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<front> <front>
<title abbrev="RTP Media Congestion Control Requirements ">Congestion <title abbrev="RTP Media Congestion Control Requirements">Congestion Co
Control Requirements for Interactive Real-Time Media</title> ntrol Requirements for Interactive Real-Time Media</title>
<seriesInfo name="RFC" value="8836" stream="IETF"/>
<author fullname="Randell Jesup" initials="R." surname="Jesup"> <author fullname="Randell Jesup" initials="R." surname="Jesup">
<organization>Mozilla</organization> <organization showOnFrontPage="true">Mozilla</organization>
<address> <address>
<postal> <postal>
<street></street> <street/>
<country>United States of America</country>
<country>USA</country>
</postal> </postal>
<email>randell-ietf@jesup.org</email> <email>randell-ietf@jesup.org</email>
</address> </address>
</author> </author>
<author fullname="Zaheduzzaman Sarker" initials="Z." role="editor" surname="
<author fullname="Zaheduzzaman Sarker" initials="Z." role="editor" Sarker">
surname="Sarker"> <organization showOnFrontPage="true">Ericsson AB</organization>
<organization>Ericsson</organization>
<address> <address>
<postal> <postal>
<street></street> <street>Torshamnsgatan 23</street>
<city>Stockholm</city>
<city></city> <region/>
<code>164 83</code>
<region></region>
<code></code>
<country>Sweden</country> <country>Sweden</country>
</postal> </postal>
<phone>+46 10 717 37 43</phone>
<phone></phone>
<facsimile></facsimile>
<email>zaheduzzaman.sarker@ericsson.com</email> <email>zaheduzzaman.sarker@ericsson.com</email>
<uri></uri>
</address> </address>
</author> </author>
<date month="01" year="2021"/>
<date /> <keyword>Interactive multimedia</keyword>
<keyword>webrtc</keyword>
<abstract> <keyword>video communication</keyword>
<t>Congestion control is needed for all data transported across the <keyword>RTP/RTCP</keyword>
<abstract pn="section-abstract">
<t indent="0" pn="section-abstract-1">Congestion control is needed for all
data transported across the
Internet, in order to promote fair usage and prevent congestion Internet, in order to promote fair usage and prevent congestion
collapse. The requirements for interactive, point-to-point real-time collapse. The requirements for interactive, point-to-point real-time
multimedia, which needs low-delay, semi-reliable data delivery, are multimedia, which needs low-delay, semi-reliable data delivery, are
different from the requirements for bulk transfer like FTP or bursty different from the requirements for bulk transfer like FTP or bursty
transfers like Web pages. Due to an increasing amount of RTP-based transfers like web pages. Due to an increasing amount of RTP-based
real-time media traffic on the Internet (e.g. with the introduction of real-time media traffic on the Internet (e.g., with the introduction of
the Web Real-Time Communication (WebRTC)), it is especially important to the Web Real-Time Communication (WebRTC)), it is especially important to
ensure that this kind of traffic is congestion controlled.</t> ensure that this kind of traffic is congestion controlled.</t>
<t indent="0" pn="section-abstract-2">This document describes a set of req
<t>This document describes a set of requirements that can be used to uirements that can be used to
evaluate other congestion control mechanisms in order to figure out evaluate other congestion control mechanisms in order to figure out
their fitness for this purpose, and in particular to provide a set of their fitness for this purpose, and in particular to provide a set of
possible requirements for real-time media congestion avoidance possible requirements for a real-time media congestion avoidance
technique.</t> technique.</t>
</abstract> </abstract>
<boilerplate>
<note title="Requirements Language"> <section anchor="status-of-memo" numbered="false" removeInRFC="false" toc=
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "exclude" pn="section-boilerplate.1">
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this <name slugifiedName="name-status-of-this-memo">Status of This Memo</name
document are to be interpreted as described in <xref >
target="RFC2119">RFC 2119</xref>. The terms are presented in many cases <t indent="0" pn="section-boilerplate.1-1">
using lowercase for readability.</t> This document is not an Internet Standards Track specification; it i
</note> s
published for informational purposes.
</t>
<t indent="0" pn="section-boilerplate.1-2">
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are candidates for any level of Internet
Standard; see Section 2 of RFC 7841.
</t>
<t indent="0" pn="section-boilerplate.1-3">
Information about the current status of this document, any
errata, and how to provide feedback on it may be obtained at
<eref target="https://www.rfc-editor.org/info/rfc8836" brackets="non
e"/>.
</t>
</section>
<section anchor="copyright" numbered="false" removeInRFC="false" toc="excl
ude" pn="section-boilerplate.2">
<name slugifiedName="name-copyright-notice">Copyright Notice</name>
<t indent="0" pn="section-boilerplate.2-1">
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
</t>
<t indent="0" pn="section-boilerplate.2-2">
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(<eref target="https://trustee.ietf.org/license-info" brackets="none
"/>) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with
respect to this document. Code Components extracted from this
document must include Simplified BSD License text as described in
Section 4.e of the Trust Legal Provisions and are provided without
warranty as described in the Simplified BSD License.
</t>
</section>
</boilerplate>
<toc>
<section anchor="toc" numbered="false" removeInRFC="false" toc="exclude" p
n="section-toc.1">
<name slugifiedName="name-table-of-contents">Table of Contents</name>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="section-to
c.1-1">
<li pn="section-toc.1-1.1">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.1.1"><xref der
ivedContent="1" format="counter" sectionFormat="of" target="section-1"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-introduction">
Introduction</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.1.2">
<li pn="section-toc.1-1.1.2.1">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.1.2.1.1"><
xref derivedContent="1.1" format="counter" sectionFormat="of" target="section-1.
1"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-re
quirements-language">Requirements Language</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.2">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.2.1"><xref der
ivedContent="2" format="counter" sectionFormat="of" target="section-2"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-requirements">
Requirements</xref></t>
</li>
<li pn="section-toc.1-1.3">
<t indent="0" pn="section-toc.1-1.3.1"><xref derivedContent="3" form
at="counter" sectionFormat="of" target="section-3"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-deficiencies-of-existing-me">Defic
iencies of Existing Mechanisms</xref></t>
</li>
<li pn="section-toc.1-1.4">
<t indent="0" pn="section-toc.1-1.4.1"><xref derivedContent="4" form
at="counter" sectionFormat="of" target="section-4"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-iana-considerations">IANA Consider
ations</xref></t>
</li>
<li pn="section-toc.1-1.5">
<t indent="0" pn="section-toc.1-1.5.1"><xref derivedContent="5" form
at="counter" sectionFormat="of" target="section-5"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-security-considerations">Security
Considerations</xref></t>
</li>
<li pn="section-toc.1-1.6">
<t indent="0" pn="section-toc.1-1.6.1"><xref derivedContent="6" form
at="counter" sectionFormat="of" target="section-6"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-references">References</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.6.2">
<li pn="section-toc.1-1.6.2.1">
<t indent="0" pn="section-toc.1-1.6.2.1.1"><xref derivedContent=
"6.1" format="counter" sectionFormat="of" target="section-6.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-normative-references">
Normative References</xref></t>
</li>
<li pn="section-toc.1-1.6.2.2">
<t indent="0" pn="section-toc.1-1.6.2.2.1"><xref derivedContent=
"6.2" format="counter" sectionFormat="of" target="section-6.2"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-informative-references
">Informative References</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.7">
<t indent="0" pn="section-toc.1-1.7.1"><xref derivedContent="" forma
t="none" sectionFormat="of" target="section-appendix.a"/><xref derivedContent=""
format="title" sectionFormat="of" target="name-acknowledgements">Acknowledgemen
ts</xref></t>
</li>
<li pn="section-toc.1-1.8">
<t indent="0" pn="section-toc.1-1.8.1"><xref derivedContent="" forma
t="none" sectionFormat="of" target="section-appendix.b"/><xref derivedContent=""
format="title" sectionFormat="of" target="name-authors-addresses">Authors' Addr
esses</xref></t>
</li>
</ul>
</section>
</toc>
</front> </front>
<middle> <middle>
<section title="Introduction"> <section numbered="true" toc="include" removeInRFC="false" pn="section-1">
<t>Most of today's TCP congestion control schemes were developed with a <name slugifiedName="name-introduction">Introduction</name>
focus on an use of the Internet for reliable bulk transfer of <t indent="0" pn="section-1-1">Most of today's TCP congestion control sche
mes were developed with a
focus on a use of the Internet for reliable bulk transfer of
non-time-critical data, such as transfer of large files. They have also non-time-critical data, such as transfer of large files. They have also
been used successfully to govern the reliable transfer of smaller chunks been used successfully to govern the reliable transfer of smaller chunks
of data in as short a time as possible, such as when fetching Web of data in as short a time as possible, such as when fetching web
pages.</t> pages.</t>
<t indent="0" pn="section-1-2">These algorithms have also been used for tr
<t>These algorithms have also been used for transfer of media streams ansfer of media streams
that are viewed in a non-interactive manner, such as "streaming" video, that are viewed in a non-interactive manner, such as "streaming" video,
where having the data ready when the viewer wants it is important, but where having the data ready when the viewer wants it is important, but
the exact timing of the delivery is not.</t> the exact timing of the delivery is not.</t>
<t indent="0" pn="section-1-3">When handling real-time interactive media,
<t>When doing real-time interactive media, the requirements are the requirements are
different; one needs to provide the data continuously, within a very different. One needs to provide the data continuously, within a very
limited time window (no more than 100s of milliseconds end-to-end limited time window (no more delay than hundreds of milliseconds
delay), the sources of data may be able to adapt the amount of data that end-to-end). In addition, the sources of data may be able to adapt the
needs sending within fairly wide margins but can be rate limited by the amount of data that needs sending within fairly wide margins, but they can
application- even not always have data to send, and may tolerate some be rate limited by the
amount of packet loss, but since the data is generated in real-time, application -- even not always having data to send. They may tolerate some
amount of packet loss, but since the data is generated in real time,
sending "future" data is impossible, and since it's consumed in sending "future" data is impossible, and since it's consumed in
real-time, data delivered late is commonly useless.</t> real time, data delivered late is commonly useless.</t>
<t indent="0" pn="section-1-4">While the requirements for real-time intera
<t>While the requirements for real-time interactive media differ from ctive media differ from
the requirements for the other flow types, these other flow types will the requirements for the other flow types, these other flow types will
be present in the network. The congestion control algorithm for be present in the network. The congestion control algorithm for
real-time interactive media must work properly when these other flow real-time interactive media must work properly when these other flow
types are present as cross traffic on the network.</t> types are present as cross traffic on the network.</t>
<t indent="0" pn="section-1-5">One particular protocol portfolio being dev
<t>One particular protocol portfolio being developed for this use case eloped for this use case
is WebRTC <xref target="I-D.ietf-rtcweb-overview"></xref>, where one is WebRTC <xref target="RFC8825" format="default" sectionFormat="of" deriv
edContent="RFC8825"/>, where one
envisions sending multiple flows using the Real-time Transport Protocol envisions sending multiple flows using the Real-time Transport Protocol
(RTP) <xref target="RFC3550"></xref> between two peers, in conjunction (RTP) <xref target="RFC3550" format="default" sectionFormat="of" derivedCo ntent="RFC3550"/> between two peers, in conjunction
with data flows, all at the same time, without having special with data flows, all at the same time, without having special
arrangements with the intervening service providers. As RTP does not arrangements with the intervening service providers. As RTP does not
provide any congestion control mechanism; a set of circuit breakers, provide any congestion control mechanism, a set of circuit breakers,
such as <xref target="I-D.ietf-avtcore-rtp-circuit-breakers"></xref>, such as those described in <xref target="RFC8083" format="default" section
Format="of" derivedContent="RFC8083"/>,
are required to protect the network from excessive congestion caused by are required to protect the network from excessive congestion caused by
the non-congestion controlled flows. When the real-time interactive non-congestion-controlled flows. When the real-time interactive
media is congestion controlled, it is recommended that the congestion media is congestion controlled, it is recommended that the
control mechanism operates within the constraints defined by these congestion control mechanism operate within the constraints defined by
circuit breakers when circuit breaker is present and that it should not these
cause congestion collapse when circuit breaker is not implemented.</t> circuit breakers when a circuit breaker is present and that it should not
cause congestion collapse when a circuit breaker is not implemented.</t>
<t>Given that this use case is the focus of this document, use cases <t indent="0" pn="section-1-6">Given that this use case is the focus of th
involving non-interactive media such as video streaming, and use cases is document, use cases
involving non-interactive media such as video streaming and those
using multicast/broadcast-type technologies, are out of scope.</t> using multicast/broadcast-type technologies, are out of scope.</t>
<t indent="0" pn="section-1-7">The terminology defined in <xref target="RF
<t>The terminology defined in <xref C8825" format="default" sectionFormat="of" derivedContent="RFC8825"/>
target="I-D.ietf-rtcweb-overview"></xref> is used in this memo.</t> is used in this memo.</t>
<section numbered="true" toc="include" removeInRFC="false" pn="section-1.1
">
<name slugifiedName="name-requirements-language">Requirements Language</
name>
<t indent="0" pn="section-1.1-1">The key words "<bcp14>MUST</bcp14>", "<
bcp14>MUST NOT</bcp14>",
"<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>",
"<bcp14>SHALL NOT</bcp14>", "<bcp14>SHOULD</bcp14>",
"<bcp14>SHOULD NOT</bcp14>",
"<bcp14>RECOMMENDED</bcp14>",
"<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document
are to be interpreted as described in BCP 14
<xref target="RFC2119" format="default" sectionFormat="of" derivedContent
="RFC2119"/>.</t>
</section>
</section> </section>
<section numbered="true" toc="include" removeInRFC="false" pn="section-2">
<section title="Requirements"> <name slugifiedName="name-requirements">Requirements</name>
<t><list style="numbers"> <ol spacing="normal" type="1" indent="adaptive" start="1" pn="section-2-1"
<t>The congestion control algorithm must attempt to provide >
<li pn="section-2-1.1" derivedCounter="1.">
<t indent="0" pn="section-2-1.1.1">The congestion control algorithm <b
cp14>MUST</bcp14> attempt to provide
as-low-as-possible-delay transit for interactive real-time traffic as-low-as-possible-delay transit for interactive real-time traffic
while still providing a useful amount of bandwidth. There may be while still providing a useful amount of bandwidth. There may be
lower limits on the amount of bandwidth that is useful, but this is lower limits on the amount of bandwidth that is useful, but this is
largely application-specific and the application may be able to largely application specific, and the application may be able to
modify or remove flows in order allow some useful flows to get modify or remove flows in order to allow some useful flows to get
enough bandwidth. (Example: not enough bandwidth for low-latency enough bandwidth. For example, although there might not be enough band
video+audio, but enough for audio-only.) <list style="letters"> width
<t>Jitter (variation in the bitrate over short time scales) also for low-latency video+audio, there could be enough for audio only.
is relevant, though moderate amounts of jitter will be absorbed </t>
<ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.1.2">
<li pn="section-2-1.1.2.1" derivedCounter="a.">Jitter (variation in
the bitrate over short timescales) is also
relevant, though moderate amounts of jitter will be absorbed
by jitter buffers. Transit delay should be considered to track by jitter buffers. Transit delay should be considered to track
the short-term maximums of delay including jitter.</t>
<t>It should provide this as-low-as-possible-delay transit and the short-term maximums of delay, including jitter.</li>
<li pn="section-2-1.1.2.2" derivedCounter="b.">The algorithm should
provide this as-low-as-possible-delay transit and
minimize self-induced latency even when faced with intermediate minimize self-induced latency even when faced with intermediate
bottlenecks and competing flows. Competing flows may limit bottlenecks and competing flows. Competing flows may limit
what's possible to achieve.</t> what's possible to achieve.</li>
<li pn="section-2-1.1.2.3" derivedCounter="c.">The algorithm should
<t>It should be resilience to the effects of the events, such as be resilient to the effects of events, such as
routing changes, which may alter or remove bottlenecks or change routing changes, which may alter or remove bottlenecks or change
the bandwidth available especially if there is a reduction in the bandwidth available, especially if there is a reduction in
available bandwidth or increase in observed delay. It is available bandwidth or increase in observed delay. It is
expected that the mechanism reacts quickly to the such events to expected that the mechanism reacts quickly to such events to
avoid delay buildup. In the context of this memo, a 'quick' avoid delay buildup. In the context of this memo, a "quick"
reaction is on the order of a few RTTs, subject to the reaction is on the order of a few RTTs, subject to the
constraints of the media codec, but is likely within a second. constraints of the media codec, but is likely within a second.
Reaction on the next RTT is explicitly not required, since many Reaction on the next RTT is explicitly not required, since many
codecs cannot adapt their sending rate that quickly, but equally codecs cannot adapt their sending rate that quickly, but
response cannot be arbitrarily delayed.</t> at the same time a response cannot be arbitrarily delayed.</li>
<li pn="section-2-1.1.2.4" derivedCounter="d.">The algorithm should
<t>It should react quickly to handle both local and remote react quickly to handle both local and remote
interface changes (WLAN to 3G data, etc) which may radically interface changes (e.g., WLAN to 3G data) that may radically
change the bandwidth available or bottlenecks, especially if change the bandwidth available or bottlenecks, especially if
there is a reduction in available bandwidth or increase in there is a reduction in available bandwidth or an increase in
bottleneck delay. It is assumed that an interface change can bottleneck delay. It is assumed that an interface change can
generate a notification to the algorithm.</t> generate a notification to the algorithm.</li>
<li pn="section-2-1.1.2.5" derivedCounter="e.">The real-time interac
<t>The real-time interactive media applications can be rate tive media applications can be rate
limited. This means the offered loads can be less than the limited. This means the offered loads can be less than the
available bandwidth at any given moment, and may vary available bandwidth at any given moment and may vary
dramatically over time, including dropping to no load and then dramatically over time, including dropping to no load and then
resuming a high load, such as in a mute/unmute operation. Hence, resuming a high load, such as in a mute/unmute operation. Hence,
the algorithm must be designed to handle such behavior from the algorithm must be designed to handle such behavior from
media source or application. Note that the reaction time between a media source or application. Note that the reaction time between
a change in the bandwidth available from the algorithm and a a change in the bandwidth available from the algorithm and a
change in the offered load is variable, and may be different change in the offered load is variable, and it may be different
when increasing versus decreasing.</t> when increasing versus decreasing.</li>
<li pn="section-2-1.1.2.6" derivedCounter="f.">The algorithm is requ
<t>The algorithm requires to avoid building up queues when ired to avoid building up queues when
competing with short-term bursts of traffic (for example, competing with short-term bursts of traffic (for example,
traffic generated by web-browsing) which can quickly saturate a traffic generated by web browsing), which can quickly saturate a
local-bottleneck router or link, but also clear quickly. The local-bottleneck router or link but clear quickly. The
algorithm should also react quickly to regain its previous share algorithm should also react quickly to regain its previous share
of the bandwidth when the local-bottleneck or link is of the bandwidth when the local bottleneck or link is
cleared.</t> cleared.</li>
<li pn="section-2-1.1.2.7" derivedCounter="g.">Similarly, periodic b
<t>Similarly periodic bursty flows such as MPEG DASH <xref ursty flows such as MPEG DASH <xref target="MPEG_DASH" format="default" sectionF
target="MPEG_DASH"></xref> or proprietary media streaming ormat="of" derivedContent="MPEG_DASH"/> or proprietary media
algorithms may compete in bursts with the algorithm, and may not streaming
algorithms may compete in bursts with the algorithm and may not
be adaptive within a burst. They are often layered on top of TCP be adaptive within a burst. They are often layered on top of TCP
but use TCP in a bursty manner that can interact poorly with but use TCP in a bursty manner that can interact poorly with
competing flows during the bursts. The algorithm must not competing flows during the bursts. The algorithm must not
increase the already existing delay buildup during those bursts. increase the already existing delay buildup during those bursts.
Note that this competing traffic may be on a shared access link, Note that this competing traffic may be on a shared access link,
or the traffic burst may cause a shift in the location of the or the traffic burst may cause a shift in the location of the
bottleneck for the duration of the burst.</t> bottleneck for the duration of the burst.</li>
</list></t> </ol>
</li>
<t>The algorithm must be fair to other flows, both real-time flows <li pn="section-2-1.2" derivedCounter="2.">
(such as other instances of itself), and TCP flows, both long-lived <t indent="0" pn="section-2-1.2.1">The algorithm <bcp14>MUST</bcp14> b
and bursts such as the traffic generated by a typical web browsing e fair to other flows, both real-time flows
session. Note that 'fair' is a rather hard-to-define term. It should (such as other instances of itself) and TCP flows, both long-lived flo
be fair with itself, giving fair share of the bandwidth to multiple ws
and bursts such as the traffic generated by a typical web-browsing
session. Note that "fair" is a rather hard-to-define term. It <bcp14>S
HOULD</bcp14>
be fair with itself, giving a fair share of the bandwidth to multiple
flows with similar RTTs, and if possible to multiple flows with flows with similar RTTs, and if possible to multiple flows with
different RTTs.<list style="letters"> different RTTs.
<t>Existing flows at a bottleneck must also be fair to new flows </t>
to that bottleneck, and must allow new flows to ramp up to a <ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.2.2">
<li pn="section-2-1.2.2.1" derivedCounter="a.">Existing flows at a b
ottleneck must also be fair to new flows
to that bottleneck and must allow new flows to ramp up to a
useful share of the bottleneck bandwidth as quickly as possible. useful share of the bottleneck bandwidth as quickly as possible.
A useful share will depend on the media types involved, total A useful share will depend on the media types involved, total
bandwidth available and the user experience requirements of a bandwidth available, and the user-experience requirements of a
particular service. Note that relative RTTs may affect the rate particular service. Note that relative RTTs may affect the rate
new flows can ramp up to a reasonable share.</t> at which new flows can ramp up to a reasonable share.</li>
</list></t> </ol>
</li>
<t>The algorithm should not starve competing TCP flows, and should <li pn="section-2-1.3" derivedCounter="3.">
as best as possible avoid starvation by TCP flows.<list <t indent="0" pn="section-2-1.3.1">The algorithm <bcp14>SHOULD NOT</bc
style="letters"> p14> starve competing TCP flows and <bcp14>SHOULD</bcp14>,
<t>The congestion control should prioritise achieving a useful as best as possible, avoid starvation by TCP flows.</t>
<ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.3.2">
<li pn="section-2-1.3.2.1" derivedCounter="a.">The congestion contro
l should prioritize achieving a useful
share of the bandwidth depending on the media types and total share of the bandwidth depending on the media types and total
available bandwidth over achieving as low as possible transit available bandwidth over achieving as-low-as-possible transit
delay, when these two requirements are in conflict.</t> delay, when these two requirements are in conflict.</li>
</list></t> </ol>
</li>
<t>The algorithm should as quickly as possible adapt to initial <li pn="section-2-1.4" derivedCounter="4.">
network conditions at the start of a flow. This should occur both if <t indent="0" pn="section-2-1.4.1">The algorithm <bcp14>SHOULD</bcp14>
adapt as quickly as possible to initial
network conditions at the start of a flow. This <bcp14>SHOULD</bcp14>
occur whether
the initial bandwidth is above or below the bottleneck bandwidth. the initial bandwidth is above or below the bottleneck bandwidth.
<list style="letters"> </t>
<t>The algorithm should allow different modes of adaptation for <ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
example,the startup adaptation may be faster than adaptation 2-1.4.2">
<li pn="section-2-1.4.2.1" derivedCounter="a.">The algorithm should
allow different modes of adaptation; for
example, the startup adaptation may be faster than adaptation
later in a flow. It should allow for both slow-start operation later in a flow. It should allow for both slow-start operation
(adapt up) and history-based startup (start at a point expected (adapt up) and history-based startup (start at a point expected
to be at or below channel bandwidth from historical information, to be at or below channel bandwidth from historical information,
which may need to adapt down quickly if the initial guess is which may need to adapt down quickly if the initial guess is
wrong). Starting too low and/or adapting up too slowly can cause wrong). Starting too low and/or adapting up too slowly can cause
a critical point in a personal communication to be poor a critical point in a personal communication to be poor
("Hello!"). Starting over-bandwidth causes other problems for ("Hello!").
Starting too high above the available bandwidth causes other probl
ems for
user experience, so there's a tension here. Alternative methods user experience, so there's a tension here. Alternative methods
to help startup like probing during setup with dummy data may be to help startup, such as probing during setup with dummy data, may
useful in some applications; in some cases there will be a be
useful in some applications; in some cases, there will be a
considerable gap in time between flow creation and the initial considerable gap in time between flow creation and the initial
flow of data. Again, A flow may need to change adaptation rates flow of data. Again, a flow may need to change adaptation rates
due to network conditions or changes in the provided flows (such due to network conditions or changes in the provided flows (such
as un-muting or sending data after a gap).</t> as unmuting or sending data after a gap).</li>
</list></t> </ol>
</li>
<t>The algorithm should be stable if the RTP streams are halted or <li pn="section-2-1.5" derivedCounter="5.">
discontinuous (for example - Voice Activity Detection). <list <t indent="0" pn="section-2-1.5.1">The algorithm <bcp14>SHOULD</bcp14>
style="letters"> be stable if the RTP streams are halted or
<t>After stream resumption, the algorithm should attempt to discontinuous (for example, when using Voice Activity Detection). </t>
<ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.5.2">
<li pn="section-2-1.5.2.1" derivedCounter="a.">After stream resumpti
on, the algorithm should attempt to
rapidly regain its previous share of the bandwidth; the rapidly regain its previous share of the bandwidth; the
aggressiveness with which this is done will decay with the aggressiveness with which this is done will decay with the
length of the pause.</t> length of the pause.</li>
</list></t> </ol>
</li>
<t>The algorithm should where possible merge information across <li pn="section-2-1.6" derivedCounter="6.">
multiple RTP streams sent between two endpoints, when those RTP <t indent="0" pn="section-2-1.6.1">Where possible, the algorithm <bcp1
4>SHOULD</bcp14> merge information across
multiple RTP streams sent between two endpoints when those RTP
streams share a common bottleneck, whether or not those streams are streams share a common bottleneck, whether or not those streams are
multiplexed onto the same ports, in order to allow congestion multiplexed onto the same ports. This will allow congestion
control of the set of streams together instead of as multiple control of the set of streams together instead of as multiple
independent streams. This allows better overall bandwidth independent streams. It will also allow better overall bandwidth
management, faster response to changing conditions, and fairer management, faster response to changing conditions, and fairer
sharing of bandwidth with other network users.<list style="letters"> sharing of bandwidth with other network users.</t>
<t>The algorithm should also share information and adaptation <ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.6.2">
<li pn="section-2-1.6.2.1" derivedCounter="a.">The algorithm should
also share information and adaptation
with other non-RTP flows between the same endpoints, such as a with other non-RTP flows between the same endpoints, such as a
WebRTC DataChannel <xref WebRTC data channel <xref target="RFC8831" format="default" sectio
target="I-D.ietf-rtcweb-data-channel"></xref>, when nFormat="of" derivedContent="RFC8831"/>, when
possible.</t> possible.</li>
<li pn="section-2-1.6.2.2" derivedCounter="b.">When there are multip
<t>When there are multiple streams across the same 5-tuple le streams across the same 5-tuple
coordinating their bandwidth use and congestion control, the coordinating their bandwidth use and congestion control, the
algorithm should allow the application to control the relative algorithm should allow the application to control the relative
split of available bandwidth. The most correlated bandwidth split of available bandwidth. The most correlated bandwidth
usage would be with other flows on the same 5-tuple, but there usage would be with other flows on the same 5-tuple, but there
may be use in coordinating measurement and control of the local may be use in coordinating measurement and control of the local
link(s). Use of information about previous flows, especially on link(s). Use of information about previous flows, especially on
the same 5-tuple, may be useful input to the algorithm, the same 5-tuple, may be useful input to the algorithm,
especially to startup performance of a new flow.</t> especially regarding startup performance of a new flow.</li>
</list></t> </ol>
</li>
<t>The algorithm should not require any special support from network <li pn="section-2-1.7" derivedCounter="7.">
elements to convey congestion related information to be functional. <t indent="0" pn="section-2-1.7.1">The algorithm <bcp14>SHOULD NOT</bc
As much as possible, it should leverage available information about p14> require any special support from network
elements to be able to convey congestion-related information.
As much as possible, it <bcp14>SHOULD</bcp14> leverage available infor
mation about
the incoming flow to provide feedback to the sender. Examples of the incoming flow to provide feedback to the sender. Examples of
this information are the packet arrival times, acknowledgements and this information are the packet arrival times, acknowledgements and
feedback, packet timestamps, and packet losses, Explicit Congestion feedback, packet timestamps, packet losses, and Explicit Congestion
Notification (ECN) <xref target="RFC3168"></xref>; all of these can Notification (ECN) <xref target="RFC3168" format="default" sectionForm
at="of" derivedContent="RFC3168"/>; all of these can
provide information about the state of the path and any bottlenecks. provide information about the state of the path and any bottlenecks.
However, the use of available information is algorithm However, the use of available information is algorithm
dependent.<list style="letters"> dependent.</t>
<t>Extra information could be added to the packets to provide <ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.7.2">
<li pn="section-2-1.7.2.1" derivedCounter="a.">Extra information cou
ld be added to the packets to provide
more detailed information on actual send times (as opposed to more detailed information on actual send times (as opposed to
sampling times), but should not be required.</t> sampling times), but such information should not be required.</li>
</list></t> </ol>
</li>
<t>Since the assumption here is a set of RTP streams, the <li pn="section-2-1.8" derivedCounter="8.">
backchannel typically should be done via RTCP<xref <t indent="0" pn="section-2-1.8.1">Since the assumption here is a set
target="RFC3550"></xref>; one alternative would be to include it of RTP streams, the
instead in a reverse RTP channel using header extensions.<list backchannel typically <bcp14>SHOULD</bcp14> be done via the RTP Contro
style="letters"> l Protocol
<t>In order to react sufficiently quickly when using RTCP for a (RTCP) <xref target="RFC3550" format="default" sectionFormat="of" deriv
backchannel, an RTP profile such as RTP/AVPF <xref edContent="RFC3550"/>; instead, one alternative
target="RFC4585"></xref> or RTP/SAVPF <xref would be to include it
target="RFC5124"></xref> that allows sufficiently frequent in a reverse-RTP channel using header extensions.</t>
<ol spacing="normal" type="a" indent="adaptive" start="1" pn="section-
2-1.8.2">
<li pn="section-2-1.8.2.1" derivedCounter="a.">In order to react suf
ficiently quickly when using RTCP for a
backchannel, an RTP profile such as RTP/AVPF <xref target="RFC4585
" format="default" sectionFormat="of" derivedContent="RFC4585"/> or RTP/SAVPF <x
ref target="RFC5124" format="default" sectionFormat="of" derivedContent="RFC5124
"/> that allows sufficiently frequent
feedback must be used. Note that in some cases, backchannel feedback must be used. Note that in some cases, backchannel
messages may be delayed until the RTCP channel can be allocated messages may be delayed until the RTCP channel can be allocated
enough bandwidth, even under AVPF rules. This may also imply enough bandwidth, even under AVPF rules. This may also imply
negotiating a higher maximum percentage for RTCP data or negotiating a higher maximum percentage for RTCP data or
allowing solutions to violate or modify the rules specified for allowing solutions to violate or modify the rules specified for
AVPF.</t> AVPF.</li>
<li pn="section-2-1.8.2.2" derivedCounter="b.">Bandwidth for the fee
<t>Bandwidth for the feedback messages should be minimized (such dback messages should be minimized
as via RFC 5506 <xref target="RFC5506"></xref>to allow RTCP using techniques such as those in <xref target="RFC5506" format="defa
without Sender Reports/Receiver Reports)</t> ult" sectionFormat="of" derivedContent="RFC5506"/>, to allow RTCP
without Sender/Receiver Reports.</li>
<t>Backchannel data should be minimized to avoid taking too much <li pn="section-2-1.8.2.3" derivedCounter="c.">Backchannel data shou
ld be minimized to avoid taking too much
reverse-channel bandwidth (since this will often be used in a reverse-channel bandwidth (since this will often be used in a
bidirectional set of flows). In areas of stability, backchannel bidirectional set of flows). In areas of stability, backchannel
data may be sent more infrequently so long as algorithm data may be sent more infrequently so long as algorithm
stability and fairness are maintained. When the channel is stability and fairness are maintained. When the channel is
unstable or has not yet reached equilibrium after a change, unstable or has not yet reached equilibrium after a change,
backchannel feedback may be more frequent and use more backchannel feedback may be more frequent and use more
reverse-channel bandwidth. This is an area with considerable reverse-channel bandwidth. This is an area with considerable
flexibility of design, and different approaches to backchannel flexibility of design, and different approaches to backchannel
messages and frequency are expected to be evaluated.</t> messages and frequency are expected to be evaluated.</li>
</list></t> </ol>
</li>
<t>Flows managed by this algorithm and flows competing against at a <li pn="section-2-1.9" derivedCounter="9.">
bottleneck may have different DSCP<xref target="RFC5865"></xref> <t indent="0" pn="section-2-1.9.1">Flows managed by this algorithm and
markings depending on the type of traffic, or may be subject to flows competing against each
other at a
bottleneck may have different Differentiated Services Code Point
(DSCP) <xref target="RFC5865" format="default" sectionFormat="of" deriv
edContent="RFC5865"/>
markings depending on the type of traffic or may be subject to
flow-based QoS. A particular bottleneck or section of the network flow-based QoS. A particular bottleneck or section of the network
path may or may not honor DSCP markings. The algorithm should path may or may not honor DSCP markings. The algorithm <bcp14>SHOULD</
attempt to leverage DSCP markings when they're available.<list bcp14>
style="letters"> attempt to leverage DSCP markings when they're available.</t>
<t>In WebRTC, a division of packets into 4 classes is envisioned </li>
in order of priority: faster-than-audio, audio, video, <li pn="section-2-1.10" derivedCounter="10.">The algorithm <bcp14>SHOULD
best-effort, and bulk-transfer. Typically the flows managed by </bcp14> sense the unexpected lack of backchannel
this algorithm would be audio or video in that hierarchy, and information as a possible indication of a channel-overuse problem
feedback flows would be faster-than-audio.</t>
</list></t>
<t>The algorithm should sense the unexpected lack of backchannel
information as a possible indication of a channel overuse problem
and react accordingly to avoid burst events causing a congestion and react accordingly to avoid burst events causing a congestion
collapse.</t> collapse.</li>
<li pn="section-2-1.11" derivedCounter="11.">The algorithm <bcp14>SHOULD
<t>The algorithm should be stable and maintain low-delay when faced </bcp14> be stable and maintain low delay when faced
with Active Queue Management (AQM) algorithms. Also note that these with Active Queue Management (AQM) algorithms. Also note that these
algorithms may apply across multiple queues in the bottleneck, or to algorithms may apply across multiple queues in the bottleneck or to
a single queue</t> a single queue.</li>
</list></t> </ol>
</section> </section>
<section numbered="true" toc="include" removeInRFC="false" pn="section-3">
<section title="Deficiencies of existing mechanisms "> <name slugifiedName="name-deficiencies-of-existing-me">Deficiencies of Exi
<t>Among the existing congestion control mechanisms TCP Friendly Rate sting Mechanisms</name>
Control (TFRC) <xref target="RFC5348"></xref> is the one which claims to <t indent="0" pn="section-3-1">Among the existing congestion control mecha
be suitable for real-time interactive media. TFRC is, an equation based, nisms, TCP Friendly Rate
congestion control mechanism which provides reasonably fair share of the Control (TFRC) <xref target="RFC5348" format="default" sectionFormat="of"
derivedContent="RFC5348"/> is the one that claims to
be suitable for real-time interactive media. TFRC is an equation-based
congestion control mechanism that provides a reasonably fair share of
bandwidth when competing with TCP flows and offers much lower throughput bandwidth when competing with TCP flows and offers much lower throughput
variations than TCP. This is achieved by a slower response to the variations than TCP. This is achieved by a slower response to the
available bandwidth change than TCP. TFRC is designed to perform best available bandwidth change than TCP. TFRC is designed to perform best
with applications which has fixed packet size and does not have fixed with applications that have a fixed packet size and do not have a fixed
period between sending packets.</t> period between sending packets.</t>
<t indent="0" pn="section-3-2">TFRC detects loss events and reacts to cong
<t>TFRC operates on detecting loss events and reacts to loss caused by estion-caused loss by
congestion by reducing its sending rate. It allows applications to reducing its sending rate. It allows applications to
increase the sending rate until loss is observed in the flows. As it is increase the sending rate until loss is observed in the flows. As
noted in IAB/IRTF report <xref target="RFC7295"></xref> large buffers noted in IAB/IRTF report <xref target="RFC7295" format="default" sectionFo
are available in the network elements which introduces additional delay rmat="of" derivedContent="RFC7295"/>, large buffers
in the communication, it becomes important to take all possible are available in the network elements, which introduce additional delay
congestion indications into considerations. Looking at the current in the communication. It becomes important to take all possible
Internet deployment, TFRC's only consideration of loss events as congestion indications into consideration. Looking at the current
congestion indication can be considered as biggest lacking.</t> Internet deployment, TFRC's biggest deficiency is that it only considers
loss events as a congestion indication.
<t>A typical real-time interactive communication includes live encoded </t>
audio and video flow(s). In such a communication scenario an audio <t indent="0" pn="section-3-3">A typical real-time interactive communicati
source typically needs fixed interval between packets, needs to vary on includes live-encoded
their segment size instead of their packet rate in response to audio and video flow(s). In such a communication scenario, an audio
congestion and sends smaller packets, a variant of TFRC , Small-Packet source typically needs a fixed interval between packets and needs to
TFRC (TFRC-SP) <xref target="RFC4828"></xref> addresses the issues vary the segment size of the packets instead of the packet rate in
related to such kind of sources ; a video source generally varies video response to congestion; therefore, it sends smaller packets.
frame sizes, can produce large frames which need to be further A variant of TFRC, Small-Packet
TFRC (TFRC-SP) <xref target="RFC4828" format="default" sectionFormat="of"
derivedContent="RFC4828"/>, addresses the issues
related to such kind of sources. A video source generally varies video
frame sizes, can produce large frames that need to be further
fragmented to fit into path Maximum Transmission Unit (MTU) size, and fragmented to fit into path Maximum Transmission Unit (MTU) size, and
have almost fixed interval between producing frames under a certain has an almost fixed interval between producing frames under a certain
frame rate, TFRC is known to be less optimal when using with such video frame rate. TFRC is known to be less optimal when using such video
sources.</t> sources.</t>
<t indent="0" pn="section-3-4">There are also some mismatches between TFRC
<t>There are also some mismatches between TFRC's design assumptions and 's design assumptions and
how the media sources in a typical real-time interactive application how the media sources in a typical real-time interactive application
works. TFRC is design to maintain smooth sending rate however media work. TFRC is designed to maintain a smooth sending rate; however, media
sources can change rates in steps for both rate increase and rate sources can change rates in steps for both rate increase and rate
decrease. TFRC can operate in two modes - i) Bytes per second and ii) decrease. TFRC can operate in two modes: i) bytes per second and ii)
packets per second, where typical real-time interactive media sources packets per second, where typical real-time interactive media sources
operates on bit per second. There are also limitations on how quickly operate on bit per second. There are also limitations on how quickly
the media sources can adapt to specific sending rates. The modern video the media sources can adapt to specific sending rates. Modern video
encoders can operate on a mode where they can vary the output bitrate a encoders can operate in a mode in which they can vary the output bitrate a
lot depending on the way there are configured, the current scene it is lot depending on the way they are configured, the current scene they are
encoding and more. Therefore, it is possible that the video source does encoding, and more. Therefore, it is possible that the video source will
not always output at a bitrate they are allowed to. TFRC tries to raise not always output at an allowable bitrate. TFRC tries to increase
its sending rate when transmitting at maximum allowed rate and increases its sending rate when transmitting at the maximum allowed rate, and it inc
only twice the current transmission rate hence it may create issues when reases
the video source vary their bitrates.</t> only twice the current transmission rate; hence, it may create issues when
the video sources vary their bitrates.</t>
<t>Moreover, there are number of studies on TFRC which shows it's <t indent="0" pn="section-3-5">Moreover, there are a number of studies on
limitations which includes TFRC's unfairness on low statistically TFRC that show its
multiplexed links, oscillatory behavior, performance issue in highly limitations, including TFRC's unfairness to low statistically
dynamic loss rates conditions and more <xref target="CH09"></xref>.</t> multiplexed links, oscillatory behavior, performance issues in highly
dynamic loss-rate conditions, and more <xref target="CH09" format="default
<t>Looking at all these deficiencies it can be concluded that the " sectionFormat="of" derivedContent="CH09"/>.</t>
requirements of congestion control mechanism for real-time interactive <t indent="0" pn="section-3-6">Looking at all these deficiencies, it can b
e concluded that the
requirements for a congestion control mechanism for real-time interactive
media cannot be met by TFRC as defined in the standard.</t> media cannot be met by TFRC as defined in the standard.</t>
</section> </section>
<section anchor="IANA" numbered="true" toc="include" removeInRFC="false" pn=
<section anchor="IANA" title="IANA Considerations"> "section-4">
<t>This document makes no request of IANA.</t> <name slugifiedName="name-iana-considerations">IANA Considerations</name>
<t indent="0" pn="section-4-1">This document has no IANA actions.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section> </section>
<section anchor="Security" numbered="true" toc="include" removeInRFC="false"
<section anchor="Security" title="Security Considerations"> pn="section-5">
<t>An attacker with the ability to delete, delay or insert messages in <name slugifiedName="name-security-considerations">Security Considerations
</name>
<t indent="0" pn="section-5-1">An attacker with the ability to delete, del
ay, or insert messages into
the flow can fake congestion signals, unless they are passed on a the flow can fake congestion signals, unless they are passed on a
tamper-proof path. Since some possible algorithms depend on the timing tamper-proof path. Since some possible algorithms depend on the timing
of packet arrival, even a traditional protected channel does not fully of packet arrival, even a traditional, protected channel does not fully
mitigate such attacks.</t> mitigate such attacks.</t>
<t indent="0" pn="section-5-2">An attack that reduces bandwidth is not nec
<t>An attack that reduces bandwidth is not necessarily significant, essarily significant,
since an on-path attacker could break the connection by discarding all since an on-path attacker could break the connection by discarding all
packets. Attacks that increase the perceived available bandwidth are packets. Attacks that increase the perceived available bandwidth are
conceivable, and need to be evaluated. Such attacks could result in conceivable and need to be evaluated. Such attacks could result in
starvation of competing flows and permit amplification attacks.</t> starvation of competing flows and permit amplification attacks.</t>
<t indent="0" pn="section-5-3">Algorithm designers should consider the pos
<t>Algorithm designers should consider the possibility of malicious sibility of malicious
on-path attackers.</t> on-path attackers.</t>
</section> </section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>This document is the result of discussions in various fora of the
WebRTC effort, in particular on the rtp-congestion@alvestrand.no mailing
list. Many people contributed their thoughts to this.</t>
</section>
</middle> </middle>
<back> <back>
<references title="Normative References"> <references pn="section-6">
<?rfc include="reference.RFC.2119"?> <name slugifiedName="name-references">References</name>
<references pn="section-6.1">
<?rfc include='reference.RFC.3550'?> <name slugifiedName="name-normative-references">Normative References</na
me>
<?rfc include='reference.RFC.4585'?> <reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2
119" quoteTitle="true" derivedAnchor="RFC2119">
<?rfc include='reference.RFC.5124'?> <front>
<title>Key words for use in RFCs to Indicate Requirement Levels</tit
<?rfc include='reference.I-D.ietf-rtcweb-overview'?> le>
</references> <author initials="S." surname="Bradner" fullname="S. Bradner">
<organization showOnFrontPage="true"/>
<references title="Informative References"> </author>
<?rfc include='reference.RFC.3168'?> <date year="1997" month="March"/>
<abstract>
<?rfc include='reference.RFC.5506'?> <t indent="0">In many standards track documents several words are
used to signify the requirements in the specification. These words are often ca
<?rfc include='reference.RFC.5865'?> pitalized. This document defines these words as they should be interpreted in IE
TF documents. This document specifies an Internet Best Current Practices for th
<?rfc include='reference.RFC.5348'?> e Internet Community, and requests discussion and suggestions for improvements.<
/t>
<?rfc include='reference.RFC.4828'?> </abstract>
</front>
<?rfc include='reference.RFC.7295'?> <seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="2119"/>
<?rfc include='reference.I-D.ietf-rtcweb-data-channel'?> <seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>
<?rfc include='reference.I-D.ietf-avtcore-rtp-circuit-breakers'?> <reference anchor="RFC3550" target="https://www.rfc-editor.org/info/rfc3
550" quoteTitle="true" derivedAnchor="RFC3550">
<reference anchor="MPEG_DASH"> <front>
<front> <title>RTP: A Transport Protocol for Real-Time Applications</title>
<title>Dynamic adaptive streaming over HTTP (DASH) -- Part 1: Media <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne
presentation description and segment formats</title> ">
<organization showOnFrontPage="true"/>
<author></author> </author>
<author initials="S." surname="Casner" fullname="S. Casner">
<date month="April" year="2012" /> <organization showOnFrontPage="true"/>
</front> </author>
<author initials="R." surname="Frederick" fullname="R. Frederick">
<format target="http://standards.iso.org/ittf/PubliclyAvailableStandards <organization showOnFrontPage="true"/>
/c057623_ISO_IEC_23009-1_2012.zip" </author>
type="TXT" /> <author initials="V." surname="Jacobson" fullname="V. Jacobson">
</reference> <organization showOnFrontPage="true"/>
</author>
<reference anchor="CH09"> <date year="2003" month="July"/>
<front> <abstract>
<title>Designing TCP-Friendly Window-based Congestion Control for <t indent="0">This memorandum describes RTP, the real-time transpo
Real-time Multimedia Applications</title> rt protocol. RTP provides end-to-end network transport functions suitable for a
pplications transmitting real-time data, such as audio, video or simulation data
<author fullname="Soo-Hyun Choi" initials="S" surname="Choi"> , over multicast or unicast network services. RTP does not address resource res
<organization></organization> ervation and does not guarantee quality-of- service for real-time services. The
</author> data transport is augmented by a control protocol (RTCP) to allow monitoring of
the data delivery in a manner scalable to large multicast networks, and to prov
<author fullname="Mark Handley" initials="M" surname="Handley"> ide minimal control and identification functionality. RTP and RTCP are designed
<organization></organization> to be independent of the underlying transport and network layers. The protocol
supports the use of RTP-level translators and mixers. Most of the text in this
<address> memorandum is identical to RFC 1889 which it obsoletes. There are no changes in
<postal> the packet formats on the wire, only changes to the rules and algorithms govern
<street></street> ing how the protocol is used. The biggest change is an enhancement to the scalab
le timer algorithm for calculating when to send RTCP packets in order to minimiz
<city></city> e transmission in excess of the intended rate when many participants join a sess
ion simultaneously. [STANDARDS-TRACK]</t>
<region></region> </abstract>
</front>
<code></code> <seriesInfo name="STD" value="64"/>
<seriesInfo name="RFC" value="3550"/>
<country></country> <seriesInfo name="DOI" value="10.17487/RFC3550"/>
</postal> </reference>
<reference anchor="RFC4585" target="https://www.rfc-editor.org/info/rfc4
<phone></phone> 585" quoteTitle="true" derivedAnchor="RFC4585">
<front>
<facsimile></facsimile> <title>Extended RTP Profile for Real-time Transport Control Protocol
(RTCP)-Based Feedback (RTP/AVPF)</title>
<email></email> <author initials="J." surname="Ott" fullname="J. Ott">
<organization showOnFrontPage="true"/>
<uri></uri> </author>
</address> <author initials="S." surname="Wenger" fullname="S. Wenger">
</author> <organization showOnFrontPage="true"/>
</author>
<date day="21" month="May" year="2009" /> <author initials="N." surname="Sato" fullname="N. Sato">
</front> <organization showOnFrontPage="true"/>
</author>
<seriesInfo name="PFLDNeT 2009 Workshop" value="" /> <author initials="C." surname="Burmeister" fullname="C. Burmeister">
<organization showOnFrontPage="true"/>
<format target="www.hpcc.jp/pfldnet2009/Program_files/1569199301.pdf" </author>
type="PDF" /> <author initials="J." surname="Rey" fullname="J. Rey">
</reference> <organization showOnFrontPage="true"/>
</author>
<date year="2006" month="July"/>
<abstract>
<t indent="0">Real-time media streams that use RTP are, to some de
gree, resilient against packet losses. Receivers may use the base mechanisms of
the Real-time Transport Control Protocol (RTCP) to report packet reception stat
istics and thus allow a sender to adapt its transmission behavior in the mid-ter
m. This is the sole means for feedback and feedback-based error repair (besides
a few codec-specific mechanisms). This document defines an extension to the Au
dio-visual Profile (AVP) that enables receivers to provide, statistically, more
immediate feedback to the senders and thus allows for short-term adaptation and
efficient feedback-based repair mechanisms to be implemented. This early feedba
ck profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves
scalability to large groups. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="4585"/>
<seriesInfo name="DOI" value="10.17487/RFC4585"/>
</reference>
<reference anchor="RFC5124" target="https://www.rfc-editor.org/info/rfc5
124" quoteTitle="true" derivedAnchor="RFC5124">
<front>
<title>Extended Secure RTP Profile for Real-time Transport Control P
rotocol (RTCP)-Based Feedback (RTP/SAVPF)</title>
<author initials="J." surname="Ott" fullname="J. Ott">
<organization showOnFrontPage="true"/>
</author>
<author initials="E." surname="Carrara" fullname="E. Carrara">
<organization showOnFrontPage="true"/>
</author>
<date year="2008" month="February"/>
<abstract>
<t indent="0">An RTP profile (SAVP) for secure real-time communica
tions and another profile (AVPF) to provide timely feedback from the receivers t
o a sender are defined in RFC 3711 and RFC 4585, respectively. This memo specif
ies the combination of both profiles to enable secure RTP communications with fe
edback. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5124"/>
<seriesInfo name="DOI" value="10.17487/RFC5124"/>
</reference>
<reference anchor="RFC8825" target="https://www.rfc-editor.org/info/rfc8
825" quoteTitle="true" derivedAnchor="RFC8825">
<front>
<title>Overview: Real-Time Protocols for Browser-Based Applications<
/title>
<author initials="H." surname="Alvestrand" fullname="Harald T. Alves
trand">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8825"/>
<seriesInfo name="DOI" value="10.17487/RFC8825"/>
</reference>
</references>
<references pn="section-6.2">
<name slugifiedName="name-informative-references">Informative References
</name>
<reference anchor="CH09" quoteTitle="true" derivedAnchor="CH09">
<front>
<title>Designing TCP-Friendly Window-based Congestion Control for Re
al-time Multimedia Applications</title>
<author fullname="Soo-Hyun Choi" initials="S" surname="Choi">
<organization showOnFrontPage="true"/>
</author>
<author fullname="Mark Handley" initials="M" surname="Handley">
<organization showOnFrontPage="true"/>
</author>
<date month="May" year="2009"/>
</front>
<refcontent>Proceedings of PFLDNeT</refcontent>
</reference>
<reference anchor="MPEG_DASH" target="https://www.iso.org/standard/79329
.html" quoteTitle="true" derivedAnchor="MPEG_DASH">
<front>
<title>Information Technology -- Dynamic adaptive streaming over HTT
P (DASH) -- Part 1: Media presentation description and segment formats</title>
<author>
<organization showOnFrontPage="true">ISO</organization>
</author>
<date month="December" year="2019"/>
</front>
<seriesInfo name="ISO/IEC" value="23009-1:2019"/>
</reference>
<reference anchor="RFC3168" target="https://www.rfc-editor.org/info/rfc3
168" quoteTitle="true" derivedAnchor="RFC3168">
<front>
<title>The Addition of Explicit Congestion Notification (ECN) to IP<
/title>
<author initials="K." surname="Ramakrishnan" fullname="K. Ramakrishn
an">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Floyd" fullname="S. Floyd">
<organization showOnFrontPage="true"/>
</author>
<author initials="D." surname="Black" fullname="D. Black">
<organization showOnFrontPage="true"/>
</author>
<date year="2001" month="September"/>
<abstract>
<t indent="0">This memo specifies the incorporation of ECN (Explic
it Congestion Notification) to TCP and IP, including ECN's use of two bits in th
e IP header. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="3168"/>
<seriesInfo name="DOI" value="10.17487/RFC3168"/>
</reference>
<reference anchor="RFC4828" target="https://www.rfc-editor.org/info/rfc4
828" quoteTitle="true" derivedAnchor="RFC4828">
<front>
<title>TCP Friendly Rate Control (TFRC): The Small-Packet (SP) Varia
nt</title>
<author initials="S." surname="Floyd" fullname="S. Floyd">
<organization showOnFrontPage="true"/>
</author>
<author initials="E." surname="Kohler" fullname="E. Kohler">
<organization showOnFrontPage="true"/>
</author>
<date year="2007" month="April"/>
<abstract>
<t indent="0">This document proposes a mechanism for further exper
imentation, but not for widespread deployment at this time in the global Interne
t.</t>
<t indent="0">TCP-Friendly Rate Control (TFRC) is a congestion con
trol mechanism for unicast flows operating in a best-effort Internet environment
(RFC 3448). TFRC was intended for applications that use a fixed packet size, a
nd was designed to be reasonably fair when competing for bandwidth with TCP conn
ections using the same packet size. This document proposes TFRC-SP, a Small-Pac
ket (SP) variant of TFRC, that is designed for applications that send small pack
ets. The design goal for TFRC-SP is to achieve the same bandwidth in bps (bits
per second) as a TCP flow using packets of up to 1500 bytes. TFRC-SP enforces a
minimum interval of 10 ms between data packets to prevent a single flow from se
nding small packets arbitrarily frequently.</t>
<t indent="0">Flows using TFRC-SP compete reasonably fairly with l
arge-packet TCP and TFRC flows in environments where large-packet flows and smal
l-packet flows experience similar packet drop rates. However, in environments w
here small-packet flows experience lower packet drop rates than large-packet flo
ws (e.g., with Drop-Tail queues in units of bytes), TFRC-SP can receive consider
ably more than its share of the bandwidth. This memo defines an Experimental Pr
otocol for the Internet community.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="4828"/>
<seriesInfo name="DOI" value="10.17487/RFC4828"/>
</reference>
<reference anchor="RFC5348" target="https://www.rfc-editor.org/info/rfc5
348" quoteTitle="true" derivedAnchor="RFC5348">
<front>
<title>TCP Friendly Rate Control (TFRC): Protocol Specification</tit
le>
<author initials="S." surname="Floyd" fullname="S. Floyd">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Handley" fullname="M. Handley">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Padhye" fullname="J. Padhye">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Widmer" fullname="J. Widmer">
<organization showOnFrontPage="true"/>
</author>
<date year="2008" month="September"/>
<abstract>
<t indent="0">This document specifies TCP Friendly Rate Control (T
FRC). TFRC is a congestion control mechanism for unicast flows operating in a b
est-effort Internet environment. It is reasonably fair when competing for bandw
idth with TCP flows, but has a much lower variation of throughput over time comp
ared with TCP, making it more suitable for applications such as streaming media
where a relatively smooth sending rate is of importance.</t>
<t indent="0">This document obsoletes RFC 3448 and updates RFC 434
2. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5348"/>
<seriesInfo name="DOI" value="10.17487/RFC5348"/>
</reference>
<reference anchor="RFC5506" target="https://www.rfc-editor.org/info/rfc5
506" quoteTitle="true" derivedAnchor="RFC5506">
<front>
<title>Support for Reduced-Size Real-Time Transport Control Protocol
(RTCP): Opportunities and Consequences</title>
<author initials="I." surname="Johansson" fullname="I. Johansson">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Westerlund" fullname="M. Westerlund">
<organization showOnFrontPage="true"/>
</author>
<date year="2009" month="April"/>
<abstract>
<t indent="0">This memo discusses benefits and issues that arise w
hen allowing Real-time Transport Protocol (RTCP) packets to be transmitted with
reduced size. The size can be reduced if the rules on how to create compound pa
ckets outlined in RFC 3550 are removed or changed. Based on that analysis, this
memo defines certain changes to the rules to allow feedback messages to be sent
as Reduced-Size RTCP packets under certain conditions when using the RTP/AVPF (
Real-time Transport Protocol / Audio-Visual Profile with Feedback) profile (RFC
4585). This document updates RFC 3550, RFC 3711, and RFC 4585. [STANDARDS-TRACK
]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5506"/>
<seriesInfo name="DOI" value="10.17487/RFC5506"/>
</reference>
<reference anchor="RFC5865" target="https://www.rfc-editor.org/info/rfc5
865" quoteTitle="true" derivedAnchor="RFC5865">
<front>
<title>A Differentiated Services Code Point (DSCP) for Capacity-Admi
tted Traffic</title>
<author initials="F." surname="Baker" fullname="F. Baker">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Polk" fullname="J. Polk">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Dolly" fullname="M. Dolly">
<organization showOnFrontPage="true"/>
</author>
<date year="2010" month="May"/>
<abstract>
<t indent="0">This document requests one Differentiated Services C
ode Point (DSCP) from the Internet Assigned Numbers Authority (IANA) for a class
of real-time traffic. This traffic class conforms to the Expedited Forwarding
Per-Hop Behavior. This traffic is also admitted by the network using a Call Adm
ission Control (CAC) procedure involving authentication, authorization, and capa
city admission. This differs from a real-time traffic class that conforms to th
e Expedited Forwarding Per-Hop Behavior but is not subject to capacity admission
or subject to very coarse capacity admission. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5865"/>
<seriesInfo name="DOI" value="10.17487/RFC5865"/>
</reference>
<reference anchor="RFC7295" target="https://www.rfc-editor.org/info/rfc7
295" quoteTitle="true" derivedAnchor="RFC7295">
<front>
<title>Report from the IAB/IRTF Workshop on Congestion Control for I
nteractive Real-Time Communication</title>
<author initials="H." surname="Tschofenig" fullname="H. Tschofenig">
<organization showOnFrontPage="true"/>
</author>
<author initials="L." surname="Eggert" fullname="L. Eggert">
<organization showOnFrontPage="true"/>
</author>
<author initials="Z." surname="Sarker" fullname="Z. Sarker">
<organization showOnFrontPage="true"/>
</author>
<date year="2014" month="July"/>
<abstract>
<t indent="0">This document provides a summary of the IAB/IRTF Wor
kshop on 'Congestion Control for Interactive Real-Time Communication', which too
k place in Vancouver, Canada, on July 28, 2012. The main goal of the workshop w
as to foster a discussion on congestion control mechanisms for interactive real-
time communication. This report summarizes the discussions and lists recommenda
tions to the Internet Engineering Task Force (IETF) community.</t>
<t indent="0">The views and positions in this report are those of
the workshop participants and do not necessarily reflect the views and positions
of the authors, the Internet Architecture Board (IAB), or the Internet Research
Task Force (IRTF).</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7295"/>
<seriesInfo name="DOI" value="10.17487/RFC7295"/>
</reference>
<reference anchor="RFC8083" target="https://www.rfc-editor.org/info/rfc8
083" quoteTitle="true" derivedAnchor="RFC8083">
<front>
<title>Multimedia Congestion Control: Circuit Breakers for Unicast R
TP Sessions</title>
<author initials="C." surname="Perkins" fullname="C. Perkins">
<organization showOnFrontPage="true"/>
</author>
<author initials="V." surname="Singh" fullname="V. Singh">
<organization showOnFrontPage="true"/>
</author>
<date year="2017" month="March"/>
<abstract>
<t indent="0">The Real-time Transport Protocol (RTP) is widely use
d in telephony, video conferencing, and telepresence applications. Such applica
tions are often run on best-effort UDP/IP networks. If congestion control is no
t implemented in these applications, then network congestion can lead to uncontr
olled packet loss and a resulting deterioration of the user's multimedia experie
nce. The congestion control algorithm acts as a safety measure by stopping RTP
flows from using excessive resources and protecting the network from overload.
At the time of this writing, however, while there are several proprietary soluti
ons, there is no standard algorithm for congestion control of interactive RTP fl
ows.</t>
<t indent="0">This document does not propose a congestion control
algorithm. It instead defines a minimal set of RTP circuit breakers: conditions
under which an RTP sender needs to stop transmitting media data to protect the
network from excessive congestion. It is expected that, in the absence of long-
lived excessive congestion, RTP applications running on best-effort IP networks
will be able to operate without triggering these circuit breakers. To avoid tri
ggering the RTP circuit breaker, any Standards Track congestion control algorith
ms defined for RTP will need to operate within the envelope set by these RTP cir
cuit breaker algorithms.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="8083"/>
<seriesInfo name="DOI" value="10.17487/RFC8083"/>
</reference>
<reference anchor="RFC8831" target="https://www.rfc-editor.org/info/rfc8
831" quoteTitle="true" derivedAnchor="RFC8831">
<front>
<title>WebRTC Data Channels</title>
<author initials="R" surname="Jesup" fullname="Randell Jesup">
<organization showOnFrontPage="true"/>
</author>
<author initials="S" surname="Loreto" fullname="Salvatore Loreto">
<organization showOnFrontPage="true"/>
</author>
<author initials="M" surname="Tüxen" fullname="Michael Tüxen">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8831"/>
<seriesInfo name="DOI" value="10.17487/RFC8831"/>
</reference>
</references>
</references> </references>
<section anchor="Acknowledgements" numbered="false" toc="include" removeInRF
C="false" pn="section-appendix.a">
<name slugifiedName="name-acknowledgements">Acknowledgements</name>
<t indent="0" pn="section-appendix.a-1">This document is the result of dis
cussions in various fora of the
WebRTC effort, in particular on the &lt;rtp-congestion@alvestrand.no&gt; m
ailing
list. Many people contributed their thoughts to this.</t>
</section>
<section anchor="authors-addresses" numbered="false" removeInRFC="false" toc
="include" pn="section-appendix.b">
<name slugifiedName="name-authors-addresses">Authors' Addresses</name>
<author fullname="Randell Jesup" initials="R." surname="Jesup">
<organization showOnFrontPage="true">Mozilla</organization>
<address>
<postal>
<street/>
<country>United States of America</country>
</postal>
<email>randell-ietf@jesup.org</email>
</address>
</author>
<author fullname="Zaheduzzaman Sarker" initials="Z." role="editor" surname
="Sarker">
<organization showOnFrontPage="true">Ericsson AB</organization>
<address>
<postal>
<street>Torshamnsgatan 23</street>
<city>Stockholm</city>
<region/>
<code>164 83</code>
<country>Sweden</country>
</postal>
<phone>+46 10 717 37 43</phone>
<email>zaheduzzaman.sarker@ericsson.com</email>
</address>
</author>
</section>
</back> </back>
</rfc> </rfc>
 End of changes. 82 change blocks. 
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