draft-ietf-rmcat-cc-requirements-07.txt   draft-ietf-rmcat-cc-requirements-08.txt 
Network Working Group R. Jesup Network Working Group R. Jesup
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Informational Z. Sarker, Ed. Intended status: Informational Z. Sarker, Ed.
Expires: April 30, 2015 Ericsson Expires: May 15, 2015 Ericsson
October 27, 2014 November 11, 2014
Congestion Control Requirements for Interactive Real-Time Media Congestion Control Requirements for Interactive Real-Time Media
draft-ietf-rmcat-cc-requirements-07 draft-ietf-rmcat-cc-requirements-08
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. Due to an increasing amount of RTP-based transfers like Web pages. Due to an increasing amount of RTP-based
real-time media traffic on the Internet (e.g. with the introduction real-time media traffic on the Internet (e.g. with the introduction
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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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 April 30, 2015. This Internet-Draft will expire on May 15, 2015.
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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publication of this document. Please review these documents publication of this document. Please review these documents
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
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. Deficiencies of existing mechanisms . . . . . . . . . . . . . 8 3. Deficiencies of existing mechanisms . . . . . . . . . . . . . 8
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
5. Security Considerations . . . . . . . . . . . . . . . . . . . 9 5. Security Considerations . . . . . . . . . . . . . . . . . . . 9
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9 6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 10
7. References . . . . . . . . . . . . . . . . . . . . . . . . . 10 7. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
7.1. Normative References . . . . . . . . . . . . . . . . . . 10 7.1. Normative References . . . . . . . . . . . . . . . . . . 10
7.2. Informative References . . . . . . . . . . . . . . . . . 10 7.2. Informative References . . . . . . . . . . . . . . . . . 10
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11
1. Introduction 1. Introduction
Most of today's TCP congestion control schemes were developed with a 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- 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 time-critical data, such as transfer of large files. They have also
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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 but can be rate limited that needs sending within fairly wide margins but can be rate limited
by the application- even not always have data to send, and may by the application- even not always have data to send, and may
tolerate some amount of packet loss, but since the data is generated tolerate some amount of packet loss, but since the data is generated
in real-time, sending "future" data is impossible, and since it's in real-time, sending "future" data is impossible, and since it's
consumed in real-time, data delivered late is commonly useless. consumed in real-time, data delivered late is commonly useless.
While the requirements for real-time interactive differ from the While the requirements for real-time interactive media differ from
requirements for the other flow types, these other flow types will be the requirements for the other flow types, these other flow types
present in the network. The congestion control algorithm for real- will be present in the network. The congestion control algorithm for
time interactive media must work properly when these other flow types real-time interactive media must work properly when these other flow
are present as cross traffic on the network. types are present as cross traffic on the network.
One particular protocol portofolio being developed for this use case One particular protocol portfolio 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 flows using the Real-time Transport Protocol (RTP) [RFC3550] multiple flows using the Real-time Transport Protocol (RTP) [RFC3550]
between two peers, in conjunction with data flows, all at the same between two peers, in conjunction with data flows, all at the same
time, without having special arrangements with the intervening time, without having special arrangements with the intervening
service providers. service providers. As RTP does not provide any congestion control
mechanism; a set of circuit breakers, such as
[I-D.ietf-avtcore-rtp-circuit-breakers], are required to protect the
network from excessive congestion caused by the non-congestion
controlled flows. When the real-time interactive media is congestion
controlled, it is recommended that the congestion control mechanism
operates within the constraints defined by these circuit breakers
when circuit breaker is present and that it should not cause
congestion collapse when circuit breaker is not implemented.
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 non-interactive media such as video streaming, and use involving non-interactive media such as video streaming, and use
cases using multicast/broadcast-type technologies, are out of scope. cases using multicast/broadcast-type technologies, are out of 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
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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 interactive real-time traffic as-possible-delay transit for interactive real-time traffic
while still providing a useful amount of bandwidth. There may while still providing a useful amount of bandwidth. There may
be lower limits on the amount of bandwidth that is useful, but be lower limits on the amount of bandwidth that is useful, but
this is largely application-specific and the application may be this is largely application-specific and the application may be
able to modify or remove flows in order allow some useful flows able to modify or remove flows in order allow some useful flows
to get enough bandwidth. (Example: not enough bandwidth for to get enough bandwidth. (Example: not enough bandwidth for
low-latency video+audio, but enough for audio-only.) low-latency video+audio, but enough for audio-only.)
A. Jitter (variation in the bitrate over short timescales) also A. Jitter (variation in the bitrate over short timescales) also
is relevant, though moderate amounts of jitter will be is relevant, though moderate amounts of jitter will be
absorbed by jitter buffers. Transit delay should be absorbed by jitter buffers. Transit delay should be
considered to track the short-term maximums of delay considered to track the short-term maximums of delay
including jitter. including jitter.
B. It should provide this as-low-as-possible-delay transit even B. It should provide this as-low-as-possible-delay transit and
when faced with intermediate bottlenecks and competing minimize self-induced latency even when faced with
flows. Competing flows may limit what's possible to intermediate bottlenecks and competing flows. Competing
achieve. flows may limit what's possible to achieve.
C. It should handle routing changes which may alter or remove C. It should handle the effect of routing changes which may
bottlenecks or change the bandwidth available especially if alter or remove bottlenecks or change the bandwidth
there is a reduction in available bandwidth or increase in available especially if there is a reduction in available
observed delay. It is expected that the mechanism reacts to bandwidth or increase in observed delay. It is expected
the routing change events in a way that avoids delay that the mechanism reacts quickly to the routing change
buildup. events to avoid delay buildup. In the context of this memo,
a 'quick' reaction is on the order of a few RTTs, subject to
the constraints of the media codec, but is likely within a
second. Reaction on the next RTT is explicitly not
required, since many codecs cannot adapt their sending rate
that quickly, but equally response cannot be arbitrarily
delayed.
D. It should handle both local and remote interface changes D. It should react quickly to handle both local and remote
(WLAN to 3G data, etc) which may radically change the interface changes (WLAN to 3G data, etc) which may radically
bandwidth available or bottlenecks, especially if there is a change the bandwidth available or bottlenecks, especially if
reduction in available bandwidth or increase in bottleneck there is a reduction in available bandwidth or increase in
delay. It is assumed that an interface change can generate bottleneck delay. It is assumed that an interface change
a notification to the algorithm. can generate a notification to the algorithm.
E. The algorithm must consider the case where offered loads are E. The algorithm must consider the case where offered loads are
less than the available bandwidth at any given moment, and less than the available bandwidth at any given moment, and
may vary dramatically over time, including dropping to no may vary dramatically over time, including dropping to no
load and then resuming a high load, such as in a mute/unmute load and then resuming a high load, such as in a mute/unmute
operation. Note that the reaction time between a change in operation. Note that 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 when the offered load is variable, and may be different when
increasing versus decreasing. increasing versus decreasing.
F. The algorithm requires to avoid building up queues when F. The algorithm requires to avoid building up queues when
competing with short-term bursts of traffic (for example, competing with short-term bursts of traffic (for example,
traffic generated by web-browsing) which can quickly traffic generated by web-browsing) which can quickly
saturate a local-bottleneck router or link, but also clear saturate a local-bottleneck router or link, but also clear
quickly. The algorithm should also attempt to regain its quickly. The algorithm should also react quickly to regain
previous share of the bandwidth when the local-bottleneck or its previous share of the bandwidth when the local-
link is cleared. bottleneck or link is cleared.
G. Similarly periodic bursty flows such as MPEG DASH G. 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 layered on top of adaptive within a burst. They are often layered on top of
TCP. Due to non-adaptiveness of the competing traffic as TCP but use TCP in a bursty manner that can interact poorly
such, the algorithm must not increase the already existing with competing flows during the bursts. The algorithm must
delay buildup during those bursts. Note that this competing not increase the already existing delay buildup during those
traffic may on a shared access link, or the traffic burst bursts. Note that this competing traffic may be on a shared
may cause a shift in the location of the bottleneck for the access link, or the traffic burst 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 real-time flows 2. The algorithm must be fair to other flows, both real-time 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 fair with itself, giving fair share of the term. It should be fair with itself, giving fair share of the
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 as quickly up to a useful share of the bottleneck bandwidth as quickly
as possible. A useful share will depend on the media types as possible. A useful share will depend on the media types
involved and total bandwidth available. Note that relative involved, total bandwidth available and the user experience
requirements of a particular service. Note that relative
RTTs may affect the rate new flows can ramp up to a RTTs may affect the rate new flows can ramp up to a
reasonable share. reasonable share.
3. The algorithm should not starve competing TCP flows, and should 3. The algorithm should not starve competing TCP flows, and should
as best as possible avoid starvation by TCP flows. as best as possible avoid starvation by TCP flows.
A. The congestion control should prioritise achieving a useful A. The congestion control should prioritise achieving a useful
share of the bandwidth depending on the media types and share of the bandwidth depending on the media types and
total available bandwidth over achieving as low as possible total available bandwidth over achieving as low as possible
transit delay, when these two requirements are in conflict. transit delay, when these two requirements are in conflict.
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and control of the local link(s). Use of information about and control of the local link(s). Use of information about
previous flows, especially on the same 5-tuple, may be previous flows, especially on the same 5-tuple, may be
useful input to the algorithm, especially to startup useful input to the algorithm, especially to startup
performance of a new flow. performance of a new flow.
7. The algorithm should not require any special support from 7. The algorithm should not require any special support from
network elements to convey congestion related information to be network elements to convey congestion related information to be
functional. As much as possible, it should leverage available functional. As much as possible, it should leverage available
information about the incoming flow to provide feedback to the information about the incoming flow to provide feedback to the
sender. Examples of this information are the packet arrival sender. Examples of this information are the packet arrival
times, acknowledgments and feedback, packet timestamps, and times, acknowledgements and feedback, packet timestamps, and
packet losses, Explicit Congestion Notification (ECN) [RFC3168]; packet losses, Explicit Congestion Notification (ECN) [RFC3168];
all of these can provide information about the state of the path all of these can provide information about the state of the path
and any bottlenecks. However, the use of available information and any bottlenecks. However, the use of available information
is algorithm dependent. is algorithm dependent.
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.
8. Since the assumption here is a set of RTP streams, the 8. Since the assumption here is a set of RTP streams, the
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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
hierarchy, and feedback flows would be faster-than-audio. hierarchy, and feedback flows would be faster-than-audio.
10. 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.
11. The algorithm should be stable and low-delay when faced with 11. The algorithm should be stable and maintain low-delay when faced
Active Queue Management (AQM) algorithms. Also note that these with Active Queue Management (AQM) algorithms. Also note that
algorithms may apply across multiple queues in the bottleneck, these algorithms may apply across multiple queues in the
or to a single queue bottleneck, or to a single queue
3. Deficiencies of existing mechanisms 3. Deficiencies of existing mechanisms
Among the existing congestion control mechanisms TCP Friendly Rate Among the existing congestion control mechanisms TCP Friendly Rate
Control (TFRC) [RFC5348] is the one which claims to be suitable for Control (TFRC) [RFC5348] is the one which claims to be suitable for
real-time interactive media. TFRC is, an equation based, congestion real-time interactive media. TFRC is, an equation based, congestion
control mechanism which provides reasonably fair share of the control mechanism which provides reasonably fair share of the
bandwidth when competing with TCP flows and offers much lower bandwidth when competing with TCP flows and offers much lower
throughput variations than TCP. This is achieved by a slower throughput variations than TCP. This is achieved by a slower
response to the available bandwidth change than TCP. TFRC is response to the available bandwidth change than TCP. TFRC is
designed to perform best with applications which has fixed packet designed to perform best with applications which has fixed packet
size and does not have fixed period between sending packets. size and does not have fixed period between sending packets.
TFRC operates on detecting loss events and reacts to loss caused by TFRC operates on detecting loss events and reacts to loss caused by
congestion by reducing its sending rate. It allows applications to congestion by reducing its sending rate. It allows applications to
increase the sending rate until loss is observed in the flows. As it increase the sending rate until loss is observed in the flows. As it
is noted in IAB/IRTF report [RFC7295] large buffers are available in is noted in IAB/IRTF report [RFC7295] large buffers are available in
the network elements which introduces additional delay in the the network elements which introduces additional delay in the
communication, it becomes important to take all possible congestion communication, it becomes important to take all possible congestion
indications into considerations. TFRC's only consideration of loss indications into considerations. Looking at the current Internet
events as congestion indication can be considered as biggest lacking deployment, TFRC's only consideration of loss events as congestion
looking at the current Internet deployment. indication can be considered as biggest lacking.
A typical real-time interactive communication includes live encoded A typical real-time interactive communication includes live encoded
audio and video flow(s). In such a communication scenario an audio audio and video flow(s). In such a communication scenario an audio
source typically needs fixed interval between packets, needs to vary source typically needs fixed interval between packets, needs to vary
their segment size instead of their packet rate in response to their segment size instead of their packet rate in response to
congestion and sends smaller packets, a variance of TFRC , Small- congestion and sends smaller packets, a variant of TFRC , Small-
Packet TFRC (TFRC-SP) [RFC4828] addresses the issues related to such Packet TFRC (TFRC-SP) [RFC4828] addresses the issues related to such
kind of sources ; a video source generally varies video frame sizes, kind of sources ; a video source generally varies video frame sizes,
can produce large frames which need to be further fragmented to fit can produce large frames which need to be further fragmented to fit
into path Maximum Transmission Unit (MTU) size, and have almost fixed into path Maximum Transmission Unit (MTU) size, and have almost fixed
interval between producing frames under a certain frame rate, TFRC is interval between producing frames under a certain frame rate, TFRC is
known to be less optimal when using with such video sources. known to be less optimal when using with such video sources.
There are also some mismatches between TFRC's design assumptions and There are also some mismatches between TFRC's design assumptions and
how the media sources in a typical real-time interactive application how the media sources in a typical real-time interactive application
works. TFRC is design to maintain smooth sending rate however media works. TFRC is design to maintain smooth sending rate however media
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5. Security Considerations 5. 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
the flow can fake congestion signals, unless they are passed on a the flow can fake congestion signals, unless they are passed on a
tamper-proof path. Since some possible algorithms depend on the tamper-proof path. Since some possible algorithms depend on the
timing of packet arrival, even a traditional protected channel does timing of packet arrival, even a traditional protected channel does
not fully mitigate such attacks. not fully mitigate such attacks.
An attack that reduces bandwidth is not necessarily significant, An attack that reduces bandwidth is not necessarily significant,
since an on-path attacker could break the connection by discarding since an on-path attacker could break the connection by discarding
all packets. Attacks that increase the percieved available bandwidth all packets. Attacks that increase the perceived available bandwidth
are concievable, and need to be evaluated. are conceivable, and need to be evaluated.
Algorithm designers should consider the possibility of malicious on- Algorithm designers should consider the possibility of malicious on-
path attackers. path attackers.
6. Acknowledgements 6. Acknowledgements
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.
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[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.
7.2. Informative References 7.2. Informative References
[CH09] Choi, S. and M. Handley, "Designing TCP-Friendly Window- [CH09] Choi, S. and M. Handley, "Designing TCP-Friendly Window-
based Congestion Control for Real-time Multimedia based Congestion Control for Real-time Multimedia
Applications", PFLDNeT 2009 Workshop , May 2009. Applications", PFLDNeT 2009 Workshop , May 2009.
[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-07 (work in progress),
October 2014.
[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-12 (work in Channels", draft-ietf-rtcweb-data-channel-12 (work in
progress), September 2014. progress), September 2014.
[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.
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