Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: Informational Ericsson
Expires: June 10, 2019 C. Perkins
University of Glasgow
H. Alvestrand
Google
R. Even
Huawei
December 07, 2018
Guidelines for using the Multiplexing Features of RTP to Support
Multiple Media Streams
draft-ietf-avtcore-multiplex-guidelines-07
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
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
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Internet-Drafts are draft documents valid for a maximum of six months
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on June 10, 2019.
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Copyright Notice
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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) . . . . . . . . . . . . . 9
3.2.4. RTP Payload Type . . . . . . . . . . . . . . . . . . 10
3.3. Issues Related to RTP Topologies . . . . . . . . . . . . 11
3.4. Issues Related to RTP and RTCP Protocol . . . . . . . . . 12
3.4.1. The RTP Specification . . . . . . . . . . . . . . . . 12
3.4.2. Multiple SSRCs in a Session . . . . . . . . . . . . . 14
3.4.3. Binding Related Sources . . . . . . . . . . . . . . . 14
3.4.4. Forward Error Correction . . . . . . . . . . . . . . 16
4. Considerations for RTP Multiplexing . . . . . . . . . . . . . 17
4.1. Interworking Considerations . . . . . . . . . . . . . . . 17
4.1.1. Application Interworking . . . . . . . . . . . . . . 17
4.1.2. RTP Translator Interworking . . . . . . . . . . . . . 17
4.1.3. Gateway Interworking . . . . . . . . . . . . . . . . 18
4.1.4. Multiple SSRC Legacy Considerations . . . . . . . . . 19
4.2. Network Considerations . . . . . . . . . . . . . . . . . 20
4.2.1. Quality of Service . . . . . . . . . . . . . . . . . 20
4.2.2. NAT and Firewall Traversal . . . . . . . . . . . . . 20
4.2.3. Multicast . . . . . . . . . . . . . . . . . . . . . . 22
4.3. Security and Key Management Considerations . . . . . . . 23
4.3.1. Security Context Scope . . . . . . . . . . . . . . . 24
4.3.2. Key Management for Multi-party sessions . . . . . . . 24
4.3.3. Complexity Implications . . . . . . . . . . . . . . . 25
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5. RTP Multiplexing Design Choices . . . . . . . . . . . . . . . 25
5.1. Multiple Media Types in one Session . . . . . . . . . . . 25
5.2. Multiple SSRCs of the Same Media Type . . . . . . . . . . 26
5.3. Multiple Sessions for one Media type . . . . . . . . . . 28
5.4. Single SSRC per Endpoint . . . . . . . . . . . . . . . . 29
5.5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 30
6. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . 31
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 32
8. Security Considerations . . . . . . . . . . . . . . . . . . . 32
9. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 32
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 32
10.1. Normative References . . . . . . . . . . . . . . . . . . 33
10.2. Informative References . . . . . . . . . . . . . . . . . 33
Appendix A. Dismissing Payload Type Multiplexing . . . . . . . . 37
Appendix B. Signalling Considerations . . . . . . . . . . . . . 39
B.1. Session Oriented Properties . . . . . . . . . . . . . . . 39
B.2. SDP Prevents Multiple Media Types . . . . . . . . . . . . 40
B.3. Signalling RTP stream Usage . . . . . . . . . . . . . . . 40
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 41
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, the question for any RTP
application designer is 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 has also
been exposed. The authors also hope that clarification on the
usefulness of some functionalities in RTP will result in more
complete implementations in the future.
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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 that come
from a particular usage of the RTP multiplexing points. The document
will 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 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 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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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, 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 when using layered encodings)
o A retransmission stream might repeat some parts of the content of
another RTP stream
o An FEC stream might provide material that can be used to repair
another RTP stream
o Alternative encodings, for instance using different codecs for the
same audio stream
o Alternative formats, for instance multiple resolutions of the same
video stream
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
reason, might not be the choice suitable for another reason. The
clearest understanding is associated with multiplexing multiple media
sources of the same media type. However, all reasons warrant
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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. 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:
|
| packets
+-- v
| +------------+
| | Socket | Transport Protocol Demultiplexing
| +------------+
| || ||
RTP | RTP/ || |+-----> SCTP ( ...and any other protocols)
Session | RTCP || +------> STUN (multiplexed using same port)
+-- ||
+-- ||
| (split by SSRC)
| || || ||
| || || ||
RTP | +--+ +--+ +--+
Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, etc.
| +--+ +--+ +--+
+-- | | |
(select decoder based on PT)
+-- | / |
| +----+ |
| / | |
Payload | +---+ +---+ +---+
Formats | |Dec| |Dec| |Dec| Decoders
| +---+ +---+ +---+
+--
Figure 1: RTP Demultiplexing Process
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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 RTP sessions must be
possible to separate both across different endpoints and within 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
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. 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 RTP endpoint can identify and delimit an RTP session from other
RTP sessions by receiving the multiple UDP flows used as identified
based on their UDP source and destination IP addresses and UDP port
numbers. Another example is SDP media descriptions (the "m=" line
and the following associated lines) signals the transport flow and
RTP session configuration for the endpoints part of the RTP session.
SDP grouping framework [RFC5888] allows labeling of the media
descriptions, for example used so that RTP Session Groups can be
created. With Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)[I-D.ietf-mmusic-sdp-bundle-negotiation],
multiple media descriptions where each represents the RTP streams
sent or received for a media source are part of a common RTP session.
The RTP protocol makes no normative statements about the relationship
between different RTP sessions, however the applications that use
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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 an 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
multiple SSRC identifiers if it sends multiple RTP streams.
Endpoints that are both RTP streams sender and RTP receiver use the
same SSRC 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 RTP sending endpoint that changes its network transport
address during a session have to choose a new SSRC identifier to
avoid being interpreted as looped source, unless the transport layer
mechanism, e.g ICE [RFC8445], handles such 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].
Note that RTP sequence number and RTP timestamp are scoped by the
SSRC and thus specific per RTP stream.
An SSRC identifier is used by different type of sources as well as
receivers:
Real Media Source: Connected to a "physical" media source, for
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example a camera or microphone.
Conceptual Media Source: A media source with some attributed
property generated by some network node, for example a filtering
function in an RTP mixer that provides the most active speaker
based on some criteria, or a mix representing a set of other
sources.
RTP receiver: A receiver that does not generate any RTP stream
themself (e.g. an endpoint or middlebox only receiving in an RTP
session). It still needs an SSRC for use as source in RTCP
reports.
Note that an endpoint that generates more than one media type, e.g.
a conference participant sending both audio and video, need not (and
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. 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, so the new SSRC is anyway
effectively 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, if the corresponding SSRC was in the
header of one of the packets that contributed to the mix.
It is not possible, in general, to extract media represented by an
individual CSRC since it is typically the result of a media mixing
(merge) operation by an RTP mixer on the individual media streams
corresponding to the CSRC identifiers. 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
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multiplexing point and will be disregarded in the rest of this
document.
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 used 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 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 whatever can
be 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 used 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. If it is acceptable to send multiple formats of the same media
source as separate RTP streams (with separate SSRC), simulcast
[I-D.ietf-mmusic-sdp-simulcast] can be used.
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; this is not recommended since a specific
payload type value can be used in multiple bundled "m=" sections
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[I-D.ietf-mmusic-sdp-bundle-negotiation]. This association is only
possible if unique RTP payload type numbers are used in each context.
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?
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
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gateway, for example a need to monitor the RTP streams. If the
stream transport characteristics are changed by the translator,
appropriate media handling can require thorough understanding of the
application logic, specifically any congestion control or media
adaptation.
The point to point topology can contain one to many RTP sessions with
one to many 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. Let's review
Section 5.2 of [RFC3550], reproduced below:
"For efficient protocol processing, the number of multiplexing points
should be minimised, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which
is different for each RTP session. For example, in a teleconference
composed of audio and video media encoded separately, each medium
SHOULD be carried in a separate RTP session with its own destination
transport address.
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields. Interleaving packets with different RTP media types but
using the same SSRC would introduce several problems:
1. If, say, two audio streams shared the same RTP session and the
same SSRC value, and one were to change encodings and thus
acquire a different RTP payload type, there would be no general
way of identifying which stream had changed encodings.
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
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require different sequence number spaces to tell which payload
type suffered packet loss.
3. The RTCP sender and receiver reports (see Section 6.4) can only
describe one timing and sequence number space per SSRC and do not
carry a payload type field.
4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
5. Carrying multiple media in one RTP session precludes: the use of
different network paths or network resource allocations if
appropriate; reception of a subset of the media if desired, for
example just audio if video would exceed the available bandwidth;
and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations.
Using a different SSRC for each medium but sending them in the same
RTP session would avoid the first three problems but not the last
two.
On the other hand, multiplexing multiple related sources of the same
medium in one RTP session using different SSRC values is the norm for
multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It might also be
appropriate to multiplex streams of the same medium using different
SSRC values in other scenarios where the last two problems do not
apply."
Let's consider one argument at a time. The first argument is for
using different SSRC for each individual RTP stream, which is
fundamental to RTP operation.
The second argument is advocating against demultiplexing RTP streams
within a session based on their RTP payload type numbers, which still
stands as can been seen by the extensive list of issues found in
Appendix A.
The third argument is yet another argument against payload type
multiplexing.
The fourth argument is against multiplexing RTP packets that require
different handling into the same session. As we saw in the
discussion of RTP mixers, the RTP mixer must embed application logic
to handle streams anyway; the separation of streams according to
stream type is just another piece of application logic, which might
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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; most other types of applications need
application-specific logic to perform the mix correctly.
The fifth argument discusses network aspects that we will discuss
more below 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]
provides a detailed analysis of the potential issues in having
multiple media types in the same RTP session. This 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.
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
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o Signalling based (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.
These has previously been considered primarily as grouping RTP
sessions, [I-D.ietf-mmusic-sdp-bundle-negotiation] groups multiple
media descriptions as a single RTP session.
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.
This supports a lot of use cases. All these solutions have
shortcomings in cases where the session's dynamic properties are such
that it is difficult or resource consuming to keep the list of
related SSRCs up to date.
An RTP/RTCP-based solution is to use the RTCP SDES CNAME to bind the
RTP streams to an endpoint or synchronization context. For
applications with a single RTP stream per type (Media, Source or
Redundancy) this is sufficient independent if one or more RTP
sessions are used. However, some applications choose not to use it
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 this method. This
method may work but might have some downsides in RTP sessions with
many participating SSRCs. When an SSRC collision occurs, this will
force one to change SSRC in all RTP sessions and thus resynchronize
all of them instead of only the single media stream having the
collision. Therefore, it is not recommended to use identical SSRC
values to relate RTP streams.
Another solution to bind SSRCs is an implicit method used by RTP
Retransmission [RFC4588] when doing retransmissions in the same RTP
session as the source RTP stream. 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
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the same CNAME. This limits a requester to having only one
outstanding 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
"rtp-stream-id" 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.
It can be noted that Section 8.3 of the RTP Specification [RFC3550]
recommends using a single SSRC space across all RTP sessions for
layered coding. Based on the experience so far however, we recommend
to use a solution doing explicit binding between the RTP streams so
what the used SSRC values are do not matter. That way solutions
using multiple RTP streams in a single RTP session and multiple RTP
sessions uses the same solution.
3.4.4. Forward Error Correction
There exist a number of Forward Error Correction (FEC) based schemes
for how to reduce the packet loss of the original streams. Most of
the FEC schemes will protect a single source flow. The protection is
achieved by transmitting a certain amount of redundant information
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 also needs to be transmitted as its
own media stream, or in some cases, instead of the original media
stream. Thus, many of these schemes 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 transport
level.
In usages involving multicast, having the FEC information on its own
multicast group allows for similar flexibility. This is especially
useful when receivers see very heterogeneous packet loss rates.
Those receivers that are not seeing packet loss don't need to join
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the multicast group with the FEC data, and so avoid the overhead of
receiving unnecessary FEC packets, for example.
4. Considerations for RTP Multiplexing
4.1. Interworking Considerations
There are several different kinds of interworking, and this section
discusses two; interworking between different applications including
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.
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 an
RTP Translator interworking (RTP bridging), where the gateway acts as
an RTP Translator, with the two applications being members of the
same RTP session; or Gateway Interworking with RTP termination, where
there are independent RTP sessions running from each interconnected
application to the gateway.
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, the mapping of SDP "m=" lines between both sides requires
that the order in bundled and not bundled sides will be the same to
allow routing without mapping, it is possible for the RTP translator
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to map the RTP streams between both sides. There are also challenges
with SSRC collision handling since there may be a collision on the
SSRC multiplexing side but the RTP session multiplexing side will not
be aware of any collision unless SSRC translation is applied on the
RTP translator. 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 no participants are using
the less flexible application when the RTP sessions are being
created.
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 uses any security mechanism, e.g., in the form
of SRTP, the gateway needs to be able to decrypt incoming packets and
re-encrypt them 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, of course.
Other tasks a gateway might need to apply include 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.
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This fact reveals the potential for these gateways to 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 functions, like SRTP, and as can be seen from
above, they incur both additional risk due to the requirement to have
the gateway in the security association between the endpoints (unless
the gateway is on the transport level), and 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 two-time pads 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 with a
mix of the other participants to each participant. 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.
When establishing RTP sessions that can contain endpoints that aren't
updated to handle multiple streams following these recommendations, a
particular application can have issues with multiple SSRCs within a
single session. These issues include:
1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously.
3. Be capable of rendering multiple streams simultaneously.
This indicates that gateways attempting to interconnect to this class
of devices has to make sure that only one RTP stream of each 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.
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4.2. Network Considerations
The RTP multiplexing choice has impact on network level mechanisms
that need to be considered by the implementer.
4.2.1. Quality of Service
When it comes to Quality of Service mechanisms, they 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 packet marking based scheme, the
method of multiplexing will not affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between
the multiplexing methods. Additional SSRC will result in all RTP
streams being part of the same 5-tuple (protocol, source address,
destination address, source port, destination port) which is the most
common selector for flow based QoS.
It must also be noted that packet marking based QoS mechanisms can
have limitations. A general observation is that different
Differentiated Services Code Points (DSCP) can be assigned to
different packets within a flow as well as within an RTP stream.
However, care must be taken when considering which forwarding
behaviours that are applied on path due to these DSCPs. In some
cases the forwarding behaviour can result in packet reordering. 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 network there exist a large number of middleboxes. The
ones that normally have most impact on RTP are Network Address
Translators (NAT) and Firewalls (FW).
Below we 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
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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.
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, it is worth pointing out that a real-time video
conference session with audio and video is likely using less than
10 UDP flows, compared to certain web applications that can use
100+ TCP flows to various servers from a single browser instance.
NAT Traversal 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 how much extra time it takes after finding the
first valid candidate pair following the specified ICE procedures
are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing
timer. That assumes a message in one direction, and then an
immediate triggered 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 needed extra time is to in parallel establish the, in
most cases 2-3, additional flows. However, packet loss causes
extra delays, at least 100 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
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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. There exist some potential that
deeply inspecting firewalls will have similar legacy issues with
multiple SSRCs as some stack implementations.
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 are also the RTP/RTCP extension to
better support Source Specific Multicast (SSM) [RFC5760]. Another
application is the Many to Many communication, which RTP [RFC3550]
was originally built to support. But the multicast semantics do
result in a certain number of limitations.
One limitation is that for any group, sender side adaptation to the
actual receiver properties causes degradation for all participants to
what is supported by the receiver with the worst conditions among the
group participants. For broadcast type of applications 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
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media flows used for adapting to network conditions. It is also
common that streams that improve transport robustness are sent in
their own multicast group to allow for interworking with legacy or to
support different levels of protection.
For many to many applications there is different needs. Here it will
depend on how the actual application is realized what is the most
appropriate choice. With sender side congestion control there might
not exist any benefit with using multiple RTP session.
The properties of a broadcast application using RTP multicast:
1. Uses a group of RTP sessions, not 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 exists.
Both broadcast and many to many multicast applications do share a
signalling requirement; all of the participants will need to have 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. It is to be noted that 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 communications are also
challenging. First, the key-management and the security protocol
needs 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
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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 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. At least unless TESLA source authentication
[RFC4383], which adds delay to achieve source authentication. In
most cases symmetric ciphers provide sufficient security properties,
but there are a few cases where this does create issues.
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
keys to the excluded party.
A second case is when using security as an enforcing mechanism for
differentiation. Take for example a scalable layer or a high quality
simulcast version that only premium users are allowed to access. The
mechanism preventing a receiver from getting the high quality stream
can be based on the stream being encrypted with a key that user can't
access without paying premium, having the key-management limit access
to the key.
SRTP [RFC3711] has no special functions for dealing with different
sets of master keys for different SSRCs. 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.
4.3.2. Key Management for Multi-party sessions
Performing key-management for multi-party session can be a challenge.
This section considers some of the issues.
Multi-party sessions, such as transport translator based sessions and
multicast sessions, cannot use Security Description [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 media plane unicast
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counterpart (to which DTLS messages would be sent) is the transport
translator. 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.
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. Where 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 have zero,
one or more media sources per media type. Resulting in a number of
RTP streams of various media types and both source and redundancy
type.
The Pros:
1. Single RTP session which implies:
* Minimal NAT/FW state.
* Minimal NAT/FW Traversal Cost.
* Fate-sharing for all media flows.
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2. Can handle dynamic allocations of RTP streams well on an RTP
level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be
signalled.
3. Minimal overhead for security association establishment.
The Cons:
a. Less suitable for interworking with other applications that uses
individual RTP sessions per media type or multiple sessions for a
single media type, due to the potential need of SSRC translation.
b. Negotiation of bandwidth for the different media types is
currently only possible using RID [I-D.ietf-mmusic-rid] in SDP.
c. Not suitable for Split Component Terminal (see Section 3.10 of
[RFC7667]).
d. Flow-based QoS cannot 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. 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. Additional concern with legacy implementations that do not
support the RTP specification fully when it comes to handling
multiple SSRC per endpoint, as also multiple simultaneous media
types needs to be handled.
h. If the applications need finer control over which session
participants that are included in different sets of security
associations, most key-management will have difficulties
establishing such a session.
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
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video, with a corresponding need for two listening ports when using
RTP/RTCP multiplexing.
The Pros:
1. Works well with Split Component Terminal (see Section 3.10 of
[RFC7667]) where the split is per media type.
2. 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.
4. Low overhead for security association establishment.
The Cons:
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
* Increased NAT/FW Traversal Cost in both processing and delay.
b. 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. Not possible to control security association for sets of RTP
streams within the same media type with today's key- management
mechanisms, unless these are split into different RTP sessions.
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 when it comes to negotiation in the
signalling of the properties for the individual media type, and also
enables flow based QoS prioritisation between media types. It
handles multi-party session well, independently of multicast or
centralised transport distribution, as additional sources can
dynamically enter and leave the session.
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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. Some typical
reasons for going to this design are scalability over multicast,
simulcast, need for extended QoS prioritisation of RTP streams due to
their usage in the application, or the need for fine- grained
signalling using today's tools.
The Pros:
1. 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, in case multiple different usages exist.
3. Less need for SSRC specific explicit signalling for each media
stream and thus reduced need for explicit and timely signalling.
4. Enables detailed QoS prioritisation for flow-based mechanisms.
5. 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 Cons:
a. Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type.
b. Increased amount of session configuration state.
c. For RTP streams that are part of scalability, simulcast or
transport robustness, a method to bind sources across multiple
RTP sessions is needed.
d. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per endpoint.
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e. Higher overhead for security association establishment due to the
increased number of RTP sessions.
f. If the applications need finer control than on RTP session level
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, they will have to establish additional RTP
sessions.
The Pros:
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 bit-rates for each RTP stream, especially
using today's tools in SDP.
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 Cons:
a. The number of RTP sessions grows directly in proportion with the
number of RTP streams, which has the implications:
* Linear growth of the amount of NAT/FW state with number of RTP
streams.
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* Increased delay and resource consumption from NAT/FW
traversal.
* Likely larger signalling message and signalling processing
requirement due to the amount of session related information.
* Higher potential for a single RTP stream to fail during
transport between the endpoints.
b. When the number of RTP sessions grows, the amount of explicit
state for relating RTP stream also grows, depending on how the
application needs to relate RTP streams.
c. 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.
d. For applications where the RTP stream usage is highly dynamic,
i.e. entering and leaving, the amount of signalling can grow
high. Issues can also arise from the timely establishment of
additional RTP sessions.
e. 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 that 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 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 applications can tax the
signalling system and make timely media establishment more difficult.
5.5. Summary
There are some clear similarities between these designs. 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.
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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 dynamic sessions. 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 or specification writers.
Do not require us of the same SSRC value across RTP sessions: As dis
cussed 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:
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.
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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.
Transport Support Extensions: 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 sessions 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] and
any applicable RTP profile [RFC3551],[RFC4585],[RFC3711], 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 needs 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.
9. Contributors
Hui Zheng (Marvin) from Huawei contributed to WG draft versions -04
and -05 of the document.
10. References
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10.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[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>.
10.2. Informative References
[ALF] Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols", SIGCOMM
Symposium on Communications Architectures and
Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE
Computer Communications Review, Vol. 20(4), September
1990.
[I-D.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-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.
[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-53 (work in progress), September 2018.
[I-D.ietf-mmusic-sdp-simulcast]
Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
"Using Simulcast in SDP and RTP Sessions", draft-ietf-
mmusic-sdp-simulcast-13 (work in progress), June 2018.
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[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-09 (work in progress),
October 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>.
[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>.
[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>.
[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>.
[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>.
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[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>.
[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>.
[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>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
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[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>.
[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>.
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 things related to
multiple RTP streams. If one attempts to use Payload type
multiplexing beyond its defined usage, that has well known negative
effects on RTP. To use payload type as the single discriminator for
multiple streams implies that all the different RTP streams are being
sent with the same SSRC, thus using the same timestamp and sequence
number space. This has many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, 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.
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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 streams in the same sequence number space makes
it impossible to determine which payload type, which stream a
packet loss relates to, and thus to which stream to potentially
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
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.
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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 hugely
important for anyone building complete applications, so it is
deserving of discussion.
The issues raised here need to be addressed in the WGs that deal with
signalling; they cannot be addressed by tweaking, extending or
profiling RTP.
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 around
RTP sessions, or at least in the case of SDP the media description.
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 is used.
o Which SSRC that will use which RTP payload types (this will be
visible from the first media packet, but is sometimes useful to
know before packet arrival).
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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]
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 has 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.
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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.
Authors' Addresses
Magnus Westerlund
Ericsson
Torshamsgatan 23
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Bo Burman
Ericsson
Gronlandsgatan 31
SE-164 80 Kista
Sweden
Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
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Harald Tveit Alvestrand
Google
Kungsbron 2
Stockholm 11122
Sweden
Email: harald@alvestrand.no
Roni Even
Huawei
Email: roni.even@huawei.com
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