Multi-Stream Media ConferencingEricssonFarogatan 6SE-164 80 KistaSweden+46 10 714 82 87magnus.westerlund@ericsson.comEricssonFarogatan 6SE-164 80 KistaSweden+46 10 714 13 11bo.burman@ericsson.comThis memo describes a multimedia multi-party conferencing
architecture based on use of multiple Real-Time Transport Protocol (RTP)
streams.Multimedia multi-party conferencing is being improved through the use
of multiple media streams per media type. At the same time, the number
of different types of end-points that are capable of participating in a
multimedia conference increases, and so does the number of access types
that end-points may use to connect. This memo describes a number of use
cases relevant to that scenario. The use cases aims to use as high media
quality as possible, while making as efficient use of available
resources as possible and accommodating a high degree of end-point and
network diversity.The use cases in this memo should handle diversity in end-point
capability such as media quality, processing power, and network
interface. The perceived end-user media quality is in turn impacted by
media stream bitrate and also number of streams per media type, as well
as end-to-end delay, which should also be taken into account. Diversity
in network capacity, network quality, and user preferences based on
location, device, etc, should also be handled. These properties should
be expected to change between conference sessions and even within the
same session.Different types of conference Topologies must be supported, including
centralized, multicast, and point-to-point. It should also be possible
to use cascaded, centralized conferences as well as combinations of
unicast and multicast. The relevant RTP base topologies are described in
.Use cases including an RTP Mixer should avoid adding delay or
reducing quality by forwarding streams as unmodified as possible, with
reduced processing requirements in the RTP Mixer as added advantage.The conference use cases should strive for continuous presence,
presenting media from as many participants as reasonably possible. At
the same time, it should strive to use as high quality media as possible
from each participant. The bandwidth used for the conference should at
the same time be as low as possible. Those three requirements are in
conflict and there is no generally accepted, optimum trade-off.The number of participants in the conference shall assumed to be
unlimited, and it may thus not be possible to create a continuous
presence experience with media from all participants being presented to
all other participants, and only media from a limited sub-set of
participants may be presented to each individual participant. This is
seen as the hardest conferencing use case and most other use cases
should also be covered by that use case. How many other participants'
media that are presented simultaneously on a certain end-point is
allowed to vary and may depend on a number of preconditions.The sub-sections below describe multi-stream conferencing use cases
relevant to each RTP base topology. Each use case is constructed as to
cover as many requirements as possible and must consequently include a
number of different end-point types.In the figures below, the involved end-points are for clarity drawn
as only sending or only receiving, but may in a real scenario of course
have the possibility to both send and receive.The common figure legends for the end-point capability categories
used in all use cases are:Dual channel, high quality, for example conference
room client. Streams in figures are denoted by '||' or '='. The term
'channel' is chosen as a generic term and may apply to any media. A
channel is related to a single media source in the sender and a
media sink in the receiver. For video media, the channel source is
typically a camera and the sink a screen or window. For audio, the
channel source is typically a microphone and the sink a loudspeaker.
A dual channel sending end-point thus has two independent media
sources that can be sent simultaneously to a receiving end-point.
Nothing prevents those dual channels from using other qualities than
'high', but that is chosen as an example in this memo to simplify
the description.High quality, single channel, for example meeting
room client. Streams in figures are denoted by '|' or '-'.Medium quality, single channel, for example desktop
client. Streams in figures are denoted by ':' or '..'.Low quality, single channel, for example mobile
client. Streams in figures are denoted by '''.It is of course possible to divide end-points into more
categories, but the chosen ones should make it possible to highlight
most of the relevant topics.Note also that the use cases are kept as separated and clean-cut as
possible to simplify the description. Most use cases, especially the
RTP Mixer ones, could be combined into
larger, more complex scenarios.The point to point case is close to trivial, since it is assumed
that capability exchange during setup will be able to negotiate the
best quality between the two end-points. When the number of channels
differ between sender and receiver, the channel aspects of apply.Each link between the RTP Mixer and an end-point is in principle a
Point to Point connection as described
above. One major difference from a Point to Point Connection is that
the RTP Mixer should represent and act according to some combination
of the wishes and needs of multiple end-points on the other side of
the Mixer. That may include handling of conflicting or partly
conflicting requirements, and the way to resolve those is not
generally defined but will typically depend on RTP Mixer design,
configuration, applied policies, or some combination.This use case is the only feasible one when end-points have
incompatible codecs, and transcoding is in that case always
necessary and cannot be avoided. The incompatibility is not
necessarily only due to different codec types, but may also be
caused by limited codec capacity, or limitations in media stream
transport between the end-points. This use case is trivial in the
sense that the Mixer is assumed to always be capable of creating a
dedicated media mix to each receiving end-point. It may be used as
an overall strategy, or as part of other use cases.A special case of transcoding is when the sender is configured to
use scalable encoding and receivers do not support scalability.
Transcoding of scalable streams to non-scalable streams is often a
less complex operation than transcoding in general.A low quality end-point has per definition the lowest media
quality in the conference, and as a sender it is assumed that all
other end-points can receive and present the media without
restrictions. Some of the more capable end-points will have to
choose how to present the received media in the best way, but it can
always be presented.As a receiver, a low quality end-point is only capable of
receiving streams from other low quality end-points (without
transcoding).It is conceivable that a receiver of a certain quality category
(not only low quality) can receive higher quality streams and reduce
the quality locally such that it is feasible for presentation, but
that will in general waste both bandwidth and processing
resources.Similar to above, the medium quality sender media stream is
assumed to be possible to receive without restrictions in all but
the low quality end-point. For the medium quality media stream to
reach also the low quality end-point, there are three options that
are described in the sub-sections below.When receiving, a medium quality end-point is capable of
receiving other medium quality streams, as well as low quality
streams (without transcoding).Transcode the stream when sent towards low quality receivers,
as described above.
This could sometimes be feasible quality-wise, especially if the
quality difference between the medium quality and the low quality
streams are large, making the reduced medium quality stream be
relatively close quality-wise to un-encoded low quality media.
End-to-end delay will however always suffer.Encode both a medium quality and a low quality stream from the
same un-encoded source data and simulcast them. The RTP Mixer can,
without having to transcode, forward the low quality stream
towards the low quality end-points, and forward the medium quality
stream towards all other end-points.As a variant of simulcast, if it is possible to use a scalable
codec, create a scalable stream with one low quality sub-stream
and one sub-stream that together with the low quality sub-stream
can reconstruct a medium quality stream. Similar but not identical
to the above, the RTP Mixer can, without having to transcode,
forward the low quality sub-stream towards the low quality
end-points, and forward both the low quality and the medium
quality sub-streams (jointly describing a medium quality stream)
to all other end-points.This use case is very similar to the medium quality end-point case above.
The difference is that there are now two different (sample)
categories of end-points that cannot receive the high quality stream
instead of one category. Simulcast and scalable streams must thus be
extended to three versions or three sub-streams, respectively.As a receiver, all streams from other end-points can be received.
The only exception is when multiple streams from a single end-point
are used, such as from D.Similar to Medium
Quality, just that the transcoding needs to produce two
different qualities instead of one.Similar to Medium
Quality, just that three instead of two simulcast streams
need to be sent.Similar to Medium
Quality, just that three instead of two scalable layers are
used.Again, this use case is very similar to the one above. The major difference is
that this end-point is capable of sending and receiving dual, high
quality streams where each stream has to be treated in a similar way
to the previous section.When using multiple inter-related media, such as video with
corresponding audio, those media streams need not only be
synchronized time-wise, just as for single channel end-points, but
their spatial relation need also be established. For example, a left
camera with an attached microphone and a right camera with an
attached microphone. In general it is likely always desirable to be
able to relate streams from a multi-channel end-point in a defined
way, representing related sub-parts of a larger scene, both
intra-media and inter-media. Description and signaling of stream
relations is a complex problem in itself, which is currently work in progress in CLUE WG
and will not be elaborated further in this memo.Another major, additional, aspect to account for is that the RTP
Mixer needs to choose how to map dual (or multiple) streams onto a
single stream, when forwarding towards end-points that has fewer
receive channels than the sender. This problem is similar to
choosing a limited set of participants from a potentially unlimited
set, which is described below.A dual-channel (or multi-channel) receiving end-point that is
receiving fewer simultaneous channel streams from a sending
end-point than the maximum possible this end-point can handle, will
have to decide which one(s) of the available receive channels should
be used for each received stream. This decision can also be made by
the RTP Mixer, if it knows the concept of multi-channel clients, has
information about how many simultaneous channels the individual
receiver supports, and knows how those channels should be
related.There may of course be end-points that have capability for more
than two simultaneous channels. It is also possible to envision
end-points where the number of receive channels differ from the
number of send channels.When a Mixer cannot forward all available streams to each client,
it has to choose a small set out of a potentially very large set of
streams in the conference. Multiple strategies are possible for that
choice. The Mixer may use different strategies towards different
receivers, depending on for example their capabilities or
preferences.Another dimension of selection exist when the conference contains
end-points with different number of channels (multiple media streams
of the same media type). On the Mixer receive side, it may be
necessary to select a few streams from a multi-stream media. On the
Mixer send side, it may be necessary to select to which channel in a
multi-stream capable receiver a certain stream should be sent.The choice of source may be either algorithmic (pre-configured)
or manual (user controlled from one or more end-points). Speech
activity is a commonly used algorithmic measure to choose which
participants' media streams to forward, but it is not the only
conceivable measure. Which and how many streams to select can either
be based on some algorithm included in the RTP Mixer (for example
the N most active speakers), or it can be controlled by the
conference owner, or even by the receiving users individually.The figure below depicts the case where a receiving end-point
explicitly selects streams through signaling (*) to the Mixer. Both
the media senders and their streams are numbered for clarity, and
the conceptual signaling message is contained in {...}.To be able to make an informed choice on what streams to select,
the user at the receiving end-point will need information about
which conference participant correspond to which stream, and
possibly also other meta-information about the streams and sending
end-points.When it is desirable that the RTP Mixer selects and forwards a larger number
of simultaneous streams than what the receiving end-point can
support, the Mixer has the option to make a composition of multiple
streams onto fewer streams, possibly to only a single stream. Which
and how many streams to compose is typically based on the selection,
as described in the previous section above.The composition operation is basically independent from
selection. In general, Mixer composition requires transcoding. A
Mixer composition use case example is depicted below. To simplify
the picture, only a single receiver is included. Also, both the
media senders and their streams are numbered for clarity. In the
below example, the Mixer has chosen to compose stream 1, 3 and 4
into the single stream sent to the receiving end-point.An alternative to make composition in the Mixer is to let the
end-point do local composition by sending it multiple, un-composed,
streams. This could avoid transcoding at the cost of sending
multiple streams, which is depicted below, where stream 2, 3 and 5
are sent to the receiving client for local composition, as an
example.When the senders offer streams of multiple qualities, either the
mixer or the local composition can select and combine media of
different qualities. Use of multiple qualities could help optimizing
resource utilization for transport, decoding and rendering. In the
figure below, one high quality (3), one medium quality (4) and two
low qualities (1 and 2) are selected for local composition, as an
example.This local composition scenario can be further enhanced by the
Mixer providing different quality streams to the receiver, based on
the Mixer selection algorithm. One example could be to let the Mixer
forward the stream from the most active speaker as a high quality
stream, and forward the less active speakers as lower quality
streams. To use this in the local composition, the receiving
end-point must know the streams' different roles, which requires a
stream role agreement between the Mixer and the receiving end-point.
In the figure above, this can be achieved by tagging for example the
leftmost stream from the mixer as having the "most active" role. The
role agreement can be made through signaling between Mixer and
receiving end-point.When the receiving end-point supports reception and presentation
of several channels (for example has several screens), it is
possible to combine Mixer composition with local composition of
multiple un-composed streams by sending one or more composed streams
and one or more un-composed streams from the Mixer.In the multicast or multi-unicast case, each media stream from a
single sender will reach multiple receivers unmodified. This can be
achieved by multicast addressing, or by multi-unicast and RTP Translators.This use case is similar to when an RTP Mixer is neither performing
composition nor source selection, but is forwarding all
streams (and qualities, if more than one) to all receivers. The entire
load of stream and quality selection for presentation is put on the
receiving end-point.In the figure below, multi-unicast through the use of an RTP
Translator is depicted since the figure becomes clearer than with a
full mesh multicast. The figure illustrates non-scalable streams, but
it is of course also possible to multicast scalable streams.This section discusses how RTP
transport could be used with the scenarios discussed in the
previous section. It complements, extends and partially presents an
alternative solution to what is described in .It is assumed that each RTP media stream in the use cases in the
previous section is identified by an SSRC. There already exist methods
to convey information about each media stream
and sending end-point in a conference. Those methods also
provide means to correlate that information with the stream SSRC. This
information could be sufficient for a receiving end-point to make
informed media stream selection
decisions.When the RTP Mixer associates a role
to a stream, for example "most active speaker" or "leftmost
channel", it is possible to associate that additional property to an
SSRC belonging to the Mixer, while also keeping the original SSRC in
the RTP packet as CSRC. This way, it is possible to apply special
treatment to received streams based on their SSRC without losing the
ability to identify the original source, using existing RTP
functionality.It is also possible in a receiving end-point to let each role (and
Mixer SSRC) map towards a specific media decoder, since that Mixer
SSRC would rarely (if ever) change during a session other than due to
SSRC conflicts, while the CSRC would typically change every time a new
stream is selected for that specific role, for example "active
speaker".Note that when simulcast is used, different simulcast versions can
typically use different SSRC. When scalable coding is used, different
layers can sometimes be sent within a single SSRC using a single
Payload Type and thus cannot be distinguished on RTP level.
Identification of different layers will then have to be codec
specific. Some scalable codecs can also send different layers on
separate SSRC or using separate Payload Types.End-points and Mixers supporting
multiple channels need to know how many simultaneous channels
that can be accepted in a receiver and will be used from a sender.
Assuming that each channel is sent as a single SSRC, there should be
signaling that limits
the number of SSRC in an RTP session. This maps well with the
above suggested relation between SSRC and media decoders, since the
suggested limitation then also expresses the maximum simultaneously
available decoding resources.When representing a specific media source in several different
qualities and when using simulcast to
transport them rather than as scalable layers contained in a
single stream, those separate streams need to be signaled as simulcast
versions, in order for the receiver to be able to apply correct selection logic.When a conference is configured to let individual users at receiving end-points
choose which streams to receive, responsive selection signaling
between end-point and RTP Mixer is needed to initiate the
stream selection. This selection is applicable to the media streams
included in a compositions also.Note that when simulcast is used, different simulcast versions can
typically use different SSRC. When scalable coding is used, different
layers can sometimes be sent within a single SSRC using a single
Payload Type and thus cannot use the parts of SDP signaling that
relies on those identifiers. Identification of different layers will
then have to be codec specific. Some scalable codecs can also send
different layers on separate SSRC or using separate Payload Types.When it is desirable to minimize the number of UDP ports used by an
end-point, for example to reduce the resources for NAT and firewall
traversal, it should be possible to send all media
streams from all RTP sessions on a single UDP port. This should
preferably be done without losing any important RTP functionality.
Transport resource priority and Quality of Service handling are
typically performed based on 5-tuple (source and destination addresses
and ports, and protocol), which together with desired differentiation
of media stream priority can require use of more than one UDP port
(5-tuple).In a conference use case with multiple sending end-points and where
the receiving end-points do not make use of all available streams,
there is a risk that some of the sent streams are not used by any
receiver. The probability for this increases when end-points provide
multiple streams of different qualities. The need for a certain stream
can change very quickly, for example when the need is based on
conditions of other streams such as speech activity. To save bandwidth
and processing resources in the sending end-point, it would thus be
desirable for an RTP Mixer to be able to quickly turn off or pause individual
streams that are no longer used in any media mix sent to
receiving end-points, and even more importantly be able to quickly
resume needed streams when they are needed again.In use cases where multiple media
streams are used in a single RTP session, when SDP is used as
signaling protocol, and specifically when the number of streams depends on the SDP
negotiation outcome, the currently defined bandwidth signaling
attribute is only capable of describing the maximum possible bandwidth
usage for the most demanding alternative. It would be desirable to
express
bandwidth requirements in a more precise way.While any RTP stream relations such as for example spatial
co-location of related audio and video streams should be possible to
express in session signaling or other application signaling protocol,
there may be times when it is desirable that RTP stream SSRC relations
such as simulcast alternatives or related FEC streams can be seen
directly in the RTP or RTCP streams. This would allow for processing
of media and related streams in middle boxes, without the need to have
access to all higher layer signaling. Keeping protocol layer
separation will enable some architectural freedom and may ease future
extensions.Any security considerations relevant to this memo are described in
the RFCs and drafts referenced in the RTP
Usage section.This document makes no request of IANA.Note to RFC Editor: this section may be removed on publication as an
RFC.