Network Working Group J. Lennox
Internet-Draft Vidyo
Intended status: Informational K. Gross
Expires: May 10, 2014 AVA
S. Nandakumar
G. Salgueiro
Cisco Systems
B. Burman
Ericsson
November 06, 2013
A Taxonomy of Grouping Semantics and Mechanisms for Real-Time Transport
Protocol (RTP) Sources
draft-ietf-avtext-rtp-grouping-taxonomy-00
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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and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on May 10, 2014.
Copyright Notice
Copyright (c) 2013 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 . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Concepts . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Media Chain . . . . . . . . . . . . . . . . . . . . . . . 4
2.1.1. Physical Stimulus . . . . . . . . . . . . . . . . . . 7
2.1.2. Media Capture . . . . . . . . . . . . . . . . . . . . 7
2.1.3. Raw Stream . . . . . . . . . . . . . . . . . . . . . 7
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 . . . . . . . . . . . . . . . . . . 10
2.1.9. Media Packetizer . . . . . . . . . . . . . . . . . . 10
2.1.10. Packet Stream . . . . . . . . . . . . . . . . . . . . 11
2.1.11. Media Redundancy . . . . . . . . . . . . . . . . . . 12
2.1.12. Redundancy Packet Stream . . . . . . . . . . . . . . 12
2.1.13. Media Transport . . . . . . . . . . . . . . . . . . . 12
2.1.14. Received Packet Stream . . . . . . . . . . . . . . . 15
2.1.15. Received Redundandy Packet Stream . . . . . . . . . . 15
2.1.16. Media Repair . . . . . . . . . . . . . . . . . . . . 15
2.1.17. Repaired Packet Stream . . . . . . . . . . . . . . . 15
2.1.18. Media Depacketizer . . . . . . . . . . . . . . . . . 15
2.1.19. Received Encoded Stream . . . . . . . . . . . . . . . 15
2.1.20. Media Decoder . . . . . . . . . . . . . . . . . . . . 16
2.1.21. Received Source Stream . . . . . . . . . . . . . . . 16
2.1.22. Media Sink . . . . . . . . . . . . . . . . . . . . . 16
2.1.23. Received Raw Stream . . . . . . . . . . . . . . . . . 16
2.1.24. Media Render . . . . . . . . . . . . . . . . . . . . 16
2.2. Communication Entities . . . . . . . . . . . . . . . . . 17
2.2.1. End Point . . . . . . . . . . . . . . . . . . . . . . 17
2.2.2. RTP Session . . . . . . . . . . . . . . . . . . . . . 17
2.2.3. Participant . . . . . . . . . . . . . . . . . . . . . 18
2.2.4. Multimedia Session . . . . . . . . . . . . . . . . . 19
2.2.5. Communication Session . . . . . . . . . . . . . . . . 19
3. Relations at Different Levels . . . . . . . . . . . . . . . . 20
3.1. Media Source Relations . . . . . . . . . . . . . . . . . 20
3.1.1. Synchronization Context . . . . . . . . . . . . . . . 20
3.1.2. End Point . . . . . . . . . . . . . . . . . . . . . . 21
3.1.3. Participant . . . . . . . . . . . . . . . . . . . . . 22
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3.1.4. WebRTC MediaStream . . . . . . . . . . . . . . . . . 22
3.2. Packetization Time Relations . . . . . . . . . . . . . . 22
3.2.1. Single Stream Transport of SVC . . . . . . . . . . . 23
3.2.2. Multi-Channel Audio . . . . . . . . . . . . . . . . . 23
3.2.3. Redundancy Format . . . . . . . . . . . . . . . . . . 23
3.3. Packet Stream Relations . . . . . . . . . . . . . . . . . 24
3.3.1. Simulcast . . . . . . . . . . . . . . . . . . . . . . 24
3.3.2. Layered Multi-Stream Transmission . . . . . . . . . . 25
3.3.3. Robustness and Repair . . . . . . . . . . . . . . . . 26
3.3.4. Packet Stream Separation . . . . . . . . . . . . . . 29
3.4. Multiple RTP Sessions over one Media Transport . . . . . 30
4. Topologies and Communication Entities . . . . . . . . . . . . 30
4.1. Point-to-Point Communication . . . . . . . . . . . . . . 31
4.2. Central Conferencing . . . . . . . . . . . . . . . . . . 32
4.3. Full Mesh Conferencing . . . . . . . . . . . . . . . . . 33
4.4. Source-Specific Multicast . . . . . . . . . . . . . . . . 36
5. Security Considerations . . . . . . . . . . . . . . . . . . . 37
6. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . 38
7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 38
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 38
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 38
9.1. Normative References . . . . . . . . . . . . . . . . . . 38
9.2. Informative References . . . . . . . . . . . . . . . . . 38
Appendix A. Changes From Earlier Versions . . . . . . . . . . . 40
A.1. Modifications Between Version -02 and -03 . . . . . . . . 40
A.2. Modifications Between Version -01 and -02 . . . . . . . . 40
A.3. Modifications Between Version -00 and -01 . . . . . . . . 40
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 41
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].
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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
This section contains the concepts that can be involved in taking a
sequence of physical world stimulus (sound waves, photons, key-
strokes) at a sender side and transport them 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-
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
transformation (with the exception of Physical Stimulus
(Section 2.1.1) that can have no 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.
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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 introduces the concepts for the sender side (Figure 1).
This covers physical stimulus until media packets are emitted onto
the network.
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 Packet Stream
Source Packet 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 2.1.13.2.
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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
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 Packet Streams.
+--------------------+ +--------------------+
| Media Transport | | Media Transport |
+--------------------+ +--------------------+
| |
Received Packet Stream Received Redundancy PS
| |
| +-------------------+
V V
+--------------------+
| Media Repair |
+--------------------+
|
Repaired Packet Stream
V
+--------------------+
| Media Depacketizer |
+--------------------+
|
Received Encoded Stream
V
+--------------------+
| Media Decoder |
+--------------------+
|
Received Source Stream
V
+--------------------+
| Media Sink |--> Synchronization Information
+--------------------+
|
Received Raw Stream
V
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+--------------------+
| Media Renderer |
+--------------------+
|
V
Physical Stimulus
Figure 2: Receiver Side Concepts of the Media Chain
2.1.1. Physical Stimulus
The physical stimulus is a physical event that can be captured and
provided as media to a receiver. 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
The process of transforming the Physical Stimulus (Section 2.1.1)
into captured media. 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.
2.1.2.1. Alternate Usages
The CLUE WG uses the term "Capture Device" to identify a physical
capture device.
WebRTC WG uses the term "Recording Device" to refer to the locally
available capture devices in an end-system.
2.1.2.2. 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
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The time progressing stream of digitally sampled information, usually
periodically sampled, provided by a Media Capture (Section 2.1.2).
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.
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
2.1.4.1. Alternate Usages
The CLUE WG uses the term "Media Capture" for this purpose. A CLUE
Media Capture is identified via indexed notation. The terms Audio
Capture and Video Capture are used to identify Audio Sources and
Video Sources respectively. Concepts such as "Capture Scene",
"Capture Scene Entry" and "Capture" provide a flexible framework to
represent media captured spanning spatial regions.
The WebRTC WG defines the term "RtcMediaStreamTrack" to refer to a
Media Source. An "RtcMediaStreamTrack" is identified by the ID
attribute.
Typically a Media Source is mapped to a single m=line via the Session
Description Protocol (SDP) [RFC4566] unless mechanisms such as
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Source-Specific attributes are in place [RFC5576]. In the latter
cases, an m=line can represent either multiple Media Sources,
multiple Packet Streams (Section 2.1.10), or both.
2.1.4.2. 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.
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 3.3.2.
Source Stream
|
V
+--------------------------+
| Scalable Media Encoder |
+--------------------------+
| | ... |
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V V V
Encoded Dependent Dependent
Stream Stream Stream
Figure 4: Scalable Media Encoder Input and Outputs
2.1.6.1. Alternate Usages
Within the SDP usage, an SDP media description (m=line) describes
part of the necessary configuration required for encoding purposes.
CLUE's "Capture Encoding" provides specific encoding configuration
for this purpose.
2.1.6.2. 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.
2.1.7.1. 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.
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.
2.1.8.1. 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
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more sequences of packets, normally RTP packets, and output Source
Packet 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
Packet Stream. One such example is the packetization when using SVC,
as in Single Stream Transport (SST) usage of the payload format both
an Encoded Stream as well as Dependent Streams are packetized in a
single Source Packet Stream using a single SSRC.
The Media Packetizer can also produce multiple Packet Streams, for
example when Encoded and/or Dependent Streams are distributed over
multiple Packet Streams, possibly in different RTP sessions.
2.1.9.1. Alternate Usages
An RTP sender is part of the Media Packetizer.
2.1.9.2. 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 Packet Streams.
2.1.10. Packet Stream
A stream of RTP packets containing media data, source or redundant.
The Packet 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 Packet Stream is a packet 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 applied to cope with network transport losses.
Compare this with the Redundancy Packet Stream (Section 2.1.12).
2.1.10.1. Alternate Usages
The term "Stream" is used by the CLUE WG to define an encoded Media
Source sent via RTP. "Capture Encoding", "Encoding Groups" are
defined to capture specific details of the encoding scheme.
RFC3550 [RFC3550] uses the terms media stream, audio stream, video
stream and streams of (RTP) packets interchangeably. It defines the
SSRC as the "The source of a stream of RTP packets, ..."
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The equivalent mapping of a Packet Stream in SDP [RFC4566] is defined
per usage. For example, each Media Description (m=line) and
associated attributes can describe one Packet Stream OR properties
for multiple Packet Streams OR for an RTP session (via [RFC5576]
mechanisms for example).
2.1.10.2. Characteristics
o Each Packet 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 Packet Stream can have one and only
one SSRC.
o Each Packet Stream defines a unique RTP sequence numbering and
timing space.
o Several Packet Streams may map to a single Media Source via the
source transformations.
o Several Packet 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 Packet 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 Packet Stream
A Packet Stream (Section 2.1.10) that contains no original source
data, only redundant data that may be combined with one or more
Received Packet Stream (Section 2.1.14) to produce Repaired Packet
Streams (Section 2.1.17).
2.1.13. Media Transport
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A Media Transport defines the transformation that the Packet 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].
2.1.13.1. Characteristics
o Media Transport transmits Packet Streams of RTP Packets from a
source transport address to a destination transport address.
2.1.13.2. Media Stream Decomposition
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).
Packet Stream
|
V
+--------------------------+
| Media Transport Sender |
+--------------------------+
|
Sent Packet Stream
V
+--------------------------+
| Network Transport |
+--------------------------+
|
Transported Packet Stream
V
+--------------------------+
| Media Transport Receiver |
+--------------------------+
|
V
Received Packet Stream
Figure 5: Decomposition of Media Transport
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2.1.13.2.1. 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 Packet Stream and emits the packets onto the
network using the transport association established for this Media
Transport thus creating a Sent Packet Stream (Section 2.1.13.2.2).
In this process it transforms the Packet 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 Packet Stream.
2.1.13.2.2. Sent Packet Stream
The Sent Packet Stream is the Packet Stream as entering the first hop
of the network path to its destination. The Sent Packet 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)).
2.1.13.2.3. Network Transport
Network Transport is the transformation that the Sent Packet Stream
(Section 2.1.13.2.2) 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 Packet Stream
(Section 2.1.13.2.4) at the exit of the network path.
2.1.13.2.4. Transported Packet Stream
The Packet Stream that is emitted out of the network path at the
destination, subjected to the Network Transport's transformation
(Section 2.1.13.2.3).
2.1.13.2.5. Media Transport Receiver
The receiver End-Point's (Section 2.2.1) transformation of the
Transported Packet Stream (Section 2.1.13.2.4) by its reception
process that result in the Received Packet Stream (Section 2.1.14).
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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2.1.14. Received Packet Stream
The Packet 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.
2.1.15. Received Redundandy Packet Stream
The Redundancy Packet Stream (Section 2.1.12) resulting from the
Media Transport's transformation, i.e. subjected to packet loss,
packet corruption, and varying transmission delay from sender to
receiver.
2.1.16. Media Repair
A Transformation that takes as input one or more Source Packet
Streams (Section 2.1.10) as well as Redundancy Packet 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 2.1.21) after Media Decoder
(Section 2.1.20). The output is a Repaired Packet Stream
(Section 2.1.17).
2.1.17. Repaired Packet Stream
A Received Packet Stream (Section 2.1.14) for which Received
Redundancy Packet Stream (Section 2.1.15) information has been used
to try to re-create the Packet Stream (Section 2.1.10) as it was
before Media Transport (Section 2.1.13).
2.1.18. Media Depacketizer
A Media Depacketizer takes one or more Packet 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 Packet Streams.
2.1.19. Received Encoded Stream
The received version of an Encoded Stream (Section 2.1.7).
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2.1.20. 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).
2.1.20.1. Alternate Usages
Within the context of SDP, an m=line describes the necessary
configuration and identification (RTP Payload Types) required to
decode either one or more incoming Media Streams.
2.1.20.2. 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.
2.1.21. Received Source Stream
The received version of a Source Stream (Section 2.1.5).
2.1.22. 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 2.1.24). 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.
2.1.22.1. 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.
2.1.23. Received Raw Stream
The received version of a Raw Stream (Section 2.1.3).
2.1.24. Media Render
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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.
2.1.24.1. 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.
2.2.1. End Point
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".
2.2.1.1. Alternate Usages
The CLUE Working Group (WG) uses the terms "Media Provider" and
"Media Consumer" to describes aspects of End Point pertaining to
sending and receiving functionalities.
2.2.1.2. Characteristics
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 3.1.1). Thus one End Point
can have multiple CNAMEs. 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.
2.2.2. RTP Session
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 Packet Streams. Within an RTP
session, every participant can find meta-data and control information
(over RTCP) about all the Packet Streams in the RTP session. The
bandwidth of the RTCP control channel is shared between all
participants within an RTP Session.
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2.2.2.1. Alternate Usages
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.
2.2.2.2. Characteristics
o Typically, an RTP Session can carry one ore more Packet 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.
2.2.3.1. 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).
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o A participant can have several associated transport flows,
including several separate local transport addresses for those
transport flows.
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.
2.2.4.1. Alternate Usages
RFC4566 [RFC4566] defines a multimedia session as a set of multimedia
senders and receivers and the data streams flowing from senders to
receivers.
RFC3550 [RFC3550] defines it as set of concurrent RTP sessions among
a common group of participants. For example, a video conference
(which is a multimedia session) may contain an audio RTP session and
a video RTP session.
2.2.4.2. Characteristics
o A Multimedia Session can be composed of several parallel RTP
Sessions with potentially multiple Packet Streams per RTP Session.
o Each participant in a Multimedia Session can have a multitude of
Media Captures and Media Rendering devices.
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.
2.2.5.1. Alternate Usages
The Session Description Protocol (SDP) [RFC4566] defines a multimedia
session as a set of multimedia senders and receivers and the data
streams flowing from senders to receivers. In that definition it is
however not clear if a multimedia session includes both the sender's
and the receiver's view of the same RTP Packet Stream.
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2.2.5.2. 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 Packet 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.
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 3.3.1) needs to determine relations at Packet Stream level,
however the reason to relate Packet Streams is that multiple Media
Encoders use the same Media Source, i.e. to be able to identify a
common Media Source.
3.1. Media Source Relations
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 application logic as well as media
handling purposes. These cases are further discussed below.
3.1.1. 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.
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A single Multimedia Session can contain media from one or more
Synchronization Contexts. 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.
3.1.1.1. 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.
3.1.1.2. 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.
3.1.1.3. CLUE Scenes
In CLUE "Capture Scene", "Capture Scene Entry" and "Captures" define
an implied Synchronization Context.
3.1.1.4. 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.
3.1.1.5. 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.
3.1.2. End Point
Some applications requires knowledge of what Media Sources originate
from a particular End Point (Section 2.2.1). This can include such
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decisions as packet routing between parts of the topology, knowing
the End Point origin of the Packet Streams.
In RTP, this identification has been overloaded with the
Synchronization Context through the usage of the source description
CNAME 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.
3.1.3. 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.
3.1.4. WebRTC MediaStream
An RtcMediaStream, in addition to requiring a single Synchronization
Context as discussed above, is also an explicit grouping of a set of
Media Sources, as identified by RtcMediaStreamTracks, within the
RtcMediaStream.
3.2. Packetization Time Relations
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 Packet Streams
(Section 2.1.10). These are caused by grouping together or
distributing these different types of streams into Packet Streams.
This section will look at such relationships.
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3.2.1. Single Stream Transport of SVC
Scalable Video Coding [RFC6190] has a mode of operation where Encoded
Streams and Dependent Streams from the SVC Media Encoder is grouped
together in a single Source Packet Stream using the SVC RTP Payload
format.
3.2.2. 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 then independently encoded by a
Media Encoder and the different Encoded Streams are then packetized
together in a time synchronized way into a single Source Packet
Stream using the used codec's RTP Payload format. Example of such
codecs are, PCMA and PCMU [RFC3551], AMR [RFC4867], and G.719
[RFC5404].
3.2.3. 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
different Media Encoder to encode the same Media Source as the
primary, as depicted below in Figure 6.
+--------------------+
| Media Source |
+--------------------+
|
Source Stream
|
+------------------------+
| |
V V
+--------------------+ +--------------------+
| Media Encoder | | Media Encoder |
+--------------------+ +--------------------+
| |
| +------------+
Encoded Stream | Time Delay |
| +------------+
| |
| +------------------+
V V
+--------------------+
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| Media Packetizer |
+--------------------+
|
V
Packet Stream
Figure 6: 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 6) 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.
3.3. Packet Stream Relations
This section discusses various cases of relationships among Packet
Streams. This is a common relation to handle in RTP due to that
Packet Streams are separate and have their own SSRC, implying
independent sequence numbers and timestamp spaces. The underlying
reasons for the Packet Stream relationships are different, as can be
seen in the cases below. The different Packet Streams can be handled
within the same RTP Session or different RTP Sessions to accomplish
different transport goals. This separation of Packet Streams is
further discussed in Section 3.3.4.
3.3.1. 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 Packet
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 3.3.4.
+----------------+
| Media Source |
+----------------+
Source Stream |
+----------------------+----------------------+
| | |
v v v
+------------------+ +------------------+ +------------------+
| Media Encoder | | Media Encoder | | Media Encoder |
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+------------------+ +------------------+ +------------------+
| Encoded | Encoded | Encoded
| Stream | Stream | Stream
v v v
+------------------+ +------------------+ +------------------+
| Media Packetizer | | Media Packetizer | | Media Packetizer |
+------------------+ +------------------+ +------------------+
| Source | Source | Source
| Packet | Packet | Packet
| Stream | Stream | Stream
+-----------------+ | +-----------------+
| | |
V V V
+-------------------+
| Media Transport |
+-------------------+
Figure 7: Example of Media Source Simulcast
The simulcast relation between the Packet Streams is the common Media
Source. In addition, to be able to identify the common Media Source,
a receiver of the Packet 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.
3.3.2. Layered Multi-Stream Transmission
Multi-stream transmission (MST) is a mechanism by which different
portions of a layered encoding of a Source Stream are sent using
separate Packet Streams (sometimes in separate RTP sessions). MSTs
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
dependency. 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 Packet Streams, i.e.
SSRCs, and the third layer is sent on a separate Media Transport,
i.e. a different RTP Session.
+----------------+
| Media Source |
+----------------+
|
|
V
+---------------------------------------------------------+
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| Media Encoder |
+---------------------------------------------------------+
| | |
Encoded Stream Dependent Stream Dependent Stream
| | |
V V V
+----------------+ +----------------+ +----------------+
|Media Packetizer| |Media Packetizer| |Media Packetizer|
+----------------+ +----------------+ +----------------+
| | |
Packet Stream Packet Stream Packet Stream
| | |
+------+ +------+ |
| | |
V V V
+-----------------+ +-----------------+
| Media Transport | | Media Transport |
+-----------------+ +-----------------+
Figure 8: Example of Media Source Layered Dependency
The SVC MST 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 Packet Stream per Media
Encoder and a single Media Source in each RTP Session, common SSRC
and CNAMEs can be used to identify the common Media Source. When
multiple Packet Streams are sent from one Media Encoder in the same
RTP Session, 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 Packet Streams.
3.3.3. Robustness and Repair
Packet Streams may be protected by Redundancy Packet Streams during
transport. Several approaches listed below can achieve the same
result;
o Duplication of the original Packet Stream
o Duplication of the original Packet Stream with a time offset,
o Forward Error Correction (FEC) techniques, and
o Retransmission of lost packets (either globally or selectively).
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3.3.3.1. RTP Retransmission
The figure below (Figure 9) represents an example where a Media
Source's Source Packet Stream is protected by a retransmission (RTX)
flow [RFC4588]. In this example the Source Packet Stream and the
Redundancy Packet Stream share the same Media Transport.
+--------------------+
| Media Source |
+--------------------+
|
V
+--------------------+
| Media Encoder |
+--------------------+
| Retransmission
Encoded Stream +--------+ +---- Request
V | V V
+--------------------+ | +--------------------+
| Media Packetizer | | | RTP Retransmission |
+--------------------+ | +--------------------+
| | |
+------------+ Redundancy Packet Stream
Source Packet Stream |
| |
+---------+ +---------+
| |
V V
+-----------------+
| Media Transport |
+-----------------+
Figure 9: Example of Media Source Retransmission Flows
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The RTP Retransmission example (Figure 9) helps illustrate that this
mechanism works purely on the Source Packet Stream. The RTP
Retransmission transform buffers the sent Source Packet Stream and
upon requests emits a retransmitted packet with some extra payload
header as a Redundancy Packet Stream. The RTP Retransmission
mechanism [RFC4588] is specified so that there is a one to one
relation between the Source Packet Stream and the Redundancy Packet
Stream. Thus a Redundancy Packet Stream needs to be associated with
its Source Packet Stream upon being received. This is done based on
CNAME selectors and heuristics to match requested packets for a given
Source Packet Stream with the original sequence number in the payload
of any new Redundancy Packet Stream using the RTX payload format. In
cases where the Redundancy Packet Stream is sent in a separate RTP
Session from the Source Packet Stream, these sessions are related,
e.g. using the SDP Media Grouping's [RFC5888] FID semantics.
3.3.3.2. Forward Error Correction
The figure below (Figure 10) represents an example where two Media
Sources' Source Packet Streams are protected by FEC. Source Packet
Stream A has a Media Redundancy transformation in FEC Encoder 1.
This produces a Redundancy Packet Stream 1, that is only related to
Source Packet Stream A. The FEC Encoder 2, however takes two Source
Packet Streams (A and B) and produces a Redundancy Packet Stream 2
that protects them together, i.e. Redundancy Packet Stream 2 relate
to two Source Packet Streams (a FEC group). FEC decoding, when
needed due to packet loss or packet corruption at the receiver,
requires knowledge about which Source Packet Streams that the FEC
encoding was based on.
In Figure 10 all Packet Streams are sent on the same Media Transport.
This is however not the only possible choice. Numerous combinations
exist for spreading these Packet 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 |
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+--------------------+ +--------------------+
| |
Source Packet Stream A Source Packet Stream B
| |
+-----+-------+-------------+ +-------+------+
| V V V |
| +---------------+ +---------------+ |
| | FEC Encoder 1 | | FEC Encoder 2 | |
| +---------------+ +---------------+ |
| | | |
| Redundancy PS 1 Redundancy PS 2 |
V V V V
+----------------------------------------------------------+
| Media Transport |
+----------------------------------------------------------+
Figure 10: Example of FEC Flows
As FEC Encoding exists in various forms, the methods for relating FEC
Redundancy Packet Streams with its source information in Source
Packet Streams are many. The XOR based RTP FEC Payload format
[RFC5109] is defined in such a way that a Redundancy Packet Stream
has a one to one relation with a Source Packet Stream. In fact, the
RFC requires the Redundancy Packet Stream to use the same SSRC as the
Source Packet 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 Packet Stream is to know the related Source Packet Stream,
including its SSRC.
3.3.4. Packet Stream Separation
Packet Streams can be separated exclusively based on their SSRCs or
at the RTP Session level or at the Multi-Media Session level as
explained below.
When the Packet 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 Packet Streams.
On the other hand, when Packet 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 Packet Streams are intended for different Media
Transports.
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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 signalling
level relations have been established using semantics from Grouping
of Media lines framework [RFC5888]. Examples of this are RTP
Retransmission [RFC4588], SVC Multi Stream Transmission [RFC6190] and
XOR Based FEC [RFC5109]. RTCP CNAME explicitly relates Packet
Streams across different RTP Sessions, as explained in the previous
section. Such a relationship can be used to perform inter-media
synchronization.
Packet 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.
3.4. 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 Packet 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
signalling design and when defining terminology.
4. Topologies and Communication Entities
This Section reviews some communication topologies and looks at the
relationship among the communication entities that are defined in
Section 2.2. This section doesn't deal with discussions about the
streams and their relation to the transport. Instead, it covers the
aspects that enable the transport of those streams. For example, the
Media Transports (Section 2.1.13) that exists between the End Points
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(Section 2.2.1) that are part of an RTP session (Section 2.2.2) and
their relationship to the Multi-Media Session (Section 2.2.4) between
Participants (Section 2.2.3) and the established Communication
session (Section 2.2.5) are explained.
4.1. Point-to-Point Communication
Figure 11 shows a very basic point-to-point communication session
between A and B. It uses two different audio and video RTP sessions
between A's and B's end points. Assume that the Multi-media session
shared by the participants is established using SIP (i.e., there is a
SIP Dialog between A and B). The high level representation of this
communication scenario can be demonstrated using Figure 11.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 11: Point to Point Communication
However, this picture gets slightly more complex when redrawn using
the communication entities concepts defined earlier in this document.
+-----------------------------------------------------------+
| Communication Session |
| |
| +----------------+ +----------------+ |
| | Participant A | +-------------+ | Participant B | |
| | | | Multi-Media | | | |
| | +-------------+|<=>| Session |<=>|+-------------+ | |
| | | End Point A || |(SIP Dialog) | || End Point B | | |
| | | || +-------------+ || | | |
| | | +-----------++---------------------++-----------+ | | |
| | | | RTP Session| | | | | |
| | | | Audio |---Media Transport-->| | | | |
| | | | |<--Media Transport---| | | | |
| | | +-----------++---------------------++-----------+ | | |
| | | || || | | |
| | | +-----------++---------------------++-----------+ | | |
| | | | RTP Session| | | | | |
| | | | Video |---Media Transport-->| | | | |
| | | | |<--Media Transport---| | | | |
| | | +-----------++---------------------++-----------+ | | |
| | +-------------+| |+-------------+ | |
| +----------------+ +----------------+ |
+-----------------------------------------------------------+
Figure 12: Point to Point Communication Session with two RTP Sessions
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Figure 12 shows the two RTP Sessions only exist between the two End
Points A and B and over their respective Media Transports. The
Multi-Media Session establishes the association between the two
Participants and configures these RTP sessions and the Media
Transports that are used.
4.2. Central Conferencing
This section looks at the central conferencing communication
topology, where a number of participants, like A, B, C, and D in
Figure 13, communicate using an RTP mixer.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 13: Centralized Conferincing using an RTP Mixer
In this case each of the Participants establish their Multi-media
session with the Conference Bridge. Thus, negotiation for the
establishment of the used RTP sessions and their configuration
happens between these entities. The participants have their End
Points (A, B, C, D) and the Conference Bridge has the host running
the RTP mixer, referred to as End Point M in Figure 14. However,
despite the individual establishment of four Multi-Media Sessions and
the corresponding Media Transports for each of the RTP sessions
between the respective End Points and the Conference Bridge, there is
actually only two RTP sessions. One for audio and one for Video, as
these RTP sessions are, in this topology, shared between all the
Participants.
+-------------------------------------------------------------------+
| Communication Session |
| |
| +----------------+ +----------------+ |
| | Participant A | +-------------+ | Conference | |
| | | | Multi-Media | | Bridge | |
| | +-------------+|<=====>| Session A |<=====>|+-------------+ | |
| | | End Point A || |(SIP Dialog) | || End Point M | | |
| | | || +-------------+ || | | |
| | | +-----------++-----------------------------++-----------+ | | |
| | | | RTP Session| | | | | |
| | | | Audio |-------Media Transport------>| | | | |
| | | | |<------Media Transport-------| | | | |
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| | | +-----------++-----------------------------++------+ | | | |
| | | || || | | | | |
| | | +-----------++-----------------------------++----+ | | | | |
| | | | RTP Session| | | | | | | |
| | | | Video |-------Media Transport------>| | | | | | |
| | | | |<------Media Transport-------| | | | | | |
| | | +-----------++-----------------------------++ | | | | | |
| | +-------------+| || | | | | | |
| +----------------+ || | | | | | |
| || | | | | | |
| +----------------+ || | | | | | |
| | Participant B | +-------------+ || | | | | | |
| | | | Multi-Media | || | | | | | |
| | +-------------+|<=====>| Session B |<=====>|| | | | | | |
| | | End Point B || |(SIP Dialog) | || | | | | | |
| | | || +-------------+ || | | | | | |
| | | +-----------++-----------------------------++ | | | | | |
| | | | RTP Session| | | | | | | |
| | | | Video |-------Media Transport------>| | | | | | |
| | | | |<------Media Transport-------| | | | | | |
| | | +-----------++-----------------------------++----+ | | | | |
| | | || || | | | | |
| | | +-----------++-----------------------------++------+ | | | |
| | | | RTP Session| | | | | |
| | | | Audio |-------Media Transport------>| | | | |
| | | | |<------Media Transport-------| | | | |
| | | +-----------++-----------------------------++-----------+ | | |
| | +-------------+| |+-------------+ | |
| +----------------+ +----------------+ |
+-------------------------------------------------------------------+
Figure 14: Central Conferencing with Two Participants A and B
communicating over a Conference Bridge
It is important to stress that in the case of Figure 14, it might
appear that the the Multi-Media Sessions context is scoped between A
and B over M. This might not be always true and they can have
contexts that extend further. In this case the RTP session, its
common SSRC space goes beyond what occurs between A and M and B and M
respectively.
4.3. Full Mesh Conferencing
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This section looks at the case where the three Participants (A, B and
C) wish to communicate. They establish individual Multi-Media
Sessions and RTP sessions between themselves and the other two peers.
Thus, each providing two copies of their media to every other
participant. Figure 15 shows a high level representation of such a
topology.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 15: Full Mesh Conferencing with three Participants A, B and C
In this particular case there are two aspects worth noting. The
first is there will be multiple Multi-Media Sessions per
Communication Session between the participants. This, however,
hasn't been true in the earlier examples; the Centralized
Conferencing inSection 4.2 being the exception. The second aspect is
consideration of whether one needs to maintain relationships between
entities and concepts, for example MediaSources, between these
different Multi-Media Sessions and between Packet Streams in the
independent RTP sessions configured by those Multi-Media Sessions.
+-----------------------------------------+
| Participant A |
+----------+ | +--------------------------------------+|
| Multi- | | | End Point A ||
| Media |<======>| | ||
| Session | | |+-------+ +-------+ +-------+ ||
| 1 | | || RTP 1 |<----| MS A1 |---->| RTP 2 | ||
+----------+ | || | +-------+ | | ||
^^ | +|-------|-------------------|-------|-+|
|| +--|-------|-------------------|-------|--+
|| | | ^^ | |
VV | | || | |
+-------------------------|-------|----+ || | |
| Participant B | | | VV | |
| +-----------------------|-------|---+| +----------+ | |
| | End Point B +----->| | || | Multi- | | |
| | | +-------+ || | Media | | |
| | +-------+ | +-------+ || | Session | | |
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| | | MS B1 |------+----->| RTP 3 | || | 2 | | |
| | +-------+ | | || +----------+ | |
| +-----------------------|-------|---+| ^^ | |
+-------------------------|-------|----+ || | |
^^ | | || | |
|| | | VV | |
|| +--|-------|-------------------|-------|--+
VV | | | Participant C | | |
+----------+ | +|-------|-------------------|-------|-+|
| Multi- | | || | End Point C | | ||
| Media |<======>| |+-------+ +-------+ ||
| Session | | | ^ +-------+ ^ ||
| 3 | | | +---------| MS C1 |---------+ ||
+----------+ | | +-------+ ||
| +--------------------------------------+|
+-----------------------------------------+
Figure 16: Full Mesh Conferencing between three Participants A, B and
C
For the sake of clarity, Figure 16 above does not include all these
concepts. The Media Sources (MS) from a given End Point is sent to
the two peers. This requires encoding and Media Packetization to
enable the Packet Streams to be sent over Media Transports in the
context of the RTP sessions depicted. The RTP sessions 1, 2, and 3
are independent, and established in the context of each of the Multi-
Media Sessions 1, 2 and 3. The joint communication session the full
figure represents (not shown here as it was Figure 14 in order to
save space), however, combines the received representations of the
peers' Media Sources and plays them back.
It is noteworthy that the full mesh conferencing topologies described
here have the potential for creating loops. For example, if one
compares the above full mesh with a mixing three party communication
session as depicted in (Figure 17). In this example A's Media Source
A1 is sent to B over a Multi-Media Session (A-B). In B the Media
Source A1 is mixed with Media Source B1 and the resulting Media
Source (MS AB) is sent to C over a Multi-Media Session (B-C). If C
and A would establish a Multi-Media Session (A-C) and C would act in
the same role as B, then A would receive a Media Source from C that
contains a mix of A, B and C's individual Media Sources. This would
result in A playing out a time delay version of its own signal (i.e.,
the system has created an echo path).
+--------------+ +--------------+ +--------------+
| A | | B +-------+ | | C |
| | | | MS B1 | | | |
| | | +-------+ | | |
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| +-------+ | | | | | |
| | MS A1 |----|--->|-----+ MS AB -|--->| |
| +-------+ | | | | |
+--------------+ +--------------+ +--------------+
Figure 17: Mixing Three Party Communication Session
The looping issue can be avoided, detected or prevented using two
general methods. The first method is to use great care when setting
up and establishing the communication session if participants have
any mixing or forwarding capacity, so that one doesn't end up getting
back a partial or full representation of one's own media believing it
is someone else's. The other method is to maintain some unique
identifiers at the communication session level for all Media Sources
and ensure that any Packet Streams received identify those Media
Sources that contributed to the content of the Packet Stream.
4.4. Source-Specific Multicast
In one-to-many media distribution cases (e.g., IPTV), where one Media
Sender or a set of Media Senders is allowed to send Packet Streams on
a particular Source-Specific Multicast (SSM) group to many receivers
(R), there are some different aspects to consider. Figure 18
presents a high level SSM system for RTP/RTCP defined in [RFC5760].
In this case, several Media Senders sends their Packet Streams to the
Distribution Source, which is the only one allowed to send to the SSM
group. The Receivers joining the SSM group can provide RTCP feedback
on its reception by sending unicast feedback to a Feedback Target
(FT).
+--------+ +-----+
|Media | | | Source-Specific
|Sender 1|<----->| D S | Multicast (SSM)
+--------+ | I O | +--+----------------> R(1)
| S U | | | |
+--------+ | T R | | +-----------> R(2) |
|Media |<----->| R C |->+ | : | |
|Sender 2| | I E | | +------> R(n-1) | |
+--------+ | B | | | | | |
: | U | +--+--> R(n) | | |
: | T +-| | | | |
: | I | |<---------+ | | |
+--------+ | O |F|<---------------+ | |
|Media | | N |T|<--------------------+ |
|Sender M|<----->| | |<-------------------------+
+--------+ +-----+ RTCP Unicast
FT = Feedback Target
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Figure 18: Source-Specific Multicast Communication Topology
Here the Media Transport from the Distribution Source to all the SSM
receivers (R) have the same 5-tuple, but in reality have different
paths. Also, the Multi-Media Sessions between the Distribution
Source and the individual receivers are normally identical. This is
due to one-way communication from the Distribution Source to the
receiver of configuration information. This is information typically
embedded in Electronic Program Guides (EPGs), distributed by the
Session Announcement Protocol (SAP) [RFC2974] or other one-way
protocols. In some cases load balancing occurs, for example, by
providing the receiver with a set of Feedback Targets and then it
randomly selects one out of the set.
This scenario varies significantly from previously described
communication topologies due to the asymmetric nature of the RTP
Session context across the Distribution Source. The Distribution
Source forms a focal point in collecting the unicasted RTCP feedback
from the receivers and then re-distributing it to the Media Senders.
Each Media Sender and the Distribution Source establish their own
Multi-Media Session Context for the underlying RTP Sessions but with
shared RTCP context across all the receivers.
To improve the readability,Figure 18 intentionally hides the details
of the various entities . Expanding on this, one can think of Media
Senders being part of one or more Multi-Media Sessions grouped under
a Communication Session. The Media Sender in this scenario refers to
the Media Packetizer transformation Section 2.1.9. The Packet Stream
generated by such a Media Sender can be part of its own RTP Session
or can be multiplexed with other Packet Streams within an End Point.
The latter case requires careful consideration since the re-
distributed RTCP packets now correspond to a single RTP Session
Context across all the Media Senders.
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.
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.
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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, and Harald Alvestrand.
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. References
9.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[UML] Object Management Group, "OMG Unified Modeling Language
(OMG UML), Superstructure, V2.2", OMG formal/2009-02-02,
February 2009.
9.2. Informative References
[I-D.ietf-avtcore-clksrc]
Williams, A., Gross, K., Brandenburg, R., and H. Stokking,
"RTP Clock Source Signalling", draft-ietf-avtcore-
clksrc-07 (work in progress), October 2013.
[I-D.ietf-clue-framework]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", draft-ietf-clue-
framework-12 (work in progress), October 2013.
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[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-05 (work in progress), October 2013.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-08 (work
in progress), September 2013.
[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.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
2002.
[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.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
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[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.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760, February 2010.
[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.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
Appendix A. Changes From Earlier Versions
NOTE TO RFC EDITOR: Please remove this section prior to publication.
A.1. 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.2. 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.3. Modifications Between Version -00 and -01
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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
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
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Bo Burman
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com
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