One document matched: draft-ietf-avtcore-rtp-multi-stream-07.xml
<?xml version="1.0" encoding="US-ASCII"?>
<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
<?rfc toc="yes" ?>
<?rfc strict="yes" ?>
<?rfc compact="yes" ?>
<?rfc sortrefs="yes" ?>
<?rfc symrefs="yes" ?>
<rfc category="std" docName="draft-ietf-avtcore-rtp-multi-stream-07"
ipr="trust200902" updates="3550, 4585">
<front>
<title abbrev="Multiple Media Streams in an RTP Session">Sending Multiple
Media Streams in a Single RTP Session</title>
<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>USA</country>
</postal>
<email>jonathan@vidyo.com</email>
</address>
</author>
<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="Qin Wu" initials="Q." surname="Wu">
<organization>Huawei</organization>
<address>
<postal>
<street>101 Software Avenue, Yuhua District</street>
<city>Nanjing, Jiangsu 210012</city>
<country>China</country>
</postal>
<email>sunseawq@huawei.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>
<date/>
<area>RAI</area>
<workgroup>AVTCORE</workgroup>
<keyword>I-D</keyword>
<keyword>Internet-Draft</keyword>
<!-- TODO: more keywords -->
<abstract>
<t>This memo expands and clarifies the behaviour of Real-time Transport
Protocol (RTP) endpoints that use multiple synchronization sources
(SSRCs). This occurs, for example, when an endpoint sends multiple media
streams in a single RTP session. This memo updates RFC 3550 with regards
to handling multiple SSRCs per endpoint in RTP sessions, with a
particular focus on RTCP behaviour. It also updates RFC 4585 to update
and clarify the calculation of the timeout of SSRCs and the inclusion of
feedback messages.</t>
</abstract>
</front>
<middle>
<section anchor="introduction" title="Introduction">
<t>At the time the <xref target="RFC3550">Real-Time Transport Protocol
(RTP)</xref> was originally designed, and for quite some time after,
endpoints in RTP sessions typically only transmitted a single media
stream, and thus used a single synchronization source (SSRC) per RTP
session, where separate RTP sessions were typically used for each
distinct media type. Recently, however, a number of scenarios have
emerged in which endpoints wish to send multiple RTP media streams,
distinguished by distinct RTP synchronization source (SSRC) identifiers,
in a single RTP session. These are outlined in <xref
target="usecases"/>. Although the initial design of RTP did consider
such scenarios, the specification was not consistently written with such
use cases in mind. The specifications are thus somewhat unclear.</t>
<t>This memo updates <xref target="RFC3550"/> to clarify behaviour in
use cases where endpoints use multiple SSRCs. It also updates <xref
target="RFC4585"/> in regards to the timeout of inactive SSRCs to
resolve problematic behaviour as well as clarifying the inclusion of
feedback messages.</t>
</section>
<section title="Terminology">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in <xref
target="RFC2119">RFC 2119</xref> and indicate requirement levels for
compliant implementations.</t>
</section>
<section anchor="usecases" title="Use Cases For Multi-Stream Endpoints">
<t>This section discusses several use cases that have motivated the
development of endpoints that sends RTP data using multiple SSRCs in a
single RTP session.</t>
<section anchor="telepresence"
title="Endpoints with Multiple Capture Devices">
<t>The most straightforward motivation for an endpoint to send
multiple simultaneous RTP streams in a session is the scenario where
an endpoint has multiple capture devices, and thus media sources, of
the same media type and characteristics. For example, telepresence
endpoints, of the type described by the <xref
target="I-D.ietf-clue-framework">CLUE Telepresence Framework</xref>,
often have multiple cameras or microphones covering various areas of a
room, and hence send several RTP streams.</t>
</section>
<section title="Multiple Media Types in a Single RTP Session">
<t>Recent work has updated <xref
target="I-D.ietf-avtcore-multi-media-rtp-session"> RTP</xref> and
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"> SDP</xref> to
remove the historical assumption in RTP that media sources of
different media types would always be sent on different RTP sessions.
In this work, a single endpoint's audio and video RTP media streams
(for example) are instead sent in a single RTP session to reduce the
number of transport layer flows used.</t>
</section>
<section title="Multiple Stream Mixers">
<t>There are several RTP topologies which can involve a central device
that itself generates multiple RTP media streams in a session. An
example is a mixer providing centralized compositing for a
multi-capture scenario like that described in <xref
target="telepresence"/>. In this case, the centralized node is
behaving much like a multi-capturer endpoint, generating several
similar and related sources.</t>
<t>A more complex example is the selective forwarding middlebox,
described in Section 3.7 of <xref
target="I-D.ietf-avtcore-rtp-topologies-update"/>. This is a middlebox
that receives media streams from several endpoints, and then
selectively forwards modified versions of some RTP streams toward the
other endpoints to which it is connected. For each connected endpoint,
a separate media source appears in the session for every other source
connected to the middlebox, "projected" from the original streams, but
at any given time many of them can appear to be inactive (and thus are
receivers, not senders, in RTP). This sort of device is closer to
being an RTP mixer than an RTP translator, in that it terminates RTCP
reporting about the mixed streams, and it can re-write SSRCs,
timestamps, and sequence numbers, as well as the contents of the RTP
payloads, and can turn sources on and off at will without appearing to
be generating packet loss. Each projected stream will typically
preserve its original RTCP source description (SDES) information.</t>
</section>
<section title="Multiple SSRCs for a Single Media Source">
<t>There are also several cases where a single media source results in
the usage of multiple SSRCs within the same RTP session. Transport
robustness tools like <xref target="RFC4588">RTP Retransmission</xref>
result in multiple SSRCs, one with source data, and another with the
repair data. Scalable encoders and their RTP payload formats, like
H.264's extension for Scalable Video Coding<xref target="RFC6190">
(SVC)</xref> can be transmitted in a configuration where the scalable
layers are distributed over multiple SSRCs within the same session, to
enable RTP packet stream level (SSRC) selection and routing in
conferencing middleboxes.</t>
</section>
</section>
<section title="Use of RTP by endpoints that send multiple media streams">
<t>Every RTP endpoint will have an allocated share of the available
session bandwidth, as determined by signalling and congestion control.
The endpoint MUST keep its total media sending rate within this share.
However, endpoints that send multiple media streams do not necessarily
need to subdivide their share of the available bandwidth independently
or uniformly to each media stream and its SSRCs. In particular, an
endpoint can vary the allocation to different streams depending on their
needs, and can dynamically change the bandwidth allocated to different
SSRCs (for example, by using a variable rate codec), provided the total
sending rate does not exceed its allocated share. This includes enabling
or disabling media streams and their redundancy streams as more or less
bandwidth becomes available.</t>
</section>
<section title="Use of RTCP by Endpoints that send multiple media streams">
<t>The RTP Control Protocol (RTCP) is defined in Section 6 of <xref
target="RFC3550"/>. The description of the protocol is phrased in terms
of the behaviour of "participants" in an RTP session, under the
assumption that each endpoint is a participant with a single SSRC.
However, for correct operation in cases where endpoints can send
multiple media streams, the specification needs to be interpreted with
each SSRC counting as a participant in the session, so that an endpoint
that has multiple SSRCs counts as multiple participants. The following
describes several concrete cases where this applies.</t>
<section anchor="reporting" title="RTCP Reporting Requirement">
<t>An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a
separate participant in the RTP session, sending RTCP reports for each
of its SSRCs in every RTCP reporting interval. If the mechanism in
<xref target="I-D.ietf-avtcore-rtp-multi-stream-optimisation"/> is not
used, then each SSRC will send RTCP reports for all other SSRCs,
including those co-located at the same endpoint.</t>
<t>If the endpoint has some SSRCs that are sending data and some that
are only receivers, then they will receive different shares of the
RTCP bandwidth and calculate different base RTCP reporting intervals.
Otherwise, all SSRCs at an endpoint will calculate the same base RTCP
reporting interval. The actual reporting intervals for each SSRC are
randomised in the usual way, but reports can be aggregated as
described in <xref target="compound"/>.</t>
</section>
<section title="Initial Reporting Interval">
<t>When a participant joins a unicast session, the following text from
Section 6.2 of <xref target="RFC3550"/> is relevant: "For unicast
sessions... the delay before sending the initial compound RTCP packet
MAY be zero." The basic assumption is that this also ought to apply in
the case of multiple SSRCs. Caution has to be exercised, however, when
an endpoint (or middlebox) with a large number of SSRCs joins a
unicast session, since immediate transmission of many RTCP reports can
create a significant burst of traffic, leading to transient congestion
and packet loss due to queue overflows.</t>
<t>To ensure that the initial burst of traffic generated by an RTP
endpoint is no larger than would be generated by a TCP connection, an
RTP endpoint MUST NOT send more than four compound RTCP packets with
zero initial delay when it joins a session. Each of those initial
compound RTCP packets MAY include aggregated reports from multiple
SSRCs, provided the total compound RTCP packet size does not exceed
the MTU, and the avg_rtcp_packet_size is maintained as in <xref
target="avg_rtcp_size"/>. Aggregating reports from several SSRCs in
the initial compound RTCP packets allows a substantial number of SSRCs
to report immediately. Endpoints SHOULD prioritize reports on SSRCs
that are likely to be most immediately useful, e.g., for SSRCs that
are initially senders.</t>
<t>An endpoint that needs to report on more SSRCs than will fit into
the four compound RTCP reports that can be sent immediately MUST send
the other reports later, following the usual RTCP timing rules
including timer reconsideration. Those reports MAY be aggregated as
described in <xref target="compound"/>.</t>
<t><list style="empty">
<t>Note: The above is based on an TCP initial window of 4 packets,
not the larger initial windows which there is an ongoing
experiment with. The reason for this is a desire to be
conservative as an RTP endpoint will also in many cases commence
RTP transmission at the same time as these initial RTCP packets
are sent.</t>
</list></t>
</section>
<section anchor="compound"
title="Aggregation of Reports into Compound RTCP Packets">
<t>As outlined in <xref target="reporting"/>, an endpoint with
multiple SSRCs has to treat each SSRC as a separate participant when
it comes to sending RTCP reports. This will lead to each SSRC sending
a compound RTCP packet in each reporting interval. Since these packets
are coming from the same endpoint, it might reasonably be expected
that they can be aggregated to reduce overheads. Indeed, Section 6.1
of <xref target="RFC3550"/> allows RTP translators and mixers to
aggregate packets in similar circumstances:</t>
<t><list style="empty">
<t>"It is RECOMMENDED that translators and mixers combine
individual RTCP packets from the multiple sources they are
forwarding into one compound packet whenever feasible in order to
amortize the packet overhead (see Section 7). An example RTCP
compound packet as might be produced by a mixer is shown in Fig.
1. If the overall length of a compound packet would exceed the MTU
of the network path, it SHOULD be segmented into multiple shorter
compound packets to be transmitted in separate packets of the
underlying protocol. This does not impair the RTCP bandwidth
estimation because each compound packet represents at least one
distinct participant. Note that each of the compound packets MUST
begin with an SR or RR packet."</t>
</list></t>
<t>This allows RTP translators and mixers to generate compound RTCP
packets that contain multiple SR or RR packets from different SSRCs,
as well as any of the other packet types. There are no restrictions on
the order in which the RTCP packets can occur within the compound
packet, except the regular rule that the compound RTCP packet starts
with an SR or RR packet. Due to this rule, correctly implemented RTP
endpoints will be able to handle compound RTCP packets that contain
RTCP packets relating to multiple SSRCs.</t>
<t>Accordingly, endpoints that use multiple SSRCs MAY aggregate the
RTCP packets sent by their different SSRCs into compound RTCP packets,
provided 1) the resulting compound RTCP packets begin with an SR or RR
packet; 2) they maintain the average RTCP packet size as described in
<xref target="avg_rtcp_size"/>; and 3) they schedule packet
transmission and manage aggregation as described in <xref
target="agg"/>.</t>
<section anchor="avg_rtcp_size" title="Maintaining AVG_RTCP_SIZE">
<t>The RTCP scheduling algorithm in <xref target="RFC3550"/> works
on a per-SSRC basis. Each SSRC sends a single compound RTCP packet
in each RTCP reporting interval. When an endpoint uses multiple
SSRCs, it is desirable to aggregate the compound RTCP packets sent
by its SSRCs, reducing the overhead by forming a larger compound
RTCP packet. This aggregation can be done as described in <xref
target="agg"/>, provided the average RTCP packet size calculation is
updated as follows.</t>
<t>Participants in an RTP session update their estimate of the
average RTCP packet size (avg_rtcp_size) each time they send or
receive an RTCP packet (see Section 6.3.3 of <xref
target="RFC3550"/>). When a compound RTCP packet that contains RTCP
packets from several SSRCs is sent or received, the avg_rtcp_size
estimate for each SSRC that is reported upon is updated using
div_packet_size rather than the actual packet size:</t>
<figure>
<artwork><![CDATA[
avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size
]]></artwork>
</figure>
<t>where div_packet_size is packet_size divided by the number of
SSRCs reporting in that compound packet. The number of SSRCs
reporting in a compound packet is determined by counting the number
of different SSRCs that are the source of Sender Report (SR) or
Receiver Report (RR) RTCP packets within the compound RTCP packet.
Non-compound RTCP packets (i.e., RTCP packets that do not contain an
SR or RR packet <xref target="RFC5506"/>) are considered to report
on a single SSRC.</t>
<t>An SSRC that doesn't follow the above rule, and instead uses the
full RTCP compound packet size to calculate avg_rtcp_size, will
derive an RTCP reporting interval that is overly large by a factor
that is proportional to the number of SSRCs aggregated into compound
RTCP packets and the size of set of SSRCs being aggregated relative
to the total number of participants. This increased RTCP reporting
interval can cause premature timeouts if it is more than five times
the interval chosen by the SSRCs that understand compound RTCP that
aggregate reports from many SSRCs. A 1500 octet MTU can fit five
typical size reports into a compound RTCP packet, so this is a real
concern if endpoints aggregate RTCP reports from multiple SSRCs.</t>
<t>The issue raised in the previous paragraph is mitigated by the
modification in timeout behaviour specified in <xref
target="sec-timeout"/>. This mitigation is in place in those cases
where the RTCP bandwidth is sufficiently high that an endpoint,
using an avg_rtcp_size calculated without taking into account the
number of reporting SSRCs, can transmit more frequently than
approximately every 5 seconds. Note, however, that the non-modified
endpoint's RTCP reporting is still negatively impacted even if the
premature timeout of its SSRCs are avoided. If compatibility with
non-updated endpoints is a concern, the number of reports from
different SSRCs aggregated into a single compound RTCP packet SHOULD
either be limited to two reports, or aggregation ought not used at
all. This will limit the non-updated endpoint's RTCP reporting
interval to be no larger than twice the RTCP reporting interval that
would be chosen by an endpoint following this specification.</t>
</section>
<section anchor="agg"
title="Scheduling RTCP with Multiple Reporting SSRCs">
<t>When implementing RTCP packet scheduling for cases where multiple
reporting SSRCs are aggregating their RTCP packets in the same
compound packet there are a number of challenges. First of all, we
have the goal of not changing the general properties of the RTCP
packet transmissions, which include the general inter-packet
distribution, and the behaviour for dealing with flash joins as well
as other dynamic events.</t>
<t>The below specified mechanism deals with:<list style="symbols">
<t>That one can't have a-priori knowledge about which RTCP
packets are to be sent, or their size, prior to generating the
packets. In which case, the time from generation to transmission
ought to be as short as possible to minimize the information
that becomes stale.</t>
<t>That one has an MTU limit, that one ought to avoid exceeding,
as that requires lower-layer fragmentation (e.g., IP
fragmentation) which impacts the packets' probability of
reaching the receiver(s).</t>
</list></t>
<t>The below text modifies and extends the behavior defined in
Section 6.3 of <xref target="RFC3550"/>, and in Section 3.5.3 of
<xref target="RFC4585"/> if the AVPF or SAVPF profile is used,
regarding actions to take when scheduling and sending an RTCP
packet. It uses the variable names tn, tp, tc, T and Td defined in
Section 6.3 of <xref target="RFC3550"/>. The variable T_rr_last is
defined in <xref target="RFC4585"/>.</t>
<t>Schedule all the endpoint's local SSRCs individually for
transmission using the regular calculation of tn for the profile
being used. Each time an SSRC's tn timer expires, do the regular
reconsideration and, if applicable, T_rr_int based suppression. If
the result indicates that an RTCP packet is to be sent and the
transmission is a regular RTCP packet:<list style="numbers">
<t>Consider if an additional SSRC can be added. That
consideration is done by picking the SSRC which has the tn value
closest in time to the current time (tc).</t>
<t>Calculate how much space for RTCP packets would be needed to
add that SSRC.</t>
<t>If the considered SSRC's RTCP Packets fit within the lower
layer datagram's Maximum Transmission Unit, taking the necessary
protocol headers and the space consumed by prior SSRCs into
account, then add that SSRC's RTCP packets to the compound
packet and go again to Step 1.</t>
<t>Otherwise, if the considered SSRC's RTCP Packets will not fit
within the compound packet, then transmit the generated compound
packet.</t>
<t>Update the RTCP Parameters for each SSRC that has been
included in the sent RTCP packet. The previous RTCP transmit
time (tp) value for each SSRC MUST be updated as follows:<list
style="letters">
<t>For the first SSRC set the transmission time (tt) to
tc.</t>
<t>For any additional SSRC calculate the transmission time
that each of these SSRCs would have had it not been
aggregated and given the current existing session context.
This value is derived by taking this SSRC's tn value and
performing reconsideration and updating tn until tp + T
<= tn, then set tt = tn. If AVPF or SAVPF is being used,
then T_rr_int based suppression MUST NOT be used in this
calcualtion.</t>
<t>Calculate average transmission time (tt_avg) using the tt
of all the SSRCs included in the packet.</t>
<t>Now update tp for all the sent SSRCs to tt_avg.</t>
<t>If AVPF or SAVPF profile is being used update T_rr_last
to tt_avg.</t>
</list></t>
<t>For the sent SSRCs calculate new tn values based on the
updated parameters and reschedule the timers.</t>
</list></t>
<t>When using AVPF or SAVPF profile, when following the scheduling
algorithm for regular transmission in Section 3.5.3 then the case of
T_rr_interval == 0, as well as option 1, 2a and 2b for T_rr_interval
!= 0, results in transmission of a regular RTCP packet that follows
the above and updates the necessary variables. However, when the
transmission is suppressed per 2c, then tp is updated to tc, as no
aggregation has taken place.</t>
<t>Reverse reconsideration needs to be performed as specified in
<xref target="RFC3550">RTP</xref>. It is important to note that
under the above algorithm when performing reconsideration, the value
of tp can actually be larger than tc. However, that still has the
desired effect of proportionally pulling the tp value towards tc (as
well as tn) as the group size shrinks in direct proportion the
reduced group size.</t>
<t>The above algorithm has been shown in simulations to maintain the
inter-RTCP-packet transmission distribution for the SSRCs and
consume the same amount of bandwidth as non-aggregated packets in
RTP sessions. With this algorithm the actual transmission interval
for any SSRC triggering an RTCP compound packet transmission is
following the regular transmission rules. The value tp is set to
somewhere in the interval [0,1.5/1.21828*Td] ahead of tc. The actual
value is average of one instance of tc and the randomized
transmission times of the additional SSRCs, thus the lower range of
the interval is more probable. This setting is performed to
compensate for the bias that is otherwise introduced by picking the
shortest tn value out of the N SSRCs included in aggregate.</t>
<t>The algorithm also handles the cases where the number of SSRCs
that can be included in an aggregated packet varies. An SSRC that
previously was aggregated and fails to fit in a packet still has its
own transmission scheduled according to normal rules. Thus, it will
trigger a transmission in due time, or the SSRC will be included in
another aggregate. The algorithm's behaviour under SSRC group size
changes is as follows:</t>
<t><list style="hanging">
<t
hangText="RTP sessions where the number of SSRC are growing:">When
the group size is growing, the Td values grow in proportion to
the number of new SSRCs in the group. When reconsideration is
done when the timer for the tn expires, that SSRC will
reconsider the transmission and with a certain probability
reschedule the tn timer. This part of the reconsideration
algorithm is only impacted by the above algorithm by having tp
values that were in the future instead of set to the time of the
actual last transmission at the time of updating tp.</t>
<t
hangText="RTP sessions where the number of SSRC are shrinking:">When
the group shrinks, reverse reconsideration moves the tp and tn
values towards tc proportionally to the number of SSRCs that
leave the session compared to the total number of participants
when they left. The setting of the tp value forward in time
related to the tc could be believed to have negative effect.
However, the reason for this setting is to compensate for bias
caused by picking the shortest tn out of the N aggregated. This
bias remains over a reduction in the number of SSRCs. The
reverse reconsideration compensates the reduction independently
of aggregation being used or not. The negative effect that can
occur on removing an SSRC is that the most favourable tn
belonged to the removed SSRC. The impact of this is limited to
delaying the transmission, in the worst case, one reporting
interval.</t>
</list></t>
<t>In conclusion the investigations performed has found no
significant negative impact on the scheduling algorithm.</t>
</section>
</section>
<section title="Use of RTP/AVPF Feedback">
<t>This section discusses the transmission of RTP/AVPF feedback
packets when the transmitting endpoint has multiple SSRCs.</t>
<section title="Choice of SSRC for Feedback Packets">
<t>When an RTP/AVPF endpoint has multiple SSRCs, it can choose what
SSRC to use as the source for the RTCP feedback packets it sends.
Several factors can affect that choice: <list style="symbols">
<t>RTCP feedback packets relating to a particular media type
SHOULD be sent by an SSRC that receives that media type. For
example, when audio and video are multiplexed onto a single RTP
session, endpoints will use their audio SSRC to send feedback on
the audio received from other participants.</t>
<t>RTCP feedback packets and RTCP codec control messages that
are notifications or indications regarding RTP data processed by
an endpoint MUST be sent from the SSRC used by that RTP data.
This includes notifications that relate to a previously received
request or command <xref target="RFC4585"/><xref
target="RFC5104"/>.</t>
<t>If separate SSRCs are used to send and receive media, then
the corresponding SSRC SHOULD be used for feedback, since they
have differing RTCP bandwidth fractions. This can also affect
the consideration if the SSRC can be used in immediate mode or
not.</t>
<t>Some RTCP feedback packet types require consistency in the
SSRC used. For example, if a <xref target="RFC5104">TMMBR
limitation</xref> is set by an SSRC, the same SSRC needs to be
used to remove the limitation.</t>
<t>If several SSRCs are suitable for sending feedback, if might
be desirable to use an SSRC that allows the sending of feedback
as an early RTCP packet.</t>
</list></t>
<t>When an RTCP feedback packet is sent as part of a compound RTCP
packet that aggregates reports from multiple SSRCs, there is no
requirement that the compound packet contains an SR or RR packet
generated by the sender of the RTCP feedback packet. For
reduced-size RTCP packets, aggregation of RTCP feedback packets from
multiple sources is not limited further than Section 4.2.2 of <xref
target="RFC5506"/>.</t>
</section>
<section title="Scheduling an RTCP Feedback Packet">
<t>When an SSRC has a need to transmit a feedback packet in early
mode it follows the scheduling rules defined in Section 3.5 in <xref
target="RFC4585">RTP/AVPF</xref>. When following these rules the
following clarifications need to be taken into account:<list
style="symbols">
<t>Whether a session is considered to be point-to-point or
multiparty is not based on the number of SSRCs, but the number
of endpoints one directly interacts with in the RTP session.
This is determined by counting the number of CNAMEs used by the
SSRCs received. A RTP session MUST be considered multiparty if
more than one CNAME is received, unless signalling explicitly
indicates that the session is to be handled as point to point,
or <xref
target="I-D.ietf-avtcore-rtp-multi-stream-optimisation">RTCP
reporting groups</xref> are used. If RTCP reporting groups
are used, the classification is solely based on whether the endpoint
receives a single reporting group, indicating point to
point, or if multiple reporting groups are received (or
a mixture of sources using and sources not using
reporting groups), which is
classified as multiparty. Note that contributing sources (CSRCs)
can be bound to any number of different CNAMEs and do not affect
the determination of whether the session is multiparty.
Similarly, SSRC/CSRC values that are only seen in the source
field of an SDES packet do not affect this determination.</t>
<t>Note that when checking if there is already a scheduled
compound RTCP packet containing feedback messages (Step 2 in
Section 3.5.2), that check is done considering all local
SSRCs.</t>
<t>If the SSRC is not allowed to send an early RTCP packet,
then the feedback message MAY be queued for transmission as part
of any early or regular scheduled transmission that can occur
within the maximum useful lifetime of the feedback message
(T_max_fb_delay). This modifies the behaviour in bullet 4a) in
Section 3.5.2 of <xref target="RFC4585"/>.</t>
</list></t>
<t>The above rule for determining if a RTP session is to be
considered point-to-point or multiparty is simple and
straightforward and works in most cases. The goal with the above
classification is to determine if the resources associated with RTP
and RTCP are shared with only one peer or multiple other endpoints.
This is significant as it affects the impact and the necessary
processing and resource consumption. Relying on only CNAME will
result in classifying some few situations where one might actually
have only one peer as a multiparty situation. The known situations
are the following ones:<list style="hanging">
<t
hangText="Endpoint with multiple synchronization contexts:">An
endpoint that is part of a point-to-point session can have
multiple synchronization contexts, for example due to forwarding
an external media source into a interactive real-time
conversation. In this case the classification will consider the
peer as two endpoints, while the actual RTP/RTCP transmission
will be under the control of one endpoint.</t>
<t hangText="Selective Forwarding Middlebox:">The SFM as defined
in <xref target="I-D.ietf-avtcore-rtp-topologies-update">Section
3.7 of </xref> has control over the transmission and
configurations between itself and each peer endpoint
individually. It also fully controls the RTCP packets being
forwarded between the individual legs. Thus, this type of
middlebox can be compared to the RTP mixer, which uses its own
SSRCs to mix or select the media it forwards, that will be
classified as a point-to-point RTP session by the above
rule.</t>
</list>In the above cases it is very reasonable to use <xref
target="I-D.ietf-avtcore-rtp-multi-stream-optimisation">RTCP
reporting groups</xref>. If that extension is used, an endpoint can
indicate that the multitude of CNAMEs are in fact under a single
endpoint or middlebox control by using only a single reporting
group.</t>
<t>The above rules will also classify some sessions where the
endpoint is connected to an RTP mixer as being point to point. For
example the mixer could act as gateway to an Any Source Multicast
based RTP session for the discussed endpoint. However, this will in
most cases be okay, as the RTP mixer provides separation between the
two parts of the session. The responsibility falls on the mixer to
act accordingly in each domain.</t>
<t>Note: The above usage of point-to-point or multiparty as
classifiers is actually misleading, but we maintain these labels
to match what is used in <xref target="RFC4585"/> as this ensures
that the right algorithms are applied.</t>
<t>To conclude we note that in some cases signalling can be used to
override the rule when it would result in the wrong
classification.</t>
</section>
</section>
</section>
<section title="RTCP Considerations for Streams with Disparate Rates">
<t>An RTP session has a single set of parameters that configure the
session bandwidth. These are the RTCP sender and receiver fractions
(e.g., 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 base RTCP reporting
interval, before randomisation, will be the same for every sending SSRC
in an RTP session. Similarly, every receiving SSRC in an RTP session
will have the same base reporting interval, although this can differ
from the reporting interval chosen by sending SSRCs. This uniform RTCP
reporting interval for all SSRCs can result in RTCP reports being sent
more often, or too seldom, than is considered desirable for a RTP
stream.</t>
<t>For example, consider a scenario when an audio flow sending at tens
of kilobits per second is multiplexed into an RTP session with a
multi-megabit high quality video flow. If the session bandwidth is
configured based on the video sending rate, and the default RTCP
bandwidth fraction of 5% of the session bandwidth is used, it is likely
that the RTCP bandwidth will exceed the audio sending rate. If the
reduced minimum RTCP interval described in Section 6.2 of <xref
target="RFC3550"/> is then used in the session, as appropriate for video
where rapid feedback on damaged I-frames is wanted, the uniform
reporting interval for all senders could mean that audio sources are
expected to send RTCP packets more often than they send audio data
packets. This bandwidth mismatch can be reduced by careful tuning of the
RTCP parameters, especially trr_int when the RTP/AVPF profile is used,
cannot be avoided entirely, as it is inherent in the design of the RTCP
timing rules, and affects all RTP sessions that contain flows with
greatly mismatched bandwidth.</t>
<t>Different media rates or desired RTCP behaviours can also occur
between SSRCs carrying the same media type. A common case in multiparty
conferencing is when only one or two video source are shown in higher
resolution, while the others are shown as small thumbnails, with the
choice of which is shown in high resolution being voice activity
controlled. Here the differences are both in actual media rate and in
choices for what feedback messages might be needed. Other examples of
differences that can exist are due to the intended usage of a media
source. A media source carrying the video of the speaker in a conference
is different from a document camera. Basic parameters that can differ in
this case are frame-rate, acceptable end-to-end delay, and the SNR
fidelity of the image. These differences affect not only the needed
bit-rates, but also possible transmission behaviours, usable repair
mechanisms, what feedback messages the control and repair requires, the
transmission requirements on those feedback messages, and monitoring of
the RTP stream delivery.</t>
<t>Sending multiple media types in a single RTP session causes that
session to contain more SSRCs than if each media type was sent in a
separate RTP session. For example, if two participants each send an
audio and a video flow in a single RTP session, that session will
comprise four SSRCs, but if separate RTP sessions had been used for
audio and video, each of those two RTP sessions would comprise only two
SSRCs. Sending multiple media streams in an RTP session hence increases
the amount of cross reporting between the SSRCs, as each SSRC reports on
all other SSRCs in the session. This increases the size of the RTCP
reports, causing them to be sent less often than would be the case if
separate RTP sessions where used for a given RTCP bandwidth.</t>
<t>Finally, when an RTP session contains multiple media types, it is
important to note that the RTCP reception quality reports, feedback
messages, and extended report blocks used might not be applicable to all
media types. Endpoints will need to consider the media type of each SSRC
only send or process reports and feedback that apply to that particular
SSRC and its media type. Signalling solutions might have shortcomings
when it comes to indicating that a particular set of RTCP reports or
feedback messages only apply to a particular media type within an RTP
session.</t>
<t>From an RTCP perspective, therefore, it can be seen that there are
advantages to using separate RTP sessions for each media stream, rather
than sending multiple media streams in a single RTP session. However,
these are frequently offset by the need to reduce port use, to ease
NAT/firewall traversal, achieved by combining media streams into a
single RTP session. The following sections consider some of the issues
with using RTCP in sessions with multiple media streams in more
detail.</t>
<section anchor="sec-timeout-ssrc" title="Timing out SSRCs">
<t>Various issues have been identified with timing out SSRC values
when sending multiple media streams in an RTP session.</t>
<section anchor="sec-avpf-bug"
title="Problems with RTP/AVPF the T_rr_interval Parameter">
<t>The RTP/AVPF profile includes a method to prevent RTCP reports
from being sent too often. This mechanism is described in Section
3.5.3 of <xref target="RFC4585"/>, and is controlled by the
T_rr_interval parameter. It works as follows. When a regular RTCP
report is sent, a new random value, T_rr_current_interval, is
generated, drawn evenly in the range 0.5 to 1.5 times T_rr_interval.
If a regular RTCP packet is to be sent earlier then
T_rr_current_interval seconds after the previous regular RTCP
packet, and there are no feedback messages to be sent, then that
regular RTCP packet is suppressed, and the next regular RTCP packet
is scheduled. The T_rr_current_interval is recalculated each time a
regular RTCP packet is sent. The benefit of suppression is that it
avoids wasting bandwidth when there is nothing requiring frequent
RTCP transmissions, but still allows utilization of the configured
bandwidth when feedback is needed.</t>
<t>Unfortunately this suppression mechanism skews the distribution
of the RTCP sending intervals compared to the regular RTCP reporting
intervals. The standard RTCP timing rules, including reconsideration
and the compensation factor, result in the intervals between sending
RTCP packets having a distribution that is skewed towards the upper
end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the
deterministic calculated RTCP reporting interval. With Td = 5s this
distribution covers the range [2.052s, 6.156s]. In comparison, the
RTP/AVPF suppression rules act in an interval that is 0.5 to 1.5
times T_rr_interval; for T_rr_interval = 5s this is [2.5s,
7.5s].</t>
<t>The effect of this is that the time between consecutive RTCP
packets when using T_rr_interval suppression can become large. The
maximum time interval between sending one regular RTCP packet and
the next, when T_rr_interval is being used, occurs when
T_rr_current_interval takes its maximum value and a regular RTCP
packet is suppressed at the end of the suppression period, then the
next regular RTCP packet is scheduled after its largest possible
reporting interval. Taking the worst case of the two intervals gives
a maximum time between two RTCP reports of 1.5*T_rr_interval +
1.5/1.21828*Td.</t>
<t>This behaviour can be surprising when Td and T_rr_interval have
the same value. That is, when T_rr_interval is configured to match
the regular RTCP reporting interval. In this case, one might expect
that regular RTCP packets are sent according to their usual
schedule, but feedback packets can be sent early. However, the
above-mentioned issue results in the RTCP packets actually being
sent in the range [0.5*Td, 2.731*Td] with a highly non-uniform
distribution, rather than the range [0.41*Td, 1.23*Td]. This is
perhaps unexpected, but is not a problem in itself. However, when
coupled with packet loss, it raises the issue of premature
timeout.</t>
</section>
<section anchor="sec-timeout" title="Avoiding Premature Timeout">
<t>In <xref target="RFC3550">RTP/AVP</xref> the timeout behaviour is
simple, and is 5 times Td, where Td is calculated with a Tmin value
of 5 seconds. In other words, if the configured RTCP bandwidth
allows for an average RTCP reporting interval shorter than 5
seconds, the timeout is 25 seconds of no activity from the SSRC (RTP
or RTCP), otherwise the timeout is 5 average reporting
intervals.</t>
<t><xref target="RFC4585">RTP/AVPF</xref> introduces different
timeout behaviours depending on the value of T_rr_interval. When
T_rr_interval is 0, it uses the same timeout calculation as RTP/AVP.
However, when T_rr_interval is non-zero, it replaces Tmin in the
timeout calculation, most likely to speed up detection of timed out
SSRCs. However, using a non-zero T_rr_interval has two consequences
for RTP behaviour.</t>
<t>First, due to suppression, the number of RTP and RTCP packets
sent by an SSRC that is not an active RTP sender can become very
low, because of the issue discussed in <xref
target="sec-avpf-bug"/>. As the RTCP packet interval can be as long
as 2.73*Td, then during a 5*Td time period an endpoint might in fact
transmit only a single RTCP packet. The long intervals result in
fewer RTCP packets, to a point where a single RTCP packet loss can
sometimes result in timing out an SSRC.</t>
<t>Second, the RTP/AVPF changes to the timeout rules reduce
robustness to misconfiguration. It is common to use RTP/AVPF
configured such that RTCP packets can be sent frequently, to allow
rapid feedback, however this makes timeouts very sensitive to
T_rr_interval. For example, if two SSRCs are configured one with
T_rr_interval = 0.1s and the other with T_rr_interval = 0.6s, then
this small difference will result in the SSRC with the shorter
T_rr_interval timing out the other if it stops sending RTP packets,
since the other RTCP reporting interval is more than five times its
own. When RTP/AVP is used, or RTP/AVPF with T_rr_interval = 0, this
is a non-issue, as the timeout period will be 25s, and differences
between configured RTCP bandwidth can only cause premature timeouts
when the reporting intervals are greater than 5s and differ by a
factor of five. To limit the scope for such problematic
misconfiguration, we propose an update to the RTP/AVPF timeout rules
in <xref target="sec-rtcp-timeout-spec"/>.</t>
</section>
<section title="Interoperability Between RTP/AVP and RTP/AVPF">
<t>If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or
their secure variants) are combined within a single RTP session, and
the RTP/AVPF endpoints use a non-zero T_rr_interval that is
significantly below 5 seconds, there is a risk that the RTP/AVPF
endpoints will prematurely timeout the SSRCs of the RTP/AVP
endpoints, due to their different RTCP timeout rules. Conversely, if
the RTP/AVPF endpoints use a T_rr_interval that is significant
larger than 5 seconds, there is a risk that the RTP/AVP endpoints
will timeout the SSRCs of the RTP/AVPF endpoints.</t>
<t>Mixing endpoints using two different RTP profiles within a single
RTP session is NOT RECOMMENDED. However, if mixed RTP profiles are
used, and the RTP/AVPF endpoints are not updated to follow <xref
target="sec-rtcp-timeout-spec"/> of this memo, then the RTP/AVPF
session SHOULD be configured to use T_rr_interval = 4 seconds to
avoid premature timeouts.</t>
<t>The choice of T_rr_interval = 4 seconds for interoperability
might appear strange. Intuitively, this value ought to be 5 seconds,
to make both the RTP/AVP and RTP/AVPF use the same timeout period.
However, the behaviour outlined in <xref target="sec-avpf-bug"/>
shows that actual RTP/AVPF reporting intervals can be longer than
expected. Setting T_rr_interval = 4 seconds gives actual RTCP
intervals near to those expected by RTP/AVP, ensuring
interoperability.</t>
</section>
<section anchor="sec-rtcp-timeout-spec"
title="Updated SSRC Timeout Rules">
<t>To ensure interoperability and avoid premature timeouts, all
SSRCs in an RTP session MUST use the same timeout behaviour.
However, previous specification are inconsistent in this regard. To
avoid interoperability issues, this memo updates the timeout rules
as follows: <list style="symbols">
<t>For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
the timeout interval SHALL be calculated using a multiplier of
five times the deterministic RTCP reporting interval. That is,
the timeout interval SHALL be 5*Td.</t>
<t>For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles,
calculation of Td, for the purpose of calculating the
participant timeout only, SHALL be done using a Tmin value of 5
seconds and not the reduced minimal interval, even if the
reduced minimum interval is used to calculate RTCP packet
transmission intervals.</t>
</list> This changes the behaviour for the RTP/AVPF or RTP/SAVPF
profiles when T_rr_interval != 0, a behaviour defined in Section
3.5.4 of RFC 4585, i.e. Tmin in the Td calculation is the
T_rr_interval.</t>
</section>
</section>
<section anchor="sec-rtcp-tuning" title="Tuning RTCP transmissions">
<t>This sub-section discusses what tuning can be done to reduce the
downsides of the shared RTCP packet intervals. First, it is considered
what possibilities exist for the <xref target="RFC3551">RTP/AVP</xref>
profile, then what additional tools are provided by <xref
target="RFC4585">RTP/AVPF</xref>.</t>
<section title="RTP/AVP and RTP/SAVP">
<t>When using the RTP/AVP or RTP/SAVP profiles, the options for
tuning the RTCP reporting intervals are limited to the RTCP sender
and receiver bandwidth, and whether the minimum RTCP interval is
scaled according to the bandwidth. As the scheduling algorithm
includes both randomisation and reconsideration, one cannot simply
calculate the expected average transmission interval using the
formula for Td given in Section 6.3.1 of <xref target="RFC3550"/>.
However, by considering the inputs to that expression, and the
randomisation and reconsideration rules, we can begin to understand
the behaviour of the RTCP transmission interval.</t>
<t>Let's start with some basic observations:<list style="letters">
<t>Unless the scaled minimum RTCP interval is used, then Td
prior to randomization and reconsideration can never be less
than Tmin. The default value of Tmin is 5 seconds.</t>
<t>If the scaled minimum RTCP interval is used, Td can become as
low as 360 divided by RTP Session bandwidth in kilobits per
second. In SDP the RTP session bandwidth is signalled using a
"b=AS" line. An RTP Session bandwidth of 72kbps results in Tmin
being 5 seconds. An RTP session bandwidth of 360kbps of course
gives a Tmin of 1 second, and to achieve a Tmin equal to once
every frame for a 25 frame-per-second video stream requires an
RTP session bandwidth of 9Mbps. Use of the RTP/AVPF or RTP/SAVPF
profile allows more frequent RTCP reports for the same
bandwidth, as discussed below.</t>
<t>The value of Td scales with the number of SSRCs and the
average size of the RTCP reports, to keep the overall RTCP
bandwidth constant.</t>
<t>The actual transmission interval for a Td value is in the
range [0.5*Td/1.21828,1.5*Td/1.21828], and the distribution is
skewed, due to reconsideration, with the majority of the
probability mass being above Td. This means, for example, that
for Td = 5s, the actual transmission interval will be
distributed in the range [2.052s, 6.156s], and tending towards
the upper half of the interval. Note that Tmin parameter limits
the value of Td before randomisation and reconsideration are
applied, so the actual transmission interval will cover a range
extending below Tmin.</t>
</list></t>
<t>Given the above, we can calculate the number of SSRCs, n, that an
RTP session with 5% of the session bandwidth assigned to RTCP can
support while maintaining Td equal to Tmin. This will tell us how
many media streams we can report on, keeping the RTCP overhead
within acceptable bounds. We make two assumptions that simplify the
calculation: that all SSRCs are senders, and that they all send
compound RTCP packets comprising an SR packet with n-1 report
blocks, followed by an SDES packet containing a 16 octet CNAME value
<xref target="RFC7022"/> (such RTCP packets will vary in size
between 54 and 798 octets depending on n, up to the maximum of 31
report blocks that can be included in an SR packet). If we put this
packet size, and a 5% RTCP bandwidth fraction into the RTCP interval
calculation in Section 6.3.1 of <xref target="RFC3550"/>, and
calculate the value of n needed to give Td = Tmin for the scaled
minimum interval, we find n=9 SSRCs can be supported (irrespective
of the interval, due to the way the reporting interval scales with
the session bandwidth). We see that to support more SSRCs, we need
to increase the RTCP bandwidth fraction from 5%; changing the
session bandwidth does not help due to the limit of Tmin.</t>
<!-- csp: see page 1593 of notes for the above calculation -->
<t>To conclude, with RTP/AVP and RTP/SAVP the key limitation for
small unicast sessions is going to be the Tmin value. Thus the RTP
session bandwidth configured in RTCP has to be sufficiently high to
reach the reporting goals the application has following the rules
for the scaled minimal RTCP interval.</t>
</section>
<section title="RTP/AVPF and RTP/SAVPF">
<t>When using RTP/AVPF or RTP/SAVPF, we have a powerful additional
tool for tuning RTCP transmissions: the T_rr_interval parameter. Use
of this parameter allows short RTCP reporting intervals;
alternatively it gives the ability to sent frequent RTCP feedback
without sending frequent regular RTCP reports.</t>
<t>The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval
set to a value greater than zero but smaller than Tmin allows more
frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a
given RTCP bandwidth. This happens because Tmin is set to zero after
the transmission of the initial RTCP report, causing the reporting
interval for later packet to be determined by the usual RTCP
bandwidth-based calculation, with Tmin=0, and the T_rr_interval.
This has the effect that we are no longer restricted by the minimal
interval (whether the default 5 second minimum, or the reduced
minimum interval). Rather, the RTCP bandwidth and the T_rr_interval
are the governing factors, allowing faster feedback. Applications
that care about rapid regular RTCP feedback ought to consider using
the RTP/AVPF or RTP/SAVPF profile, even if they don't use the
feedback features of that profile.</t>
<t>The use of the RTP/AVPF or RTP/SAVPF profile allows RTCP feedback
packets to be sent frequently, without also requiring regular RTCP
reports to be sent frequently, since T_rr_interval limits the rate
at which regular RTCP packets can be sent, while still permitting
RTCP feedback packets to be sent. Applications that can use feedback
packets for some media streams, e.g., video streams, but don't want
frequent regular reporting for other media streams, can 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. They could then use feedback packets, which will include RTCP
SR/RR packets unless reduced size RTCP feedback packets <xref
target="RFC5506"/> are used, for the video reporting. This allows
the available RTCP bandwidth to be devoted on the feedback that
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. Indeed, it might make this
choice even more important, as this is more likely to affect the
RTCP behaviour and performance than when using the RTP/AVP or
RTP/SAVP profile, as there are fewer limitations affecting the RTCP
transmission.</t>
<t>When T_rr_interval is non-zero, there are configurations that
need to be avoided. If the RTCP bandwidth chosen is such that the Td
value is smaller than, but close to, T_rr_interval, then the actual
regular RTCP packet transmission interval can become very large, as
discussed in <xref target="sec-avpf-bug"/>. 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 the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0
has utility, and results in a behaviour where the RTCP transmission
is only limited by the bandwidth, i.e., no Tmin limitations at all.
This allows more frequent regular RTCP reporting than can be
achieved using the RTP/AVP profile. Many configurations of RTCP will
not consume all the bandwidth that they have been configured to use,
but this configuration will consume what it has been given. Note
that the same behaviour will be achieved as long as T_rr_interval is
smaller than 1/3 of Td as that prevents T_rr_interval from affecting
the transmission.</t>
<t>There exists no method for using different regular RTCP reporting
intervals depending on the media type or individual media stream,
other than using a separate RTP session for each type or stream.</t>
</section>
</section>
</section>
<section anchor="security" title="Security Considerations">
<t>When using the secure RTP protocol (RTP/SAVP) <xref
target="RFC3711"/>, or the secure variant of the feedback profile
(RTP/SAVPF) <xref target="RFC5124"/>, the cryptographic context of a
compound secure RTCP packet is the SSRC of the sender of the first RTCP
(sub-)packet. This could matter in some cases, especially for keying
mechanisms such as <xref target="RFC3830">Mikey</xref> which allow use
of per-SSRC keying.</t>
<t>Otherwise, the standard security considerations of RTP apply; sending
multiple media streams from a single endpoint in a single RTP session
does not appear to have different security consequences than sending the
same number of media streams spread across different RTP sessions.</t>
</section>
<section anchor="iana" title="IANA Considerations">
<t>No IANA actions are needed.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include='reference.RFC.2119'?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.3711'?>
<?rfc include='reference.RFC.4585'?>
<?rfc include='reference.RFC.5124'?>
<?rfc include='reference.RFC.5506'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.3551'?>
<?rfc include='reference.RFC.3830'?>
<?rfc include='reference.RFC.4588'?>
<?rfc include='reference.RFC.5104'?>
<?rfc include='reference.RFC.6190'?>
<?rfc include='reference.RFC.7022'?>
<?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-clue-framework'?>
<?rfc include='reference.I-D.ietf-avtcore-multi-media-rtp-session'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-multi-stream-optimisation'?>
</references>
</back>
</rfc>
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