| Network Working Group | J. Lennox |
| Internet-Draft | Vidyo |
| Intended status: Informational | K. Gross |
| Expires: August 18, 2014 | AVA |
| S. Nandakumar | |
| G. Salgueiro | |
| Cisco Systems | |
| B. Burman | |
| Ericsson | |
| February 14, 2014 |
A Taxonomy of Grouping Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources
draft-ietf-avtext-rtp-grouping-taxonomy-01
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.
This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 18, 2014.
Copyright (c) 2014 IETF Trust and the persons identified as the document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
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].
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.
In the context of this memo, Media is a sequence of synthetic or Physical Stimulus [physical-stimulus] (sound waves, photons, key-strokes), represented in digital form. Synthesized Media is typically generated directly in the digital domain.
This section contains the concepts that can be involved in taking Media at a sender side and transporting it to a receiver, which may recover a sequence of physical stimulus. This chain of concepts is of two main types, streams and transformations. Streams are time-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:
It is also important to remember that this is a conceptual model. Thus real-world implementations may look different and have different structure.
To provide a basic understanding of the relationships in the chain we below first introduce the concepts for the sender side [fig-sender-chain]. 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.
Below we review a receiver media chain [fig-receiver-chain] 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
+--------------------+
| Media Renderer |
+--------------------+
|
V
Physical Stimulus
Figure 2: Receiver Side Concepts of the Media Chain
The physical stimulus is a physical event that can be measured and converted to digital form by an appropriate sensor or transducer. This include sound waves making up audio, photons in a light field that is visible, or other excitations or interactions with sensors, like keystrokes on a keyboard.
Media Capture is the process of transforming the Physical Stimulus [physical-stimulus] into digital Media using an appropriate sensor or transducer. The Media Capture performs a digital sampling of the physical stimulus, usually periodically, and outputs this in some representation as a Raw Stream [raw-stream]. 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.
Alternate usages:
Characteristics:
The time progressing stream of digitally sampled information, usually periodically sampled and provided by a Media Capture [media-capture]. A Raw Stream can also contain synthesized Media that may not require any explicit Media Capture, since it is already in an appropriate digital form.
A Media Source is the logical source of a reference clock synchronized, time progressing, digital media stream, called a Source Stream [source-stream]. This transformation takes one or more Raw Streams [raw-stream] 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 [fig-media-source-mixer], 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
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 Source-Specific attributes are in place [RFC5576]. In the latter cases, an m=line can represent either multiple Media Sources, multiple Packet Streams [packet-stream], or both.
Characteristics:
A time progressing stream of digital samples that has been synchronized with a reference clock and comes from particular Media Source [media-source].
A Media Encoder is a transform that is responsible for encoding the media data from a Source Stream [source-stream] into another representation, usually more compact, that is output as an Encoded Stream [encoded-stream].
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 [dependent-stream] 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 |
+--------------------------+
| | ... |
V V V
Encoded Dependent Dependent
Stream Stream Stream
Figure 4: Scalable Media Encoder Input and Outputs
There are also other variants of encoders, like so-called Multiple Description Coding (MDC). Such Media Encoder produce multiple independent and thus individually decodable Encoded Streams that are possible to combine into a Received Source Stream that is somehow a better representation of the original Source Stream than using only a single Encoded Stream.
Alternate usages:
Characteristics:
A stream of time synchronized encoded media that can be independently decoded.
Characteristics:
A stream of time synchronized encoded media fragments that are dependent on one or more Encoded Streams [encoded-stream] and zero or more Dependent Streams to be possible to decode.
Characteristics:
The transformation of taking one or more Encoded [encoded-stream] or Dependent Stream [dependent-stream] and put their content into one or more sequences of packets, normally RTP packets, and output Source Packet Streams [packet-stream]. 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 SST packetization when using SVC [sstmst].
The Media Packetizer can also produce multiple Packet Streams, for example when Encoded and/or Dependent Streams are distributed over multiple Packet Streams. One example of this is MST packetization when using SVC [sstmst].
Alternate usages:
Characteristics:
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 [redundancy-packet-stream].
Alternate usages:
Characteristics:
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.
A Packet Stream [packet-stream] that contains no original source data, only redundant data that may be combined with one or more Received Packet Stream [received-packet-stream] to produce Repaired Packet Streams [repaired-packet-stream].
A Media Transport defines the transformation that the Packet Streams [packet-stream] are subjected to by the end-to-end transport from one RTP sender to one specific RTP receiver (an RTP session may contain multiple RTP receivers per sender). Each Media Transport is defined by a transport association that is identified by a 5-tuple (source address, source port, destination address, destination port, transport protocol). Each transport association normally contains only a single RTP session, although a proposal exists for sending multiple RTP sessions over one transport association [I-D.westerlund-avtcore-transport-multiplexing].
Characteristics:
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 [fig-media-transport].
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
The first transformation within the Media Transport [media-transport] is the Media Transport Sender, where the sending End-Point [end-point] 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 [sent-packet-stream]. 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.
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)).
Network Transport is the transformation that the Sent Packet Stream [sent-packet-stream] 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 [transported-packet-stream] at the exit of the network path.
The Packet Stream that is emitted out of the network path at the destination, subjected to the Network Transport's transformation [network-transport].
The receiver End-Point's [end-point] transformation of the Transported Packet Stream [transported-packet-stream] by its reception process that result in the Received Packet Stream [received-packet-stream]. 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.
The Packet Stream [packet-stream] 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.
The Redundancy Packet Stream [redundancy-packet-stream] resulting from the Media Transport's transformation, i.e. subjected to packet loss, packet corruption, and varying transmission delay from sender to receiver.
A Transformation that takes as input one or more Source Packet Streams [packet-stream] as well as Redundancy Packet Streams [redundancy-packet-stream] and attempts to combine them to counter the transformations introduced by the Media Transport [media-transport] to minimize the difference between the Source Stream [source-stream] and the Received Source Stream [received-source-stream] after Media Decoder [media-decoder]. The output is a Repaired Packet Stream [repaired-packet-stream].
A Received Packet Stream [received-packet-stream] for which Received Redundancy Packet Stream [received-redundancy-ps] information has been used to try to re-create the Packet Stream [packet-stream] as it was before Media Transport [media-transport].
A Media Depacketizer takes one or more Packet Streams [packet-stream] and depacketizes them and attempts to reconstitute the Encoded Streams [encoded-stream] or Dependent Streams [dependent-stream] present in those Packet Streams.
It should be noted that in practical implementations, the Media Depacketizer and the Media Decoder may be tightly coupled and share information to improve or optimize the overall decoding process in various ways. It is however not expected that there would be any benefit in defining a taxonomy for those detailed (and likely very implementation-dependent) steps.
The received version of an Encoded Stream [encoded-stream].
A Media Decoder is a transformation that is responsible for decoding Encoded Streams [encoded-stream] and any Dependent Streams [dependent-stream] into a Source Stream [source-stream].
It should be noted that in practical implementations, the Media Decoder and the Media Depacketizer may be tightly coupled and share information to improve or optimize the overall decoding process in various ways. It is however not expected that there would be any benefit in defining a taxonomy for those detailed (and likely very implementation-dependent) steps.
Alternate usages:
Characteristics:
The received version of a Source Stream [source-stream].
The Media Sink receives a Source Stream [source-stream] 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 [raw-stream] that is sent in synchronization with the output from other Media Sinks to a Media Render [media-render]. The media sink may also be connected with a Media Source [media-source] and be used as part of a conceptual Media Source.
Characteristics:
The received version of a Raw Stream [raw-stream].
A Media Render takes a Raw Stream [raw-stream] and converts it into Physical Stimulus [physical-stimulus] that a human user can perceive. Examples of such devices are screens, D/A converters connected to amplifiers and loudspeakers.
Characteristics:
This section contains concept for entities involved in the communication.
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".
Alternate usages:
Characteristics:
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.
Alternate usages:
Characteristics:
A participant is an entity reachable by a single signaling address, and is thus related more to the signaling context than to the media context.
Characteristics:
A multimedia session is an association among a group of participants engaged in the communication via one or more RTP Sessions [rtp-session]. It defines logical relationships among Media Sources [media-source] that appear in multiple RTP Sessions.
Alternate usages:
Characteristics:
A Communication Session is an association among group of participants communicating with each other via a set of Multimedia Sessions.
Alternate usages:
Characteristics:
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.
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.
Media Sources [media-source] are commonly grouped and related to an End Point [end-point] or a Participant [participant]. This occurs for several reasons; both application logic as well as media handling purposes. These cases are further discussed below.
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.
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.
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.
[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.
In CLUE "Capture Scene", "Capture Scene Entry" and "Captures" define an implied Synchronization Context.
The WebRTC WG defines "RtcMediaStream" with one or more "RtcMediaStreamTracks". All tracks in a "RtcMediaStream" are intended to be possible to synchronize when rendered.
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.
Some applications requires knowledge of what Media Sources originate from a particular End Point [end-point]. This can include such 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.
In communication scenarios, it is commonly needed to know which Media Sources that originate from which Participant [participant]. 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.
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.
At RTP Packetization time, there exists a possibility for a number of different types of relationships between Encoded Streams [encoded-stream], Dependent Streams [dependent-stream] and Packet Streams [packet-stream]. These are caused by grouping together or distributing these different types of streams into Packet Streams. This section will look at such relationships.
Scalable Video Coding [RFC6190] has a mode of operation called Single Session Transmission (SST), where Encoded Streams and Dependent Streams from the SVC Media Encoder are sent in a single RTP Session [rtp-session] using the SVC RTP Payload format. There is another mode of operation where Encoded Streams and Dependent Streams are distributed across multiple RTP Sessions, called Multi-Session Transmission (MST). Regardless if used with SST or MST, as they are defined, each of those RTP Sessions may contain one or more Packet Streams (SSRC) per Media Source.
To elaborate, what could be called SST-SingleStream (SST-SS) uses a single Packet Stream in a single RTP Session to send all Encoded and Dependent Streams. Similarly, SST-MultiStream (SST-MS) uses multiple Packet Streams in a single RTP Session to send the Encoded and Dependent Streams. MST-SS uses a single Packet Stream in each of multiple RTP Sessions and MST-MS uses multiple Packet Streams in each of the multiple RTP Sessions:
| Single RTP Session | Multiple RTP Sessions | |
|---|---|---|
| Single Packet Stream | SST-SS | MST-SS |
| Multiple Packet Streams | SST-MS | MST-MS |
There exist a number of RTP payload formats that can carry multi-channel audio, despite the codec being a mono encoder. Multi-channel audio can be viewed as multiple Media Sources sharing a common Synchronization Context. These are independently encoded by a Media Encoder and the different Encoded Streams are then packetized together in a time synchronized way into a single Source 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].
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
+--------------------+
| 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 [fig-red-rfc2198] 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.
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.
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 |
+------------------+ +------------------+ +------------------+
| 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.
Layered Multi-Stream (LMS) 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). LMSs are useful for receiver control of layered media.
A Media Source represented as an Encoded Stream and multiple Dependent Streams constitutes a Media Source that has layered dependencies. The figure below represents an example of a Media Source that is encoded into three dependent layers, where two layers are sent on the same Media Transport using different 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
+---------------------------------------------------------+
| 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
As an example, the SVC MST [sstmst] 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 (MST-SS [sstmst]), 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 (SST-MS), then CNAME is the only currently specified RTP identifier that can be used. In cases where multiple Media Encoders use multiple Media Sources sharing Synchronization Context, and thus having a common CNAME, additional heuristics need to be applied to create the MST relationship between the Packet Streams.
Packet Streams may be protected by Redundancy Packet Streams during transport. Several approaches listed below can achieve the same result;
The figure below [fig-rtx] 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
The RTP Retransmission example [fig-rtx] 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.
The figure below [fig-fec] 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 |
+--------------------+ +--------------------+
| |
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.
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.
Several mechanisms that use RTP Session-based separation rely on it to enable an implicit grouping mechanism expressing the relationship. The solutions have been based on using the same SSRC value in the different RTP Sessions to implicitly indicate their relation. That way, no explicit RTP level mechanism has been needed, only signaling level relations have been established using semantics from Grouping of Media lines framework [RFC5888]. Examples of this are RTP Retransmission [RFC4588], SVC Multi-Session Transmission [RFC6190] and XOR Based FEC [RFC5109]. RTCP CNAME explicitly relates 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.
[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 signaling design and when defining terminology.
This section reviews some communication topologies and looks at the relationship among the communication entities that are defined in Section 2.2. It does not 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 [media-transport] that exists between the End Points [end-point] that are part of an RTP session [rtp-session] and their relationship to the Multi-Media Session [multimedia-session] between Participants [participant] and the established Communication session [comm-session] are explained.
The text provided below is neither any exhaustive listing of possible topologies, nor does it cover all topologies described in [I-D.ietf-avtcore-rtp-topologies-update].
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
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.
This section looks at the centralized 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-------| | | | | | | | +-----------++-----------------------------++------+ | | | | | | | || || | | | | | | | | +-----------++-----------------------------++----+ | | | | | | | | | 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: Centralized 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 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.
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 Media Sources, 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 | | |
| | | 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 [fig-three-relay]. 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 | | | | | | | +-------+ | | | | +-------+ | | | | | | | | 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.
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
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.
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.
This document has many concepts borrowed from several documents such as WebRTC [I-D.ietf-rtcweb-overview], CLUE [I-D.ietf-clue-framework], Multiplexing Architecture [I-D.westerlund-avtcore-transport-multiplexing]. The authors would like to thank all the authors of each of those documents.
The authors would also like to acknowledge the insights, guidance and contributions of Magnus Westerlund, Roni Even, Paul Kyzivat, Colin Perkins, Keith Drage, Harald Alvestrand, and Alex Eleftheriadis.
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.
This document makes no request of IANA.
| [RFC3550] | Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. |
NOTE TO RFC EDITOR: Please remove this section prior to publication.