Guidelines for
Using the Multiplexing Features of RTP to Support Multiple Media
StreamsEricssonTorshamnsgatan 23164 80KistaSwedenmagnus.westerlund@ericsson.comEricssonGronlandsgatan 31164 60KistaSwedenbo.burman@ericsson.comUniversity of GlasgowSchool of Computing ScienceGlasgowG12 8QQUnited Kingdomcsp@csperkins.orgGoogleKungsbron 2Stockholm11122Swedenharald@alvestrand.noron.even.tlv@gmail.comSimulcastThe 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.IntroductionThe Real-time Transport Protocol (RTP)
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. From the beginning, RTP was designed for
multiple participants in a communication session. It supports many
topology paradigms and usages, as defined in
. RTP has several multiplexing points designed
for different purposes; these points enable support of multiple RTP streams
and switching between different encoding or packetization techniques for the
media. By using multiple RTP sessions, sets of RTP streams can be
structured for efficient processing or identification. Thus,
to meet an application's needs, an RTP application designer needs to understand how best to use the RTP
session, the RTP stream identifier (synchronization source (SSRC)), and the RTP payload type.There has 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) from
which streams of media need to be sent simultaneously. Consequently, questions are raised
regarding the most appropriate RTP usage. The limitations in some
implementations, RTP/RTCP extensions, and signaling have also been
exposed. This document aims to clarify the usefulness
of some functionalities in RTP that, hopefully, will result in future
implementations that are more complete.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. This document
provides some guidelines and recommends against some usages as
being unsuitable, in general or for particular purposes.This document starts with some definitions and then goes into
existing RTP functionalities around multiplexing. Both the desired
behavior and the implications of a particular behavior depend on
which topologies are used; therefore, this topic requires some
consideration. We then discuss some choices regarding multiplexing
behavior and the impacts of those choices. Some designs of RTP usage
are also discussed. Finally, some
guidelines and examples are provided.DefinitionsTerminologyThe definitions in are referenced normatively.The taxonomy defined in
is referenced normatively.The following terms and abbreviations are used in this document:
Multi-party:
Communication that includes multiple endpoints.
In this document, "multi-party" will be used to refer to scenarios where
more than two endpoints communicate.
Multiplexing:
An operation that takes 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 (demultiplexed) 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 include 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
synchronization 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 synchronized for playout.
Signaling:
The process of configuring endpoints to participate in one or
more RTP sessions.
Focus of This DocumentThis document is focused on issues that affect RTP. Thus, issues
that involve signaling 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 signaling protocols -- are
mentioned only as examples in order to describe the RTP issues more
precisely.This document assumes that 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 RTP.RTP Multiplexing OverviewReasons for Multiplexing and Grouping RTP StreamsThere are several reasons why an endpoint might choose to send
multiple media streams. In the discussion below, please keep in mind
that the reasons for having multiple RTP streams vary and include, but
are not limited to, the following:
There might be multiple media sources.
Multiple RTP streams might be needed to represent one media
source, for example:
To carry different layers of a scalable encoding of a media source
Alternative encodings during simulcast, using different codecs for the
same audio stream
Alternative formats during simulcast, multiple resolutions of the same
video stream
A retransmission stream might repeat some parts of the content of
another RTP stream.
A Forward Error Correction (FEC) stream might provide material that
can be used to repair another RTP stream.
For each of these reasons, it is necessary to decide whether each
additional RTP stream is sent within the same RTP session as the other
RTP streams or it is necessary to use additional RTP sessions to
group the RTP streams. For a combination of reasons, the suitable choice for one situation might not
be the suitable choice for another situation. The choice is easiest
when multiplexing multiple media sources of the same
media type. However, all reasons warrant discussion and clarification
regarding how to deal with them. As the discussion below will show,
a single solution does not suit all purposes.
To utilize RTP well and as efficiently as
possible, both are needed.
The real issue is knowing when to create multiple RTP sessions versus when to
send multiple RTP streams in a single RTP session.RTP Multiplexing PointsThis section describes the multiplexing points present in RTP
that can be used to distinguish RTP streams and groups of RTP
streams. outlines
the process of demultiplexing incoming RTP
streams, starting with one or more sockets representing the 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 Session Traversal
Utilities for NAT (STUN)
and DTLS-SRTP on the same transport as
described in .
The Processing and Buffering (PB)
step in terminates
RTP/RTCP and prepares the
RTP payload for input to the decoder.RTP SessionAn RTP session is the highest semantic layer in RTP
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 SSRC space
. That is, if a group of participants are each
aware of the SSRC identifiers belonging to the other
participants, then those participants are in a single RTP session. A
participant can become aware of an SSRC identifier by
receiving an RTP packet containing the identifier in the SSRC field or
contributing source (CSRC) list,
by receiving an RTCP packet listing it in an SSRC field, or through
signaling (e.g., the Session Description Protocol (SDP)
"a=ssrc:" attribute
). 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 signaling.For RTP session separation within a single endpoint, RTP relies on
the underlying transport layer and the signaling 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 signaling 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
their 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 that multiple IP/UDP flows will be
required. Another example is SDP media descriptions (the "m=" line and the
subsequent associated lines) that signal the transport flow and RTP session
configuration for the endpoint's part of the RTP session. The SDP grouping
framework
allows labeling of the media descriptions to be used so that
RTP Session Groups can be created. Through the use of
"Negotiating Media Multiplexing Using the
Session Description Protocol (SDP)",
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.RTP makes no normative statements about the
relationship between different RTP sessions; however, applications
that use more than one RTP session need to understand how the
different RTP sessions that they create relate to one another.Synchronization Source (SSRC)An 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
multiple SSRC identifiers if it sends multiple RTP streams. Endpoints
that function as 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 signaling. Unless presignaled, e.g.,
using the SDP "a=ssrc:" attribute
, 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
and
, 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 a looped source, unless
a mechanism providing a virtual transport (such as Interactive
Connectivity Establishment (ICE)
) abstracts the changes.SSRC identifiers that belong to the same synchronization context
(i.e., that represent RTP streams that can be synchronized using
information in RTCP SR packets) use identical CNAME chunks in
corresponding RTCP source description (SDES) packets. SDP signaling can also be used to
provide explicit SSRC grouping
.In some cases, the same SSRC identifier value is used to relate
streams in two different RTP sessions, such as in RTP retransmission
. This is to be avoided, since there is no
guarantee that SSRC values are unique across RTP sessions. In the
case of RTP retransmission
,
it is recommended to use explicit binding of the source RTP
stream and the redundancy stream, e.g., using the RepairedRtpStreamId
RTCP SDES item . The
RepairedRtpStreamId is a rather recent mechanism, so one cannot expect
older applications to follow this recommendation.
Note that the RTP sequence number and RTP timestamp are scoped by the
SSRC and are 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.
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. Using RTCP compound
packets containing the CNAME SDES item is the designated method for
binding 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 synchronization context and can be
synchronized 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 its SSRC number due to an SSRC
conflict. Using the media identification (MID) extension
helps to identify
which media source the new SSRC represents, and using the
restriction identifier (RID) extension
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.Contributing Source (CSRC)The 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 Multipoint Control Unit (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 where
only a single CSRC is indicated, as this represents the forwarding of an RTP
stream that might have been modified. The RTP header extension ("A Real-time Transport Protocol (RTP)
Header Extension for Mixer-to-Client Audio Level Indication")
expands on the receiver's information about a packet with a CSRC list.
Due to these restrictions, a CSRC will not be considered a fully
qualified multiplexing point and will be disregarded in the rest of
this document.RTP Payload TypeEach RTP stream utilizes 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
for statically allocated payload types but ought to be explicitly
defined in signaling, such as SDP, for both static and dynamic
payload types. The term "format" here includes those aspects described
by out-of-band signaling 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 .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 instant in time in the RTP stream's
timestamp timeline, 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
. 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
can be used. Several additional constraints, other than those mentioned
above, need to be met to enable this usage, one of which is that the
combined payload sizes of the different payload types ought not exceed
the transport MTU.Other aspects of using the RTP payload format are described in
"How to Write an RTP Payload Format".The payload type is not a multiplexing point at the RTP layer (see
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 signaling, which in general requires that unique RTP payload
type numbers be used in each context. Using MID, e.g., when bundling "m=" sections
,
can replace the payload type as a signaling association, and unique
RTP payload types are then no longer required for that purpose.Issues Related to RTP TopologiesThe impact of how RTP multiplexing is performed will in general
vary with how the RTP session participants are interconnected,
as described in
"RTP Topologies".Even the most basic use case -- "Topo-Point-to-Point" as described in
-- raises a number of
considerations, which are
discussed in detail in the following sections. They range over such
aspects as the following:
Does my communication peer support RTP as defined with multiple
SSRCs per RTP session?
Do I need network differentiation in the form of QoS
()?
Can the application more easily process and handle the media
streams if they are in different RTP sessions?
Do I need to use additional RTP streams for RTP retransmission or FEC?
For some point-to-multipoint topologies (e.g., Topo-ASM and
Topo-SSM
), multicast is used to interconnect the
session participants. Special considerations (documented in
) are then needed, as multicast is a
one-to-many distribution system.Sometimes, an RTP communication session can end up in a situation where the
communicating peers are not compatible, for various reasons:
No common media codec for a media type, thus requiring transcoding.
Different support for multiple RTP streams and RTP sessions.
Usage of different media transport protocols (i.e., one peer
uses RTP, but the other peer uses a different transport protocol).
Usage of different transport protocols, e.g., UDP, the Datagram
Congestion Control Protocol (DCCP), or TCP.
Different security solutions (e.g., IPsec, TLS, DTLS, or the
Secure Real-time Transport Protocol (SRTP)) with
different keying mechanisms.
These compatibility issues can often be resolved by the inclusion of a
translator between the two peers -- the Topo-PtP-Translator, as
described in
. The translator's main purpose is to make the
peers look compatible to each other. There can also be reasons other
than compatibility for inserting 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 require a thorough understanding of aspects ranging from congestion control
and media-level adaptations 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.Issues Related to RTP and RTCPUsing 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 that
need to be taken into account.The RTP SpecificationRFC 3550 contains some
recommendations and a numbered list () of five arguments regarding different
aspects of RTP multiplexing. Please review . Five important aspects are
quoted below.
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.
This argument advocates the use of different SSRCs for each individual RTP
stream, as this is fundamental to RTP operation.
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.
This argument advocates against demultiplexing RTP
streams within a session based only on their RTP payload type numbers;
it still stands, as can be seen by the extensive list of issues
discussed in .
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.
This argument is yet another argument against payload type
multiplexing.
An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
This argument advocates 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 the 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.
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.
This argument discusses network aspects that are described in
. It also goes into aspects of
implementation, like split component terminals (see
) -- endpoints where different processes or
interconnected devices handle different aspects of the whole
multimedia session.
To summarize, RFC 3550's view on multiplexing is to use unique SSRCs
for anything that is its own media/packet stream and 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. "Sending
Multiple Types of Media in a Single RTP Session"
updates RFC 3550 to allow multiple media types in an RTP session
and provides a detailed analysis of the potential benefits
and issues related to having
multiple media types in the same RTP session. Thus, 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.Multiple SSRCs in a SessionUsing multiple SSRCs at one endpoint in an RTP session requires
that some unclear aspects of the RTP specification be resolved. These
items could potentially lead to some interoperability issues as
well as some potential significant inefficiencies, as further
discussed in "Sending Multiple RTP Streams in a Single RTP Session"
. An RTP
application designer should consider these issues and the
application's possible impact caused by a 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.Binding Related SourcesA 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:
RTP/RTCP based
Signaling based, e.g., SDP
Grouping related RTP sessions
Grouping SSRCs within an RTP session
Most solutions are explicit, but some implicit methods have also
been applied to the problem.The SDP-based signaling solutions are:
SDP media description grouping:
The SDP grouping framework uses various semantics to group any number of
media descriptions. SDP media description grouping has primarily
been used to group RTP sessions,
but in combination with ,
it can also group multiple media descriptions within a single RTP
session.
SDP media multiplexing:
"Negotiating Media
Multiplexing Using the Session Description Protocol (SDP)"
uses information taken from both SDP and RTCP to associate RTP streams to SDP media
descriptions. This allows both SDP and RTCP to group RTP streams belonging to
an SDP media description and group multiple SDP media
descriptions into a single RTP session.
SDP SSRC grouping:
"Source-Specific Media Attributes in
the Session Description Protocol (SDP)" includes a solution for grouping
SSRCs in the same
way that 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.One RTP/RTCP-based grouping solution is to use the RTCP SDES CNAME to bind
related RTP streams to an endpoint or a synchronization context. For
applications with a single RTP stream per type (media, source, or
redundancy stream), the CNAME is sufficient for that purpose, independent of whether one or more RTP sessions
are used. However, some applications choose not to use a 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
,
when configured to use multiple RTP sessions, and generic FEC
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 a change of that SSRC in all RTP
sessions and will thus resynchronize all of the streams instead of only the single
media stream experiencing the collision.Another method for implicitly binding SSRCs is used by RTP
retransmission
when using the same RTP session as the source RTP stream for retransmissions.
A receiver that is missing a packet issues an RTP retransmission
request and then awaits a new SSRC carrying the RTP retransmission
payload, where that SSRC is from the same CNAME. This limits a
requester to having only one outstanding retransmission request on any
new SSRCs per endpoint."RTP Payload Format Restrictions"
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 signaling 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
streams use RID
identifiers. The redundancy stream uses the RepairedRtpStreamId
SDES item and RTP header extension to declare the RtpStreamId
value of the source stream to create the binding.Experience has shown that an explicit binding between the RTP streams,
agnostic of SSRC values, behaves well. That way, solutions using
multiple RTP streams in a single RTP session and in multiple RTP sessions
will use the same type of binding.Forward Error CorrectionThere exist a number of FEC-based schemes designed to mitigate packet loss in the original streams.
Most of the FEC schemes protect a single source flow. This
protection is achieved by transmitting a certain amount of redundant
information that is encoded such that it can repair one or more
instances of packet
loss 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 utilize the FEC information. By placing
it in a separate RTP session and if separating RTP sessions at the
transport level, FEC can easily be ignored at the transport level,
without considering any RTP-layer information.In usages involving multicast, sending FEC information in a separate 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 suitable FEC data repair
capabilities.Considerations for RTP MultiplexingInterworking ConsiderationsThere are several different kinds of interworking, and this section
discusses two: interworking directly between different applications and
the 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.Application InterworkingIt is not uncommon that applications or services of similar but not
identical usage, especially those intended for interactive
communication, encounter a situation where one wants 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
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 trust model being used. Secondly, the applications
need to use media streams in a way that makes sense in both applications.
RTP Translator InterworkingFrom 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 regarding the 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
, 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 related to
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.Gateway InterworkingWhen one terminates RTP sessions at the gateway, there are certain
tasks that the gateway has to carry out:
Generating appropriate RTCP reports for all RTP streams (possibly
based on incoming RTCP reports) originating from SSRCs controlled by
the gateway.
Handling SSRC collision resolution in each application's RTP sessions.
Signaling, choosing, and policing appropriate bitrates 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 then 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 and 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 a security mechanism like SRTP, the gateway and the
necessary trust in it by the peers pose an additional risk to
communication security. The gateway also incurs additional
complexities in the
form of the decrypt-encrypt cycles needed for each forwarded packet.
SRTP, due to its keying structure, also requires that each RTP session
need different master keys, as the use of the same key in two RTP
sessions can, for some ciphers, result in a reuse of a one-time pad that
completely breaks the confidentiality of the packets.Legacy Considerations for Multiple SSRCsHistorically, 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 provides to each participant a single stream
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:
The need to handle more than one stream simultaneously rather than
replacing an already-existing stream with a new one.
Being capable of decoding multiple streams simultaneously.
Being 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.Network ConsiderationsThe RTP implementer needs to consider that the RTP multiplexing choice
also impacts network-level mechanisms.Quality of ServiceQoS mechanisms are either flow based or packet marking
based. RSVP
is an example of a flow-based mechanism, while Diffserv
is an example of a packet-marking-based mechanism.For a flow-based scheme, additional SSRCs will receive the
same QoS as all other 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.For a packet-marking-based scheme, the method of multiplexing will
not affect the possibility of using QoS. Different
Differentiated Services Code Points (DSCPs) can be assigned to
different packets within a transport flow (5-tuple) as well as within an RTP stream,
assuming the usage of UDP or other transport protocols that do not have issues
with packet reordering within the transport flow (5-tuple).
To avoid packet-reordering issues, packets belonging to the same RTP
flow should limit their use of DSCPs to packets whose corresponding
Per-Hop Behavior (PHB) do not enable reordering. If the transport protocol being used assumes in&nbhy;order
delivery of packets (e.g., TCP and the Stream Control Transmission
Protocol (SCTP)),
then a single DSCP should be used.
For more discussion on this topic, see .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 full
functionality.NAT and Firewall TraversalIn today's networks, there exist a large number of middleboxes. Those
that normally have the most impact on RTP are Network Address
Translators (NATs) and Firewalls (FWs).Below, we analyze and comment on the impact of requiring more
underlying transport flows in the presence of NATs and FWs:
Endpoint 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 a
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
the 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 is the maximum number of simultaneous
RTP sessions a single endpoint can use. In practice, implementations
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&wj;/Home Office (SOHO) NATs, the most limited resource is
memory or processing. For large-scale NATs serving many internal
endpoints, available external ports are likely the scarce resource.
Port limitations are primarily a problem for larger centralized 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.
Extra Delay Added by NAT Traversal:
Performing the NAT/FW traversal takes a
certain amount of time for each flow. 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
return message in the opposite direction to confirm reachability.
It isn't more, because 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
establish the additional flows (two or three, in most cases)
in parallel. However, packet
loss causes extra delays of at least 500 ms (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 will succeed but one or more of the additional
flows will fail.
The risk of this happening is hard to quantify but should be fairly
low, as one flow from the same interfaces has just been successfully
established. Thus, only such rare events as NAT resource overload,
selecting particular port numbers that are filtered, etc., ought to be
reasons for failure.
Deep Packet Inspection and Multiple Streams:
FWs differ in how
deeply they inspect packets.
Previous experience using FWs and Session Border Gateways
(SBGs) with RTP shows that there is a significant risk that
the FWs and SBGs will reject RTP sessions that use multiple SSRCs.
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 demultiplexing layer is added between RTP and the transport
protocol. At the time of this writing, no such mechanism was defined.MulticastMulticast groups provide a powerful tool for a number of real-time
applications, especially those that desire broadcast-like
behaviors with one endpoint transmitting to a large number of
receivers, like in IPTV. An RTP/RTCP extension to
better support Source-Specific Multicast (SSM)
is also available. Many-to-many communication, which RTP
was originally built to support, has several limitations in common with
multicast.One limitation is that, for any group, sender-side adaptations 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 the 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 in 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 applications 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 multicast sessions and RTP
sessions that result from the use of receiver-side congestion control.The properties of a broadcast application using RTP multicast are
as follows:
The application 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.
Within each RTP session, the number of RTP receivers is likely to
be much larger than the number of RTP senders.
The application needs signaling functions to identify the
relationships between RTP sessions.
The application needs signaling or RTP/RTCP functions to identify
the relationships between SSRCs in different RTP sessions when
more complex relations than those that can be expressed by the CNAME exist.
Both broadcast and many-to-many multicast applications share a
signaling 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
is not appropriate for ensuring this property in a broadcast/multicast
context. The signaling 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 mechanism and the security protocol need
to support group communication. Second, source authentication requires
special solutions. For more discussion on this topic, please review "Options for Securing RTP Sessions".Security and Key-Management ConsiderationsWhen dealing with point-to-point two-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 multi-party sessions that might warrant consideration. For general
information regarding possible methods of securing RTP, please review
.Security Context ScopeWhen using SRTP
,
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. As a result, no one in a multi-party
session can be certain that a received packet was really sent by the
claimed sender and not by another party having access to the key. The
single SRTP algorithm not having this property is Timed
Efficient Stream Loss-Tolerant Authentication (TESLA) source
authentication . However, TESLA adds delay
to achieve source authentication. In most cases, symmetric ciphers
provide sufficient security properties, but in a few cases they can 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 rekey without disclosing the
new keys to the excluded party.A second case is when security is used 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 users can't access without paying a premium,
using the key-management mechanism to limit access to the key.As specified in , SRTP uses
unique keys per SSRC;
however, the original assumption was a single-session master key from
which SSRC-specific RTP and RTCP keys were derived. However, that
assumption was proven incorrect, as the application usage and
the developed key-management mechanisms have chosen many different
methods for ensuring unique keys per SSRC. The key-management functions have different
abilities to establish different sets of keys, normally on a
per-endpoint basis. For example, DTLS-SRTP
and Security Descriptions
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 the chosen key-management method and due to
the integration
of particular implementations of the key-management method and SRTP.Key Management for Multi-party SessionsThe capabilities of the key-management method combined with the RTP multiplexing
choices affect the resulting security properties, control over the
secured media, and who has access to it.Multi-party sessions contain at least one RTP stream from each active
participant. Depending on the multi-party topology
,
each participant can both send and receive multiple RTP streams.
Transport translator-based sessions (Topo-Trn-Translator) and multicast
sessions (Topo-ASM) can use neither Security Descriptions
nor DTLS-SRTP
without an extension, because each endpoint provides its own set of
keys. In
centralized conferences, the signaling 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
or a solution based on Multimedia Internet KEYing (MIKEY) that allows for
keying all session participants with the same master key is needed.Privacy-Enhanced RTP Conferencing (PERC) also enables a different
trust model with semi-trusted media-switching RTP middleboxes
.Complexity ImplicationsThere can be complex interactions between the choice of
multiplexing and topology and the security functions. This becomes especially
evident in RTP topologies having any type of middlebox that processes
or modifies RTP/RTCP packets. While the overhead of
an RTP translator or mixer rewriting an SSRC value in the RTP packet
of an unencrypted session is low, the cost is higher when using cryptographic
security functions. For example, if using SRTP
, the actual security context and exact crypto
key are determined by the SSRC field value. If one changes the
SSRC value, 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.RTP Multiplexing Design ChoicesThis section discusses how some RTP multiplexing design choices can
be used in applications to achieve certain goals and summarizes the
implications of such choices. The benefits and downsides of each
design are also discussed.Multiple Media Types in One SessionThis 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 multiple media sources per media type, resulting in a number of RTP
streams of various media types for both source and redundancy streams.Advantages:
Only a single RTP session is used, which implies:
Minimal need to keep NAT/FW state.
Minimal NAT/FW traversal cost.
Fate-sharing for all media flows.
Minimal overhead for security association establishment.
Dynamic allocation of RTP streams can be handled almost entirely
at the RTP level.
The extent to which this allocation can be kept at the RTP level depends on the application's needs
for an explicit indication of stream usage and in how timely a
fashion that information can be signaled.
Disadvantages:
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 collisions and thus a potential
need for SSRC translation.
Negotiation of individual bandwidths for the different media types is
currently only possible in SDP when using RID
.
It is not suitable for split component terminals (see
).
Flow-based QoS cannot be used to provide separate treatment of RTP
streams compared to others in the single RTP session.
If there is significant asymmetry between the RTP streams' RTCP
reporting needs, there are some challenges related to configuration and usage
to avoid wasting RTCP reporting on the RTP stream that does not need
such frequent reporting.
It is not suitable for applications where some receivers like to receive
only a subset of the RTP streams, especially if multicast or a transport
translator is being used.
There are some additional concerns regarding legacy implementations that do
not support the RTP specification fully when it comes to handling multiple
SSRCs per endpoint, as multiple simultaneous media types are sent as
separate SSRCs in the same RTP session.
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.
Multiple SSRCs of the Same Media TypeIn this design, each RTP session serves only a single media type.
The RTP session can contain multiple RTP streams, from either a single
endpoint or 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
.Advantages:
It works well with split component terminals (see ) where the
split is per media type.
It enables flow-based QoS with different prioritization levels between media
types.
For applications with dynamic usage of RTP streams (i.e.,
streams are 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 signaling of meta-information about each SSRC. In simple
cases, this allows for unsignaled RTP streams where session-level
information and an RTCP SDES item (e.g., CNAME) are
sufficient. In the more complex cases where more source-specific metadata needs to be
signaled, the SSRC can be associated with an intermediate identifier,
e.g., the MID conveyed as an SDES item as defined in
.
The overhead of security association establishment is low.
Disadvantages:
A slightly higher number of RTP sessions are needed, compared
to multiple media types in one session
(). This implies the following:
More NAT/FW state is needed.
The cost of NAT/FW traversal is increased in terms of both processing and delay.
There is some potential for concern regarding legacy implementations that don't
support the RTP specification fully when it comes to handling multiple
SSRCs per endpoint.
It is not possible to control security associations 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 the same usage, this structure provides efficiency gains in
the amount
of network state used and provides more fate-sharing with other media
flows of the same type. At the same time, it still maintains
almost all functionalities for the negotiation signaling of properties per
individual media type and also
enables flow-based QoS prioritization between media types. It handles
multi-party sessions well, independently of multicast or centralized
transport distribution, as additional sources can dynamically enter
and leave the session.Multiple Sessions for One Media TypeThis design goes one step further than the design discussed in
by also using multiple RTP sessions for a single media type. The main
reason for going in this direction is that the RTP application needs
separation of the RTP streams according to their usage, such as, for example, scalability
over multicast, simulcast, the need for extended QoS prioritization, or the need
for fine-grained signaling using RTP session-focused signaling tools.Advantages:
This design is more suitable for multicast usage where receivers can individually
select which RTP sessions they want to participate in, assuming
that each
RTP session has its own multicast group.
When multiple different usages exist, the application can
indicate its usage of the RTP streams at the RTP
session level.
There is less need for SSRC-specific explicit signaling for each media
stream and thus a reduced need for explicit and timely signaling when
RTP streams are added or removed.
It enables detailed QoS prioritization for flow-based mechanisms.
It works well with split component terminals (see
).
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 mechanisms.
Disadvantages:
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.
For RTP streams that are part of scalability, simulcast, or
transport robustness, a method for binding sources across multiple RTP
sessions is needed.
There is some potential for concern regarding legacy implementations that
don't support the RTP specification fully when it comes to handling
multiple SSRCs per endpoint.
The overhead of security association establishment is higher, due
to the increased number of RTP sessions.
If the applications need finer control over which participants
in a given RTP session are included in different sets of
security associations, most of today's key-management mechanisms
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 that use 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.Single SSRC per EndpointIn 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 either unidirectionally
(i.e., one SSRC sends an RTP stream that is received by the other
SSRC) or bidirectionally (i.e., the two SSRCs both send an RTP
stream and receive the RTP stream sent by the other endpoint).
If the application needs additional media flows
between the endpoints, it will have to establish additional RTP
sessions.Advantages:
This design has great potential for interoperability with legacy
applications, as it will
not tax any RTP stack implementations.
The signaling system makes it possible to negotiate and describe
the exact formats and bitrates for each RTP stream, especially
using today's tools in SDP.
It is possible to control security associations per RTP stream with
current key-management functions, since each RTP stream is directly related to
an RTP session and the most commonly used keying mechanisms operate on a
per-session basis.
Disadvantages:
The amount of NAT/FW state grows linearly with the number
of RTP streams.
NAT/FW traversal increases delay and resource consumption.
There are likely more signaling message and signaling processing
requirements due to the increased amount of session-related information.
There is higher potential for a single RTP stream to fail during
transport between the endpoints, due to the need for a separate
NAT/FW traversal for every RTP stream, since there is only one stream per session.
The amount of explicit state for relating RTP streams grows, depending
on how the application relates RTP streams.
Port consumption might become a problem for centralized
services, where the central node's port or 5-tuple filter consumption
grows rapidly with the number of sessions.
For applications where RTP stream usage is highly dynamic,
i.e., entities frequently enter and leave sessions, the amount of signaling can become high. Issues
can also arise from the need for timely establishment of additional RTP
sessions.
If, against the recommendation in , 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 the same usage in the
application, needs its own RTP session. In addition, the dynamic
behavior that can arise in multi-party applications can tax the
signaling system and make timely media establishment more difficult.SummaryBoth the "single SSRC per endpoint" () and "multiple media types in one
session" () cases require full explicit signaling of the media
stream relationships. 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 extremes when it comes to the 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
allow for some implicit mapping of the role or usage of the RTP
streams based on which RTP session they appear in. Thus, they potentially
allow for less signaling and, in particular, reduce the need for
real-time signaling in sessions with a dynamically changing number
of RTP streams. They also represent points
between the first two designs when it comes to the amount of RTP
sessions established, i.e., they represent an attempt to balance the
amount of RTP sessions with the functionality the communication
session provides at both the network level and the signaling level.GuidelinesThis section contains a number of multi-stream guidelines for
implementers, system designers, and specification writers.
Do not require the use of the same SSRC value across RTP sessions:
As discussed in ,
there are downsides to 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 relationship,
in either RTP&wj;/RTCP or the signaling 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, in the case of a telepresence room
where there are three cameras and each camera captures two persons
sitting at the table, we suggest that each camera send its own RTP stream within
a single RTP session.
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 be put in different RTP sessions. This includes the
case where different participants want different subsets of the set of
RTP streams.
Use grouping when using multiple RTP sessions:
When
using multiple RTP session solutions, it is suggested to explicitly
group the involved RTP sessions when needed using a signaling
mechanism -- for example, see "The Session
Description Protocol (SDP) Grouping Framework" -- using some appropriate grouping semantics.
Ensure that 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 for 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
whether interoperability is better served by defining a single solution or
providing both options.
Provide adequate 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.
IANA ConsiderationsThis document has no IANA actions.Security ConsiderationsThe security considerations discussed in the RTP specification
;
any applicable RTP profile
;
and the extensions for sending multiple media types in a single RTP
session
, RID
, BUNDLE
,
, and
apply if selected and thus need to be considered in the evaluation. discusses the security implications of choosing
multiple SSRCs vs. multiple RTP sessions.ReferencesNormative ReferencesSending Multiple Types of Media in a Single RTP SessionRTP Payload Format RestrictionsNegotiating Media Multiplexing Using the Session Description Protocol (SDP)RTP Stream Identifier Source Description (SDES)Encrypted Key Transport for DTLS and Secure RTPcompanycompanycompanycompanycompanyInformative ReferencesA Solution Framework for Private Media in Privacy-Enhanced RTP Conferencing (PERC)XEP-0166: JingleDismissing Payload Type MultiplexingThis 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, as
discussed below.
To use the 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.
The many effects of using payload type multiplexing are as follows:
Constraints are placed on the 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.
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 be done relatively easily
on the sender side by ensuring that the fragments of each RTP stream
are sent in sequence.
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&wj;/t140 payload format
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.
Sending multiple media streams in the same sequence number space
makes it
impossible to determine which media stream lost a packet.
Such a scenario causes difficulties, since the receiver cannot determine to which stream it should
apply packet-loss concealment or other stream-specific
loss-mitigation mechanisms.
If RTP retransmission
is used and packet loss occurs, 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 to tell which missing packet or packets belong to the
stream or streams of interest, and all lost packets need to be requested, wasting bandwidth.
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.
The current RTCP media control messages specification
is oriented around controlling particular media flows,
i.e., requests are done by addressing a particular SSRC. Such mechanisms
would need to be redefined to support payload type multiplexing.
The number of payload types is 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
where a number of payload types are blocked due to the overlap between
RTP and RTCP.
At times, there is a need to group multiplexed streams. This is
currently possible for RTP sessions and SSRCs, but there is no
defined way to group payload types.
It is currently not possible to signal bandwidth requirements per
RTP stream when using payload type multiplexing.
Most existing SDP media-level attributes cannot be applied on a
per-payload-type basis and would require redefinition in that context.
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.
Signaling ConsiderationsSignaling 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 some issues here that need to be addressed when using some form of signaling to establish RTP sessions. These
issues cannot be addressed by simply tweaking, extending, or profiling
RTP; rather, they require a dedicated and in-depth look at the signaling
primitives that set up the RTP sessions.There exist various signaling solutions for establishing RTP
sessions. Many are based on SDP
;
however, SDP functionality is also dependent on the signaling
protocols carrying the SDP. The Real-Time Streaming Protocol (RTSP)
and the Session Announcement Protocol (SAP)
both use SDP in a declarative fashion, while SIP
uses SDP with the additional definition of offer/answer
. The impact on signaling,
and especially on SDP,
needs to be considered, as it can greatly affect how to deploy a
certain multiplexing point choice.Session-Oriented PropertiesOne aspect of existing signaling protocols is that they are focused on
RTP sessions or, in the case of SDP, the concept of media
descriptions. A number of things are signaled at the media
description level, but those are not necessarily strictly bound to
an RTP session and could be of interest for signaling, especially
for a particular RTP stream (SSRC) within the session.
The following properties have been
identified as being potentially useful for signaling, and not only
at the RTP session level:
Bitrate and/or bandwidth can be specified today only as an
aggregate limit, or as a common "any RTP stream" limit, unless
either codec-specific bandwidth limiting or
RTCP signaling using Temporary Maximum Media Stream Bit Rate
Request (TMMBR) messages is used.
Which SSRC will use which RTP payload type (this information will be
visible in the first media packet but is sometimes useful to have
before the packet arrives).
Some of these issues are clearly SDP's problem rather than RTP
limitations. However, if the aim is to deploy a solution that uses
several SSRCs and contains several sets of RTP streams with
different properties (encoding/packetization parameters, bitrate,
etc.), putting each set in a different RTP session would directly
enable negotiation of the parameters for each set. If insisting on
additional SSRCs only, a number of signaling extensions are needed to
clarify that there are multiple sets of RTP streams with different
properties and that they in fact need to be kept different, since a
single set will not satisfy the application's requirements.For some parameters, such as RTP payload type, resolution, and
frame rate, an SSRC-linked mechanism has been proposed in
.SDP Prevents Multiple Media TypesSDP uses the "m=" line to both delineate an RTP session and specify
the top-level 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 "a=rtpmap:" attribute. This binding has to be loosened in
order to use SDP to describe RTP sessions containing multiple
top-level media types.
describes how to let multiple SDP media descriptions use a single
underlying transport in SDP, which allows the definition of one RTP session
with different top-level media types.Signaling RTP Stream UsageRTP streams being transported in RTP have a particular usage in
an RTP application. In many applications to date, this usage of the RTP
stream is implicitly signaled. 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 somehow signal this usage. The signaling that is used will have to
identify the RTP streams affected by their RTP-level identifiers,
which means that they have to be identified by either their session or
their SSRC + session.In some applications, the receiver cannot utilize the RTP stream at
all before it has received the signaling message describing the RTP
stream and its usage. In other applications, there exists a default
handling method 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, signaling needs to identify both the session
and the SSRC.If this signaling 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 signaling to
know how to treat RTP streams with different usages in the right
fashion.AcknowledgmentsThe authors would like to acknowledge and thank , , , , , and for review and comments.Contributors contributed to WG draft versions -04
and -05 of the document.