Network Working Group J. Lennox
Internet-Draft Vidyo
Intended status: Informational K. Gross
Expires: December 29, 2014 AVA
S. Nandakumar
G. Salgueiro
Cisco Systems
B. Burman
Ericsson
June 27, 2014
A Taxonomy of Grouping Semantics and Mechanisms for Real-Time Transport
Protocol (RTP) Sources
draft-ietf-avtext-rtp-grouping-taxonomy-02
Abstract
The terminology about, and associations among, Real-Time Transport
Protocol (RTP) sources can be complex and somewhat opaque. This
document describes a number of existing and proposed relationships
among RTP sources, and attempts to define common terminology for
discussing protocol entities and their relationships.
Status of This Memo
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provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on December 29, 2014.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Concepts . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Media Chain . . . . . . . . . . . . . . . . . . . . . . . 4
2.1.1. Physical Stimulus . . . . . . . . . . . . . . . . . . 8
2.1.2. Media Capture . . . . . . . . . . . . . . . . . . . . 8
2.1.3. Raw Stream . . . . . . . . . . . . . . . . . . . . . 8
2.1.4. Media Source . . . . . . . . . . . . . . . . . . . . 8
2.1.5. Source Stream . . . . . . . . . . . . . . . . . . . . 9
2.1.6. Media Encoder . . . . . . . . . . . . . . . . . . . . 9
2.1.7. Encoded Stream . . . . . . . . . . . . . . . . . . . 10
2.1.8. Dependent Stream . . . . . . . . . . . . . . . . . . 11
2.1.9. Media Packetizer . . . . . . . . . . . . . . . . . . 11
2.1.10. RTP Stream . . . . . . . . . . . . . . . . . . . . . 11
2.1.11. Media Redundancy . . . . . . . . . . . . . . . . . . 12
2.1.12. Redundancy RTP Stream . . . . . . . . . . . . . . . . 12
2.1.13. Media Transport . . . . . . . . . . . . . . . . . . . 13
2.1.14. Media Transport Sender . . . . . . . . . . . . . . . 14
2.1.15. Sent RTP Stream . . . . . . . . . . . . . . . . . . . 14
2.1.16. Network Transport . . . . . . . . . . . . . . . . . . 14
2.1.17. Transported RTP Stream . . . . . . . . . . . . . . . 14
2.1.18. Media Transport Receiver . . . . . . . . . . . . . . 14
2.1.19. Received RTP Stream . . . . . . . . . . . . . . . . . 15
2.1.20. Received Redundancy RTP Stream . . . . . . . . . . . 15
2.1.21. Media Repair . . . . . . . . . . . . . . . . . . . . 15
2.1.22. Repaired RTP Stream . . . . . . . . . . . . . . . . . 15
2.1.23. Media Depacketizer . . . . . . . . . . . . . . . . . 15
2.1.24. Received Encoded Stream . . . . . . . . . . . . . . . 16
2.1.25. Media Decoder . . . . . . . . . . . . . . . . . . . . 16
2.1.26. Received Source Stream . . . . . . . . . . . . . . . 16
2.1.27. Media Sink . . . . . . . . . . . . . . . . . . . . . 16
2.1.28. Received Raw Stream . . . . . . . . . . . . . . . . . 17
2.1.29. Media Render . . . . . . . . . . . . . . . . . . . . 17
2.2. Communication Entities . . . . . . . . . . . . . . . . . 17
2.2.1. End Point . . . . . . . . . . . . . . . . . . . . . . 18
2.2.2. RTP Session . . . . . . . . . . . . . . . . . . . . . 18
2.2.3. Participant . . . . . . . . . . . . . . . . . . . . . 19
2.2.4. Multimedia Session . . . . . . . . . . . . . . . . . 20
2.2.5. Communication Session . . . . . . . . . . . . . . . . 20
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3. Relations at Different Levels . . . . . . . . . . . . . . . . 21
3.1. Synchronization Context . . . . . . . . . . . . . . . . . 22
3.1.1. RTCP CNAME . . . . . . . . . . . . . . . . . . . . . 22
3.1.2. Clock Source Signaling . . . . . . . . . . . . . . . 22
3.1.3. Implicitly via RtcMediaStream . . . . . . . . . . . . 22
3.1.4. Explicitly via SDP Mechanisms . . . . . . . . . . . . 22
3.2. End Point . . . . . . . . . . . . . . . . . . . . . . . . 22
3.3. Participant . . . . . . . . . . . . . . . . . . . . . . . 23
3.4. RtcMediaStream . . . . . . . . . . . . . . . . . . . . . 23
3.5. Single- and Multi-Session Transmission of SVC . . . . . . 23
3.6. Multi-Channel Audio . . . . . . . . . . . . . . . . . . . 24
3.7. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 24
3.8. Layered Multi-Stream . . . . . . . . . . . . . . . . . . 25
3.9. RTP Stream Duplication . . . . . . . . . . . . . . . . . 27
3.10. Redundancy Format . . . . . . . . . . . . . . . . . . . . 27
3.11. RTP Retransmission . . . . . . . . . . . . . . . . . . . 28
3.12. Forward Error Correction . . . . . . . . . . . . . . . . 29
3.13. RTP Stream Separation . . . . . . . . . . . . . . . . . . 31
3.14. Multiple RTP Sessions over one Media Transport . . . . . 32
4. Mapping from Existing Terms . . . . . . . . . . . . . . . . . 32
4.1. Audio Capture . . . . . . . . . . . . . . . . . . . . . . 32
4.2. Capture Device . . . . . . . . . . . . . . . . . . . . . 32
4.3. Capture Encoding . . . . . . . . . . . . . . . . . . . . 32
4.4. Capture Scene . . . . . . . . . . . . . . . . . . . . . . 33
4.5. Endpoint . . . . . . . . . . . . . . . . . . . . . . . . 33
4.6. Individual Encoding . . . . . . . . . . . . . . . . . . . 33
4.7. Multipoint Control Unit (MCU) . . . . . . . . . . . . . . 33
4.8. Media Capture . . . . . . . . . . . . . . . . . . . . . . 33
4.9. Media Consumer . . . . . . . . . . . . . . . . . . . . . 33
4.10. Media Description . . . . . . . . . . . . . . . . . . . . 33
4.11. Media Provider . . . . . . . . . . . . . . . . . . . . . 34
4.12. Media Stream . . . . . . . . . . . . . . . . . . . . . . 34
4.13. Multimedia Session . . . . . . . . . . . . . . . . . . . 34
4.14. Recording Device . . . . . . . . . . . . . . . . . . . . 34
4.15. RtcMediaStream . . . . . . . . . . . . . . . . . . . . . 34
4.16. RtcMediaStreamTrack . . . . . . . . . . . . . . . . . . . 35
4.17. RTP Sender . . . . . . . . . . . . . . . . . . . . . . . 35
4.18. RTP Session . . . . . . . . . . . . . . . . . . . . . . . 35
4.19. SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . 35
4.20. Stream . . . . . . . . . . . . . . . . . . . . . . . . . 35
4.21. Video Capture . . . . . . . . . . . . . . . . . . . . . . 35
5. Security Considerations . . . . . . . . . . . . . . . . . . . 35
6. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . 36
7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 36
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 36
9. Informative References . . . . . . . . . . . . . . . . . . . 36
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 38
A.1. Modifications Between WG Version -01 and -02 . . . . . . 38
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A.2. Modifications Between WG Version -00 and -01 . . . . . . 39
A.3. Modifications Between Version -02 and -03 . . . . . . . . 40
A.4. Modifications Between Version -01 and -02 . . . . . . . . 40
A.5. Modifications Between Version -00 and -01 . . . . . . . . 40
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 40
1. Introduction
The existing taxonomy of sources in RTP is often regarded as
confusing and inconsistent. Consequently, a deep understanding of
how the different terms relate to each other becomes a real
challenge. Frequently cited examples of this confusion are (1) how
different protocols that make use of RTP use the same terms to
signify different things and (2) how the complexities addressed at
one layer are often glossed over or ignored at another.
This document attempts to provide some clarity by reviewing the
semantics of various aspects of sources in RTP. As an organizing
mechanism, it approaches this by describing various ways that RTP
sources can be grouped and associated together.
All non-specific references to ControLling mUltiple streams for
tElepresence (CLUE) in this document map to [I-D.ietf-clue-framework]
and all references to Web Real-Time Communications (WebRTC) map to
[I-D.ietf-rtcweb-overview].
2. Concepts
This section defines concepts that serve to identify and name various
transformations and streams in a given RTP usage. For each concept
an attempt is made to list any alternate definitions and usages that
co-exist today along with various characteristics that further
describes the concept. These concepts are divided into two
categories, one related to the chain of streams and transformations
that media can be subject to, the other for entities involved in the
communication.
2.1. Media Chain
In the context of this memo, Media is a sequence of synthetic or
Physical Stimulus (Section 2.1.1) (sound waves, photons, key-
strokes), represented in digital form. Synthesized Media is
typically generated directly in the digital domain.
This section contains the concepts that can be involved in taking
Media at a sender side and transporting it to a receiver, which may
recover a sequence of physical stimulus. This chain of concepts is
of two main types, streams and transformations. Streams are time-
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based sequences of samples of the physical stimulus in various
representations, while transformations changes the representation of
the streams in some way.
The below examples are basic ones and it is important to keep in mind
that this conceptual model enables more complex usages. Some will be
further discussed in later sections of this document. In general the
following applies to this model:
o A transformation may have zero or more inputs and one or more
outputs.
o A stream is of some type.
o A stream has one source transformation and one or more sink
transformations (with the exception of Physical Stimulus
(Section 2.1.1) that may lack source or sink transformation).
o Streams can be forwarded from a transformation output to any
number of inputs on other transformations that support that type.
o If the output of a transformation is sent to multiple
transformations, those streams will be identical; it takes a
transformation to make them different.
o There are no formal limitations on how streams are connected to
transformations, this may include loops if required by a
particular transformation.
It is also important to remember that this is a conceptual model.
Thus real-world implementations may look different and have different
structure.
To provide a basic understanding of the relationships in the chain we
below first introduce the concepts for the sender side (Figure 1).
This covers physical stimulus until media packets are emitted onto
the network.
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Physical Stimulus
|
V
+--------------------+
| Media Capture |
+--------------------+
|
Raw Stream
V
+--------------------+
| Media Source |<- Synchronization Timing
+--------------------+
|
Source Stream
V
+--------------------+
| Media Encoder |
+--------------------+
|
Encoded Stream +-----------+
V | V
+--------------------+ | +--------------------+
| Media Packetizer | | | Media Redundancy |
+--------------------+ | +--------------------+
| | |
+------------+ Redundancy RTP Stream
Source RTP Stream |
V V
+--------------------+ +--------------------+
| Media Transport | | Media Transport |
+--------------------+ +--------------------+
Figure 1: Sender Side Concepts in the Media Chain
In Figure 1 we have included a branched chain to cover the concepts
for using redundancy to improve the reliability of the transport.
The Media Transport concept is an aggregate that is decomposed below
in Section 2.1.13.
Below we review a receiver media chain (Figure 2) matching the sender
side to look at the inverse transformations and their attempts to
recover possibly identical streams as in the sender chain. Note that
the streams out of a reverse transformation, like the Source Stream
out the Media Decoder are in many cases not the same as the
corresponding ones on the sender side, thus they are prefixed with a
"Received" to denote a potentially modified version. The reason for
not being the same lies in the transformations that can be of
irreversible type. For example, lossy source coding in the Media
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Encoder prevents the Source Stream out of the Media Decoder to be the
same as the one fed into the Media Encoder. Other reasons include
packet loss or late loss in the Media Transport transformation that
even Media Repair, if used, fails to repair. It should be noted that
some transformations are not always present, like Media Repair that
cannot operate without Redundancy RTP Streams.
+--------------------+ +--------------------+
| Media Transport | | Media Transport |
+--------------------+ +--------------------+
| |
Received RTP Stream Received Redundancy RTP Stream
| |
| +-------------------+
V V
+--------------------+
| Media Repair |
+--------------------+
|
Repaired RTP Stream
V
+--------------------+
| Media Depacketizer |
+--------------------+
|
Received Encoded Stream
V
+--------------------+
| Media Decoder |
+--------------------+
|
Received Source Stream
V
+--------------------+
| Media Sink |--> Synchronization Information
+--------------------+
|
Received Raw Stream
V
+--------------------+
| Media Renderer |
+--------------------+
|
V
Physical Stimulus
Figure 2: Receiver Side Concepts of the Media Chain
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2.1.1. Physical Stimulus
The physical stimulus is a physical event that can be measured and
converted to digital form by an appropriate sensor or transducer.
This include sound waves making up audio, photons in a light field
that is visible, or other excitations or interactions with sensors,
like keystrokes on a keyboard.
2.1.2. Media Capture
Media Capture is the process of transforming the Physical Stimulus
(Section 2.1.1) into digital Media using an appropriate sensor or
transducer. The Media Capture performs a digital sampling of the
physical stimulus, usually periodically, and outputs this in some
representation as a Raw Stream (Section 2.1.3). This data is due to
its periodical sampling, or at least being timed asynchronous events,
some form of a stream of media data. The Media Capture is normally
instantiated in some type of device, i.e. media capture device.
Examples of different types of media capturing devices are digital
cameras, microphones connected to A/D converters, or keyboards.
Characteristics:
o A Media Capture is identified either by hardware/manufacturer ID
or via a session-scoped device identifier as mandated by the
application usage.
o A Media Capture can generate an Encoded Stream (Section 2.1.7) if
the capture device support such a configuration.
2.1.3. Raw Stream
The time progressing stream of digitally sampled information, usually
periodically sampled and provided by a Media Capture (Section 2.1.2).
A Raw Stream can also contain synthesized Media that may not require
any explicit Media Capture, since it is already in an appropriate
digital form.
2.1.4. Media Source
A Media Source is the logical source of a reference clock
synchronized, time progressing, digital media stream, called a Source
Stream (Section 2.1.5). This transformation takes one or more Raw
Streams (Section 2.1.3) and provides a Source Stream as output. This
output has been synchronized with some reference clock, even if just
a system local wall clock.
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The output can be of different types. One type is directly
associated with a particular Media Capture's Raw Stream. Others are
more conceptual sources, like an audio mix of multiple Raw Streams
(Figure 3), a mixed selection of the three loudest inputs regarding
speech activity, a selection of a particular video based on the
current speaker, i.e. typically based on other Media Sources.
Raw Raw Raw
Stream Stream Stream
| | |
V V V
+--------------------------+
| Media Source |<-- Reference Clock
| Mixer |
+--------------------------+
|
V
Source Stream
Figure 3: Conceptual Media Source in form of Audio Mixer
Characteristics:
o At any point, it can represent a physical captured source or
conceptual source.
2.1.5. Source Stream
A time progressing stream of digital samples that has been
synchronized with a reference clock and comes from particular Media
Source (Section 2.1.4).
2.1.6. Media Encoder
A Media Encoder is a transform that is responsible for encoding the
media data from a Source Stream (Section 2.1.5) into another
representation, usually more compact, that is output as an Encoded
Stream (Section 2.1.7).
The Media Encoder step commonly includes pre-encoding
transformations, such as scaling, resampling etc. The Media Encoder
can have a significant number of configuration options that affects
the properties of the encoded stream. This include properties such
as bit-rate, start points for decoding, resolution, bandwidth or
other fidelity affecting properties. The actually used codec is also
an important factor in many communication systems, not only its
parameters.
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Scalable Media Encoders need special mentioning as they produce
multiple outputs that are potentially of different types. A scalable
Media Encoder takes one input Source Stream and encodes it into
multiple output streams of two different types; at least one Encoded
Stream that is independently decodable and one or more Dependent
Streams (Section 2.1.8) that requires at least one Encoded Stream and
zero or more Dependent Streams to be possible to decode. A Dependent
Stream's dependency is one of the grouping relations this document
discusses further in Section 3.8.
Source Stream
|
V
+--------------------------+
| Scalable Media Encoder |
+--------------------------+
| | ... |
V V V
Encoded Dependent Dependent
Stream Stream Stream
Figure 4: Scalable Media Encoder Input and Outputs
There are also other variants of encoders, like so-called Multiple
Description Coding (MDC). Such Media Encoder produce multiple
independent and thus individually decodable Encoded Streams that are
possible to combine into a Received Source Stream that is somehow a
better representation of the original Source Stream than using only a
single Encoded Stream.
Characteristics:
o A Media Source can be multiply encoded by different Media Encoders
to provide various encoded representations.
2.1.7. Encoded Stream
A stream of time synchronized encoded media that can be independently
decoded.
Characteristics:
o Due to temporal dependencies, an Encoded Stream may have
limitations in where decoding can be started. These entry points,
for example Intra frames from a video encoder, may require
identification and their generation may be event based or
configured to occur periodically.
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2.1.8. Dependent Stream
A stream of time synchronized encoded media fragments that are
dependent on one or more Encoded Streams (Section 2.1.7) and zero or
more Dependent Streams to be possible to decode.
Characteristics:
o Each Dependent Stream has a set of dependencies. These
dependencies must be understood by the parties in a multi-media
session that intend to use a Dependent Stream.
2.1.9. Media Packetizer
The transformation of taking one or more Encoded (Section 2.1.7) or
Dependent Stream (Section 2.1.8) and put their content into one or
more sequences of packets, normally RTP packets, and output Source
RTP Streams (Section 2.1.10). This step includes both generating RTP
payloads as well as RTP packets.
The Media Packetizer can use multiple inputs when producing a single
RTP Stream. One such example is SST packetization when using SVC
(Section 3.5).
The Media Packetizer can also produce multiple RTP Streams, for
example when Encoded and/or Dependent Streams are distributed over
multiple RTP Streams. One example of this is MST packetization when
using SVC (Section 3.5).
Characteristics:
o The Media Packetizer will select which Synchronization source(s)
(SSRC) [RFC3550] in which RTP sessions that are used.
o Media Packetizer can combine multiple Encoded or Dependent Streams
into one or more RTP Streams.
2.1.10. RTP Stream
A stream of RTP packets containing media data, source or redundant.
The RTP Stream is identified by an SSRC belonging to a particular RTP
session. The RTP session is identified as discussed in
Section 2.2.2.
A Source RTP Stream is a RTP Stream containing at least some content
from an Encoded Stream. Source material is any media material that
is produced for transport over RTP without any additional redundancy
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applied to cope with network transport losses. Compare this with the
Redundancy RTP Stream (Section 2.1.12).
Characteristics:
o Each RTP Stream is identified by a unique Synchronization source
(SSRC) [RFC3550] that is carried in every RTP and RTP Control
Protocol (RTCP) packet header in a specific RTP session context.
o At any given point in time, a RTP Stream can have one and only one
SSRC. SSRC collision and clock rate change [RFC7160] are examples
of valid reasons to change SSRC for a RTP Stream, since the RTP
Stream itself is not changed in any significant way, only the
identifying SSRC number.
o Each RTP Stream defines a unique RTP sequence numbering and timing
space.
o Several RTP Streams may map to a single Media Source via the
source transformations.
o Several RTP Streams can be carried over a single RTP Session.
2.1.11. Media Redundancy
Media redundancy is a transformation that generates redundant or
repair packets sent out as a Redundancy RTP Stream to mitigate
network transport impairments, like packet loss and delay.
The Media Redundancy exists in many flavors; they may be generating
independent Repair Streams that are used in addition to the Source
Stream (RTP Retransmission [RFC4588] and some FEC [RFC5109]), they
may generate a new Source Stream by combining redundancy information
with source information (Using XOR FEC [RFC5109] as a redundancy
payload [RFC2198]), or completely replace the source information with
only redundancy packets.
2.1.12. Redundancy RTP Stream
A RTP Stream (Section 2.1.10) that contains no original source data,
only redundant data that may be combined with one or more Received
RTP Stream (Section 2.1.19) to produce Repaired RTP Streams
(Section 2.1.22).
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2.1.13. Media Transport
A Media Transport defines the transformation that the RTP Streams
(Section 2.1.10) are subjected to by the end-to-end transport from
one RTP sender to one specific RTP receiver (an RTP session may
contain multiple RTP receivers per sender). Each Media Transport is
defined by a transport association that is identified by a 5-tuple
(source address, source port, destination address, destination port,
transport protocol). Each transport association normally contains
only a single RTP session, although a proposal exists for sending
multiple RTP sessions over one transport association
[I-D.westerlund-avtcore-transport-multiplexing].
Characteristics:
o Media Transport transmits RTP Streams of RTP Packets from a source
transport address to a destination transport address.
The Media Transport concept sometimes needs to be decomposed into
more steps to enable discussion of what a sender emits that gets
transformed by the network before it is received by the receiver.
Thus we provide also this Media Transport decomposition (Figure 5).
RTP Stream
|
V
+--------------------------+
| Media Transport Sender |
+--------------------------+
|
Sent RTP Stream
V
+--------------------------+
| Network Transport |
+--------------------------+
|
Transported RTP Stream
V
+--------------------------+
| Media Transport Receiver |
+--------------------------+
|
V
Received RTP Stream
Figure 5: Decomposition of Media Transport
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2.1.14. Media Transport Sender
The first transformation within the Media Transport (Section 2.1.13)
is the Media Transport Sender, where the sending End-Point
(Section 2.2.1) takes a RTP Stream and emits the packets onto the
network using the transport association established for this Media
Transport thus creating a Sent RTP Stream (Section 2.1.15). In this
process it transforms the RTP Stream in several ways. First, it
gains the necessary protocol headers for the transport association,
for example IP and UDP headers, thus forming IP/UDP/RTP packets. In
addition, the Media Transport Sender may queue, pace or otherwise
affect how the packets are emitted onto the network. Thus adding
delay, jitter and inter packet spacings that characterize the Sent
RTP Stream.
2.1.15. Sent RTP Stream
The Sent RTP Stream is the RTP Stream as entering the first hop of
the network path to its destination. The Sent RTP Stream is
identified using network transport addresses, like for IP/UDP the
5-tuple (source IP address, source port, destination IP address,
destination port, and protocol (UDP)).
2.1.16. Network Transport
Network Transport is the transformation that the Sent RTP Stream
(Section 2.1.15) is subjected to by traveling from the source to the
destination through the network. These transformations include, loss
of some packets, varying delay on a per packet basis, packet
duplication, and packet header or data corruption. These
transformations produces a Transported RTP Stream (Section 2.1.17) at
the exit of the network path.
2.1.17. Transported RTP Stream
The RTP Stream that is emitted out of the network path at the
destination, subjected to the Network Transport's transformation
(Section 2.1.16).
2.1.18. Media Transport Receiver
The receiver End-Point's (Section 2.2.1) transformation of the
Transported RTP Stream (Section 2.1.17) by its reception process that
result in the Received RTP Stream (Section 2.1.19). This
transformation includes transport checksums being verified and if
non-matching, causing discarding of the corrupted packet. Other
transformations can include delay variations in receiving a packet on
the network interface and providing it to the application.
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2.1.19. Received RTP Stream
The RTP Stream (Section 2.1.10) resulting from the Media Transport's
transformation, i.e. subjected to packet loss, packet corruption,
packet duplication and varying transmission delay from sender to
receiver.
2.1.20. Received Redundancy RTP Stream
The Redundancy RTP Stream (Section 2.1.12) resulting from the Media
Transport transformation, i.e. subjected to packet loss, packet
corruption, and varying transmission delay from sender to receiver.
2.1.21. Media Repair
A Transformation that takes as input one or more Source RTP Streams
(Section 2.1.10) as well as Redundancy RTP Streams (Section 2.1.12)
and attempts to combine them to counter the transformations
introduced by the Media Transport (Section 2.1.13) to minimize the
difference between the Source Stream (Section 2.1.5) and the Received
Source Stream (Section 2.1.26) after Media Decoder (Section 2.1.25).
The output is a Repaired RTP Stream (Section 2.1.22).
2.1.22. Repaired RTP Stream
A Received RTP Stream (Section 2.1.19) for which Received Redundancy
RTP Stream (Section 2.1.20) information has been used to try to re-
create the RTP Stream (Section 2.1.10) as it was before Media
Transport (Section 2.1.13).
2.1.23. Media Depacketizer
A Media Depacketizer takes one or more RTP Streams (Section 2.1.10)
and depacketizes them and attempts to reconstitute the Encoded
Streams (Section 2.1.7) or Dependent Streams (Section 2.1.8) present
in those RTP Streams.
It should be noted that in practical implementations, the Media
Depacketizer and the Media Decoder may be tightly coupled and share
information to improve or optimize the overall decoding process in
various ways. It is however not expected that there would be any
benefit in defining a taxonomy for those detailed (and likely very
implementation-dependent) steps.
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2.1.24. Received Encoded Stream
The received version of an Encoded Stream (Section 2.1.7).
2.1.25. Media Decoder
A Media Decoder is a transformation that is responsible for decoding
Encoded Streams (Section 2.1.7) and any Dependent Streams
(Section 2.1.8) into a Source Stream (Section 2.1.5).
It should be noted that in practical implementations, the Media
Decoder and the Media Depacketizer may be tightly coupled and share
information to improve or optimize the overall decoding process in
various ways. It is however not expected that there would be any
benefit in defining a taxonomy for those detailed (and likely very
implementation-dependent) steps.
Characteristics:
o A Media Decoder is the entity that will have to deal with any
errors in the encoded streams that resulted from corruptions or
failures to repair packet losses. This as a media decoder
generally is forced to produce some output periodically. It thus
commonly includes concealment methods.
2.1.26. Received Source Stream
The received version of a Source Stream (Section 2.1.5).
2.1.27. Media Sink
The Media Sink receives a Source Stream (Section 2.1.5) that
contains, usually periodically, sampled media data together with
associated synchronization information. Depending on application,
this Source Stream then needs to be transformed into a Raw Stream
(Section 2.1.3) that is sent in synchronization with the output from
other Media Sinks to a Media Render (Section 2.1.29). The media sink
may also be connected with a Media Source (Section 2.1.4) and be used
as part of a conceptual Media Source.
Characteristics:
o The Media Sink can further transform the Source Stream into a
representation that is suitable for rendering on the Media Render
as defined by the application or system-wide configuration. This
include sample scaling, level adjustments etc.
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2.1.28. Received Raw Stream
The received version of a Raw Stream (Section 2.1.3).
2.1.29. Media Render
A Media Render takes a Raw Stream (Section 2.1.3) and converts it
into Physical Stimulus (Section 2.1.1) that a human user can
perceive. Examples of such devices are screens, D/A converters
connected to amplifiers and loudspeakers.
Characteristics:
o An End Point can potentially have multiple Media Renders for each
media type.
2.2. Communication Entities
This section contains concept for entities involved in the
communication.
+----------------------------------------------------------+
| Communication Session |
| |
| +----------------+ +----------------+ |
| | Participant A | +------------+ | Participant B | |
| | | | Multimedia | | | |
| | +-------------+|<=>| Session |<=>|+-------------+ | |
| | | End Point A || | | || End Point B | | |
| | | || +------------+ || | | |
| | | +-----------++--------------------++-----------+ | | |
| | | | RTP Session| | | | | |
| | | | Audio |--Media Transport-->| | | | |
| | | | |<--Media Transport--| | | | |
| | | +-----------++--------------------++-----------+ | | |
| | | || || | | |
| | | +-----------++--------------------++-----------+ | | |
| | | | RTP Session| | | | | |
| | | | Video |--Media Transport-->| | | | |
| | | | |<--Media Transport--| | | | |
| | | +-----------++--------------------++-----------+ | | |
| | +-------------+| |+-------------+ | |
| +----------------+ +----------------+ |
+----------------------------------------------------------+
Figure 6: Example Point to Point Communication Session with two RTP
Sessions
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The figure above shows a high-level example representation of a very
basic point-to-point Communication Session between Participants A and
B. It uses two different audio and video RTP Sessions between A's
and B's End Points, using separate Media Transports for those RTP
Sessions. The Multimedia Session shared by the participants can for
example be established using SIP (i.e., there is a SIP Dialog between
A and B). The terms used in that figure are further elaborated in
the sub-sections below.
2.2.1. End Point
Editor's note: Consider if a single word, "Endpoint", is
preferable
A single addressable entity sending or receiving RTP packets. It may
be decomposed into several functional blocks, but as long as it
behaves as a single RTP stack entity it is classified as a single
"End Point".
Characteristics:
o End Points can be identified in several different ways. While
RTCP Canonical Names (CNAMEs) [RFC3550] provide a globally unique
and stable identification mechanism for the duration of the
Communication Session (see Section 2.2.5), their validity applies
exclusively within a Synchronization Context (Section 3.1). Thus
one End Point can handle multiple CNAMEs, each of which can be
shared among a set of End Points belonging to the same Participant
(Section 2.2.3). Therefore, mechanisms outside the scope of RTP,
such as application defined mechanisms, must be used to ensure End
Point identification when outside this Synchronization Context.
o An End Point can be associated with at most one Participant
(Section 2.2.3) at any single point in time.
o In some contexts, an End Point would typically correspond to a
single "host".
2.2.2. RTP Session
Editor's note: Re-consider if this is really a Communication
Entity, or if it is rather an existing concept that should be
described in Section 4.
An RTP session is an association among a group of participants
communicating with RTP. It is a group communications channel which
can potentially carry a number of RTP Streams. Within an RTP
session, every participant can find meta-data and control information
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(over RTCP) about all the RTP Streams in the RTP session. The
bandwidth of the RTCP control channel is shared between all
participants within an RTP Session.
Characteristics:
o Typically, an RTP Session can carry one ore more RTP Streams.
o An RTP Session shares a single SSRC space as defined in RFC3550
[RFC3550]. That is, the End Points participating in an RTP
Session can see an SSRC identifier transmitted by any of the other
End Points. An End Point can receive an SSRC either as SSRC or as
a Contributing source (CSRC) in RTP and RTCP packets, as defined
by the endpoints' network interconnection topology.
o An RTP Session uses at least two Media Transports
(Section 2.1.13), one for sending and one for receiving.
Commonly, the receiving one is the reverse direction of the same
one as used for sending. An RTP Session may use many Media
Transports and these define the session's network interconnection
topology. A single Media Transport can normally not transport
more than one RTP Session, unless a solution for multiplexing
multiple RTP sessions over a single Media Transport is used. One
example of such a scheme is Multiple RTP Sessions on a Single
Lower-Layer Transport
[I-D.westerlund-avtcore-transport-multiplexing].
o Multiple RTP Sessions can be related.
2.2.3. Participant
A Participant is an entity reachable by a single signaling address,
and is thus related more to the signaling context than to the media
context.
Characteristics:
o A single signaling-addressable entity, using an application-
specific signaling address space, for example a SIP URI.
o A Participant can have several Multimedia Sessions
(Section 2.2.4).
o A Participant can have several associated End Points
(Section 2.2.1).
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2.2.4. Multimedia Session
A multimedia session is an association among a group of participants
engaged in the communication via one or more RTP Sessions
(Section 2.2.2). It defines logical relationships among Media
Sources (Section 2.1.4) that appear in multiple RTP Sessions.
Characteristics:
o A Multimedia Session can be composed of several parallel RTP
Sessions with potentially multiple RTP Streams per RTP Session.
o Each participant in a Multimedia Session can have a multitude of
Media Captures and Media Rendering devices.
o A single Multimedia Session can contain media from one or more
Synchronization Contexts (Section 3.1). An example of that is a
Multimedia Session containing one set of audio and video for
communication purposes belonging to one Synchronization Context,
and another set of audio and video for presentation purposes (like
playing a video file) with a separate Synchronization Context that
has no strong timing relationship and need not be strictly
synchronized with the audio and video used for communication.
2.2.5. Communication Session
A Communication Session is an association among group of participants
communicating with each other via a set of Multimedia Sessions.
Characteristics:
o Each participant in a Communication Session is identified via an
application-specific signaling address.
o A Communication Session is composed of at least one Multimedia
Session per participant, involving one or more parallel RTP
Sessions with potentially multiple RTP Streams per RTP Session.
For example, in a full mesh communication, the Communication Session
consists of a set of separate Multimedia Sessions between each pair
of Participants. Another example is a centralized conference, where
the Communication Session consists of a set of Multimedia Sessions
between each Participant and the conference handler.
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3. Relations at Different Levels
This section uses the concepts from previous section and look at
different types of relationships among them. These relationships
occur at different levels and for different purposes. The section is
organized such as to look at the level where a relation is required.
The reason for the relationship may exist at another step in the
media handling chain. For example, using Simulcast (discussed in
Section 3.7) needs to determine relations at RTP Stream level,
however the reason to relate RTP Streams is that multiple Media
Encoders use the same Media Source, i.e. to be able to identify a
common Media Source.
Media Sources (Section 2.1.4) are commonly grouped and related to an
End Point (Section 2.2.1) or a Participant (Section 2.2.3). This
occurs for several reasons; both due to application logic as well as
for media handling purposes.
At RTP Packetization time, there exists a possibility for a number of
different types of relationships between Encoded Streams
(Section 2.1.7), Dependent Streams (Section 2.1.8) and RTP Streams
(Section 2.1.10). These are caused by grouping together or
distributing these different types of streams into RTP Streams.
The resulting RTP Streams will thus also have relations. This is a
common relation to handle in RTP due to that RTP Streams are separate
and have their own SSRC, implying independent sequence numbers and
timestamp spaces. The underlying reasons for the RTP Stream
relationships are different, as can be seen in the sub-sections
below.
RTP Streams may be protected by Redundancy RTP Streams during
transport. Several approaches listed below can be used to create
Redundancy RTP Streams;
o Duplication of the original RTP Stream
o Duplication of the original RTP Stream with a time offset,
o Forward Error Correction (FEC) techniques, and
o Retransmission of lost packets (either globally or selectively).
The different RTP Streams can be transported within the same RTP
Session or in different RTP Sessions to accomplish different
transport goals. This explicit separation of RTP Streams is further
discussed in Section 3.13.
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3.1. Synchronization Context
A Synchronization Context defines a requirement on a strong timing
relationship between the Media Sources, typically requiring alignment
of clock sources. Such relationship can be identified in multiple
ways as listed below. A single Media Source can only belong to a
single Synchronization Context, since it is assumed that a single
Media Source can only have a single media clock and requiring
alignment to several Synchronization Contexts (and thus reference
clocks) will effectively merge those into a single Synchronization
Context.
3.1.1. RTCP CNAME
RFC3550 [RFC3550] describes Inter-media synchronization between RTP
Sessions based on RTCP CNAME, RTP and Network Time Protocol (NTP)
[RFC5905] formatted timestamps of a reference clock. As indicated in
[I-D.ietf-avtcore-clksrc], despite using NTP format timestamps, it is
not required that the clock be synchronized to an NTP source.
3.1.2. Clock Source Signaling
[I-D.ietf-avtcore-clksrc] provides a mechanism to signal the clock
source in SDP both for the reference clock as well as the media
clock, thus allowing a Synchronization Context to be defined beyond
the one defined by the usage of CNAME source descriptions.
3.1.3. Implicitly via RtcMediaStream
The WebRTC WG defines "RtcMediaStream" with one or more
"RtcMediaStreamTracks". All tracks in a "RtcMediaStream" are
intended to be possible to synchronize when rendered.
3.1.4. Explicitly via SDP Mechanisms
RFC5888 [RFC5888] defines m=line grouping mechanism called "Lip
Synchronization (LS)" for establishing the synchronization
requirement across m=lines when they map to individual sources.
RFC5576 [RFC5576] extends the above mechanism when multiple media
sources are described by a single m=line.
3.2. End Point
Some applications requires knowledge of what Media Sources originate
from a particular End Point (Section 2.2.1). This can include such
decisions as packet routing between parts of the topology, knowing
the End Point origin of the RTP Streams.
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In RTP, this identification has been overloaded with the
Synchronization Context (Section 3.1) through the usage of the RTCP
source description CNAME (Section 3.1.1) item. This works for some
usages, but sometimes it breaks down. For example, if an End Point
has two sets of Media Sources that have different Synchronization
Contexts, like the audio and video of the human participant as well
as a set of Media Sources of audio and video for a shared movie.
Thus, an End Point may have multiple CNAMEs. The CNAMEs or the Media
Sources themselves can be related to the End Point.
3.3. Participant
In communication scenarios, it is commonly needed to know which Media
Sources that originate from which Participant (Section 2.2.3). Thus
enabling the application to for example display Participant Identity
information correctly associated with the Media Sources. This
association is currently handled through the signaling solution to
point at a specific Multimedia Session where the Media Sources may be
explicitly or implicitly tied to a particular End Point.
Participant information becomes more problematic due to Media Sources
that are generated through mixing or other conceptual processing of
Raw Streams or Source Streams that originate from different
Participants. This type of Media Sources can thus have a dynamically
varying set of origins and Participants. RTP contains the concept of
Contributing Sources (CSRC) that carries such information about the
previous step origin of the included media content on RTP level.
3.4. RtcMediaStream
An RtcMediaStream in WebRTC is an explicit grouping of a set of Media
Sources (RtcMediaStreamTracks) that share a common identifier and a
single Synchronization Context (Section 3.1).
3.5. Single- and Multi-Session Transmission of SVC
Scalable Video Coding [RFC6190] has a mode of operation called Single
Session Transmission (SST), where Encoded Streams and Dependent
Streams from the SVC Media Encoder are sent in a single RTP Session
(Section 2.2.2) using the SVC RTP Payload format. There is another
mode of operation where Encoded Streams and Dependent Streams are
distributed across multiple RTP Sessions, called Multi-Session
Transmission (MST). SST denotes one or more RTP Streams (SSRC) per
Media Source in a single RTP Session. MST denotes one or more RTP
Streams (SSRC) per Media Source in each of multiple RTP Sessions.
This is not always clear from the SVC payload format text [RFC6190],
but is what existing deployments of that RFC have implemented.
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To elaborate, what could be called SST-SingleStream (SST-SS) uses a
single RTP Stream in a single RTP Session to send all Encoded and
Dependent Streams from a single Media Source. Similarly, SST-
MultiStream (SST-MS) uses a single RTP Stream per Media Source in a
single RTP Session to send the Encoded and Dependent Streams. MST-SS
uses a single RTP Stream in each of multiple RTP Sessions, where each
RTP Stream can originate from any one of possibly multiple Media
Sources. Finally, MST-MS uses multiple RTP Streams in each of the
multiple RTP Sessions, where each RTP Stream can originate from any
one of possibly multiple Media Sources. This is summarized below:
+--------------------------+------------------+---------------------+
| RTP Streams per Media | Single RTP | Multiple RTP |
| Source | Session | Sessions |
+--------------------------+------------------+---------------------+
| Single | SST-SS | MST-SS |
| Multiple | SST-MS | MST-MS |
+--------------------------+------------------+---------------------+
Table 1: SST / MST Summary
3.6. Multi-Channel Audio
There exist a number of RTP payload formats that can carry multi-
channel audio, despite the codec being a mono encoder. Multi-channel
audio can be viewed as multiple Media Sources sharing a common
Synchronization Context. These are independently encoded by a Media
Encoder and the different Encoded Streams are then packetized
together in a time synchronized way into a single Source RTP Stream
using the used codec's RTP Payload format. Example of such codecs
are, PCMA and PCMU [RFC3551], AMR [RFC4867], and G.719 [RFC5404].
3.7. Simulcast
A Media Source represented as multiple independent Encoded Streams
constitutes a simulcast of that Media Source. Figure 7 below
represents an example of a Media Source that is encoded into three
separate and different Simulcast streams, that are in turn sent on
the same Media Transport flow. When using Simulcast, the RTP Streams
may be sharing RTP Session and Media Transport, or be separated on
different RTP Sessions and Media Transports, or be any combination of
these two. It is other considerations that affect which usage is
desirable, as discussed in Section 3.13.
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+----------------+
| Media Source |
+----------------+
Source Stream |
+----------------------+----------------------+
| | |
V V V
+------------------+ +------------------+ +------------------+
| Media Encoder | | Media Encoder | | Media Encoder |
+------------------+ +------------------+ +------------------+
| Encoded | Encoded | Encoded
| Stream | Stream | Stream
V V V
+------------------+ +------------------+ +------------------+
| Media Packetizer | | Media Packetizer | | Media Packetizer |
+------------------+ +------------------+ +------------------+
| Source | Source | Source
| RTP | RTP | RTP
| Stream | Stream | Stream
+-----------------+ | +-----------------+
| | |
V V V
+-------------------+
| Media Transport |
+-------------------+
Figure 7: Example of Media Source Simulcast
The simulcast relation between the RTP Streams is the common Media
Source. In addition, to be able to identify the common Media Source,
a receiver of the RTP Stream may need to know which configuration or
encoding goals that lay behind the produced Encoded Stream and its
properties. This to enable selection of the stream that is most
useful in the application at that moment.
3.8. Layered Multi-Stream
Layered Multi-Stream (LMS) is a mechanism by which different portions
of a layered encoding of a Source Stream are sent using separate RTP
Streams (sometimes in separate RTP Sessions). LMSs are useful for
receiver control of layered media.
A Media Source represented as an Encoded Stream and multiple
Dependent Streams constitutes a Media Source that has layered
dependencies. The figure below represents an example of a Media
Source that is encoded into three dependent layers, where two layers
are sent on the same Media Transport using different RTP Streams,
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i.e. SSRCs, and the third layer is sent on a separate Media
Transport, i.e. a different RTP Session.
+----------------+
| Media Source |
+----------------+
|
|
V
+---------------------------------------------------------+
| Media Encoder |
+---------------------------------------------------------+
| | |
Encoded Stream Dependent Stream Dependent Stream
| | |
V V V
+----------------+ +----------------+ +----------------+
|Media Packetizer| |Media Packetizer| |Media Packetizer|
+----------------+ +----------------+ +----------------+
| | |
RTP Stream RTP Stream RTP Stream
| | |
+------+ +------+ |
| | |
V V V
+-----------------+ +-----------------+
| Media Transport | | Media Transport |
+-----------------+ +-----------------+
Figure 8: Example of Media Source Layered Dependency
As an example, the SVC MST (Section 3.5) relation needs to identify
the common Media Encoder origin for the Encoded and Dependent
Streams. The SVC RTP Payload RFC is not particularly explicit about
how this relation is to be implemented. When using different RTP
Sessions, thus different Media Transports, and as long as there is
only one RTP Stream per Media Encoder and a single Media Source in
each RTP Session (MST-SS (Section 3.5)), common SSRC and CNAMEs can
be used to identify the common Media Source. When multiple RTP
Streams are sent from one Media Encoder in the same RTP Session (SST-
MS), then CNAME is the only currently specified RTP identifier that
can be used. In cases where multiple Media Encoders use multiple
Media Sources sharing Synchronization Context, and thus having a
common CNAME, additional heuristics need to be applied to create the
MST relationship between the RTP Streams.
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3.9. RTP Stream Duplication
RTP Stream Duplication [RFC7198], using the same or different Media
Transports, and optionally also delaying the duplicate [RFC7197],
offers a simple way to protect media flows from packet loss in some
cases. It is a specific type of redundancy and all but one Source
RTP Stream (Section 2.1.10) are effectively Redundancy RTP Streams
(Section 2.1.12), but since both Source and Redundant RTP Streams are
the same it does not matter which is which. This can also be seen as
a specific type of Simulcast (Section 3.7) that transmits the same
Encoded Stream (Section 2.1.7) multiple times.
+----------------+
| Media Source |
+----------------+
Source Stream |
V
+----------------+
| Media Encoder |
+----------------+
Encoded Stream |
+-----------+-----------+
| |
V V
+------------------+ +------------------+
| Media Packetizer | | Media Packetizer |
+------------------+ +------------------+
Source | RTP Stream Source | RTP Stream
| V
| +-------------+
| | Delay (opt) |
| +-------------+
| |
+-----------+-----------+
|
V
+-------------------+
| Media Transport |
+-------------------+
Figure 9: Example of RTP Stream Duplication
3.10. Redundancy Format
The RTP Payload for Redundant Audio Data [RFC2198] defines how one
can transport redundant audio data together with primary data in the
same RTP payload. The redundant data can be a time delayed version
of the primary or another time delayed Encoded Stream using a
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different Media Encoder to encode the same Media Source as the
primary, as depicted below in Figure 10.
+--------------------+
| Media Source |
+--------------------+
|
Source Stream
|
+------------------------+
| |
V V
+--------------------+ +--------------------+
| Media Encoder | | Media Encoder |
+--------------------+ +--------------------+
| |
| +------------+
Encoded Stream | Time Delay |
| +------------+
| |
| +------------------+
V V
+--------------------+
| Media Packetizer |
+--------------------+
|
V
RTP Stream
Figure 10: Concept for usage of Audio Redundancy with different Media
Encoders
The Redundancy format is thus providing the necessary meta
information to correctly relate different parts of the same Encoded
Stream, or in the case depicted above (Figure 10) relate the Received
Source Stream fragments coming out of different Media Decoders to be
able to combine them together into a less erroneous Source Stream.
3.11. RTP Retransmission
The figure below (Figure 11) represents an example where a Media
Source's Source RTP Stream is protected by a retransmission (RTX)
flow [RFC4588]. In this example the Source RTP Stream and the
Redundancy RTP Stream share the same Media Transport.
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+--------------------+
| Media Source |
+--------------------+
|
V
+--------------------+
| Media Encoder |
+--------------------+
| Retransmission
Encoded Stream +--------+ +---- Request
V | V V
+--------------------+ | +--------------------+
| Media Packetizer | | | RTP Retransmission |
+--------------------+ | +--------------------+
| | |
+------------+ Redundancy RTP Stream
Source RTP Stream |
| |
+---------+ +---------+
| |
V V
+-----------------+
| Media Transport |
+-----------------+
Figure 11: Example of Media Source Retransmission Flows
The RTP Retransmission example (Figure 11) helps illustrate that this
mechanism works purely on the Source RTP Stream. The RTP
Retransmission transform buffers the sent Source RTP Stream and upon
requests emits a retransmitted packet with some extra payload header
as a Redundancy RTP Stream. The RTP Retransmission mechanism
[RFC4588] is specified so that there is a one to one relation between
the Source RTP Stream and the Redundancy RTP Stream. Thus a
Redundancy RTP Stream needs to be associated with its Source RTP
Stream upon being received. This is done based on CNAME selectors
and heuristics to match requested packets for a given Source RTP
Stream with the original sequence number in the payload of any new
Redundancy RTP Stream using the RTX payload format. In cases where
the Redundancy RTP Stream is sent in a separate RTP Session from the
Source RTP Stream, these sessions are related, e.g. using the SDP
Media Grouping's [RFC5888] FID semantics.
3.12. Forward Error Correction
The figure below (Figure 12) represents an example where two Media
Sources' Source RTP Streams are protected by FEC. Source RTP Stream
A has a Media Redundancy transformation in FEC Encoder 1. This
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produces a Redundancy RTP Stream 1, that is only related to Source
RTP Stream A. The FEC Encoder 2, however takes two Source RTP
Streams (A and B) and produces a Redundancy RTP Stream 2 that
protects them together, i.e. Redundancy RTP Stream 2 relate to two
Source RTP Streams (a FEC group). FEC decoding, when needed due to
packet loss or packet corruption at the receiver, requires knowledge
about which Source RTP Streams that the FEC encoding was based on.
In Figure 12 all RTP Streams are sent on the same Media Transport.
This is however not the only possible choice. Numerous combinations
exist for spreading these RTP Streams over different Media Transports
to achieve the communication application's goal.
+--------------------+ +--------------------+
| Media Source A | | Media Source B |
+--------------------+ +--------------------+
| |
V V
+--------------------+ +--------------------+
| Media Encoder A | | Media Encoder B |
+--------------------+ +--------------------+
| |
Encoded Stream Encoded Stream
V V
+--------------------+ +--------------------+
| Media Packetizer A | | Media Packetizer B |
+--------------------+ +--------------------+
| |
Source RTP Stream A Source RTP Stream B
| |
+-----+---------+-------------+ +---+---+
| V V V |
| +---------------+ +---------------+ |
| | FEC Encoder 1 | | FEC Encoder 2 | |
| +---------------+ +---------------+ |
| Redundancy | Redundancy | |
| RTP Stream 1 | RTP Stream 2 | |
V V V V
+----------------------------------------------------------+
| Media Transport |
+----------------------------------------------------------+
Figure 12: Example of FEC Flows
As FEC Encoding exists in various forms, the methods for relating FEC
Redundancy RTP Streams with its source information in Source RTP
Streams are many. The XOR based RTP FEC Payload format [RFC5109] is
defined in such a way that a Redundancy RTP Stream has a one to one
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relation with a Source RTP Stream. In fact, the RFC requires the
Redundancy RTP Stream to use the same SSRC as the Source RTP Stream.
This requires to either use a separate RTP session or to use the
Redundancy RTP Payload format [RFC2198]. The underlying relation
requirement for this FEC format and a particular Redundancy RTP
Stream is to know the related Source RTP Stream, including its SSRC.
3.13. RTP Stream Separation
RTP Streams can be separated exclusively based on their SSRCs, at the
RTP Session level, or at the Multi-Media Session level.
When the RTP Streams that have a relationship are all sent in the
same RTP Session and are uniquely identified based on their SSRC
only, it is termed an SSRC-Only Based Separation. Such streams can
be related via RTCP CNAME to identify that the streams belong to the
same End Point. [RFC5576]-based approaches, when used, can
explicitly relate various such RTP Streams.
On the other hand, when RTP Streams that are related but are sent in
the context of different RTP Sessions to achieve separation, it is
known as RTP Session-based separation. This is commonly used when
the different RTP Streams are intended for different Media
Transports.
Several mechanisms that use RTP Session-based separation rely on it
to enable an implicit grouping mechanism expressing the relationship.
The solutions have been based on using the same SSRC value in the
different RTP Sessions to implicitly indicate their relation. That
way, no explicit RTP level mechanism has been needed, only signaling
level relations have been established using semantics from Grouping
of Media lines framework [RFC5888]. Examples of this are RTP
Retransmission [RFC4588], SVC Multi-Session Transmission [RFC6190]
and XOR Based FEC [RFC5109]. RTCP CNAME explicitly relates RTP
Streams across different RTP Sessions, as explained in the previous
section. Such a relationship can be used to perform inter-media
synchronization.
RTP Streams that are related and need to be associated can be part of
different Multimedia Sessions, rather than just different RTP
sessions within the same Multimedia Session context. This puts
further demand on the scope of the mechanism(s) and its handling of
identifiers used for expressing the relationships.
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3.14. Multiple RTP Sessions over one Media Transport
[I-D.westerlund-avtcore-transport-multiplexing] describes a mechanism
that allow several RTP Sessions to be carried over a single
underlying Media Transport. The main reasons for doing this are
related to the impact of using one or more Media Transports. Thus
using a common network path or potentially have different ones.
There is reduced need for NAT/FW traversal resources and no need for
flow based QoS.
However, Multiple RTP Sessions over one Media Transport makes it
clear that a single Media Transport 5-tuple is not sufficient to
express which RTP Session context a particular RTP Stream exists in.
Complexities in the relationship between Media Transports and RTP
Session already exist as one RTP Session contains multiple Media
Transports, e.g. even a Peer-to-Peer RTP Session with RTP/RTCP
Multiplexing requires two Media Transports, one in each direction.
The relationship between Media Transports and RTP Sessions as well as
additional levels of identifiers need to be considered in both
signaling design and when defining terminology.
4. Mapping from Existing Terms
This section describes a selected set of terms from some relevant
IETF RFC and Internet Drafts (at the time of writing), using the
concepts from previous sections.
4.1. Audio Capture
Telepresence specifications from CLUE WG uses this term to describe
an audio Media Source (Section 2.1.4).
4.2. Capture Device
Telepresence specifications from CLUE WG use this term to identify a
physical entity performing a Media Capture (Section 2.1.2)
transformation.
4.3. Capture Encoding
Telepresence specifications from CLUE WG uses this term to describe
an Encoded Stream (Section 2.1.7) related to CLUE specific semantic
information.
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4.4. Capture Scene
Telepresence specifications from CLUE WG uses this term to describe a
set of spatially related Media Sources (Section 2.1.4).
4.5. Endpoint
Telepresence specifications from CLUE WG use this term to describe
exactly one Participant (Section 2.2.3) and one or more End Points
(Section 2.2.1).
4.6. Individual Encoding
Telepresence specifications from CLUE WG use this term to describe
the configuration information needed to perform a Media Encoder
(Section 2.1.6) transformation.
4.7. Multipoint Control Unit (MCU)
This term is commonly used to describe the central node in any type
of star topology [I-D.ietf-avtcore-rtp-topologies-update] conference.
It describes a device that includes one Participant (Section 2.2.3)
(usually corresponding to a so-called conference focus) and one or
more related End Points (Section 2.2.1) (sometimes one or more per
conference participant).
4.8. Media Capture
Telepresence specifications from CLUE WG uses this term to describe
either a Media Capture (Section 2.1.2) or a Media Source
(Section 2.1.4), depending on in which context the term is used.
4.9. Media Consumer
Telepresence specifications from CLUE WG use this term to describe
the media receiving part of an End Point (Section 2.2.1).
4.10. Media Description
A single Source Description Protocol (SDP) [RFC4566] media
description (or media block; an m-line and all subsequent lines until
the next m-line or the end of the SDP) describes part of the
necessary configuration and identification information needed for a
Media Encoder transformation, as well as the necessary configuration
and identification information for the Media Decoder to be able to
correctly interpret a received RTP Stream.
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A Media Description typically relates to a single Media Source. This
is for example an explicit restriction in WebRTC. However, nothing
prevents that the same Media Description (and same RTP Session) is
re-used for multiple Media Sources
[I-D.ietf-avtcore-rtp-multi-stream]. It can thus describe properties
of one or more RTP Streams, and can also describe properties valid
for an entire RTP Session (via [RFC5576] mechanisms, for example).
4.11. Media Provider
Telepresence specifications from CLUE WG use this term to describe
the media sending part of an End Point (Section 2.2.1).
4.12. Media Stream
RTP [RFC3550] uses media stream, audio stream, video stream, and
stream of (RTP) packets interchangeably, which are all RTP Streams.
4.13. Multimedia Session
SDP [RFC4566] defines a multimedia session as a set of multimedia
senders and receivers and the data streams flowing from senders to
receivers, which would correspond to a set of End Points and the RTP
Streams that flow between them. In this memo, Multimedia Session
also assumes those End Points belong to a set of Participants that
are engaged in communication via a set of related RTP Streams.
RTP [RFC3550] defines a multimedia session as a set of concurrent RTP
Sessions among a common group of participants. For example, a video
conference may contain an audio RTP Session and a video RTP Session.
This would correspond to a group of Participants (each using one or
more End Points) sharing a set of concurrent RTP Sessions. In this
memo, Multimedia Session also defines those RTP Sessions to have some
relation and be part of a communication among the Participants.
4.14. Recording Device
WebRTC specifications use this term to refer to locally available
entities performing a Media Capture (Section 2.1.2) transformation.
4.15. RtcMediaStream
A WebRTC RtcMediaStreamTrack is a set of Media Sources
(Section 2.1.4) sharing the same Synchronization Context
(Section 3.1).
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4.16. RtcMediaStreamTrack
A WebRTC RtcMediaStreamTrack is a Media Source (Section 2.1.4).
4.17. RTP Sender
RTP [RFC3550] uses this term, which can be seen as the RTP protocol
part of a Media Packetizer (Section 2.1.9).
4.18. RTP Session
Within the context of SDP, a singe m=line can map to a single RTP
Session or multiple m=lines can map to a single RTP Session. The
latter is enabled via multiplexing schemes such as BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation], for example, which allows
mapping of multiple m=lines to a single RTP Session.
Editor's note: Consider if the contents of Section 2.2.2 should be
moved here, or if this section should be kept and refer to the
above.
4.19. SSRC
RTP [RFC3550] defines this as "the source of a stream of RTP
packets", which indicates that an SSRC is not only a unique
identifier for the Encoded Stream (Section 2.1.7) carried in those
packets, but is also effectively used as a term to denote a Media
Packetizer (Section 2.1.9).
4.20. Stream
Telepresence specifications from CLUE WG use this term to describe an
RTP Stream (Section 2.1.10).
4.21. Video Capture
Telepresence specifications from CLUE WG uses this term to describe a
video Media Source (Section 2.1.4).
5. Security Considerations
This document simply tries to clarify the confusion prevalent in RTP
taxonomy because of inconsistent usage by multiple technologies and
protocols making use of the RTP protocol. It does not introduce any
new security considerations beyond those already well documented in
the RTP protocol [RFC3550] and each of the many respective
specifications of the various protocols making use of it.
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Hopefully having a well-defined common terminology and understanding
of the complexities of the RTP architecture will help lead us to
better standards, avoiding security problems.
6. Acknowledgement
This document has many concepts borrowed from several documents such
as WebRTC [I-D.ietf-rtcweb-overview], CLUE [I-D.ietf-clue-framework],
Multiplexing Architecture
[I-D.westerlund-avtcore-transport-multiplexing]. The authors would
like to thank all the authors of each of those documents.
The authors would also like to acknowledge the insights, guidance and
contributions of Magnus Westerlund, Roni Even, Paul Kyzivat, Colin
Perkins, Keith Drage, Harald Alvestrand, and Alex Eleftheriadis.
7. Contributors
Magnus Westerlund has contributed the concept model for the media
chain using transformations and streams model, including rewriting
pre-existing concepts into this model and adding missing concepts.
The first proposal for updating the relationships and the topologies
based on this concept was also performed by Magnus.
8. IANA Considerations
This document makes no request of IANA.
9. Informative References
[I-D.ietf-avtcore-clksrc]
Williams, A., Gross, K., Brandenburg, R., and H. Stokking,
"RTP Clock Source Signalling", draft-ietf-avtcore-
clksrc-11 (work in progress), March 2014.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-04 (work in progress),
May 2014.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-02 (work in progress),
May 2014.
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[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue-
framework-15 (work in progress), May 2014.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-07 (work in progress), April 2014.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-10
(work in progress), June 2014.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
Sessions onto a Single Lower-Layer Transport", draft-
westerlund-avtcore-transport-multiplexing-07 (work in
progress), October 2013.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, April 2007.
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[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for
G.719", RFC 5404, January 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC5905] Mills, D., Martin, J., Burbank, J., and W. Kasch, "Network
Time Protocol Version 4: Protocol and Algorithms
Specification", RFC 5905, June 2010.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
May 2011.
[RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, April 2014.
[RFC7197] Begen, A., Cai, Y., and H. Ou, "Duplication Delay
Attribute in the Session Description Protocol", RFC 7197,
April 2014.
[RFC7198] Begen, A. and C. Perkins, "Duplicating RTP Streams", RFC
7198, April 2014.
Appendix A. Changes From Earlier Versions
NOTE TO RFC EDITOR: Please remove this section prior to publication.
A.1. Modifications Between WG Version -01 and -02
o Major re-structure
o Moved media chain Media Transport detailing up one section level
o Collapsed level 2 sub-sections of section 3 and thus moved level 3
sub-sections up one level, gathering some introductory text into
the beginning of section 3
o Added that not only SSRC collision, but also a clock rate change
[RFC7160] is a valid reason to change SSRC value for an RTP stream
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o Added a sub-section on clock source signaling
o Added a sub-section on RTP stream duplication
o Elaborated a bit in section 2.2.1 on the relation between End
Points, Participants and CNAMEs
o Elaborated a bit in section 2.2.4 on Multimedia Session and
synchronization contexts
o Removed the section on CLUE scenes defining an implicit
synchronization context, since it was incorrect
o Clarified text on SVC SST and MST according to list discussions
o Removed the entire topology section to avoid possible
inconsistencies or duplications with draft-ietf-avtcore-rtp-
topologies-update, but saved one example overview figure of
Communication Entities into that section
o Added a section 4 on mapping from existing terms with one sub-
section per term, mainly by moving text from sections 2 and 3
o Changed all occurrences of Packet Stream to RTP Stream
o Moved all normative references to informative, since this is an
informative document
o Added references to RFC 7160, RFC 7197 and RFC 7198, and removed
unused references
A.2. Modifications Between WG Version -00 and -01
o WG version -00 text is identical to individual draft -03
o Amended description of SVC SST and MST encodings with respect to
concepts defined in this text
o Removed UML as normative reference, since the text no longer uses
any UML notation
o Removed a number of level 4 sections and moved out text to the
level above
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A.3. Modifications Between Version -02 and -03
o Section 4 rewritten (and new communication topologies added) to
reflect the major updates to Sections 1-3
o Section 8 removed (carryover from initial -00 draft)
o General clean up of text, grammar and nits
A.4. Modifications Between Version -01 and -02
o Section 2 rewritten to add both streams and transformations in the
media chain.
o Section 3 rewritten to focus on exposing relationships.
A.5. Modifications Between Version -00 and -01
o Too many to list
o Added new authors
o Updated content organization and presentation
Authors' Addresses
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
Kevin Gross
AVA Networks, LLC
Boulder, CO
US
Email: kevin.gross@avanw.com
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Suhas Nandakumar
Cisco Systems
170 West Tasman Drive
San Jose, CA 95134
US
Email: snandaku@cisco.com
Gonzalo Salgueiro
Cisco Systems
7200-12 Kit Creek Road
Research Triangle Park, NC 27709
US
Email: gsalguei@cisco.com
Bo Burman
Ericsson
Kistavagen 25
SE-164 80 Kista
Sweden
Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com
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