One document matched: draft-ietf-rmcat-cc-requirements-07.xml
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<rfc category="info" docName="draft-ietf-rmcat-cc-requirements-07"
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<front>
<title abbrev="RMCAT congestion requirements">Congestion Control
Requirements for Interactive Real-Time Media</title>
<author fullname="Randell Jesup" initials="R." surname="Jesup">
<organization>Mozilla</organization>
<address>
<postal>
<street></street>
<country>USA</country>
</postal>
<email>randell-ietf@jesup.org</email>
</address>
</author>
<author fullname="Zaheduzzaman Sarker" initials="Z." role="editor"
surname="Sarker">
<organization>Ericsson</organization>
<address>
<postal>
<street></street>
<city></city>
<region></region>
<code></code>
<country>Sweden</country>
</postal>
<phone></phone>
<facsimile></facsimile>
<email>zaheduzzaman.sarker@ericsson.com</email>
<uri></uri>
</address>
</author>
<date />
<abstract>
<t>Congestion control is needed for all data transported across the
Internet, in order to promote fair usage and prevent congestion
collapse. The requirements for interactive, point-to-point real-time
multimedia, which needs low-delay, semi-reliable data delivery, are
different from the requirements for bulk transfer like FTP or bursty
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
the Web Real-Time Communication (WebRTC)), it is especially important to
ensure that this kind of traffic is congestion controlled.</t>
<t>This document describes a set of requirements that can be used to
evaluate other congestion control mechanisms in order to figure out
their fitness for this purpose, and in particular to provide a set of
possible requirements for real-time media congestion avoidance
technique.</t>
</abstract>
<note title="Requirements Language">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in <xref
target="RFC2119">RFC 2119</xref>. The terms are presented in many cases
using lowercase for readability.</t>
</note>
</front>
<middle>
<section title="Introduction">
<t>Most of today's TCP congestion control schemes were developed with a
focus on an use of the Internet for reliable bulk transfer of
non-time-critical data, such as transfer of large files. They have also
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
pages.</t>
<t>These algorithms have also been used for transfer of media streams
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
the exact timing of the delivery is not.</t>
<t>When doing real-time interactive media, the requirements are
different; one needs to provide the data continuously, within a very
limited time window (no more than 100s of milliseconds end-to-end
delay), the sources of data may be able to adapt the amount of data that
needs sending within fairly wide margins but can be rate limited by the
application- even not always have data to send, and 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
real-time, data delivered late is commonly useless.</t>
<t>While the requirements for real-time interactive differ from the
requirements for the other flow types, these other flow types will be
present in the network. The congestion control algorithm for real-time
interactive media must work properly when these other flow types are
present as cross traffic on the network.</t>
<t>One particular protocol portofolio being developed for this use case
is WebRTC <xref target="I-D.ietf-rtcweb-overview"></xref>, where one
envisions sending multiple flows using the Real-time Transport Protocol
(RTP) <xref target="RFC3550"></xref> between two peers, in conjunction
with data flows, all at the same time, without having special
arrangements with the intervening service providers.</t>
<t>Given that this use case is the focus of this document, use cases
involving non-interactive media such as video streaming, and use cases
using multicast/broadcast-type technologies, are out of scope.</t>
<t>The terminology defined in <xref
target="I-D.ietf-rtcweb-overview"></xref> is used in this memo.</t>
</section>
<section title="Requirements">
<t><list style="numbers">
<t>The congestion control algorithm must attempt to provide
as-low-as-possible-delay transit for interactive real-time traffic
while still providing a useful amount of bandwidth. There may be
lower limits on the amount of bandwidth that is useful, but this is
largely application-specific and the application may be able to
modify or remove flows in order allow some useful flows to get
enough bandwidth. (Example: not enough bandwidth for low-latency
video+audio, but enough for audio-only.) <list style="letters">
<t>Jitter (variation in the bitrate over short timescales) also
is relevant, though moderate amounts of jitter will be absorbed
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 even
when faced with intermediate bottlenecks and competing flows.
Competing flows may limit what's possible to achieve.</t>
<t>It should handle routing changes which may alter or remove
bottlenecks or change the bandwidth available especially if
there is a reduction in available bandwidth or increase in
observed delay. It is expected that the mechanism reacts to the
routing change events in a way that avoids delay buildup.</t>
<t>It should handle both local and remote interface changes
(WLAN to 3G data, etc) which may radically change the bandwidth
available or bottlenecks, especially if there is a reduction in
available bandwidth or increase in bottleneck delay. It is
assumed that an interface change can generate a notification to
the algorithm.</t>
<t>The algorithm must consider the case where offered loads are
less than the available bandwidth at any given moment, and may
vary dramatically over time, including dropping to no load and
then resuming a high load, such as in a mute/unmute operation.
Note that the reaction time between a change in the bandwidth
available from the algorithm and a change in the offered load is
variable, and may be different when increasing versus
decreasing.</t>
<t>The algorithm requires to avoid building up queues when
competing with short-term bursts of traffic (for example,
traffic generated by web-browsing) which can quickly saturate a
local-bottleneck router or link, but also clear quickly. The
algorithm should also attempt to regain its previous share of
the bandwidth when the local-bottleneck or link is cleared.</t>
<t>Similarly periodic bursty flows such as MPEG DASH <xref
target="MPEG_DASH"></xref> or proprietary media 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. Due to non-adaptiveness of the competing traffic as such,
the algorithm must not increase the already existing delay
buildup during those bursts. Note that this competing traffic
may on a shared access link, or the traffic burst may cause a
shift in the location of the bottleneck for the duration of the
burst.</t>
</list></t>
<t>The algorithm must be fair to other flows, both real-time flows
(such as other instances of itself), and TCP flows, both long-lived
and bursts such as the traffic generated by a typical web browsing
session. Note that 'fair' is a rather hard-to-define term. It should
be fair with itself, giving fair share of the bandwidth to multiple
flows with similar RTTs, and if possible to multiple flows with
different RTTs.<list style="letters">
<t>Existing flows at a bottleneck 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.
A useful share will depend on the media types involved and total
bandwidth available. Note that relative RTTs may affect the rate
new flows can ramp up to a reasonable share.</t>
</list></t>
<t>The algorithm should not starve competing TCP flows, and should
as best as possible avoid starvation by TCP flows.<list
style="letters">
<t>The congestion control should prioritise achieving a useful
share of the bandwidth depending on the media types and total
available bandwidth over achieving as low as possible transit
delay, when these two requirements are in conflict.</t>
</list></t>
<t>The algorithm should as quickly as possible adapt to initial
network conditions at the start of a flow. This should occur both if
the initial bandwidth is above or below the bottleneck bandwidth.
<list style="letters">
<t>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
(adapt up) and history-based startup (start at a point expected
to be at or below channel bandwidth from historical information,
which may need to adapt down quickly if the initial guess is
wrong). Starting too low and/or adapting up too slowly can cause
a critical point in a personal communication to be poor
("Hello!"). Starting over-bandwidth causes other problems for
user experience, so there's a tension here. Alternative methods
to help startup like probing during setup with dummy data may be
useful in some applications; in some cases there will be a
considerable gap in time between flow creation and the initial
flow of data. Again, A flow may need to change adaptation rates
due to network conditions or changes in the provided flows (such
as un-muting or sending data after a gap).</t>
</list></t>
<t>The algorithm should be stable if the RTP streams are halted or
discontinuous (for example - Voice Activity Detection). <list
style="letters">
<t>After stream resumption, the algorithm should attempt to
rapidly regain its previous share of the bandwidth; the
aggressiveness with which this is done will decay with the
length of the pause.</t>
</list></t>
<t>The algorithm should where possible merge information across
multiple RTP streams sent between two endpoints, when those RTP
streams share a common bottleneck, whether or not those streams are
multiplexed onto the same ports, in order to allow congestion
control of the set of streams together instead of as multiple
independent streams. This allows better overall bandwidth
management, faster response to changing conditions, and fairer
sharing of bandwidth with other network users.<list style="letters">
<t>The algorithm should also share information and adaptation
with other non-RTP flows between the same endpoints, such as a
WebRTC DataChannel <xref
target="I-D.ietf-rtcweb-data-channel"></xref>, when
possible.</t>
<t>When there are multiple streams across the same 5-tuple
coordinating their bandwidth use and congestion control, the
algorithm should allow the application to control the relative
split of available bandwidth.The most correlated bandwidth usage
would be with other flows on the same 5-tuple, but there may be
use in coordinating measurement and control of the local
link(s). Use of information about previous flows, especially on
the same 5-tuple, may be useful input to the algorithm,
especially to startup performance of a new flow.</t>
</list></t>
<t>The algorithm should not require any special support from network
elements to convey congestion related information to be functional.
As much as possible, it should leverage available information about
the incoming flow to provide feedback to the sender. Examples of
this information are the packet arrival times, acknowledgments and
feedback, packet timestamps, and packet losses, Explicit Congestion
Notification (ECN) <xref target="RFC3168"></xref>; all of these can
provide information about the state of the path and any bottlenecks.
However, the use of available information is algorithm
dependent.<list style="letters">
<t>Extra information could be added to the packets to provide
more detailed information on actual send times (as opposed to
sampling times), but should not be required.</t>
</list></t>
<t>Since the assumption here is a set of RTP streams, the
backchannel typically should be done via RTCP<xref
target="RFC3550"></xref>; one alternative would be to include it
instead in a reverse RTP channel using header extensions.<list
style="letters">
<t>In order to react sufficiently quickly when using RTCP for a
backchannel, an RTP profile such as RTP/AVPF <xref
target="RFC4585"></xref> or RTP/SAVPF <xref
target="RFC5124"></xref> that allows sufficiently frequent
feedback must be used. Note that in some cases, backchannel
messages may be delayed until the RTCP channel can be allocated
enough bandwidth, even under AVPF rules. This may also imply
negotiating a higher maximum percentage for RTCP data or
allowing solutions to violate or modify the rules specified for
AVPF.</t>
<t>Bandwidth for the feedback messages should be minimized (such
as via RFC 5506 <xref target="RFC5506"></xref>to allow RTCP
without Sender Reports/Receiver Reports)</t>
<t>Backchannel data should be minimized to avoid taking too much
reverse-channel bandwidth (since this will often be used in a
bidirectional set of flows). In areas of stability, backchannel
data may be sent more infrequently so long as algorithm
stability and fairness are maintained. When the channel is
unstable or has not yet reached equilibrium after a change,
backchannel feedback may be more frequent and use more
reverse-channel bandwidth. This is an area with considerable
flexibility of design, and different approaches to backchannel
messages and frequency are expected to be evaluated.</t>
</list></t>
<t>Flows managed by this algorithm and flows competing against at a
bottleneck may have different DSCP<xref target="RFC5865"></xref>
markings depending on the type of traffic, or may be subject to
flow-based QoS. A particular bottleneck or section of the network
path may or may not honor DSCP markings. The algorithm should
attempt to leverage DSCP markings when they're available.<list
style="letters">
<t>In WebRTC, a division of packets into 4 classes is envisioned
in order of priority: faster-than-audio, audio, video,
best-effort, and bulk-transfer. Typically the flows managed by
this algorithm would be audio or video in that hierarchy, and
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
collapse.</t>
<t>The algorithm should be stable and low-delay when faced with
Active Queue Management (AQM) algorithms. Also note that these
algorithms may apply across multiple queues in the bottleneck, or to
a single queue</t>
</list></t>
</section>
<section title="Deficiencies of existing mechanisms ">
<t>Among the existing congestion control mechanisms TCP Friendly Rate
Control (TFRC) <xref target="RFC5348"></xref> is the one which claims to
be suitable for real-time interactive media. TFRC is, an equation based,
congestion control mechanism which provides reasonably fair share of the
bandwidth when competing with TCP flows and offers much lower throughput
variations than TCP. This is achieved by a slower response to the
available bandwidth change than TCP. TFRC is designed to perform best
with applications which has fixed packet size and does not have fixed
period between sending packets.</t>
<t>TFRC operates on detecting loss events and reacts to loss caused by
congestion by reducing its sending rate. It allows applications to
increase the sending rate until loss is observed in the flows. As it is
noted in IAB/IRTF report <xref target="RFC7295"></xref> large buffers
are available in the network elements which introduces additional delay
in the communication, it becomes important to take all possible
congestion indications into considerations. TFRC's only consideration of
loss events as congestion indication can be considered as biggest
lacking looking at the current Internet deployment.</t>
<t>A typical real-time interactive communication includes live encoded
audio and video flow(s). In such a communication scenario an audio
source typically needs fixed interval between packets, needs to vary
their segment size instead of their packet rate in response to
congestion and sends smaller packets, a variance of TFRC , Small-Packet
TFRC (TFRC-SP) <xref target="RFC4828"></xref> addresses the issues
related to such kind of sources ; a video source generally varies video
frame sizes, can produce large frames which need to be further
fragmented to fit into path Maximum Transmission Unit (MTU) size, and
have almost fixed interval between producing frames under a certain
frame rate, TFRC is known to be less optimal when using with such video
sources.</t>
<t>There are also some mismatches between TFRC's design assumptions and
how the media sources in a typical real-time interactive application
works. TFRC is design to maintain smooth sending rate however media
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)
packets per second, where typical real-time interactive media sources
operates on bit per second. There are also limitations on how quickly
the media sources can adapt to specific sending rates. The modern video
encoders can operate on a mode where they can vary the output bitrate a
lot depending on the way there are configured, the current scene it is
encoding and more. Therefore, it is possible that the video source does
not always output at a bitrate they are allowed to. TFRC tries to raise
its sending rate when transmitting at maximum allowed rate and increases
only twice the current transmission rate hence it may create issues when
the video source vary their bitrates.</t>
<t>Moreover, there are number of studies on TFRC which shows it's
limitations which includes TFRC's unfairness on low statistically
multiplexed links, oscillatory behavior, performance issue in highly
dynamic loss rates conditions and more <xref target="CH09"></xref>.</t>
<t>Looking at all these deficiencies it can be concluded that the
requirements of congestion control mechanism for real-time interactive
media cannot be met by TFRC as defined in the standard.</t>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>An attacker with the ability to delete, delay or insert messages in
the flow can fake congestion signals, unless they are passed on a
tamper-proof path. Since some possible algorithms depend on the timing
of packet arrival, even a traditional protected channel does not fully
mitigate such attacks.</t>
<t>An attack that reduces bandwidth is not necessarily significant,
since an on-path attacker could break the connection by discarding all
packets. Attacks that increase the percieved available bandwidth are
concievable, and need to be evaluated.</t>
<t>Algorithm designers should consider the possibility of malicious
on-path attackers.</t>
</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>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.4585'?>
<?rfc include='reference.RFC.5124'?>
<?rfc include='reference.I-D.ietf-rtcweb-overview'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.3168'?>
<?rfc include='reference.RFC.5506'?>
<?rfc include='reference.RFC.5865'?>
<?rfc include='reference.RFC.5348'?>
<?rfc include='reference.RFC.4828'?>
<?rfc include='reference.RFC.7295'?>
<?rfc include='reference.I-D.ietf-rtcweb-data-channel'?>
<reference anchor="MPEG_DASH">
<front>
<title>Dynamic adaptive streaming over HTTP (DASH) -- Part 1: Media
presentation description and segment formats</title>
<author></author>
<date month="April" year="2012" />
</front>
<format target="http://standards.iso.org/ittf/PubliclyAvailableStandards/c057623_ISO_IEC_23009-1_2012.zip"
type="TXT" />
</reference>
<reference anchor="CH09">
<front>
<title>Designing TCP-Friendly Window-based Congestion Control for
Real-time Multimedia Applications</title>
<author fullname="Soo-Hyun Choi" initials="S" surname="Choi">
<organization></organization>
</author>
<author fullname="Mark Handley" initials="M" surname="Handley">
<organization></organization>
<address>
<postal>
<street></street>
<city></city>
<region></region>
<code></code>
<country></country>
</postal>
<phone></phone>
<facsimile></facsimile>
<email></email>
<uri></uri>
</address>
</author>
<date day="21" month="May" year="2009" />
</front>
<seriesInfo name="PFLDNeT 2009 Workshop" value="" />
<format target="www.hpcc.jp/pfldnet2009/Program_files/1569199301.pdf"
type="PDF" />
</reference>
</references>
</back>
</rfc>
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