Internet DRAFT - draft-ietf-avtcore-multiplex-guidelines
draft-ietf-avtcore-multiplex-guidelines
Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: Informational Ericsson
Expires: December 18, 2020 C. Perkins
University of Glasgow
H. Alvestrand
Google
R. Even
June 16, 2020
Guidelines for using the Multiplexing Features of RTP to Support
Multiple Media Streams
draft-ietf-avtcore-multiplex-guidelines-12
Abstract
The Real-time Transport Protocol (RTP) is a flexible protocol that
can be used in a wide range of applications, networks, and system
topologies. That flexibility makes for wide applicability, but can
complicate the application design process. One particular design
question that has received much attention is how to support multiple
media streams in RTP. This memo discusses the available options and
design trade-offs, and provides guidelines on how to use the
multiplexing features of RTP to support multiple media streams.
Status of This Memo
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This Internet-Draft will expire on December 18, 2020.
Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4
2.2. Subjects Out of Scope . . . . . . . . . . . . . . . . . . 5
3. RTP Multiplexing Overview . . . . . . . . . . . . . . . . . . 5
3.1. Reasons for Multiplexing and Grouping RTP Streams . . . . 5
3.2. RTP Multiplexing Points . . . . . . . . . . . . . . . . . 6
3.2.1. RTP Session . . . . . . . . . . . . . . . . . . . . . 7
3.2.2. Synchronisation Source (SSRC) . . . . . . . . . . . . 8
3.2.3. Contributing Source (CSRC) . . . . . . . . . . . . . 10
3.2.4. RTP Payload Type . . . . . . . . . . . . . . . . . . 11
3.3. Issues Related to RTP Topologies . . . . . . . . . . . . 12
3.4. Issues Related to RTP and RTCP Protocol . . . . . . . . . 13
3.4.1. The RTP Specification . . . . . . . . . . . . . . . . 13
3.4.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 15
3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 15
3.4.4. Forward Error Correction . . . . . . . . . . . . . . 17
4. Considerations for RTP Multiplexing . . . . . . . . . . . . . 17
4.1. Interworking Considerations . . . . . . . . . . . . . . . 17
4.1.1. Application Interworking . . . . . . . . . . . . . . 17
4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 18
4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 19
4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 20
4.2. Network Considerations . . . . . . . . . . . . . . . . . 20
4.2.1. Quality of Service . . . . . . . . . . . . . . . . . 20
4.2.2. NAT and Firewall Traversal . . . . . . . . . . . . . 21
4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 23
4.3. Security and Key Management Considerations . . . . . . . 24
4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24
4.3.2. Key Management for Multi-party Sessions . . . . . . . 25
4.3.3. Complexity Implications . . . . . . . . . . . . . . . 26
5. RTP Multiplexing Design Choices . . . . . . . . . . . . . . . 26
5.1. Multiple Media Types in One Session . . . . . . . . . . . 26
5.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 28
5.3. Multiple Sessions for One Media Type . . . . . . . . . . 29
5.4. Single SSRC per Endpoint . . . . . . . . . . . . . . . . 30
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5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 32
6. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . 32
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
8. Security Considerations . . . . . . . . . . . . . . . . . . . 33
9. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 34
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 34
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 34
11.1. Normative References . . . . . . . . . . . . . . . . . . 34
11.2. Informative References . . . . . . . . . . . . . . . . . 35
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 39
Appendix B. Signalling Considerations . . . . . . . . . . . . . 40
B.1. Session Oriented Properties . . . . . . . . . . . . . . . 41
B.2. SDP Prevents Multiple Media Types . . . . . . . . . . . . 42
B.3. Signalling RTP Stream Usage . . . . . . . . . . . . . . . 42
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 43
1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] is a commonly used
protocol for real-time media transport. It is a protocol that
provides great flexibility and can support a large set of different
applications. RTP was from the beginning designed for multiple
participants in a communication session. It supports many topology
paradigms and usages, as defined in [RFC7667]. RTP has several
multiplexing points designed for different purposes. These enable
support of multiple RTP streams and switching between different
encoding or packetization of the media. By using multiple RTP
sessions, sets of RTP streams can be structured for efficient
processing or identification. Thus, an RTP application designer
needs to understand how to best use the RTP session, the RTP stream
identifier (SSRC), and the RTP payload type to meet the application's
needs.
There have been increased interest in more advanced usage of RTP.
For example, multiple RTP streams can be used when a single endpoint
has multiple media sources (like multiple cameras or microphones)
that need to be sent simultaneously. Consequently, questions are
raised regarding the most appropriate RTP usage. The limitations in
some implementations, RTP/RTCP extensions, and signalling have also
been exposed. This document aims to clarify the usefulness of some
functionalities in RTP which will hopefully result in more complete
implementations in the future.
The purpose of this document is to provide clear information about
the possibilities of RTP when it comes to multiplexing. The RTP
application designer needs to understand the implications arising
from a particular usage of the RTP multiplexing points. The document
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will provide some guidelines and recommend against some usages as
being unsuitable, in general or for particular purposes.
The document starts with some definitions and then goes into the
existing RTP functionalities around multiplexing. Both the desired
behaviour and the implications of a particular behaviour depend on
which topologies are used, which requires some consideration. This
is followed by a discussion of some choices in multiplexing behaviour
and their impacts. Some designs of RTP usage are discussed.
Finally, some guidelines and examples are provided.
2. Definitions
2.1. Terminology
The definitions in Section 3 of [RFC3550] are referenced normatively.
The taxonomy defined in [RFC7656] is referenced normatively.
The following terms and abbreviations are used in this document:
Multiparty: A communication situation including multiple endpoints.
In this document, it will be used to refer to situations where
more than two endpoints communicate.
Multiplexing: The operation of taking multiple entities as input,
aggregating them onto some common resource while keeping the
individual entities addressable such that they can later be fully
and unambiguously separated (de-multiplexed) again.
RTP Receiver: An Endpoint or Middlebox receiving RTP streams and
RTCP messages. It uses at least one SSRC to send RTCP messages.
An RTP Receiver may also be an RTP Sender.
RTP Sender: An Endpoint sending one or more RTP streams, but also
sending RTCP messages.
RTP Session Group: One or more RTP sessions that are used together
to perform some function. Examples are multiple RTP sessions used
to carry different layers of a layered encoding. In an RTP
Session Group, CNAMEs are assumed to be valid across all RTP
sessions, and designate synchronisation contexts that can cross
RTP sessions; i.e. SSRCs that map to a common CNAME can be assumed
to have RTCP Sender Report (SR) timing information derived from a
common clock such that they can be synchronised for playout.
Signalling: The process of configuring endpoints to participate in
one or more RTP sessions.
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Note: The above definitions of RTP Receiver and RTP Sender are
consistent with the usage in [RFC3550].
2.2. Subjects Out of Scope
This document is focused on issues that affect RTP. Thus, issues
that involve signalling protocols, such as whether SIP [RFC3261],
Jingle [JINGLE] or some other protocol is in use for session
configuration, the particular syntaxes used to define RTP session
properties, or the constraints imposed by particular choices in the
signalling protocols, are mentioned only as examples in order to
describe the RTP issues more precisely.
This document assumes the applications will use RTCP. While there
are applications that don't send RTCP, they do not conform to the RTP
specification, and thus can be regarded as reusing the RTP packet
format but not implementing the RTP protocol.
3. RTP Multiplexing Overview
3.1. Reasons for Multiplexing and Grouping RTP Streams
There are several reasons why an endpoint might choose to send
multiple media streams. In the below discussion, please keep in mind
that the reasons for having multiple RTP streams vary and include but
are not limited to the following:
o Multiple media sources
o Multiple RTP streams might be needed to represent one media source
for instance:
* To carry different layers of an scalable encoding of a media
source
* Alternative encodings during simulcast, for instance using
different codecs for the same audio stream
* Alternative formats during simulcast, for instance multiple
resolutions of the same video stream
o A retransmission stream might repeat some parts of the content of
another RTP stream
o A Forward Error Correction (FEC) stream might provide material
that can be used to repair another RTP stream
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For each of these reasons, it is necessary to decide if each
additional RTP stream is sent within the same RTP session as the
other RTP streams, or if it is necessary to use additional RTP
sessions to group the RTP streams. The choice suitable for one
situation, might not be the choice suitable in another situation or
combination of reasons. The clearest understanding is associated
with multiplexing multiple media sources of the same media type.
However, all reasons warrant discussion and clarification on how to
deal with them. As the discussion below will show, in reality we
cannot choose a single one of SSRC or RTP session multiplexing
solutions for all purposes. To utilise RTP well and as efficiently
as possible, both are needed. The real issue is finding the right
guidance on when to create additional RTP sessions and when
additional RTP streams in the same RTP session is the right choice.
3.2. RTP Multiplexing Points
This section describes the multiplexing points present in the RTP
protocol that can be used to distinguish RTP streams and groups of
RTP streams. Figure 1 outlines the process of demultiplexing
incoming RTP streams starting already at the socket representing
reception of one or more transport flows, e.g. based on the UDP
destination port. It also demultiplexes RTP/RTCP from any other
protocols, such as STUN [RFC5389] and DTLS-SRTP [RFC5764] on the same
transport as described in [RFC7983]. The Processing and Buffering
(PB) step of Figure 1 terminates the RTP/RTCP protocol and prepares
the RTP payload for input to the decoder.
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| | |
| | | packets
+-- v v v
| +------------+
| | Socket(s) | Transport Protocol Demultiplexing
| +------------+
| || ||
RTP | RTP/ || |+-----> DTLS (SRTP Keying, SCTP, etc)
Session | RTCP || +------> STUN (multiplexed using same port)
+-- ||
+-- ||
| ++(split by SSRC)-++---> Identify SSRC collision
| || || || ||
| (associate with signalling by MID/RID)
| vv vv vv vv
RTP | +--+ +--+ +--+ +--+ Jitter buffer,
Streams | |PB| |PB| |PB| |PB| process RTCP, etc.
| +--+ +--+ +--+ +--+
+-- | | | |
(select decoder based on PT)
+-- | / | /
| +-----+ | /
| / | |/
Payload | v v v
Formats | +---+ +---+ +---+
| |Dec| |Dec| |Dec| Decoders
| +---+ +---+ +---+
+--
Figure 1: RTP Demultiplexing Process
3.2.1. RTP Session
An RTP session is the highest semantic layer in the RTP protocol, and
represents an association between a group of communicating endpoints.
RTP does not contain a session identifier, yet different RTP sessions
must be possible to identify both across a set of different endpoints
and from the perspective of a single endpoint.
For RTP session separation across endpoints, the set of participants
that form an RTP session is defined as those that share a single
synchronisation source space [RFC3550]. That is, if a group of
participants are each aware of the synchronisation source identifiers
belonging to the other participants, then those participants are in a
single RTP session. A participant can become aware of a
synchronisation source identifier by receiving an RTP packet
containing it in the SSRC field or CSRC list, by receiving an RTCP
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packet mentioning it in an SSRC field, or through signalling (e.g.,
the Session Description Protocol (SDP) [RFC4566] "a=ssrc:" attribute
[RFC5576]). Thus, the scope of an RTP session is determined by the
participants' network interconnection topology, in combination with
RTP and RTCP forwarding strategies deployed by the endpoints and any
middleboxes, and by the signalling.
For RTP session separation within a single endpoint RTP relies on the
underlying transport layer, and on the signalling to identify RTP
sessions in a manner that is meaningful to the application. A single
endpoint can have one or more transport flows for the same RTP
session, and a single RTP session can span multiple transport layer
flows even if all endpoints use a single transport layer flow per
endpoint for that RTP session. The signalling layer might give RTP
sessions an explicit identifier, or the identification might be
implicit based on the addresses and ports used. Accordingly, a
single RTP session can have multiple associated identifiers, explicit
and implicit, belonging to different contexts. For example, when
running RTP on top of UDP/IP, an endpoint can identify and delimit an
RTP session from other RTP sessions by their UDP source and
destination IP addresses and UDP port numbers. A single RTP session
can be using multiple IP/UDP flows for receiving and/or sending RTP
packets to other endpoints or middleboxes, even if the endpoint does
not have multiple IP addresses. Using multiple IP addresses only
makes it more likely to require multiple IP/UDP flows. Another
example is SDP media descriptions (the "m=" line and the following
associated lines) that signal the transport flow and RTP session
configuration for the endpoint's part of the RTP session. The SDP
grouping framework [RFC5888] allows labeling of the media
descriptions to be used so that RTP Session Groups can be created.
Through use of Negotiating Media Multiplexing Using the Session
Description Protocol (SDP) [I-D.ietf-mmusic-sdp-bundle-negotiation],
multiple media descriptions become part of a common RTP session where
each media description represents the RTP streams sent or received
for a media source.
The RTP protocol makes no normative statements about the relationship
between different RTP sessions, however the applications that use
more than one RTP session will have some higher layer understanding
of the relationship between the sessions they create.
3.2.2. Synchronisation Source (SSRC)
A synchronisation source (SSRC) identifies a source of an RTP stream,
or an RTP receiver when sending RTCP. Every endpoint has at least
one SSRC identifier, even if it does not send RTP packets. RTP
endpoints that are only RTP receivers still send RTCP and use their
SSRC identifiers in the RTCP packets they send. An endpoint can have
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multiple SSRC identifiers if it sends multiple RTP streams.
Endpoints that are both RTP sender and RTP receiver use the same
SSRC(s) in both roles.
The SSRC is a 32-bit identifier. It is present in every RTP and RTCP
packet header, and in the payload of some RTCP packet types. It can
also be present in SDP signalling. Unless pre-signalled, e.g. using
the SDP "a=ssrc:" attribute [RFC5576], the SSRC is chosen at random.
It is not dependent on the network address of the endpoint, and is
intended to be unique within an RTP session. SSRC collisions can
occur, and are handled as specified in [RFC3550] and [RFC5576],
resulting in the SSRC of the colliding RTP streams or receivers
changing. An endpoint that changes its network transport address
during a session has to choose a new SSRC identifier to avoid being
interpreted as looped source, unless a mechanism providing a virtual
transport (such as ICE [RFC8445]) abstracts the changes.
SSRC identifiers that belong to the same synchronisation context
(i.e., that represent RTP streams that can be synchronised using
information in RTCP SR packets) use identical CNAME chunks in
corresponding RTCP SDES packets. SDP signalling can also be used to
provide explicit SSRC grouping [RFC5576].
In some cases, the same SSRC identifier value is used to relate
streams in two different RTP sessions, such as in RTP retransmission
[RFC4588]. This is to be avoided since there is no guarantee that
SSRC values are unique across RTP sessions. For the RTP
retransmission [RFC4588] case it is recommended to use explicit
binding of the source RTP stream and the redundancy stream, e.g.
using the RepairedRtpStreamId RTCP SDES item [I-D.ietf-avtext-rid].
The RepairedRtpStreamId is a rather recent mechanism, so one cannot
expect older applications to follow this recommendation.
Note that RTP sequence number and RTP timestamp are scoped by the
SSRC and thus specific per RTP stream.
Different types of entities use an SSRC to identify themselves, as
follows:
A real media source: Uses the SSRC to identify a "physical" media
source.
A conceptual media source: Uses the SSRC to identify the result of
applying some filtering function in a 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.
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An RTP receiver: Uses the SSRC to identify itself as the source of
its RTCP reports.
An endpoint that generates more than one media type, e.g. a
conference participant sending both audio and video, need not (and,
indeed, 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 endpoint. The main property attributed to SSRCs associated with
the same CNAME is that they are from a particular synchronisation
context and can be synchronised at playback.
An RTP receiver receiving a previously unseen SSRC value will
interpret it as a new source. It might in fact be a previously
existing source that had to change SSRC number due to an SSRC
conflict. Use of the MID extension
[I-D.ietf-mmusic-sdp-bundle-negotiation] helps to identify which
media source the new SSRC represents and use of the RID extension
[I-D.ietf-mmusic-rid] helps to identify what encoding or redundancy
stream it represents, even though the SSRC changed. However, the
originator of the previous SSRC ought to have ended the conflicting
source by sending an RTCP BYE for it prior to starting to send with
the new SSRC, making the new SSRC a new source.
3.2.3. Contributing Source (CSRC)
The Contributing Source (CSRC) is not a separate identifier. Rather
an SSRC identifier is listed as a CSRC in the RTP header of a packet
generated by an RTP mixer or video MCU/switch, if the corresponding
SSRC was in the header of one of the packets that contributed to the
output.
It is not possible, in general, to extract media represented by an
individual CSRC since it is typically the result of a media merge
(e.g. mix) operation on the individual media streams corresponding to
the CSRC identifiers. The exception is the case when only a single
CSRC is indicated as this represent forwarding of an RTP stream,
possibly modified. The RTP header extension for Mixer-to-Client
Audio Level Indication [RFC6465] expands on the receiver's
information about a packet with a CSRC list. Due to these
restrictions, CSRC will not be considered a fully qualified
multiplexing point and will be disregarded in the rest of this
document.
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3.2.4. RTP Payload Type
Each RTP stream utilises one or more RTP payload formats. An RTP
payload format describes how the output of a particular media codec
is framed and encoded into RTP packets. The payload format is
identified by the payload type (PT) field in the RTP packet header.
The combination of SSRC and PT therefore identifies a specific RTP
stream in a specific encoding format. The format definition can be
taken from [RFC3551] for statically allocated payload types, but
ought to be explicitly defined in signalling, such as SDP, both for
static and dynamic payload types. The term "format" here includes
those aspects described by out-of-band signalling means; 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 RTP payload type is scoped by the sending endpoint within an RTP
session. PT has the same meaning across all RTP streams in an RTP
session. All SSRCs sent from a single endpoint share the same
payload type definitions. The RTP payload type is designed such that
only a single payload type is valid at any time instant in the RTP
stream's timestamp time line, effectively time-multiplexing different
payload types if any change occurs. The payload type can change on a
per-packet basis for an SSRC, for example a speech codec making use
of generic comfort noise [RFC3389]. If there is a true need to send
multiple payload types for the same SSRC that are valid for the same
instant, 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 is that the combined payload sizes
of the different payload types ought not exceed the transport MTU.
Other aspects of RTP payload format use are described in How to Write
an RTP Payload Format [RFC8088].
The payload type is not a multiplexing point at the RTP layer (see
Appendix A for a detailed discussion of why using the payload type as
an RTP multiplexing point does not work). The RTP payload type is,
however, used to determine how to consume and decode an RTP stream.
The RTP payload type number is sometimes used to associate an RTP
stream with the signalling, which in general requires that unique RTP
payload type numbers are used in each context. Use of MID, e.g. when
bundling "m=" sections [I-D.ietf-mmusic-sdp-bundle-negotiation], can
replace the payload type as signalling association and unique RTP
payload types are then no longer required for that purpose.
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3.3. Issues Related to RTP Topologies
The impact of how RTP multiplexing is performed will in general vary
with how the RTP session participants are interconnected, described
by RTP Topology [RFC7667].
Even the most basic use case, denoted Topo-Point-to-Point in
[RFC7667], raises a number of considerations that are discussed in
detail in following sections. They range over such aspects as:
o Does my communication peer support RTP as defined with multiple
SSRCs per RTP session?
o Do I need network differentiation in form of QoS (Section 4.2.1)?
o Can the application more easily process and handle the media
streams if they are in different RTP sessions?
o Do I need to use additional RTP streams for RTP retransmission or
FEC?
For some point to multi-point topologies (e.g. Topo-ASM and Topo-SSM
in [RFC7667]), multicast is used to interconnect the session
participants. Special considerations (documented in Section 4.2.3)
are then needed as multicast is a one-to-many distribution system.
Sometimes an RTP communication can end up in a situation when the
communicating peers are not compatible for various reasons:
o No common media codec for a media type thus requiring transcoding.
o Different support for multiple RTP streams and RTP sessions.
o Usage of different media transport protocols, i.e., RTP or other.
o Usage of different transport protocols, e.g., UDP, DCCP, or TCP.
o Different security solutions, e.g., IPsec, TLS, DTLS, or SRTP with
different keying mechanisms.
In many situations this is resolved by the inclusion of a translator
between the two peers, as described by Topo-PtP-Translator in
[RFC7667]. The translator's main purpose is to make the peers look
compatible to each other. There can also be other reasons than
compatibility to insert a translator in the form of a middlebox or
gateway, for example a need to monitor the RTP streams. Beware that
changing the stream transport characteristics in the translator can
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require thorough understanding of aspects from congestion control and
media adaptation to application-layer semantics.
Within the uses enabled by the RTP standard, the point to point
topology can contain one or more RTP sessions with one or more media
sources per session, each having one or more RTP streams per media
source.
3.4. Issues Related to RTP and RTCP Protocol
Using multiple RTP streams is a well-supported feature of RTP.
However, for most implementers or people writing RTP/RTCP
applications or extensions attempting to apply multiple streams, it
can be unclear when it is most appropriate to add an additional RTP
stream in an existing RTP session and when it is better to use
multiple RTP sessions. This section discusses the various
considerations needed.
3.4.1. The RTP Specification
RFC 3550 contains some recommendations and a bullet list with 5
arguments for different aspects of RTP multiplexing. Please review
Section 5.2 of [RFC3550]. Five important aspects are quoted below.
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.
The first argument is to use different SSRC for each individual RTP
stream, which is fundamental to RTP operation.
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.
The second argument is advocating against demultiplexing RTP streams
within a session based only on their RTP payload type numbers, which
still stands as can been seen by the extensive list of issues found
in Appendix A.
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.
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The third argument is yet another argument against payload type
multiplexing.
4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
The fourth argument is against multiplexing RTP packets that require
different handling into the same session. In most cases the RTP
mixer must embed application logic to handle streams; the separation
of streams according to stream type is just another piece of
application logic, which might or might not be appropriate for a
particular application. One type of application that can mix
different media sources blindly is the audio-only telephone bridge,
although the ability to do that comes from the well-defined scenario
that is aided by use of a single media type, even though individual
streams may use incompatible codec types; most other types of
applications need application-specific logic to perform the mix
correctly.
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.
The fifth argument discusses network aspects that are described in
Section 4.2. It also goes into aspects of implementation, like Split
Component Terminal (see Section 3.10 of [RFC7667]) endpoints 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 to use
different RTP sessions for media streams that don't share a media
type. This document supports the first point; it is very valid. The
latter needs further discussion, as imposing a single solution on all
usages of RTP is inappropriate. "Multiple Media Types in an RTP
Session specification" [I-D.ietf-avtcore-multi-media-rtp-session]
updates RFC 3550 to allow multiple media types in a RTP session. It
also provides a detailed analysis of the potential benefits and
issues in having multiple media types in the same RTP session. Thus,
that document provides a wider scope for an RTP session and considers
multiple media types in one RTP session as a possible choice for the
RTP application designer.
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3.4.2. Multiple SSRCs in a Session
Using multiple SSRCs at one endpoint in an RTP session requires
resolving some unclear aspects of the RTP specification. These could
potentially lead to some interoperability issues as well as some
potential significant inefficiencies, as further discussed in "RTP
Considerations for Endpoints Sending Multiple Media Streams"
[RFC8108]. An RTP application designer should consider these issues
and the possible application impact from lack of appropriate RTP
handling or optimization in the peer endpoints.
Using multiple RTP sessions can potentially mitigate application
issues caused by multiple SSRCs in an RTP session.
3.4.3. Binding Related Sources
A common problem in a number of various RTP extensions has been how
to bind related RTP streams together. This issue is common to both
using additional SSRCs and multiple RTP sessions.
The solutions can be divided into a few groups:
o RTP/RTCP based
o Signalling based, e.g. SDP
o Grouping related RTP sessions
o Grouping SSRCs within an RTP session
Most solutions are explicit, but some implicit methods have also been
applied to the problem.
The SDP-based signalling solutions are:
SDP Media Description Grouping: The SDP Grouping Framework [RFC5888]
uses various semantics to group any number of media descriptions.
This has primarily been grouping RTP sessions, but in combination
with [I-D.ietf-mmusic-sdp-bundle-negotiation] it can also group
multiple media descriptions within a single RTP session.
SDP Media Multiplexing: Negotiating Media Multiplexing Using the
Session Description Protocol (SDP)
[I-D.ietf-mmusic-sdp-bundle-negotiation]
uses both SDP and RTCP information to associate RTP streams to SDP
media descriptions. This allows both to group RTP streams
belonging to an SDP media description, and to group multiple SDP
media descriptions into a single RTP session.
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SDP SSRC grouping: Source-Specific Media Attributes in SDP [RFC5576]
includes a solution for grouping SSRCs the same way as the
Grouping framework groups Media Descriptions.
The above grouping constructs support many use cases. Those
solutions have shortcomings in cases where the session's dynamic
properties are such that it is difficult or a drain on resources to
keep the list of related SSRCs up to date.
An RTP/RTCP-based grouping solution is to use the RTCP SDES CNAME to
bind related RTP streams to an endpoint or to a synchronization
context. For applications with a single RTP stream per type (media,
source or redundancy stream), CNAME is sufficient for that purpose
independently of whether one or more RTP sessions are used. However,
some applications choose not to use CNAME because of perceived
complexity or a desire not to implement RTCP and instead use the same
SSRC value to bind related RTP streams across multiple RTP sessions.
RTP Retransmission [RFC4588] in multiple RTP session mode and Generic
FEC [RFC5109] both use the CNAME method to relate the RTP streams,
which may work but might have some downsides in RTP sessions with
many participating SSRCs. It is not recommended to use identical
SSRC values across RTP sessions to relate RTP streams; When an SSRC
collision occurs, this will force change of that SSRC in all RTP
sessions and thus resynchronize all of them instead of only the
single media stream having the collision.
Another method to implicitly bind SSRCs is used by RTP Retransmission
[RFC4588] when using the same RTP session as the source RTP stream
for retransmissions. The receiver missing a packet issues an RTP
retransmission request, and then awaits a new SSRC carrying the RTP
retransmission payload and where that SSRC is from the same CNAME.
This limits a requester to having only one outstanding retransmission
request on any new source SSRCs per endpoint.
RTP Payload Format Restrictions [I-D.ietf-mmusic-rid] provides an
RTP/RTCP based mechanism to unambiguously identify the RTP streams
within an RTP session and restrict the streams' payload format
parameters in a codec-agnostic way beyond what is provided with the
regular payload types. The mapping is done by specifying an "a=rid"
value in the SDP offer/answer signalling and having the corresponding
RtpStreamId value as an SDES item and an RTP header extension. The
RID solution also includes a solution for binding redundancy RTP
streams to their original source RTP streams, given that those use
RID identifiers.
Experience has found that an explicit binding between the RTP
streams, agnostic of SSRC values, behaves well. That way, solutions
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using multiple RTP streams in a single RTP session and in multiple
RTP sessions will use the same type of binding.
3.4.4. Forward Error Correction
There exist a number of Forward Error Correction (FEC) based schemes
for how to mitigate packet loss in the original streams. Most of the
FEC schemes protect a single source flow. The protection is achieved
by transmitting a certain amount of redundant information that is
encoded such that it can repair one or more packet losses over the
set of packets the redundant information protects. This sequence of
redundant information needs to be transmitted as its own media
stream, or in some cases, instead of the original media stream.
Thus, many of these schemes create a need for binding related flows
as discussed above. Looking at the history of these schemes, there
are schemes using multiple SSRCs and schemes using multiple RTP
sessions, and some schemes that support both modes of operation.
Using multiple RTP sessions supports the case where some set of
receivers might not be able to utilise the FEC information. By
placing it in a separate RTP session and if separating RTP sessions
on transport level, FEC can easily be ignored already on the
transport level, without considering any RTP layer information.
In usages involving multicast, having the FEC information on its own
multicast group allows for similar flexibility. This is especially
useful when receivers see heterogeneous packet loss rates. A
receiver can decide, based on measurement of experienced packet loss
rates, whether to join a multicast group with the suitable FEC data
repair capabilities.
4. Considerations for RTP Multiplexing
4.1. Interworking Considerations
There are several different kinds of interworking, and this section
discusses two; interworking directly between different applications,
and interworking of applications through an RTP Translator. The
discussion includes the implications of potentially different RTP
multiplexing point choices and limitations that have to be considered
when working with some legacy applications.
4.1.1. Application Interworking
It is not uncommon that applications or services of similar but not
identical usage, especially the ones intended for interactive
communication, encounter a situation where one want to interconnect
two or more of these applications.
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In these cases, one ends up in a situation where one might use a
gateway to interconnect applications. This gateway must then either
change the multiplexing structure or adhere to the respective
limitations in each application.
There are two fundamental approaches to building a gateway: using RTP
Translator interworking (RTP bridging), where the gateway acts as an
RTP Translator with the two interconnected applications being members
of the same RTP session; or using Gateway Interworking with RTP
termination, where there are independent RTP sessions between each
interconnected application and the gateway.
For interworking to be feasible, any security solution in use needs
to be compatible and capable of exchanging keys with either the peer
or the gateway under the used trust model. Secondly, the
applications need to use media streams in a way that makes sense in
both applications.
4.1.2. RTP Translator Interworking
From an RTP perspective, the RTP Translator approach could work if
all the applications are using the same codecs with the same payload
types, have made the same multiplexing choices, and have the same
capabilities in number of simultaneous RTP streams combined with the
same set of RTP/RTCP extensions being supported. Unfortunately, this
might not always be true.
When a gateway is implemented via an RTP Translator, an important
consideration is if the two applications being interconnected need to
use the same approach to multiplexing. If one side is using RTP
session multiplexing and the other is using SSRC multiplexing with
BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], it may be possible
for the RTP translator to map the RTP streams between both sides
using some method, e.g. based on the number and order of SDP "m="
lines from each side. There are also challenges with SSRC collision
handling since, unless SSRC translation is applied on the RTP
translator, there may be a collision on the SSRC multiplexing side
that the RTP session multiplexing side will not be aware of.
Furthermore, if one of the applications is capable of working in
several modes (such as being able to use additional RTP streams in
one RTP session or multiple RTP sessions at will), and the other one
is not, successful interconnection depends on locking the more
flexible application into the operating mode where interconnection
can be successful, even if none of the participants are using the
less flexible application when the RTP sessions are being created.
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4.1.3. Gateway Interworking
When one terminates RTP sessions at the gateway, there are certain
tasks that the gateway has to carry out:
o Generating appropriate RTCP reports for all RTP streams (possibly
based on incoming RTCP reports), originating from SSRCs controlled
by the gateway.
o Handling SSRC collision resolution in each application's RTP
sessions.
o Signalling, choosing, and policing appropriate bit-rates for each
session.
For applications that use any security mechanism, e.g., in the form
of SRTP, the gateway needs to be able to decrypt and verify source
integrity of the incoming packets, and re-encrypt, integrity protect,
and sign the packets as the peer in the other application's security
context. This is necessary even if all that's needed is a simple
remapping of SSRC numbers. If this is done, the gateway also needs
to be a member of the security contexts of both sides, and thus a
trusted entity.
The gateway might also need to apply transcoding (for incompatible
codec types), media-level adaptations that cannot be solved through
media negotiation (such as rescaling for incompatible video size
requirements), suppression of content that is known not to be handled
in the destination application, or the addition or removal of
redundancy coding or scalability layers to fit the needs of the
destination domain.
From the above, we can see that the gateway needs to have an intimate
knowledge of the application requirements; a gateway is by its nature
application specific, not a commodity product.
These gateways might therefore potentially block application
evolution by blocking RTP and RTCP extensions that the applications
have been extended with but that are unknown to the gateway.
If one uses security mechanism, like SRTP, the gateway and the
necessary trust in it by the peers is an additional risk to the
communication security. The gateway also incur additional
complexities in form of the decrypt-encrypt cycles needed for each
forwarded packet. SRTP, due to its keying structure, also requires
that each RTP session needs different master keys, as use of the same
key in two RTP sessions can for some ciphers result in a reuse of a
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one-time pad that completely breaks the confidentiality of the
packets.
4.1.4. Multiple SSRC Legacy Considerations
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 endpoint and media type (typically audio or
video). Even in conferencing applications, especially voice-only,
the conference focus or bridge has provided a single stream to each
participant containing a mix of the other participants. It is also
common to have individual RTP sessions between each endpoint and the
RTP mixer, meaning that the mixer functions as an RTP-terminating
gateway.
Applications and systems that aren't updated to handle multiple
streams following these recommendations can have issues with
participating in RTP sessions containing multiple SSRCs within a
single session, such as:
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.
This indicates that gateways attempting to interconnect to this class
of devices have to make sure that only one RTP stream of each media
type gets delivered to the endpoint if it's expecting only one, and
that the multiplexing format is what the device expects. It is
highly unlikely that RTP translator-based interworking can be made to
function successfully in such a context.
4.2. Network Considerations
The RTP implementer needs to consider that the RTP multiplexing
choice also impacts network level mechanisms.
4.2.1. Quality of Service
Quality of Service mechanisms are either flow based or packet marking
based. RSVP [RFC2205] is an example of a flow based mechanism, while
Diff-Serv [RFC2474] is an example of a packet marking based one.
For a flow based scheme, additional SSRC will receive the same QoS as
all other RTP streams being part of the same 5-tuple (protocol,
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source address, destination address, source port, destination port),
which is the most common selector for flow based QoS.
For a packet marking based scheme, the method of multiplexing will
not affect the possibility to use QoS. Different Differentiated
Services Code Points (DSCP) can be assigned to different packets
within a transport flow (5-Tuple) as well as within an RTP stream,
assuming usage of UDP or other transport protocol that do not have
issues with packet reordering within the transport flow (5-tuple).
To avoid packet reording issues, packets belonging to the same RTP
flow should limits its use of DSCP to those whose corresponding Per
Hop Behavior (PHB) that do not enable reordering. If the transport
protocol used assumes in order delivery of packet, such as TCP and
SCTP, then a single DSCP should be used. For more discussion of this
see [RFC7657].
The method for assigning marking to packets can impact what number of
RTP sessions to choose. If this marking is done using a network
ingress function, it can have issues discriminating the different RTP
streams. The network API on the endpoint also needs to be capable of
setting the marking on a per-packet basis to reach the full
functionality.
4.2.2. NAT and Firewall Traversal
In today's networks 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 analyse and comment on the impact of requiring more
underlying transport flows in the presence of NATs and Firewalls:
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 endpoints 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 endpoint 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 endpoint can use. In practice, implementation will
probably benefit from using more server ports to simplify
implementation or avoid performance bottlenecks.
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NAT State: If an endpoint 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 endpoints, available external ports are likely the scarce
resource. Port limitations is primarily a problem for larger
centralised NATs where endpoint 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, as a comparison, a real-time video conference session
with audio and video likely uses 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 Extra Delay: Performing 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 additional NAT/FW traversal time after finding
the first valid candidate pair following the specified ICE
procedures is 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the
pacing timer. That assumes a message in one direction,
immediately followed by a check back. The reason it isn't more,
is that ICE first finds one candidate pair that works prior to
attempting to establish multiple flows. Thus, there is no extra
time until one has found a working candidate pair. Based on that
working pair, the extra time is needed to in parallel establish
the, in most cases 2-3, additional flows. However, packet loss
causes extra delays, at least 500 ms, which is the minimal
retransmission timer for ICE.
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
ought to 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., ought to be reasons for failure.
Deep Packet Inspection and Multiple Streams: Firewalls differ in how
deeply they inspect packets. Due to all previous issues with
firewall and Session Boarder Gateways (SBG) with RTP transport
media e.g. in Voice over IP (VoIP) systems, there exists a
significant risk that deeply inspecting firewalls will have
similar legacy issues with multiple SSRCs as some RTP stack
implementations.
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Using additional RTP streams in the same RTP session and transport
flow does not introduce any additional NAT traversal complexities per
RTP stream. This can be compared with normally one or two additional
transport flows per RTP session when using multiple RTP sessions.
Additional lower layer transport flows will be needed, unless an
explicit de-multiplexing layer is added between RTP and the transport
protocol. At time of writing no such mechanism was defined.
4.2.3. Multicast
Multicast groups provides a powerful tool for a number of real-time
applications, especially the ones that desire broadcast-like
behaviours with one endpoint transmitting to a large number of
receivers, like in IPTV. There is also the RTP/RTCP extension to
better support Source Specific Multicast (SSM) [RFC5760]. Many-to-
many communication, which RTP [RFC3550] was originally built to
support, has several limitations in common with multicast.
One limitation is that, for any group, sender side adaptation with
the intent to suit all receivers would have to adapt to the most
limited receiver experiencing the worst conditions among the group
participants, which imposes degradation for all participants. For
broadcast-type applications with a large number of receivers, 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.
In addition, the transport flow considerations in multicast are a bit
different from unicast; NATs with port translation are not useful in
the multicast environment, meaning that the entire port range of each
multicast address is available for distinguishing between RTP
sessions.
Thus, when using broadcast applications it appears easiest and most
straightforward to use multiple RTP sessions for sending different
media flows used for adapting to network conditions. It is also
common that streams improving transport robustness are sent in their
own multicast group to allow for interworking with legacy or to
support different levels of protection.
Many-to-many applications have different needs and the most
appropriate multiplexing choice will depend on how the actual
application is realized. Multicast applications that are capable of
using sender side congestion control can avoid the use of multiple
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multicast sessions and RTP sessions that result from use of receiver
side congestion control.
The properties of a broadcast application using RTP multicast:
1. Uses a group of RTP sessions, not just one. Each endpoint will
need to be a member of a number of RTP sessions in order to
perform well.
2. Within each RTP session, the number of RTP receivers is likely to
be much larger than the number of RTP senders.
3. The applications need signalling functions to identify the
relationships between RTP sessions.
4. The applications need signalling or RTP/RTCP functions to
identify the relationships between SSRCs in different RTP
sessions when needs beyond CNAME exist.
Both broadcast and many-to-many multicast applications share a
signalling requirement; all of the participants need the same RTP and
payload type configuration. Otherwise, A could for example be using
payload type 97 as the video codec H.264 while B thinks it is MPEG-2.
SDP offer/answer [RFC3264] is not appropriate for ensuring this
property in broadcast/multicast context. The signalling aspects of
broadcast/multicast are not explored further in this memo.
Security solutions for this type of group communication are also
challenging. First, the key-management and the security protocol
need to support group communication. Second, source authentication
requires special solutions. For more discussion on this please
review Options for Securing RTP Sessions [RFC7201].
4.3. Security and Key Management Considerations
When dealing with point-to-point, 2-member RTP sessions only, there
are few security issues that are relevant to the choice of having one
RTP session or multiple RTP sessions. However, there are a few
aspects of multiparty sessions that might warrant consideration. For
general information of possible methods of securing RTP, please
review RTP Security Options [RFC7201].
4.3.1. 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 are (so far) built around symmetric keys, the
receiver will need to have the same key as the sender. This results
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in that no one in a multi-party session can be certain that a
received packet really was sent by the claimed sender and not by
another party having access to the key. The single SRTP algorithm
not having this propery is the TESLA source authentication [RFC4383].
However, TESLA adds delay to achieve source authentication. In most
cases, symmetric ciphers provide sufficient security properties but
create issues in a few cases.
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 RTP
streams. This requires that everyone re-keys without disclosing the
new keys to the excluded party.
A second case is when using security as an enforcing mechanism for
stream access differentiation between different receivers. Take for
example a scalable layer or a high quality simulcast version that
only users paying a premium 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, using the key-management to limit access to
the key.
SRTP [RFC3711] as specified uses per SSRC unique keys, however the
original assumption was a single session master key from which SSRC
specific RTP and RTCP keys where derived. However, that assumption
was proven incorrect, as the application usage and the developed key-
mamangement mechanisms have chosen many different methods for
ensuring SSRC unique keys. The key-management functions have
different capabilities to establish different sets of keys, normally
on a per-endpoint basis. For example, DTLS-SRTP [RFC5764] and
Security Descriptions [RFC4568] establish different keys for outgoing
and incoming traffic from an endpoint. This key usage has to be
written into the cryptographic context, possibly associated with
different SSRCs. Thus, limitations do exist depending on chosen key-
management method and due to integration of particular
implementations of the key-management and SRTP.
4.3.2. Key Management for Multi-party Sessions
The capabilities of the key-management combined with the RTP
multiplexing choices affects the resulting security properties,
control over the secured media, and who have access to it.
Multi-party sessions contain at least one RTP stream from each active
participant. Depending on the multi-party topology [RFC7667], each
participant can both send and receive multiple RTP streams.
Transport translator-based sessions (Topo-Trn-Translator) and
multicast sessions (Topo-ASM), can neither use Security Description
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[RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each
endpoint provides its set of keys. In centralised conferences, the
signalling counterpart is a conference server, and the transport
translator is the media plane unicast counterpart (to which DTLS
messages would be sent). Thus, an extension like Encrypted Key
Transport [I-D.ietf-perc-srtp-ekt-diet] or a MIKEY [RFC3830] based
solution that allows for keying all session participants with the
same master key is needed.
Privacy Enchanced RTP Conferencing (PERC) also enables a different
trust model with semi-trusted media switching RTP middleboxes
[I-D.ietf-perc-private-media-framework].
4.3.3. Complexity Implications
The usage of security functions can surface complexity implications
from the choice of multiplexing and topology. This becomes
especially evident in RTP topologies having any type of middlebox
that processes or modifies RTP/RTCP packets. While there is very
small overhead for an RTP translator or mixer to rewrite an SSRC
value in the RTP packet of an unencrypted session, the cost is higher
when using cryptographic security functions. For example, if using
SRTP [RFC3711], the actual security context and exact crypto key are
determined by the SSRC field value. If one changes SSRC, the
encryption and authentication must use another key. Thus, changing
the SSRC value implies a decryption using the old SSRC and its
security context, followed by an encryption using the new one.
5. RTP Multiplexing Design Choices
This section discusses how some RTP multiplexing design choices can
be used in applications to achieve certain goals, and a summary of
the implications of such choices. For each design there is
discussion of benefits and downsides.
5.1. Multiple Media Types in One Session
This design uses a single RTP session for multiple different media
types, like audio and video, and possibly also transport robustness
mechanisms like FEC or retransmission. An endpoint can send zero,
one or more media sources per media type, resulting in a number of
RTP streams of various media types for both source and redundancy
streams.
The Advantages:
1. Only a single RTP session is used, which implies:
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* Minimal need to keep NAT/FW state.
* Minimal NAT/FW-traversal cost.
* Fate-sharing for all media flows.
* Minimal overhead for security association establishment.
2. Dynamic allocation of RTP streams can be handled almost entirely
at RTP level. How localized this can be kept to RTP level
depends on the application's needs for explicit indication of the
stream usage and how timely that can be signalled.
The Disadvantages:
a. It is less suitable for interworking with other applications that
use individual RTP sessions per media type or multiple sessions
for a single media type, due to the risk of SSRC collision and
thus potential need for SSRC translation.
b. Negotiation of individual bandwidths for the different media
types is currently only possible in SDP when using RID
[I-D.ietf-mmusic-rid].
c. It is not suitable for Split Component Terminal (see Section 3.10
of [RFC7667]).
d. Flow-based QoS cannot be used to provide separate treatment of
RTP streams compared to others in the single RTP session.
e. If there is significant asymmetry between the RTP streams' RTCP
reporting needs, there are some challenges in configuration and
usage to avoid wasting RTCP reporting on the RTP stream that does
not need that frequent reporting.
f. It is not suitable for applications where some receivers like to
receive only a subset of the RTP streams, especially if multicast
or transport translator is being used.
g. There is some additional concern with legacy implementations that
do not support the RTP specification fully when it comes to
handling multiple SSRC per endpoint, as multiple simultaneous
media types are sent as separate SSRC in the same RTP session.
h. If the applications need finer control over which session
participants are included in different sets of security
associations, most key-management mechanisms will have
difficulties establishing such a session.
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5.2. Multiple SSRCs of the Same Media Type
In this design, each RTP session serves only a single media type.
The RTP session can contain multiple RTP streams, either from a
single endpoint or from multiple endpoints. This commonly creates a
low number of RTP sessions, typically only one for audio and one for
video, with a corresponding need for two listening ports when using
RTP/RTCP multiplexing [RFC5761].
The Advantages
1. It works well with Split Component Terminal (see Section 3.10 of
[RFC7667]) where the split is per media type.
2. It enables flow-based QoS with different prioritisation between
media types.
3. For applications with dynamic usage of RTP streams, i.e.
frequently added and removed, having much of the state associated
with the RTP session rather than per individual SSRC can avoid
the need for in-session signalling of meta-information about each
SSRC. In the simple cases this allows for unsignalled RTP
streams where session level information and RTCP SDES item (e.g.
CNAME) are suffient. In the more complex cases where more
source-specific metadata needs to be signalled the SSRC can be
associated with an intermediate identifier, e.g. the MID conveyed
as an SDES item as defined in Section 15 of
[I-D.ietf-mmusic-sdp-bundle-negotiation].
4. There is low overhead for security association establishment.
The Disadvantages
a. There are a slightly higher number of RTP sessions needed
compared to Multiple Media Types in one Session Section 5.1.
This implies:
* More NAT/FW state is needed.
* There is increased NAT/FW-traversal cost in both processing
and delay.
b. There is some potential for concern with legacy implementations
that don't support the RTP specification fully when it comes to
handling multiple SSRC per endpoint.
c. It is not possible to control security association for sets of
RTP streams within the same media type with today's key-
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management mechanisms, unless these are split into different RTP
sessions (Section 5.3).
For RTP applications where all RTP streams of the same media type
share same usage, this structure provides efficiency gains in amount
of network state used and provides more fate sharing with other media
flows of the same type. At the same time, it is still maintaining
almost all functionalities for the negotiation signaling of
properties per individual media type, and also enables flow based QoS
prioritisation between media types. It handles multi-party sessions
well, independently of multicast or centralised transport
distribution, as additional sources can dynamically enter and leave
the session.
5.3. Multiple Sessions for One Media Type
This design goes one step further than above (Section 5.2) by using
multiple RTP sessions also for a single media type. The main reason
for going in this direction is that the RTP application needs
separation of the RTP streams due to their usage, such as e.g.
scalability over multicast, simulcast, need for extended QoS
prioritisation, or the need for fine-grained signalling using RTP
session-focused signalling tools.
The Advantages:
1. This is more suitable for multicast usage where receivers can
individually select which RTP sessions they want to participate
in, assuming each RTP session has its own multicast group.
2. The application can indicate its usage of the RTP streams on RTP
session level, when multiple different usages exist.
3. There is less need for SSRC-specific explicit signalling for each
media stream and thus reduced need for explicit and timely
signalling when RTP streams are added or removed.
4. It enables detailed QoS prioritisation for flow-based mechanisms.
5. It works well with Split Component Terminal (see Section 3.10 of
[RFC7667]).
6. The scope for who is included in a security association can be
structured around the different RTP sessions, thus enabling such
functionality with existing key-management.
The Disadvantages:
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a. There is an increased amount of session configuration state
compared to Multiple SSRCs of the Same Media Type, due to the
increased amount of RTP sessions.
b. For RTP streams that are part of scalability, simulcast or
transport robustness, a method to bind sources across multiple
RTP sessions is needed.
c. There is some potential for concern with legacy implementations
that don't support the RTP specification fully when it comes to
handling multiple SSRC per endpoint.
d. There is higher overhead for security association establishment,
due to the increased number of RTP sessions.
e. If the applications need more fine-grained control than per RTP
session over which participants that are included in different
sets of security associations, most of today's key-management
will have difficulties establishing such a session.
For more complex RTP applications that have several different usages
for RTP streams of the same media type, or uses scalability or
simulcast, this solution can enable those functions at the cost of
increased overhead associated with the additional sessions. This
type of structure is suitable for more advanced applications as well
as multicast-based applications requiring differentiation to
different participants.
5.4. Single SSRC per Endpoint
In this design each endpoint in a point-to-point session has only a
single SSRC, thus the RTP session contains only two SSRCs, one local
and one remote. This session can be used both unidirectional, i.e.
only a single RTP stream, or bi-directional, i.e. both endpoints have
one RTP stream each. If the application needs additional media flows
between the endpoints, it will have to establish additional RTP
sessions.
The Advantages:
1. This design has great legacy interoperability potential as it
will not tax any RTP stack implementations.
2. The signalling has good possibilities to negotiate and describe
the exact formats and bitrates for each RTP stream, especially
using today's tools in SDP.
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3. It is possible to control security association per RTP stream
with current key-management, since each RTP stream is directly
related to an RTP session, and the most used keying mechanisms
operates on a per-session basis.
The Disadvantages:
a. There is a linear growth of the amount of NAT/FW state with
number of RTP streams.
b. There is increased delay and resource consumption from NAT/FW-
traversal.
c. There are likely larger signalling message and signalling
processing requirements due to the increased amount of session-
related information.
d. There is higher potential for a single RTP stream to fail during
transport between the endpoints, due to the need for separate
NAT/FW- traversal for every RTP stream since there is only one
stream per session.
e. The amount of explicit state for relating RTP streams grows,
depending on how the application relates RTP streams.
f. The port consumption might become a problem for centralised
services, where the central node's port or 5-tuple filter
consumption grows rapidly with the number of sessions.
g. For applications where the RTP stream usage is highly dynamic,
i.e. entering and leaving, the amount of signalling can become
high. Issues can also arise from the need for timely
establishment of additional RTP sessions.
h. If, against the recommendation, the same SSRC value is reused in
multiple RTP sessions rather than being randomly chosen,
interworking with applications that use a different multiplexing
structure will require SSRC translation.
RTP applications with a strong need to interwork with legacy RTP
applications can potentially benefit from this structure. However, a
large number of media descriptions in SDP can also run into issues
with existing implementations. For any application needing a larger
number of media flows, the overhead can become very significant.
This structure is also not suitable for non-mixed multi-party
sessions, as any given RTP stream from each participant, although
having same usage in the application, needs its own RTP session. In
addition, the dynamic behaviour that can arise in multi-party
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applications can tax the signalling system and make timely media
establishment more difficult.
5.5. Summary
Both the "Single SSRC per Endpoint" and the "Multiple Media Types in
One Session" are cases that require full explicit signalling of the
media stream relations. However, they operate on two different
levels where the first primarily enables session level binding, and
the second needs SSRC level binding. From another perspective, the
two solutions are the two extreme points when it comes to number of
RTP sessions needed.
The two other designs, "Multiple SSRCs of the Same Media Type" and
"Multiple Sessions for One Media Type", are two examples that
primarily allows for some implicit mapping of the role or usage of
the RTP streams based on which RTP session they appear in. It thus
potentially allows for less signalling and in particular reduces the
need for real-time signalling in sessions with dynamically changing
number of RTP streams. They also represent points in-between the
first two designs when it comes to amount of RTP sessions
established, i.e. representing an attempt to balance the amount of
RTP sessions with the functionality the communication session
provides both on network level and on signalling level.
6. Guidelines
This section contains a number of multi-stream guidelines for
implementers, system designers, or specification writers.
Do not require use of the same SSRC value across RTP sessions:
As discussed in Section 3.4.3 there exist drawbacks in using the
same SSRC in multiple RTP sessions as a mechanism to bind related
RTP streams together. It is instead recommended to use a
mechanism to explicitly signal the relation, either in RTP/RTCP or
in the signalling mechanism used to establish the RTP session(s).
Use additional RTP streams for additional media sources: In the
cases where an RTP endpoint needs to transmit additional RTP
streams of the same media type in the application, with the same
processing requirements at the network and RTP layers, it is
suggested to send them 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 RTP stream within a single RTP session is suggested.
Use additional RTP sessions for streams with different requirements:
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When RTP streams have different processing requirements from the
network or the RTP layer at the endpoints, it is suggested that
the different types of streams are put in different RTP sessions.
This includes the case where different participants want different
subsets of the set of RTP streams.
When using multiple RTP sessions, use grouping: When using multiple
RTP session solutions, it is suggested to explicitly group the
involved RTP sessions when needed using a signalling mechanism,
for example The Session Description Protocol (SDP) Grouping
Framework [RFC5888], using some appropriate grouping semantics.
RTP/RTCP Extensions Support Multiple RTP Streams as Well as Multiple
RTP Sessions:
When defining an RTP or RTCP extension, the creator needs to
consider if this extension is applicable to use with additional
SSRCs and multiple RTP sessions. Any extension intended to be
generic must support both. Extensions that are not as generally
applicable will have to consider if interoperability is better
served by defining a single solution or providing both options.
Extensions for Transport Support: When defining new RTP/RTCP
extensions intended for transport support, like the retransmission
or FEC mechanisms, they must include support for both multiple RTP
streams in the same RTP session and multiple RTP sessions, such
that application developers can choose freely from the set of
mechanisms without concerning themselves with which of the
multiplexing choices a particular solution supports.
7. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section can be removed on publication as an
RFC.
8. Security Considerations
The security considerations of the RTP specification [RFC3550], any
applicable RTP profile [RFC3551],[RFC4585],[RFC3711], and the
extensions for sending multiple media types in a single RTP session
[I-D.ietf-avtcore-multi-media-rtp-session], RID
[I-D.ietf-mmusic-rid], BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation], [RFC5760], [RFC5761], apply
if selected and thus need to be considered in the evaluation.
There is discussion of the security implications of choosing multiple
SSRC vs multiple RTP sessions in Section 4.3.
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9. Contributors
Hui Zheng (Marvin) contributed to WG draft versions -04 and -05 of
the document.
10. Acknowledgments
The Authors like to acknowledge and thank Cullen Jennings, Dale R
Worley, Huang Yihong (Rachel), Benjamin Kaduk, Mirja Kuehlewind, and
Vijay Gurbani for review and comments.
11. References
11.1. Normative References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-13 (work in
progress), December 2015.
[I-D.ietf-mmusic-rid]
Roach, A., "RTP Payload Format Restrictions", draft-ietf-
mmusic-rid-15 (work in progress), May 2018.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-54 (work in progress), December 2018.
[I-D.ietf-perc-srtp-ekt-diet]
Jennings, C., Mattsson, J., McGrew, D., Wing, D., and F.
Andreasen, "Encrypted Key Transport for DTLS and Secure
RTP", draft-ietf-perc-srtp-ekt-diet-11 (work in progress),
January 2020.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
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[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<https://www.rfc-editor.org/info/rfc4585>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<https://www.rfc-editor.org/info/rfc5576>.
[RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
Protocol (RTCP) Extensions for Single-Source Multicast
Sessions with Unicast Feedback", RFC 5760,
DOI 10.17487/RFC5760, February 2010,
<https://www.rfc-editor.org/info/rfc5760>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/info/rfc7656>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
11.2. Informative References
[I-D.ietf-avtext-rid]
Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
Identifier Source Description (SDES)", draft-ietf-avtext-
rid-09 (work in progress), October 2016.
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[I-D.ietf-perc-private-media-framework]
Jones, P., Benham, D., and C. Groves, "A Solution
Framework for Private Media in Privacy Enhanced RTP
Conferencing (PERC)", draft-ietf-perc-private-media-
framework-12 (work in progress), June 2019.
[JINGLE] Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
S., and J. Hildebrand, "XEP-0166: Jingle", XMPP.org
https://xmpp.org/extensions/xep-0166.html, September 2018.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
DOI 10.17487/RFC2198, September 1997,
<https://www.rfc-editor.org/info/rfc2198>.
[RFC2205] Braden, R., Ed., Zhang, L., Berson, S., Herzog, S., and S.
Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
Functional Specification", RFC 2205, DOI 10.17487/RFC2205,
September 1997, <https://www.rfc-editor.org/info/rfc2205>.
[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,
DOI 10.17487/RFC2474, December 1998,
<https://www.rfc-editor.org/info/rfc2474>.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
October 2000, <https://www.rfc-editor.org/info/rfc2974>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
September 2002, <https://www.rfc-editor.org/info/rfc3389>.
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[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
DOI 10.17487/RFC3830, August 2004,
<https://www.rfc-editor.org/info/rfc3830>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<https://www.rfc-editor.org/info/rfc4103>.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383,
DOI 10.17487/RFC4383, February 2006,
<https://www.rfc-editor.org/info/rfc4383>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<https://www.rfc-editor.org/info/rfc4568>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <https://www.rfc-editor.org/info/rfc5109>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008,
<https://www.rfc-editor.org/info/rfc5389>.
[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,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
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[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010,
<https://www.rfc-editor.org/info/rfc5888>.
[RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
time Transport Protocol (RTP) Header Extension for Mixer-
to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011,
<https://www.rfc-editor.org/info/rfc6465>.
[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
<https://www.rfc-editor.org/info/rfc7201>.
[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
(Diffserv) and Real-Time Communication", RFC 7657,
DOI 10.17487/RFC7657, November 2015,
<https://www.rfc-editor.org/info/rfc7657>.
[RFC7826] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
and M. Stiemerling, Ed., "Real-Time Streaming Protocol
Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December
2016, <https://www.rfc-editor.org/info/rfc7826>.
[RFC7983] Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme
Updates for Secure Real-time Transport Protocol (SRTP)
Extension for Datagram Transport Layer Security (DTLS)",
RFC 7983, DOI 10.17487/RFC7983, September 2016,
<https://www.rfc-editor.org/info/rfc7983>.
[RFC8088] Westerlund, M., "How to Write an RTP Payload Format",
RFC 8088, DOI 10.17487/RFC8088, May 2017,
<https://www.rfc-editor.org/info/rfc8088>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
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Appendix A. Dismissing Payload Type Multiplexing
This section documents a number of reasons why using the payload type
as a multiplexing point is unsuitable for most issues related to
multiple RTP streams. Attempting to use Payload type multiplexing
beyond its defined usage has well known negative effects on RTP
discussed below. To use payload type as the single discriminator for
multiple streams implies that all the different RTP streams are being
sent with the same SSRC, thus using the same timestamp and sequence
number space. This has many effects:
1. Putting constraints on RTP timestamp rate for the multiplexed
media. For example, RTP 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 RTP timestamp rate. When
this is not possible, payload type multiplexing cannot be used.
2. Many RTP payload formats can fragment a media object over
multiple RTP 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 RTP 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
a repair mechanism within a single media context. The text/
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 RTP stream before being
used with payload type multiplexing.
4. Sending multiple media streams in the same sequence number space
makes it impossible to determine which media stream lost a
packet. This as the payload type that is used for demultiplex
the media stream is not received. Thus, causing the receiver
difficulties in determining which stream to apply packet loss
concealment or other stream-specific loss mitigation mechanisms.
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
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way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets need be requested, wasting
bandwidth.
6. The current RTCP feedback mechanisms are built around providing
feedback on RTP 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 RTP payload type 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 RTP 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.
12. A legacy endpoint that does not understand the indication that
different RTP payload types are different RTP streams might be
slightly confused by the large amount of possibly overlapping or
identically defined RTP payload types.
Appendix B. Signalling Considerations
Signalling is not an architectural consideration for RTP itself, so
this discussion has been moved to an appendix. However, it is
extremely important for anyone building complete applications, so it
is deserving of discussion.
We document salient issues here that need to be addressed by the WGs
that use some form of signaling to establish RTP sessions. These
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issues cannot simply be addressed by tweaking, extending, or
profiling RTP, but require a dedicated and indepth look at the
signaling primitives that set up the RTP sessions.
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. RTSP
[RFC7826] and SAP [RFC2974] both use SDP in a declarative fashion,
while 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.
B.1. Session Oriented Properties
One aspect of the existing signalling is that it is focused on RTP
sessions, or in the case of SDP, the media description concept.
There are a number of things that are signalled on media description
level but those are not necessarily strictly bound to an RTP session
and could be of interest to signal specifically for a particular RTP
stream (SSRC) 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 as a common
"any RTP stream" limit, unless either codec-specific bandwidth
limiting or RTCP signalling using TMMBR [RFC5104] is used.
o Which SSRC that will use which RTP payload type (this will be
visible from the first media packet, but is sometimes useful to
know before packet arrival).
Some of these issues are clearly SDP's problem rather than RTP
limitations. However, if the aim is to deploy an solution using
additional SSRCs that contains several sets of RTP 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
additional SSRC only, a number of signalling extensions are needed to
clarify that there are multiple sets of RTP streams with different
properties and that they need in fact be kept different, since a
single set will not satisfy the application's requirements.
For some parameters, such as RTP payload type, resolution and
framerate, a SSRC-linked mechanism has been proposed in
[I-D.ietf-mmusic-rid]
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B.2. SDP Prevents Multiple Media Types
SDP chose to use the m= line both to delineate an RTP session and to
specify the top level of the MIME media type; audio, video, text,
image, application. This media type is used as the top-level media
type for identifying the actual payload format and is bound to a
particular payload type using the rtpmap attribute. This binding has
to be loosened in order to use SDP to describe RTP sessions
containing multiple MIME top level types.
[I-D.ietf-mmusic-sdp-bundle-negotiation] describes how to let
multiple SDP media descriptions use a single underlying transport in
SDP, which allows to define one RTP session with media types having
different MIME top level types.
B.3. Signalling RTP Stream Usage
RTP streams being transported in RTP have some particular usage in an
RTP application. This usage of the RTP stream is in many
applications so far implicitly signalled. For example, an
application might choose to take all incoming audio RTP streams, mix
them and play them out. However, in more advanced applications that
use multiple RTP streams there will be more than a single usage or
purpose among the set of RTP streams being sent or received. RTP
applications will need to signal this usage somehow. The signalling
used will have to identify the RTP streams affected by their RTP-
level identifiers, which means that they have to be identified either
by their session or by their SSRC + session.
In some applications, the receiver cannot utilise the RTP stream at
all before it has received the signalling message describing the RTP
stream and its usage. In other applications, there exists a default
handling that is appropriate.
If all RTP streams in an RTP session are to be treated in the same
way, identifying the session is enough. If SSRCs in a session are to
be treated differently, signalling needs to identify both the session
and the SSRC.
If this signalling affects how any RTP central node, like an RTP
mixer or translator that selects, mixes or processes streams, treats
the streams, the node will also need to receive the same signalling
to know how to treat RTP streams with different usage in the right
fashion.
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Authors' Addresses
Magnus Westerlund
Ericsson
Torshamnsgatan 23
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Bo Burman
Ericsson
Gronlandsgatan 31
SE-164 60 Kista
Sweden
Email: bo.burman@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Harald Tveit Alvestrand
Google
Kungsbron 2
Stockholm 11122
Sweden
Email: harald@alvestrand.no
Roni Even
Email: ron.even.tlv@gmail.com
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