One document matched: draft-ietf-mmusic-sdp-simulcast-03.xml
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<rfc category="std" docName="draft-ietf-mmusic-sdp-simulcast-03"
ipr="trust200902" submissionType="IETF">
<front>
<title abbrev="Simulcast">Using Simulcast in SDP and RTP Sessions</title>
<author fullname="Bo Burman" initials="B." surname="Burman">
<organization>Ericsson</organization>
<address>
<postal>
<street>Kistavagen 25</street>
<city>SE-164 80 Stockholm</city>
<region/>
<code/>
<country>Sweden</country>
</postal>
<phone/>
<facsimile/>
<email>bo.burman@ericsson.com</email>
<uri/>
</address>
</author>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 2</street>
<city>SE-164 80 Stockholm</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Suhas Nandakumar" initials="S." surname="Nandakumar">
<organization>Cisco</organization>
<address>
<postal>
<street>170 West Tasman Drive</street>
<city>San Jose</city>
<region>CA</region>
<code>95134</code>
<country>USA</country>
</postal>
<phone/>
<facsimile/>
<email>snandaku@cisco.com</email>
<uri/>
</address>
</author>
<author fullname="Mo Zanaty" initials="M." surname="Zanaty">
<organization>Cisco</organization>
<address>
<postal>
<street>170 West Tasman Drive</street>
<city>San Jose</city>
<region>CA</region>
<code>95134</code>
<country>USA</country>
</postal>
<phone/>
<facsimile/>
<email>mzanaty@cisco.com</email>
<uri/>
</address>
</author>
<date day="19" month="October" year="2015"/>
<abstract>
<t>In some application scenarios it may be desirable to send multiple
differently encoded versions of the same media source in different RTP
streams. This is called simulcast. This document discusses the best way
of accomplishing simulcast in RTP and how to signal it in SDP. A
solution is defined by making an extension to SDP, and using RTP/RTCP
identification methods to relate RTP streams belonging to the same media
source. The SDP extension consists of a new media level SDP attribute
that expresses capability to send and/or receive simulcast RTP streams.
RTP/RTCP identification using either payload types or a separately
defined method for RTP stream configuration are defined.</t>
</abstract>
</front>
<middle>
<section anchor="sec-intro" title="Introduction">
<t>Most of today's multiparty video conference solutions make use of
centralized servers to reduce the bandwidth and CPU consumption in the
endpoints. Those servers receive RTP streams from each participant and
send some suitable set of possibly modified RTP streams to the rest of
the participants, which usually have heterogeneous capabilities (screen
size, CPU, bandwidth, codec, etc). One of the biggest issues is how to
perform RTP stream adaptation to different participants' constraints
with the minimum possible impact on both video quality and server
performance.</t>
<t>Simulcast is defined in this memo as the act of simultaneously
sending multiple different encoded streams of the same media source,
e.g. the same video source encoded with different video encoder types or
image resolutions. This can be done in several ways and for different
purposes. This document focuses on the case where it is desirable to
provide a media source as multiple encoded streams over <xref
target="RFC3550">RTP</xref> towards an intermediary so that the
intermediary can provide the wanted functionality by selecting which RTP
stream(s) to forward to other participants in the session, and more
specifically how the identification and grouping of the involved RTP
streams are done. From an RTP perspective, simulcast is a specific
application of the aspects discussed in <xref
target="I-D.ietf-avtcore-multiplex-guidelines">RTP Multiplexing
Guidelines</xref>.<!--MW: What is referenced in the above sentence by "the aspects discussed"--></t>
<t>This document describes a few scenarios where it is motivated to use
simulcast, and also defines the needed SDP signaling for it.</t>
</section>
<section anchor="sec-definitions" title="Definitions">
<t/>
<section title="Terminology">
<t>This document makes use of the terminology defined in <xref
target="I-D.ietf-avtext-rtp-grouping-taxonomy">RTP Taxonomy</xref>,
<xref target="RFC5117">RTP Topology</xref> and <xref
target="I-D.ietf-avtcore-rtp-topologies-update">RTP Topologies
Update</xref>. In addition, the following terms are used:<list
style="hanging">
<t hangText="RTP Mixer:">An RTP middle node, defined in <xref
target="RFC5117"/> (Section 3.4: Topo-Mixer), further elaborated
and extended with other topologies in <xref
target="I-D.ietf-avtcore-rtp-topologies-update"/> (Section 3.6 to
3.9).</t>
<t hangText="RTP Switch:">A common short term for the terms
"switching RTP mixer", "source projecting middlebox", and "video
switching MCU" as discussed in <xref
target="I-D.ietf-avtcore-rtp-topologies-update"/>.</t>
<t hangText="Simulcast Stream:">One Encoded Stream or Dependent
Stream from a set of concurrently transmitted Encoded Streams and
optional Dependent Streams, all sharing a common Media Source, as
defined in <xref target="I-D.ietf-avtext-rtp-grouping-taxonomy"/>.
Decoding a Dependent Stream also requires the related (Dependent
and) Encoded Stream(s), but in the context of simulcast that is
considered a property of the Dependent Stream constituting the
simulcast stream. For example, HD and thumbnail video simulcast
versions of a single Media Source sent concurrently as separate
RTP Streams.</t>
<t hangText="Simulcast Format:">Different formats of a simulcast
stream serve the same purpose as alternative RTP payload types in
non-simulcast SDP, to allow multiple alternative media formats for
a given RTP Stream. As for multiple RTP payload types on the
m-line, any one of the alternative formats can be used at a given
point in time, but not more than one (based on RTP timestamp), and
what format is used can change dynamically from one RTP packet to
another. For example, if all participants in a group video call
can decode H.264 and H.265 video, but only some can encode H.265,
both H.264 and H.265 can be kept as alternative formats, and the
format may dynamically switch between H.264 and H.265 as different
participants become active speaker.</t>
</list></t>
<!--MW: In the above we should be able to remove RFC 5117 in the next version
as updated topologies are in AUTH48.-->
</section>
<section title="Requirements Language">
<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">RFC 2119</xref>.</t>
</section>
</section>
<section anchor="sec-use-cases" title="Use Cases">
<t>Many use cases of simulcast as described in this document relate to a
multi-party communication session where one or more central nodes are
used to adapt the view of the communication session towards individual
participants, and facilitate the media transport between participants.
Thus, these cases targets the RTP Mixer type of topology.</t>
<t>There are two principle approaches for an RTP Mixer to provide this
adapted view of the communication session to each receiving
participant:<list style="symbols">
<t>Transcoding (decoding and re-encoding) received RTP streams with
characteristics adapted to each receiving participant. This often
include mixing or composition of media sources from multiple
participants into a mixed media source originated by the RTP Mixer.
The main advantage of this approach is that it achieves close to
optimal adaptation to individual receiving participants. The main
disadvantages are that it can be very computationally expensive to
the RTP Mixer and typically also degrades media Quality of
Experience (QoE) such as end-to-end delay for the receiving
participants.</t>
<t>Switching a subset of all received RTP streams or sub-streams to
each receiving participant, where the used subset is typically
specific to each receiving participant. The main advantages of this
approach are that it is computationally cheap to the RTP Mixer and
it has very limited impact on media QoE. The main disadvantage is
that it can be difficult to combine a subset of received RTP streams
into a perfect fit to the resource situation of a receiving
participant.</t>
</list></t>
<t>The use of simulcast relates to the latter approach, where it is more
important to reduce the load on the RTP Mixer and/or minimize QoE impact
than to achieve an optimal adaptation of resource usage.</t>
<section anchor="sec-diverse-receivers"
title="Reaching a Diverse Set of Receivers">
<t>The media sources provided by a sending participant potentially
need to reach several receiving participants that differ in terms of
available resources. The receiver resources that typically differ
include, but are not limited to:<list style="hanging">
<t hangText="Codec:">This includes codec type (such as SDP MIME
type) and can include codec configuration options (e.g. SDP fmtp
parameters). A couple of codec resources that differ only in codec
configuration will be "different" if they are somehow not
"compatible", like if they differ in video codec profile, or the
transport packetization configuration.</t>
<t hangText="Sampling:">This relates to how the media source is
sampled, in spatial as well as in temporal domain. For video
streams, spatial sampling affects image resolution and temporal
sampling affects video frame rate. For audio, spatial sampling
relates to the number of audio channels and temporal sampling
affects audio bandwidth. This may be used to suit different
rendering capabilities or needs at the receiving endpoints, as
well as a method to achieve different transport capabilities,
bitrates and eventually QoE by controlling the amount of source
data.</t>
<t hangText="Bitrate:">This relates to the amount of bits spent
per second to transmit the media source as an RTP stream, which
typically also affects the Quality of Experience (QoE) for the
receiving user.</t>
</list>Letting the sending participant create a simulcast of a few
differently configured RTP streams per media source can be a good
tradeoff when using an RTP switch as middlebox, instead of sending a
single RTP stream and using an RTP mixer to create individual
transcodings to each receiving participant.</t>
<t>This requires that the receiving participants can be categorized in
terms of available resources and that the sending participant can
choose a matching configuration for a single RTP stream per category
and media source.</t>
<t>For example, assume for simplicity a set of receiving participants
that differ only in that some have support to receive Codec A, and the
others have support to receive Codec B. Further assume that the
sending participant can send both Codec A and B. It can then reach all
receivers by creating two simulcasted RTP streams from each media
source; one for Codec A and one for Codec B.</t>
<t>In another simple example, a set of receiving participants differ
only in screen resolution; some are able to display video with at most
360p resolution and some support 720p resolution. A sending
participant can then reach all receivers by creating a simulcast of
RTP streams with 360p and 720p resolution for each sent video media
source.</t>
<t>In more elaborate cases, the receiving participants differ both in
available sampling and bitrate, and maybe also codec, and it is up to
the RTP switch to find a good trade-off in which simulcasted stream to
choose for each intended receiver. It is also the responsibility of
the RTP switch to negotiate a good fit of simulcast streams with the
sending participant.</t>
<t>The maximum number of simulcasted RTP streams that can be sent is
mainly limited by the amount of processing and uplink network
resources available to the sending participant.</t>
</section>
<section anchor="sec-application-specific"
title="Application Specific Media Source Handling">
<t>The application logic that controls the communication session may
include special handling of some media sources. It is for example
commonly the case that the media from a sending participant is not
sent back to itself.</t>
<t>It is also common that a currently active speaker participant is
shown in larger size or higher quality than other participants (the
sampling or bitrate aspects of <xref
target="sec-diverse-receivers"/>). Not sending the active speaker
media back to itself means there is some other participant's media
that instead has to receive special handling towards the active
speaker; typically the previous active speaker. This way, the
previously active speaker is needed both in larger size (to current
active speaker) and in small size (to the rest of the participants),
which can be solved with a simulcast from the previously active
speaker to the RTP switch.</t>
</section>
<section anchor="sec-receiver-preferences"
title="Receiver Media Source Preferences">
<t>The application logic that controls the communication session may
allow receiving participants to apply preferences to the
characteristics of the RTP stream they receive, for example in terms
of the aspects listed in <xref target="sec-diverse-receivers"/>.
Sending a simulcast of RTP streams is one way of accommodating
receivers with conflicting or otherwise incompatible preferences.</t>
</section>
</section>
<section anchor="sec-requirements" title="Requirements">
<t>The following requirements need to be met to support the use cases in
previous sections:<list style="hanging">
<t anchor="req-1" hangText="REQ-1:">Identification. It must be
possible to identify a set of simulcasted RTP streams as originating
from the same media source:<list style="hanging">
<t anchor="req-1.1" hangText="REQ-1.1:">In SDP signaling.</t>
<t anchor="req-1.2" hangText="REQ-1.2:">On RTP/RTCP level.</t>
</list></t>
<t anchor="req-2" hangText="REQ-2:">Transport usage. The solution
must work when using:<list style="hanging">
<t anchor="req-2.1" hangText="REQ-2.1:">Legacy SDP with separate
media transports per SDP media description.</t>
<t anchor="req-2.2" hangText="REQ-2.2:"><xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Bundled</xref>
SDP media descriptions.</t>
</list></t>
<t anchor="req-3" hangText="REQ-3:">Capability negotiation. It must
be possible that:<list style="hanging">
<t anchor="req-3.1" hangText="REQ-3.1:">Sender can express
capability of sending simulcast.</t>
<t anchor="req-3.2" hangText="REQ-3.2:">Receiver can express
capability of receiving simulcast.</t>
<t anchor="req-3.3" hangText="REQ-3.3:">Sender can express
maximum number of simulcast streams that can be provided.</t>
<t anchor="req-3.4" hangText="REQ-3.4:">Receiver can express
maximum number of simulcast streams that can be received.</t>
<t anchor="req-3.5" hangText="REQ-3.5:">Sender can detail the
characteristics of the simulcast streams that can be
provided.</t>
<t anchor="req-3.6" hangText="REQ-3.6:">Receiver can detail the
characteristics of the simulcast streams that it prefers to
receive.</t>
</list></t>
<t anchor="req-4" hangText="REQ-4:">Distinguishing features. It must
be possible to have different simulcast streams use different codec
parameters, as can be expressed by SDP format values and RTP payload
types.</t>
<t anchor="req-5" hangText="REQ-5:">Compatibility. It must be
possible to use simulcast in combination with other RTP mechanisms
that generate additional RTP streams:<list style="hanging">
<t anchor="req-5.1" hangText="REQ-5.1:"><xref
target="RFC4588">RTP Retransmission</xref>.</t>
<t anchor="req-5.2" hangText="REQ-5.2:"><xref
target="RFC5109">RTP Forward Error Correction</xref>.</t>
<t anchor="req-5.3" hangText="REQ-5.3:">Related payload types
such as audio Comfort Noise and/or DTMF.</t>
</list></t>
<t anchor="req-6" hangText="REQ-6:">Interoperability. The solution
must be possible to use in:<list style="hanging">
<t anchor="req-6.1" hangText="REQ-6.1:">Interworking with
non-simulcast legacy clients using a single media source per
media type.</t>
<t anchor="req-6.2" hangText="REQ-6.2:">WebRTC "Unified Plan"
environment with a single media source per SDP media
description.</t>
</list></t>
</list></t>
</section>
<section anchor="sec-solution-overview" title="Overview">
<t>As an overview, the above requirements are met by signaling simulcast
capability and configurations in <xref target="RFC4566">SDP</xref>:<list
style="symbols">
<t>An offer or answer can contain a number of simulcast streams,
separate for send and receive directions.</t>
<t>An offer or answer can contain multiple, alternative simulcast
streams in the same fashion as multiple, alternative codecs can be
offered in a media description.</t>
<t>A single media source per SDP media description is assumed, which
is aligned with the concepts defined in <xref
target="I-D.ietf-avtext-rtp-grouping-taxonomy"/> and will
specifically work in a WebRTC context, both with and without <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref>
grouping.</t>
<t>The codec configuration for a simulcast stream can be expressed
in two alternative ways, with complementing drawbacks and
benefits:<list style="symbols">
<t>Through existing SDP formats (corresponding to RTP payload
types), enabling the use of simulcast with a minimum set of
additions to existing SDP specifications.</t>
<t>Through use of a separately specified <xref
target="I-D.pthatcher-mmusic-rid">RTP-level identification
mechanism</xref>, which complements and effectively extends the
available simulcast stream identification and configuration
possibilities provided by using SDP formats.</t>
</list></t>
<t>It is possible, but not required to use <xref
target="RFC5576">source-specific signaling</xref> with the proposed
solution.</t>
</list></t>
</section>
<section anchor="sec-solution" title="Detailed Description">
<t>This section further details the overview <xref
target="sec-solution-overview">above</xref>.</t>
<section anchor="sec-cap" title="Simulcast Capability">
<t>Simulcast capability is expressed as a new media level SDP
attribute, "a=simulcast". For each desired direction (send/recv), the
simulcast attribute defines a list of simulcast streams (separated by
semicolons), each of which is a list of simulcast formats (separated
by commas). The meaning of the attribute on SDP session level is
undefined and MUST NOT be used. The <xref target="RFC5234">ABNF</xref>
for this attribute is:</t>
<figure align="center" anchor="fig-abnf" title="ABNF for Simulcast">
<artwork align="left"><![CDATA[
sc-attr = "a=simulcast:" 1*2( WSP sc-str-list ) [WSP sc-pause-list]
sc-str-list = sc-dir WSP sc-id-type "=" sc-alt-list *( ";" sc-alt-list )
sc-pause-list = "paused=" sc-alt-list
sc-dir = "send" / "recv"
sc-id-type = "pt" / "rid" / token
sc-alt-list = sc-id *( "," sc-id )
sc-id = fmt / rid-identifier / token
; WSP defined in [RFC5234]
; fmt, token defined in [RFC4566]
; rid-identifier defined in [I-D.pthatcher-mmusic-rid]
]]></artwork>
</figure>
<t/>
<t>There are separate and independent sets of parameters for simulcast
in send and receive directions. When listing multiple directions, each
direction MUST NOT occur more than once on the same line.</t>
<t>Two simulcast stream identification methods are defined; "pt" using
RTP payload type (SDP format), and "rid" using an additional <xref
target="I-D.pthatcher-mmusic-rid">RTP-level identification
mechanism</xref>. Different identification methods MUST NOT be used
for different directions on a single "a=simulcast" line.
Implementations that support both identification methods MAY include
one "a=simulcast" line for each identification method for the same
"m="-line. Multiple "a=simulcast" lines with the same identification
method MUST NOT be used for a single "m="-line.</t>
<t>Attribute parameters are grouped by direction and consist of a
listing of simulcast stream identifications to be used. The number of
(non-alternative, see below) identifications in the list sets a limit
to the number of supported simulcast streams in that direction. The
order of the listed simulcast versions in the "send" direction
suggests a proposed order of preference, in decreasing order: the
stream listed first is the most preferred <xref
target="sec-diverse-receivers"/>, and subsequent streams have
progressively lower preference. The order of the listed simulcast
streams in the "recv" direction expresses a preference which simulcast
streams that are preferred, with the leftmost being most preferred.
This can be of importance if the number of actually sent simulcast
streams have to be reduced for some reason.</t>
<t>Formats that have explicit <xref
target="RFC5583">dependencies</xref> <xref
target="I-D.pthatcher-mmusic-rid"/> to other formats (even in the same
media description) MAY be listed as different simulcast streams.</t>
<t>Alternative simulcast formats MAY be specified as part of the
attribute parameters by expressing each simulcast stream as a
comma-separated list of alternative format identifiers. In this case,
there MUST NOT be any capability restriction in what alternative
formats can be used across different simulcast streams, like requiring
all simulcast streams to use the same codec format alternative. The
order of the format alternatives within a simulcast stream is
significant; the alternatives are listed from (left) most preferred to
(right) least preferred. For the use of simulcast, this overrides the
normal codec preference as expressed by format type ordering on the
"m="-line, using regular SDP rules. This is to enable a separation of
general codec preferences and simulcast stream configuration
preferences.</t>
<t>A simulcast stream can use a codec defined such that the same RTP
SSRC can change RTP payload type multiple times during a session,
possibly even on a per-packet basis. A typical example can be a speech
codec that makes use of <xref target="RFC3389">Comfort Noise</xref>
and/or <xref target="RFC4733">DTMF</xref> formats. In those cases,
such "related" formats MUST NOT be listed explicitly in the attribute
parameters, since they are not strictly simulcast streams of the media
source, but rather a specific way of generating the RTP stream of a
single simulcast stream with varying RTP payload type. Instead, only a
single simulcast stream identification MUST be used per simulcast
stream or alternative simulcast format (if there are such) in the SDP.
The used simulcast stream identification SHOULD be the codec format
most relevant to the media description, if possible to identify, for
example the audio codec rather than the DTMF. What codec format to
choose in the case of switching between multiple equally "important"
formats is left open, but it is assumed that in the presence of such
strong relation it does not matter which is chosen.</t>
<t>If <xref target="I-D.ietf-avtext-rtp-stream-pause">RTP stream
pause/resume</xref> is supported, the optional "paused=" parameter MAY
be used in conjunction with "rid" simulcast stream identification to
specify that a certain simulcast stream is initially paused already
from start of the RTP session. In this case, support for RTP stream
pause/resume MUST also be included under the same "m="-line listing
"a=simulcast". Initially paused simulcast streams MUST NOT be used
with "pt" identification. Initially paused simulcast streams are
resumed as described by the RTP pause/resume specification.</t>
<t>An initially paused simulcast stream in "send" direction MUST be
considered equivalent to an unsolicited locally paused stream, and be
handled accordingly.</t>
<t>An initially paused simulcast stream in "recv" direction SHOULD
cause the remote RTP sender to put the stream as unsolicited locally
paused, unless there are other RTP stream receivers that do not mark
the simulcast stream as initially paused. The reason to require an
initially paused "recv" stream to be considered locally paused by the
remote RTP sender, instead of making it equivalent to implicitly
sending a pause request, is because the pausing RTP sender cannot know
which SSRC owns the restriction when TMMBR/TMMBN are used for
pause/resume signaling since the RTP receiver's SSRC in send direction
is not known yet.</t>
<t>Use of the <xref target="RFC2198">redundant audio data</xref>
format could be seen as a form of simulcast for loss protection
purposes, but is not considered conflicting with the mechanisms
described in this memo and MAY therefore be used as any other format.
In this case the "red" format, rather than the carried formats, SHOULD
be the one to list as a simulcast stream on the "a=simulcast"
line.</t>
<section anchor="sec-declarative" title="Declarative Use">
<t>When used as a declarative media description, a=simulcast "recv"
direction formats indicates the configured end point's required
capability to recognize and receive a specified set of RTP streams
as simulcast streams. In the same fashion, a=simulcast "send"
direction requests the end point to send a specified set of RTP
streams as simulcast streams.</t>
<t>If multiple simulcast formats are listed, it means that the
configured end point MUST be prepared to receive any of the "recv"
formats, and MAY send any of the "send" formats for that simulcast
stream.</t>
<t><list style="empty">
<t>Editor's note: The RID identification mechanism currently
lacks a declarative use definition. As declarative use may also
not follow unified plan with a single media source per
'"m="-line, it is uncertain if declarative can be defined for
the mechanism in its current shape.</t>
</list></t>
</section>
<section anchor="sec-offer-answer" title="Offer/Answer Use">
<t>An offerer wanting to use simulcast SHALL include the
"a=simulcast" attribute in the offer. An offerer that receives an
answer without "a=simulcast" MUST NOT use simulcast towards the
answerer. An offerer that receives an answer with "a=simulcast" not
listing a direction or without any simulcast stream identifications
in a specified direction MUST NOT use simulcast in that
direction.</t>
<t>An answerer that does not understand the concept of simulcast
will also not know the attribute and will remove it in the SDP
answer, as defined in existing <xref target="RFC3264">SDP
Offer/Answer</xref> procedures.</t>
<t>An answerer that does understand the attribute and that wants to
support simulcast in an indicated direction SHALL reverse
directionality of the unidirectional direction parameters; "send"
becomes "recv" and vice versa, and include it in the answer. Note
that, like all other use of SDP format tags ("pt:") for the send
direction in Offer/Answer, format tags related to the simulcast
stream identification send direction in an offer are placeholders
that refer to information in the offer SDP, and the actual formats
that will be used on the wire (including RTP Payload Format numbers)
depends on information included in the SDP answer.</t>
<t>An offerer listing a set of receive simulcast streams and/or
alternative formats in the offer MUST be prepared to receive RTP
streams for any of those simulcast streams and/or alternative
formats from the answerer.</t>
<t>An answerer that receives an offer with simulcast containing an
"a=simulcast" attribute listing alternative formats for simulcast
streams MAY keep all the alternatives in the answer, but it MAY also
choose to remove any non-desirable alternatives per simulcast stream
in the answer. The answerer MUST NOT add any alternatives that were
not present in the offer.</t>
<t>An answerer that receives an offer with simulcast that lists a
number of simulcast streams, MAY reduce the number of simulcast
streams in the answer, but MUST NOT add simulcast streams.</t>
<t>An offerer that receives an answer where some simulcast formats
are kept MUST be prepared to receive any of the kept send direction
alternatives, and MAY send any of the kept receive direction
alternatives from the answer. Similarly, the answerer MUST be
prepared to receive any of the kept receive direction alternatives,
and MAY send any of the kept send direction alternatives in the
answer.</t>
<t>The offerer and answerer MUST NOT send more than a single
alternative format at a time (based on RTP timestamps) per simulcast
stream, but MAY change format on a per-RTP packet basis. This
corresponds to the existing (non-simulcast) SDP offer/answer case
when multiple formats are included on the "m="-line in the SDP
answer.</t>
<t>An offerer that receives an answer where some of the simulcast
streams are removed MAY release the corresponding resources (codec,
transport, etc) in its receive direction and MUST NOT send any RTP
streams corresponding to the removed simulcast streams.</t>
<t>Simulcast streams or formats using undefined simulcast stream
identifications MUST NOT be used as valid simulcast streams by an
RTP stream receiver.</t>
<t>An offerer that is capable of using both simulcast stream
identification methods MAY include one "a=simulcast" line per
identification method in the offer. Note that it is in general not
expected that the "pt" identification method will provide feature
parity with the "rid" method, and the different "a=simulcast" lines
can therefore express different use of simulcast functionality.
However, for some configurations the different identification
methods can be equivalent.</t>
<t>An answerer receiving an offer listing both simulcast stream
identification methods MUST choose only one and remove the other
from the answer. An answerer not supporting a simulcast stream
identification method in the offer MUST remove the non-supported
"a=simulcast" line from the answer, possibly falling back to not
using simulcast at all.</t>
<t>The media formats and corresponding characteristics of encoded
streams used in a simulcast SHOULD be chosen such that they are
different. If this difference is not required, <xref
target="RFC7104">RTP duplication</xref> procedures SHOULD be
considered instead of simulcast.</t>
<t><list style="empty">
<t>Note: The inclusion of "a=simulcast" or the use of simulcast
does not change any of the interpretation or Offer/Answer
procedures for other SDP attributes, like "a=fmtp" or
"a=rid".</t>
</list></t>
</section>
</section>
<section anchor="sec-relating" title="Relating Simulcast Streams">
<t>As long as there is only a single media source per SDP media
description, simulcast RTP streams can be related on RTP level through
the RTP payload type and (optionally) <xref
target="I-D.pthatcher-mmusic-rid">RID</xref>, as specified in the SDP
<xref target="sec-cap">"a=simulcast" attribute </xref> parameters.
When using <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">BUNDLE</xref> with
multiple SDP media descriptions to specify a single RTP session, there
is an identification mechanism that allows relating RTP streams back
to individual media descriptions, after which the above RTP payload
type and RID relations can be used.</t>
<t>BUNDLE's MID is an RTCP source description (SDES) item. To ensure
rapid initial reception, required to correctly process the RTP
streams, it is also defined as an <xref target="RFC5285">RTP header
extension</xref>.</t>
</section>
<section anchor="sec-ex" title="Signaling Examples">
<t>These examples describe a client to video conference service, using
a centralized media topology with an RTP mixer.</t>
<figure align="center" anchor="fig-mixer-four-party"
title="Four-party Mixer-based Conference">
<artwork align="center"><![CDATA[
+---+ +-----------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| F |<---->| |<---->| J |
+---+ +-----------+ +---+]]></artwork>
</figure>
<section anchor="sec-ex-unified-plan" title="Unified Plan Client">
<t>Alice is calling in to the mixer with a simulcast-enabled Unified
Plan client capable of a single media source per media type. The
client can send a simulcast of 2 video resolutions and frame rates:
HD 1280x720p 30fps and thumbnail 320x180p 15fps. This is defined
below using the <xref target="RFC6236">"imageattr"</xref>. Media
formats (RTP payload types) are used as simulcast stream
identification. Alice's Offer:</t>
<figure align="center" anchor="fig-up-offer"
title="Unified Plan Simulcast Offer">
<artwork align="left"><![CDATA[
v=0
o=alice 2362969037 2362969040 IN IP4 192.0.2.156
s=Simulcast Enabled Unified Plan Client
t=0 0
c=IN IP4 192.0.2.156
m=audio 49200 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 49300 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=simulcast: send pt=97;98 recv pt=97
]]></artwork>
</figure>
<t>The only thing in the SDP that indicates simulcast capability is
the line in the video media description containing the "simulcast"
attribute. The included format parameters indicates that sent
simulcast streams can differ in video resolution.</t>
<t>The Answer from the server indicates that it too is simulcast
capable. Should it not have been simulcast capable, the
"a=simulcast" line would not have been present and communication
would have started with the media negotiated in the SDP.</t>
<figure align="center" anchor="fig-up-answer"
title="Unified Plan Simulcast Answer">
<artwork align="left"><![CDATA[
v=0
o=server 823479283 1209384938 IN IP4 192.0.2.2
s=Answer to Simulcast Enabled Unified Plan Client
t=0 0
c=IN IP4 192.0.2.43
m=audio 49672 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 49674 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=rtpmap:98 H264/90000
a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
a=simulcast: recv pt=97;98 send pt=97
]]></artwork>
</figure>
<t>Since the server is the simulcast media receiver, it reverses the
direction of the "simulcast" attribute parameters.</t>
</section>
<section anchor="sec-ex-multi-source" title="Multi-Source Client">
<t>Fred is calling in to the same conference as in the example above
with a two-camera, two-display system, thus capable of handling two
separate media sources in each direction, where each media source is
simulcast-enabled in the send direction. Fred's client is restricted
to a single media source per media description.</t>
<t>The first two simulcast streams for the first media source use
different codecs, <xref target="RFC6190">H264-SVC</xref> and <xref
target="RFC6184">H264</xref>. These two simulcast streams also have
a temporal dependency. Two different video codecs, <xref
target="I-D.ietf-payload-vp8">VP8</xref> and H264, are offered as
alternatives for the third simulcast stream for the first media
source. RID is used as simulcast stream identification, reducing the
number of media formats needed. Only the highest fidelity simulcast
stream are sent from start, the lower fidelity streams being
initially paused.</t>
<t>The second media source is offered with three different simulcast
streams. All video streams of this second media source are loss
protected by <xref target="RFC4588">RTP retransmission</xref>. RID
is used as simulcast stream identification. Also here, all but the
highest fidelity simulcast stream are initially paused.</t>
<t>Fred's client is also using BUNDLE to send all RTP streams from
all media descriptions in the same RTP session on a single media
transport. Although using many different simulcast streams in this
example, use of RID as simulcast stream identification enables use
of a low number of RTP payload types. Note that the use of both
BUNDLE and RID recommends using the <xref target="RFC5285">RTP
header extension</xref> for carrying these fields.</t>
<figure anchor="fig-ms-offer"
title="Fred's Multi-Source Simulcast Offer">
<artwork><![CDATA[
v=0
o=fred 238947129 823479223 IN IP4 192.0.2.125
s=Offer from Simulcast Enabled Multi-Source Client
t=0 0
c=IN IP4 192.0.2.125
a=group:BUNDLE foo bar zen
m=audio 49200 RTP/AVP 99
a=mid:foo
a=rtpmap:99 G722/8000
m=video 49600 RTP/AVPF 100 101 103
a=mid:bar
a=rtpmap:100 H264-SVC/90000
a=rtpmap:101 H264/90000
a=rtpmap:103 VP8/90000
a=fmtp:100 profile-level-id=42400d; max-fs=3600; max-mbps=108000; \
mst-mode=NI-TC
a=fmtp:101 profile-level-id=42c00d; max-fs=3600; max-mbps=54000
a=fmtp:103 max-fs=900; max-fr=30
a=rid:1 send pt=100;max-width=1280;max-height=720;max-fr=60;depend=2
a=rid:2 send pt=101;max-width=1280;max-height=720;max-fr=30
a=rid:3 send pt=101;max-width=640;max-height=360
a=rid:4 send pt=103;max-width=640;max-height=360
a=depend:100 lay bar:101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:rid
a=rtcp-fb:* ccm pause nowait
a=simulcast: send rid=1;2;4,3 paused=2,3,4
m=video 49602 RTP/AVPF 96 104
a=mid:zen
a=rtpmap:96 VP8/90000
a=fmtp:96 max-fs=3600; max-fr=30
a=rtpmap:104 rtx/90000
a=fmtp:104 apt=96;rtx-time=200
a=rid:5 send pt=96;max-fs=921600;max-fr=30
a=rid:6 send pt=96;max-fs=614400;max-fr=15
a=rid:7 send pt=96;max-fs=230400;max-fr=30
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:2 urn:ietf:params:rtp-hdrext:rid
a=rtcp-fb:* ccm pause nowait
a=simulcast: send rid=5;6;7 paused=6,7
]]></artwork>
</figure>
<t><list style="empty">
<t>Note: Empty lines in the SDP above are added only for
readability and would not be present in an actual SDP.</t>
</list></t>
</section>
</section>
</section>
<section anchor="sec-network-aspects" title="Network Aspects">
<t>Simulcast is in this memo defined as the act of sending multiple
alternative encoded streams of the same underlying media source. When
transmitting multiple independent streams that originate from the same
source, it could potentially be done in several different ways using
RTP. A general discussion on considerations for use of the different RTP
multiplexing alternatives can be found in <xref
target="I-D.ietf-avtcore-multiplex-guidelines">Guidelines for
Multiplexing in RTP</xref>. Discussion and clarification on how to
handle multiple streams in an RTP session can be found in <xref
target="I-D.ietf-avtcore-rtp-multi-stream"/>.</t>
<t>The network aspects that are relevant for simulcast are:<list
style="hanging">
<t hangText="Quality of Service:">When using simulcast it might be
of interest to prioritize a particular simulcast stream, rather than
applying equal treatment to all streams. For example, lower bit-rate
streams may be prioritized over higher bit-rate streams to minimize
congestion or packet losses in the low bit-rate streams. Thus, there
is a benefit to use a simulcast solution that supports QoS as good
as possible.</t>
<t hangText="NAT/FW Traversal:">Using multiple RTP sessions incurs
more cost for NAT/FW traversal unless they can re-use the same
transport flow, which can be achieved by <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation">Multiplexing
Negotiation Using SDP Port Numbers</xref>.</t>
</list></t>
<t/>
</section>
<section anchor="sec-limitation" title="Limitations">
<t>The chosen approach has a few limitations that are described in this
section. Some relate to the use of a single RTP session for all
simulcast formats of a media source, while others relate to the two
different simulcast stream identification methods.</t>
<section title="Single RTP Session">
<t>The limitations in this section come from sending all simulcast
streams related to a media source under the same SDP media
description, which also means they are sent in the same RTP
session.</t>
<t>It is not possible to use different simulcast streams on different
transports, limiting the possibilities to apply different QoS to
different simulcast streams. When using unicast, QoS mechanisms based
on individual packet marking are feasible, since they do not require
separation of simulcast streams into different RTP sessions to apply
different QoS.</t>
<t>It is not possible to separate different simulcast streams into
different multicast groups to allow a multicast receiver to pick the
stream it wants, rather than receive all of them. In this case, the
only reasonable implementation is to use different RTP sessions for
each multicast group so that reporting and other RTCP functions
operate as intended.</t>
</section>
<section title="SDP Format Identification">
<t>The limitations in this section come from and thus apply only when
using SDP format (RTP payload type) as simulcast stream identification
method.</t>
<t>The available RTP payload type number space may not be sufficient
when many different media formats and/or simulcast streams are used in
the SDP. This can be particularly prominent when BUNDLE is used, and
for any technology that adds to the number of required RTP payload
types in a multiplicative way, such as for example adding <xref
target="RFC4588">RTP retransmission</xref> and <xref
target="RFC5109">Forward Error Correction</xref>. <xref
target="I-D.ietf-payload-flexible-fec-scheme">Flexible FEC
Scheme</xref> can be used for RTP retransmissions and would avoid the
double consumption of the PT space that <xref target="RFC4588">RTP
Retransmission</xref> causes. </t>
<t>Only existing SDP attributes and parameters can be used to define
codec configuration for a simulcast format. Any codec that does not
define a sufficient set of codec parameters in "a=fmtp", or can make
use of other SDP attributes, may not be capable of expressing the
desired <xref target="sec-diverse-receivers">simulcast format
dimensions</xref> with necessary precision, or not at all. One example
of this is the ability to separate simulcast formats by bandwidth for
codecs lacking a codec-specific bandwidth parameter, since the SDP
"b="-line covers all RTP payload types listed on an "m="-line. </t>
<t>A simulcast stream signaled as initially paused is not possible to
resume by a remote peer, because it cannot know which target SSRC to
use in the <xref target="I-D.ietf-avtext-rtp-stream-pause">RESUME
message</xref>.</t>
</section>
<section title="RID Identification">
<t>The limitations in this section come from and thus apply only when
using RID as simulcast stream identification method.</t>
<t>Use of the additional "a=rid"-line in SDP and the corresponding RID
RTCP SDES item and RTP header extension requires some additional
implementation complexity, and incurs some extra bandwidth cost to
carry the RID RTCP SDES item and RTP header extension.</t>
</section>
</section>
<section anchor="sec-iana" title="IANA Considerations">
<t>This document requests to register a new SDP attribute,
simulcast.</t>
<t>Formal registrations to be written.</t>
</section>
<section anchor="sec-security" title="Security Considerations">
<t>The simulcast capability, configuration attributes and parameters are
vulnerable to attacks in signaling.</t>
<t>A false inclusion of the "a=simulcast" attribute may result in
simultaneous transmission of multiple RTP streams that would otherwise
not be generated. The impact is limited by the media description joint
bandwidth, shared by all simulcast streams irrespective of their number.
There may however be a large number of unwanted RTP streams that will
impact the share of bandwidth allocated for the originally wanted RTP
stream.</t>
<t>A hostile removal of the "a=simulcast" attribute will result in
simulcast not being used.</t>
<t>Neither of the above will likely have any major consequences and can
be mitigated by signaling that is at least integrity and source
authenticated to prevent an attacker to change it.</t>
</section>
<section title="Contributors">
<t>Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have
contributed with important material to the first versions of this
document. Robert Hansen and Cullen Jennings, from Cisco, and Peter
Thatcher, from Google, contributed significantly to subsequent
versions.</t>
</section>
<section anchor="sec-ack" title="Acknowledgements">
<t/>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.4566'?>
<?rfc include='reference.RFC.5109'?>
<?rfc include='reference.RFC.5234'?>
<?rfc include='reference.RFC.7104'?>
<?rfc include='reference.I-D.pthatcher-mmusic-rid'?>
<?rfc include='reference.I-D.ietf-avtext-rtp-stream-pause'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.2198'?>
<?rfc include='reference.RFC.3264'?>
<?rfc include='reference.RFC.3389'?>
<?rfc include='reference.RFC.4588'?>
<?rfc include='reference.RFC.4733'?>
<?rfc include='reference.RFC.5117'?>
<?rfc include='reference.RFC.5285'?>
<?rfc include='reference.RFC.5576'?>
<?rfc include='reference.RFC.5583'?>
<?rfc include='reference.RFC.6184'?>
<?rfc include='reference.RFC.6190'?>
<?rfc include='reference.RFC.6236'?>
<?rfc include='reference.I-D.ietf-avtcore-multiplex-guidelines'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-multi-stream'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-topologies-update'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
<?rfc include='reference.I-D.ietf-avtext-rtp-grouping-taxonomy'?>
<?rfc include='reference.I-D.ietf-payload-vp8'?>
<?rfc include='reference.I-D.ietf-payload-flexible-fec-scheme'?>
</references>
<section title="Changes From Earlier Versions">
<t>NOTE TO RFC EDITOR: Please remove this section prior to
publication.</t>
<section title="Modifications Between WG Version -02 and -03">
<t><list style="symbols">
<t>Removed text on multicast / broadcast from use cases, since it
is not supported by the solution.</t>
<t>Removed explicit references to unified plan draft.</t>
<t>Added possibility to initiate simulcast streams in paused
mode.</t>
<t>Enabled an offerer to offer multiple stream identification (pt
or rid) methods and have the answerer choose which to use.</t>
<t>Added a preference indication also in send direction
offers.</t>
<t>Added a section on limitations of the current proposal,
including identification method specific limitations.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -01 and -02">
<t><list style="symbols">
<t>Relying on the new RID solution for codec constraints and
configuration identification. This has resulted in changes in
syntax to identify if pt or RID is used to describe the simulcast
stream.</t>
<t>Renamed simulcast version and simulcast version alternative to
simulcast stream and simulcast format respectively, and improved
definitions for them.</t>
<t>Clarification that it is possible to switch between simulcast
version alternatives, but that only a single one be used at any
point in time.</t>
<t>Changed the definition so that ordering of simulcast formats
for a specific simulcast stream do have a preference order.</t>
</list></t>
</section>
<section title="Modifications Between WG Version -00 and -01">
<t><list style="symbols">
<t>No changes. Only preventing expiry.</t>
</list></t>
</section>
<section title="Modifications Between Individual Version -00 and WG Version -00">
<t><list style="symbols">
<t>Added this appendix.</t>
</list></t>
</section>
</section>
</back>
</rfc>
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