AVTCORE WG M. Westerlund
Internet-Draft Ericsson
Updates: 3550, 3551 (if approved) C. Perkins
Intended status: Standards Track University of Glasgow
Expires: August 29, 2013 J. Lennox
Vidyo
February 25, 2013
Multiple Media Types in an RTP Session
draft-ietf-avtcore-multi-media-rtp-session-02
Abstract
This document specifies how an RTP session can contain media streams
with media from multiple media types such as audio, video, and text.
This has been restricted by the RTP Specification, and thus this
document updates RFC 3550 and RFC 3551 to enable this behaviour for
applications that satisfy the applicability for using multiple media
types in a single RTP session.
Status of this Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on August 29, 2013.
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to this document. Code Components extracted from this document must
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. NAT and Firewalls . . . . . . . . . . . . . . . . . . . . 4
3.2. No Transport Level QoS . . . . . . . . . . . . . . . . . . 5
3.3. Architectural Equality . . . . . . . . . . . . . . . . . . 5
4. Overview of Solution . . . . . . . . . . . . . . . . . . . . . 5
5. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 6
5.1. Usage of the RTP session . . . . . . . . . . . . . . . . . 6
5.2. Signalled Support . . . . . . . . . . . . . . . . . . . . 7
5.3. Homogeneous Multi-party . . . . . . . . . . . . . . . . . 7
5.4. Reduced number of Payload Types . . . . . . . . . . . . . 8
5.5. Stream Differentiation . . . . . . . . . . . . . . . . . . 8
5.6. Non-compatible Extensions . . . . . . . . . . . . . . . . 9
6. RTP Session Specification . . . . . . . . . . . . . . . . . . 9
6.1. RTP Session . . . . . . . . . . . . . . . . . . . . . . . 10
6.2. Sender Source Restrictions . . . . . . . . . . . . . . . . 12
6.3. Payload Type Applicability . . . . . . . . . . . . . . . . 12
6.4. RTCP . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
6.4.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . 14
6.4.2. Tuning RTCP transmissions . . . . . . . . . . . . . . 14
7. Extension Considerations . . . . . . . . . . . . . . . . . . . 17
7.1. RTP Retransmission . . . . . . . . . . . . . . . . . . . . 17
7.2. Generic FEC . . . . . . . . . . . . . . . . . . . . . . . 18
8. Signalling . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8.1. SDP-Based Signalling . . . . . . . . . . . . . . . . . . . 19
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
10. Security Considerations . . . . . . . . . . . . . . . . . . . 19
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
12.1. Normative References . . . . . . . . . . . . . . . . . . . 20
12.2. Informative References . . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
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1. Introduction
When the Real-time Transport Protocol (RTP) [RFC3550] was designed,
close to 20 years ago, IP networks were very different compared to
the ones in 2013 when this is written. The almost ubiquitous
deployment of Network Address Translators (NAT) and Firewalls has
increased the cost and likely-hood of communication failure when
using many different transport flows. Thus there exists a pressure
to reduce the number of concurrent transport flows.
RTP [RFC3550] recommends against sending several different types of
media, for example audio and video, in a single RTP session. The RTP
profile for Audio and Video Conferences with Minimal Control (RTP/
AVP) [RFC3551] mandates a similar restriction. The motivation for
these limitations is partly to allow lower layer Quality of Service
(QoS) mechanisms to be used, and partly due to limitations of the
RTCP timing rules that assumes all media in a session to have similar
bandwidth. The Session Description Protocol (SDP) [RFC4566], as one
of the dominant signalling method for establishing RTP session, has
enforced this rule, simply by not allowing multiple media types for a
given receiver destination or set of ICE candidates, which is the
most common method to determine which RTP session the packets are
intended for.
The fact that these limitations have been in place for so long a
time, in addition to RFC 3550 being written without fully considering
multiple media types in an RTP session, does result in a number of
considerations being needed when allowing this behaviour. This
document provides such considerations regarding applicability as well
as functionality, including normative specification of behaviour.
First, some basic definitions are provided. This is followed by a
background that discusses the motivation in more detail. A overview
of the solution of how to provide multiple media types in one RTP
session is then presented. Next is the formal applicability this
specification have followed by the normative specification. This is
followed by a discussion how some RTP/RTCP Extensions is expected to
function in the case of multiple media types in one RTP session. A
specification of the requirements on signalling from this
specification and a look how this is realized in SDP using Bundle
[I-D.ietf-mmusic-sdp-bundle-negotiation]. The document ends with the
security considerations.
2. Definitions
The following terms are used with supplied definitions:
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Endpoint: A single entity sending or receiving RTP packets. It can
be decomposed into several functional blocks, but as long as it
behaves as a single RTP stack entity it is classified as a single
endpoint.
Media Stream: A sequence of RTP packets using a single SSRC that
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
Media Type: Audio, video, text or application whose form and meaning
are defined by a specific real-time application.
QoS: Quality of Service, i.e. network mechanisms that intended to
ensure that the packets within a flow or with a specific marking
are transported with certain properties.
RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That
is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints. An endpoint can receive an SSRC
either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the endpoints' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by endpoints and any interconnecting middle nodes.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Motivation
This section discusses in more detail the main motivations why
allowing multiple media types in the same RTP session is suitable.
3.1. NAT and Firewalls
The existence of NATs and Firewalls at almost all Internet access has
had implications on protocols like RTP that were designed to use
multiple transport flows. First of all, the NAT/FW traversal
solution needs to ensure that all these transport flows are
established. This has three consequences:
1. Increased delay to perform the transport flow establishment
2. The more transport flows, the more state and the more resource
consumption in the NAT and Firewalls. When the resource
consumption in NAT/FWs reaches their limits, unexpected
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behaviours usually occur.
3. More transport flows means a higher risk that some transport flow
fails to be established, thus preventing the application to
communicate.
Using fewer transport flows reduces the risk of communication
failure, improved establishment behaviour and less load on NAT and
Firewalls.
3.2. No Transport Level QoS
Many RTP-using applications don't utilize any network level Quality
of Service functions. Nor do they expect or desire any separation in
network treatment of its media packets, independent of whether they
are audio, video or text. When an application has no such desire, it
doesn't need to provide a transport flow structure that simplifies
flow based QoS.
3.3. Architectural Equality
For applications that don't require different lower-layer QoS for
different media types, and that have no special requirements for RTP
extensions or RTCP reporting, the requirement to separate different
media into different RTP sessions might seem unnecessary. Provided
the application accepts that all media flows will get similar RTCP
reporting, using the same RTP session for several types of media at
once appears a reasonable choice. The architecture ought to be
agnostic about the type of media being carried in an RTP session to
the extent possible given the constraints of the protocol.
4. Overview of Solution
The goal of the solution is to enable each RTP session to contain
more than just one media type. This includes having multiple RTP
sessions containing a given media type, for example having three
sessions containing both video and audio.
The solution is quite straightforward. The first step is to override
the SHOULD and SHOULD NOT language of the RTP specification
[RFC3550]. Similar change is needed to a sentence in Section 6 of
[RFC3551] that states that "different media types SHALL NOT be
interleaved or multiplexed within a single RTP Session". This is
resolved by appropriate exception clauses given that this
specification and its applicability is followed.
Within an RTP session where multiple media types have been configured
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for use, an SSRC can only send one type of media during its lifetime
(i.e., it can switch between different audio codecs, since those are
both the same type of media, but cannot switch between audio and
video). Different SSRCs MUST be used for the different media
sources, the same way multiple media sources of the same media type
already have to do. The payload type will inform a receiver which
media type the SSRC is being used for. Thus the payload type MUST be
unique across all of the payload configurations independent of media
type that is used in the RTP session.
Some few extra considerations within the RTP sessions also needs to
be considered. RTCP bandwidth and regular reporting suppression
(RTP/AVPF and RTP/SAVPF) SHOULD be configured to reduce the impact
for bit-rate variations between streams and media types. It is also
clarified how timeout calculations are to be done to avoid any
issues. Certain payload types like FEC also need additional rules.
The final important part of the solution to this is to use signalling
and ensure that agreement on using multiple media types in an RTP
session exists, and how that then is configured. This memo describes
some existing requirements, while an external reference defines how
this is accomplished in SDP.
5. Applicability
This specification has limited applicability, and anyone intending to
use it needs to ensure that their application and usage meets the
below criteria.
5.1. Usage of the RTP session
Before choosing to use this specification, an application implementer
needs to ensure that they don't have a need for different RTP
sessions between the media types for some reason. The main rule is
that if one expects to have equal treatment of all media packets,
then this specification might be suitable. The equal treatment
include anything from network level up to RTCP reporting and
feedback. The document Guidelines for using the Multiplexing
Features of RTP [I-D.westerlund-avtcore-multiplex-architecture] gives
more detailed guidance on aspects to consider when choosing how to
use RTP and specifically sessions. RTP-using applications that need
or would prefer multiple RTP sessions, but do not require the
functionalities or behaviours that multiple transport flows give, can
consider using Multiple RTP Sessions on a Single Lower-Layer
Transport [I-D.westerlund-avtcore-transport-multiplexing]. It needs
to be noted that some difference in treatment is still possible to
achieve, for example marking based QoS, or RTCP feedback traffic for
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only some media streams.
The second important consideration is the resulting behaviour when
media flows to be sent within a single RTP session does not have
similar bandwidth. There are limitations in the RTCP timing rules,
and this implies a common RTCP reporting interval across all
participants in a session. If an RTP session contains flows with
very different bandwidths, for example low-rate audio coupled with
high-quality video, this can result in either excessive or
insufficient RTCP for some flows, depending how the RTCP session
bandwidth, and hence reporting interval, is configured. This is
discussed further in Section 6.4.
5.2. Signalled Support
Usage of this specification is not compatible with anyone following
RFC 3550 and intending to have different RTP sessions for each media
type. Therefore there needs to be mutual agreement to use multiple
media types in one RTP session by all participants within that RTP
session. This agreement has to be determined using signalling in
most cases.
This requirement can be a problem for signalling solutions that can't
negotiate with all participants. For declarative signalling
solutions, mandating that the session is using multiple media types
in one RTP session can be a way of attempting to ensure that all
participants in the RTP session follow the requirement. However, for
signalling solutions that lack methods for enforcing that a receiver
supports a specific feature, this can still cause issues.
5.3. Homogeneous Multi-party
In multiparty communication scenarios it is important to separate two
different cases. One case is where the RTP session contains multiple
participants in a common RTP session. This occurs for example in Any
Source Multicast (ASM) and Transport Translator topologies as defined
in RTP Topologies [RFC5117]. It can also occur in some
implementations of RTP mixers that share the same SSRC/CSRC space
across all participants. The second case is when the RTP session is
terminated in a middlebox and the other participants sources are
projected or switched into each RTP session and rewritten on RTP
header level including SSRC mappings.
For the first case, with a common RTP session or at least shared
SSRC/CSRC values, all participants in multiparty communication are
REQUIRED to support multiple media types in an RTP session. An
participant using two or more RTP sessions towards a multiparty
session can't be collapsed into a single session with multiple media
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types. The reason is that in case of multiple RTP sessions, the same
SSRC value can be use in both RTP sessions without any issues, but
when collapsed to a single session there is an SSRC collision. In
addition some collisions can't be represented in the multiple
separate RTP sessions. For example, in a session with audio and
video, an SSRC value used for video will not show up in the Audio RTP
session at the participant using multiple RTP sessions, and thus not
trigger any collision handling. Thus any application using this type
of RTP session structure MUST have a homogeneous support for multiple
media types in one RTP session, or be forced to insert a translator
node between that participant and the rest of the RTP session.
For the second case of separate RTP sessions for each multiparty
participant and a central node it is possible to have a mix of single
RTP session users and multiple RTP session users as long as one is
willing to remap the SSRCs used by a participant with multiple RTP
sessions into non-used values in the single RTP session SSRC space
for each of the participants using a single RTP session with multiple
media types. It can be noted that this type of implementation has to
understand all types of RTP/RTCP extension being used in the RTP
sessions to correctly be able to translate them between the RTP
sessions. It can also negatively impact the possibility for loop
detection, as SSRC/CSRC can't be used to detect the loops, instead
some other media stream identity name space that is common across all
interconnect parts are needed.
5.4. Reduced number of Payload Types
An RTP session with multiple media types in it have only a single
7-bit Payload Type range for all its payload types. Within the 128
available values, only 96 or less if "Multiplexing RTP Data and
Control Packets on a Single Port" [RFC5761] is used, all the
different RTP payload configurations for all the media types need to
fit in the available space. For most applications this will not be a
real problem, but the limitation exists and could be encountered.
5.5. Stream Differentiation
If network level differentiation of the media streams of different
media types are desired using this specification can cause severe
limitations. All media streams in an RTP session, independent of the
media type, will be sent over the same underlying transport flow.
Any flow-based Quality of Service (QoS) mechanism will be unable to
provide differentiated treatment between different media types, e.g.
to prioritize audio over video. If differentiated treatment is
desired using flow-based QoS, separate RTP sessions over different
underlying transport flows needs to be used.
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Any marking-based QoS scheme like DiffServ is not affected unless a
network ingress marks based on flows, in which case the same
considerations as for flow based QoS applies.
5.6. Non-compatible Extensions
There exist some RTP and RTCP extensions that rely on the existence
of multiple RTP sessions. If the goal of using an RTP session with
multiple media types is to have only a single RTP session, then these
extensions can't be used. If one has no need to have different RTP
sessions for the media types but is willing to have multiple RTP
sessions, one for the main media transmission and one for the
extension, they can be used. It is to be noted that this assumes
that it is possible to get the extension working when the related RTP
session contains multiple media types.
Identified RTP/RTCP extensions that require multiple RTP Sessions
are:
RTP Retransmission: RTP Retransmission [RFC4588] has a session
multiplexed mode. It also has a SSRC multiplexed mode that can be
used instead. So use the mode that is suitable for the RTP
application.
XOR-Based FEC: The RTP Payload Format for Generic Forward Error
Correction [RFC5109] and its predecessor [RFC2733] requires a
separate RTP session unless the FEC data is carried in RTP Payload
for Redundant Audio Data [RFC2198]. However, using the Generic
FEC with the Redundancy payload has another set of restrictions,
see Section 7.2.
Note that the Source-Specific Media Attributes [RFC5576]
specification defines an SDP syntax (the "FEC" semantic of the
"ssrc-group" attribute) to signal FEC relationships between
multiple media streams within a single RTP session. However, this
can't be used as the FEC repair packets need to have the same SSRC
value as the source packets being protected. [RFC5576] does not
normatively update and resolve that restriction. There is ongoing
work on an ULP extension to allow it be use FEC streams within the
same RTP Session as the source stream
[I-D.lennox-payload-ulp-ssrc-mux].
6. RTP Session Specification
This section defines what needs to be done or avoided to make an RTP
session with multiple media types function without issues.
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6.1. RTP Session
Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications"
[RFC3550] states:
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.
This specification changes both of these sentences. The first
sentence is changed to:
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, unless
specification [RFCXXXX] is followed and the application meets the
applicability constraints.
The second sentence is changed to:
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields, unless multiplexed based on both SSRC and payload type and
usage meets what Multiple Media Types in an RTP Session [RFCXXXX]
specifies.
Second paragraph of Section 6 in RTP Profile for Audio and Video
Conferences with Minimal Control [RFC3551] says:
The payload types currently defined in this profile are assigned
to exactly one of three categories or media types: audio only,
video only and those combining audio and video. The media types
are marked in Tables 4 and 5 as "A", "V" and "AV", respectively.
Payload types of different media types SHALL NOT be interleaved or
multiplexed within a single RTP session, but multiple RTP sessions
MAY be used in parallel to send multiple media types. An RTP
source MAY change payload types within the same media type during
a session. See the section "Multiplexing RTP Sessions" of RFC
3550 for additional explanation.
This specifications purpose is to violate that existing SHALL NOT
under certain conditions. Thus also this sentence has to be changed
to allow for multiple media type's payload types in the same session.
The above sentence is changed to:
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Payload types of different media types SHALL NOT be interleaved or
multiplexed within a single RTP session unless as specified and
under the restriction in Multiple Media Types in an RTP Session
[RFCXXXX]. Multiple RTP sessions MAY be used in parallel to send
multiple media types.
RFC-Editor Note: Please replace RFCXXXX with the RFC number of this
specification when assigned.
We can now go on and discuss the five bullets that are motivating the
previous in Section 5.2 of the RTP Specification [RFC3550]. They are
repeated here for the reader's convenience:
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
require different sequence number spaces to tell which payload
type suffered packet loss.
3. The RTCP sender and receiver reports (see Section 6.4 of RFC
3550) 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.
Bullets 1 to 3 are all related to that each media source has to use
one or more unique SSRCs to avoid these issues as mandated below
(Section 6.2). Bullet 4 can be served by two arguments, first of all
each SSRC will be associated with a specific media type, communicated
through the RTP payload type, allowing a middlebox to do media type
specific operations. The second argument is that in many contexts
blind combining without additional contexts are anyway not suitable.
Regarding bullet 5 this is a understood and explicitly stated
applicability limitations for the method described in this document.
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6.2. Sender Source Restrictions
A SSRC in the RTP session MUST only send one media type (audio,
video, text etc.) during the SSRC's lifetime. The main motivation is
that a given SSRC has its own RTP timestamp and sequence number
spaces. The same way that you can't send two streams of encoded
audio on the same SSRC, you can't send one audio and one video
encoding on the same SSRC. Each media encoding when made into an RTP
stream needs to have the sole control over the sequence number and
timestamp space. If not, one would not be able to detect packet loss
for that particular stream. Nor can one easily determine which clock
rate a particular SSRCs timestamp will increase with. For additional
arguments why RTP payload type based multiplexing of multiple media
streams doesn't work see Appendix A in
[I-D.westerlund-avtcore-multiplex-architecture].
6.3. Payload Type Applicability
Most Payload Types have a native media type, like an audio codec is
natural belonging to the audio media type. However, there exist a
number of RTP payload types that don't have a native media type. For
example, transport robustness mechanisms like RTP Retransmission
[RFC4588] and Generic FEC [RFC5109] inherit their media type from
what they protect. RTP Retransmission is explicitly bound to the
payload type it is protecting, and thus will inherit it. However
Generic FEC is a excellent example of an RTP payload type that has no
natural media type. The media type for what it protects is not
relevant as it is the recovered RTP packets that have a particular
media type, and thus Generic FEC is best categorized as an
application media type.
The above discussion is relevant to what limitations exist for RTP
payload type usage within an RTP session that has multiple media
types. In fact this document (Section 7.2) suggest that for usage of
Generic FEC (XOR-based) as defined in RFC 5109 can actually use a
single media type when used with independent RTP sessions for source
and repair data.
Note a particular SSRC carrying Generic FEC will clearly only
protect a specific SSRC and thus that instance is bound to the
SSRC's media type. For this specific case, it is possible to have
one be applicable to both. However, in cases when the signalling
is setup to enable fall back to using separate RTP sessions, then
using a different media type, e.g. application, than the media
being protected can create issues.
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6.4. RTCP
An RTP session has a single set of parameters that configure the
session bandwidth, the RTCP sender and receiver fractions (e.g., via
the SDP "b=RR:" and "b=RS: lines), and the parameters of the RTP/AVPF
profile [RFC4585] (e.g., trr-int) if that profile (or its secure
extension, RTP/SAVPF [RFC5124]) is used. As a consequence, the RTCP
reporting interval will be the same for every SSRC in an RTP session.
This uniform RTCP reporting interval can result in RTCP reports being
sent more often than is considered desirable for a particular media
type. For example, if an audio flow is multiplexed with a high
quality video flow where the session bandwidth is configured to match
the video bandwidth, this can result in the RTCP packets having a
greater bandwidth allocation than the audio data rate. If the
reduced minimum RTCP interval described in Section 6.2 of [RFC3550]
is used in the session, which might be appropriate for video where
rapid feedback is wanted, the audio sources could be expected to send
RTCP packets more often than they send audio data packets. This is
most likely undesirable, and while the mismatch can be reduced
through careful tuning of the RTCP parameters, particularly trr_int
in RTP/AVPF sessions, it is inherent in the design of the RTCP timing
rules, and affects all RTP sessions containing flows with mismatched
bandwidth.
Having multiple media types in one RTP session also results in more
SSRCs being present in this RTP session. This increasing the amount
of cross reporting between the SSRCs. From an RTCP perspective, two
RTP sessions with half the number of SSRCs in each will be slightly
more efficient. If someone needs either the higher efficiency due to
the lesser number of SSRCs or the fact that one can't tailor RTCP
usage per media type, they need to use independent RTP sessions.
When it comes to handling multiple SSRCs in an RTP session there is a
clarification under discussion in Real-Time Transport Protocol (RTP)
Considerations for Multi-Stream Endpoints
[I-D.lennox-avtcore-rtp-multi-stream]. When it comes to configuring
RTCP the need for regular periodic reporting needs to be weighted
against any feedback or control messages being sent. The
applications using RTP/AVPF or RTP/SAVPF are RECOMMENDED to consider
setting trr-int parameter to a value suitable for the applications
needs, thus potentially reducing the need for regular reporting and
thus releasing more bandwidth for use for feedback or control.
Another aspect of an RTP session with multiple media types is that
the used RTCP packets, RTCP Feedback Messages, or RTCP XR metrics
used might not be applicable to all media types. Instead all RTP/
RTCP endpoints need to correlate the media type of the SSRC being
referenced in an messages/packet and only use those that apply to
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that particular SSRC and its media type. Signalling solutions might
have shortcomings when it comes to indicate that a particular set of
RTCP reports or feedback messages only apply to a particular media
type within an RTP session.
6.4.1. Timing out SSRCs
All used SSRCs in the RTP session MUST use the same timeout behaviour
to avoid premature timeouts. This will depend on the RTP profile and
its configuration. The RTP specification provides several options
that can influence the values used when calculating the time-
interval, to avoid such issues when using this specification we make
clarification on the calculations.
For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval = 0
the timeout interval SHALL be calculated using a multiplier of 5,
i.e. the timeout interval becomes 5*Td. The Td calculation SHALL be
done using a Tmin value of 5 seconds, not the reduced minimal
interval even if used to calculate RTCP packet transmission
intervals. If using either the RTP/AVPF or RTP/SAVPF profiles with
T_rr_interval != 0 then the calculation as specified in Section 3.5.4
of RFC 4585 SHALL be used with a multiplier of 5, i.e. Tmin in the
Td calculation is the T_rr_interval.
Note: If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or
their secure variants) are combined in a single RTP session, and the
RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly
lower than 5 seconds, then there is a risk that the RTP/AVP endpoints
will prematurely timeout the RTP/AVPF endpoints due to their
different RTCP timeout intervals. Since an RTP session can only use
a single RTP profile, this issue ought never occur. If such mixed
RTP profiles are used, however, the RTP/AVPF session MUST NOT use a
non-zero T_rr_interval that is smaller than 5 seconds.
(tbd: it has been suggested that a minimum non-zero T_rr_interval of
4 seconds is more appropriate, due to the nature of the timing
rules).
6.4.2. Tuning RTCP transmissions
This sub-section discusses what tuning can be done to reduce
downsides of the shared RTCP packet intervals.
When using the RTP/AVP or RTP/SAVP profile the tuning one can do is
very limited. The controls one has are very limited to the RTCP
bandwidth values and if one scales the minimum RTCP interval
according to the bandwidth. As the scheduling algorithm includes
both random factors and reconsideration, one can't simply calculate
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the expected average transmission interval using formula for Td. But
it does indicate the important factors affecting the transmission
interval, namely the RTCP bandwidth available for the role (Active
Sender or Participant), the average RTCP packet size and the number
of SSRCs classified in the relevant role. Note, that if the ratio of
senders to total number of session participants are larger than the
ratio of RTCP bandwidth for senders in relation to the total RTCP
bandwidth, then senders and receivers are treated together.
Lets start with some basic observations:
a. Unless scaled minimum RTCP interval is used, then Td prior to
randomization and reconsideration can never be less than 5
seconds (assuming default Tmin of 5 seconds).
b. If scaled minimum RTCP interval is used Td can become as low as
360 divided by RTP Session bandwidth in kilobits. In SDP the RTP
session bandwidth is signalled using b=AS. A RTP Session
bandwidth of 72 kbps results in Tmin being 5 seconds. A RTP
session bandwidth of 360 kbps of course gives a Tmin of 1 second,
and to achieve a Tmin equal to once every frame for a 25 Hz video
stream requires an RTP session bandwidth of 9 Mbps! (The use of
the RTP/AVPF or RTP/SAVPF profile allows smaller Tmin, and hence
more frequent RTCP report, as discussed below).
c. Lets calculate the number (n) of SSRCs in the RTP session that 5%
of the session bandwidth can support to yield a Td value equal to
Tmin with minimal scaling. For this calculation we have to make
two assumptions. The first is that we will consider most or all
SSRC being senders resulting in everyone sharing the available
bandwidth. Secondly we will select an average RTCP packet size.
This packet will consist of an SR, containing (n-1) report blocks
up to 31 report blocks, a SDES item with at least a CNAME (17
bytes value) in it. Such a basic packet will be 800 bytes for
n>=32. With these parameters, and as the bandwidth goes up the
time interval is proportionally decreased (due to minimal
scaling), thus all the example bandwidths 72 kbps, 360 kbps and 9
Mbps all support 9 SSRCs.
d. The actual transmission interval for a Td value is [0.5*Td/
1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the
interval is actually [2.052,6.156] and the distribution is not
uniform, it is an exponential increasing one. The probability
for sending at time X, given it is within the interval, is
probability of picking X in the interval times the probability to
randomly picking a number that is <=X within the interval with an
uniform probability distribution. This results in that the
majority of the probability mass is above the Td value.
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To conclude, with RTP/AVP and RTP/SAVP the key limitation for small
unicast sessions are going to be the Tmin value. Thus the RTP
session bandwidth configured in RTCP has to be sufficient high to
reach the reporting goals the application has following the rules for
scaled minimal RTCP interval.
When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional
tool, the setting of the T_rr_interval which has several effects on
the RTCP reporting. First of all as Tmin is set to 0 after the
initial transmission and regular reporting interval is instead
affected of the regular bandwidth based calculation and the
T_rr_interval. This has the affect that we are no longer restricted
by the minimal interval or even the scaling rule for the minimal
rule. Instead the RTCP bandwidth and the T_rr_interval is the
governing factors. Now it also becomes important to separate between
the applications need for regular reports and RTCP feedback packet
types. In both regular RTCP mode, as in Early RTCP Mode, the usage
of the T_rr_Interval prevents regular RTCP packets, i.e. packets
without any Feedback packets to be sent more often than
T_rr_interval. This value is a hard as no regular RTCP packet can be
sent less than T_rr_interval after the previous regular packet
packet.
So for applications that has a use for feedback packets for some
media streams, for example video packets but don't want to frequent
regular reporting for audio could configure the T_rr_interval to a
value so that the regular reporting for both audio and video is at a
level that is considered acceptable for the audio. Then use feedback
packets, which will include RTCP SR/RR packets, unless reduced-size
RTCP feedback packets [RFC5506] are used, and can include other
report information in addition to the feedback packet that needs to
be sent. That way the available RTCP bandwidth can be focused for
use, which provides the most utility for the application.
Using T_rr_interval still requires one to determine suitable values
for the RTCP bandwidth value, in fact it might make it even more
important, as one is more likely to affect the RTCP behaviour and
performance, than when using RTP/AVP, as their is fewer limitations
affecting the RTCP transmission.
When using T_rr_interval, i.e. having it be non zero, there are
configurations that have to be avoided. If the resulting Td value is
smaller but close to T_rr_interval then the interval in which the
actual regular RTCP packet transmission falls into becomes very
large, from 0.5 times T_rr_interval up to 2.73 times the
T_rr_interval. Therefore for configuration where one intends to have
Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted
at values less than 1/4th of T_rr_interval which results in that the
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range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].
With RTP/AVPF using T_rr_interval of 0 or with another low value,
which will be significantly lower than Td still has its utility and
different behaviour compared to RTP/AVP. This avoids the Tmin
limitations of RTP/AVP, thus allowing more frequent regular RTCP
reporting. In fact this will result that the RTCP traffic becomes as
high as the configured values.
(tbd: a future version of this memo will include examples of how to
choose RTCP parameters for common scenarios)
There exist no method within the specification for using different
regular RTCP reporting interval depending on media type or individual
media stream.
7. Extension Considerations
This section discusses the impact on some RTP/RTCP extensions due to
usage of multiple media types in on RTP session. Only extensions
where something worth noting has been included.
7.1. RTP Retransmission
SSRC-multiplexed RTP retransmission [RFC4588] is actually very
straightforward. Each retransmission RTP payload type is explicitly
connected to an associated payload type. If retransmission is only
to be used with a subset of all payload types, this is not a problem,
as it will be evident from the retransmission payload types which
payload types that have retransmission enabled for them.
Session-multiplexed RTP retransmission is also possible to use where
an retransmission session contains the retransmissions of the
associated payload types in the source RTP session. The only
difference to previously is that the source RTP session is one which
contains multiple media types. Thus it is even more likely that only
a subset of the source RTP session's payload types and SSRCs are
actually retransmitted.
Open Issue: When using SDP to signal retransmission for one RTP
session with multiple media types and one RTP session for the
retransmission data will cause a situation where one will have
multiple m= lines grouped using FID and the ones belonging to
respective RTP session being grouped using BUNDLE. This usage might
contradict both the FID semantics [RFC5888] and an assumption in the
RTP retransmission specification [RFC4588].
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7.2. Generic FEC
The RTP Payload Format for Generic Forward Error Correction
[RFC5109], and also its predecessor [RFC2733], requires some
considerations, and they are different depending on what type of
configuration of usage one has.
Independent RTP Sessions, i.e. where source and repair data are sent
in different RTP sessions. As this mode of configuration requires
different RTP session, there has to be at least one RTP session for
source data, this session can be one using multiple media types. The
repair session only needs one RTP Payload type indicating repair
data, i.e. x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733
is used. The media type in this session is not relevant and can in
theory be any of the defined ones. It is RECOMMENDED that one uses
"Application".
In stream, using RTP Payload for Redundant Audio Data [RFC2198]
combining repair and source data in the same packets. This is
possible to use within a single RTP session. However, the usage and
configuration of the payload types can create an issue. First of all
it might be necessary to have one payload type per media type for the
FEC repair data payload format, i.e. one for audio/ulpfec and one for
text/ulpfec if audio and text are combined in an RTP session.
Secondly each combination of source payload and its FEC repair data
has to be an explicit configured payload type. This has potential
for making the limitation of RTP payload types available into a real
issue.
8. Signalling
The Signalling requirements
Establishing an RTP session with multiple media types requires
signalling. This signalling needs to fulfil the following
requirements:
1. Ensure that any participant in the RTP session is aware that this
is an RTP session with multiple media types.
2. Ensure that the payload types in use in the RTP session are using
unique values, with no overlap between the media types.
3. Configure the RTP session level parameters, such as RTCP RR and
RS bandwidth, AVPF trr-int, underlying transport, the RTCP
extensions in use, and security parameters, commonly for the RTP
session.
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4. RTP and RTCP functions that can be bound to a particular media
type SHOULD be reused when possible also for other media types,
instead of having to be configured for multiple code-points.
Note: In some cases one will not have a choice but to use
multiple configurations.
8.1. SDP-Based Signalling
The signalling of multiple media types in one RTP session in SDP is
specified in "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers"
[I-D.ietf-mmusic-sdp-bundle-negotiation].
9. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as
an RFC.
10. Security Considerations
Having an RTP session with multiple media types doesn't change the
methods for securing a particular RTP session. One possible
difference is that the different media have often had different
security requirements. When combining multiple media types in one
session, their security requirements also have to be combined by
selecting the most demanding for each property. Thus having multiple
media types can result in increased overhead for security for some
media types to ensure that all requirements are meet.
Otherwise, the recommendations for how to configure and RTP session
do not add any additional requirements compared to normal RTP, except
for the need to be able to ensure that the participants are aware
that it is a multiple media type session. If not that is ensured it
can cause issues in the RTP session for both the unaware and the
aware one. Similar issues can also be produced in an normal RTP
session by creating configurations for different end-points that
doesn't match each other.
11. Acknowledgements
The authors would like to thank Christer Holmberg, Gunnar Hellstroem,
and Charles Eckel for the feedback on the document.
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12. References
12.1. Normative References
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-03 (work in
progress), February 2013.
[I-D.lennox-avtcore-rtp-multi-stream]
Lennox, J. and M. Westerlund, "Real-Time Transport
Protocol (RTP) Considerations for Endpoints Sending
Multiple Media Streams",
draft-lennox-avtcore-rtp-multi-stream-01 (work in
progress), October 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
12.2. Informative References
[I-D.lennox-payload-ulp-ssrc-mux]
Lennox, J., "Supporting Source-Multiplexing of the Real-
Time Transport Protocol (RTP) Payload for Generic Forward
Error Correction", draft-lennox-payload-ulp-ssrc-mux-00
(work in progress), February 2013.
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., Perkins, C., and H.
Alvestrand, "Guidelines for using the Multiplexing
Features of RTP",
draft-westerlund-avtcore-multiplex-architecture-02 (work
in progress), July 2012.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport",
draft-westerlund-avtcore-transport-multiplexing-04 (work
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in progress), October 2012.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", RFC 2733,
December 1999.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[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,
July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
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Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
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