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In: MissingRef
Network Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Informational October 7, 2014
Expires: April 10, 2015
Congestion Control Requirements For RMCAT
draft-ietf-rmcat-cc-requirements-06
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 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[I-D.ietf-rtcweb-overview]), it is especially important to
ensure that this kind of traffic is congestion controlled.
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.
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].
The terms are presented in many cases using lowercase for
readability.
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
working documents as Internet-Drafts. The list of current Internet-
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
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time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 10, 2015.
Copyright Notice
Copyright (c) 2014 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
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
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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.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3
3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
4. Security Considerations . . . . . . . . . . . . . . . . . . . 8
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.1. Normative References . . . . . . . . . . . . . . . . . . 8
6.2. Informative References . . . . . . . . . . . . . . . . . 9
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9
1. Introduction
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.
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
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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.
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.
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 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. 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.)
A. 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.
B. 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.
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C. 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.
D. 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.
E. 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.
F. 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.
G. Similarly periodic bursty flows such as MPEG DASH
[MPEG_DASH] 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.
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. 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.
A. 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.
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Note that relative RTTs may affect the rate new flows can
ramp up to a reasonable share.
3. The algorithm should not starve competing TCP flows, and should
as best as possible avoid starvation by TCP flows.
A. An algorithm may be more successful at avoiding starvation
from short-lived TCP than long-lived/saturating TCP flows.
B. In order to avoid starvation other goals may need to be
compromised (such as delay).
4. 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!"). Starting over-
bandwidth causes other problems for user experience, so
there's a tension here.
B. 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.
C. 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).
5. It should be stable if the RTP streams are halted or
discontinuous (Voice Activity Detection/Discontinuous
Transmission).
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.
6. 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
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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
[I-D.welzl-rmcat-coupled-cc].
A. If possible, it should also share information and adaptation
with other non-RTP flows between the same endpoints, such as
a WebRTC DataChannel[I-D.ietf-rtcweb-data-channel]
B. 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).
C. 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.
D. 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.
7. The algorithm should not require any special support from
network elements (Explicit Congestion Notification (ECN)
[RFC3168], 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.
A. 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.
B. When additional input signals such as ECN are available,
they should be utilized if possible.
8. 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.
A. In order to react sufficiently quickly when using RTCP for a
backchannel, an RTP profile such as RTP/AVPF [RFC4585] or
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RTP/SAVPF [RFC5124] that allows sufficiently frequent
feedback 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. Bandwidth for the feedback messages should be minimized
(such as via RFC 5506 [RFC5506]to allow RTCP without Sender
Reports/Receiver Reports)
D. Header extensions would avoid the RTCP timing rules issues,
and allow the application to allocate bandwidth as needed
for the congestion algorithm.
E. 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.
9. Flows managed by this algorithm and flows competing against at a
bottleneck may have different DSCP[RFC5865] 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.
A. 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.
10. 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.
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11. 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
3. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
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
Browser-based Applications", draft-ietf-rtcweb-overview-11
(work in progress), August 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[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.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
6.2. Informative References
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-12 (work in
progress), September 2014.
[I-D.welzl-rmcat-coupled-cc]
Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
control for RTP media", draft-welzl-rmcat-coupled-cc-03
(work in progress), May 2014.
[MPEG_DASH]
"Dynamic adaptive streaming over HTTP (DASH) -- Part 1:
Media presentation description and segment formats", April
2012.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5865] Baker, F., Polk, J., and M. Dolly, "A Differentiated
Services Code Point (DSCP) for Capacity-Admitted Traffic",
RFC 5865, May 2010.
Author's Address
Randell Jesup
Mozilla
USA
Email: randell-ietf@jesup.org
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