One document matched: draft-ietf-rmcat-cc-requirements-06.xml
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<rfc category="info" docName="draft-ietf-rmcat-cc-requirements-06"
ipr="trust200902">
<front>
<title abbrev="RMCAT congestion requirements">Congestion Control
Requirements For RMCAT</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>
<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
WebRTC<xref target="I-D.ietf-rtcweb-overview"></xref>), 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 realtime 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, 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 RMCAT differ from the requirements for
the other flow types, these other flow types will be present in the
network. The RMCAT congestion control algorithm 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 RTP-based flows 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 noninteractive 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 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,
and react quickly, especially if there is a reduction in
available bandwidth or increase in observed delay.</t>
<t>It should handle interface changes (WLAN to 3G data,
etc) which may radically change the bandwidth available
or bottlenecks, and react quickly, 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 offered load may be 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 operation. 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 must not overreact to short-term bursts (such
as web-browsing) which can quickly saturate a local-bottleneck
router or link, but also clear quickly, and should recover
quickly when the burst ends. This is inherently at odds with
the need to react quickly-enough to avoid queue buildup.</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. The algorithm must
avoid too much delay buildup during those bursts, and
quickly recover. 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 realtime
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 roughly equal 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
quickly. 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>An algorithm may be more successful at avoiding
starvation from short-lived TCP than long-lived/saturating
TCP flows.</t>
<t>In order to avoid starvation other goals may need to
be compromised (such as delay).</t>
</list></t>
<t>The algorithm should quickly 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 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.</t>
<t>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.</t>
<t>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>It should be stable if the RTP streams are halted or
discontinuous (Voice Activity Detection/Discontinuous Transmission).
<list style="letters">
<t>After a resumption of RTP data it may adapt more
quickly (similar to the start of a flow), and previous
bandwidth estimates may need to be aged or thrown
away.</t>
</list></t>
<t>The algorithm should where possible merge information across
multiple RTP streams between the same endpoints, whether or not
they're multiplexed on 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.
Alternatively, it should work with an external bandwidth control
framework to coordinate bandwidth usage across a bottleneck, such
as draft-welzl-rmcat-coupled-cc <xref
target="I-D.welzl-rmcat-coupled-cc"></xref>.<list
style="letters">
<t>If possible, it 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></t>
<t>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).</t>
<t>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>
<t>When there are multiple streams across the same
5-tuple coordinating their bandwidth use and congestion
control, it should be possible for the application to
control the relative split of available bandwidth.</t>
</list></t>
<t>The algorithm should not require any special support from
network elements (Explicit Congestion Notification (ECN)
<xref target="RFC3168"></xref>, etc). 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
ECN, packet arrival times, acknowledgments and feedback, packet
timestamps, and packet losses; all of these can provide
information about the state of the path and any bottlenecks.<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>
<t>When additional input signals such as ECN are available, they
should be utilized if possible.</t>
</list></t>
<t>Since the assumption here is a set of RTP streams, the
backchannel typically should be done via RTCP; 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.</t>
<t>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 RMCAT 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>Header extensions would avoid the RTCP timing rules issues,
and allow the application to allocate bandwidth as needed for
the congestion algorithm.</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
heirarchy, 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 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.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.I-D.ietf-rtcweb-data-channel'?>
<?rfc include='reference.I-D.welzl-rmcat-coupled-cc'?>
<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 type="TXT" target="http://standards.iso.org/ittf/PubliclyAvailableStandards/c057623_ISO_IEC_23009-1_2012.zip" />
</reference>
</references>
</back>
</rfc>
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