3. Concepts of Inter-Relations
This section uses the concepts from previous sections and looks at
different types of relationships among them. These relationships
occur at different abstraction levels and for different purposes, but
the reason for the needed relationship at a certain step in the media
handling chain may exist at another step. For example, the use of
simulcast (Section 3.6) implies a need to determine relations at the
RTP stream level, but the underlying reason is that multiple media
encoders use the same media source, i.e., to be able to identify a
common media source.
3.1. Synchronization Context
A synchronization context defines a requirement for a strong timing
relationship between the media sources, typically requiring alignment
of clock sources. Such a 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] describes inter-media synchronization between RTP sessions
based on RTCP CNAME, RTP, and timestamps of a reference clock
formatted using the Network Time Protocol (NTP) [RFC5905]. As
indicated in [RFC7273], despite using NTP format timestamps, it is
not required that the clock be synchronized to an NTP source.
3.1.2. Clock Source Signaling
[RFC7273] provides a mechanism to signal the clock source in the
Session Description Protocol (SDP) [RFC4566] 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
WebRTC defines RtcMediaStream with one or more RtcMediaStreamTracks.
All tracks in a RtcMediaStream are intended to be synchronized when
rendered, implying that they must be generated such that
synchronization is possible.
3.1.4. Explicitly via SDP Mechanisms
The SDP Grouping Framework [RFC5888] defines an "m=" line
(Section 4.2) grouping mechanism called Lip Synchronization (with LS
identification-tag) for establishing the synchronization requirement
across "m=" lines when they map to individual sources.
Source-Specific Media Attributes in SDP [RFC5576] extends the above
mechanism when multiple media sources are described by a single "m="
line.
3.2. Endpoint
Some applications require knowledge of what media sources originate
from a particular endpoint (Section 2.2.1). This can include such
decisions as packet routing between parts of the topology, knowing
the endpoint origin of the RTP streams.
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). This works for some
usages, but in others it breaks down. For example, if an endpoint
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,
CNAME would not be an appropriate identification for that endpoint.
Therefore, an endpoint may have multiple CNAMEs. The CNAMEs or the
media sources themselves can be related to the endpoint.
3.3. Participant
In communication scenarios, information about which media sources
originate from which participant (Section 2.2.3) is commonly needed.
One reason is, for example, to enable the application to correctly
display participant identity information associated with the media
sources. This association is handled through signaling to point at a
specific multimedia session where the media sources may be explicitly
or implicitly tied to a particular endpoint.
Participant information becomes more problematic when there are media
sources that are generated through mixing or other conceptual
processing of raw streams or source streams that originate from
different participants. These types of media sources can thus have a
dynamically varying set of origins and participants. RTP contains
the concept of CSRC that carries information about the previous step
origin of the included media content on the 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. Multi-Channel Audio
There exist a number of RTP payload formats that can carry multi-
channel audio, despite the codec being a single-channel (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
packetized together in a time-synchronized way into a single source
RTP stream, using the used codec's RTP payload format. Examples of
codecs that support multi-channel audio are PCMA and PCMU [RFC3551],
Adaptive Multi Rate (AMR) [RFC4867], and G.719 [RFC5404].
3.6. Simulcast
A media source represented as multiple independent encoded streams
constitutes a simulcast [SDP-SIMULCAST] or Modification Detection
Code (MDC) of that media source. Figure 8 shows an example of a
media source that is encoded into three separate simulcast streams,
that are in turn sent on the same media transport flow. When using
simulcast, the RTP streams may be sharing an RTP session and media
transport, or be separated on different RTP sessions and media
transports, or be any combination of these two. One major reason to
use separate media transports is to make use of different quality of
service (QoS) for the different source RTP streams. Some
considerations on separating related RTP streams are discussed in
Section 3.12.
+----------------+
| 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 8: 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 lay behind the produced encoded stream and its
properties. This enables selection of the stream that is most useful
in the application at that moment.
3.7. Layered Multi-Stream
Layered Multi-Stream (LMS) is a mechanism by which different portions
of a layered or scalable 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. Figure 9 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, i.e., SSRCs,
and the third layer is sent on a separate media transport.
+----------------+
| 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 9: Example of Media Source Layered Dependency
It is sometimes useful to make a distinction between using a single
media transport or multiple separate media transports when (in both
cases) using multiple RTP streams to carry encoded streams and
dependent streams for a media source. Therefore, the following new
terminology is defined here:
SRST: Single RTP stream on a Single media Transport
MRST: Multiple RTP streams on a Single media Transport
MRMT: Multiple RTP streams on Multiple media Transports
MRST and MRMT relations need to identify the common media encoder
origin for the encoded and dependent streams. When using different
RTP sessions (MRMT), a single RTP stream per media encoder, and a
single media source in each RTP session, common SSRCs 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
(MRST), 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 have
a common CNAME, additional heuristics or identification need to be
applied to create the MRST or MRMT relationships between the RTP
streams.
3.8. 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 (see Figure 10). This is a specific type of redundancy. 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 one is which. This
can also be seen as a specific type of simulcast (Section 3.6) 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 10: Example of RTP Stream Duplication
3.9. Redundancy Format
"RTP Payload for Redundant Audio Data" [RFC2198] defines a transport
for 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 in Figure 11.
+--------------------+
| Media Source |
+--------------------+
|
Source Stream
|
+------------------------+
| |
V V
+--------------------+ +--------------------+
| Media Encoder | | Media Encoder |
+--------------------+ +--------------------+
| |
| +------------+
Encoded Stream | Time Delay |
| +------------+
| |
| +------------------+
V V
+--------------------+
| Media Packetizer |
+--------------------+
|
V
RTP Stream
Figure 11: 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. The case depicted above (Figure 11) relates the received
source stream fragments coming out of different media decoders, to be
able to combine them together into a less erroneous source stream.
3.10. RTP Retransmission
Figure 12 shows 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.
+--------------------+
| 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 12: Example of Media Source Retransmission Flows
The RTP retransmission example (Figure 12) illustrates that this
mechanism works purely on the source RTP stream. The RTP
retransmission transforms buffers from the sent source RTP stream
and, upon request, emits a retransmitted packet with an extra payload
header as a redundancy RTP stream. The RTP retransmission mechanism
[RFC4588] is specified such that there is a one-to-one relation
between the source RTP stream and the redundancy RTP stream.
Therefore, a redundancy RTP stream needs to be associated with its
source RTP stream. 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 different RTP session than the
source RTP stream, the RTP session relation is signaled by using the
SDP media grouping's [RFC5888] Flow Identification (FID
identification-tag) semantics.
3.11. Forward Error Correction
Figure 13 shows an example where two media sources' source RTP
streams are protected by FEC. Source RTP stream A has an RTP-based
redundancy transformation in FEC encoder 1. This 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 jointly,
i.e., redundancy RTP stream 2 relates 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 13, 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 13: Example of FEC Redundancy RTP Streams
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
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 the use of either a separate RTP session or 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.
3.12. RTP Stream Separation
RTP streams can be separated exclusively based on their SSRCs, at the
RTP session level, or at the multimedia 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 endpoint. SSRC-based approaches [RFC5576], when used, can
explicitly relate various such RTP streams.
On the other hand, when RTP streams that are related 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
as a 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 the media-line
grouping 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.
3.13. Multiple RTP Sessions over one Media Transport
[TRANSPORT-MULTIPLEX] describes a mechanism that allows 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 (using a common network path or potentially
having different ones). The fewer media transports used, the less
need for NAT/firewall traversal resources and smaller number of flow-
based QoS.
However, multiple RTP sessions over one media transport imply that a
single media transport 5-tuple is not sufficient to express in which
RTP session context a particular RTP stream exists. Complexities in
the relationship between media transports and RTP sessions 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 needs 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
RFCs and Internet-Drafts (at the time of writing), using the concepts
from previous sections.
4.1. Telepresence Terms
The terms in this subsection are used in the context of CLUE
[CLUE-FRAME]. Note that some terms listed in this subsection use the
same names as terms defined elsewhere in this document. Unless
explicitly stated (as "RTP Taxonomy") and in this subsection, they
are to be read as references to the CLUE-specific term within this
subsection.
4.1.1. Audio Capture
Defined in CLUE as a Media Capture (Section 4.1.7) for audio.
Describes an audio media source (Section 2.1.4).
4.1.2. Capture Device
Defined in CLUE as a device that converts physical input into an
electrical signal. Identifies a physical entity performing an RTP
Taxonomy media capture (Section 2.1.2) transformation.
4.1.3. Capture Encoding
Defined in CLUE as a specific Encoding (Section 4.1.6) of a Media
Capture (Section 4.1.7). Describes an encoded stream (Section 2.1.7)
related to CLUE-specific semantic information.
4.1.4. Capture Scene
Defined in CLUE as a structure representing a spatial region captured
by one or more Capture Devices (Section 4.1.2), each capturing media
representing a portion of the region. Describes a set of spatially
related media sources (Section 2.1.4).
4.1.5. Endpoint
Defined in CLUE as a CLUE-capable device that is the logical point of
final termination through receiving, decoding, and rendering and/or
initiation through capturing, encoding, and sending of media Streams
(Section 4.1.10). CLUE further defines it to consist of one or more
physical devices with source and sink media streams, and exactly one
participant [RFC4353]. Describes exactly one participant
(Section 2.2.3) and one or more RTP Taxonomy endpoints
(Section 2.2.1).
4.1.6. Individual Encoding
Defined in CLUE as a set of parameters representing a way to encode a
Media Capture (Section 4.1.7) to become a Capture Encoding
(Section 4.1.3). Describes the configuration information needed to
perform a media encoder (Section 2.1.6) transformation.
4.1.7. Media Capture
Defined in CLUE as a source of media, such as from one or more
Capture Devices (Section 4.1.2) or constructed from other media
Streams (Section 4.1.10). Describes either an RTP Taxonomy media
capture (Section 2.1.2) or a media source (Section 2.1.4), depending
on in which context the term is used.
4.1.8. Media Consumer
Defined in CLUE as a CLUE-capable device that intends to receive
Capture Encodings (Section 4.1.3). Describes the media receiving
part of an RTP Taxonomy endpoint (Section 2.2.1).
4.1.9. Media Provider
Defined in CLUE as a CLUE-capable device that intends to send Capture
Encodings (Section 4.1.3). Describes the media sending part of an
RTP Taxonomy endpoint (Section 2.2.1).
4.1.10. Stream
Defined in CLUE as a Capture Encoding (Section 4.1.3) sent from a
Media Provider (Section 4.1.9) to a Media Consumer (Section 4.1.8)
via RTP. Describes an RTP stream (Section 2.1.10).
4.1.11. Video Capture
Defined in CLUE as a Media Capture (Section 4.1.7) for video.
Describes a video media source (Section 2.1.4).
4.2. Media Description
A single Session 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.
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
reused for multiple media sources [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).
4.3. Media Stream
RTP [RFC3550] uses media stream, audio stream, video stream, and a
stream of (RTP) packets interchangeably, which are all RTP streams.
4.4. Multimedia Conference
A Multimedia Conference is a communication session (Section 2.2.5)
between two or more participants (Section 2.2.3), along with the
software they are using to communicate.
4.5. 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 endpoints and the RTP
streams that flow between them. In this document, multimedia session
(Section 2.2.4) also assumes those endpoints 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 endpoints) sharing a set of concurrent RTP sessions. In this
document, multimedia session also defines those RTP sessions to have
some relation and be part of a communication among the participants.
4.6. Multipoint Control Unit (MCU)
This term is commonly used to describe the central node in any type
of star topology [RTP-TOPOLOGIES] 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 endpoints
(Section 2.2.1) (sometimes one or more per conference participant).
4.7. Multi-Session Transmission (MST)
One of two transmission modes defined in H.264-based SVC [RFC6190],
the other mode being a Single-Session Transmission (SST)
(Section 4.14). In Multi-Session Transmission (MST), the SVC media
encoder sends encoded streams and dependent streams distributed
across two or more RTP streams in one or more RTP sessions. The term
"MST" is ambiguous in RFC 6190, especially since the name indicates
the use of multiple "sessions", while MST-type packetization is in
fact required whenever two or more RTP streams are used for the
encoded and dependent streams, regardless if those are sent in one or
more RTP sessions. Corresponds either to MRST or MRMT (Section 3.7)
stream relations defined in this document. The SVC RTP payload RFC
[RFC6190] is not particularly explicit about how the common media
encoder (Section 2.1.6) relation between encoded streams
(Section 2.1.7) and dependent streams (Section 2.1.8) is to be
implemented.
4.8. Recording Device
WebRTC specifications use this term to refer to locally available
entities performing a media capture (Section 2.1.2) transformation.
4.9. RtcMediaStream
A WebRTC RtcMediaStream is a set of media sources (Section 2.1.4)
sharing the same synchronization context (Section 3.1).
4.10. RtcMediaStreamTrack
A WebRTC RtcMediaStreamTrack is a media source (Section 2.1.4).
4.11. RTP Receiver
RTP [RFC3550] uses this term, which can be seen as the RTP protocol
part of a media depacketizer (Section 2.1.27).
4.12. RTP Sender
RTP [RFC3550] uses this term, which can be seen as the RTP protocol
part of a media packetizer (Section 2.1.9).
4.13. RTP Session
Within the context of SDP, a singe "m=" line can map to a single RTP
session (Section 2.2.2), or multiple "m=" lines can map to a single
RTP session. The latter is enabled via multiplexing schemes such as
BUNDLE [SDP-BUNDLE], for example, which allows mapping of multiple
"m=" lines to a single RTP session.
4.14. Single-Session Transmission (SST)
One of two transmission modes defined in H.264-based SVC [RFC6190],
the other mode being MST (Section 4.7). In SST, the SVC media
encoder sends encoded streams (Section 2.1.7) and dependent streams
(Section 2.1.8) combined into a single RTP stream (Section 2.1.10) in
a single RTP session (Section 2.2.2), using the SVC RTP payload
format. The term "SST" is ambiguous in RFC 6190, in that it
sometimes refers to the use of a single RTP stream, like in sections
relating to packetization, and sometimes appears to refer to use of a
single RTP session, like in the context of discussing SDP. Closely
corresponds to SRST (Section 3.7) defined in this document.
4.15. 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). In [RFC3550], it is stated that "a
synchronization source may change its data format, e.g., audio
encoding, over time". The related encoded stream data format in an
RTP stream (Section 2.1.10) is identified by the RTP payload type.
Changing the data format for an encoded stream effectively also
changes what media encoder (Section 2.1.6) is used for the encoded
stream. No ambiguity is introduced to SSRC as an encoded stream
identifier by allowing RTP payload type changes, as long as only a
single RTP payload type is valid for any given RTP Timestamp. This
is aligned with and further described by Section 5.2 of [RFC3550].
5. Security Considerations
The purpose of this document is to make clarifications and reduce 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.
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. Informative References
[CLUE-FRAME]
Duckworth, M., Pepperell, A., and S. Wenger, "Framework
for Telepresence Multi-Streams", Work in Progress,
draft-ietf-clue-framework-22, April 2015.
[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,
DOI 10.17487/RFC2198, September 1997,
<http://www.rfc-editor.org/info/rfc2198>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[SDP-BUNDLE]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", Work in Progress,
draft-ietf-mmusic-sdp-bundle-negotiation-23, July 2015.
[SDP-SIMULCAST]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", Work in
Progress, draft-ietf-mmusic-sdp-simulcast-01, July 2015.
[TRANSPORT-MULTIPLEX]
Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
Sessions onto a Single Lower-Layer Transport", Work in
Progress, draft-westerlund-avtcore-transport-multiplexing-
07, October 2013.
[WEBRTC-OVERVIEW]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", Work in Progress,
draft-ietf-rtcweb-overview-14, June 2015.
Acknowledgements
This document has many concepts borrowed from several documents such
as WebRTC [WEBRTC-OVERVIEW], CLUE [CLUE-FRAME], and Multiplexing
Architecture [TRANSPORT-MULTIPLEX]. 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, Alex Eleftheriadis, Mo
Zanaty, Stephan Wenger, and Bernard Aboba.
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.
Authors' Addresses
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
United States
Email: jonathan@vidyo.com
Kevin Gross
AVA Networks, LLC
Boulder, CO
United States
Email: kevin.gross@avanw.com
Suhas Nandakumar
Cisco Systems
170 West Tasman Drive
San Jose, CA 95134
United States
Email: snandaku@cisco.com
Gonzalo Salgueiro
Cisco Systems
7200-12 Kit Creek Road
Research Triangle Park, NC 27709
United States
Email: gsalguei@cisco.com
Bo Burman (editor)
Ericsson
Kistavagen 25
SE-16480 Stockholm
Sweden
Email: bo.burman@ericsson.com
