mmusic-trickle-ice-00 as the WG document for the milestone?


BRs,
Ari & Flemming (MMUSIC co-chairs)

From enrico.marocco@telecomitalia.it  Tue Feb 26 06:52:49 2013
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Subject: Re: [MMUSIC] Milestone and WG item for Incremental provisioning of candidates for ICE (aka Trickle ICE)
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--------------ms090206000708030707090103
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I do believe this is something that really needs to be documented, as at
this point interworking with existing trickling-ICE implementations
degenerates very quickly in blind guessing.

I also support adopting the one individual draft on the table, on which
the authors have already put quite a lot of effort.

Enrico

On 2/26/13 1:46 PM, Ari Ker=E4nen wrote:
> MMUSIC,
>=20
> Incremental provisioning of candidates for ICE (also known as Trickle=20
> ICE) has been actively discussed on the list and presented at the=20
> previous meeting and Boston interim. Based on the discussions, there=20
> appears to be need and interest to work on this topic.
>=20
> Currently we have no milestone or WG document for the work so the chair=
s=20
> would like to ask the working group's opinion on this.
>=20
> Would you support having an MMUSIC milestone for this work?
>=20
> At the moment there is one document=20
> (http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00) that woul=
d=20
> be suitable as the WG item for this milestone.
>=20
> If you support having the new milestone, would you support adopting=20
> draft-ivov-mmusic-trickle-ice-00 as the WG document for the milestone?
>=20
>=20
> BRs,
> Ari & Flemming (MMUSIC co-chairs)
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20



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--------------ms090206000708030707090103--

From christer.holmberg@ericsson.com  Tue Feb 26 10:39:05 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] FW: New Version Notification	for draft-ejzak-mmusic-bundle-alternatives-01.txt
Thread-Index: AQHOE6/5apv3pLaYfUOB37EkyfYHjZiLOY7wgADTArA=
Date: Tue, 26 Feb 2013 18:39:02 +0000
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Subject: Re: [MMUSIC] FW: New Version Notification	for	draft-ejzak-mmusic-bundle-alternatives-01.txt
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Hi Richard,

A few comments on the -01 version.

----------

Section 1:

Q1: You say that, if the first line of the offer is rejected, and one has t=
o wait for the second offer, it may cause a setup time delay. There may be =
cases where the application doesn't want to session to succeed in the first=
 place in that case, so we can have text which recommends adding an "essent=
ial" (if available) m- line first. Typically I guess it would be an audio m=
- line.

----------

Section 2:

Q2: You indicate, that if an empty re-INVITE is sent to the WebRTC client, =
it will create a BUNDLE offer. We can simply say that, unless there is an B=
UNDLE indicator (yes, we may have to define a feature tag, option tag, etc)=
, the WebRTC should not create a BUNDLE offer - it should create a legacy o=
ffer, with different ports.

----------

Section 3.1:

Q3: The text says that, if the answerer does not use BUNDLE, the offerer wi=
ll have to send a second offer. But, you don't mention that this would also=
 add delay to the session establishment, as the answerer can't do anything =
with the m- lines with unspecified address.

----------

Section 3.1:

Q4: The text says:

"The middle box will not allocate resources for these m lines
 when forwarding the SDP offer, and the SDP answerer will usually
 respond with the unspecified address for these m lines.  Even if the
 SDP answer includes valid connection information for these m lines,
 the middle box will still not allocate separate transport flows for
 them."

As I've said before, you still DO expect the intermediary to take those m- =
lines into consideration when calculating bandwidth, checking media types, =
payload type information, crypto information etc etc etc.

----------

Section 3.1:

Q5: The text says:

"If the SDP answerer chooses to reject the first m line(s) in the
 bundle group in the SDP offer (by setting port in the m line to
 zero), it places the intended port, connection, candidate and DTLS
 crypto information for the bundle in the first valid m line of the
 SDP answer and includes the unspecified address for all other m lines
 in the bundle group.  Because of this, it is understood by both
 endpoints that the 5-tuple connection, port and DTLS crypto
 information is to be based on the first valid m line in the SDP offer
 and the first valid m line in the SDP answer."

So, assume that in the offer the first m- line contains a valid address, bu=
t in the answer the second m- line contains a valid address (because the an=
swerer rejected the first m- line).

Now, in this case, based on the text in previous section, an intermediary w=
ill not reserve resources for any of the m- lines:

- It will not reserve resources based on the first m- line, as it is reject=
ed
- It will not reserve resources based on the second m- line, as the offer c=
ontained an unspecified address

Or, do you expect media to be sent in one direction using one m- line, and =
in another direction using another m- line?

----------

Regards,

Christer

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Subject: Re: [MMUSIC] Milestone and WG item for Incremental provisioning of candidates for ICE (aka Trickle ICE)
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> From: Ari KerC$nen <ari.keranen@ericsson.com>

> At the moment there is one document 
> (http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00) that would 
> be suitable as the WG item for this milestone.
> 
> If you support having the new milestone, would you support adopting 
> draft-ivov-mmusic-trickle-ice-00 as the WG document for the milestone?

I support defining "trickle ICE", and I support adopting
draft-ivov-mmusic-trickle-ice as a basis for defining it.

Dale

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From: worley@ariadne.com (Dale R. Worley)
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Another bundling proposal - Christer's comments
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> From: Christer Holmberg <christer.holmberg@ericsson.com>
> 
> >> Second, you use the non-bundle m- lines for codec negotiation etc. 
> >> But, previously, when MMT was discussed, I believe that people 
> >> indicated that they do not want to "refer" to other m- lines from the 
> >> MMT m- line. Instead,  the information should be explicitly provided 
> >> in the MMT m- line (which is the reason why the SDP size gets pretty 
> >> big with MMT, as most information is duplicated).
> >
> > I've heard that mentioned before, but I've never seen the argument
> > in favor of it.  Can someone tell me what it is, or provide a
> > pointer?
> 
> It was indicated in a meeting (at the moment I can't remember which
> meeting, though), but I never checked whether it was added to the
> minutes. I guess it will be on audio recording.
> 
> But, as far as I remember, there were never much discussion about
> it. It was an open issue on one of my slides, and people simply
> indicated support for adding explicit information, rather the
> referencing.

If this desire exists, I'd like to hear technical reasons why it is
desired.  Adding payload types to the bundle media description makes
it more difficult to ensure that a non-supporting answerer rejects the
bundle media description.

Dale

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From: worley@ariadne.com (Dale R. Worley)
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To: mmusic@ietf.org
Subject: [MMUSIC] Encapsulation and demultiplexing
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It turns out the demultiplexing STUN, RTP/RTCP/SRTP/SRTCP, SCTP, and
DTLS (presumably encapsulating SCTP) is straightforward, as long as we
ensure that SCTP (outside of DTLS) uses a restricted range of port
numbers.

There are two versions of encapsulation now on the table:


The KUMQUAT encapsulation (draft-worley-sdp-bundle) adds four octets
to the beginning of an RTP packet to make it into a new RTP packet.
The new packet has a chosen PT to identify the encapsulation, and uses
the 2-octet "sequence" field of the first word of the RTP header to
carry the indexes of the constituent media description and of the
transport flow (in case the MD has multiple transport flows).

It's not clear that there is a need to have an encapsulation carefully
reproduce the structure of an RTP packet.  Indeed, the encapsulation
doesn't need to be very careful to make packets that look like RTP
packets:  The fixed header of an RTP packet is only three 4-octet
words, and the second and third words are 32-bit binary numbers, so
KUMQUAT would be just as effective if it merely prepended a 4-octet
word that resembled the first word of an RTP header.

Using this simplified KUMQUAT encapsulation, encapsulation of SCTP is
straightforward:  Just prepend the KUMQUAT header word.

If RTP is being encapsulated with KUMQUAT, there is no reason not to
encapsulate SCTP or SCTP-over-DTLS with it also.  Encapsulation allows
demultiplexing without requiring distinct SCTP port numbers between
media descriptions, and the octet savings of not encapsulating SCTP is
small (given the relative size of the RTP).


The SHIM encapsulation
(draft-westerlund-avtcore-transport-multiplexing) uses a two-octet
prefix containing a 14-bit "session ID".  Each SDP media description
has an attribute to specify the session IDs to be used to encapsulate
RTP and the RTCP for that media description.

The remaining two bits of the two-octet prefix are fixed as "10",
presumably so the first octet values have the same range as those of
RTP/RTCPv2.

Although the draft does not say so, an SCTP packet stream can be a
constituent session of a SHIM encapsulation in the same way as an RTP
packet stream.


One requirement question that should be examined is whether, if
encapsulation is done, there is any value in making the encapsulated
packet look like a RTP packet, or whether there is any benefit to
preserving four-octet alignment of RTP fields.

The major advantage of an encapsulated solution over an unencapsulated
solution is that an encapsulated solution does not require the
constituent media descriptions to have disjoint sets of payload types
(or SCTP port numbers).  Since previous uses of SDP have no
interaction between the payload types of different media descriptions,
existing entities may have difficulty coordinating payload types
between media descriptions.  E.g., if the entity is assembling its
media streams from several sources, and thus assembling its SDP from
several constituent SDPs, the disjointness restriction may require the
entity to recode the payload type numbers of constituent transport
flows in order to multiplex them.

Dale

From richard.ejzak@alcatel-lucent.com  Tue Feb 26 12:41:41 2013
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Hi Christer,
Thanks for your comments.  See below.

Richard

> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> A few comments on the -01 version.
>=20
> ----------
>=20
> Section 1:
>=20
> Q1: You say that, if the first line of the offer is rejected, and one
> has to wait for the second offer, it may cause a setup time delay.
> There may be cases where the application doesn't want to session to
> succeed in the first place in that case, so we can have text which
> recommends adding an "essential" (if available) m- line first.
> Typically I guess it would be an audio m- line.

SDP doesn't really have the concept of an "essential" m line, so this recom=
mendation reduces but does not eliminate the possibility that this will hap=
pen.

>=20
> ----------
>=20
> Section 2:
>=20
> Q2: You indicate, that if an empty re-INVITE is sent to the WebRTC
> client, it will create a BUNDLE offer. We can simply say that, unless
> there is an BUNDLE indicator (yes, we may have to define a feature tag,
> option tag, etc), the WebRTC should not create a BUNDLE offer - it
> should create a legacy offer, with different ports.

If we added a feature tag for this it would only cover some use cases while=
 others would still not be covered.  A better solution is one that is less =
likely to cause failure in any use case and more likely to work with at mos=
t a second offer/answer.  That is at least my current assessment of the ris=
k.

Just to be clear regarding your terminology, you mean by BUNDLE offer one i=
n which all ports are the same (corresponding to the 2nd and subsequent off=
ers in the latest BUNDLE draft) and a legacy offer is still an SDP offering=
 BUNDLE but with different ports (corresponding to the first offer in your =
draft)?  Or do you mean a fully legacy offer without BUNDLE?

>=20
> ----------
>=20
> Section 3.1:
>=20
> Q3: The text says that, if the answerer does not use BUNDLE, the
> offerer will have to send a second offer. But, you don't mention that
> this would also add delay to the session establishment, as the answerer
> can't do anything with the m- lines with unspecified address.

You are correct.  I thought this was clear from context but should have men=
tioned it.  Fortunately with this option a second offer is only needed for =
the non-bundle (lower priority) case.  Note that as the default in my prefe=
rred hybrid proposal (section 3.3) I propose to use the same first offer fo=
rmat as the current first BUNDLE offer to avoid this potential extra offer/=
answer.

>=20
> ----------
>=20
> Section 3.1:
>=20
> Q4: The text says:
>=20
> "The middle box will not allocate resources for these m lines  when
> forwarding the SDP offer, and the SDP answerer will usually  respond
> with the unspecified address for these m lines.  Even if the  SDP
> answer includes valid connection information for these m lines,  the
> middle box will still not allocate separate transport flows for  them."
>=20
> As I've said before, you still DO expect the intermediary to take those
> m- lines into consideration when calculating bandwidth, checking media
> types, payload type information, crypto information etc etc etc.

Agreed.  I should have emphasized this point in the draft based on your ear=
lier comment.  My bad.  Since this aspect is inherited from the current BUN=
DLE draft, along with many other things, I forgot to mention it.

>=20
> ----------
>=20
> Section 3.1:
>=20
> Q5: The text says:
>=20
> "If the SDP answerer chooses to reject the first m line(s) in the
> bundle group in the SDP offer (by setting port in the m line to  zero),
> it places the intended port, connection, candidate and DTLS  crypto
> information for the bundle in the first valid m line of the  SDP answer
> and includes the unspecified address for all other m lines  in the
> bundle group.  Because of this, it is understood by both  endpoints
> that the 5-tuple connection, port and DTLS crypto  information is to be
> based on the first valid m line in the SDP offer  and the first valid m
> line in the SDP answer."
>=20
> So, assume that in the offer the first m- line contains a valid
> address, but in the answer the second m- line contains a valid address
> (because the answerer rejected the first m- line).
>=20
> Now, in this case, based on the text in previous section, an
> intermediary will not reserve resources for any of the m- lines:
>=20
> - It will not reserve resources based on the first m- line, as it is
> rejected
> - It will not reserve resources based on the second m- line, as the
> offer contained an unspecified address
>=20
> Or, do you expect media to be sent in one direction using one m- line,
> and in another direction using another m- line?

This (as you pointed out in Q1, rare) case requires a second offer/answer f=
or simple intermediaries that do not understand BUNDLE but are needed, e.g.=
, to open pinholes.  While the idea of using different m line offer/answer =
pairs in the 2 directions is intriguing, I think this is a non-starter sinc=
e we will have non-symmetric RTP and it would be difficult to switch back t=
o a single m line representation.

This is not my preferred option for the first offer in a session.  My prefe=
rred option is based on section 3.2, where the offer is formatted exactly a=
s in the current BUNDLE draft but the answer differs.  A second offer is st=
ill needed if the answerer rejects the first m line in the offer, but this =
should not usually happen (see Q1).

>=20
> ----------
>=20
> Regards,
>=20
> Christer

From christer.holmberg@ericsson.com  Wed Feb 27 02:01:40 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [MMUSIC] Another bundling proposal - Christer's comments
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Another bundling proposal - Christer's comments
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Hi,

>>>> Second, you use the non-bundle m- lines for codec negotiation etc.=20
>>>> But, previously, when MMT was discussed, I believe that people=20
>>>> indicated that they do not want to "refer" to other m- lines from=20
>>>> the MMT m- line. Instead,  the information should be explicitly=20
>>>> provided in the MMT m- line (which is the reason why the SDP size=20
>>>> gets pretty big with MMT, as most information is duplicated).
>>>
>>> I've heard that mentioned before, but I've never seen the argument=20
>>> in favor of it.  Can someone tell me what it is, or provide a=20
>>> pointer?
>>=20
>>It was indicated in a meeting (at the moment I can't remember which=20
>>meeting, though), but I never checked whether it was added to the=20
>>minutes. I guess it will be on audio recording.
>>=20
>>But, as far as I remember, there were never much discussion about it.=20
>>It was an open issue on one of my slides, and people simply indicated=20
>>support for adding explicit information, rather the referencing.
>>
>If this desire exists, I'd like to hear technical reasons why it is desire=
d. =20

As far as I remember, there were no real technical reasons - it was mostly =
"cosmetic".

But, one issue is if there is an intermediary (e.g. transcoder) that modifi=
es the payload type/codec information in the non-BUNDLE m- lines, but leave=
s the BUNDLE m- line untouched and passes the "kumquat" payload transparent=
ly. The offerer and answerer would not have different payload type/codec in=
formation.

(Of course, if you assume that such intermediary would always remove the KU=
MQUAT grouping attribute value, the case won't occur.)

>Adding payload types to the bundle media description makes it more difficu=
lt to ensure that a non-supporting answerer rejects the bundle media descri=
ption.

It depends on how it's done. You could still only indicate "kumquat" on the=
 m- line, but then provide the encapsulated payload type information using =
SDP attributes.

Regards,

Christer


From fluffy@iii.ca  Wed Feb 27 07:39:05 2013
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From: Cullen Jennings <fluffy@iii.ca>
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Subject: [MMUSIC] Request for agenda time for draft-jennings-rtcweb-plan
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I would like to request agenda time for the following MMUSIC-related =
topics from draft-jennings-rtcweb-plan:

- The multi-render attribute
- Simulcast/layered coding/FEC m-line representations.
- MSID structure and use for stream identification

I believe I will need 40 minutes for these topics. It would be best if =
that were Friday so that it follows the RTCWEB discussion on the big =
picture.

Cullen=

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From: Suhas Nandakumar <suhasietf@gmail.com>
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Subject: [MMUSIC] Request for agenda time for draft-nandakumar-mmusic-sdp-mux-attributes
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Hi,

I would like to request agenda time for the following draft:

 draft-nandakumar-mmusic-sdp-mux-attributes

Due to the complexity and depth of analysis, I believe I will
need 60 minutes.

Suhas

--089e0112c512c323f104d6b6a162
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<div><span style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-ser=
if;font-size:13px">Hi,=A0</span></div><span style=3D"font-family:&#39;Segoe=
 UI&#39;,Helvetica,Arial,sans-serif;font-size:13px"><div><span style=3D"fon=
t-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;font-size:13px"><br>
</span></div>I would like to request agenda time for the following draft:</=
span><br style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif=
;font-size:13px"><br style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Aria=
l,sans-serif;font-size:13px">
<span style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;fo=
nt-size:13px">=A0draft-nandakumar-mmusic-sdp-mux-attributes</span><br style=
=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;font-size:13p=
x">
<br style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;font=
-size:13px"><span style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,s=
ans-serif;font-size:13px">Due to the complexity and depth of analysis, I be=
lieve I will</span><br style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Ar=
ial,sans-serif;font-size:13px">
<span style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;fo=
nt-size:13px">need 60 minutes.</span><br style=3D"font-family:&#39;Segoe UI=
&#39;,Helvetica,Arial,sans-serif;font-size:13px"><br style=3D"font-family:&=
#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;font-size:13px">
<span style=3D"font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif;fo=
nt-size:13px">Suhas</span>

--089e0112c512c323f104d6b6a162--

From suhasietf@gmail.com  Wed Feb 27 07:50:41 2013
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From: Suhas Nandakumar <suhasietf@gmail.com>
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Subject: [MMUSIC] Agenda Time request for draft-nandakumar-mmusic-sdp-mux-attributes
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--047d7b86cea463ca9404d6b6bc6b
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Hi,

I would like to request agenda time for the following draft:

 draft-nandakumar-mmusic-sdp-mux-attributes

Due to the complexity and depth of analysis, I believe I will
need 60 minutes.


Thanks
Suhas
PS: Resending due to some network issues. Hope it reaches this time

--047d7b86cea463ca9404d6b6bc6b
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<div style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13=
px;background-color:rgb(255,255,255)"><span style=3D"font-family:&#39;Segoe=
 UI&#39;,Helvetica,Arial,sans-serif">Hi,=A0</span></div><span style=3D"colo=
r:rgb(34,34,34);font-size:13px;background-color:rgb(255,255,255);font-famil=
y:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif"><div>
<br></div>I would like to request agenda time for the following draft:</spa=
n><br style=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb(255,=
255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif"><br sty=
le=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb(255,255,255);=
font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif">
<span style=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb(255,=
255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif">=A0draf=
t-nandakumar-mmusic-sdp-mux-attributes</span><br style=3D"color:rgb(34,34,3=
4);font-size:13px;background-color:rgb(255,255,255);font-family:&#39;Segoe =
UI&#39;,Helvetica,Arial,sans-serif">
<br style=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb(255,25=
5,255);font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif"><span sty=
le=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb(255,255,255);=
font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif">Due to the compl=
exity and depth of analysis, I believe I will</span><br style=3D"color:rgb(=
34,34,34);font-size:13px;background-color:rgb(255,255,255);font-family:&#39=
;Segoe UI&#39;,Helvetica,Arial,sans-serif">
<span style=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb(255,=
255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif">need 60=
 minutes.</span><div><span style=3D"color:rgb(34,34,34);font-size:13px;back=
ground-color:rgb(255,255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Aria=
l,sans-serif"><br>
</span></div><div><span style=3D"color:rgb(34,34,34);font-size:13px;backgro=
und-color:rgb(255,255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Arial,s=
ans-serif"><br></span></div><div><span style=3D"color:rgb(34,34,34);font-si=
ze:13px;background-color:rgb(255,255,255);font-family:&#39;Segoe UI&#39;,He=
lvetica,Arial,sans-serif">Thanks</span></div>
<div><span style=3D"color:rgb(34,34,34);font-size:13px;background-color:rgb=
(255,255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Arial,sans-serif">Su=
has</span></div><div><span style=3D"color:rgb(34,34,34);font-size:13px;back=
ground-color:rgb(255,255,255);font-family:&#39;Segoe UI&#39;,Helvetica,Aria=
l,sans-serif">PS: Resending due to some network issues. Hope it reaches thi=
s time</span></div>

--047d7b86cea463ca9404d6b6bc6b--

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Subject: [MMUSIC] RFC 6871 on Session Description Protocol (SDP) Media Capabilities Negotiation
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6871

        Title:      Session Description Protocol (SDP) Media 
                    Capabilities Negotiation 
        Author:     R. Gilman, R. Even,
                    F. Andreasen
        Status:     Standards Track
        Stream:     IETF
        Date:       February 2013
        Mailbox:    bob_gilman@comcast.net, 
                    roni.even@mail01.huawei.com, 
                    fandreas@cisco.com
        Pages:      55
        Characters: 124829
        Updates:    RFC5939

        I-D Tag:    draft-ietf-mmusic-sdp-media-capabilities-17.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6871.txt

Session Description Protocol (SDP) capability negotiation provides a
general framework for indicating and negotiating capabilities in SDP.
The base framework defines only capabilities for negotiating
transport protocols and attributes.  This documents extends the
framework by defining media capabilities that can be used to
negotiate media types and their associated parameters.

This document updates the IANA Considerations of RFC 5939.

This document is a product of the Multiparty Multimedia Session Control Working Group of the IETF.

This is now a Proposed Standard.

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

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specifically noted otherwise on the RFC itself, all RFCs are for
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The RFC Editor Team
Association Management Solutions, LLC



From fandreas@cisco.com  Wed Feb 27 12:22:23 2013
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Subject: [MMUSIC] MMUSIC Draft Agenda for IETF 86
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The MMUSIC draft agenda for IETF 86 has been posted at:

     https://datatracker.ietf.org/meeting/86/agenda/mmusic/

Please take a look and let us know if you have any questions or comments.

Thanks

     Ari & Flemming

From worley@shell01.TheWorld.com  Wed Feb 27 13:57:53 2013
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: mmusic@ietf.org
Subject: [MMUSIC] Better way to not-really-offer constituent media descriptions
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This is a revision of how how constituent MDs are presented in the
offer in draft-worley-sdp-bundle.  Instead of offering them with a
non-zero port and null address (as if they were on-hold in the old
manner), offer them with a real port and address (but only provide ICE
candidates that are already known, and don't allocate a TURN relay).

(The idea that we can have the answerer suppress the constituent MDs
rather than the offerer is taken from
draft-ejzak-mmusic-bundle-alternatives-01.)

        o=- 2890844526 2890844526 IN IP4 host.example.com
        c=IN IP4 10.0.1.1

        a=group:KUMQUAT bundle con1 con2

        m=audio 10002 RTP/AVP 0 8 97
        c=IN IP4 0.0.0.0
        a=mid:con1
        a=rtcp-mux
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:97 iLBC/8000
        a=candidate:0 1 UDP 2113601791 10.0.1.1 10002 typ host

        m=video 10004 RTP/AVP 31 32
        c=IN IP4 0.0.0.0
        a=mid:con2
        a=rtcp-mux
        a=rtpmap:31 H261/90000
        a=rtpmap:32 MPV/90000
        a=candidate:0 1 UDP 2113601791 10.0.1.1 10004 typ host

        m=audio 10000 RTP/AVP 127
        a=mid:bundle
        a=rtcp-mux
        a=rtpmap:127 kumquat
        a=candidate:0 1 UDP 2113601791 10.0.1.1 10000 typ host
        a=candidate:1 1 UDP 1694194431 198.51.100.32 51000 typ srflx \
            raddr 10.0.1.1 rport 10000

If the answerer understands bundling, the answer is as before, with a
non-zero port but a null address for the constituent MDs.  (I assume
that intermediate devices realize that a null address means that it
will see no media on that transport flow.)

           o=- 2890844526 2890844526 IN IP4 answer.example.com
           c=IN IP4 10.0.2.1

           a=group:KUMQUAT bundle con1 con2

           m=audio 9 RTP/AVP 0 8 97
           c=IN IP4 0.0.0.0
           a=mid:con1
           a=rtcp-mux
           a=rtpmap:0 PCMU/8000
           a=rtpmap:8 PCMA/8000
           a=rtpmap:97 iLBC/8000

           m=video 9 RTP/AVP 31 32
           c=IN IP4 0.0.0.0
           a=mid:con2
           a=rtcp-mux
           a=rtpmap:31 H261/90000
           a=rtpmap:32 MPV/90000

           m=audio 20000 RTP/AVP 127
           a=mid:bundle
           a=rtcp-mux
           a=rtpmap:127 kumquat
           a=candidate:0 1 UDP 2113601791 10.0.2.1 20000 typ host
           a=candidate:1 1 UDP 1694194431 198.51.100.35 51090 typ srflx \
               raddr 10.0.2.1 rport 20000

If the answerer does not understand bundling, since the offer includes
real transport addresses, there is no risk that answerer wil not see
the constituent MDs as being offered.  The answer now has real
transport address information for the constituent MDs, including full
ICE candidates.  But since the offer didn't include the full ICE
candidates, it's possible that media communications won't be
established.  The offerer will have to update its offer with complete
ICE information, which may require allocating more TURN relays.

Non-supporting answer:

           o=- 2890844526 2890844526 IN IP4 answer.example.com
           c=IN IP4 10.0.2.1

           m=audio 20000 RTP/AVP 0 8 97
           c=IN IP4 10.0.2.1
           a=rtcp-mux
           a=rtpmap:0 PCMU/8000
           a=rtpmap:8 PCMA/8000
           a=rtpmap:97 iLBC/8000
           a=candidate:0 1 UDP 2113601791 10.0.2.1 20000 typ host
           a=candidate:1 1 UDP 1694194431 198.51.100.35 51090 typ srflx \
               raddr 10.0.2.1 rport 20000

           m=audio 20002 RTP/AVP 31 32
           c=IN IP4 10.0.2.1
           a=rtcp-mux
           a=rtpmap:31 H261/90000
           a=rtpmap:32 MPV/90000
           a=candidate:0 1 UDP 2113601791 10.0.2.1 20002 typ host
           a=candidate:1 1 UDP 1694194431 198.51.100.35 51092 typ srflx \
               raddr 10.0.2.1 rport 20002

           m=audio 0 RTP/AVP 127

Updated offer, with further ICE candidates:

           o=- 2890844526 2890844527 IN IP4 host.example.com
           c=IN IP4 10.0.1.1

           m=audio 10000 RTP/AVP 0 8 97
           c=IN IP4 10.0.1.1
           a=mid:con1
           a=rtcp-mux
           a=rtpmap:0 PCMU/8000
           a=rtpmap:8 PCMA/8000
           a=rtpmap:97 iLBC/8000
           a=candidate:0 1 UDP 2113601791 10.0.1.1 10000 typ host
           a=candidate:1 1 UDP 1694194431 198.51.100.32 51000 typ srflx \
               raddr 10.0.1.1 rport 10000

           m=video 10002 RTP/AVP 31 32
           c=IN IP4 10.0.1.1
           a=mid:con2
           a=rtcp-mux
           a=rtpmap:31 H261/90000
           a=rtpmap:32 MPV/90000
           a=candidate:0 1 UDP 2113601791 10.0.1.1 10002 typ host
           a=candidate:1 1 UDP 1694194431 198.51.100.32 51002 typ srflx \
               raddr 10.0.1.1 rport 10002

           m=audio 0 RTP/AVP 127
           a=mid:bundle

The updated answer is the same as the first answer.

There also a fast-start alternative where the offerer provides full
ICE candidates for the constituent MDs.  This avoids the delay of
updating the offer when interoperating with a non-supporting answerer
and full ICE candidates are needed, but may require pre-allocating
additional TURN relays.

Dale

From worley@shell01.TheWorld.com  Wed Feb 27 13:58:44 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: mmusic@ietf.org
Subject: [MMUSIC] Where to apply encryption?
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Current bundling proposals seem to expect that the packets on the wire
will be either SRTP/SRTCP or SCTP-within-DTLS.  Of course, this
provides encryption of the carried media.

But it seems to me that it would be more straightforward to multiplex
RTP/RTCP and SCTP packets, and than as a lower layer, have one DTLS
association that encrypts all of those packets indifferently.  It
would also provide privacy regarding the number and types of the
bundled media streams.

But my knowledge of crypto is thin, and maybe there's a reason that
using one DTLS association to encrypt the multiplexed packet stream
wouldn't work as well.

Dale

From martin.thomson@gmail.com  Wed Feb 27 14:11:29 2013
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Subject: Re: [MMUSIC] Where to apply encryption?
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There are no security reasons why this wouldn't work.  Some
applications already do this.

The actual problems are manifold:

 - SRTP was designed to have a low per-packet overhead.  The DTLS
record layer has a larger per-packet overhead.
 - SRTP exposes certain attributes in clear text for intermediaries to
use.  Intentionally.  Using DTLS would lose this.
 - SRTP enjoys wide support.  This would harm interoperability with
existing communication devices, forcing the use of more complex
gateways.

--Martin

On 27 February 2013 13:58, Dale R. Worley <worley@ariadne.com> wrote:
> Current bundling proposals seem to expect that the packets on the wire
> will be either SRTP/SRTCP or SCTP-within-DTLS.  Of course, this
> provides encryption of the carried media.
>
> But it seems to me that it would be more straightforward to multiplex
> RTP/RTCP and SCTP packets, and than as a lower layer, have one DTLS
> association that encrypts all of those packets indifferently.  It
> would also provide privacy regarding the number and types of the
> bundled media streams.
>
> But my knowledge of crypto is thin, and maybe there's a reason that
> using one DTLS association to encrypt the multiplexed packet stream
> wouldn't work as well.
>
> Dale
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From: Richard Barnes <rlb@ipv.sx>
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Subject: Re: [MMUSIC] Where to apply encryption?
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Yes.  Back in the days of the ZRTP / DTLS wars, the possibility of just
using DTLS was discussed, but morphed into DTLS-SRTP for the reasons Martin
describes.

On Wednesday, February 27, 2013, Martin Thomson wrote:

> There are no security reasons why this wouldn't work.  Some
> applications already do this.
>
> The actual problems are manifold:
>
>  - SRTP was designed to have a low per-packet overhead.  The DTLS
> record layer has a larger per-packet overhead.
>  - SRTP exposes certain attributes in clear text for intermediaries to
> use.  Intentionally.  Using DTLS would lose this.
>  - SRTP enjoys wide support.  This would harm interoperability with
> existing communication devices, forcing the use of more complex
> gateways.
>
> --Martin
>
> On 27 February 2013 13:58, Dale R. Worley <worley@ariadne.com<javascript:;>>
> wrote:
> > Current bundling proposals seem to expect that the packets on the wire
> > will be either SRTP/SRTCP or SCTP-within-DTLS.  Of course, this
> > provides encryption of the carried media.
> >
> > But it seems to me that it would be more straightforward to multiplex
> > RTP/RTCP and SCTP packets, and than as a lower layer, have one DTLS
> > association that encrypts all of those packets indifferently.  It
> > would also provide privacy regarding the number and types of the
> > bundled media streams.
> >
> > But my knowledge of crypto is thin, and maybe there's a reason that
> > using one DTLS association to encrypt the multiplexed packet stream
> > wouldn't work as well.
> >
> > Dale
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org <javascript:;>
> > https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org <javascript:;>
> https://www.ietf.org/mailman/listinfo/mmusic
>

--e89a8ff1c2fc85353e04d6bc6d2c
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Yes. <span></span>=A0Back in the days of the ZRTP / DTLS wars, the possibil=
ity of just using DTLS was discussed, but morphed into DTLS-SRTP for the re=
asons Martin describes.=A0<br><br>On Wednesday, February 27, 2013, Martin T=
homson  wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">There are no security reasons why this would=
n&#39;t work. =A0Some<br>
applications already do this.<br>
<br>
The actual problems are manifold:<br>
<br>
=A0- SRTP was designed to have a low per-packet overhead. =A0The DTLS<br>
record layer has a larger per-packet overhead.<br>
=A0- SRTP exposes certain attributes in clear text for intermediaries to<br=
>
use. =A0Intentionally. =A0Using DTLS would lose this.<br>
=A0- SRTP enjoys wide support. =A0This would harm interoperability with<br>
existing communication devices, forcing the use of more complex<br>
gateways.<br>
<br>
--Martin<br>
<br>
On 27 February 2013 13:58, Dale R. Worley &lt;<a href=3D"javascript:;" oncl=
ick=3D"_e(event, &#39;cvml&#39;, &#39;worley@ariadne.com&#39;)">worley@aria=
dne.com</a>&gt; wrote:<br>
&gt; Current bundling proposals seem to expect that the packets on the wire=
<br>
&gt; will be either SRTP/SRTCP or SCTP-within-DTLS. =A0Of course, this<br>
&gt; provides encryption of the carried media.<br>
&gt;<br>
&gt; But it seems to me that it would be more straightforward to multiplex<=
br>
&gt; RTP/RTCP and SCTP packets, and than as a lower layer, have one DTLS<br=
>
&gt; association that encrypts all of those packets indifferently. =A0It<br=
>
&gt; would also provide privacy regarding the number and types of the<br>
&gt; bundled media streams.<br>
&gt;<br>
&gt; But my knowledge of crypto is thin, and maybe there&#39;s a reason tha=
t<br>
&gt; using one DTLS association to encrypt the multiplexed packet stream<br=
>
&gt; wouldn&#39;t work as well.<br>
&gt;<br>
&gt; Dale<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmu=
sic@ietf.org&#39;)">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;mmusic@i=
etf.org&#39;)">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote>

--e89a8ff1c2fc85353e04d6bc6d2c--

From pkyzivat@alum.mit.edu  Wed Feb 27 14:47:55 2013
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Subject: Re: [MMUSIC] Better way to not-really-offer constituent media descriptions
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On 2/28/13 5:56 AM, Dale R. Worley wrote:

> (The idea that we can have the answerer suppress the constituent MDs
> rather than the offerer is taken from
> draft-ejzak-mmusic-bundle-alternatives-01.)
...
> If the answerer understands bundling, the answer is as before, with a
> non-zero port but a null address for the constituent MDs.  (I assume
> that intermediate devices realize that a null address means that it
> will see no media on that transport flow.)

If an O/A exchange has been concluded and the result is that one end has 
a valid address, and the other end has null (zero/.invalid) address, 
can't that be construed as establishing a oneway media stream where RTCP 
is also oneway?

I don't think this has ever been discussed.

	Thanks,
	Paul


From mary.ietf.barnes@gmail.com  Wed Feb 27 16:15:59 2013
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Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 86
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I think it would be much clearer and more accurate if you would
separate out the documents under the "RTCWEB related drafts" into
those that are more general multi-stream related documents and those
that are RTCWEB specific - e.g., draft-jennings-rtcweb-plan-01 is
clearly an RTCWEB related draft. Whereas, many of the other drafts are
proposing solutions that might well be useful for other applications
such as CLUE.  I think it's really important that the MMUSIC WG keeps
in mind that RTCWEB requirements are not the only ones that need to be
considered in the context of the multi-stream work.

Thanks,
Mary

On Wed, Feb 27, 2013 at 2:22 PM, Flemming Andreasen <fandreas@cisco.com> wrote:
> The MMUSIC draft agenda for IETF 86 has been posted at:
>
>     https://datatracker.ietf.org/meeting/86/agenda/mmusic/
>
> Please take a look and let us know if you have any questions or comments.
>
> Thanks
>
>     Ari & Flemming
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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To: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 86
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On Wed, Feb 27, 2013 at 3:53 PM, Mary Barnes <mary.ietf.barnes@gmail.com> wrote:
> I think it would be much clearer and more accurate if you would
> separate out the documents under the "RTCWEB related drafts" into
> those that are more general multi-stream related documents and those
> that are RTCWEB specific - e.g., draft-jennings-rtcweb-plan-01 is
> clearly an RTCWEB related draft.

Actually, that document says this:

   "The requirements listed here are a collection of requirements that
   have come from WebRTC, CLUE, and the general community that uses RTP
   for interactive communications based on Offer/Answer."

So filename may not be the perfect indicator of author intent.

Ted

> Whereas, many of the other drafts are
> proposing solutions that might well be useful for other applications
> such as CLUE.  I think it's really important that the MMUSIC WG keeps
> in mind that RTCWEB requirements are not the only ones that need to be
> considered in the context of the multi-stream work.
>
> Thanks,
> Mary
>
> On Wed, Feb 27, 2013 at 2:22 PM, Flemming Andreasen <fandreas@cisco.com> wrote:
>> The MMUSIC draft agenda for IETF 86 has been posted at:
>>
>>     https://datatracker.ietf.org/meeting/86/agenda/mmusic/
>>
>> Please take a look and let us know if you have any questions or comments.
>>
>> Thanks
>>
>>     Ari & Flemming
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 86
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On Feb 27, 2013, at 4:34 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> Actually, that document says this:
>=20
>   "The requirements listed here are a collection of requirements that
>   have come from WebRTC, CLUE, and the general community that uses RTP
>   for interactive communications based on Offer/Answer."
>=20
> So filename may not be the perfect indicator of author intent.

plan is pretty much a grab bag list of things so a bunch of it might =
relate to all things using RTP/SDP while others are more or less rtcweb =
specific and at lest one is probably only specific to CLUE. It's hard to =
name.=

From mary.ietf.barnes@gmail.com  Thu Feb 28 06:21:34 2013
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On Wed, Feb 27, 2013 at 6:34 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
> On Wed, Feb 27, 2013 at 3:53 PM, Mary Barnes <mary.ietf.barnes@gmail.com> wrote:
>> I think it would be much clearer and more accurate if you would
>> separate out the documents under the "RTCWEB related drafts" into
>> those that are more general multi-stream related documents and those
>> that are RTCWEB specific - e.g., draft-jennings-rtcweb-plan-01 is
>> clearly an RTCWEB related draft.
>
> Actually, that document says this:
>
>    "The requirements listed here are a collection of requirements that
>    have come from WebRTC, CLUE, and the general community that uses RTP
>    for interactive communications based on Offer/Answer."
>
> So filename may not be the perfect indicator of author intent.
>
> Ted
[MB] Yeah - I had just assumed that was rtcweb only stuff.  But, that
just re-enforces my point that I don't believe this section should be
referenced as "RTCWEB related drafts" - either include CLUE in the
header or make the topic more general. I actually prefer the former as
I think it's really important for more of the CLUE participants to get
engaged in these discussions and understand that the decisions in
MMUSIC need CLUE input and may impact the overall CLUE solution. [/MB]
>
>> Whereas, many of the other drafts are
>> proposing solutions that might well be useful for other applications
>> such as CLUE.  I think it's really important that the MMUSIC WG keeps
>> in mind that RTCWEB requirements are not the only ones that need to be
>> considered in the context of the multi-stream work.
>>
>> Thanks,
>> Mary
>>
>> On Wed, Feb 27, 2013 at 2:22 PM, Flemming Andreasen <fandreas@cisco.com> wrote:
>>> The MMUSIC draft agenda for IETF 86 has been posted at:
>>>
>>>     https://datatracker.ietf.org/meeting/86/agenda/mmusic/
>>>
>>> Please take a look and let us know if you have any questions or comments.
>>>
>>> Thanks
>>>
>>>     Ari & Flemming
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic

From worley@shell01.TheWorld.com  Thu Feb 28 11:48:16 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [MMUSIC] Better way to not-really-offer constituent media	descriptions
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> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> 
> If an O/A exchange has been concluded and the result is that one end has 
> a valid address, and the other end has null (zero/.invalid) address, 
> can't that be construed as establishing a oneway media stream where RTCP 
> is also oneway?

Aye, that's true, assuming the old model "null address = on-hold".  So
you'd need an SDP update to clear that up after connecting with a
supporting endpoint.  Although the update wouldn't be needed to get
media flowing.

Dale

From dwing@cisco.com  Thu Feb 28 14:52:01 2013
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From: "Dan Wing" <dwing@cisco.com>
To: "'Dale R. Worley'" <worley@ariadne.com>, <mmusic@ietf.org>
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Subject: Re: [MMUSIC] Where to apply encryption?
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Dale R. Worley
> Sent: Wednesday, February 27, 2013 1:58 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] Where to apply encryption?
> 
> Current bundling proposals seem to expect that the packets on the wire
> will be either SRTP/SRTCP or SCTP-within-DTLS.  Of course, this provides
> encryption of the carried media.
> 
> But it seems to me that it would be more straightforward to multiplex
> RTP/RTCP and SCTP packets, and than as a lower layer, have one DTLS
> association that encrypts all of those packets indifferently.  It would
> also provide privacy regarding the number and types of the bundled media
> streams.

SRTP (RFC3711) was carefully and purposefully designed so the RTP header 
remains in the clear, and DTLS-SRTP was also purposefully designed so
after the DTLS-SRTP handshake establishes the SRTP keys, the SRTP packets
flow over the wire exactly like they would for any other keying 
mechanism (ZRTP, MIKEY-*, Security Descriptions).

By sending SRTP packets with their RTP header in the clear, it allows:
  * the RTP header to be compressed (cRTP [RFC2508], ECRTP [RFC3545])
  * analysis devices to see where, on a network path, SRTP packets 
    are lost, delayed, or get mis-ordered.
  * analysis devices to see SRTCP receiver reports and sender reports,
    which are typically authenticated but in the clear.

> But my knowledge of crypto is thin, and maybe there's a reason that
> using one DTLS association to encrypt the multiplexed packet stream
> wouldn't work as well.

-d

> Dale
> _______________________________________________
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