Internet DRAFT - draft-burman-rtcweb-mmusic-media-structure
draft-burman-rtcweb-mmusic-media-structure
Network Working Group B. Burman
Internet-Draft M. Westerlund
Intended status: Informational Ericsson
Expires: August 4, 2013 January 31, 2013
Multi-Media Concepts and Relations
draft-burman-rtcweb-mmusic-media-structure-00
Abstract
There are currently significant efforts ongoing in IETF regarding
more advanced multi-media functionalities, such as the work related
to RTCWEB and CLUE. This work includes use cases for both multi-
party communication and multiple media streams from an individual
end-point. The usage of scalable encoding or simulcast encoding as
well as different types of transport mechanisms have created
additional needs to correctly identify different types of resources
and describe their relations to achieve intended functionalities.
The different usages have both commonalities and differences in needs
and behavior. This document attempts to review some usages and
identify commonalities and needs. It then continues to highlight
important aspects that need to be considered in the definition of
these usages.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on August 4, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Existing RTP Usages . . . . . . . . . . . . . . . . . . . 4
3.1.1. Basic VoIP call . . . . . . . . . . . . . . . . . . . 4
3.1.2. Audio and Video Conference . . . . . . . . . . . . . . 5
3.1.3. Audio and Video Switched Conference . . . . . . . . . 7
3.2. WebRTC . . . . . . . . . . . . . . . . . . . . . . . . . . 8
3.2.1. Mesh-based Multi-party . . . . . . . . . . . . . . . . 9
3.2.2. Multi-source Endpoints . . . . . . . . . . . . . . . . 10
3.2.3. Media Relaying . . . . . . . . . . . . . . . . . . . . 11
3.2.4. Usage of Simulcast . . . . . . . . . . . . . . . . . . 11
3.3. CLUE Telepresence . . . . . . . . . . . . . . . . . . . . 13
3.3.1. Telepresence Functionality . . . . . . . . . . . . . . 13
3.3.2. Distributed Endpoint . . . . . . . . . . . . . . . . . 14
4. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.1. Commonalities in Use Cases . . . . . . . . . . . . . . . . 14
4.1.1. Media Source . . . . . . . . . . . . . . . . . . . . . 14
4.1.2. Encodings . . . . . . . . . . . . . . . . . . . . . . 16
4.1.3. Synchronization contexts . . . . . . . . . . . . . . . 17
4.1.4. Distributed Endpoints . . . . . . . . . . . . . . . . 18
4.2. Identified WebRTC issues . . . . . . . . . . . . . . . . . 18
4.3. Relevant to SDP evolution . . . . . . . . . . . . . . . . 19
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20
6. Security Considerations . . . . . . . . . . . . . . . . . . . 21
7. Informative References . . . . . . . . . . . . . . . . . . . . 21
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
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1. Introduction
This document concerns itself with the conceptual structures that can
be found in different logical levels of a multi-media communication,
from transport aspects to high-level needs of the communication
application. The intention is to provide considerations and guidance
that can be used when discussing how to resolve issues in the RTCWEB
and CLUE related standardization. Typical use cases for those WG
have commonalities that likely should be addressed similarly and in a
way that allows to align them.
The document starts with going deeper in the motivation why this has
become an important problem at this time. This is followed by
studies of some use cases and what concepts they contain, and
concludes with a discussion of observed commonalities and important
aspects to consider.
2. Motivation
There has arisen a number of new needs and requirements lately from
work such as WebRTC/RTCWEB [I-D.ietf-rtcweb-overview] and CLUE
[I-D.ietf-clue-framework]. The applications considered in those WG
has surfaced new requirements on the usage of both RTP [RFC3550] and
existing signalling solutions.
The main application aspects that have created new needs are:
o Multiple Media Streams from an end-point. The fact that an end-
point may have multiple media capture devices, such as cameras or
microphone mixes.
o Group communications involving multiple end-points. This is
realized using both mesh based connections as well as centralized
conference nodes. These creating a need for dealing with multiple
endpoints and/or multiple streams with different origins from a
transport peer.
o Media Stream Adaptation, both to adjust network resource
consumption as well as to handle varying end-point capabilities in
group communication.
o Transport mechanisms including both higher levels of aggregation
[I-D.ietf-mmusic-sdp-bundle-negotiation]
[I-D.ietf-avtcore-multi-media-rtp-session] and the use of
application-level transport repair mechanisms such as forward
error correction (FEC) and/or retransmission.
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The presence of multiple media resources or components creates a need
to identify, handle and group those resources across multiple
different instantiations or alternatives.
3. Use Cases
3.1. Existing RTP Usages
There are many different existing RTP usages. This section brings up
some that we deem interesting in comparison to the other use cases.
3.1.1. Basic VoIP call
This use case is intended to function as a base-line to contrast
against the rest of the use cases.
The communication context is an audio-only bi-directional
communication between two users, Alice and Bob. This communication
uses a single multi-media session that can be established in a number
of ways, but let's assume SIP/SDP [RFC3261][RFC3264]. This multi-
media session contains two end-points, one for Alice and one for Bob.
Each end-point has an audio capture device that is used to create a
single audio media source at each end-point.
+-------+ +-------+
| Alice |<------->| Bob |
+-------+ +-------+
Figure 1: Point-to-point Audio
The session establishment (SIP/SDP) negotiates the intent to
communicate over RTP using only the audio media type. Inherent in
the application is an assumption of only a single media source in
each direction. The boundaries for the encodings are represented
using RTP Payload types in conjunction with the SDP bandwidth
parameter (b=). The session establishment is also used to negotiate
that RTP will be used, thus resulting in that an RTP session will be
created for the audio. The underlying transport flows, in this case
a single bi-directional UDP flow for RTP, another for RTCP, is
configured by each end-point providing its' IP address and port,
which becomes source or destination depending on in which direction
the packet is sent.
The RTP session will have two RTP media streams, one in each
direction, which carries the encoding of the media source the sending
implementation has chosen based on the boundaries established by the
RTP payload types and other SDP parameters, e.g. codec, and bit-
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rates. The streams are in the RTP context identified by their SSRCs.
3.1.2. Audio and Video Conference
This use case is a multi-party use case with a central conference
node performing media mixing. It also includes two media types, both
audio and video. The high level topology of the communication
session is the following:
+-------+ +------------+ +-------+
| |<-Audio->| |<--Audio-->| |
| Alice | | | | Bob |
| |<-Video->| |<--Video-->| |
+-------+ | | +-------+
| Mixer |
+-------+ | | +-------+
| |<-Audio->| |<--Audio-->| |
|Charlie| | | | David |
| |<-Video->| |<--Video-->| |
+-------+ +------------+ +-------+
Figure 2: Audio and Video Conference with Centralized Mixing
The communication session is a multi-party conference including the
four users Alice, Bob, Charlie, and David. This communication
session contains four end-points and one middlebox (the Mixer). The
communication session is established using four different multi-media
sessions; one each between the user's endpoints and the middlebox.
Each of these multi-media sessions uses a session establishment
method, like SIP/SDP.
Looking at a single multi-media session between a user, e.g. Alice,
and the Mixer, there exist two media types, audio and video. Alice
has two capture devices, one video camera giving her a video media
source, and an audio capture device giving an audio media source.
These two media sources are captured in the same room by the same
end-point and thus have a strong timing relationship, requiring
inter-media synchronization at playback to provide the correct
fidelity. Thus Alice's endpoint has a synchronization context that
both her media sources use.
These two media sources are encoded using encoding parameters within
the boundaries that has been agreed between the end-point and the
Mixer using the session establishment. As has been common practice,
each media type will use its own RTP session between the end-point
and the mixer. Thus a single audio stream using a single SSRC will
flow from Alice to the Mixer in the Audio RTP session and a single
video stream will flow in the Video RTP session. Using this division
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in separate RTP sessions, the bandwidth of both audio and video can
be unambiguously and separately negotiated by the SDP bandwidth
attributes exchanged between the end-points and the mixer. Each RTP
session is using its own Transport Flows. The common synchronization
context across Alice's two media streams is identified by binding
both streams to the same CNAME, generated by Alice's endpoint.
The mixer does not have any physical capture devices, instead it
creates conceptual media sources. It provides two media sources
towards Alice; one audio being a mix of the audio from Bob, Charlie
and David, the second one being a conceptual video source that
contains a selection of one of the other video sources received from
Bob, Charlie, or David depending on who is speaking. The Mixer's
audio and video sources are provided in an encoding using a codec
that is supported by both Alice's endpoint and the mixer. These
streams are identified by a single SSRC in the respective RTP
session.
The mixer will have its own synchronization context and it will
inject the media from Bob, Charlie and David in a synchronized way
into the mixer's synchronization context to maintain the inter-media
synchronization of the original media sources.
The mixer establishes independent multimedia sessions with each of
the participant's endpoints. The mixer will in most cases also have
unique conceptual media sources for each of the endpoints. This as
audio mixes and video selections typically exclude media sources
originating from the receiving end-point. For example, Bob's audio
mix will be a mix of Alice, Charlie and David, and will not contain
Bob's own audio.
This use case may need unique user identities across the whole
communication session. An example functionality of this is a
participant list which includes audio energy levels showing who is
speaking within the audio mix. If that information is carried in RTP
using the RTP header extension for Mixer to audio clients [RFC6465]
then contributing source identities in the form of CSRC need to be
bound to the other end-point's media sources or user identities.
This despite the fact that each RTP session towards a particular
user's endpoint is terminated in the RTP mixer. This points out the
need for identifiers that exist in multiple multi-media session
contexts. In most cases this can easily be solved by the application
having identities tailored specifically for its own needs, but some
applications will benefit from having access to some commonly defined
structure for media source identities.
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3.1.3. Audio and Video Switched Conference
This use case is similar to the one above (Section 3.1.2), with the
difference that the mixer does not mix media streams by decoding,
mixing and re-encoding them, but rather switches a selection of
received media more or less unmodified towards receiving end-points.
This difference may not be very apparent to the end-user, but the
main motivations to eliminate the mixing operation and switch rather
than mix are:
o Lower processing requirements in the mixer.
o Lower complexity in the mixer.
o Higher media quality at the receiver given a certain media
bitrate.
o Lower end-to-end media delay.
Without the mixing operation, the mixer has limited ability to create
conceptual media sources that are customized for each receiver. The
reasons for such customizations comes from sender and receiver
differences in available resources and preferences:
o Presenting multiple conference users simultaneously, like in a
video mosaic.
o Alignment of sent media quality to receivers presentation needs.
o Alignment of codec type and configuration between sender and
receiver.
o Alignment of encoded bitrate to the available end-to-end link
bandwidth.
To enable elimination of the mixing operation, media sent to the
mixer must sufficiently well meet the above constraints for all
intended receivers. There are several ways to achieve this. One way
is to, by some system-wide design, ensure that all senders and
receivers are basically identical in all the above aspects. This may
however prove unrealistic when variations in conditions and end-
points are too large. Another way is to let a sender provide a
(small) set of alternative representations for each sent media
source, enough to sufficiently well cover the expected range of
variation. If those media source representations, encodings, are
independent from one another, they constitute a Simulcast of the
media source. If an encoding is instead dependent on and thus
requires reception of one or more other encodings, the representation
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of the media source jointly achieved by all dependent encodings is
said to be Scalable. Simulcast and Scalable encoding can also be
combined.
Both Simulcast and Scalable encodings result in that a single media
source generates multiple RTP media streams of the same media type.
The division of bandwidth between the Simulcast or Scalable streams
for a single media source is application specific and will vary. The
total bandwidth for a Simulcast or a Scalable source is the sum of
all included RTP media streams. Since all streams in a Simulcast or
Scalable source originate from the same capture device, they are
closely related and should thus share synchronization context.
The first and second customizations listed above, presenting multiple
conference users simultaneously, aligned with the presentation needs
in the receiver, can also be achieved without mixing operation by
simply sending appropriate quality media from those users
individually to each receiver. The total bandwidth of this user
presentation aggregate is the sum of all included RTP media streams.
Audio and video from a single user share synchronization context and
can be synchronized. Streams that originate from different users do
not have the same synchronization context, which is acceptable since
they do not need to be synchronized, but just presented jointly.
An actual mixer device need not be either mixing-only or switching-
only, but may implement both mixing and switching and may also choose
dynamically what to do for a specific media and a specific receiving
user on a case-by-case basis or based on some policy.
3.2. WebRTC
This section brings up two different instantiations of WebRTC
[ref-webrtc10] that stresses different aspects. But let's start with
reviewing some important aspects of WebRTC and the MediaStream
[ref-media-capture] API.
In WebRTC, an application gets access to a media source by calling
getUserMedia(), which creates a MediaStream [ref-media-capture] (note
the capitalization). A MediaStream consists of zero or more
MediaStreamTracks, where each MediaStreamTrack is associated with a
media source. These locally generated MediaStreams and their tracks
are connected to local media sources, which can be media devices such
as video cameras or microphones, but can also be files.
An WebRTC PeerConnection (PC) is an association between two endpoints
that is capable of communicating media from one end to the other.
The PC concept includes establishment procedures, including media
negotiation. Thus a PC is an instantiation of a Multimedia Session.
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When one end-point adds a MediaStream to a PC, the other endpoint
will by default receive an encoded representation of the MediaStream
and the active MediaStreamTracks.
3.2.1. Mesh-based Multi-party
This is a use case of WebRTC which establishes a multi-party
communication session by establishing an individual PC with each
participant in the communication session.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 3: WebRTC Mesh-based Multi-party
Users A, B and C want to have a joint communication session. This
communication session is created using a Web-application without any
central conference functionality. Instead, it uses a mesh of
PeerConnections to connect each participant's endpoint with the other
endpoints. In this example, three double-ended connections are
required to connect the three participants, and each endpoint has two
PCs.
This is an audio and video communication and each end-point has one
video camera and one microphone as media sources. Each endpoint
creates its own MediaStream with one video MediaStreamTrack and one
audio MediaStreamTrack. The endpoints add their MediaStream to both
of their PCs.
Let's now focus on a single PC; in this case the one established
between A and B. During the establishment of this PC, the two
endpoints agree to use only a single transport flow for all media
types, thus a single RTP session is created between A and B. A's
MediaStream has one audio media source that is encoded according to
the boundaries established by the PeerConnection establishment
signalling, which includes the RTP payload types and thus Codecs
supported as well as bit-rate boundaries. The encoding of A's media
source is then sent in an RTP stream identified by a unique SSRC. In
this case, as there are two media sources at A, two encodings will be
created which will be transmitted using two different RTP streams
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with their respective SSRC. Both these streams will reference the
same synchronization context through a common CNAME identifier used
by A. B will have the same configuration, thus resulting in at least
four SSRC being used in the RTP session part of the A-B PC.
Depending on the configuration of the two PCs that A has, i.e. the
A-B and the A-C ones, A could potentially reuse the encoding of a
media source in both contexts, under certain conditions. First, a
common codec and configuration needs to exist and the boundaries for
these configurations must allow a common work point. In addition,
the required bandwidth capacity needs to be available over the paths
used by the different PCs. Both of those conditions are not always
true. Thus it is quite likely that the endpoint will sometimes
instead be required to produce two different encodings of the same
media source.
If an application needs to reference the media from a particular
endpoint, it can use the MediaStream and MediaStreamTrack as they
point back to the media sources at a particular endpoint. This as
the MediaStream has a scope that is not PeerConnection specific.
The programmer can however implement this differently while
supporting the same use case. In this case the programmer creates
two MediaStreams that each have MediaStreamTracks that share common
media sources. This can be done either by calling getUserMedia()
twice, or by cloning the MediaStream obtained by the only
getUserMedia() call. In this example the result is two MediaStreams
that are connected to different PCs. From an identity perspective,
the two MediaStreams are different but share common media sources.
This fact is currently not made explicit in the API.
3.2.2. Multi-source Endpoints
This section concerns itself with endpoints that have more than one
media source for a particular media type. A straightforward example
would be a laptop with a built in video camera used to capture the
user and a second video camera, for example attached by USB, that is
used to capture something else the user wants to show. Both these
cameras are typically present in the same sound field, so it will be
common to have only a single audio media source.
A possible way of representing this is to have two MediaStreams, one
with the built in camera and the audio, and a second one with the USB
camera and the audio. Each MediaStream is intended to be played with
audio and video synchronized, but the user (local or remote) or
application is likely to switch between the two captures.
It becomes important for a receiving endpoint that it can determine
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that the audio in the two MediaStreams have the same synchronization
context. Otherwise a receiver may playback the same media source
twice, with some time overlap, at a switch between playing the two
MediaStreams. Being able to determine that they are the same media
source further allow for removing redundancy by having a single
encoding if appropriate for both MediaStreamTracks.
3.2.3. Media Relaying
WebRTC endpoints can relay a received MediaStream from one PC to
another by the simple API level maneuver of adding the received
MediaStream to the other PC. To realize this in the implementation
is more complex. This can also cause some issues from a media
perspective. If an application spanning across multiple endpoints
that relay media between each other makes a mistake, a media loop can
be created. Media Loops could become a significant issue. For
example could an audio echo be created, i.e. an endpoint receives its
own media without detecting that it is its own media and plays it
back with some delay. In case a WebRTC endpoint produces a
conceptual media source by mixing incoming MediaStreams, if there is
no loop detection, a feedback loop can be created.
RTP has loop detection to detect and handle such cases within a
single RTP session. However, in the context of WebRTC, the RTP
session is local to the PC and thus cannot rely on the RTP level loop
detection. Instead, if this protection is needed on the WebRTC
MediaStream level, it could for example be achieved by having media
source identifiers that can be preserved between the different
MediaStreams in the PCs.
When relaying media and in case one receives multiple encodings of
the same source it is beneficial to know that. For example, if one
encoding arrives with a delay of 80 ms and another with 450 ms, being
able to choose the one with 80 ms and not be forced to delay all
media sources from the same synchronization context to the most
delayed source improves performance.
3.2.4. Usage of Simulcast
In this section we look at a use case applying simulcast from each
user's endpoint to a central conference node to avoid the need for an
individual encoding to each receiving endpoint. Instead, the central
node chooses which of the available encodings that is forwarded to a
particular receiver, like in Section 3.1.3.
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+-----------+ +------------+ Enc2 +---+
| A +-Enc1|----->| |----->| B |
| | | | | +---+
| Src-+-Enc2|----->| | Enc1 +---+
+-----------+ | Mixer |----->| C |
| | +---+
| | Enc2 +---+
| |----->| D |
+------------+ +---+
Figure 4
In this Communication Session there are four users with endpoints and
one middlebox (The Mixer). This is an audio and video communication
session. The audio source is not simulcasted and the endpoint only
needs to produce a single encoding. For the video source, each
endpoint will produce multiple encodings (Enc1 and Enc2 in Figure 4)
and transfer them simultaneously to the mixer. The mixer picks the
most appropriate encoding for the path from the mixer to each
receiving client.
Currently there exists no specified way in WebRTC to realise the
above, although use-cases and requirements discuss simulcast
functionality. The authors believe there exist two possible solution
alternatives in the WebRTC context:
Multiple Encodings within a PeerConnection: The endpoint that wants
to provide a simulcast creates one or more MediaStreams with the
media sources it wants to transmit over a particular PC. The
WebRTC API provides functionality to enable multiple encodings to
be produced for a particular MediaStreamTrack and have possibility
to configure the desired quality levels and/or differences for
each of the encodings.
Using Multiple PeerConnections: There exist capabilities to both
negotiate and control the codec, bit-rate, video resolution,
frame-rate, etc of a particular MediaStreamTrack in the context of
one PeerConnection. Thus one method to provide multiple encodings
is to establish multiple PeerConnections between A and the Mixer,
where each PC is configured to provide the desired quality. Note
that this solution comes in two flavors from an application
perspective. One is that the same MediaStream object is added to
the two PeerConnections. The second is that two different
MediaStream objects, with the same number of MediaStreamTracks and
representing the same sources, are created (e.g by cloning), one
of them added to the first PeerConnection and the second one to
the second PeerConnection.
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Both of these solutions share a common requirement, the need to
separate the received RTP streams not only based on media source, but
also on the encoding. However, on an API level the solutions appear
different. For Multiple Encodings within the context of a PC, the
receiver will need new access methods for accessing and manipulating
the different encodings. Using multiple PC instead requires that one
can easily determine the shared (simulcasted) media source despite
receiving it in multiple MediaStreams on different PCs. If the same
MediaStream is added to both PC's the id's of the MediaStream and
MediaStreamTracks will be the same, while they will be different if
different MediaStream's (but representing the same sources) are added
to the two PC's.
3.3. CLUE Telepresence
The CLUE framework [I-D.ietf-clue-framework] and use case
[I-D.ietf-clue-telepresence-use-cases] documents make use of most, if
not all, media concepts that were already discussed in previous
sections, and adds a few more.
3.3.1. Telepresence Functionality
A communicating CLUE Endpoint can, compared to other types of
Endpoints, be characterized by using multiple media resources:
o Multiple capture devices, such as cameras or microphones,
generating the media for a media source.
o Multiple render devices, such as displays or speakers.
o Multiple Media Types, such as audio, video and presentation
streams.
o Multiple remote Endpoints, since conference is a typical use case.
o Multiple Encodings (encoded representations) of a media source.
o Multiple Media Streams representing multiple media sources.
To make the multitude of resources more manageable, CLUE introduces
some additional structures. For example, related media sources in a
multimedia session are grouped into Scenes, which can generally be
represented in different ways, described by alternative Scene
Entries. CLUE explicitly separates the concept of a media source
from the encoded representations of it and a single media source can
be used to create multiple Encodings. It is also possible in CLUE to
account for constraints in resource handling, like limitations in
possible Encoding combinations due to physical device implementation.
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The number of media resources typically differ between Endpoints.
Specifically, the number of available media resources of a certain
type used for sending at the sender side typically does not match the
number of corresponding media resources used for receiving at the
receiver side. Some selection process must thus be applied either at
the sender or the receiver to select a subset of resources to be
used. Hence, each resource that need to be part of that selection
process must have some identification and characterization that can
be understood by the selecting party. In the CLUE model, the sender
(Provider) announces available resources and the receiver (Consumer)
chooses what to receive. This choice is made independently in the
two directions of a bi-directional communication.
3.3.2. Distributed Endpoint
The definition of a single CLUE Endpoint in the framework
[I-D.ietf-clue-framework] says it can consist of several physical
devices with source and sink media streams. This means that each
logical node of such distributed Endpoint can have a separate
transport interface, and thus that media sources originating from the
same Endpoint can have different transport addresses.
4. Discussion
This section discusses some conclusions the authors make based on the
use cases. First we will discuss commonalities between use cases.
Secondly we will provide a summary of issues we see affect WebRTC.
Lastly we consider aspects that need to be considered in the SDP
evolution that is ongoing.
4.1. Commonalities in Use Cases
The above use cases illustrate a couple of concepts that are not well
defined, nor have they fully specified standard mechanisms or
behaviors. This section contains a discussion of such concepts,
which the authors believe are useful in more than one context and
thus should be defined to provide a common function when needed by
multi-media communication applications.
4.1.1. Media Source
In several of the above use cases there exist a need for a separation
between the media source, the particular encoding and its transport
stream. In vanilla RTP there exist a one-to-one mapping between
these; one media source is encoded in one particular way and
transported as one RTP stream using a single SSRC in a particular RTP
session.
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The reason for not keeping a strict one-to-one mapping, allowing the
media source to be identified separately from the RTP media stream
(SSRC), varies depending on the application's needs and the desired
functionalities:
Simulcast: Simulcast is a functionality to provide multiple
simultaneous encodings of the same media source. As each encoding
is independent of the other, in contrast to scalable encoding,
independent transport streams for each encoding is needed. The
receiver of a simulcast stream will need to be able to explicitly
identify each encoding upon reception, as well as which media
source it is an encoding of. This is especially important in a
context of multiple media sources being provided from the same
endpoint.
Mesh-based communication: When a communication application
implements multi-party communication through a mesh of transport
flows, there exist a need for tracking the original media source,
especially when relaying between nodes is possible. It is likely
that the encodings provided over the different transports are
different. If an application uses relaying between different
transports, an endpoint may, intentionally or not, receive
multiple encodings of the same media source over the same or
different transports. Some applications can handle the needed
identification, but some can benefit from a standardized method to
identify sources.
The second argument above can be generalized into a common need in
applications that utilize multiple multimedia sessions, such as
multiple PeerConnections or multiple SIP/SDP-established RTP
sessions, to form a larger communication session between multiple
endpoints. These applications commonly need to track media sources
that occur in more than one multimedia session.
Looking at both CLUE and WebRTC, they appear to contain their own
variants of the concept that was above denoted a media source. In
CLUE it is called Media Capture. In WebRTC each MediaStreamTrack is
identifiable, however, several MediaStreamTracks can share the actual
source, and there is no way for the application to realize this
currently. The identification of sources is being discussed, and
there is a proposal [ref-leithead] that introduces the concept 'Track
Source'. Thus, in this document we see the media source as the
generalized commonality between these two concepts. Giving each
media source a unique identifier in the communication session/context
that is reused in all the PeerConnections or SIP/SDP-established RTP
sessions would enable loop detection, correctly associate alternative
encodings and provide a common name across the endpoints for
application logic to reference the actual media source rather than a
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particular encoding or transport stream.
It is arguable if the application should really know a long term
persistent source identification, such as based on hardware
identities, for example due to fingerprinting issues, and it would
likely be better to use an anonymous identification that is still
unique in a sufficiently wide context, for example within the
communication application instance.
4.1.2. Encodings
An Encoding is a particular encoded representation of a particular
media source. In the context of RTP and Signalling, a particular
encoding must fit the established parameters, such as RTP payload
types, media bandwidths, and other more or less codec-specific media
constraints such as resolution, frame-rate, fidelity, audio
bandwidth, etc.
In the context of an application, it appears that there are primarily
two considerations around the use of multiple encodings.
The first is how many and what their defining parameters are. This
may require to be negotiated, something the existing signalling
solutions, like SDP, currently lack support for. For example in SDP,
there exist no way to express that you would like to receive three
different encodings of a particular video source. In addition, if
you for example prefer these three encodings to be 720p/25 Hz,
360p/25 Hz and 180p/12.5 Hz, and even if you could define RTP payload
types with these constraints, they must be linked to RTP streams
carrying the encodings of the particular source. Also, for some RTP
payload types there exist difficulties to express encoding
characteristics with the desired granularity. The number of RTP
payload types that can be used for a particular potential encoding
can also be a constraint, especially as a single RTP payload type
could well be used for all three target resolutions and frame rates
in the example. Using multiple encodings might even be desirable for
multi-party conferences that switches video, rather than composites
and re-encodes it. It might be that SDP is not the most suitable
place to negotiate this. From an application perspective, utilizing
clients that have standardized APIs or protocols to control them,
there exist a need for the application to express what it prefers in
number of encodings as well as what their primary target parameters
are.
Secondly, some applications may need explicit indication of what
encoding a particular stream represents. In some cases this can be
deduced based on information such as RTP payload types and parameters
received in the media stream, but such implicit information will not
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always be detailed enough and it may also be time-consuming to
extract. For example, in SDP there is currently limitations for
binding the relevant information about a particular encoding to the
corresponding RTP stream, unless only a single RTP stream is defined
per media description (m= line).
The CLUE framework explicitly discusses encodings as constraints that
are applied when transforming a media source (capture) into what CLUE
calls a capture encoding. This includes both explicit identification
as well as a set of boundary parameters such as maximum width,
height, frame rate as well as bandwidth. In WebRTC nothing related
has yet been defined, and we note this as an issue that needs to be
resolved. This as the authors expect that support for multiple
encodings will be required to enable simulcast and scalability.
4.1.3. Synchronization contexts
The shortcomings around synchronization contexts appears rather
limited. In RTP, each RTP media stream is associated with a
particular synchronization context through the CNAME session
description item. The main concerns here are likely twofold.
The first concern is to avoid unnecessary creation of new contexts,
and rather correctly associate with the contexts that actually exist.
For example, WebRTC MediaStreams are defined so that all
MediaStreamTracks within a particular MediaStream shall be
synchronized. An easy method for meeting this would be to assign a
new CNAME for each MediaStream. However, that would ignore the fact
that several media sources from the same synchronization context may
appear in different combinations across several MediaStreams. Thus
all these MediaStreams should share synchronization context to avoid
playback glitches, like playing back different instantiations of a
single media source out of sync because the media source was shared
between two different MediaStreams.
The second problem is that synchronization context identification in
RTP, i.e. CNAME, is overloaded as an endpoint identifier. As an
example, consider an endpoint that has two synchronization contexts;
one for audio and video in the room and another for an audio and
video presentation stream, like the output of an DVD player. Relying
on that an endpoint has only a single synchronization context and
CNAME may be incorrect and could create issues that an application
designer as well as RTP and signalling extension specifications need
to watch out for.
CLUE discusses so far quite little about synchronization, but clearly
intends to enable lip synchronization between captures that have that
relation. The second issue is however quite likely to be encountered
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in CLUE due to explicit inclusion of the Scene concept, where
different Scenes do not require to share the same synchronization
context, but is rather intended for situations where Scenes cannot
share synchronization context.
4.1.4. Distributed Endpoints
When an endpoint consists of multiple nodes, the added complexity is
often local to that endpoint, which is appropriate. However, some
few properties of distributed endpoints needs to be tolerated by all
entities in a multimedia communication session. The main item is to
not assume that a single endpoint will only use a single network
address. This is a dangerous assumption even for non-distributed
endpoints due to multi-homing and the common deployment of NATs,
especially large scale NATs which in worst case uses multiple
addresses for a single endpoint's transport flows.
Distributed endpoints are brought up in the CLUE context. They are
not specifically discussed in the WebRTC context, instead the desire
for transport level aggregation makes such endpoints problematic.
However, WebRTC does allow for fallback to media type specific
transport flows and can thus without issues support distributed
endpoints.
4.2. Identified WebRTC issues
In the process of identifying commonalities and differences between
the different use cases we have identified what to us appears to be
issues in the current specification of WebRTC that needs to be
reviewed.
1. If simulcast or scalability are to be supported at all, the
WebRTC API will need to find a method to deal more explicitly
with the existence of different encodings and how these are
configured, accessed and referenced. For simulcast, the authors
see a quite straightforward solution where each PeerConnection is
only allowed to contain a single encoding for a specific media
source and the desired quality level can be negotiated for the
full PeerConnection. When multiple encodings are desired,
multiple PeerConnections with differences in configuration are
established. That would only require that the underlying media
source can explicitly be indicated and tracked by the receiver.
2. The current API structure allows to have multiple MediaStreams
with fully or partially overlapping media sources. This,
combined with multiple PeerConnections and the likely possibility
to do relaying, there appears to exist a significant need to
determine the underlying media source, despite receiving
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different MediaStreams with particular media sources encoded in
different ways. It is proposed that MediaSources are made
possible to identify uniquely across multiple PeerConnections in
the context of the communication application. It is however
likely that while being unique in a sufficiently large context,
the identification should also be anonymous to avoid
fingerprinting issues, similar to the situation discussed in
Section 4.1.1.
3. Implementations of the MediaStream API must be careful in how
they name and deal with synchronization contexts, so that the
actual underlying synchronization context is preserved when
possible. It should be noted that cannot be done when a
MediaStream is created that contains media sources from multiple
synchronization contexts. This will instead require
resynchronization of contributing sources, creation of a new
synchronization context, and inserting the sources into that
synchronization context.
These issues need to be discussed and an appropriate way to resolve
them must be chosen.
4.3. Relevant to SDP evolution
The joint MMUSIC / RTCWeb WGs interim meeting in February 2013 will
discuss a number of SDP related issues around the handling of
multiple sources; the aggregation of multiple media types over the
same RTP session as well as RTP sharing its transport flow not only
with ICE/STUN but also with the WebRTC data channel using SCTP/DTLS/
UDP. These issues will potentially result in a significant impact on
SDP. It may also impact other ongoing work as well as existing
usages and applications, making these discussions difficult.
The above use cases and discussion points to the existence of a
number of commonalities between WebRTC and CLUE, and that a solution
should preferably be usable by both. It is a very open question how
much functionality CLUE requires from SDP, as CLUE WG plans to
develop a protocol with a different usage model. The appropriate
division in functionality between SDP and this protocol is currently
unknown.
Based on this document, it is possible to express some protocol
requirements when negotiating multimedia sessions and their media
configurations. Note that this is written as requirements to
consider, given that one believes this functionality is needed in
SDP.
The Requirements:
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Encoding negotiation: For Simulcast and Scalability in applications,
it must be possible to negotiate the number and the boundary
conditions for the desired encodings created from a particular
media source.
Media Resource Identification: SDP-based applications that need
explicit information about media sources, multiple encodings and
their related RTP media streams could benefit from a common way of
providing this information. This need can result in multiple
different actual requirements. Some require a common, explicit
identification of media sources across multiple signalling
contexts. Some may require explicit indication of which set of
encodings that has the same media source and thus which sets of
RTP media streams (SSRCs) that are related to a particular media
source.
RTP media stream parameters: With a greater heterogeneity of the
possible encodings and their boundary conditions, situations may
arise where some or sets of RTP media streams will need to have
specific sets of parameters associated with them, compared to
other (sets of) RTP media streams.
The above are general requirements and in some cases the appropriate
point to address the requirement may not even be SDP. For example,
media source identification could primarily be put in an RTCP Session
Description (SDES) item, and only when so required by the application
also be included in the signalling.
The discussion in this document has impact on the high level decision
regarding how to relate RTP media streams to SDP media descriptions.
However, as it is currently presenting concepts rather than giving
concrete proposals on how to enable these concepts as extensions to
SDP or other protocols, it is difficult to determine the actual
impact that a high level solution will have. However, the authors
are convinced that neither of the directions will prevent the
definition of suitable concepts in SDP.
5. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
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6. Security Considerations
The realization of the proposed concepts and the resolution will have
security considerations. However, at this stage it is unclear if any
has not already common considerations regarding preserving privacy,
confidentiality and ensure integrity to prevent denial of service or
quality degradations.
7. Informative References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Multiple
Media Types in an RTP Session",
draft-ietf-avtcore-multi-media-rtp-session-01 (work in
progress), October 2012.
[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams",
draft-ietf-clue-framework-08 (work in progress),
December 2012.
[I-D.ietf-clue-telepresence-use-cases]
Romanow, A., Botzko, S., Duckworth, M., Even, R., and I.
Communications, "Use Cases for Telepresence Multi-
streams", draft-ietf-clue-telepresence-use-cases-04 (work
in progress), August 2012.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-01 (work in
progress), August 2012.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-05 (work
in progress), December 2012.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011.
[ref-leithead]
Microsoft, "Proposal: Media Capture and Streams Settings
API v6, https://dvcs.w3.org/hg/dap/raw-file/tip/
media-stream-capture/proposals/
SettingsAPI_proposal_v6.html", December 2012.
[ref-media-capture]
"Media Capture and Streams,
http://dev.w3.org/2011/webrtc/editor/getusermedia.html",
December 2012.
[ref-webrtc10]
"WebRTC 1.0: Real-time Communication Between Browsers,
http://dev.w3.org/2011/webrtc/editor/webrtc.html",
January 2013.
Authors' Addresses
Bo Burman
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
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