Using Simulcast in SDP and RTP Sessions
draft-ietf-mmusic-sdp-simulcast-04
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 8853.
|
|
|---|---|---|---|
| Authors | Bo Burman , Magnus Westerlund , Suhas Nandakumar , Mo Zanaty | ||
| Last updated | 2016-02-03 | ||
| Replaces | draft-burman-mmusic-sdp-simulcast | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
| Document shepherd | Flemming Andreasen | ||
| IESG | IESG state | Became RFC 8853 (Proposed Standard) | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
draft-ietf-mmusic-sdp-simulcast-04
Network Working Group B. Burman
Internet-Draft M. Westerlund
Intended status: Standards Track Ericsson
Expires: August 6, 2016 S. Nandakumar
M. Zanaty
Cisco
February 3, 2016
Using Simulcast in SDP and RTP Sessions
draft-ietf-mmusic-sdp-simulcast-04
Abstract
In some application scenarios it may be desirable to send multiple
differently encoded versions of the same media source in different
RTP streams. This is called simulcast. This document describes how
to accomplish simulcast in RTP and how to signal it in SDP. The
described solution uses an RTP/RTCP identification method to identify
RTP streams belonging to the same media source, and makes an
extension to SDP to relate those RTP streams as being different
simulcast formats of that media source. The SDP extension consists
of a new media level SDP attribute that expresses capability to send
and/or receive simulcast RTP streams.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 6, 2016.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 3
2.2. Requirements Language . . . . . . . . . . . . . . . . . . 4
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Reaching a Diverse Set of Receivers . . . . . . . . . . . 5
3.2. Application Specific Media Source Handling . . . . . . . 6
3.3. Receiver Media Source Preferences . . . . . . . . . . . . 7
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 7
5. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 8
6. Detailed Description . . . . . . . . . . . . . . . . . . . . 9
6.1. Simulcast Attribute . . . . . . . . . . . . . . . . . . . 9
6.2. Simulcast Capability . . . . . . . . . . . . . . . . . . 11
6.2.1. Declarative Use . . . . . . . . . . . . . . . . . . . 13
6.2.2. Offer/Answer Use . . . . . . . . . . . . . . . . . . 13
6.3. Relating Simulcast Streams . . . . . . . . . . . . . . . 15
6.4. Signaling Examples . . . . . . . . . . . . . . . . . . . 15
6.4.1. Single-Source Client . . . . . . . . . . . . . . . . 16
6.4.2. Multi-Source Client . . . . . . . . . . . . . . . . . 17
7. Network Aspects . . . . . . . . . . . . . . . . . . . . . . . 20
7.1. Bitrate Adaptation . . . . . . . . . . . . . . . . . . . 20
8. Limitations . . . . . . . . . . . . . . . . . . . . . . . . . 21
8.1. Single RTP Session . . . . . . . . . . . . . . . . . . . 21
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21
10. Security Considerations . . . . . . . . . . . . . . . . . . . 21
11. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 22
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 22
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 22
13.1. Normative References . . . . . . . . . . . . . . . . . . 22
13.2. Informative References . . . . . . . . . . . . . . . . . 23
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 25
A.1. Modifications Between WG Version -03 and -04 . . . . . . 25
A.2. Modifications Between WG Version -02 and -03 . . . . . . 26
A.3. Modifications Between WG Version -01 and -02 . . . . . . 26
A.4. Modifications Between WG Version -00 and -01 . . . . . . 26
A.5. Modifications Between Individual Version -00 and WG
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Version -00 . . . . . . . . . . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 26
1. Introduction
Most of today's multiparty video conference solutions make use of
centralized servers to reduce the bandwidth and CPU consumption in
the endpoints. Those servers receive RTP streams from each
participant and send some suitable set of possibly modified RTP
streams to the rest of the participants, which usually have
heterogeneous capabilities (screen size, CPU, bandwidth, codec, etc).
One of the biggest issues is how to perform RTP stream adaptation to
different participants' constraints with the minimum possible impact
on both video quality and server performance.
Simulcast is defined in this memo as the act of simultaneously
sending multiple different encoded streams of the same media source,
e.g. the same video source encoded with different video encoder types
or image resolutions. This can be done in several ways and for
different purposes. This document focuses on the case where it is
desirable to provide a media source as multiple encoded streams over
RTP [RFC3550] towards an intermediary so that the intermediary can
provide the wanted functionality by selecting which RTP stream(s) to
forward to other participants in the session, and more specifically
how the identification and grouping of the involved RTP streams are
done.
This document describes a few scenarios where it is motivated to use
simulcast, and also defines the needed RTP/RTCP and SDP signaling for
it.
2. Definitions
2.1. Terminology
This document makes use of the terminology defined in RTP Taxonomy
[RFC7656], and RTP Topologies [RFC7667]. In addition, the following
terms are used:
RTP Mixer: An RTP middle node, defined in [RFC7667] (Section 3.6 to
3.9).
RTP Switch: A common short term for the terms "switching RTP mixer",
"source projecting middlebox", and "video switching MCU" as
discussed in [RFC7667].
Simulcast Stream: One Encoded Stream or Dependent Stream from a set
of concurrently transmitted Encoded Streams and optional Dependent
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Streams, all sharing a common Media Source, as defined in
[RFC7656]. Decoding a Dependent Stream also requires the related
(Dependent and) Encoded Stream(s), but in the context of simulcast
that is considered a property of the Dependent Stream constituting
the simulcast stream. For example, HD and thumbnail video
simulcast versions of a single Media Source sent concurrently as
separate RTP Streams.
Simulcast Format: Different formats of a simulcast stream serve the
same purpose as alternative RTP payload types in non-simulcast
SDP, to allow multiple alternative media formats for a given RTP
Stream. As for multiple RTP payload types on the m-line, any one
of the alternative formats can be used at a given point in time,
but not more than one (based on RTP timestamp), and what format is
used can change dynamically from one RTP packet to another. For
example, if all participants in a group video call can decode
H.264 and H.265 video, but only some can encode H.265, both H.264
and H.265 can be kept as alternative formats, and the format may
dynamically switch between H.264 and H.265 as different
participants become active speaker.
2.2. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. Use Cases
Many use cases of simulcast as described in this document relate to a
multi-party communication session where one or more central nodes are
used to adapt the view of the communication session towards
individual participants, and facilitate the media transport between
participants. Thus, these cases targets the RTP Mixer type of
topology.
There are two principle approaches for an RTP Mixer to provide this
adapted view of the communication session to each receiving
participant:
o Transcoding (decoding and re-encoding) received RTP streams with
characteristics adapted to each receiving participant. This often
include mixing or composition of media sources from multiple
participants into a mixed media source originated by the RTP
Mixer. The main advantage of this approach is that it achieves
close to optimal adaptation to individual receiving participants.
The main disadvantages are that it can be very computationally
expensive to the RTP Mixer and typically also degrades media
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Quality of Experience (QoE) such as end-to-end delay for the
receiving participants.
o Switching a subset of all received RTP streams or sub-streams to
each receiving participant, where the used subset is typically
specific to each receiving participant. The main advantages of
this approach are that it is computationally cheap to the RTP
Mixer and it has very limited impact on media QoE. The main
disadvantage is that it can be difficult to combine a subset of
received RTP streams into a perfect fit to the resource situation
of a receiving participant.
The use of simulcast relates to the latter approach, where it is more
important to reduce the load on the RTP Mixer and/or minimize QoE
impact than to achieve an optimal adaptation of resource usage.
3.1. Reaching a Diverse Set of Receivers
The media sources provided by a sending participant potentially need
to reach several receiving participants that differ in terms of
available resources. The receiver resources that typically differ
include, but are not limited to:
Codec: This includes codec type (such as SDP MIME type) and can
include codec configuration options (e.g. SDP fmtp parameters).
A couple of codec resources that differ only in codec
configuration will be "different" if they are somehow not
"compatible", like if they differ in video codec profile, or the
transport packetization configuration.
Sampling: This relates to how the media source is sampled, in
spatial as well as in temporal domain. For video streams, spatial
sampling affects image resolution and temporal sampling affects
video frame rate. For audio, spatial sampling relates to the
number of audio channels and temporal sampling affects audio
bandwidth. This may be used to suit different rendering
capabilities or needs at the receiving endpoints, as well as a
method to achieve different transport capabilities, bitrates and
eventually QoE by controlling the amount of source data.
Bitrate: This relates to the amount of bits spent per second to
transmit the media source as an RTP stream, which typically also
affects the Quality of Experience (QoE) for the receiving user.
Letting the sending participant create a simulcast of a few
differently configured RTP streams per media source can be a good
tradeoff when using an RTP switch as middlebox, instead of sending a
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single RTP stream and using an RTP mixer to create individual
transcodings to each receiving participant.
This requires that the receiving participants can be categorized in
terms of available resources and that the sending participant can
choose a matching configuration for a single RTP stream per category
and media source.
For example, assume for simplicity a set of receiving participants
that differ only in that some have support to receive Codec A, and
the others have support to receive Codec B. Further assume that the
sending participant can send both Codec A and B. It can then reach
all receivers by creating two simulcasted RTP streams from each media
source; one for Codec A and one for Codec B.
In another simple example, a set of receiving participants differ
only in screen resolution; some are able to display video with at
most 360p resolution and some support 720p resolution. A sending
participant can then reach all receivers by creating a simulcast of
RTP streams with 360p and 720p resolution for each sent video media
source.
In more elaborate cases, the receiving participants differ both in
available sampling and bitrate, and maybe also codec, and it is up to
the RTP switch to find a good trade-off in which simulcasted stream
to choose for each intended receiver. It is also the responsibility
of the RTP switch to negotiate a good fit of simulcast streams with
the sending participant.
The maximum number of simulcasted RTP streams that can be sent is
mainly limited by the amount of processing and uplink network
resources available to the sending participant.
3.2. Application Specific Media Source Handling
The application logic that controls the communication session may
include special handling of some media sources. It is for example
commonly the case that the media from a sending participant is not
sent back to itself.
It is also common that a currently active speaker participant is
shown in larger size or higher quality than other participants (the
sampling or bitrate aspects of Section 3.1). Not sending the active
speaker media back to itself means there is some other participant's
media that instead has to receive special handling towards the active
speaker; typically the previous active speaker. This way, the
previously active speaker is needed both in larger size (to current
active speaker) and in small size (to the rest of the participants),
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which can be solved with a simulcast from the previously active
speaker to the RTP switch.
3.3. Receiver Media Source Preferences
The application logic that controls the communication session may
allow receiving participants to apply preferences to the
characteristics of the RTP stream they receive, for example in terms
of the aspects listed in Section 3.1. Sending a simulcast of RTP
streams is one way of accommodating receivers with conflicting or
otherwise incompatible preferences.
4. Requirements
The following requirements need to be met to support the use cases in
previous sections:
Editor's note: Consider adding an explicit requirement that the
solution supports use of simulcast even when using multiple codecs
and multiple redundant RTP streams per defined codec (or something
similar), since this is really an existing requirement and should
also fully motivate the use of RID as identification mechanism.
REQ-1: Identification. It must be possible to identify a set of
simulcasted RTP streams as originating from the same media source:
REQ-1.1: In SDP signaling.
REQ-1.2: On RTP/RTCP level.
REQ-2: Transport usage. The solution must work when using:
REQ-2.1: Legacy SDP with separate media transports per SDP media
description.
REQ-2.2: Bundled [I-D.ietf-mmusic-sdp-bundle-negotiation] SDP
media descriptions.
REQ-3: Capability negotiation. It must be possible that:
REQ-3.1: Sender can express capability of sending simulcast.
REQ-3.2: Receiver can express capability of receiving simulcast.
REQ-3.3: Sender can express maximum number of simulcast streams
that can be provided.
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REQ-3.4: Receiver can express maximum number of simulcast streams
that can be received.
REQ-3.5: Sender can detail the characteristics of the simulcast
streams that can be provided.
REQ-3.6: Receiver can detail the characteristics of the simulcast
streams that it prefers to receive.
REQ-4: Distinguishing features. It must be possible to have
different simulcast streams use different codec parameters, as can
be expressed by SDP format values and RTP payload types.
REQ-5: Compatibility. It must be possible to use simulcast in
combination with other RTP mechanisms that generate additional RTP
streams:
REQ-5.1: RTP Retransmission [RFC4588].
REQ-5.2: RTP Forward Error Correction [RFC5109].
REQ-5.3: Related payload types such as audio Comfort Noise and/or
DTMF.
REQ-6: Interoperability. The solution must be possible to use in:
REQ-6.1: Interworking with non-simulcast legacy clients using a
single media source per media type.
REQ-6.2: WebRTC environment with a single media source per SDP
media description.
5. Overview
As an overview, the above requirements are met by signaling simulcast
capability and configurations in SDP [RFC4566]:
o An offer or answer can contain a number of simulcast streams,
separate for send and receive directions.
o An offer or answer can contain multiple, alternative simulcast
stream formats in the same fashion as multiple, alternative codecs
can be offered in a media description.
o A single media source per SDP media description is assumed, which
is aligned with the concepts defined in [RFC7656] and will
specifically work in a WebRTC context, both with and without
BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] grouping.
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o The codec configuration for a simulcast stream is expressed
through use of a separately specified RTP-level identification
mechanism [I-D.ietf-mmusic-rid][I-D.roach-avtext-rid], which
complements and effectively extends the available simulcast stream
identification and configuration possibilities that could be
provided by using only SDP formats.
o It is possible, but not required to use source-specific signaling
[RFC5576] with the proposed solution.
6. Detailed Description
This section further details the overview above (Section 5). First,
formal syntax is provided (Section 6.1), followed by the rest of the
SDP attribute definition in Section 6.2. Relating Simulcast Streams
(Section 6.3) provides the definition of the RTP/RTCP mechanisms
used. The section is concluded with a number of examples.
6.1. Simulcast Attribute
Name: simulcast
Value: sc-value
Usage Level: media
Charset Dependent: no
Multiplex Category: NORMAL
Syntax [RFC5234]:
sc-attr = "a=simulcast:" sc-value
sc-value = sc-str-list [SP sc-str-list]
sc-str-list = sc-dir SP sc-alt-list *( ";" sc-alt-list )
sc-dir = "send" / "recv"
sc-alt-list = sc-id *( "," sc-id )
sc-id-paused = "~"
sc-id = [sc-id-paused] rid-identifier / token
; SP defined in [RFC5234]
; token defined in [RFC4566]
; rid-identifier defined in [I-D.ietf-mmusic-rid]
Figure 1: ABNF for Simulcast
The "a=simulcast" attribute has a parameter in the form of one or two
simulcast stream descriptions, each consisting of a direction ("send"
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or "recv"), followed by a list of one or more simulcast streams.
Each simulcast stream in that list is separated by a semicolon (";").
Each simulcast stream can in turn be offered in one or more
alternative formats, where each simulcast stream alternative is
separated by a comma (","). The simulcast stream alternative MUST be
described in the form of a RID, as described by
[I-D.ietf-mmusic-rid]. Each simulcast stream can be initially paused
[I-D.ietf-avtext-rtp-stream-pause], indicated by prepending a "~" to
the simulcast stream. In case there are simulcast stream
alternatives, pause can be specified individually for each
alternative. The reason to allow separate initial pause states for
each simulcast stream alternative is that pause capability can be
specified individually for each RTP payload type referenced by a RID,
which makes it infeasible to pause RID where any of the related RTP
payload type(s) do not have pause capability.
Examples:
a=simulcast:send 1,2,3;~4,~5 recv 1;~2,~5
a=simulcast:recv 1;4,5 send 1;2
Figure 2: Simulcast Examples
Above are two examples of different "a=simulcast" lines.
The first line is an example offer to send two simulcast streams and
to receive two simulcast streams. The first simulcast stream in send
direction can be sent as three different alternatives (1, 2, 3), and
the second simulcast stream in send direction can be sent as two
different alternatives (4, 5). All second stream send alternatives
are offered as initially paused. The first simulcast stream in
receive direction has no alternatives (only 1). The second simulcast
stream in receive direction has two alternatives (2, 5) that are both
offered as initially paused.
The second line is an example answer to the first line, accepting to
send and receive the two offered simulcast streams, however send and
receive directions are specified in opposite order compared to the
first line, which lets the answer keep the same order of simulcast
streams in the SDP as in the offer, even though directionality is
reversed. This example answer has removed all offered alternatives
for the first simulcast stream (keeping only 1), but kept alternative
formats for the second simulcast stream in receive direction (4, 5).
The answer accepts to send two simulcast streams, without
alternatives. The answer does not accept initial pause of any
simulcast streams, in either direction. More examples can be found
in Section 6.4.
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6.2. Simulcast Capability
Simulcast capability is expressed as a new media level SDP attribute,
"a=simulcast" (Section 6.1), with multiplex category
[I-D.ietf-mmusic-sdp-mux-attributes] NORMAL.
For each desired direction (send/recv), the simulcast attribute
defines a list of simulcast streams (separated by semicolons), each
of which is a list of simulcast formats (separated by commas). The
meaning of the attribute on SDP session level is undefined and MUST
NOT be used.
The meaning of including multiple "a=simulcast" lines in a single SDP
media description is undefined and MUST NOT be used. There are
separate and independent sets of parameters for simulcast in send and
receive directions. When listing multiple directions, each direction
MUST NOT occur more than once on the same line.
The different simulcast streams MUST be identified through the RTP-
level "RID" identification mechanism [I-D.ietf-mmusic-rid].
Attribute parameters are grouped by direction and consist of a
listing of simulcast stream identifications to be used. The number
of (non-alternative, see below) identifications in the list sets a
limit to the number of supported simulcast streams in that direction.
The order of the listed simulcast versions in the "send" direction
suggests a proposed order of preference, in decreasing order: the
stream listed first is the most preferred Section 3.1, and subsequent
streams have progressively lower preference. The order of the listed
simulcast streams in the "recv" direction expresses a preference
which simulcast streams that are preferred, with the leftmost being
most preferred. This can be of importance if the number of actually
sent simulcast streams have to be reduced for some reason.
Formats that have explicit dependencies [RFC5583]
[I-D.ietf-mmusic-rid] to other formats (even in the same media
description) MAY be listed as different simulcast streams.
Alternative simulcast formats MAY be specified as part of the
attribute parameters by expressing each simulcast stream as a comma-
separated list of alternative format identifiers. In this case, it
is not possible to align what alternative formats that are used
between different simulcast streams, like requiring all simulcast
streams to use alternatives with the same codec format. The order of
the format alternatives within a simulcast stream is significant; the
alternatives are listed from (left) most preferred to (right) least
preferred. For the use of simulcast, this overrides the normal codec
preference as expressed by format type ordering on the "m="-line,
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using regular SDP rules. This is to enable a separation of general
codec preferences and simulcast stream configuration preferences.
A simulcast stream can use a codec defined such that the same RTP
SSRC can change RTP payload type multiple times during a session,
possibly even on a per-packet basis. A typical example can be a
speech codec that makes use of Comfort Noise [RFC3389] and/or DTMF
[RFC4733] formats. In those cases, such "related" formats MUST NOT
be listed explicitly in the attribute parameters, since they are not
strictly simulcast streams of the media source, but rather a specific
way of generating the RTP stream of a single simulcast stream with
varying RTP payload type. Instead, only a single simulcast stream
identification MUST be used per simulcast stream or alternative
simulcast format (if there are such) in the SDP.
If RTP stream pause/resume [I-D.ietf-avtext-rtp-stream-pause] is
supported, any simulcast stream identification MAY be prefixed by a
"~" character to indicate that the corresponding simulcast stream is
initially paused already from start of the RTP session. In this
case, support for RTP stream pause/resume MUST also be included under
the same "m="-line listing "a=simulcast". If the simulcast stream is
specified as a list of alternative formats, the indication is
prepended to the first format of the list and applies to whatever
alternative that is eventually chosen. All RTP payload types related
to such initially paused simulcast stream MUST be listed in the SDP
as pause/resume capable as specified by
[I-D.ietf-avtext-rtp-stream-pause].
An initially paused simulcast stream in "send" direction MUST be
considered equivalent to an unsolicited locally paused stream, and be
handled accordingly. Initially paused simulcast streams are resumed
as described by the RTP pause/resume specification. An RTP stream
receiver that wishes to resume an unsolicited locally paused stream
needs to know the SSRC of that stream. The SSRC of an initially
paused simulcast stream can be obtained from an RTP stream sender
RTCP Sender Report (SR) including both the desired SSRC as "SSRC of
sender", and the stream RID identification as an RID RTCP SDES item.
Including an initially paused simulcast stream in "recv" direction in
an SDP towards an RTP sender, SHOULD cause the remote RTP sender to
put the stream as unsolicited locally paused, unless there are other
RTP stream receivers that do not mark the simulcast stream as
initially paused. The reason to require an initially paused "recv"
stream to be considered locally paused by the remote RTP sender,
instead of making it equivalent to implicitly sending a pause
request, is because the pausing RTP sender cannot know which SSRC
owns the restriction when TMMBR/TMMBN are used for pause/resume
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signaling since the RTP receiver's SSRC in send direction is not
known yet.
Use of the redundant audio data [RFC2198] format could be seen as a
form of simulcast for loss protection purposes, but is not considered
conflicting with the mechanisms described in this memo and MAY
therefore be used as any other format. In this case the "red"
format, rather than the carried formats, SHOULD be the one to list as
a simulcast stream on the "a=simulcast" line.
The media formats and corresponding characteristics of simulcast
streams SHOULD be chosen such that they are different. If this
difference is not required, RTP duplication [RFC7104] procedures
SHOULD be considered instead of simulcast.
6.2.1. Declarative Use
When used as a declarative media description, "a=simulcast" line
"recv" direction formats indicate the configured end point's required
capability to recognize and receive a specified set of RTP streams as
simulcast streams. In the same fashion, "a=simulcast" line "send"
direction requests the end point to send a specified set of RTP
streams as simulcast streams.
If multiple simulcast formats are listed, it means that the
configured end point MUST be prepared to receive any of the "recv"
formats, and MAY send any of the "send" formats for that simulcast
stream.
Editor's note: It may not be beneficial for declarative use to be
limited to a single media source per "m=" line, as elaborated
further in Section 8.
6.2.2. Offer/Answer Use
An offerer wanting to use simulcast SHALL include the "a=simulcast"
attribute in the offer. An offerer that receives an answer without
"a=simulcast" MUST NOT use simulcast towards the answerer. An
offerer that receives an answer with "a=simulcast" without any
simulcast stream identifications in a specified direction MUST NOT
use simulcast in that direction.
An answerer that does not understand the concept of simulcast will
also not know the attribute and will remove it in the SDP answer, as
defined in existing SDP Offer/Answer [RFC3264] procedures.
An answerer that does understand the attribute and that wants to
support simulcast in an indicated direction SHALL reverse
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directionality of the unidirectional direction parameters; "send"
becomes "recv" and vice versa, and include it in the answer.
An offerer listing a set of receive simulcast streams and/or
alternative formats in the offer MUST be prepared to receive RTP
streams for any of those simulcast streams and/or alternative formats
from the answerer.
An answerer that receives an offer with simulcast containing an
"a=simulcast" attribute listing alternative formats for simulcast
streams MAY keep all the alternatives in the answer, but it MAY also
choose to remove any non-desirable alternatives per simulcast stream
in the answer. The answerer MUST NOT add any alternatives that were
not present in the offer.
An answerer that receives an offer with simulcast that lists a number
of simulcast streams, MAY reduce the number of simulcast streams in
the answer, but MUST NOT add simulcast streams.
An offerer that receives an answer where some simulcast formats are
kept MUST be prepared to receive any of the kept send direction
alternatives, and MAY send any of the kept receive direction
alternatives from the answer. Similarly, the answerer MUST be
prepared to receive any of the kept receive direction alternatives,
and MAY send any of the kept send direction alternatives in the
answer.
The offerer and answerer MUST NOT send more than a single alternative
format at a time (based on RTP timestamps) per simulcast stream, but
MAY change format on a per-RTP packet basis. This corresponds to the
existing (non-simulcast) SDP offer/answer case when multiple formats
are included on the "m=" line in the SDP answer.
An offerer that receives an answer where some of the simulcast
streams are removed MAY release the corresponding resources (codec,
transport, etc) in its receive direction and MUST NOT send any RTP
packets corresponding to the removed simulcast streams.
Simulcast streams or formats using undefined simulcast stream
identifications MUST NOT be used as valid simulcast streams by an RTP
stream receiver.
An answerer that receives an offer without RTP stream pause/resume
capability MUST NOT mark any simulcast streams as initially paused in
the answer.
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An answerer that receives an offer with RTP stream pause/resume
capability MAY mark any simulcast streams as initially paused in the
answer.
An answerer that receives indication in an offer of a simulcast
stream being initially paused , SHOULD mark that simulcast stream as
initially paused also in the answer, regardless of direction, unless
it has good reason for the stream not being initially paused.
An offerer that offered some of its simulcast streams as initially
paused and that receives an answer that does not indicate RTP stream
pause/resume capability, MUST NOT intially pause any simulcast
streams.
An offerer with RTP stream pause/resume capability that receives an
answer where some simulcast streams are marked as initially paused,
SHOULD initially pause them regardless if they were marked as
initially paused also in the offer, unless it has good reason for
those streams not being initially paused.
Note: The inclusion of "a=simulcast" or the use of simulcast does
not change any of the interpretation or Offer/Answer procedures
for other SDP attributes, like "a=fmtp" or "a=rid".
6.3. Relating Simulcast Streams
Simulcast RTP streams MUST be related on RTP level through RID
[I-D.roach-avtext-rid], as specified in the SDP "a=simulcast"
attribute (Section 6.2) parameters. This is sufficient as long as
there is only a single media source per SDP media description. When
using BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], where multiple
SDP media descriptions jointly specify a single RTP session, the SDES
MID identification mechanism in BUNDLE allows relating RTP streams
back to individual media descriptions, after which the above
described RID relations can be used. Use of the RTP header extension
[RFC5285] for both MID and RID identifications can be important to
ensure rapid initial reception, required to correctly interpret and
process the RTP streams. Implementers of this specification MUST
support RTCP source description (SDES) item and SHOULD support RTP
header extension method to signal RID on RTP level.
6.4. Signaling Examples
These examples describe a client to video conference service, using a
centralized media topology with an RTP mixer.
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+---+ +-----------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| F |<---->| |<---->| J |
+---+ +-----------+ +---+
Figure 3: Four-party Mixer-based Conference
6.4.1. Single-Source Client
Alice is calling in to the mixer with a simulcast-enabled client
capable of a single media source per media type. The client can send
a simulcast of 2 video resolutions and frame rates: HD 1280x720p
30fps and thumbnail 320x180p 15fps. This is defined below using the
"imageattr" [RFC6236]. In this example, only the "pt" RID parameter
is used, effectively achieving a 1:1 mapping between RID and media
formats (RTP payload types), to describe simulcast stream formats.
Alice's Offer:
v=0
o=alice 2362969037 2362969040 IN IP4 192.0.2.156
s=Simulcast Enabled Client
t=0 0
c=IN IP4 192.0.2.156
m=audio 49200 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 49300 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=rid:1 pt=97
a=rid:2 pt=98
a=simulcast:send 1;2 recv 1
Figure 4: Single-Source Simulcast Offer
The only thing in the SDP that indicates simulcast capability is the
line in the video media description containing the "simulcast"
attribute. The included format parameters indicates that sent
simulcast streams can differ in video resolution.
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The Answer from the server indicates that it too is simulcast
capable. Should it not have been simulcast capable, the
"a=simulcast" line would not have been present and communication
would have started with the media negotiated in the SDP.
v=0
o=server 823479283 1209384938 IN IP4 192.0.2.2
s=Answer to Simulcast Enabled Client
t=0 0
c=IN IP4 192.0.2.43
m=audio 49672 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 49674 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=rid:1 pt=97
a=rid:2 pt=98
a=simulcast:recv 1;2 send 1
Figure 5: Single-Source Simulcast Answer
Since the server is the simulcast media receiver, it reverses the
direction of the "simulcast" attribute parameters.
6.4.2. Multi-Source Client
Fred is calling in to the same conference as in the example above
with a two-camera, two-display system, thus capable of handling two
separate media sources in each direction, where each media source is
simulcast-enabled in the send direction. Fred's client is restricted
to a single media source per media description.
The first two simulcast streams for the first media source use
different codecs, H264-SVC [RFC6190] and H264 [RFC6184]. These two
simulcast streams also have a temporal dependency. Two different
video codecs, VP8 [I-D.ietf-payload-vp8] and H264, are offered as
alternatives for the third simulcast stream for the first media
source. Only the highest fidelity simulcast stream are sent from
start, the lower fidelity streams being initially paused.
The second media source is offered with three different simulcast
streams. All video streams of this second media source are loss
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protected by RTP retransmission [RFC4588]. Also here, all but the
highest fidelity simulcast stream are initially paused.
Fred's client is also using BUNDLE to send all RTP streams from all
media descriptions in the same RTP session on a single media
transport. Although using many different simulcast streams in this
example, the use of RID as simulcast stream identification enables
use of a low number of RTP payload types. Note that the use of both
BUNDLE and RID recommends using the RTP header extension [RFC5285]
for carrying these fields, which is consequently also included in the
SDP.
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v=0
o=fred 238947129 823479223 IN IP4 192.0.2.125
s=Offer from Simulcast Enabled Multi-Source Client
t=0 0
c=IN IP4 192.0.2.125
a=group:BUNDLE foo bar zen
m=audio 49200 RTP/AVP 99
a=mid:foo
a=rtpmap:99 G722/8000
m=video 49600 RTP/AVPF 100 101 103
a=mid:bar
a=rtpmap:100 H264-SVC/90000
a=rtpmap:101 H264/90000
a=rtpmap:103 VP8/90000
a=fmtp:100 profile-level-id=42400d; max-fs=3600; max-mbps=108000; \
mst-mode=NI-TC
a=fmtp:101 profile-level-id=42c00d; max-fs=3600; max-mbps=54000
a=fmtp:103 max-fs=900; max-fr=30
a=rid:1 send pt=100;max-width=1280;max-height=720;max-fr=60;depend=2
a=rid:2 send pt=101;max-width=1280;max-height=720;max-fr=30
a=rid:3 send pt=101;max-width=640;max-height=360
a=rid:4 send pt=103;max-width=640;max-height=360
a=depend:100 lay bar:101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rid
a=rtcp-fb:* ccm pause nowait
a=simulcast:send 1;2;~4,3
m=video 49602 RTP/AVPF 96 104
a=mid:zen
a=rtpmap:96 VP8/90000
a=fmtp:96 max-fs=3600; max-fr=30
a=rtpmap:104 rtx/90000
a=fmtp:104 apt=96;rtx-time=200
a=rid:5 send pt=96;max-fs=921600;max-fr=30
a=rid:6 send pt=96;max-fs=614400;max-fr=15
a=rid:7 send pt=96;max-fs=230400;max-fr=30
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rid
a=rtcp-fb:* ccm pause nowait
a=simulcast:send 5;~6;~7
Figure 6: Fred's Multi-Source Simulcast Offer
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Note: Empty lines in the SDP above are added only for readability
and would not be present in an actual SDP.
7. Network Aspects
Simulcast is in this memo defined as the act of sending multiple
alternative encoded streams of the same underlying media source.
When transmitting multiple independent streams that originate from
the same source, it could potentially be done in several different
ways using RTP. A general discussion on considerations for use of
the different RTP multiplexing alternatives can be found in
Guidelines for Multiplexing in RTP
[I-D.ietf-avtcore-multiplex-guidelines]. Discussion and
clarification on how to handle multiple streams in an RTP session can
be found in [I-D.ietf-avtcore-rtp-multi-stream].
The network aspects that are relevant for simulcast are:
Quality of Service: When using simulcast it might be of interest to
prioritize a particular simulcast stream, rather than applying
equal treatment to all streams. For example, lower bit-rate
streams may be prioritized over higher bit-rate streams to
minimize congestion or packet losses in the low bit-rate streams.
Thus, there is a benefit to use a simulcast solution with good QoS
support.
NAT/FW Traversal: Using multiple RTP sessions incurs more cost for
NAT/FW traversal unless they can re-use the same transport flow,
which can be achieved by Multiplexing Negotiation Using SDP Port
Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation].
7.1. Bitrate Adaptation
Use of multiple simulcast streams can require a significant amount of
network resources. If the amount of available network resources
varies during an RTP session such that it does not match what is
negotiated in SDP, the bitrate used by the different simulcast
streams may have to be reduced dynamically. What simulcast streams
to prioritize when allocating available bitrate among the simulcast
streams in such adaptation SHOULD be taken from the simulcast stream
order on the "a=simulcast" line. Simulcast streams that have pause/
resume capability and that would be given such low bitrate by the
adaptation process that they are considered not really useful can be
temporarily paused until the limiting condition clears.
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8. Limitations
The chosen approach has a few limitations that are described in this
section. The only one currently described relates to the use of a
single RTP session for all simulcast formats of a media source.
8.1. Single RTP Session
The limitations in this section come from sending all simulcast
streams related to a media source under the same SDP media
description, which also means they are sent in the same RTP session.
It is not possible to use different simulcast streams on different
media transports, limiting the possibilities to apply different QoS
to different simulcast streams. When using unicast, QoS mechanisms
based on individual packet marking are feasible, since they do not
require separation of simulcast streams into different RTP sessions
to apply different QoS.
It is not possible to separate different simulcast streams into
different multicast groups to allow a multicast receiver to pick the
stream it wants, rather than receive all of them. In this case, the
only reasonable implementation is to use different RTP sessions for
each multicast group so that reporting and other RTCP functions
operate as intended.
9. IANA Considerations
This document requests to register a new SDP attribute, simulcast, as
defined in Section 6.1.
10. Security Considerations
The simulcast capability, configuration attributes, and parameters
are vulnerable to attacks in signaling.
A false inclusion of the "a=simulcast" attribute may result in
simultaneous transmission of multiple RTP streams that would
otherwise not be generated. The impact is limited by the media
description joint bandwidth, shared by all simulcast streams
irrespective of their number. There may however be a large number of
unwanted RTP streams that will impact the share of bandwidth
allocated for the originally wanted RTP stream.
A hostile removal of the "a=simulcast" attribute will result in
simulcast not being used.
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Neither of the above will likely have any major consequences and can
be mitigated by signaling that is at least integrity and source
authenticated to prevent an attacker to change it.
Security considerations related to the use of RID is covered in
[I-D.ietf-mmusic-rid] and [I-D.roach-avtext-rid]. There are no
additional security concerns related to its use in this
specification.
11. Contributors
Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have
contributed with important material to the first versions of this
document. Robert Hansen and Cullen Jennings, from Cisco, Peter
Thatcher, from Google, and Adam Roach, from Mozilla, contributed
significantly to subsequent versions.
12. Acknowledgements
13. References
13.1. Normative References
[I-D.ietf-avtext-rtp-stream-pause]
Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP
Stream Pause and Resume", draft-ietf-avtext-rtp-stream-
pause-10 (work in progress), September 2015.
[I-D.ietf-mmusic-rid]
Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
Roach, A., and B. Campen, "RTP Payload Format
Constraints", draft-ietf-mmusic-rid-01 (work in progress),
February 2016.
[I-D.ietf-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-12
(work in progress), January 2016.
[I-D.roach-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Payload
Format Constraints", draft-roach-avtext-rid-01 (work in
progress), February 2016.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <http://www.rfc-editor.org/info/rfc5109>.
[RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234,
DOI 10.17487/RFC5234, January 2008,
<http://www.rfc-editor.org/info/rfc5234>.
[RFC7104] Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol", RFC 7104,
DOI 10.17487/RFC7104, January 2014,
<http://www.rfc-editor.org/info/rfc7104>.
13.2. Informative References
[I-D.ietf-avtcore-multiplex-guidelines]
Westerlund, M., Perkins, C., and H. Alvestrand,
"Guidelines for using the Multiplexing Features of RTP to
Support Multiple Media Streams", draft-ietf-avtcore-
multiplex-guidelines-03 (work in progress), October 2014.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-25 (work in progress), January 2016.
[I-D.ietf-payload-flexible-fec-scheme]
Singh, V., Begen, A., Zanaty, M., and G. Mandyam, "RTP
Payload Format for Flexible Forward Error Correction
(FEC)", draft-ietf-payload-flexible-fec-scheme-01 (work in
progress), October 2015.
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[I-D.ietf-payload-vp8]
Westin, P., Lundin, H., Glover, M., Uberti, J., and F.
Galligan, "RTP Payload Format for VP8 Video", draft-ietf-
payload-vp8-17 (work in progress), September 2015.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
DOI 10.17487/RFC2198, September 1997,
<http://www.rfc-editor.org/info/rfc2198>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<http://www.rfc-editor.org/info/rfc3264>.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <http://www.rfc-editor.org/info/rfc3389>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733,
DOI 10.17487/RFC4733, December 2006,
<http://www.rfc-editor.org/info/rfc4733>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <http://www.rfc-editor.org/info/rfc5285>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<http://www.rfc-editor.org/info/rfc5576>.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, DOI 10.17487/RFC5583, July 2009,
<http://www.rfc-editor.org/info/rfc5583>.
[RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP
Payload Format for H.264 Video", RFC 6184,
DOI 10.17487/RFC6184, May 2011,
<http://www.rfc-editor.org/info/rfc6184>.
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[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
DOI 10.17487/RFC6190, May 2011,
<http://www.rfc-editor.org/info/rfc6190>.
[RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
Attributes in the Session Description Protocol (SDP)",
RFC 6236, DOI 10.17487/RFC6236, May 2011,
<http://www.rfc-editor.org/info/rfc6236>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<http://www.rfc-editor.org/info/rfc7667>.
Appendix A. Changes From Earlier Versions
NOTE TO RFC EDITOR: Please remove this section prior to publication.
A.1. Modifications Between WG Version -03 and -04
o Changed to only use RID identification, as was consensus during
IETF 94.
o ABNF improvements.
o Clarified offer-answer rules for initially paused streams.
o Changed references for RTP topologies and RTP taxonomy documents
that are now published as RFC.
o Added reference to the new RID draft in AVTEXT.
o Re-structured section 6 to provide an easy reference by the
updated IANA section.
o Added a sub-section 7.1 with a discussion of bitrate adaptation.
o Editorial improvements.
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A.2. Modifications Between WG Version -02 and -03
o Removed text on multicast / broadcast from use cases, since it is
not supported by the solution.
o Removed explicit references to unified plan draft.
o Added possibility to initiate simulcast streams in paused mode.
o Enabled an offerer to offer multiple stream identification (pt or
rid) methods and have the answerer choose which to use.
o Added a preference indication also in send direction offers.
o Added a section on limitations of the current proposal, including
identification method specific limitations.
A.3. Modifications Between WG Version -01 and -02
o Relying on the new RID solution for codec constraints and
configuration identification. This has resulted in changes in
syntax to identify if pt or RID is used to describe the simulcast
stream.
o Renamed simulcast version and simulcast version alternative to
simulcast stream and simulcast format respectively, and improved
definitions for them.
o Clarification that it is possible to switch between simulcast
version alternatives, but that only a single one be used at any
point in time.
o Changed the definition so that ordering of simulcast formats for a
specific simulcast stream do have a preference order.
A.4. Modifications Between WG Version -00 and -01
o No changes. Only preventing expiry.
A.5. Modifications Between Individual Version -00 and WG Version -00
o Added this appendix.
Authors' Addresses
Burman, et al. Expires August 6, 2016 [Page 26]
Internet-Draft Simulcast February 2016
Bo Burman
Ericsson
Kistavagen 25
SE-164 80 Stockholm
Sweden
Email: bo.burman@ericsson.com
Magnus Westerlund
Ericsson
Farogatan 2
SE-164 80 Stockholm
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Suhas Nandakumar
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: snandaku@cisco.com
Mo Zanaty
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: mzanaty@cisco.com
Burman, et al. Expires August 6, 2016 [Page 27]