One document matched: draft-jesup-rmcat-reqs-01.xml
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<rfc category="info" docName="draft-jesup-rmcat-reqs-01"
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 day="25" month="Feb" year="2013" />
<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 by low-delay, semi-reliable data delivery, are
different from the requirements for bulk transfer like FTP or bursty
transfers like Web pages, and the TCP algorithms are not suitable for
this traffic.</t>
<t>This document attempts to describe 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 proposals coming out of the RMCAT Working Group.</t>
<t>This document is derived from draft-jesup-rtp-congestion-reqs
<xref target="I-D.jesup-rtp-congestion-reqs"></xref>.</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>.</t>
</note>
</front>
<middle>
<section title="Introduction">
<t>The traditional TCP congestion control requirements were developed in
order to promote efficient 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 fast as possible" mode, 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 useless.</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 YouTube-like 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, even when faced
with intermediate bottlenecks and competing flows. 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>It should also deal well with routing changes and interface
changes (WiFi to 3G data, etc) which may radically change the
bandwidth available.</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. <list
style="letters">
<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.</t>
<t>We will need make some evaluation of fairness, but
deciding what is "fair" is a tough question and likely to be
partially subjective, but we should specify some of the
inputs needed in order to select among algorithms and
tunings presented as options.</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. <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 data channel</t>
</list></t>
<t>The algorithm should not require any special support from
network elements (ECN, etc). As much as possible, it should
leverage existing information about the incoming flows to provide
feedback to the sender. Examples of this information are the
packet arrival times, acknowledgments and feedback, packet
timestamps, packet sizes, packet losses. 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.<list style="letters">
<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; the alternative
would be to include it in a reverse RTP channel using header
extensions. <list style="letters">
<t>In order to react sufficiently quickly, the AVPF/SAVPF RTP
profile<xref target="RFC4585"> </xref> 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>Note that RTCP is of course unreliable</t>
<t>Bandwidth for the feedback messages should be minimized (such
as via RFC 5506 <xref target="RFC5506"></xref>to allow RTCP
without SR/RR)</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 fequency are expected to be
evaluated.</t>
</list></t>
<t>Where possible and helpful, the algorithm should leverage and
piggyback on other RTP/RTCP communications, such as SR/RR,
rctp-fb PLI, RPSI, SLI or application-specific NACK messages
(such as for loss information), and also reverse-direction
RTP.</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>It should attempt to avoid bandwidth 'collapse' when facing a
long-lived saturating TCP flow or flows. (I.e. a classic
delay-sensitive algorithm will reduce bandwidth to keep delay down
until the TCP flow has all the bandwidth). See the Cx-TCP algorithm
discussed in a recent Transactions On Networking <xref
target="cx-tcp"></xref> for an example of a delay-sensitive
congestion-control algorithm that transitions to a loss-based mode
when competing with TCP flows - at the cost of increased delay.</t>
<t>The algorithm should be stable and low-delay when faced with
active queue management (AQM) such as RED <xref target="RFC2309"></xref>
or CoDel <xref target="I-D.nichols-tsvwg-codel"></xref> in the channel.</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!").</t>
<t>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.</t>
</list></t>
<t>It should be evaluated in how it works both with
backbone-router bottlenecks, (asymmetric) local-loop bottlenecks,
and local-lan (WiFi/etc) bottlenecks, and in competition with
varying numbers and types of streams (TCP, TCP variants in use,
LEDBAT <xref target="I-D.ietf-ledbat-congestion"></xref>, inflexible VoIP UDP flows).</t>
<t>It should be stable if the RTP streams are halted or
discontinuous (VAD/DTX). <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>
</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.I-D.ietf-rtcweb-overview'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.2309'?>
<?rfc include='reference.RFC.5506'?>
<?rfc include='reference.I-D.jesup-rtp-congestion-reqs'?>
<?rfc include='reference.I-D.nichols-tsvwg-codel'?>
<?rfc include='reference.I-D.ietf-ledbat-congestion'?>
<reference anchor="cx-tcp">
<front>
<title>On the Fair Coexistence of Loss- and Delay-Based TCP</title>
<author fullname="Lukasz Budzisz" initials="L." surname="Budzisz">
<organization>B</organization>
</author>
<author fullname="Rade Stanojevic" initials="R."
surname="Stanojevic">
<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>
<author fullname="Arieh Schlote" initials="A." surname="Schlote">
<organization>S</organization>
<address>
<postal>
<street></street>
<city></city>
<region></region>
<code></code>
<country></country>
</postal>
<phone></phone>
<facsimile></facsimile>
<email></email>
<uri></uri>
</address>
</author>
<author fullname="Fred Baker" initials="F." surname="Baker">
<organization>F</organization>
<address>
<postal>
<street></street>
<city></city>
<region></region>
<code></code>
<country></country>
</postal>
<phone></phone>
<facsimile></facsimile>
<email></email>
<uri></uri>
</address>
</author>
<author fullname="Robert Shorten" initials="R." surname="Shorten">
<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 month="December" year="2011" />
<note title="Publisher">
<t>IEEE/ACM Transactions on Networking, vol. 19 no. 6, December
2011</t>
</note>
<note title="URL:">
<t>http://www.networks.imdea.org/Portals/7/Descargas/Publicaciones/On_the_Fair_Coexistence__-_2011_EN.pdf</t>
</note>
</front>
</reference>
</references>
</back>
</rfc>
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