Using Simulcast in Session Description Protocol (SDP) and RTP Sessions
RFC 8853
| Document | Type | RFC - Proposed Standard (January 2021; No errata) | |
|---|---|---|---|
| Authors | Bo Burman , Magnus Westerlund , Suhas Nandakumar , Mo Zanaty | ||
| Last updated | 2021-01-19 | ||
| Replaces | draft-burman-mmusic-sdp-simulcast | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html xml pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Flemming Andreasen | ||
| Shepherd write-up | Show (last changed 2018-03-06) | ||
| IESG | IESG state | RFC 8853 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Yes | ||
| Telechat date | |||
| Responsible AD | Ben Campbell | ||
| Send notices to | (None) | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | RFC-Ed-Ack | ||
Internet Engineering Task Force (IETF) B. Burman
Request for Comments: 8853 M. Westerlund
Category: Standards Track Ericsson
ISSN: 2070-1721 S. Nandakumar
M. Zanaty
Cisco
January 2021
Using Simulcast in Session Description Protocol (SDP) and RTP Sessions
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 the Session
Description Protocol (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 indicate that those RTP
streams are 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 is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8853.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
2. Definitions
2.1. Terminology
2.2. Requirements Language
3. Use Cases
3.1. Reaching a Diverse Set of Receivers
3.2. Application-Specific Media Source Handling
3.3. Receiver Media-Source Preferences
4. Overview
5. Detailed Description
5.1. Simulcast Attribute
5.2. Simulcast Capability
5.3. Offer/Answer Use
5.3.1. Generating the Initial SDP Offer
5.3.2. Creating the SDP Answer
5.3.3. Offerer Processing the SDP Answer
5.3.4. Modifying the Session
5.4. Use with Declarative SDP
5.5. Relating Simulcast Streams
5.6. Signaling Examples
5.6.1. Single-Source Client
5.6.2. Multisource Client
5.6.3. Simulcast and Redundancy
6. RTP Aspects
6.1. Outgoing from Endpoint with Media Source
6.2. RTP Middlebox to Receiver
6.2.1. Media-Switching Mixer
6.2.2. Selective Forwarding Middlebox
6.3. RTP Middlebox to RTP Middlebox
7. Network Aspects
7.1. Bitrate Adaptation
8. Limitation
9. IANA Considerations
10. Security Considerations
11. References
11.1. Normative References
11.2. Informative References
Appendix A. Requirements
Acknowledgements
Contributors
Authors' Addresses
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.
The intended scope of the defined mechanism is to support negotiation
and usage of simulcast when using SDP offer/answer and media
transport over RTP. The media transport topologies considered are
point-to-point RTP sessions, as well as centralized multiparty RTP
sessions, where a media sender will provide the simulcasted streams
to an RTP middlebox or endpoint, and middleboxes may further
distribute the simulcast streams to other middleboxes or endpoints.
Simulcast could be used point to point between middleboxes as part of
a distributed multiparty scenario. Usage of multicast or broadcast
transport is out of scope and left for future extensions.
This document describes a few scenarios that motivate the use of
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 "A Taxonomy of
Semantics and Mechanisms for Real-Time Transport Protocol (RTP)
Sources" [RFC7656] and "RTP Topologies" [RFC7667]. The following
terms are especially noted or here defined:
RTP mixer: An RTP middlebox, in the wide sense of the term,
encompassing Sections 3.6 to 3.9 of [RFC7667].
RTP session: An association among a group of participants
communicating with RTP, as defined in [RFC3550] and amended by
[RFC7656].
RTP stream: A stream of RTP packets containing media data, as
defined in [RFC7656].
RTP switch: A common short term for the terms "switching RTP mixer",
"source projecting middlebox", and "video switching Multipoint
Control Unit (MCU)", as discussed in [RFC7667].
Simulcast stream: One encoded stream or dependent stream from a set
of concurrently transmitted encoded streams and optional dependent
streams, all sharing a common media source, as defined in
[RFC7656]. 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 nonsimulcast SDP:
to allow multiple alternative media formats for a given RTP
stream. As for multiple RTP payload types on the "m=" line in
offer/answer [RFC3264], any one of the negotiated alternative
formats can be used in a single RTP stream at a given point in
time, but not more than one (based on RTP timestamp). What format
is used can change dynamically from one RTP packet to another.
2.2. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
3. Use Cases
The use cases of simulcast described in this document relate to a
multiparty 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 target the RTP mixer type of
topology.
There are two principal approaches for an RTP mixer to provide this
adapted view of the communication session to each receiving
participant:
* Transcoding (decoding and re-encoding) received RTP streams with
characteristics adapted to each receiving participant. This often
includes 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, typically degrades media Quality of
Experience (QoE) such as creating end-to-end delay for the
receiving participants, and requires the RTP mixer to have access
to media content.
* Switching a subset of all received RTP streams or substreams 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, has very limited impact on media QoE, and does not require
the RTP mixer to have (full) access to media content. The main
disadvantage is that it can be difficult to combine a subset of
received RTP streams into a perfect fit for the resource situation
of a receiving participant. It is also a disadvantage that
sending multiple RTP streams consumes more network resources from
the sending participant to the RTP mixer.
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 RTP payload format MIME
type) and can include codec configuration. A couple of codec
resources that differ only in codec configuration will be
"different" if they are somehow not "compatible", such as 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 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.
Bitrate: This relates to the number of bits sent per second to
transmit the media source as an RTP stream, which typically also
affects the 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
trade-off when using an RTP switch as middlebox, instead of sending a
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, 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 with best possible
resolution by creating a simulcast of RTP streams with 360p and 720p
resolution for each sent video media source.
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) in a receiving client.
Many conferencing systems do not send the active speaker's media back
to the sender itself, which means there is some other participant's
media that instead is forwarded to 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), 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 state preferences on the
characteristics of the RTP stream they like to 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. Overview
This memo defines SDP [RFC4566] signaling that covers the above
described simulcast use cases and functionalities. A number of
requirements for such signaling are elaborated in Appendix A.
The Restriction Identifier (RID) mechanism, as defined in [RFC8851],
enables an SDP offerer or answerer to specify a number of different
RTP stream restrictions for a rid-id by using the "a=rid" line.
Examples of such restrictions are maximum bitrate, maximum spatial
video resolution (width and height), maximum video frame rate, etc.
Each rid-id may also be restricted to use only a subset of the RTP
payload types in the associated SDP media description. Those RTP
payload types can have their own configurations and parameters
affecting what can be sent or received, using the "a=fmtp" line as
well as other SDP attributes.
A new SDP media-level attribute, "a=simulcast", is defined. The
attribute describes, independently for "send" and "receive"
directions, the number of simulcast RTP streams as well as potential
alternative formats for each simulcast RTP stream. Each simulcast
RTP stream, including alternatives, is identified using the RID
identifier (rid-id), defined in [RFC8851].
a=simulcast:send 1;2,3 recv 4
If this line is included in an SDP offer, the "send" part indicates
the offerer's capability and proposal to send two simulcast RTP
streams. Each simulcast stream is described by one or more RTP
stream identifiers (rid-ids), and each group of rid-ids for a
simulcast stream is separated by a semicolon (";"). When a simulcast
stream has multiple rid-ids that are separated by a comma (","), they
describe alternative representations for that particular simulcast
RTP stream. Thus, the "send" part shown above is interpreted as an
intention to send two simulcast RTP streams. The first simulcast RTP
stream is identified and restricted according to rid-id 1. The
second simulcast RTP stream can be sent as two alternatives,
identified and restricted according to rid-ids 2 and 3. The "recv"
part of the line shown here indicates that the offerer desires to
receive a single RTP stream (no simulcast) according to rid-id 4.
A more complete example SDP-offer media description is provided in
Figure 1.
m=video 49300 RTP/AVP 97 98 99
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=rtpmap:99 VP8/90000
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=fmtp:99 max-fs=240; max-fr=30
a=rid:1 send pt=97;max-width=1280;max-height=720
a=rid:2 send pt=98;max-width=320;max-height=180
a=rid:3 send pt=99;max-width=320;max-height=180
a=rid:4 recv pt=97
a=simulcast:send 1;2,3 recv 4
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
Figure 1: Example Simulcast Media Description in Offer
The SDP media description in Figure 1 can be interpreted at a high
level to say that the offerer is capable of sending two simulcast RTP
streams: one H.264 encoded stream in up to 720p resolution, and one
additional stream encoded as either H.264 or VP8 with a maximum
resolution of 320x180 pixels. The offerer can receive one H.264
stream with maximum 720p resolution.
The receiver of this SDP offer can generate an SDP answer that
indicates what it accepts. It uses the "a=simulcast" attribute to
indicate simulcast capability and specify what simulcast RTP streams
and alternatives to receive and/or send. An example of such an
answering "a=simulcast" attribute, corresponding to the above offer,
is:
a=simulcast:recv 1;2 send 4
With this SDP answer, the answerer indicates in the "recv" part that
it wants to receive the two simulcast RTP streams. It has removed an
alternative that it doesn't support (rid-id 3). The "send" part
confirms to the offerer that it will receive one stream for this
media source according to rid-id 4. The corresponding, more complete
example SDP answer media description could look like Figure 2.
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=rid:1 recv pt=97;max-width=1280;max-height=720
a=rid:2 recv pt=98;max-width=320;max-height=180
a=rid:4 send pt=97
a=simulcast:recv 1;2 send 4
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
Figure 2: Example Simulcast Media Description in Answer
It is assumed that a single SDP media description is used to describe
a single media source. This is aligned with the concepts defined in
[RFC7656] and will work in a WebRTC context, both with and without
BUNDLE grouping of media descriptions [RFC8843].
To summarize, the "a=simulcast" line describes "send"- and "receive"-
direction simulcast streams separately. Each direction can in turn
describe one or more simulcast streams, separated by semicolons. The
identifiers describing simulcast streams on the "a=simulcast" line
are rid-ids, as defined by "a=rid" lines in [RFC8851]. Each
simulcast stream can be offered as a list of alternative rid-ids,
with each alternative separated by a comma as shown in the example
offer in Figure 1. A detailed specification can be found in
Section 5, and more detailed examples are outlined in Section 5.6.
5. Detailed Description
This section provides further details to the overview in Section 4.
First, formal syntax is provided (Section 5.1), followed by the rest
of the SDP attribute definition in Section 5.2. "Relating Simulcast
Streams" (Section 5.5) provides the definition of the RTP/RTCP
mechanisms used. The section concludes with a number of examples.
5.1. Simulcast Attribute
This document defines a new SDP media-level "a=simulcast" attribute,
with value according to the syntax in Figure 3, which uses ABNF
[RFC5234] and its update, "Case-Sensitive String Support in ABNF"
[RFC7405]:
sc-value = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] )
sc-send = %s"send" SP sc-str-list
sc-recv = %s"recv" SP sc-str-list
sc-str-list = sc-alt-list *( ";" sc-alt-list )
sc-alt-list = sc-id *( "," sc-id )
sc-id-paused = "~"
sc-id = [sc-id-paused] rid-id
; SP defined in [RFC5234]
; rid-id defined in [RFC8851]
Figure 3: ABNF for Simulcast Value
The "a=simulcast" attribute has a parameter in the form of one or two
simulcast stream descriptions, each consisting of a direction ("send"
or "recv"), followed by a list of one or more simulcast streams.
Each simulcast stream consists of one or more alternative simulcast
formats. Each simulcast format is identified by a simulcast stream
identifier (rid-id). The rid-id MUST have the form of an RTP stream
identifier, as described by "RTP Payload Format Restrictions"
[RFC8851].
In the list of simulcast streams, each simulcast stream is separated
by a semicolon (";"). Each simulcast stream can, in turn, be offered
in one or more alternative formats, represented by rid-ids, separated
by commas (","). Each rid-id can also be specified as initially
paused [RFC7728], indicated by prepending a "~" to the rid-id. The
reason to allow separate initial pause states for each rid-id is that
pause capability can be specified individually for each RTP payload
type referenced by a rid-id. Since pause capability specified via
the "a=rtcp-fb" attribute applies only to specified payload types,
and a rid-id specified by "a=rid" can refer to multiple different
payload types, it is unfeasible to pause streams with rid-id where
any of the related RTP payload type(s) do not have pause capability.
5.2. Simulcast Capability
Simulcast capability is expressed through a new media-level SDP
attribute, "a=simulcast" (Section 5.1). The use of this attribute at
the session level is undefined. Implementations of this
specification MUST NOT use it at the session level and MUST ignore it
if received at the session level. Extensions to this specification
may define such session-level usage. Each SDP media description MUST
contain at most one "a=simulcast" line.
There are separate and independent sets of simulcast streams in the
"send" and "receive" directions. When listing multiple directions,
each direction MUST NOT occur more than once on the same line.
Simulcast streams using undefined rid-ids MUST NOT be used as valid
simulcast streams by an RTP stream receiver. The direction for a
rid-id MUST be aligned with the direction specified for the
corresponding RTP stream identifier on the "a=rid" line.
The listed number of simulcast streams for a direction sets a limit
to the number of supported simulcast streams in that direction. The
order of the listed simulcast streams in the "send" direction
suggests a proposed order of preference, in decreasing order: the
rid-id listed first is the most preferred, and subsequent streams
have progressively lower preference. The order of the listed rid-ids
in the "recv" direction expresses which simulcast streams are
preferred, with the leftmost being most preferred. This can be of
importance if the number of actually sent simulcast streams has to be
reduced for some reason.
rid-ids that have explicit dependencies [RFC5583] [RFC8851] to other
rid-ids (even in the same media description) MAY be used.
Use of more than a single, alternative simulcast format for a
simulcast stream MAY be specified as part of the attribute parameters
by expressing the simulcast stream as a comma-separated list of
alternative rid-ids. The order of the rid-id alternatives within a
simulcast stream is significant; the rid-id 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, using regular SDP rules.
This is to enable a separation of general codec preferences and
simulcast-stream configuration preferences. However, the choice of
which alternative to use per simulcast stream is independent, and
there is currently no mechanism for the offerer to force the answerer
to choose the same alternative for multiple simulcast streams.
A simulcast stream can use a codec defined such that the same RTP
synchronization source (SSRC) can change RTP payload type multiple
times during a session, possibly even on a per-packet basis. A
typical example is a speech codec that makes use of formats for
Comfort Noise [RFC3389] and/or dual-tone multifrequency (DTMF)
[RFC4733].
If RTP stream pause/resume [RFC7728] is supported, any rid-id MAY be
prefixed by a "~" character to indicate that the corresponding
simulcast stream is paused already from the start of the RTP session.
In this case, support for RTP stream pause/resume MUST also be
included under the same "m=" line where "a=simulcast" is included.
All RTP payload types related to such an initially paused simulcast
stream MUST be listed in the SDP as pause/resume capable as specified
by [RFC7728] -- e.g., by using the "*" wildcard format for "a=rtcp-
fb".
An initially paused simulcast stream in the "send" direction for the
endpoint sending the SDP MUST be considered equivalent to an
unsolicited locally paused stream and 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) or
Receiver Report (RR) that includes both the desired SSRC as initial
SSRC in the source description (SDES) chunk, optionally a MID SDES
item [RFC8843] (if used and if rid-ids are not unique across "m="
lines), and the rid-id value in an RtpStreamId RTCP SDES item
[RFC8852].
If the endpoint sending the SDP includes a "recv"-direction simulcast
stream that is initially paused, then the remote RTP sender receiving
the SDP SHOULD put its RTP stream in an unsolicited locally paused
state. The simulcast stream sender does not put the stream in the
locally paused state if there are other RTP stream receivers in the
session that do not mark the simulcast stream as initially paused.
However, in centralized conferencing, the RTP sender usually does not
see the SDP signaling from RTP receivers and cannot make this
determination. The reason for requiring that an initially paused
"recv" stream be considered locally paused by the remote RTP sender
instead of making it equivalent to implicitly sending a pause request
is that the pausing RTP sender cannot know which receiving SSRC owns
the restriction when Temporary Maximum Media Stream Bit Rate Request
(TMMBR) and Temporary Maximum Media Stream Bit Rate Notification
(TMMBN) are used for pause/resume signaling (Section 5.6 of
[RFC7728]); this is because the RTP receiver's SSRC in the "send"
direction is sometimes not yet known.
Use of the redundant audio data format [RFC2198] could be seen as a
form of simulcast for loss-protection purposes, but it 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 -- e.g., as
different SDP formats with differing "a=rtpmap" and/or "a=fmtp"
lines, or as differently defined RTP payload format restrictions. If
this difference is not required, it is RECOMMENDED to use RTP
duplication procedures [RFC7104] instead of simulcast. To avoid
complications in implementations, a single rid-id MUST NOT occur more
than once per "a=simulcast" line. Note that this does not eliminate
use of simulcast as an RTP duplication mechanism, since it is
possible to define multiple different rid-ids that are effectively
equivalent.
5.3. Offer/Answer Use
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, such as "a=fmtp" or "a=rid".
5.3.1. Generating the Initial SDP Offer
An offerer wanting to use simulcast for a media description SHALL
include one "a=simulcast" attribute in that media description in the
offer. An offerer listing a set of receive simulcast streams and/or
alternative formats as rid-ids in the offer MUST be prepared to
receive RTP streams for any of those simulcast streams and/or
alternative formats from the answerer.
5.3.2. Creating the SDP Answer
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 procedures [RFC3264]. Since SDP
session-level simulcast is undefined in this memo, an answerer that
receives an offer with the "a=simulcast" attribute on the SDP session
level SHALL remove it in the answer. An answerer that understands
the attribute but receives multiple "a=simulcast" attributes in the
same media description SHALL disable use of simulcast by removing all
"a=simulcast" lines for that media description in the answer.
An answerer that does understand the attribute and wants to support
simulcast in an indicated direction SHALL reverse directionality of
the unidirectional direction parameters -- "send" becomes "recv" and
vice versa -- and include it in the answer.
An answerer that receives an offer with simulcast containing an
"a=simulcast" attribute listing alternative rid-ids MAY keep all the
alternative rid-ids in the answer, but it MAY also choose to remove
any nondesirable alternative rid-ids in the answer. The answerer
MUST NOT add any alternative rid-ids in the "send" direction in the
answer that were not present in the offer receive direction. The
answerer MUST be prepared to receive any of the receive-direction
rid-id alternatives and MAY send any of the "send"-direction
alternatives that are part of the answer.
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 it MUST NOT add simulcast streams.
An answerer that receives an offer without RTP stream pause/resume
capability MUST NOT mark any simulcast streams as initially paused in
the answer.
An RTP stream answerer capable of pause/resume that receives an offer
with RTP stream pause/resume capability MAY mark any rid-ids that
refer to pause/resume capable formats as initially paused in the
answer.
An answerer that receives indication in an offer of a rid-id being
initially paused SHOULD mark that rid-id as initially paused also in
the answer, regardless of direction, unless it has good reason for
the rid-id not being initially paused. One reason to remove an
initial pause in the answer compared to the offer could be, for
example, that all "receive"-direction simulcast streams for a media
source the answerer accepts in the answer would otherwise be paused.
5.3.3. Offerer Processing the SDP Answer
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 rid-id in a specified direction MUST
NOT use simulcast in that direction.
An offerer that receives an answer where some rid-id alternatives are
kept MUST be prepared to receive any of the kept "send"-direction
rid-id alternatives and MAY send any of the kept "receive"-direction
rid-id alternatives.
An offerer that receives an answer where some of the rid-ids are
removed compared to the offer MAY release the corresponding resources
(codec, transport, etc) in its "receive" direction and MUST NOT send
any RTP packets corresponding to the removed rid-ids.
An offerer that offered some of its rid-ids as initially paused and
receives an answer that does not indicate RTP stream pause/resume
capability MUST NOT initially pause any simulcast streams.
An offerer with RTP stream pause/resume capability that receives an
answer where some rid-ids are marked as initially paused SHOULD
initially pause those RTP streams, even if they were marked as
initially paused also in the offer, unless it has good reason for
those RTP streams not being initially paused. One such reason could
be, for example, that the answerer would otherwise initially not
receive any media of that type at all.
5.3.4. Modifying the Session
Offers inside an existing session follow the same rules as for
initial SDP offer, with these additions:
1. rid-ids marked as initially paused in the offerer's "send"
direction SHALL reflect the offerer's opinion of the current
pause state at the time of creating the offer. This is purely
informational, and RTP stream pause/resume signaling [RFC7728] in
the ongoing session SHALL take precedence in case of any conflict
or ambiguity.
2. rid-ids marked as initially paused in the offerer's "receive"
direction SHALL (as in an initial offer) reflect the offerer's
desired rid-id pause state. Except for the case where the
offerer already paused the corresponding RTP stream through RTP
stream pause/resume [RFC7728] signaling, this is identical to the
conditions at an initial offer.
Creation of SDP answers and processing of SDP answers inside an
existing session follow the same rules as described above for initial
SDP offer/answer.
Session modification restrictions in Section 6.5 of "RTP Payload
Format Restrictions" [RFC8851] also apply.
5.4. Use with Declarative SDP
This document does not define the use of "a=simulcast" in declarative
SDP, partly because use of the simulcast format identification
[RFC8851] is not defined for use in declarative SDP. If concrete use
cases for simulcast in declarative SDP are identified in the future,
the authors of this memo expect that additional specifications will
address such use.
5.5. Relating Simulcast Streams
Simulcast RTP streams MUST be related on the RTP level through
RtpStreamId [RFC8852], as specified in the SDP "a=simulcast"
attribute (Section 5.2) parameters. This is sufficient as long as
there is only a single media source per SDP media description. When
using BUNDLE [RFC8843], where multiple SDP media descriptions jointly
specify a single RTP session, the SDES MID (Media Identification)
mechanism in BUNDLE allows relating RTP streams back to individual
media descriptions, after which the RtpStreamId relations described
above can be used. Use of the RTP header extension for the RTCP
source description items [RFC7941] for both MID and RtpStreamId
identifications can be important to ensure rapid initial reception,
required to correctly interpret and process the RTP streams.
Implementers of this specification MUST support the RTCP source
description (SDES) item method and SHOULD support RTP header
extension method to signal RtpStreamId on the RTP level.
NOTE: For the case where it is clear from SDP that the RTP PT
uniquely maps to a corresponding RtpStreamId, an RTP receiver can
use RTP PT to relate simulcast streams. This can sometimes enable
decoding even in advance of receiving RtpStreamId information in
RTCP SDES and/or RTP header extensions.
RTP streams MUST only use a single alternative rid-id at a time
(based on RTP timestamps) but MAY change format (and rid-id) on a
per-RTP packet basis. This corresponds to the existing
(nonsimulcast) SDP offer/answer case when multiple formats are
included on the "m=" line in the SDP answer, enabling per-RTP packet
change of RTP payload type.
5.6. Signaling Examples
These examples describe a client-to-video-conference service, using a
centralized media topology with an RTP mixer.
+---+ +-----------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| F |<---->| |<---->| J |
+---+ +-----------+ +---+
Figure 4: Four-Party Mixer-Based Conference
5.6.1. 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" "a=rid"
parameter is used to describe simulcast stream formats, effectively
achieving a 1:1 mapping between RtpStreamId and media formats (RTP
payload types). Alice's Offer:
v=0
o=alice 2362969037 2362969040 IN IP4 192.0.2.156
s=Simulcast-Enabled Client
c=IN IP4 192.0.2.156
t=0 0
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 send pt=97
a=rid:2 send pt=98
a=rid:3 recv pt=97
a=simulcast:send 1;2 recv 3
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
Figure 5: 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 "a=fmtp" and "a=imageattr" parameters
indicate that sent simulcast streams can differ in video resolution.
The RTP header extension for RtpStreamId is offered to avoid issues
with the initial binding between RTP streams (SSRCs) and the
RtpStreamId identifying the simulcast stream and its format.
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. Also, the
usage of the RtpStreamId RTP header extension is accepted.
v=0
o=server 823479283 1209384938 IN IP4 192.0.2.2
s=Answer to Simulcast-Enabled Client
c=IN IP4 192.0.2.43
t=0 0
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 recv pt=97
a=rid:2 recv pt=98
a=rid:3 send pt=97
a=simulcast:recv 1;2 send 3
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
Figure 6: Single-Source Simulcast Answer
Since the server is the simulcast media receiver, it reverses the
direction of the "simulcast" and "rid" attribute parameters.
5.6.2. Multisource 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 [RFC7741] and H264, are offered as alternatives for
the third simulcast stream for the first media source. Only the
highest-fidelity simulcast stream is 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
protected by RTP retransmission [RFC4588]. In addition, all but the
highest-fidelity simulcast stream are initially paused. Note that
the lower resolution is more prioritized than the medium-resolution
simulcast stream.
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 RtpStreamId as simulcast stream identification
enables use of a low number of RTP payload types. Note that when
using both BUNDLE [RFC8843] and "a=rid" [RFC8851], it is recommended
to use the RTP header extension for the RTCP source descriptions
items [RFC7941] for carrying these RTP stream-identification fields,
which is consequently also included in the SDP. Note also that for
"a=rid", the corresponding RtpStreamId SDES attribute RTP header
extension is named rtp-stream-id [RFC8852].
v=0
o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d
s=Offer from Simulcast-Enabled Multi-Source Client
c=IN IP6 2001:db8::c000:27d
t=0 0
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=216000; \
mst-mode=NI-TC
a=fmtp:101 profile-level-id=42c00d;max-fs=3600;max-mbps=108000
a=fmtp:103 max-fs=900; max-fr=30
a=rid:1 send pt=100;max-width=1280;max-height=720;max-fps=60;depend=2
a=rid:2 send pt=101;max-width=1280;max-height=720;max-fps=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:rtp-stream-id
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:1 send max-fs=921600;max-fps=30
a=rid:2 send max-fs=614400;max-fps=15
a=rid:3 send max-fs=230400;max-fps=30
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=rtcp-fb:* ccm pause nowait
a=simulcast:send 1;~3;~2
Figure 7: Fred's Multisource Simulcast Offer
5.6.3. Simulcast and Redundancy
The example in this section looks at applying simulcast with audio
and video redundancy formats. The audio media description uses codec
and bitrate restrictions, combined with the RTP payload for redundant
audio data [RFC2198] for enhanced packet-loss resilience. The video
media description applies both resolution and bitrate restrictions,
combined with Forward Error Correction (FEC) in the form of flexible
FEC [RFC8627] and RTP retransmission [RFC4588].
The audio source is offered to be sent as two simulcast streams. The
first simulcast stream is encoded with Opus, restricted to 64 kbps
(rid-id=1), and the second simulcast stream (rid-id=2) is encoded
with either G.711, or G.711 combined with linear predictive coding
(LPC) for redundancy and explicit comfort noise (CN). Both simulcast
streams include telephone-event capability. In this example, stand-
alone LPC is not offered as a possible payload type for the second
simulcast stream's RID, which could be motivated by, for example, not
providing sufficient quality.
The video source is offered to be sent as two simulcast streams, both
with two alternative simulcast formats. Redundancy and repair are
offered in the form of both flexible FEC and RTP retransmission. The
flexible FEC is not bound to any particular RTP streams and is
therefore able to be used across all RTP streams that are being sent
as part of this media description.
o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d
s=Offer from Simulcast-Enabled Client using Redundancy
c=IN IP6 2001:db8::c000:27d
t=0 0
a=group:BUNDLE foo bar
m=audio 49200 RTP/AVP 97 98 99 100 101 102
a=mid:foo
a=rtpmap:97 G711/8000
a=rtpmap:98 LPC/8000
a=rtpmap:99 OPUS/48000/1
a=rtpmap:100 RED/8000/1
a=rtpmap:101 CN/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:99 useinbandfec=1;usedtx=0
a=fmtp:100 97/98
a=fmtp:102 0-15
a=ptime:20
a=maxptime:40
a=rid:1 send pt=99,102;max-br=64000
a=rid:2 send pt=100,97,101,102
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=simulcast:send 1;2
m=video 49600 RTP/AVPF 103 104 105 106 107
a=mid:bar
a=rtpmap:103 H264/90000
a=rtpmap:104 VP8/90000
a=rtpmap:105 rtx/90000
a=rtpmap:106 rtx/90000
a=rtpmap:107 flexfec/90000
a=fmtp:103 profile-level-id=42c00d;max-fs=3600;max-mbps=108000
a=fmtp:104 max-fs=3600; max-fr=30
a=fmtp:105 apt=103;rtx-time=200
a=fmtp:106 apt=104;rtx-time=200
a=fmtp:107 repair-window=100000
a=rid:1 send pt=103;max-width=1280;max-height=720;max-fps=30
a=rid:2 send pt=104;max-width=1280;max-height=720;max-fps=30
a=rid:3 send pt=103;max-width=640;max-height=360;max-br=300000
a=rid:4 send pt=104;max-width=640;max-height=360;max-br=300000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=rtcp-fb:* ccm pause nowait
a=simulcast:send 1,2;3,4
Figure 8: Simulcast and Redundancy Example
6. RTP Aspects
This section discusses what the different entities in a simulcast
media path can expect to happen on the RTP level. This is explored
from source to sink by starting in an endpoint with a media source
that is simulcasted to an RTP middlebox. That RTP middlebox sends
media sources to other RTP middleboxes (cascaded middleboxes), as
well as selecting some simulcast format of the media source and
sending it to receiving endpoints. Different types of RTP
middleboxes and their usage of the different simulcast formats
results in several different behaviors.
6.1. Outgoing from Endpoint with Media Source
The most straightforward simulcast case is the RTP streams being
emitted from the endpoint that originates a media source. When
simulcast has been negotiated in the sending direction, the endpoint
can transmit up to the number of RTP streams needed for the
negotiated simulcast streams for that media source. Each RTP stream
(SSRC) is identified by associating it (Section 5.5) with an
RtpStreamId SDES item, transmitted in RTCP and possibly also as an
RTP header extension. In cases where multiple media sources have
been negotiated for the same RTP session and thus BUNDLE [RFC8843] is
used, the MID SDES item will also be sent, similarly to the
RtpStreamId.
Each RTP stream might not be continuously transmitted due to any of
the following reasons: temporarily paused using Pause/Resume
[RFC7728], sender-side application logic temporarily pausing it, or
lack of network resources to transmit this simulcast stream.
However, all simulcast streams that have been negotiated have active
and maintained SSRCs (at least in regular RTCP reports), even if no
RTP packets are currently transmitted. The relation between an RTP
stream (SSRC) and a particular simulcast stream is not expected to
change, except in exceptional situations such as SSRC collisions. At
SSRC changes, the usage of MID and RtpStreamId should enable the
receiver to correctly identify the RTP streams even after an SSRC
change.
6.2. RTP Middlebox to Receiver
RTP streams in a multiparty RTP session can be used in multiple
different ways when the session utilizes simulcast at least on the
media-source-to-middlebox legs. This is to a large degree due to the
different RTP middlebox behaviors, but also the needs of the
application. This text assumes that the RTP middlebox will select a
media source and choose which simulcast stream for that media source
to deliver to a specific receiver. In many cases, at most one
simulcast stream per media source will be forwarded to a particular
receiver at any instant in time, even if the selected simulcast
stream may vary. For cases where this does not hold due to
application needs, the RTP stream aspects will fall under the
middlebox-to-middlebox case (Section 6.3).
The selection of which simulcast streams to forward towards the
receiver is application specific. However, in conferencing
applications, active speaker selection is common. In case the number
of media sources possible to forward, N, is less than the total
number of media sources available in a multimedia session, the
current and previous speakers (up to N in total) are often the ones
forwarded. To avoid the need for media-specific processing to
determine the current speaker(s) in the RTP middlebox, the endpoint
providing a media source may include metadata, such as the RTP header
extension for client-to-mixer audio level indication [RFC6464].
The possibilities for stream switching are media type specific, but
for media types with significant interframe dependencies in the
encoding, like most video coding, the switching needs to be made at
suitable switching points in the media stream that breaks or
otherwise deals with the dependency structure. Even if switching
points can be included periodically, it is common to use mechanisms
like Full Intra Requests [RFC5104] to request switching points from
the endpoint performing the encoding of the media source.
Inclusion of the RtpStreamId SDES item for an SSRC in the middlebox-
to-receiver direction should only occur when use of RtpStreamId has
been negotiated in that direction. It is worth noting that one can
signal multiple RtpStreamIds when simulcast signaling indicates only
a single simulcast stream, allowing one to use all of the
RtpStreamIds as alternatives for that simulcast stream. One reason
for including the RtpStreamId in the middlebox-to-receiver direction
for an RTP stream is to let the receiver know which restrictions
apply to the currently delivered RTP stream. In case the RtpStreamId
is negotiated to be used, it is important to remember that the used
identifiers will be specific to each signaling session. Even if the
central entity can attempt to coordinate, it is likely that the
RtpStreamIds need to be translated to the leg-specific values. The
below cases will assume that RtpStreamId is not used in the mixer to
receiver direction.
6.2.1. Media-Switching Mixer
This section discusses the behavior in cases where the RTP middlebox
behaves like the media-switching mixer in RTP topologies
(Section 3.6.2 of [RFC7667]). The fundamental aspect here is that
the media sources delivered from the middlebox will be the mixer's
conceptual or functional ones. For example, one media source may be
the main speaker in high-resolution video, while a number of other
media sources are thumbnails of each participant.
The above results in the RTP stream produced by the mixer being one
that switches between a number of received incoming RTP streams for
different media sources and in different simulcast versions. The
mixer selects the media source to be sent as one of the RTP streams
and then selects among the available simulcast streams for the most
appropriate one. The selection criteria include available bandwidth
on the mixer-to-receiver path and restrictions based on the
functional usage of the RTP stream delivered to the receiver. As an
example of the latter, it is unnecessary to forward a full HD video
to a receiver if the display area is just a thumbnail. Thus,
restrictions may exist to not allow some simulcast streams to be
forwarded for some of the mixer's media sources.
This will result in a single RTP stream being used for each of the
RTP mixer's media sources. At any point in time, this RTP stream is
a selection of one particular RTP stream arriving to the mixer, where
the RTP header-field values are rewritten to provide a consistent,
single RTP stream. If the RTP mixer doesn't receive any incoming
stream matched to this media source, the SSRC will not transmit but
be kept alive using RTCP. The SSRC and thus RTP stream for the
mixer's media source is expected to be long-term stable. It will
only be changed by signaling or other disruptive events. Note that
although the above talks about a single RTP stream, there can in some
cases be multiple RTP streams carrying the selected simulcast stream
for the originating media source, including redundancy or other
auxiliary RTP streams.
The mixer may communicate the identity of the originating media
source to the receiver by including the Contributing Source (CSRC)
field with the originating media source's SSRC value. Note that due
to the possibility that the RTP mixer switches between simulcast
versions of the media source, the CSRC value may change, even if the
media source is kept the same.
It is important to note that any MID SDES item from the originating
media source needs to be removed and not be associated with the RTP
stream's SSRC. That is, there is nothing in the signaling between
the mixer and the receiver that is structured around the originating
media sources, only the mixer's media sources. If they were
associated with the SSRC, the receiver would likely believe that
there has been an SSRC collision and the RTP stream is spurious,
because it doesn't carry the identifiers used to relate it to the
correct context. However, this is not true for CSRC values, as long
as they are never used as an SSRC. In these cases, one could provide
CNAME and MID as SDES items. A receiver could use this to determine
which CSRC values that are associated with the same originating media
source.
If RtpStreamIds are used in the scenario described by this section,
it should be noted that the RtpStreamId on a particular SSRC will
change based on the actual simulcast stream selected for switching.
These RtpStreamId identifiers will be local to this leg's signaling
context. In addition, the defined RtpStreamIds and their parameters
need to cover all the media sources and simulcast streams received by
the RTP mixer that can be switched into this media source, sent by
the RTP mixer.
6.2.2. Selective Forwarding Middlebox
This section discusses the behavior in cases where the RTP middlebox
behaves like the Selective Forwarding Middlebox in RTP topologies
(Section 3.7 of [RFC7667]). Applications for this type of RTP
middlebox result in each originating media source having a
corresponding media source on the leg between the middlebox and the
receiver. A Selective Forwarding Middlebox (SFM) could go as far as
exposing all the simulcast streams for a media source; however, this
section will focus on having a single simulcast stream that can
contain any of the simulcast formats. This section will assume that
the SFM projection mechanism works on the media-source level and maps
one of the media source's simulcast streams onto one RTP stream from
the SFM to the receiver.
This usage will result in the individual RTP stream(s) for one media
source being able to switch between being active and paused, based on
the subset of media sources the SFM wants to provide the receiver for
the moment. With SFMs, there exist no reasons to use CSRC to
indicate the originating stream, as there is a one-to-one media-
source mapping. If the application requires knowing the simulcast
version received to function well, then RtpStreamId should be
negotiated on the SFM to receiver leg. Which simulcast stream that
is being forwarded is not made explicit unless RtpStreamId is used on
the leg.
Any MID SDES items being sent by the SFM to the receiver are only
those agreed between the SFM and the receiver, and no MID values from
the originating side of the SFM are to be forwarded.
An SFM could expose corresponding RTP streams for all the media
sources and their simulcast streams and then, for any media source
that is to be provided, forward one selected simulcast stream.
However, this is not recommended, as it would unnecessarily increase
the number of RTP streams and require the receiver to timely detect
switching between simulcast streams. The above usage requires the
same SFM functionality for switching, while avoiding the
uncertainties of timely detecting that an RTP stream ends. The
benefit would be that the received simulcast stream would be
implicitly provided by which RTP stream would be active for a media
source. However, using RtpStreamId to make this explicit also
exposes which alternative format is used. The conclusion is that
using one RTP stream per simulcast stream is unnecessary. The issue
with timely detecting end of streams, independent of whether they are
stopped temporarily or long term, is that there is no explicit
indication that the transmission has intentionally been stopped. The
RTCP-based pause and resume mechanism [RFC7728] includes a PAUSED
indication that provides the last RTP sequence number transmitted
prior to the pause. Due to usage, the timeliness of this solution
depends on when delivery using RTCP can occur in relation to the
transmission of the last RTP packet. If no explicit information is
provided at all, then detection based on nonincreasing RTCP SR field
values and timers need to be used to determine pause in RTP packet
delivery. As a result, when the last RTP packet arrives (if it
arrives), one usually cannot determine that this will be the last.
That it was the last is something that one learns later.
6.3. RTP Middlebox to RTP Middlebox
This relates to the transmission of simulcast streams between RTP
middleboxes or other usages where one wants to enable the delivery of
multiple simultaneous simulcast streams per media source, but the
transmitting entity is not the originating endpoint. For a
particular direction between middleboxes A and B, this looks very
similar to the originating-to-middlebox case on a media-source basis.
However, in this case, there are usually multiple media sources,
originating from multiple endpoints. This can create situations
where limitations in the number of simultaneously received media
streams can arise -- for example, due to limitation in network
bandwidth. In this case, a subset of not only the simulcast streams
but also media sources can be selected. As a result, individual RTP
streams can become paused at any point and later be resumed based on
various criteria.
The MIDs used between A and B are the ones agreed between these two
identities in signaling. The RtpStreamId values will also be
provided to ensure explicit information about which simulcast stream
they are. The RTP-stream-to-MID and -RtpStreamId associations should
here be long-term stable.
7. Network Aspects
Simulcast is in this memo defined as the act of sending multiple
alternative encoded streams of the same underlying media source.
Transmitting multiple independent streams that originate from the
same source 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 Using
the Multiplexing Features of RTP to Support Multiple Media Streams"
[RFC8872]. Discussion and clarification on how to handle multiple
streams in an RTP session can be found in [RFC8108].
The network aspects that are relevant for simulcast are:
Quality of Service (QoS): 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-
bitrate streams may be prioritized over higher-bitrate streams to
minimize congestion or packet losses in the low-bitrate streams.
Thus, there is a benefit to using a simulcast solution with good
QoS support.
NAT/FW Traversal (Network Address Translator / Firewall
Traversal): Using multiple RTP sessions incurs more cost for NAT/FW
traversal unless they can reuse the same transport flow, which can
be achieved by multiplexing negotiation using SDP port numbers
[RFC8843].
7.1. Bitrate Adaptation
Use of multiple simulcast streams can require a significant amount of
network resources. The aggregate bandwidth for all simulcast streams
for a media source (and thus SDP media description) is bounded by any
SDP "b=" line applicable to that media source. It is assumed that a
suitable congestion-control mechanism is used by the application to
ensure that it doesn't cause persistent congestion. 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. When
a simulcasting media source uses a single media transport for all of
the simulcast streams, it is likely that a joint congestion control
across all simulcast streams is used for that media source. 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 and ordering of
alternative simulcast formats (Section 5.2). 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.
8. Limitation
The chosen approach has a limitation that relates to the use of a
single RTP session for all simulcast formats of a media source, which
comes from sending all simulcast streams related to a media source
under the same SDP media description.
It is not possible to use different simulcast streams on different
media transports, which limits the possibilities for applying
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 also 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. Such simulcast usage in a multicast context is
out of scope for the current document and would require additional
specification.
9. IANA Considerations
This document registers a new media-level SDP attribute, "simulcast",
in the "att-field (media level only)" registry within the "Session
Description Protocol (SDP) Parameters" registry, according to the
procedures of [RFC4566] and [RFC8859].
Contact name, email: The IESG (iesg@ietf.org)
Attribute name: simulcast
Long-form attribute name: Simulcast stream description
Charset dependent: No
Attribute value: sc-value; see Section 5.1 of RFC 8853.
Purpose: Signals simulcast capability for a set of RTP streams
Mux category: NORMAL
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. However, there may 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.
Integrity protection and source authentication of all SDP signaling,
including simulcast attributes, can mitigate the risks of such
attacks that attempt to alter signaling.
Security considerations related to the use of "a=rid" and the
RtpStreamId SDES item are covered in [RFC8851] and [RFC8852]. There
are no additional security concerns related to their use in this
specification.
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[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, <https://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, <https://www.rfc-editor.org/info/rfc4566>.
[RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234,
DOI 10.17487/RFC5234, January 2008,
<https://www.rfc-editor.org/info/rfc5234>.
[RFC7405] Kyzivat, P., "Case-Sensitive String Support in ABNF",
RFC 7405, DOI 10.17487/RFC7405, December 2014,
<https://www.rfc-editor.org/info/rfc7405>.
[RFC7728] Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP
Stream Pause and Resume", RFC 7728, DOI 10.17487/RFC7728,
February 2016, <https://www.rfc-editor.org/info/rfc7728>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8843] Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", RFC 8843,
DOI 10.17487/RFC8843, January 2021,
<https://www.rfc-editor.org/info/rfc8843>.
[RFC8851] Roach, A.B., Ed., "RTP Payload Format Restrictions",
RFC 8851, DOI 10.17487/RFC8851, January 2021,
<https://www.rfc-editor.org/info/rfc8851>.
[RFC8852] Roach, A.B., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", RFC 8852,
DOI 10.17487/RFC8852, January 2021,
<https://www.rfc-editor.org/info/rfc8852>.
[RFC8859] Nandakumar, S., "A Framework for Session Description
Protocol (SDP) Attributes When Multiplexing", RFC 8859,
DOI 10.17487/RFC8859, January 2021,
<https://www.rfc-editor.org/info/rfc8859>.
11.2. Informative References
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J.C., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
DOI 10.17487/RFC2198, September 1997,
<https://www.rfc-editor.org/info/rfc2198>.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <https://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,
<https://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,
<https://www.rfc-editor.org/info/rfc4733>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <https://www.rfc-editor.org/info/rfc5109>.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, DOI 10.17487/RFC5583, July 2009,
<https://www.rfc-editor.org/info/rfc5583>.
[RFC6184] Wang, Y.-K., Even, R., Kristensen, T., and R. Jesup, "RTP
Payload Format for H.264 Video", RFC 6184,
DOI 10.17487/RFC6184, May 2011,
<https://www.rfc-editor.org/info/rfc6184>.
[RFC6190] Wenger, S., Wang, Y.-K., Schierl, T., and A.
Eleftheriadis, "RTP Payload Format for Scalable Video
Coding", RFC 6190, DOI 10.17487/RFC6190, May 2011,
<https://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,
<https://www.rfc-editor.org/info/rfc6236>.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011,
<https://www.rfc-editor.org/info/rfc6464>.
[RFC7104] Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol", RFC 7104,
DOI 10.17487/RFC7104, January 2014,
<https://www.rfc-editor.org/info/rfc7104>.
[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,
<https://www.rfc-editor.org/info/rfc7656>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC7741] Westin, P., Lundin, H., Glover, M., Uberti, J., and F.
Galligan, "RTP Payload Format for VP8 Video", RFC 7741,
DOI 10.17487/RFC7741, March 2016,
<https://www.rfc-editor.org/info/rfc7741>.
[RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
Header Extension for the RTP Control Protocol (RTCP)
Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
August 2016, <https://www.rfc-editor.org/info/rfc7941>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>.
[RFC8627] Zanaty, M., Singh, V., Begen, A., and G. Mandyam, "RTP
Payload Format for Flexible Forward Error Correction
(FEC)", RFC 8627, DOI 10.17487/RFC8627, July 2019,
<https://www.rfc-editor.org/info/rfc8627>.
[RFC8872] Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
and R. Even, "Guidelines for Using the Multiplexing
Features of RTP to Support Multiple Media Streams",
RFC 8872, DOI 10.17487/RFC8872, January 2021,
<https://www.rfc-editor.org/info/rfc8872>.
Appendix A. Requirements
The following requirements are met by the defined solution to support
the use cases (Section 3):
REQ-1: Identification:
REQ-1.1: It must be possible to identify a set of simulcasted RTP
streams as originating from the same media source in SDP
signaling.
REQ-1.2: An RTP endpoint must be capable of identifying the
simulcast stream that a received RTP stream is associated with,
knowing the content of the SDP signaling.
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 [RFC8843] SDP media descriptions.
REQ-3: Capability negotiation. The following must be possible:
REQ-3.1: The sender can express capability of sending simulcast.
REQ-3.2: The receiver can express capability of receiving
simulcast.
REQ-3.3: The sender can express the maximum number of simulcast
streams that can be provided.
REQ-3.4: The receiver can express the maximum number of simulcast
streams that can be received.
REQ-3.5: The sender can detail the characteristics of the
simulcast streams that can be provided.
REQ-3.6: The 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-5.4: A single simulcast stream can consist of multiple RTP
streams, to support codecs where a dependent stream is
dependent on a set of encoded and dependent streams, each
potentially carried in their own RTP stream.
REQ-6: Interoperability. The solution must be possible to use in:
REQ-6.1: Interworking with nonsimulcast legacy clients using a
single media source per media type.
REQ-6.2: WebRTC environment with a single media source per SDP
media description.
Acknowledgements
The authors would like to thank Bernard Aboba, Thomas Belling, Roni
Even, Adam Roach, Iñaki Baz Castillo, Paul Kyzivat, and Arun
Arunachalam for the feedback they provided during the development of
this document.
Contributors
Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have
contributed with important material to the first draft versions of
this document. Robert Hanton and Cullen Jennings from Cisco, Peter
Thatcher from Google, and Adam Roach from Mozilla contributed
significantly to subsequent versions.
Authors' Addresses
Bo Burman
Ericsson
Gronlandsgatan 31
SE-164 60 Stockholm
Sweden
Email: bo.burman@ericsson.com
Magnus Westerlund
Ericsson
Torshamnsgatan 23
SE-164 83 Stockholm
Sweden
Email: magnus.westerlund@ericsson.com
Suhas Nandakumar
Cisco
170 West Tasman Drive
San Jose, CA 95134
United States of America
Email: snandaku@cisco.com
Mo Zanaty
Cisco
170 West Tasman Drive
San Jose, CA 95134
United States of America
Email: mzanaty@cisco.com