draft-ietf-rmcat-cc-requirements-03.txt   draft-ietf-rmcat-cc-requirements-04.txt 
Network Working Group R. Jesup Network Working Group R. Jesup
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Informational March 31, 2014 Intended status: Informational April 19, 2014
Expires: October 2, 2014 Expires: October 21, 2014
Congestion Control Requirements For RMCAT Congestion Control Requirements For RMCAT
draft-ietf-rmcat-cc-requirements-03 draft-ietf-rmcat-cc-requirements-04
Abstract Abstract
Congestion control is needed for all data transported across the Congestion control is needed for all data transported across the
Internet, in order to promote fair usage and prevent congestion Internet, in order to promote fair usage and prevent congestion
collapse. The requirements for interactive, point-to-point real time collapse. The requirements for interactive, point-to-point real time
multimedia, which needs low-delay, semi-reliable data delivery, are multimedia, which needs low-delay, semi-reliable data delivery, are
different from the requirements for bulk transfer like FTP or bursty different from the requirements for bulk transfer like FTP or bursty
transfers like Web pages. 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 attempts to describe a set of requirements that can be This document attempts to describe a set of requirements that can be
used to evaluate other congestion control mechanisms in order to used to evaluate other congestion control mechanisms in order to
figure out their fitness for this purpose, and in particular to figure out their fitness for this purpose, and in particular to
provide a set of possible requirements for proposals coming out of provide a set of possible requirements for proposals coming out of
the RMCAT Working Group. the RMCAT Working Group.
Requirements Language Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
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This Internet-Draft will expire on October 21, 2014.
This Internet-Draft will expire on October 2, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3
3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
4. Security Considerations . . . . . . . . . . . . . . . . . . . 8 4. Security Considerations . . . . . . . . . . . . . . . . . . . 8
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.1. Normative References . . . . . . . . . . . . . . . . . . 8 6.1. Normative References . . . . . . . . . . . . . . . . . . 8
6.2. Informative References . . . . . . . . . . . . . . . . . 9 6.2. Informative References . . . . . . . . . . . . . . . . . 8
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9
1. Introduction 1. Introduction
The traditional TCP congestion control requirements were developed in Most of today's TCP congestion control schemes were developed with a
order to promote efficient use of the Internet for reliable bulk focus on an use of the Internet for reliable bulk transfer of non-
transfer of non-time-critical data, such as transfer of large files. time-critical data, such as transfer of large files. They have also
They have also been used successfully to govern the reliable transfer been used successfully to govern the reliable transfer of smaller
of smaller chunks of data in as short a time as possible, such as chunks of data in as short a time as possible, such as when fetching
when fetching Web pages. Web pages.
These algorithms have also been used for transfer of media streams These algorithms have also been used for transfer of media streams
that are viewed in a non-interactive manner, such as "streaming" that are viewed in a non-interactive manner, such as "streaming"
video, where having the data ready when the viewer wants it is video, where having the data ready when the viewer wants it is
important, but the exact timing of the delivery is not. important, but the exact timing of the delivery is not.
When doing real time interactive media, the requirements are When doing real time interactive media, the requirements are
different; one needs to provide the data continuously, within a very different; one needs to provide the data continuously, within a very
limited time window (no more than 100s of milliseconds end-to-end 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 delay), the sources of data may be able to adapt the amount of data
skipping to change at page 4, line 25 skipping to change at page 4, line 25
E. The algorithm must not overreact to short-term bursts (such E. The algorithm must not overreact to short-term bursts (such
as web-browsing) which can quickly saturate a local- as web-browsing) which can quickly saturate a local-
bottleneck router or link, but also clear quickly, and bottleneck router or link, but also clear quickly, and
should recover quickly when the burst ends. This is should recover quickly when the burst ends. This is
inherently at odds with the need to react quickly-enough to inherently at odds with the need to react quickly-enough to
avoid queue buildup. avoid queue buildup.
F. Similarly periodic bursty flows such as MPEG DASH F. Similarly periodic bursty flows such as MPEG DASH
[MPEG_DASH] or proprietary media streaming algorithms may [MPEG_DASH] or proprietary media streaming algorithms may
compete in bursts with the algorithm, and may not be compete in bursts with the algorithm, and may not be
adaptive within a burst. They are often are layered on top adaptive within a burst. They are often layered on top of
of TCP. The algorithm must avoid too much delay buildup TCP. The algorithm must avoid too much delay buildup during
during those bursts, and quickly recover. Note that this those bursts, and quickly recover. Note that this traffic
traffic may on an access link, or may cause a shift in the may on an access link, or may cause a shift in the location
location of the bottleneck for the duration of the burst. of the bottleneck for the duration of the burst.
2. The algorithm must be fair to other flows, both realtime flows 2. The algorithm must be fair to other flows, both realtime flows
(such as other instances of itself), and TCP flows, both long- (such as other instances of itself), and TCP flows, both long-
lived and bursts such as the traffic generated by a typical web lived and bursts such as the traffic generated by a typical web
browsing session. Note that 'fair' is a rather hard-to-define browsing session. Note that 'fair' is a rather hard-to-define
term. It should be self-fair with itself, giving roughly equal term. It should be self-fair with itself, giving roughly equal
bandwidth to multiple flows with similar RTTs, and if possible bandwidth to multiple flows with similar RTTs, and if possible
to multiple flows with different RTTs. to multiple flows with different RTTs.
A. Existing flows at a bottleneck must also be fair to new A. Existing flows at a bottleneck must also be fair to new
flows to that bottleneck, and must allow new flows to ramp flows to that bottleneck, and must allow new flows to ramp
up to a useful share of the bottleneck bandwidth quickly. up to a useful share of the bottleneck bandwidth quickly.
Note that relative RTTs may affect the rate new flows can Note that relative RTTs may affect the rate new flows can
ramp up to a reasonable share. ramp up to a reasonable share.
3. The algorithm should where possible merge information across 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 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 (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.
6. The algorithm should where possible merge information across
multiple RTP streams between the same endpoints, whether or not multiple RTP streams between the same endpoints, whether or not
they're multiplexed on the same ports, in order to allow they're multiplexed on the same ports, in order to allow
congestion control of the set of streams together instead of as congestion control of the set of streams together instead of as
multiple independent streams. This allows better overall multiple independent streams. This allows better overall
bandwidth management, faster response to changing conditions, bandwidth management, faster response to changing conditions,
and fairer sharing of bandwidth with other network users. and fairer sharing of bandwidth with other network users.
Alternatively, it should work with an external bandwidth control Alternatively, it should work with an external bandwidth control
framework to coordinate bandwidth usage across a bottleneck, framework to coordinate bandwidth usage across a bottleneck,
such as draft-welzl-rmcat-coupled-cc such as draft-welzl-rmcat-coupled-cc
[I-D.welzl-rmcat-coupled-cc]. [I-D.welzl-rmcat-coupled-cc].
A. If possible, it should also share information and adaptation A. If possible, it should also share information and adaptation
with other non-RTP flows between the same endpoints, such as with other non-RTP flows between the same endpoints, such as
a WebRTC DataChannel[I-D.ietf-rtcweb-data-channel] a WebRTC DataChannel[I-D.ietf-rtcweb-data-channel]
B. The most correlated bandwidth usage would be with other B. The most correlated bandwidth usage would be with other
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C. Use of information about previous flows, especially on the C. Use of information about previous flows, especially on the
same 5-tuple, may be useful input to the algorithm, same 5-tuple, may be useful input to the algorithm,
especially to startup performance of a new flow. especially to startup performance of a new flow.
D. When there are multiple streams across the same 5-tuple D. When there are multiple streams across the same 5-tuple
coordinating their bandwidth use and congestion control, it coordinating their bandwidth use and congestion control, it
should be possible for the application to control the should be possible for the application to control the
relative split of available bandwidth. relative split of available bandwidth.
4. The algorithm should not require any special support from 7. The algorithm should not require any special support from
network elements (ECN, etc). As much as possible, it should network elements (Explicit Congestion Notification (ECN)
leverage available information about the incoming flow to [RFC3168], etc). As much as possible, it should leverage
provide feedback to the sender. Examples of this information available information about the incoming flow to provide
are the ECN, packet arrival times, acknowledgments and feedback, feedback to the sender. Examples of this information are the
packet timestamps, and packet losses; all of these can provide 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. information about the state of the path and any bottlenecks.
A. Extra information could be added to the packets to provide A. Extra information could be added to the packets to provide
more detailed information on actual send times (as opposed more detailed information on actual send times (as opposed
to sampling times), but should not be required. to sampling times), but should not be required.
B. When additional input signals such as ECN are available, B. When additional input signals such as ECN are available,
they should be utilized if possible. they should be utilized if possible.
5. Since the assumption here is a set of RTP streams, the 8. Since the assumption here is a set of RTP streams, the
backchannel typically should be done via RTCP; one alternative backchannel typically should be done via RTCP; one alternative
would be to include it instead in a reverse RTP channel using would be to include it instead in a reverse RTP channel using
header extensions. header extensions.
A. In order to react sufficiently quickly when using RTCP for a A. In order to react sufficiently quickly when using RTCP for a
backchannel, an RTP profile such as RTP/AVPF [RFC4585] or backchannel, an RTP profile such as RTP/AVPF [RFC4585] or
RTP/SAVPF [RFC5124] that allows sufficiently frequent RTP/SAVPF [RFC5124] that allows sufficiently frequent
feedback MUST be used. feedback MUST be used.
B. Note that in some cases, backchannel messages may be delayed B. Note that in some cases, backchannel messages may be delayed
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used in a bidirectional set of flows). In areas of used in a bidirectional set of flows). In areas of
stability, backchannel data may be sent more infrequently so stability, backchannel data may be sent more infrequently so
long as algorithm stability and fairness are maintained. long as algorithm stability and fairness are maintained.
When the channel is unstable or has not yet reached When the channel is unstable or has not yet reached
equilibrium after a change, backchannel feedback may be more equilibrium after a change, backchannel feedback may be more
frequent and use more reverse-channel bandwidth. This is an frequent and use more reverse-channel bandwidth. This is an
area with considerable flexibility of design, and different area with considerable flexibility of design, and different
approaches to backchannel messages and frequency are approaches to backchannel messages and frequency are
expected to be evaluated. expected to be evaluated.
6. Flows managed by this algorithm and flows competing against at a 9. Flows managed by this algorithm and flows competing against at a
bottleneck may have different DSCP[RFC5865] markings depending bottleneck may have different DSCP[RFC5865] markings depending
on the type of traffic, or may be subject to flow-based QoS. A 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 particular bottleneck or section of the network path may or may
not honor DSCP markings. 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 A. In WebRTC, a division of packets into 4 classes is
envisioned in order of priority: faster-than-audio, audio, envisioned in order of priority: faster-than-audio, audio,
video, best-effort, and bulk-transfer. Typically the flows video, best-effort, and bulk-transfer. Typically the flows
managed by this algorithm would be audio or video in that managed by this algorithm would be audio or video in that
heirarchy, and feedback flows would be faster-than-audio. heirarchy, and feedback flows would be faster-than-audio.
7. The algorithm should sense the unexpected lack of backchannel 10. The algorithm should sense the unexpected lack of backchannel
information as a possible indication of a channel overuse information as a possible indication of a channel overuse
problem and react accordingly to avoid burst events causing a problem and react accordingly to avoid burst events causing a
congestion collapse. congestion collapse.
8. The algorithm should not starve competing TCP flows, and should 11. The algorithm should be stable and low-delay when faced with
as best as possible avoid starvation by TCP flows.
A. An algorithm may be more successful at avoiding starvation
from short-lived TCP long-lived/saturating TCP flows.
B. In order to avoid starvation other goals may need to be
compromised (such as delay).
9. The algorithm should be stable and low-delay when faced with
active queue management (AQM) algorithms. Also note that these active queue management (AQM) algorithms. Also note that these
algorithms may apply across multiple queues in the bottleneck, algorithms may apply across multiple queues in the bottleneck,
or to a single queue or to a single queue
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!"). 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).
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 3. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
4. Security Considerations 4. Security Considerations
An attacker with the ability to delete, delay or insert messages in An attacker with the ability to delete, delay or insert messages in
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"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006. 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008. (RTP/SAVPF)", RFC 5124, February 2008.
6.2. Informative References 6.2. Informative References
[I-D.ietf-ledbat-congestion]
Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", draft-
ietf-ledbat-congestion-10 (work in progress), September
2012.
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-07 (work in Channels", draft-ietf-rtcweb-data-channel-08 (work in
progress), February 2014. progress), April 2014.
[I-D.welzl-rmcat-coupled-cc] [I-D.welzl-rmcat-coupled-cc]
Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
control for RTP media", draft-welzl-rmcat-coupled-cc-02 control for RTP media", draft-welzl-rmcat-coupled-cc-02
(work in progress), October 2013. (work in progress), October 2013.
[MPEG_DASH] [MPEG_DASH]
"Dynamic adaptive streaming over HTTP (DASH) -- Part 1: "Dynamic adaptive streaming over HTTP (DASH) -- Part 1:
Media presentation description and segment formats", April Media presentation description and segment formats", April
2012. 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 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
[RFC5865] Baker, F., Polk, J., and M. Dolly, "A Differentiated [RFC5865] Baker, F., Polk, J., and M. Dolly, "A Differentiated
Services Code Point (DSCP) for Capacity-Admitted Traffic", Services Code Point (DSCP) for Capacity-Admitted Traffic",
RFC 5865, May 2010. RFC 5865, May 2010.
Author's Address Author's Address
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