RTP Multiplexing Architecture
draft-westerlund-avtcore-multiplex-architecture-00
The information below is for an old version of the document.
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| Authors | Magnus Westerlund , Bo Burman , Colin Perkins | ||
| Last updated | 2011-10-24 | ||
| Replaced by | draft-ietf-avtcore-multiplex-guidelines, draft-ietf-avtcore-multiplex-guidelines, RFC 8872 | ||
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draft-westerlund-avtcore-multiplex-architecture-00
Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: BCP Ericsson
Expires: April 26, 2012 C. Perkins
University of Glasgow
October 24, 2011
RTP Multiplexing Architecture
draft-westerlund-avtcore-multiplex-architecture-00
Abstract
RTP has always been a protocol that supports multiple participants
each sending their own media streams in an RTP session. Thus relying
on the three main multiplexing points in RTP; RTP session, SSRC and
Payload Type for their various needs. However, most usages of RTP
have been less complex often with a single SSRC in each direction,
with a single RTP session per media type. But the more complex
usages start to be more common and thus guidance on how to use RTP in
various complex cases are needed. This document analyzes a number of
cases and discusses the usage of the various multiplexing points and
the need for functionality when defining RTP/RTCP extensions that
utilize multiple RTP streams and multiple RTP sessions.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 26, 2012.
Copyright Notice
Copyright (c) 2011 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
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Provisions Relating to IETF Documents
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publication of this document. Please review these documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Requirements Language . . . . . . . . . . . . . . . . . . 5
2.2. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5
3. RTP Multiplex Points . . . . . . . . . . . . . . . . . . . . . 6
3.1. Session . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.2. SSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.3. CSRC . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
3.4. Payload Type . . . . . . . . . . . . . . . . . . . . . . . 8
4. Multiple Streams Alternatives . . . . . . . . . . . . . . . . 9
5. RTP Topologies and Issues . . . . . . . . . . . . . . . . . . 10
5.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 11
5.1.1. RTCP Reporting . . . . . . . . . . . . . . . . . . . . 11
5.1.2. Compound RTCP Packets . . . . . . . . . . . . . . . . 12
5.2. Point to Multipoint Using Multicast . . . . . . . . . . . 12
5.3. Point to Multipoint Using an RTP Translator . . . . . . . 14
5.4. Point to Multipoint Using an RTP Mixer . . . . . . . . . . 15
5.5. Point to Multipoint using Multiple Unicast flows . . . . . 16
5.6. Decomposited End-Point . . . . . . . . . . . . . . . . . . 16
6. Dismissing Payload Type Multiplexing . . . . . . . . . . . . . 18
7. Multiple Streams Discussion . . . . . . . . . . . . . . . . . 20
7.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 20
7.2. RTP/RTCP Aspects . . . . . . . . . . . . . . . . . . . . . 20
7.2.1. The RTP Specification . . . . . . . . . . . . . . . . 20
7.2.2. Multiple SSRC Legacy Considerations . . . . . . . . . 22
7.2.3. RTP Specification Clarifications Needed . . . . . . . 23
7.2.4. Handling Varying sets of Senders . . . . . . . . . . . 23
7.2.5. Cross Session RTCP requests . . . . . . . . . . . . . 23
7.2.6. Binding Related Sources . . . . . . . . . . . . . . . 23
7.2.7. Forward Error Correction . . . . . . . . . . . . . . . 25
7.2.8. Transport Translator Sessions . . . . . . . . . . . . 26
7.2.9. Multiple Media Types in one RTP session . . . . . . . 26
7.3. Signalling Aspects . . . . . . . . . . . . . . . . . . . . 28
7.3.1. Session Oriented Properties . . . . . . . . . . . . . 28
7.3.2. SDP Prevents Multiple Media Types . . . . . . . . . . 29
7.4. Network Apsects . . . . . . . . . . . . . . . . . . . . . 29
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7.4.1. Quality of Service . . . . . . . . . . . . . . . . . . 29
7.4.2. NAT and Firewall Traversal . . . . . . . . . . . . . . 29
7.4.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 31
7.4.4. Multiplexing multiple RTP Session on a Single
Transport . . . . . . . . . . . . . . . . . . . . . . 32
7.5. Security Aspects . . . . . . . . . . . . . . . . . . . . . 32
7.5.1. Security Context Scope . . . . . . . . . . . . . . . . 32
7.5.2. Key-Management for Multi-party session . . . . . . . . 33
8. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 33
9. RTP Specification Clarifications . . . . . . . . . . . . . . . 35
9.1. RTCP Reporting from all SSRCs . . . . . . . . . . . . . . 35
9.2. RTCP Self-reporting . . . . . . . . . . . . . . . . . . . 35
9.3. Combined RTCP Packets . . . . . . . . . . . . . . . . . . 35
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35
11. Security Considerations . . . . . . . . . . . . . . . . . . . 36
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36
13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 36
13.1. Normative References . . . . . . . . . . . . . . . . . . . 36
13.2. Informative References . . . . . . . . . . . . . . . . . . 36
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 39
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1. Introduction
This document focuses at issues of non-basic usage of RTP [RFC3550]
where multiple media sources of the same media type are sent over
RTP. Separation of different media types is another issue that will
be discussed in this document. The intended uses include for example
multiple sources from the same end-point, multiple streams from a
single media source, multiple end-points each having a source, or an
application that needs multiple representations (encodings) of a
particular source. It will be shown that these uses are inter-
related and need a common discussion to ensure consistency. In
general, usage of the RTP session and media streams will be discussed
in detail.
RTP is already designed for multiple participants in a communication
session. This is not restricted to multicast, as many believe, but
also provides functionality over unicast, using either multiple
transport flows below RTP or a network node that re-distributes the
RTP packets. The node can for example be a transport translator
(relay) that forwards the packets unchanged, a translator performing
media translation in addition to forwarding, or an RTP mixer that
creates new conceptual sources from the received streams. In
addition, multiple streams may occur when a single end-point have
multiple media sources of the same media type, like multiple cameras
or microphones that need to be sent simultaneously.
Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per end-point and media type (typically audio
and video). Even in conferencing applications, especially voice
only, the conference focus or bridge has provided a single stream
with a mix of the other participants to each participant. It is also
common to have individual RTP sessions between each end-point and the
RTP mixer.
SSRC is the RTP media stream identifier that helps to uniquely
identify media sources in RTP sessions. Even though available SSRC
space can theoretically handle more than 4 billion simultaneous
sources, the perceived need for handling multiple SSRCs in
implementations has been small. This has resulted in an installed
legacy base that isn't fully RTP specification compliant and will
have different issues if they receive multiple SSRCs of media, either
simultaneously or in sequence. These issues will manifest themselves
in various ways, either by software crashes or simply in limited
functionality, like only decoding and playing back the first or
latest received SSRC and discarding media related to any other SSRCs.
There have also arisen various cases where multiple SSRCs are used to
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represent different aspects of what is in fact a single underlying
media source. A very basic case is RTP retransmission [RFC4588]
which have one SSRC for the original stream, and a second SSRC either
in the same session or in a different session to represent the
retransmitted packets to ensure that the monitoring functions still
function. Another use case is scalable encoding, such as the RTP
payload format for Scalable Video Coding (SVC) [RFC6190], which has
an operation mode named Multiple Session Transmission (MST) that uses
one SSRC in each RTP session to send one or more scalability layers.
A third example is simulcast where a single media source is encoded
in different versions and transmitted to an RTP mixer that picks
which version to actually distribute to a given receiver part of the
RTP session.
This situation has created a need for a document that discusses the
existing possibilities in the RTP protocol and how these can and
should be used in applications. A new set of applications needing
more advanced functionalities from RTP is also emerging on the
market, such as telepresence and advanced video conferencing. Thus
furthering the need for a more common understanding of how multiple
streams are handled in RTP to ensure media plane interoperability.
The document starts with some definitions and then goes into the
existing RTP functionalities around multiplexing. Both the desired
behavior and the implications of a particular behavior depend on
which topologies are used, which requires some consideration. This
is followed by a discussion of some choices in multiplexing behavior
and their impacts. Finally, some recommendations and examples are
provided.
2. Definitions
2.1. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
2.2. Terminology
The following terms and abbreviations are used in this document:
End-point: A single entity sending or receiving RTP packets. It may
be decomposed into several functional blocks, but as long as it
behaves a single RTP stack entity it is classified as a single
end-point.
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Media Stream: A sequence of RTP packets using a single SSRC that
together carry part or all of the content of a specific Media Type
from a specific sender source within a given RTP session.
Media Aggregate: All Media Streams related to a particular Source.
Media Type: Audio, video, text or data whose form and meaning are
defined by a specific real-time application.
Source: The source of a particular media stream. Either a single
media capturing device such as a video camera, or a microphone, or
a specific output of a media production function, such as an audio
mixer, or some video editing function.
3. RTP Multiplex Points
This section describes the existing RTP tools that enable
multiplexing of different media streams and RTP functionalities.
3.1. Session
The RTP Session is the highest semantic level in RTP and contains all
of the RTP functionality.
RTP in itself does not contain any Session identifier, but relies on
the underlying transport. For example, when running RTP on top of
UDP, an RTP endpoint can identify and delimit an RTP Session from
other RTP Sessions through the UDP source and destination transport
address, consisting of network address and port number(s). Most
commonly only the destination address, i.e. all traffic received on a
particular port, is defined as belonging to a specific RTP Session.
It is worth noting that in practice a more narrow definition of the
transport flows that are related to a give RTP session is possible.
An RTP session can for example be defined as one or more 5-tuples
(Transport Protocol, Source Address, Source Port, Destination
Address, Destination Port). Any set of identifiers of RTP and RTCP
packet flows are sufficient to determine if the flow belongs to a
particular session or not.
Commonly, RTP and RTCP use separate ports and the destination
transport address is in fact an address pair, but in the case of RTP/
RTCP multiplex [RFC5761] there is only a single port.
A source that changes its source transport address during a session
must also choose a new SSRC identifier to avoid being interpreted as
a looped source.
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The set of participants considered part of the same RTP Session is
defined by[RFC3550] as those that share a single SSRC space. That
is, those participants that can see an SSRC identifier transmitted by
any one of the other participants. A participant can receive an SSRC
either as SSRC or CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the participants' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by end-points and any interconnecting middle nodes.
3.2. SSRC
The Synchronization Source (SSRC) identifier is used to identify
individual sources within an RTP Session. The SSRC number is
globally unique within an RTP Session and all RTP implementations
must be prepared to use procedures for SSRC collision handling, which
results in an SSRC number change. The SSRC number is randomly
chosen, carried in every RTP packet header and is not dependent on
network address. SSRC is also used as identifier to refer to
separate media streams in RTCP.
A media source having an SSRC identifier can be of different types:
Real: Connected to a "physical" media source, for example a camera
or microphone.
Conceptual: A source with some attributed property generated by some
network node, for example a filtering function in an RTP mixer
that provides the most active speaker based on some criteria, or a
mix representing a set of other sources.
Virtual: A source that does not generate any RTP media stream in
itself (e.g. an end-point only receiving in an RTP session), but
anyway need a sender SSRC for use as source in RTCP reports.
Note that a "multimedia source" that generates more than one media
type, e.g. a conference participant sending both audio and video,
need not (and commonly should not) use the same SSRC value across RTP
sessions. RTCP Compound packets containing the CNAME SDES item is
the designated method to bind an SSRC to a CNAME, effectively cross-
correlating SSRCs within and between RTP Sessions as coming from the
same end-point. The main property attributed to SSRCs associated
with the same CNAME is that they are from a particular
synchronization context and may be synchronized at playback. There
exist a few other methods to relate different SSRC where use of CNAME
is inappropriate, such as session-based RTP retransmission [RFC4588].
Note also that RTP sequence number and RTP timestamp are scoped by
SSRC and thus independent between different SSRCs.
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An RTP receiver receiving a previously unseen SSRC value must
interpret it as a new source. It may in fact be a previously
existing source that had to change SSRC number due to an SSRC
conflict. However, the originator of the previous SSRC should have
ended the conflicting source by sending an RTCP BYE for it prior to
starting to send with the new SSRC, so the new SSRC is anyway
effectively a new source.
Some RTP extension mechanisms already require the RTP stacks to
handle additional SSRCs, like SSRC multiplexed RTP retransmission
[RFC4588]. However, that still only requires handling a single media
decoding chain per pair of SSRCs.
3.3. CSRC
The Contributing Source (CSRC) can arguably be seen as a sub-part of
a specific SSRC and thus a multiplexing point. It is optionally
included in the RTP header, shares the SSRC number space and
specifies which set of SSRCs that has contributed to the RTP payload.
However, even though each RTP packet and SSRC can be tagged with the
contained CSRCs, the media representation of an individual CSRC is in
general not possible to extract from the RTP payload since it is
typically the result of a media mixing (merge) operation (by an RTP
mixer) on the individual media streams corresponding to the CSRC
identifiers. Due to these restrictions, CSRC will not be considered
a fully qualified multiplex point and will be disregarded in the rest
of this document.
3.4. Payload Type
The Payload Type number is also carried in every RTP packet header
and identifies what format the RTP payload has. The term "format"
here includes whatever can be described by out-of-band signaling
means for dynamic payload types, as well as the statically allocated
payload types in [RFC3551]. In SDP the term "format" includes media
type, RTP timestamp sampling rate, codec, codec configuration,
payload format configurations, and various robustness mechanisms such
as redundant encodings [RFC2198].
The meaning of a Payload Type definition (the number) is re-used
between all media streams within an RTP session, when the definition
is either static or signaled through SDP. There however do exist
cases where each end-point have different sets of payload types due
to SDP offer/answer.
Although Payload Type definitions are commonly local to an RTP
Session, there are some uses where Payload Type numbers need be
unique across RTP Sessions. This is for example the case in Media
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Decoding Dependency [RFC5583] where Payload Types are used to
describe media dependency across RTP Sessions.
Given that multiple Payload Types are defined in an RTP Session, a
media sender is free to change the Payload Type on a per packet
basis. One example of designed per-packet change of Payload Type is
a speech codec that makes use of generic Comfort Noise [RFC3389].
The RTP Payload Type in RTP is designed such that only a single
Payload Type is valid at any time instant in the SSRC's timestamp
time line, effectively time-multiplexing different Payload Types if
any switch occurs. Even when this constraint is met, having
different rates on the RTP timestamp clock for the RTP Payload Types
in use in the same RTP Session have issues such as loss of
synchronization. Payload Type clock rate switching requires some
special consideration that is described in the multiple clock rates
specification [I-D.ietf-avtext-multiple-clock-rates].
If there is a true need to send multiple Payload Types for the same
SSRC that are valid for the same RTP Timestamps, then redundant
encodings [RFC2198] can be used. Several additional constraints than
the ones mentioned above need to be met to enable this use, one of
which are that the combined payload sizes of the different Payload
Types must not exceed the transport MTU.
Other aspects of RTP payload format use are described in RTP Payload
HowTo [I-D.ietf-payload-rtp-howto].
4. Multiple Streams Alternatives
This section reviews the alternatives to enable multi-stream
handling. Let's start with describing mechanisms that could enable
multiple media streams, independent of the purpose for having
multiple streams.
SSRC Multiplexing: Each additional Media Stream gets its own SSRC
within a RTP Session.
Session Multiplexing: Using additional RTP Sessions to handle
additional Media Streams
Payload Type Multiplexing: Using different RTP payload types for
different additional streams.
Independent of the reason to use additional media streams, achieving
it using payload type multiplexing is not a good choice as can be
seen in the below section (Section 6). The RTP payload type alone is
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not suitable for cases where additional media streams are required.
Streams need their own SSRCs, so that they get their own sequence
number space. The SSRC itself is also important so that the media
stream can be referenced and reported on.
This leaves us with two choices, either using SSRC multiplexing to
have multiple SSRCs from one end-point in one RTP session, or create
additional RTP sessions to hold that additional SSRC. As the below
discussion will show, in reality we cannot choose a single one of the
two solutions. To utilize RTP well and as efficiently as possible,
both are needed. The real issue is finding the right guidance on
when to create RTP sessions and when additional SSRCs in an RTP
session is the right choice.
In the below discussion, please keep in mind that the reasons for
having multiple media streams vary and include but are not limited to
the following:
o Multiple Media Sources of the same media type
o Retransmission streams
o FEC stream
o Alternative Encoding
o Scalability layer
Thus the choice made due to one reason may not be the choice suitable
for another reason. In the above list, the different items have
different levels of maturity in the discussion on how to solve them.
The clearest understanding is associated with multiple media sources
of the same media type. However, all warrant discussion and
clarification on how to deal with them.
5. RTP Topologies and Issues
The impact of how RTP Multiplex is performed will in general vary
with how the RTP Session participants are interconnected; the RTP
Topology [RFC5117]. This section describes the topologies and
attempts to highlight the important behaviors concerning RTP
multiplexing and multi-stream handling. It lists any identified
issues regarding RTP and RTCP handling, and introduces additional
topologies that are supported by RTP beyond those included in RTP
Topologies [RFC5117]. The RTP Topologies that do not follow the RTP
specification or do not attempt to utilize the facilities of RTP are
ignored in this document.
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5.1. Point to Point
This is the most basic use case with an RTP session containing of two
end-points. Each end-point has one or more SSRCs.
+---+ +---+
| A |<------->| B |
+---+ +---+
Point to Point
5.1.1. RTCP Reporting
In cases when an end-point uses multiple SSRCs, we have found two
closely related issues. The first is if every SSRC shall report on
all other SSRC, even the ones originating from the same end-point.
The reason for this would be ensure that no monitoring function
should suspect a breakage in the RTP session.
The second issue around RTCP reporting arise when an end-point
receives one or more media streams, and when the receiving end-point
itself sends multiple SSRC in the same RTP session. As transport
statistics are gathered per end-point and shared between the nodes,
all the end-point's SSRC will report based on the same received data,
the only difference will be which SSRCs sends the report. This could
be considered unnecessary overhead, but for consistency it might be
simplest to always have all sending SSRCs send RTCP reports on all
media streams the end-point receives.
The current RTP text is silent about sending RTCP Receiver Reports
for an endpoint's own sources, but does not preclude either sending
or omitting them. The uncertainty in the expected behavior in those
cases have likely caused variations in the implementation strategy.
This could cause an interoperability issue where it is not possible
to determine if the lack of reports are a true transport issue, or
simply a result of implementation.
Although this issue is valid already for the simple point to point
case, it needs to be considered in all topologies. From the
perspective of an end-point, any solution needs to take into account
what a particular end-point can determine without explicit
information of the topology. For example, a Transport Translator
(Relay) topology will look quite similar as point to point on an RTP
level but is different. The main difference between a point to point
with two SSRC being sent from the remote end-point and a Transport
Translator with two single SSRC remote clients are that the RTT may
vary between the SSRCs (but it is not guaranteed), and that the SSRCs
may have different CNAMEs.
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5.1.2. Compound RTCP Packets
When an end-point has multiple SSRCs and it needs to send RTCP
packets on behalf of these SSRCs, the question arises if and how RTCP
packets with different source SSRCs can be sent in the same compound
packet. If it is allowed, then some consideration of the
transmission scheduling is needed.
5.2. Point to Multipoint Using Multicast
This section discusses the Point to Multi-point using Multicast to
interconnect the session participants. This needs to consider both
Any Source Multicast (ASM) and Source-Specific Multicast (SSM).
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Point to Multipoint Using Any Source Multicast
In Any Source Multicast, any of the participants can send to all the
other participants, simply by sending a packet to the multicast
group. That is not possible in Source Specific Multicast [RFC4607]
where only a single source (Distribution Source) can send to the
multicast group, creating a topology that looks like the one below:
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Source-specific
+--------+ +-----+ Multicast
|Media | | | +----------------> R(1)
|Sender 1|<----->| D S | | |
+--------+ | I O | +--+ |
| 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|<----->| | |<-------------------------+
+--------+ +-----+ Unicast
FT = Feedback Target
Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not
co-located.
Point to Multipoint using Source Specific Multicast
In this topology a number of Media Senders (1 to M) are allowed to
send media to the SSM group, sends media to the distribution source
which then forwards the media streams to the multicast group. The
media streams reach the Receivers (R(1) to R(n)). The Receiver's
RTCP cannot be sent to the multicast group. To support RTCP, an RTP
extension for SSM [RFC5760] was defined that use unicast transmission
to send RTCP from the receivers to one or more Feedback Targets (FT).
As multicast is a one to many distribution system this must be taken
into consideration. For example, the only practical method for
adapting the bit-rate sent towards a given receiver is to use a set
of multicast groups, where each multicast group represents a
particular bit-rate. The media encoding is either scalable, where
multiple layers can be combined, or simulcast where a single version
is selected. By either selecting or combing multicast groups, the
receiver can control the bit-rate sent on the path to itself. It is
also common that transport robustification is sent in its own
multicast group to allow for interworking with legacy or to support
different levels of protection.
The result of this is three common behaviors for RTP multicast:
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1. Use of multiple RTP sessions for the same media type.
2. The need for identifying RTP sessions that are related in one of
several ways.
3. The need for binding related SSRCs in different RTP sessions
together.
This indicates that Multicast is an important consideration when
working with the RTP multiplexing and multi stream architecture
questions. It is also important to note that so far there is no
special mode for basic behavior between multicast and unicast usages
of RTP. Yes, there are extensions targeted to deal with multicast
specific cases but the general applicability does need to be
considered.
5.3. Point to Multipoint Using an RTP Translator
Transport Translators (Relays) are a very important consideration for
this document as they result in an RTP session situation that is very
similar to how an ASM group RTP session would behave.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Transport Translator (Relay)
One of the most important aspects with the simple relay is that it is
both easy to implement and require minimal amount of resources as
only transport headers are rewritten, no RTP modifications nor media
transcoding occur. Thus it is most likely the cheapest and most
generally deployable method for multi-point sessions. The most
obvious downside of this basic relaying is that the translator has no
control over how many streams needs to be delivered to a receiver.
Nor can it simply select to deliver only certain streams, at least
not without new RTCP extensions to coherently handle the fact that
some middlebox temporarily stops a stream, preventing some receivers
from reporting on it. This consistency problem in RTCP reporting
needs to be handled.
The Transport Translator does not need to have an SSRC of itself, nor
need it send any RTCP reports on the flows that passes it, but it may
choose to do that.
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Use of a transport translator results in that any of the end-points
will receive multiple SSRCs over a single unicast transport flow from
the translator. That is independent of the other end-points having
only a single or several SSRCs. End-points that have multiple SSRCs
put further requirements on how SSRCs can be related or bound within
and across RTP sessions and how they can be identified on an
application level.
A Media Translator can perform a large variety of media functions
affecting the media stream passing the translator, coming from one
source and destined to a particular end-point. The media stream can
be transcoded to a different bit-rate, change to another encoder,
change the packetization of the media stream, add FEC streams, or
terminate RTP retransmissions. The latter behaviors require the
translator to use SSRCs that only exist in a particular sub-domain of
the RTP session, and it may also create additional sessions when the
mechanism applied on one side so requires.
5.4. Point to Multipoint Using an RTP Mixer
The most commonly used topology in centralized conferencing is based
on the RTP Mixer. The main reason for this is that it provides a
very consistent view of the RTP session towards each participant.
That is accomplished through the mixer having its own SSRCs and any
media sent to the participants will be sent using those SSRCs. If
the mixer wants to identify the underlying media sources for its
conceptual streams, it can identify them using CSRC. The media
stream the mixer provides can be an actual media mixing of multiple
media sources. It might also be as simple as selecting one of the
underlying sources based on some mixer policy or control signalling.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
RTP Mixer
In the case where the mixer does stream selection, an application may
in fact desire multiple simultaneous streams but only as many as the
mixer can handle. As long as the mixer and an end-point can agree on
the maximum number of streams and how the streams that are delivered
are selected, this provides very good functionality. As these
streams are forwarded using the mixer's SSRCs, there are no
inconsistencies within the session.
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5.5. Point to Multipoint using Multiple Unicast flows
Based on the RTP session definition, it is clearly possible to have a
joint RTP session over multiple transport flows like the below three
end-point joint session. In this case, A needs to send its' media
streams and RTCP packets to both B and C over their respective
transport flows. As long as all participants do the same, everyone
will have a joint view of the RTP session.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Point to Multi-Point using Multiple Unicast Transprots
This doesn't create any additional requirements beyond the need to
have multiple transport flows associated with a single RTP session.
Note that an end-point may use a single local port to receive all
these transport flows, or it might have separate local reception
ports for each of the end-points.
5.6. Decomposited End-Point
There is some possibility that an RTP end-point implementation in
fact reside on multiple devices, each with their own network address.
A very basic use case for this would be to separate audio and video
processing for a particular end-point, like a conference room, into
one device handling the audio and another handling the video being
interconnected by some control functions allowing them to behave as a
single end-point.
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+---------------------+
| End-point A |
| Local Area Network |
| +------------+ |
| +->| Audio |<+----\
| | +------------+ | \ +------+
| | +------------+ | +-->| |
| +->| Video |<+--------->| B |
| | +------------+ | +-->| |
| | +------------+ | / +------+
| +->| Control |<+----/
| +------------+ |
+---------------------+
Decomposited End-Point
In the above usage, let us assume that the RTP sessions are different
for audio and video. The audio and video parts will use a common
CNAME and also have a common clock to ensure that synchronization and
clock drift handling works despite the decomposition. However, if
the audio and video were in a single RTP session then this use case
becomes problematic. This as all transport flow receivers will need
to receive all the other media streams that are part of the session.
Thus the audio component will receive also all the video media
streams, while the video component will receive all the audio ones,
thus doubling the site's bandwidth requirements from all other
session participants. With a joint RTP session it also becomes
evident that a given end-point, as interpreted from a CNAME
perspective, has two sets of transport flows for receiving the
streams and the decomposition isn't hidden.
The requirements that can derived from the above usage is that the
transport flows for each RTP session might be under common control
but still go to what looks like different end-points based on
addresses and ports. A conclusion can also be reached that
decomposition without using separate RTP sessions has downsides and
potential for RTP/RTCP issues.
There exist another use case which might be considered as a
decomposited end-point. However, as will be shown this should be
considered a translator instead. An example of this is when an end-
point A sends a media flow to B. On the path there is a device C that
on A's behalf does something with the media streams, for example adds
an RTP session with FEC information for A's media streams. C will in
this case need to bind the new FEC streams to A's media stream by
using the same CNAME as A.
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+------+ +------+ +------+
| | | | | |
| A |------->| C |-------->| B |
| | | |---FEC-->| |
+------+ +------+ +------+
When Decomposition is a Translator
This type of functionality where C does something with the media
stream on behalf of A is clearly covered under the media translator
definition (Section 5.3).
6. Dismissing Payload Type Multiplexing
Before starting a discussion on when to use what alternative, we will
first document a number of reasons why using the payload type as a
multiplexing point for anything related to multiple streams is
unsuitable and will not be considered further.
If one attempts to use Payload type multiplexing beyond it's defined
usage, that has well known negative effects on RTP. To use Payload
type as the single discriminator for multiple streams implies that
all the different media streams are being sent with the same SSRC,
thus using the same timestamp and sequence number space. This has
many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, media streams that use different RTP
timestamp rates cannot be combined, as the timestamp values need
to be consistent across all multiplexed media frames. Thus
streams are forced to use the same rate. When this is not
possible, Payload Type multiplexing cannot be used.
2. Many RTP payload formats may fragment a media object over
multiple packets, like parts of a video frame. These payload
formats need to determine the order of the fragments to
correctly decode them. Thus it is important to ensure that all
fragments related to a frame or a similar media object are
transmitted in sequence and without interruptions within the
object. This can relatively simple be solved on the sender side
by ensuring that the fragments of each media stream are sent in
sequence.
3. Some media formats require uninterrupted sequence number space
between media parts. These are media formats where any missing
RTP sequence number will result in decoding failure or invoking
of a repair mechanism within a single media context. The text/
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T140 payload format [RFC4103] is an example of such a format.
These formats will need a sequence numbering abstraction
function between RTP and the individual media stream before
being used with Payload Type multiplexing.
4. Sending multiple streams in the same sequence number space makes
it impossible to determine which Payload Type and thus which
stream a packet loss relates to.
5. If RTP Retransmission [RFC4588] is used and there is a loss, it
is possible to ask for the missing packet(s) by SSRC and
sequence number, not by Payload Type. If only some of the
Payload Type multiplexed streams are of interest, there is no
way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets must be requested, wasting
bandwidth.
6. The current RTCP feedback mechanisms are built around providing
feedback on media streams based on stream ID (SSRC), packet
(sequence numbers) and time interval (RTP Timestamps). There is
almost never a field to indicate which Payload Type is reported,
so sending feedback for a specific media stream is difficult
without extending existing RTCP reporting.
7. The current RTCP media control messages [RFC5104] specification
is oriented around controlling particular media flows, i.e.
requests are done addressing a particular SSRC. Such mechanisms
would need to be redefined to support Payload Type multiplexing.
8. The number of payload types are inherently limited.
Accordingly, using Payload Type multiplexing limits the number
of streams that can be multiplexed and does not scale. This
limitation is exacerbated if one uses solutions like RTP and
RTCP multiplexing [RFC5761] where a number of payload types are
blocked due to the overlap between RTP and RTCP.
9. At times, there is a need to group multiplexed streams and this
is currently possible for RTP Sessions and for SSRC, but there
is no defined way to group Payload Types.
10. It is currently not possible to signal bandwidth requirements
per media stream when using Payload Type Multiplexing.
11. Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that
context.
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12. A legacy end-point that doesn't understand the indication that
different RTP payload types are different media streams may be
slightly confused by the large amount of possibly overlapping or
identically defined RTP Payload Types.
7. Multiple Streams Discussion
7.1. Introduction
Using multiple media streams is a well supported feature of RTP.
However, what can be unclear for most implementors or people writing
RTP/RTCP extensions attempting to apply multiple streams, is when it
is most appropriate to add an additional SSRC in an existing RTP
session and when it is better to use multiple RTP sessions. This
section tries to discuss the various considerations needed. The next
section then concludes with some guidelines.
7.2. RTP/RTCP Aspects
This section discusses RTP and RTCP aspects worth considering when
selecting between SSRC multiplexing and Session multiplexing.
7.2.1. The RTP Specification
RFC 3550 contains some recommendations and a bullet list with 5
arguments for different aspects of RTP multiplexing. Let's review
Section 5.2 of [RFC3550], reproduced below:
"For efficient protocol processing, the number of multiplexing points
should be minimized, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which
is different for each RTP session. For example, in a teleconference
composed of audio and video media encoded separately, each medium
SHOULD be carried in a separate RTP session with its own destination
transport address.
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields. Interleaving packets with different RTP media types but
using the same SSRC would introduce several problems:
1. If, say, two audio streams shared the same RTP session and the
same SSRC value, and one were to change encodings and thus
acquire a different RTP payload type, there would be no general
way of identifying which stream had changed encodings.
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2. An SSRC is defined to identify a single timing and sequence
number space. Interleaving multiple payload types would require
different timing spaces if the media clock rates differ and would
require different sequence number spaces to tell which payload
type suffered packet loss.
3. The RTCP sender and receiver reports (see Section 6.4) can only
describe one timing and sequence number space per SSRC and do not
carry a payload type field.
4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
5. Carrying multiple media in one RTP session precludes: the use of
different network paths or network resource allocations if
appropriate; reception of a subset of the media if desired, for
example just audio if video would exceed the available bandwidth;
and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations.
Using a different SSRC for each medium but sending them in the same
RTP session would avoid the first three problems but not the last
two.
On the other hand, multiplexing multiple related sources of the same
medium in one RTP session using different SSRC values is the norm for
multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It may also be appropriate
to multiplex streams of the same medium using different SSRC values
in other scenarios where the last two problems do not apply."
Let's consider one argument at a time. The first is an argument for
using different SSRC for each individual media stream, which still is
very applicable.
The second argument is advocating against using payload type
multiplexing, which still stands as can been seen by the extensive
list of issues found in Section 6.
The third argument is yet another argument against payload type
multiplexing.
The fourth is an argument against multiplexing media streams that
require different handling into the same session. This is to
simplify the processing at any receiver of the media stream. If all
media streams that exist in an RTP session is of one media type and
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one particular purpose, there is no need for deeper inspection of the
packets before processing them in both end-points and RTP aware
middle nodes.
The fifth argument discusses network aspects that we will discuss
more below in Section 7.4. It also goes into aspects of
implementation, like decomposed end-points where different processes
or inter-connected devices handle different aspects of the whole
multi-media session.
A summary of RFC 3550's view on multiplexing is to use unique SSRCs
for anything that is its' own media/packet stream, and secondly use
different RTP sessions for media streams that don't share media type
and purpose, to maximize flexibility when it comes to processing and
handling of the media streams.
This mostly agrees with the discussion and recommendations in this
document. However, there has been an evolution of RTP since that
text was written which needs to be reflected in the discussion.
Additional clarifications for specific cases are also needed.
7.2.2. Multiple SSRC Legacy Considerations
When establishing RTP sessions that may contain end-points that
aren't updated to handle multiple streams following these
recommendations, a particular application can have issues with
multiple SSRCs within a single session. These issues include:
1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously.
3. Be capable of rendering multiple streams simultaneously.
RTP Session multiplexing could potentially avoid these issues if
there is only a single SSRC at each end-point, and in topologies
which appears like point to point as seen the end-point. However,
forcing the usage of session multiplexing due to this reason would be
a great mistake, as it is likely that a significant set of
applications will need a combination of SSRC multiplexing of several
media sources and session multiplexing for other aspects such as
encoding alternatives, robustification or simply to support legacy.
However, this issue does need consideration when deploying multiple
media streams within an RTP session where legacy end-points may
occur.
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7.2.3. RTP Specification Clarifications Needed
The RTP specification contains a few things that are potential
interoperability issues when using multiple SSRCs within a session.
These issues are described and discussed in Section 9. These should
not be considered strong arguments against using SSRC multiplexing
when otherwise appropriate, and there are some issues we expect to be
solved in the near future.
7.2.4. Handling Varying sets of Senders
Another potential issue that needs to be considered is where a
limited set of simultaneously active sources varies within a larger
set of session members. As each media decoding chain may contain
state, it is important that this type of usage ensures that a
receiver can flush a decoding state for an inactive source and if
that source becomes active again, it does not assume that this
previous state exists.
This behavior might in certain applications be possible to limit to a
particular RTP Session and instead use multiple RTP sessions. But in
some cases it is likely unavoidable and the most appropriate thing is
to SSRC multiplex.
7.2.5. Cross Session RTCP requests
There currently exist no functionality to make truly synchronized and
atomic RTCP requests across multiple RTP Sessions. Instead separate
RTCP messages will have to be sent in each session. This gives SSRC
multiplexed streams a slight advantage as RTCP requests for different
streams in the same session can be sent in a compound RTCP packet.
Thus providing an atomic operation if different modifications of
different streams are requested at the same time.
In Session multiplexed cases, the RTCP timing rules in the sessions
and the transport aspects, such as packet loss and jitter, prevents a
receiver from relying on atomic operations, instead more robust and
forgiving mechanisms need to be used.
7.2.6. Binding Related Sources
A common problem in a number of various RTP extensions has been how
to bind together related sources. This issue is common independent
of SSRC multiplexing and Session Multiplexing, and any solution and
recommendation to the problem should work equally well for both to
avoid creating barriers between using session multiplexing and SSRC
multiplexing.
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The current solutions don't have these properties. There exist one
solution for grouping RTP session together in SDP [RFC5888] to know
which RTP session contains for example the FEC data for the source
data in another session. However, this mechanism does not work on
individual media flows and is thus not directly applicable to the
problem. The other solution is also SDP based and can group SSRCs
within a single RTP session [RFC5576]. Thus this mechanism can bind
media streams in SSRC multiplexed cases. Both solutions have the
shortcoming of being restricted to SDP based signalling and also do
not work in cases where the session's dynamic properties are such
that it is difficult or resource consuming to keep the list of
related SSRCs up to date.
One possible solution could be to mandate the same SSRC being used in
all RTP session in case of session multiplexing. We do note that
Section 8.3 of the RTP Specification [RFC3550] recommends using a
single SSRC space across all RTP sessions for layered coding.
However this recommendation has some downsides and is less applicable
beyond the field of layered coding. To use the same sender SSRC in
all RTP sessions from a particular end-point can cause issues if an
SSRC collision occurs. If the same SSRC is used as the required
binding between the streams, then all streams in the related RTP
sessions must change their SSRC. This is extra likely to cause
problems if the participant populations are different in the
different sessions. For example, in case of large number of
receivers having selected totally random SSRC values in each RTP
session as RFC 3550 specifies, a change due to a SSRC collision in
one session can then cause a new collision in another session. This
cascading effect is not severe but there is an increased risk that
this occurs for well populated sessions. In addition, being forced
to change the SSRC affects all the related media streams; instead of
having to re-synchronize only the originally conflicting stream, all
streams will suddenly need to be re-synchronized with each other.
This will prevent also the media streams not having an actual
collision from being usable during the re-synchronization and also
increases the time until synchronization is finalized. In addition,
it requires exception handling in the SSRC generation.
The above collision issue does not occur in case of having only one
SSRC space across all sessions and all participants will be part of
at least one session, like the base layer in layered encoding. In
that case the only downside is the special behavior that needs to be
well defined by anyone using this. But, having an exception behavior
where the SSRC space is common across all session an that doesn't fit
all the RTP extensions or payload formats present in the sessions is
a issue. It is possible to create a situation where the different
mechanisms can't be combined due to the non standard SSRC allocation
behavior.
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Existing mechanisms with known issues:
RTP Retransmission (RFC4588): Has two modes, one for SSRC
multiplexing and one for Session multiplexing. The session
multiplexing requires the same CNAME and mandates that the same
SSRC is used in both sessions. Using the same SSRC does work but
will potentially have issues in certain cases. In SSRC
multiplexed mode the CNAME is used, and when the first
retransmission request is sent, one must not have another
retransmission request outstanding for an SSRC which don't have a
the binding between the original SSRC and the retransmission
stream's SSRC. This works but creates some limitations that can
be avoided by a more explicit mechanism. The SDP based ssrc-group
mechanism is sufficient in this case as long as the application
can rely on the signalling based solution.
Scalable Video Coding (RFC6190): As an example of scalable coding,
SVC [RFC6190] has various modes. The Multi Session Transmission
(MST) uses Session multiplexing to separate scalability layers.
However, this specification has failed to explicit how these
layers are bound together in cases where CNAME isn't sufficient.
CNAME is no longer sufficient when more than one media source
occur within a session that have the same CNAME, for example due
to multiple video cameras capturing the same lecture hall. This
likely implies that a single SSRC space as recommend by Section
8.3 of RTP [RFC3550] is to be used.
Forward Error Correction: If some type of FEC or redundancy stream
is being sent, it needs it's own SSRC, with the exception of
constructions like redundancy encoding [RFC2198]. Thus in case of
transmitting the FEC in the same session as the source data, the
inter SSRC relation within a session is needed. In case of
sending the redundant data in a separate session from the source,
the SSRC in each session needs to be related. This occurs for
example in RFC5109 when using session separation of original and
FEC data. SSRC multiplexing is not supported, only using
redundant encoding is supported.
This issue appears to need action to harmonize and avoid future
shortcomings in extension specifications. A proposed solution for
handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname].
7.2.7. Forward Error Correction
There exist a number of Forward Error Correction (FEC) based schemes
for how to reduce the packet loss of the original streams. Most of
the FEC schemes will protect a single source flow. The protection is
achieved by transmitting a certain amount of redundant information
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that is encoded such that it can repair one or more packet loss over
the set of packets they protect. This sequence of redundant
information also needs to be transmitted as its own media stream, or
in some cases instead of the original media stream. Thus many of
these schemes creates a need for binding the related flows as
discussed above. They also create additional flows that need to be
transported. Looking at the history of these schemes, there is both
SSRC multiplexed and Session multiplexed solutions and some schemes
that support both.
Using a Session multiplexed solution provides good support for legacy
when deploying FEC or changing the scheme used so that some set of
receivers may not be able to utilize the FEC information. By placing
it in a separate RTP session, it can easily be ignored.
In usages involving multicast, having the FEC information on its own
multicast group and RTP session allows for flexibility, for example
when using Rapid Acquisition of Multicast Groups (RAMS) [RFC6285].
During the RAMS burst where data is received over unicast and where
it is possible to combine with unicast based retransmission
[RFC4588], there is no need to burst the FEC data related to the
burst of the source media streams needed to catch up with the
multicast group. This saves bandwidth to the receiver during the
burst, enabling quicker catch up. When the receiver has catched up
and joins the multicast group(s) for the source, it can at the same
time join the multicast group with the FEC information. Having the
source stream and the FEC in separate groups allow for easy
separation in the Burst/Retransmission Source (BRS) without having to
individually classify packets.
7.2.8. Transport Translator Sessions
A basic Transport Translator relays any incoming RTP and RTCP packets
to the other participants. The main difference between SSRC
multiplexing and Session multiplexing resulting from this use case is
that for SSRC multiplexing it is not possible for a particular
session participant to decide to receive a subset of media streams.
When using separate RTP sessions for the different sets of media
streams, a single participant can choose to leave one of the sessions
but not the other.
7.2.9. Multiple Media Types in one RTP session
Having different media types, like audio and video, in the same RTP
sessions is not forbidden, only recommended against as can be seen in
Section 7.2.1. When using multiple media types, there are a number
of considerations:
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Payload Type gives Media Type: This solution is dependent on getting
the media type from the Payload Type. Thus overloading this de-
multiplexing point in a receiver for two purposes. First for the
main media type and determining the processing chain, then later
for the exact configuration of the encoder and packetization.
Payload Type field limiations: The total number of Payload Types
available to use in an RTP session is fairly limited, especially
if Multiplexing RTP Data and Control Packets on a Single Port
[RFC5761] is used. For certain applications negotiating a large
set of codes and configuration may become an issue.
Don't switch media types for an SSRC: The primary reasons to avoid
switching from sending for example audio to sending video using
the same SSRC is the implications on a receiver. When this
happens, the processing chain in the receiver will have to switch
from one media type to another. As the different media type's
entire processing chains are different and are connected to
different outputs it is difficult to reuse the decoding chain,
which a normal codec change likely can. Instead the entire
processing chain has to be torn down and replaced. In addition,
there is likely a clock rate switching problem, possibly resulting
in synchronization loss at the point of switching media type if
some packet loss occurs.
RTCP Bit-rate Issues: If the media types are significantly different
in bit-rate, the RTCP bandwidth rates assigned to each source in a
session can result in interesting effects, like that the RTCP bit-
rate share for an audio stream is larger than the actual audio
bit-rate. In itself this doesn't cause any conflicts, only
potentially unnecessary overhead. It is possible to avoid this
using AVPF or SAVPF and setting trr-int parameter, which can bring
down unnecessary regular reporting while still allowing for rapid
feedback.
Decomposited end-points: Decomposited nodes that rely on the regular
network to separate audio and video to different devices do not
work well with this session setup. If they are forced to work,
all media receiver parts of a decomposited end-point will receive
all media, thus doubling the bit-rate consumption for the end-
point.
RTP Mixers and Translators: An RTP mixer or Media Translator will
also have to support this particular session setup, where it
before could rely on the RTP session to determine what processing
options should be applied to the incoming packets.
As can be seen, there is nothing in here that prevents using a single
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RTP session for multiple media types, however it does create a number
of limitations and special case implementation requirements. So
anyone considering to use this setup should carefully review if the
reasons for using a single RTP session is sufficient to motivate this
special case.
7.3. Signalling Aspects
There exist various signalling solutions for establishing RTP
sessions. Many are SDP [RFC4566] based, however SDP functionality is
also dependent on the signalling protocols carrying the SDP. Where
RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative
fashion, SIP [RFC3261] uses SDP with the additional definition of
Offer/Answer [RFC3264]. The impact on signalling and especially SDP
needs to be considered as it can greatly affect how to deploy a
certain multiplexing point choice.
7.3.1. Session Oriented Properties
One aspect of the existing signalling is that it is focused around
sessions, or at least in the case of SDP the media description.
There are a number of things that are signalled on a session level/
media description but that are not necessarily strictly bound to an
RTP session and could be of interest to signal specifically for a
particular media stream within the session. The following properties
have been identified as being potentially useful to signal not only
on RTP session level:
o Bitrate/Bandwidth exist today only at aggregate or a common any
media stream limit
o Which SSRC that will use which RTP Payload Types
Some of these issues are clearly SDP's problem rather than RTP
limitations. However, if the aim is to deploy an SSRC multiplexed
solution that contains several sets of media streams with different
properties (encoding/packetization parameter, bit-rate, etc), putting
each set in a different RTP session would directly enable negotiation
of the parameters for each set. If insisting on SSRC multiplexing, a
number of signalling extensions are needed to clarify that there are
multiple sets of media streams with different properties and that
they shall in fact be kept different, since a single set will not
satisfy the applications requirements.
This does in fact create a strong driver to use RTP session
multiplexing for any case where different sets of media streams with
different requirements exist.
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7.3.2. SDP Prevents Multiple Media Types
SDP encoded in its structure a prevention against using multiple
media types in the same RTP session. A media description in SDP can
only have a single media type; audio, video, text, image,
application. This media type is used as the top-level media type for
identifying the actual payload format bound to a particular payload
type using the rtpmap attribute. Thus a high fence against using
multiple media types in the same session was created.
There is a proposal in the MMUSIC WG for how one could allow multiple
media lines describe a single underlying transport
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus support either
one RTP session with multiple media types. There is also a solution
for multiplexing multiple RTP sessions onto the same transport
[I-D.westerlund-avtcore-single-transport-multiplexing].
7.4. Network Apsects
The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor.
7.4.1. Quality of Service
When it comes to Quality of Service mechanisms, they are either flow
based or marking based. RSVP [RFC2205] is an example of a flow based
mechanism, while Diff-Serv [RFC2474] is an example of a Marking based
one. For a marking based scheme, the method of multiplexing will not
affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between
the methods. SSRC multiplexing will result in all media streams
being part of the same 5-tuple (protocol, source address, destination
address, source port, destination port) which is the most common
selector for flow based QoS. Thus, separation of the level of QoS
between media streams is not possible. That is however possible for
session based multiplexing, where each different version can be in a
different RTP session that can be sent over different 5-tuples.
7.4.2. NAT and Firewall Traversal
In today's network there exist a large number of middleboxes. The
ones that normally have most impact on RTP are Network Address
Translators (NAT) and Firewalls (FW).
Below we analyze and comment on the impact of requiring more
underlying transport flows in the presence of NATs and Firewalls:
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End-Point Port Consumption: A given IP address only has 65536
available local ports per transport protocol for all consumers of
ports that exist on the machine. This is normally never an issue
for an end-user machine. It can become an issue for servers that
handle large number of simultaneous streams. However, if the
application uses ICE to authenticate STUN requests, a server can
serve multiple end-points from the same local port, and use the
whole 5-tuple (source and destination address, source and
destination port, protocol) as identifier of flows after having
securely bound them to the remote end-point address using the STUN
request. In theory the minimum number of media server ports
needed are the maximum number of simultaneous RTP Sessions a
single end-point may use. In practice, implementation will
probably benefit from using more server ports to simplify
implementation or avoid performance bottlenecks.
NAT State: If an end-point sits behind a NAT, each flow it generates
to an external address will result in a state that has to be kept
in the NAT. That state is a limited resource. In home or Small
Office/Home Office (SOHO) NATs, memory or processing are usually
the most limited resources. For large scale NATs serving many
internal end-points, available external ports are typically the
scarce resource. Port limitations is primarily a problem for
larger centralized NATs where end-point independent mapping
requires each flow to use one port for the external IP address.
This affects the maximum number of internal users per external IP
address. However, it is worth pointing out that a real-time video
conference session with audio and video is likely using less than
10 UDP flows, compared to certain web applications that can use
100+ TCP flows to various servers from a single browser instance.
NAT Traversal Excess Time: Making the NAT/FW traversal takes a
certain amount of time for each flow. It also takes time in a
phase of communication between accepting to communicate and the
media path being established which is fairly critical. The best
case scenario for how much extra time it can take following the
specified ICE procedures are: 1.5*RTT + Ta*(Additional_Flows-1),
where Ta is the pacing timer, which ICE specifies to be no smaller
than 20 ms. That assumes a message in one direction, and then an
immediate triggered check back. This as ICE first finds one
candidate pair that works prior to establish multiple flows.
Thus, there are no extra time until one has found a working
candidate pair. Based on that working pair the extra time is to
in parallel establish the, in most cases 2-3, additional flows.
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NAT Traversal Failure Rate: Due to the need to establish more than a
single flow through the NAT, there is some risk that establishing
the first flow succeeds but that one or more of the additional
flows fail. The risk that this happens is hard to quantify, but
it should be fairly low as one flow from the same interfaces has
just been successfully established . Thus only rare events such
as NAT resource overload, or selecting particular port numbers
that are filtered etc, should be reasons for failure.
SSRC multiplexing keeps additional media streams within one RTP
Session and does not introduce any additional NAT traversal
complexities per media stream. In contrast, the session multiplexing
is using one RTP session per media stream. Thus additional lower
layer transport flows will be required, unless an explicit de-
multiplexing layer is added between RTP and the transport protocol.
A proposal for how to multiplex multiple RTP sessions over the same
single lower layer transport exist in
[I-D.westerlund-avtcore-single-transport-multiplexing].
7.4.3. Multicast
Multicast groups provides a powerful semantics for a number of real-
time applications, especially the ones that desire broadcast-like
behaviors with one end-point transmitting to a large number of
receivers, like in IPTV. But that same semantics do result in a
certain number of limitations.
One limitation is that for any group, sender side adaptation to the
actual receiver properties causes a degradation for all participants
to what is supported by the receiver with the worst conditions among
the group participants. In most cases this is not acceptable.
Instead various receiver based solutions are employed to ensure that
the receivers achieve best possible performance. By using scalable
encoding and placing each scalability layer in a different multicast
group, the receiver can control the amount of traffic it receives.
To have each scalability layer on a different multicast group, one
RTP session per multicast group is used.
If instead a single RTP session over multiple transports were to be
deployed, i.e. multicast groups with each layer as it's own SSRC,
then very different views of the RTP session would exist. That as
one receiver may see only a single layer (SSRC), while another may
see three SSRCs if it joined three multicast groups. This would
cause disjoint RTCP reports where a management system would not be
able to determine if a receiver isn't reporting on a particular SSRC
due to that it is not a member of that multicast group, or because it
doesn't receive it as a result of a transport failure.
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Thus it appears easiest and most straightforward to use multiple RTP
sessions. In addition, the transport flow considerations in
multicast are a bit different from unicast. First of all there is no
shortage of port space, as each multicast group has its own port
space.
7.4.4. Multiplexing multiple RTP Session on a Single Transport
For applications that doesn't need flow based QoS and like to save
ports and NAT/FW traversal costs, there is a proposal for how to
achieve multiplexing of multiple RTP sessions over the same lower
layer transport
[I-D.westerlund-avtcore-single-transport-multiplexing]. Using such a
solution would allow session multiplexing without most of the
perceived downsides of additional RTP sessions creating a need for
additional transport flows.
7.5. Security Aspects
On the basic level there is no significant difference in security
when having one RTP session and having multiple. However, there are
a few more detailed considerations that might need to be considered
in certain usages.
7.5.1. Security Context Scope
When using SRTP [RFC3711] the security context scope is important and
can be a necessary differentiation in some applications. As SRTP's
crypto suites (so far) is built around symmetric keys, the receiver
will need to have the same key as the sender. This results in that
none in a multi-party session can be certain that a received packet
really was sent by the claimed sender or by another party having
access to the key. In most cases this is a sufficient security
property, but there are a few cases where this does create
situations.
The first case is when someone leaves a multi-party session and one
wants to ensure that the party that left can no longer access the
media streams. This requires that everyone re-keys without
disclosing the keys to the excluded party.
A second case is when using security as an enforcing mechanism for
differentiation. Take for example a scalable layer or a high quality
simulcast version which only premium users are allowed to access.
The mechanism preventing a receiver from getting the high quality
stream can be based on the stream being encrypted with a key that
user can't access without paying premium, having the key-management
limit access to the key.
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In the latter case it is likely easiest from signalling, transport
(if done over multicast) and security to use a different RTP session.
That way the user(s) not intended to receive a particular stream can
easily be excluded. There is no need to have SSRC specific keys,
which many of the key-management systems cannot handle.
7.5.2. Key-Management for Multi-party session
Performing key-management for Multi-party session can be a challenge.
This section considers some of the issues.
Transport translator based session cannot use Security Description
[RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each end-
point provides it's set of keys. In centralized conference, the
signalling counterpart is a conference server and the media plane
unicast counterpart (to which DTLS messages would be sent) is the
translator. Thus an extension like Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] are needed or a MIKEY [RFC3830] based
solution that allows for keying all session participants with the
same master key.
Keying of multicast transported SRTP face similar challenges as the
transport translator case.
8. Guidelines
This section contains a number of recommendations for implementors or
specification writers when it comes to handling multi-stream.
Don't Require the same SSRC across Sessions: As discussed in
Section 7.2.6 there exist drawbacks in using the same SSRC in
multiple RTP sessions as a mechanism to bind related media streams
together. Instead a mechanism to explicitly signal the relation
SHOULD be used, either in RTP/RTCP or in the used signalling
mechanism that establish the RTP session(s).
Use SSRC multiplexing for additional Media Sources: In the cases an
RTP end-point needs to transmit additional media source(s) of the
same media type and purpose in the application it is RECOMMENDED
to send them as additional SSRCs in the same RTP session. For
example a telepresence room where there are three cameras, and
each camera captures 2 persons sitting at the table, sending each
camera as its own SSRC within a single RTP session is recommended.
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Use additional RTP sessions for streams with different purposes:
When media streams have different purpose or processing
requirements it is RECOMMENDED that the different types of streams
are put in different RTP sessions.
When using Session Multiplexing use grouping: When using Session
Multiplexing solutions it is RECOMMENDED to be explicitly group
the involved RTP sessions using the signalling mechanism, for
example The Session Description Protocol (SDP) Grouping Framework.
[RFC5888]
RTP/RTCP Extensions May Support SSRC and Session Multiplexing: When
defining an RTP or RTCP extension, the creator needs to consider
if this extension is applicable in both SSRC multiplexed and
Session multiplexed usages. If it is, then any generic extensions
are RECOMMENDED to support both. Applications that are not as
generally applicable will have to consider if interoperability is
better served by defining a single solution or providing both
options.
Transport Support Extensions: When defining new RTP/RTCP extensions
intended for transport support, like the retransmission or FEC
mechanisms, they are RECOMMENDED to include support for both SSRC
and Session multiplexing so that application developers can choose
freely from the set of mechanisms without concerning themselves
with if a particular solution only supports one of the
multiplexing choices.
This discussion and guidelines points out that a small set of
extension mechanisms could greatly improve the situation when it
comes to using multiple streams independently of Session multiplexing
or SSRC multiplexing. These extensions are:
Media Source Identification: A Media source identification that can
be used to bind together media streams that are related to the
same media source. A proposal
[I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES
item SRCNAME that also can be used with the a=ssrc SDP attribute
to provide signalling layer binding information.
SSRC limiations within RTP sessions: By providing a signalling
solution that allows the signalling peers to explicitly express
both support and limitations on how many simultaneous media
streams an end-point can handle within a given RTP Session. That
ensures that usage of SSRC multiplexing occurs when supported and
without overloading an end-point. This extension is proposed in
[I-D.westerlund-avtcore-max-ssrc].
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9. RTP Specification Clarifications
This section describes a number of clarifications to the RTP
specifications that are likely necessary for aligned behavior when
RTP sessions contains more SSRCs than one local and one remote.
9.1. RTCP Reporting from all SSRCs
When one have multiple SSRC in an RTP node, then all these SSRC must
send RTCP SR or RR as long as the SSRC exist. It is not sufficient
that only one SSRC in the node sends report blocks on the incoming
RTP streams. The reason for this is that a third party monitor may
not necessarily be able to determine that all these SSRC are in fact
co-located and originate from the same stack instance that gather
report data.
9.2. RTCP Self-reporting
For any RTP node that sends more than one SSRC, there exist the
question if SSRC1 needs to report its reception of SSRC2 and vice
versa. The reason that they in fact need to report on all other
local streams as being received is report consistency. A third party
monitor that considers the full matrix of media streams and all known
SSRC reports on these media streams would detect a gap in the reports
which could be a transport issue unless identified as in fact being
sources from same node.
9.3. Combined RTCP Packets
When a node contains multiple SSRCs, it is questionable if an RTCP
compound packet can only contain RTCP packets from a single SSRC or
if multiple SSRCs can include their packets in a joint compound
packet. The high level question is a matter for any receiver
processing on what to expect. In addition to that question there is
the issue of how to use the RTCP timer rules in these cases, as the
existing rules are focused on determining when a single SSRC can
send.
10. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
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11. Security Considerations
12. Acknowledgements
13. References
13.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
13.2. Informative References
[ALF] Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols", SIGCOMM
Symposium on Communications Architectures and
Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE
Computer Communications Review, Vol. 20(4),
September 1990.
[I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011.
[I-D.ietf-avt-srtp-ekt]
McGrew, D., Andreasen, F., Wing, D., and K. Fischer,
"Encrypted Key Transport for Secure RTP",
draft-ietf-avt-srtp-ekt-02 (work in progress), March 2011.
[I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M., "Support for multiple clock rates in
an RTP session", draft-ietf-avtext-multiple-clock-rates-01
(work in progress), July 2011.
[I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-01 (work in progress),
July 2011.
[I-D.westerlund-avtcore-max-ssrc]
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Westerlund, M., Burman, B., and F. Jansson, "Multiple
Synchronization sources (SSRC) in RTP Session Signaling",
draft-westerlund-avtcore-max-ssrc (work in progress),
October 2011.
[I-D.westerlund-avtcore-single-transport-multiplexing]
Westerlund, M., "Multiple RTP Session on a Single Lower-
Layer Transport",
draft-westerlund-avtcore-transport-multiplexing (work in
progress), October 2011.
[I-D.westerlund-avtext-rtcp-sdes-srcname]
Westerlund, M., Burman, B., and P. Sandgren, "RTCP SDES
Item SRCNAME to Label Individual Sources",
draft-westerlund-avtext-rtcp-sdes-srcname (work in
progress), October 2011.
[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.
[RFC2205] Braden, B., Zhang, L., Berson, S., Herzog, S., and S.
Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
Functional Specification", RFC 2205, September 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474,
December 1998.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
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[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, June 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, July 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.
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[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding", RFC 6190,
May 2011.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
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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Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
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