Using Simulcast in SDP and RTP SessionsEricssonGronlandsgatan 31SE-164 60 StockholmSwedenbo.burman@ericsson.comEricssonTorshamnsgatan 23SE-164 83 StockholmSweden+46 10 714 82 87magnus.westerlund@ericsson.comCisco170 West Tasman DriveSan JoseCA95134USAsnandaku@cisco.comCisco170 West Tasman DriveSan JoseCA95134USAmzanaty@cisco.comIn 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.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 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 multi-party 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, as part of a
distributed multi-party scenario, be used point-to-point between
middleboxes. 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.This document makes use of the terminology defined in RTP Taxonomy, and RTP
Topologies. The following terms are especially noted or here
defined:An RTP middle node, defined in (Section 3.6 to 3.9).An association among a group of
participants communicating with RTP, as defined in and amended by .A stream of RTP packets containing media
data, as defined in .A common short term for the terms
"switching RTP mixer", "source projecting middlebox", and "video
switching MCU" as discussed in .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 . For example, HD and thumbnail
video simulcast versions of a single media source sent
concurrently as separate RTP Streams.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 in offer/answer, 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.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 when, and only
when, they appear in all capitals, as shown here.The use cases of simulcast 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 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
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, typically degrades media Quality of Experience (QoE)
such as end-to-end delay for the receiving participants, and
requires RTP Mixer access to media content.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, has
very limited impact on media QoE, and does not require RTP Mixer
(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 to 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.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: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", like if they
differ in video codec profile, or the transport packetization
configuration.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.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
tradeoff 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.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 )
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.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 .
Sending a simulcast of RTP streams is one way of accommodating
receivers with conflicting or otherwise incompatible preferences.This memo defines SDP signaling that
covers the above described simulcast use cases and functionalities. A
number of requirements for such signaling are elaborated in .The RID mechanism, as defined in , 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 framerate, 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 .If the above 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-id), 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 above "send" part 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 above line 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
below:The above SDP media description 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 answering
"a=simulcast" attribute, corresponding to the above offer, is: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: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
and will work in a WebRTC context, both with
and without BUNDLE grouping
of media descriptions.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 semicolon. The
identifiers describing simulcast streams on the "a=simulcast" line are
rid-id, as defined by "a=rid" lines in . Each simulcast stream can be offered as
a list of alternative rid-id, with each alternative separated by comma
(not in the examples above). A detailed specification can be found in
and more detailed examples are outlined in
.This section further details the overview above. First, formal syntax is provided, followed by the rest of the SDP
attribute definition in . Relating Simulcast Streams provides the
definition of the RTP/RTCP mechanisms used. The section is concluded
with a number of examples.This document defines a new SDP media-level "a=simulcast"
attribute, with value according to the following ABNF syntax and its update for Case-Sensitive String Support in ABNF:Note to RFC Editor: Replace "I-D.ietf-mmusic-rid" in the above
figure with RFC number of draft-ietf-mmusic-rid before publication
of this document.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.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 a comma (","). Each rid-id can also be specified as
initially paused, 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 an rid-id. Since
pause capability specified via the "a=rtcp-fb" attribute applies only
to specified payload types and 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.Simulcast capability is expressed through a new media level SDP attribute, "a=simulcast". 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
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-id MUST NOT be used as valid
simulcast streams by an RTP stream receiver. The direction for an
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-id in the "recv"
direction expresses 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.rid-id that have explicit dependencies to other rid-id (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-id. 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 to align the choice between alternative rid-ids
between different simulcast streams.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
and/or DTMF formats.If RTP stream pause/resume is
supported, any rid-id 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 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 , e.g. by
using the "*" wildcard format for "a=rtcp-fb".An initially paused simulcast stream in "send" direction for the
endpoint sending the SDP 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 rid-id value in an
RtpStreamId RTCP SDES
item.If the endpoint sending the SDP includes an "recv" direction
simulcast stream that is initially paused, then the remote RTP sender
receiving the SDP SHOULD put its RTP stream in a 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 signalling from RTP receivers and cannot make this
determination. 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 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 ) since the RTP receiver's SSRC
in send direction is sometimes not yet known.Use of the redundant audio data
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, 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 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-id that are
effectively equivalent.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".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-id 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 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. Since SDP session level simulcast is
undefined in this memo, an answerer that receives an offer with the
"a=simulcast" attribute on 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 that 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-id MAY keep all the
alternative rid-id in the answer, but it MAY also choose to remove
any non-desirable alternative rid-id in the answer. The answerer
MUST NOT add any alternative rid-id in 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 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 pause/resume capable answerer that receives an
offer with RTP stream pause/resume capability MAY mark any rid-id
that refer to pause/resume capable formats as initially paused in
the answer.An answerer that receives indication in an offer of an 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,
for example, be that all receive direction simulcast streams for a
media source the answerer accepts in the answer would otherwise be
paused.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-id 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-id.An offerer that offered some of its rid-id as initially paused
and that 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-id are marked as initially paused, SHOULD
initially pause those RTP streams regardless 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,
for example, be that the answerer would otherwise initially not
receive any media of that type at all.Offers inside an existing session follow the same rules as for
initial SDP offer, with these additions:rid-id 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 in the ongoing session SHALL take
precedence in case of any conflict or ambiguity.rid-id 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 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
also apply.This document does not define the use of "a=simulcast" in
declarative SDP, partly motivated by use of the simulcast format identification
not being 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.Simulcast RTP streams MUST be related on RTP level through RtpStreamId, as specified in the
SDP "a=simulcast" attribute parameters.
This is sufficient as long as there is only a single media source per
SDP media description. When using BUNDLE, 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 RtpStreamId relations can be used. Use of the RTP header extension 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 RTP level.For the case where it is clear from SDP that
RTP PT uniquely maps to corresponding RtpStreamId, an RTP receiver
can use RTP PT to relate simulcast streams. This can sometimes
enable decoding even in advance to 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 (non-simulcast)
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.These examples describe a client to video conference service, using
a centralized media topology with an RTP mixer.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". In this example, only the
"pt" "a=rid" parameter is used, effectively achieving a 1:1 mapping
between RtpStreamId and media formats (RTP payload types), to
describe simulcast stream formats. Alice's 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
indicates 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.Since the server is the simulcast media receiver, it reverses the
direction of the "simulcast" and "rid" attribute parameters.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 and H264. These two simulcast streams also have
a temporal dependency. Two different video codecs, VP8 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. Also
here, 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 the use
of both BUNDLE and
"a=rid" recommends using
the RTP header extension 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.The example in this section looks at applying simulcast with
audio and video redundancy formats. The audio media description uses
codec and bitrate restrictions, combining it with RTP Payload for Redundant Audio Data for
enhanced packet loss resilience. The video media description applies
both resolution and bitrate restrictions, combining it with FEC in
the form of Flexible FEC
and RTP Retransmission.The audio source is offered to be sent as two simulcast streams.
The first simulcast stream is encoded with Opus, restricted to 50
kbps (rid-id=5), and the second simulcast stream is encoded either
with G.711 (rid-id=7) or with G.711 combined with LPC for redundancy
(rid-id=6). In this example, stand-alone LPC is not offered as an
possible payload type for the second simulcast stream's RID, which
could e.g. be motivated by 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 possible to use across all RTP streams that are being sent
as part of this media description.This section discusses what the different entities in a simulcast
media path can expect to happen on 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
both 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.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 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 is used,
also the MID SDES item will 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, 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 SSRC (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.RTP streams in a multi-party 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,
then the RTP stream aspects will fall under the middlebox to middlebox
case .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 amount
of media sources available in an multi-media 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 meta data, such as the RTP Header
Extension for Client-to-Mixer Audio Level Indication.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 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 signalling 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 signalling 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 have as
base line that RtpStreamId is not used in the mixer to receiver
direction.This section discusses the behavior in cases where the RTP
middlebox behaves like the Media-Switching Mixer (Section 3.6.2) in
RTP Topologies. 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 that the RTP stream produced by the mixer is
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. This RTP stream is at any point in
time 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 signalling 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 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
signalling between the mixer and the receiver that is structured
around the originating media sources, only the mixer's media
sources. If they would be associated with the SSRC, the receiver
would likely believe that there has been an SSRC collision, and that
the RTP stream is spurious as 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 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
signalling 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.This section discusses the behavior in cases where the RTP
middlebox behaves like the Selective Forwarding Middlebox (Section
3.7) in RTP Topologies. Applications
for this type of RTP middlebox results in that each originating
media source will have 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 an 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 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 that the individual RTP stream(s) for
one media source can switch between being active to 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.A 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 a 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 if 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
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 non
increasing RTCP SR field values and timers need to be used to
determine pause in RTP packet delivery. This results in that one can
usually not determine when the last RTP packet arrives (if it
arrives) that this will be the last. That it was the last is
something that one learns later.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 middlebox A and B, this looks very similar to the
originating to middlebox case on a media source basis. However, in
this case there is 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. This results in that individual RTP streams can be become
paused at any point and later being resumed based on various
criteria.The MIDs used between A and B are the ones agreed between these two
identities in signalling. 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.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. Discussion and clarification on how to
handle multiple streams in an RTP session can be found in .The network aspects that are relevant for simulcast are: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 use a simulcast solution with good QoS support.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.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 . 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.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, 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 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 multicast context is out of scope for
the current document and would require additional specification.This document requests to register a new media-level SDP attribute,
"simulcast", in the "att-field (media level only)" registry within the
SDP parameters registry, according to the procedures of and .The IESG (iesg@ietf.org)simulcastSimulcast stream
descriptionNosc-value; see of RFC XXXX.Signals simulcast capability for a set of RTP
streamsNORMALNote to RFC Editor: Please replace "RFC XXXX" with the assigned
number of this RFC.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.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 "a=rid" and the
RtpStreamId SDES item is covered in
and . There are no additional
security concerns related to their use in this specification.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.The authors would like to thank Bernard Aboba, Thomas Belling, Roni
Even, Adam Roach, Inaki Baz Castillo, Paul Kyzivat, and Arun Arunachalam
for the feedback they provided during the development of this
document.The following requirements are met by the defined solution to support
the use cases:Identification:It must be possible to
identify a set of simulcasted RTP streams as originating from
the same media source in SDP signaling.An RTP endpoint must be
capable of identifying the simulcast stream a received RTP
stream is associated with, knowing the content of the SDP
signalling.Transport usage. The solution
must work when using:Legacy SDP with separate
media transports per SDP media description.Bundled
SDP media descriptions.Capability negotiation. It must
be possible that:Sender can express
capability of sending simulcast.Receiver can express
capability of receiving simulcast.Sender can express
maximum number of simulcast streams that can be provided.Receiver can express
maximum number of simulcast streams that can be received.Sender can detail the
characteristics of the simulcast streams that can be
provided.Receiver can detail the
characteristics of the simulcast streams that it prefers to
receive.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.Compatibility. It must be
possible to use simulcast in combination with other RTP mechanisms
that generate additional RTP streams:RTP Retransmission.RTP Forward Error Correction.Related payload types
such as audio Comfort Noise and/or DTMF.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.Interoperability. The solution
must be possible to use in:Interworking with
non-simulcast legacy clients using a single media source per
media type.WebRTC environment with
a single media source per SDP media description.NOTE TO RFC EDITOR: Please remove this section prior to
publication.c= and t= line order corrected in SDP examplesExamples corrected to follow RID ABNFExample now comments on priority
for second media source.Clarified a SHOULD limitation.Added urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id in
examples with RTX.ABNF now uses RFC 7405 to indicate case sensitivityVarious minor editorials and nits.Modified Normative statement regarding RTP stream duplication
in Section 5.2.Clarified assumption about use of congestion control by
applications.Changed to use RFC 8174 boilerplate instead of RFC 2119.Clarified explanation of syntax for simulcast attribute in
Section 4.Editorial clarification in Section 5.2 and 5.3.2.Various minor editorials and nits.Added new SDP example section on Simulcast and Redundancy,
including both RED (RFC2198), RTP RTX (RFC4588), and FEC
(draft-ietf-payload-flexible-fec-scheme).Removed restriction that "related" payload formats in an RTP
stream (such as CN and DTMF) must not have their own rid-id, since
there is no reason to forbid this and corresponding clarification
is made in draft-ietf-mmusic-rid.Removed any mention of source-specific signaling and the
reference to RFC5576, since draft-ietf-mmusic-rid is not defined
for source-specific signaling.Changed some SDP examples to use a=rid restrictions instead of
a=imageattr.Changed reference from the obsoleted RFC 5285 to RFC 8285.Amended overview section with a bit more explanation on the
examples, and added an rid-id alternative for one of the
streams.Removed SCID also from the Terminology section, which was
forgotten in -09 when changing SCID to rid-id.Changed SCID to rid-id, to align with ietf-draft-mmusic-rid
naming.Changed Overview to be based on examples and shortened it.Changed semantics of initially paused rid-id in modified SDP
offers from requiring it to follow actual RFC 7728 pause state to
an informational offerer's opinion at the time of offer creation,
not in any way overriding or amending RFC 7728 signaling.Replaced text on ignoring all but the first of multiple
"a=simulcast" lines in a media description with mandating that at
most one "a=simulcast" line is included.Clarified with a note that, for the case it is clear from the
SDP that RTP PT uniquely maps to RtpStreamId, an RTP receiver can
use RTP PT to relate simulcast streams.Moved Section 4 Requirements to become Appendix A.Editorial corrections and clarifications.Correcting syntax of SDP examples in section 6.6.1, as found by
Inaki Baz Castillo.Changing ABNF to only define the sc-value, not the SDP
attribute itself, as suggested by Paul Kyzivat.Changing I-D reference to newly published RFC 8108.Adding list of modifications between -06 and -07.A scope clarification, as result of the discussion with Roni
Even.A reformulation of the identification requirements for
simulcast stream.Correcting the statement related to source specific signalling
(RFC 5576) to address Roni Even's comment.Update of the last paragraph in Section 6.2 regarding simulcast
stream differences as well as forbidding multiple instances of the
same SCID within a single a=simulcast line.Removal of note in Section 6.4 as result of issue raised by
Roni Even.Use of "m=" has been changed to media description and a few
other editorial improvements and clarifications.Added section on RTP AspectsAdded a requirement (5-4) on that capability exchange must be
capable of handling multi RTP stream cases.Added extmap attribute also on first signalling example as it
is a recommended to use mechanism.Clarified the definition of the simulcast attribute and how
simulcast streams relates to simulcast formats and SCIDs.Updated References list and moved around some references
between informative and normative categories.Editorial improvements and corrections.Aligned with recent changes in draft-ietf-mmusic-rid and
draft-ietf-avtext-rid.Modified the SDP offer/answer section to follow the generally
accepted structure, also adding a brief text on modifying the
session that is aligned with draft-ietf-mmusic-rid.Improved text around simulcast stream identification (as
opposed to the simulcast stream itself) to consistently use the
acronym SCID and defined that in the Terminology section.Changed references for RTP-level pause/resume and VP8 payload
format that are now published as RFC.Improved IANA registration text.Removed unused reference to
draft-ietf-payload-flexible-fec-scheme.Editorial improvements and corrections.Changed to only use RID identification, as was consensus during
IETF 94.ABNF improvements.Clarified offer-answer rules for initially paused streams.Changed references for RTP topologies and RTP taxonomy
documents that are now published as RFC.Added reference to the new RID draft in AVTEXT.Re-structured section 6 to provide an easy reference by the
updated IANA section.Added a sub-section 7.1 with a discussion of bitrate
adaptation.Editorial improvements.Removed text on multicast / broadcast from use cases, since it
is not supported by the solution.Removed explicit references to unified plan draft.Added possibility to initiate simulcast streams in paused
mode.Enabled an offerer to offer multiple stream identification (pt
or rid) methods and have the answerer choose which to use.Added a preference indication also in send direction
offers.Added a section on limitations of the current proposal,
including identification method specific limitations.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.Renamed simulcast version and simulcast version alternative to
simulcast stream and simulcast format respectively, and improved
definitions for them.Clarification that it is possible to switch between simulcast
version alternatives, but that only a single one be used at any
point in time.Changed the definition so that ordering of simulcast formats
for a specific simulcast stream do have a preference order.No changes. Only preventing expiry.Added this appendix.