draft-ietf-rmcat-cc-requirements-02.txt   draft-ietf-rmcat-cc-requirements-03.txt 
Network Working Group R. Jesup Network Working Group R. Jesup
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Informational February 14, 2014 Intended status: Informational March 31, 2014
Expires: August 18, 2014 Expires: October 2, 2014
Congestion Control Requirements For RMCAT Congestion Control Requirements For RMCAT
draft-ietf-rmcat-cc-requirements-02 draft-ietf-rmcat-cc-requirements-03
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.
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.
This document is derived from draft-jesup-rtp-congestion-reqs
[I-D.jesup-rtp-congestion-reqs].
Requirements Language Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119]. document are to be interpreted as described in RFC 2119 [RFC2119].
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 18, 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.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
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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 . . . . . . . . . . . . . . . . . . . . . 7 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
4. Security Considerations . . . . . . . . . . . . . . . . . . . 7 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 . . . . . . . . . . . . . . . . . 8 6.2. Informative References . . . . . . . . . . . . . . . . . 9
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9
1. Introduction 1. Introduction
The traditional TCP congestion control requirements were developed in The traditional TCP congestion control requirements were developed in
order to promote efficient use of the Internet for reliable bulk order to promote efficient use of the Internet for reliable bulk
transfer of non-time-critical data, such as transfer of large files. transfer of non-time-critical data, such as transfer of large files.
They have also been used successfully to govern the reliable transfer 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 of smaller chunks of data in as short a time as possible, such as
when fetching Web pages. when fetching Web pages.
skipping to change at page 3, line 6 skipping to change at page 3, line 5
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
that needs sending within fairly wide margins, and may tolerate some that needs sending within fairly wide margins, and may tolerate some
amount of packet loss, but since the data is generated in real time, 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 sending "future" data is impossible, and since it's consumed in real
time, data delivered late is useless. time, data delivered late is commonly useless.
While the requirements for RMCAT differ from the requirements for the While the requirements for RMCAT differ from the requirements for the
other flow types, these other flow types will be present in the other flow types, these other flow types will be present in the
network. The RMCAT congestion control algorithm must work properly network. The RMCAT congestion control algorithm must work properly
when these other flow types are present as cross traffic on the when these other flow types are present as cross traffic on the
network. network.
One particular protocol portofolio being developed for this use case One particular protocol portofolio being developed for this use case
is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending
multiple RTP-based flows between two peers, in conjunction with data multiple RTP-based flows between two peers, in conjunction with data
flows, all at the same time, without having special arrangements with flows, all at the same time, without having special arrangements with
the intervening service providers. the intervening service providers.
Given that this use case is the focus of this document, use cases Given that this use case is the focus of this document, use cases
involving noninteractive media such as YouTube-like video streaming, involving noninteractive media such as video streaming, and use cases
and use cases using multicast/broadcast-type technologies, are out of using multicast/broadcast-type technologies, are out of scope.
scope.
The terminology defined in [I-D.ietf-rtcweb-overview] is used in this The terminology defined in [I-D.ietf-rtcweb-overview] is used in this
memo. memo.
2. Requirements 2. Requirements
1. The congestion control algorithm must attempt to provide as-low- 1. The congestion control algorithm must attempt to provide as-low-
as-possible-delay transit for real-time traffic while still as-possible-delay transit for real-time traffic while still
providing a useful amount of bandwidth, even when faced with providing a useful amount of bandwidth. There may be lower
intermediate bottlenecks and competing flows. There may be limits on the amount of bandwidth that is useful, but this is
lower limits on the amount of bandwidth that is useful, but this largely application-specific and the application may be able to
is largely application-specific and the application may be able modify or remove flows in order allow some useful flows to get
to modify or remove flows in order allow some useful flows to enough bandwidth. (Example: not enough bandwidth for low-
get enough bandwidth. (Example: not enough bandwidth for low-
latency video+audio, but enough for audio-only.) latency video+audio, but enough for audio-only.)
A. It should also handle routing changes and interface changes A. It should provide this as-low-as-possible-delay transit even
(WiFi to 3G data, etc) which may radically change the when faced with intermediate bottlenecks and competing
bandwidth available, and react quickly, especially if there flows. Competing flows may limit what's possible to
is a reduction in available bandwidth. achieve.
B. The offered load may be less than the available bandwidth at B. 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 bottleneck delay.
C. It should handle interface changes (WiFi 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.
D. The offered load may be less than the available bandwidth at
any given moment, and may vary dramatically over time, any given moment, and may vary dramatically over time,
including dropping to no load and then resuming a high load, including dropping to no load and then resuming a high load,
such as in a mute operation. The reaction time between a such as in a mute operation. The reaction time between a
change in the bandwidth available from the algorithm and a change in the bandwidth available from the algorithm and a
change in the offered load is variable, and may be different change in the offered load is variable, and may be different
when increasing versus decreasing. when increasing versus decreasing.
C. 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.
D. Similarly periodic bursty flows such as DASH or proprietary F. Similarly periodic bursty flows such as MPEG DASH
media streaming algorithms may compete in bursts with the [MPEG_DASH] or proprietary media streaming algorithms may
algorithm, and may not be adaptive within a burst. They are compete in bursts with the algorithm, and may not be
often are layered on top of TCP. The algorithm must avoid adaptive within a burst. They are often are layered on top
too much delay buildup during those bursts, and quickly of TCP. The algorithm must avoid too much delay buildup
recover. Note that this traffic may on an access link, or during those bursts, and quickly recover. Note that this
may cause a shift in the location of the bottleneck fir the traffic may on an access link, or may cause a shift in the
duration of the burst. location 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. term. It should be self-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 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
ramp up to a reasonable share.
3. The algorithm should where possible merge information across 3. 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 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
flows on the same 5-tuple, but there may be use in flows on the same 5-tuple, but there may be use in
coordinating measurement and control of the local link(s). coordinating measurement and control of the local link(s).
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
coordinating their bandwidth use and congestion control, it
should be possible for the application to control the
relative split of available bandwidth.
4. The algorithm should not require any special support from 4. The algorithm should not require any special support from
network elements (ECN, etc). As much as possible, it should network elements (ECN, etc). As much as possible, it should
leverage available information about the incoming flow to leverage available information about the incoming flow to
provide feedback to the sender. Examples of this information provide feedback to the sender. Examples of this information
are the ECN, packet arrival times, acknowledgments and feedback, are the ECN, packet arrival times, acknowledgments and feedback,
packet timestamps, and packet losses; all of these can provide 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
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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 5. 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 AVPF/SAVPF that allows backchannel, an RTP profile such as RTP/AVPF [RFC4585] or
sufficiently frequent feedback [RFC4585] MUST be used. RTP/SAVPF [RFC5124] that allows sufficiently frequent
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
until the RTCP channel can be allocated enough bandwidth, until the RTCP channel can be allocated enough bandwidth,
even under AVPF rules. This may also imply negotiating a even under AVPF rules. This may also imply negotiating a
higher maximum percentage for RTCP data or allowing RMCAT higher maximum percentage for RTCP data or allowing RMCAT
solutions to violate or modify the rules specified for AVPF. solutions to violate or modify the rules specified for AVPF.
C. Bandwidth for the feedback messages should be minimized C. Bandwidth for the feedback messages should be minimized
(such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR) (such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR)
skipping to change at page 6, line 9 skipping to change at page 6, line 30
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 competed against at a 6. Flows managed by this algorithm and flows competing against at a
bottleneck may have different DSCP markings depending on the bottleneck may have different DSCP[RFC5865] markings depending
type of traffic. A particular bottleneck or section of the on the type of traffic, or may be subject to flow-based QoS. A
network path may or may not honor these markings. particular bottleneck or section of the network path may or may
not honor DSCP markings.
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 7. 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
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This document is the result of discussions in various fora of the This document is the result of discussions in various fora of the
WebRTC effort, in particular on the rtp-congestion@alvestrand.no WebRTC effort, in particular on the rtp-congestion@alvestrand.no
mailing list. Many people contributed their thoughts to this. mailing list. Many people contributed their thoughts to this.
6. References 6. References
6.1. Normative References 6.1. Normative References
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-08 (work based Applications", draft-ietf-rtcweb-overview-09 (work
in progress), September 2013. in progress), February 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"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
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
6.2. Informative References 6.2. Informative References
[I-D.ietf-ledbat-congestion] [I-D.ietf-ledbat-congestion]
Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", draft- "Low Extra Delay Background Transport (LEDBAT)", draft-
ietf-ledbat-congestion-10 (work in progress), September ietf-ledbat-congestion-10 (work in progress), September
2012. 2012.
[I-D.jesup-rtp-congestion-reqs] [I-D.ietf-rtcweb-data-channel]
Jesup, R. and H. Alvestrand, "Congestion Control Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Requirements For Real Time Media", draft-jesup-rtp- Channels", draft-ietf-rtcweb-data-channel-07 (work in
congestion-reqs-00 (work in progress), March 2012. progress), February 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]
"Dynamic adaptive streaming over HTTP (DASH) -- Part 1:
Media presentation description and segment formats", April
2012.
[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
Services Code Point (DSCP) for Capacity-Admitted Traffic",
RFC 5865, May 2010.
Author's Address Author's Address
Randell Jesup Randell Jesup
Mozilla Mozilla
USA USA
Email: randell-ietf@jesup.org Email: randell-ietf@jesup.org
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