One document matched: draft-ietf-avtcore-multi-media-rtp-session-07.xml
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<rfc category="std" docName="draft-ietf-avtcore-multi-media-rtp-session-07"
ipr="trust200902" updates="3550, 3551">
<front>
<title abbrev="Multiple Media Types in an RTP Session">Sending Multiple
Types of Media in a Single 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="9" month="March" year="2015"/>
<workgroup>AVTCORE WG</workgroup>
<abstract>
<t>This document specifies how an RTP session can contain RTP 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
different to those deployed at the time of this writing. The virtually
ubiquitous deployment of Network Address Translators (NAT) and Firewalls
has since increased the cost and likely-hood of communication failure
when using many different transport flows. Hence, there is pressure to
reduce the number of concurrent transport flows used by RTP
applications.</t>
<t>The RTP specification 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> is one of the dominant signalling
methods for establishing RTP sessions, and has enforced this rule by not
allowing multiple media types for a given destination or set of ICE
candidates.</t>
<t>The fact that these limitations have been in place for so long, in
addition to RFC 3550 being written without fully considering the use of
multiple media types in an RTP session, results in a number of issues
when allowing this behaviour. This memo updates <xref target="RFC3550"/>
and <xref target="RFC3551"/> with important considerations regarding
applicability and functionality when using multiple types of media in an
RTP session, including normative specification of behaviour. This memo
makes no changes to RTP behaviour when using multiple RTP streams with
media of the same type (e.g., multiple audio streams or multiple video
streams) in a single RTP session. Instead it relies on the
clarifications in <xref
target="I-D.ietf-avtcore-rtp-multi-stream"/>.</t>
<t>This memo is structured as follows. 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 are 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
memo ends with the security considerations.</t>
</section>
<section title="Definitions">
<t><list style="hanging">
<t hangText="Media Type:">The general type of media data used by a
real-time application. The media type corresponds to the value used
in the <media> field of an SDP m= line. The media types
defined at the time of this writing are "audio", "video", "text",
"application", and "message".</t>
<t hangText="Quality of Service (QoS):">Network mechanisms that are
intended to ensure that the packets within a flow or with a specific
marking are transported with certain properties.</t>
</list></t>
<t>The terms Encoded Stream, Endpoint, Media Source, RTP Session, and
RTP Stream are used as defined in <xref
target="I-D.ietf-avtext-rtp-grouping-taxonomy"/>.</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>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>
<t>Furthermore, we note that many RTP-using applications don't utilize
any network level Quality of Service (QoS) 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>
<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 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 RTP 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.ietf-avtcore-multiplex-guidelines">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.</t>
<t><list style="empty">
<t>There is some work in progress <xref
target="I-D.westerlund-avtcore-transport-multiplexing"/> that
attempt to address a solution for RTP-using applications that need
or would prefer multiple RTP sessions, but do not require the
functionalities or behaviours that multiple transport flows
give.</t>
</list></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 RTCP requirements. 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 RTCP requirements, for example due to RTP Streams bandwidth
consumption and packet rate, 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 Relay (Transport Translator)
topologies as defined in <xref
target="I-D.ietf-avtcore-rtp-topologies-update">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 might also suffer issues due to differencies in
configured RTCP bandwidth and other parameters 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 RTP stream or media source 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 RTP streams with different
media types is desired, using this specification can cause severe
limitations. All RTP 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>Marking-based QoS schemes like DiffServ can be affected if a
network ingress is the one that performs, markings based on flows.
Endpoint marking where the network API supports marking on individual
packet level will be unaffected by this specification. However, there
exist limitations, as discussed in <xref
target="I-D.ietf-dart-dscp-rtp"/>, on how different traffic classes
can be applied on different packets or RTP streams within a single
transport flow.</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 RTP 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 RTP streams within the same RTP Session as the source
stream <xref target="I-D.lennox-payload-ulp-ssrc-mux"/>.</t>
<!--MW: What is status on the lennox draft?-->
</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 media sources 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 encoded streams of audio
on the same SSRC, you can't send one encoded audio and one encoded
video stream on the same SSRC. Each encoded stream 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 encoded 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 sources doesn't work see <xref
target="I-D.ietf-avtcore-multiplex-guidelines"/>.</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 Considerations">
<t>Guidelines for handling RTCP when sending multiple RTP streams with
disparate rates in a single RTP session are outlined in <xref
target="I-D.ietf-avtcore-rtp-multi-stream"/>. These guidelines apply
when sending multiple types of media in a single RTP session if the
different types of media have different rates.</t>
</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 anchor="sec-rtx" 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 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 the previous case is if the source RTP session is one
which contains multiple media types. This results in the
retransmission streams in the RTP session for the retransmission
having multiple associated media types.</t>
<t>When using SDP signalling for a multiple media type RTP session,
i.e. <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref>, the
session multiplexed case do require some recommendations on how to
signal this. To avoid breaking the semantics of the <xref
target="RFC5888">FID grouping as defined by </xref> each media line
can only be included in one FID group. FID is used by RTP
retransmission to indicate the SDP media lines that is a source and
retransmission pair. Thus, for SDP using BUNDLE, each original media
source (m= line) that is retransmitted needs a corresponding media
line in the retransmission RTP session. In case there are multiple
media lines for retransmission, these media lines will form a
independent BUNDLE group from the BUNDLE group with the source
streams.</t>
<t>Below is an <xref target="fig-rtx-session">SDP example</xref> which
shows the grouping structures. This example is not legal SDP and only
the most important attributes has been left in place. Note that this
SDP is not an initial BUNDLE offer. As can be seen there are two
bundle groups, one for the source RTP session and one for the
retransmissions. Then each of the media sources are grouped with its
retransmission flow using FID, resulting in three more groupings.</t>
<figure anchor="fig-rtx-session"
title="SDP example of Session Multiplexed RTP Retransmission">
<artwork><![CDATA[ a=group:BUNDLE foo bar fiz
a=group:BUNDLE zoo kelp glo
a=group:FID foo zoo
a=group:FID bar kelp
a=group:FID fiz glo
m=audio 10000 RTP/AVP 0
a=mid:foo
a=rtpmap:0 PCMU/8000
m=video 10000 RTP/AVP 31
a=mid:bar
a=rtpmap:31 H261/90000
m=video 10000 RTP/AVP 31
a=mid:fiz
a=rtpmap:31 H261/90000
m=audio 40000 RTP/AVPF 99
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=0;rtx-time=3000
a=mid:zoo
m=video 40000 RTP/AVPF 100
a=rtpmap:100 rtx/90000
a=fmtp:199 apt=31;rtx-time=3000
a=mid:kelp
m=video 40000 RTP/AVPF 100
a=rtpmap:100 rtx/90000
a=fmtp:199 apt=31;rtx-time=3000
a=mid:glo
]]></artwork>
</figure>
</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>If one uses SDP signalling with <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref>, then
the RTP session carrying the FEC streams will be its own BUNDLE group.
The media line with the source stream for the FEC and the FEC stream's
media line will be grouped using media line grouping using the FEC or
<xref target="RFC5956">FEC-FR</xref> grouping. This is very similar to
the situation that arise for RTP retransmission with session
multiplexing discussed above in<xref target="sec-rtx"/>.</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.ietf-avtcore-rtp-multi-stream'?>
</references>
<references title="Informative References">
<?rfc include='reference.I-D.ietf-avtcore-multiplex-guidelines'?>
<?rfc include='reference.I-D.ietf-avtext-rtp-grouping-taxonomy'?>
<?rfc include='reference.I-D.ietf-dart-dscp-rtp'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-topologies-update'?>
<?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.4588'?>
<?rfc include='reference.RFC.5109'?>
<?rfc include='reference.RFC.5576'?>
<?rfc include='reference.RFC.5761'?>
<?rfc include='reference.RFC.5888'?>
<?rfc include='reference.RFC.5956'?>
</references>
</back>
</rfc>
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