Internet DRAFT - draft-jesup-rmcat-reqs
draft-jesup-rmcat-reqs
Network Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Informational Feb 25, 2013
Expires: August 29, 2013
Congestion Control Requirements For RMCAT
draft-jesup-rmcat-reqs-01
Abstract
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.
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.
This document is derived from draft-jesup-rtp-congestion-reqs
[I-D.jesup-rtp-congestion-reqs].
Requirements Language
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 RFC 2119 [RFC2119].
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
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This Internet-Draft will expire on August 29, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
4. Security Considerations . . . . . . . . . . . . . . . . . . . . 8
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8
6. References . . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.1. Normative References . . . . . . . . . . . . . . . . . . . 8
6.2. Informative References . . . . . . . . . . . . . . . . . . 8
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 9
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1. Introduction
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.
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.
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.
One particular protocol portofolio being developed for this use case
is WebRTC [I-D.ietf-rtcweb-overview], 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.
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.
The terminology defined in [I-D.ietf-rtcweb-overview] is used in this
memo.
2. Requirements
1. 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-
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latency video+audio, but enough for audio-only.)
A. It should also deal well with routing changes and interface
changes (WiFi to 3G data, etc) which may radically change
the bandwidth available.
2. 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.
A. 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.
B. 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.
3. 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.
A. If possible, it should also share information and adaptation
with other non-RTP flows between the same endpoints, such as
a WebRTC data channel
4. 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.
A. When additional input signals such as ECN are available,
they should be utilized if possible.
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5. 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.
A. In order to react sufficiently quickly, the AVPF/SAVPF RTP
profile[RFC4585] must be used
B. 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.
C. Note that RTCP is of course unreliable
D. Bandwidth for the feedback messages should be minimized
(such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR)
E. Header extensions would avoid the RTCP timing rules issues,
and allow the application to allocate bandwidth as needed
for the congestion algorithm.
F. 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.
6. 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.
7. 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.
8. 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
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until the TCP flow has all the bandwidth). See the Cx-TCP
algorithm discussed in a recent Transactions On Networking
[cx-tcp] 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.
9. The algorithm should be stable and low-delay when faced with
active queue management (AQM) such as RED [RFC2309] or CoDel
[I-D.nichols-tsvwg-codel] in the channel.
10. 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.
A. 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!").
B. Starting over-bandwidth causes other problems for user
experience, so there's a tension here.
C. Alternative methods to help startup like probing during
setup with dummy data may be useful in some applications.
11. 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
[I-D.ietf-ledbat-congestion], inflexible VoIP UDP flows).
12. It should be stable if the RTP streams are halted or
discontinuous (VAD/DTX).
A. 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.
3. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
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RFC.
4. Security Considerations
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.
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.
Algorithm designers SHOULD consider the possibility of malicious on-
path attackers.
5. Acknowledgements
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.
6. References
6.1. Normative References
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-06 (work
in progress), February 2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
6.2. Informative References
[I-D.ietf-ledbat-congestion]
Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
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"Low Extra Delay Background Transport (LEDBAT)",
draft-ietf-ledbat-congestion-10 (work in progress),
September 2012.
[I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media",
draft-jesup-rtp-congestion-reqs-00 (work in progress),
March 2012.
[I-D.nichols-tsvwg-codel]
Nichols, K., "Controlled Delay Active Queue Management",
draft-nichols-tsvwg-codel-00 (work in progress),
July 2012.
[RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
S., Wroclawski, J., and L. Zhang, "Recommendations on
Queue Management and Congestion Avoidance in the
Internet", RFC 2309, April 1998.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[cx-tcp] Budzisz, L., Stanojevic, R., Schlote, A., Baker, F., and
R. Shorten, "On the Fair Coexistence of Loss- and Delay-
Based TCP", December 2011.
Author's Address
Randell Jesup
Mozilla
USA
Email: randell-ietf@jesup.org
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