One document matched: draft-ietf-avtcore-multi-media-rtp-session-02.xml
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<rfc category="std" docName="draft-ietf-avtcore-multi-media-rtp-session-02"
ipr="trust200902" updates="3550, 3551">
<front>
<title abbrev="Multiple Media Types in an RTP Session">Multiple Media
Types in an RTP Session</title>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Colin Perkins" initials="C. " surname="Perkins">
<organization>University of Glasgow</organization>
<address>
<postal>
<street>School of Computing Science</street>
<city>Glasgow</city>
<code>G12 8QQ</code>
<country>United Kingdom</country>
</postal>
<email>csp@csperkins.org</email>
</address>
</author>
<author fullname="Jonathan Lennox" initials="J." surname="Lennox">
<organization abbrev="Vidyo">Vidyo, Inc.</organization>
<address>
<postal>
<street>433 Hackensack Avenue</street>
<street>Seventh Floor</street>
<city>Hackensack</city>
<region>NJ</region>
<code>07601</code>
<country>US</country>
</postal>
<email>jonathan@vidyo.com</email>
</address>
</author>
<date day="25" month="February" year="2013"/>
<workgroup>AVTCORE WG</workgroup>
<abstract>
<t>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.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>When the <xref target="RFC3550">Real-time Transport Protocol
(RTP)</xref> 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.</t>
<t><xref target="RFC3550">RTP</xref> recommends against sending several
different types of media, for example audio and video, in a single RTP
session. The <xref target="RFC3551">RTP profile for Audio and Video
Conferences with Minimal Control (RTP/AVP)</xref> 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 <xref target="RFC4566">Session
Description Protocol (SDP)</xref>, 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.</t>
<t>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.</t>
<t>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 <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Bundle</xref>. The
document ends with the security considerations.</t>
</section>
<section title="Definitions">
<t>The following terms are used with supplied definitions:<list
style="hanging">
<t hangText="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.</t>
<t hangText="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.</t>
<t hangText="Media Type:">Audio, video, text or application whose
form and meaning are defined by a specific real-time
application.</t>
<t hangText="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.</t>
<t hangText="RTP Session:">As defined by <xref target="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.</t>
</list></t>
<t>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 <xref
target="RFC2119"/>.</t>
</section>
<section title="Motivation ">
<t>This section discusses in more detail the main motivations why
allowing multiple media types in the same RTP session is suitable.</t>
<section title="NAT and Firewalls">
<t>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:<list style="numbers">
<t>Increased delay to perform the transport flow establishment</t>
<t>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 behaviours
usually occur.</t>
<t>More transport flows means a higher risk that some transport
flow fails to be established, thus preventing the application to
communicate.</t>
</list></t>
<t>Using fewer transport flows reduces the risk of communication
failure, improved establishment behaviour and less load on NAT and
Firewalls.</t>
</section>
<section title="No Transport Level QoS">
<t>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.</t>
</section>
<section title="Architectural Equality">
<t>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.</t>
</section>
</section>
<section title="Overview of Solution">
<t>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.</t>
<t>The solution is quite straightforward. The first step is to override
the SHOULD and SHOULD NOT language of the <xref target="RFC3550">RTP
specification</xref>. Similar change is needed to a sentence in Section
6 of <xref target="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.</t>
<t>Within an RTP session where multiple media types have been configured
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.</t>
<t>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.</t>
<t>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.</t>
</section>
<section title="Applicability">
<t>This specification has limited applicability, and anyone intending to
use it needs to ensure that their application and usage meets the below
criteria.</t>
<section title="Usage of the RTP session">
<t>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 <xref
target="I-D.westerlund-avtcore-multiplex-architecture">Guidelines for
using the Multiplexing Features of RTP</xref> 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
<xref target="I-D.westerlund-avtcore-transport-multiplexing">Multiple
RTP Sessions on a Single Lower-Layer Transport</xref>. 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 only some
media streams. </t>
<t>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 <xref
target="sec.rtcp"/>.</t>
</section>
<section title="Signalled Support">
<t>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.</t>
<t>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.</t>
</section>
<section title="Homogeneous Multi-party">
<t>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 <xref target="RFC5117">RTP Topologies</xref>. 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.</t>
<t>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
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.</t>
<t>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.</t>
</section>
<section title="Reduced number of Payload Types">
<t>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 <xref
target="RFC5761">"Multiplexing RTP Data and Control Packets on a
Single Port"</xref> 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.</t>
</section>
<section title="Stream Differentiation">
<t>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. </t>
<t>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.</t>
</section>
<section title="Non-compatible Extensions">
<t>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.</t>
<t>Identified RTP/RTCP extensions that require multiple RTP Sessions
are:<list style="hanging">
<t hangText="RTP Retransmission:"><xref target="RFC4588">RTP
Retransmission</xref> 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.</t>
<t hangText="XOR-Based FEC:">The <xref target="RFC5109">RTP
Payload Format for Generic Forward Error Correction</xref> and its
predecessor <xref target="RFC2733"/> requires a separate RTP
session unless the FEC data is carried in <xref
target="RFC2198">RTP Payload for Redundant Audio Data</xref>.
However, using the Generic FEC with the Redundancy payload has
another set of restrictions, see <xref
target="sec-generic-fec"/>.</t>
<t hangText="">Note that the <xref
target="RFC5576">Source-Specific Media Attributes</xref>
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. <xref
target="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 <xref target="I-D.lennox-payload-ulp-ssrc-mux"/>. </t>
</list></t>
<t/>
</section>
</section>
<section title="RTP Session Specification">
<t>This section defines what needs to be done or avoided to make an RTP
session with multiple media types function without issues.</t>
<section title="RTP Session">
<t>Section 5.2 of <xref target="RFC3550">"RTP: A Transport Protocol
for Real-Time Applications"</xref> states:<list style="empty">
<t>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.</t>
<t>Separate audio and video streams SHOULD NOT be carried in a
single RTP session and demultiplexed based on the payload type or
SSRC fields.</t>
</list></t>
<t>This specification changes both of these sentences. The first
sentence is changed to:<list style="empty">
<t>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.</t>
</list></t>
<t>The second sentence is changed to:<list style="empty">
<t>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.</t>
</list></t>
<t>Second paragraph of Section 6 in <xref target="RFC3551">RTP Profile
for Audio and Video Conferences with Minimal Control</xref> says:</t>
<t><list style="empty">
<t>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.</t>
</list>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:<list style="empty">
<t>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.</t>
</list></t>
<t>RFC-Editor Note: Please replace RFCXXXX with the RFC number of this
specification when assigned.</t>
<t>We can now go on and discuss the five bullets that are motivating
the previous in Section 5.2 of the <xref target="RFC3550">RTP
Specification</xref>. They are repeated here for the reader's
convenience:<list style="numbers">
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>An RTP mixer would not be able to combine interleaved streams
of incompatible media into one stream.</t>
<t>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.</t>
</list></t>
<t>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 <xref
target="sec-source-restrcitctions">below</xref>. 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.</t>
</section>
<section anchor="sec-source-restrcitctions"
title="Sender Source Restrictions">
<t>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 <xref
target="I-D.westerlund-avtcore-multiplex-architecture"/>.</t>
</section>
<section title="Payload Type Applicability">
<t>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 <xref
target="RFC4588">RTP Retransmission</xref> and <xref
target="RFC5109">Generic FEC</xref> 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.</t>
<t>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 <xref target="sec-generic-fec">this document</xref>
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. <list style="hanging">
<t>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.</t>
</list></t>
</section>
<section anchor="sec.rtcp" title="RTCP ">
<t>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 <xref
target="RFC4585">RTP/AVPF profile</xref> (e.g., trr-int) if that
profile (or its <xref target="RFC5124"> secure extension,
RTP/SAVPF</xref>) 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 <xref target="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.</t>
<t>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.</t>
<t>When it comes to handling multiple SSRCs in an RTP session there is
a clarification under discussion in <xref
target="I-D.lennox-avtcore-rtp-multi-stream">Real-Time Transport
Protocol (RTP) Considerations for Multi-Stream Endpoints</xref>. 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.</t>
<t>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 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.</t>
<section title="Timing out SSRCs">
<t>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.
</t>
<t> 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.</t>
<t>
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. </t>
<t>
(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).
</t>
</section>
<section title="Tuning RTCP transmissions">
<t>This sub-section discusses what tuning can be done to reduce
downsides of the shared RTCP packet intervals.</t>
<t>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
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.</t>
<t>Lets start with some basic observations:<list style="letters">
<t>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).</t>
<t>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).</t>
<t>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. </t>
<t>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.</t>
</list></t>
<t>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. </t>
<t>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. </t>
<t>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 <xref
target="RFC5506">reduced-size RTCP feedback packets</xref> 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. </t>
<t>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. </t>
<t>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 range becomes [0.5*T_rr_interval,
1.81*T_rr_interval].</t>
<t>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.</t>
<t>
(tbd: a future version of this memo will include examples of how
to choose RTCP parameters for common scenarios)
</t>
<t>There exist no method within the specification for using
different regular RTCP reporting interval depending on media type or
individual media stream. </t>
</section>
</section>
</section>
<section title="Extension Considerations">
<t>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.</t>
<section title="RTP Retransmission">
<t>SSRC-multiplexed <xref target="RFC4588">RTP retransmission</xref>
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.</t>
<t>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.</t>
<t>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 <xref target="RFC5888">FID semantics</xref> and an
assumption in the <xref target="RFC4588">RTP retransmission
specification</xref>.</t>
</section>
<section anchor="sec-generic-fec" title="Generic FEC">
<t>The <xref target="RFC5109">RTP Payload Format for Generic Forward
Error Correction</xref>, and also its predecessor <xref
target="RFC2733"/>, requires some considerations, and they are
different depending on what type of configuration of usage one
has.</t>
<t>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".</t>
<t>In stream, using <xref target="RFC2198">RTP Payload for Redundant
Audio Data</xref> 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.</t>
</section>
</section>
<section title="Signalling">
<t>The Signalling requirements</t>
<t>Establishing an RTP session with multiple media types requires
signalling. This signalling needs to fulfil the following
requirements:<list style="numbers">
<t>Ensure that any participant in the RTP session is aware that this
is an RTP session with multiple media types.</t>
<t>Ensure that the payload types in use in the RTP session are using
unique values, with no overlap between the media types.</t>
<t>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.</t>
<t>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.</t>
</list></t>
<t/>
<section title="SDP-Based Signalling">
<t>The signalling of multiple media types in one RTP session in SDP is
specified in <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">"Multiplexing
Negotiation Using Session Description Protocol (SDP) Port
Numbers"</xref>.</t>
</section>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section is to be removed on publication as an
RFC.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>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.</t>
<t>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.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>The authors would like to thank Christer Holmberg, Gunnar
Hellström, and Charles Eckel for the feedback on the document.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.3551'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
<?rfc include='reference.I-D.lennox-avtcore-rtp-multi-stream'?>
</references>
<references title="Informative References">
<?rfc include='reference.I-D.westerlund-avtcore-multiplex-architecture'?>
<?rfc include='reference.I-D.westerlund-avtcore-transport-multiplexing'?>
<?rfc include='reference.I-D.lennox-payload-ulp-ssrc-mux'?>
<?rfc include='reference.RFC.2198'?>
<?rfc include='reference.RFC.2733'?>
<?rfc include='reference.RFC.4566'?>
<?rfc include='reference.RFC.4585'?>
<?rfc include='reference.RFC.4588'?>
<?rfc include='reference.RFC.5109'?>
<?rfc include='reference.RFC.5117'?>
<?rfc include='reference.RFC.5124'?>
<?rfc include='reference.RFC.5506'?>
<?rfc include='reference.RFC.5576'?>
<?rfc include='reference.RFC.5761'?>
<?rfc include='reference.RFC.5888'?>
</references>
</back>
</rfc>
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