draft-ietf-rmcat-cc-requirements-06.txt   draft-ietf-rmcat-cc-requirements-07.txt 
Network Working Group R. Jesup Network Working Group R. Jesup
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Informational October 7, 2014 Intended status: Informational Z. Sarker, Ed.
Expires: April 10, 2015 Expires: April 30, 2015 Ericsson
October 27, 2014
Congestion Control Requirements For RMCAT Congestion Control Requirements for Interactive Real-Time Media
draft-ietf-rmcat-cc-requirements-06 draft-ietf-rmcat-cc-requirements-07
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
of WebRTC[I-D.ietf-rtcweb-overview]), it is especially important to of the Web Real-Time Communication (WebRTC)), it is especially
ensure that this kind of traffic is congestion controlled. important to ensure that this kind of traffic is congestion
controlled.
This document describes a set of requirements that can be used to This document describes a set of requirements that can be used to
evaluate other congestion control mechanisms in order to figure out evaluate other congestion control mechanisms in order to figure out
their fitness for this purpose, and in particular to provide a set of their fitness for this purpose, and in particular to provide a set of
possible requirements for realtime media congestion avoidance possible requirements for real-time media congestion avoidance
technique. technique.
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].
The terms are presented in many cases using lowercase for The terms are presented in many cases using lowercase for
readability. readability.
skipping to change at page 2, line 7 skipping to change at page 2, line 10
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 April 10, 2015. This Internet-Draft will expire on April 30, 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
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
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. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 3. Deficiencies of existing mechanisms . . . . . . . . . . . . . 8
4. Security Considerations . . . . . . . . . . . . . . . . . . . 8 4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 5. Security Considerations . . . . . . . . . . . . . . . . . . . 9
6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9
6.1. Normative References . . . . . . . . . . . . . . . . . . 8 7. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
6.2. Informative References . . . . . . . . . . . . . . . . . 9 7.1. Normative References . . . . . . . . . . . . . . . . . . 10
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9 7.2. Informative References . . . . . . . . . . . . . . . . . 10
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
been used successfully to govern the reliable transfer of smaller been used successfully to govern the reliable transfer of smaller
chunks of data in as short a time as possible, such as when fetching chunks of data in as short a time as possible, such as 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
that needs sending within fairly wide margins, and may tolerate some that needs sending within fairly wide margins but can be rate limited
amount of packet loss, but since the data is generated in real time, by the application- even not always have data to send, and may
sending "future" data is impossible, and since it's consumed in real tolerate some amount of packet loss, but since the data is generated
time, data delivered late is commonly useless. 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 While the requirements for real-time interactive differ from the
other flow types, these other flow types will be present in the requirements for the other flow types, these other flow types will be
network. The RMCAT congestion control algorithm must work properly present in the network. The congestion control algorithm for real-
when these other flow types are present as cross traffic on the time interactive media must work properly when these other flow types
network. are present as cross traffic on the 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 flows using the Real-time Transport Protocol (RTP) [RFC3550]
flows, all at the same time, without having special arrangements with between two peers, in conjunction with data flows, all at the same
the intervening service providers. time, without having special arrangements with 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 video streaming, and use cases involving non-interactive media such as video streaming, and use
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
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 interactive real-time traffic
providing a useful amount of bandwidth. There may be lower while still providing a useful amount of bandwidth. There may
limits on the amount of bandwidth that is useful, but this is be lower limits on the amount of bandwidth that is useful, but
largely application-specific and the application may be able to this is largely application-specific and the application may be
modify or remove flows in order allow some useful flows to get able to modify or remove flows in order allow some useful flows
enough bandwidth. (Example: not enough bandwidth for low- to get enough bandwidth. (Example: not enough bandwidth for
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 even
when faced with intermediate bottlenecks and competing when faced with intermediate bottlenecks and competing
flows. Competing flows may limit what's possible to flows. Competing flows may limit what's possible to
achieve. achieve.
C. It should handle routing changes which may alter or remove C. It should handle routing changes which may alter or remove
bottlenecks or change the bandwidth available, and react bottlenecks or change the bandwidth available especially if
quickly, especially if there is a reduction in available there is a reduction in available bandwidth or increase in
bandwidth or increase in observed delay. observed delay. It is expected that the mechanism reacts to
the routing change events in a way that avoids delay
buildup.
D. It should handle interface changes (WLAN to 3G data, etc) D. It should handle both local and remote interface changes
which may radically change the bandwidth available or (WLAN to 3G data, etc) which may radically change the
bottlenecks, and react quickly, especially if there is a bandwidth available or bottlenecks, especially if there is a
reduction in available bandwidth or increase in bottleneck reduction in available bandwidth or increase in bottleneck
delay. It is assumed that an interface change can generate delay. It is assumed that an interface change can generate
a notification to the algorithm. a notification to the algorithm.
E. The offered load may be less than the available bandwidth at E. The algorithm must consider the case where offered loads are
any given moment, and may vary dramatically over time, less than the available bandwidth at any given moment, and
including dropping to no load and then resuming a high load, may vary dramatically over time, including dropping to no
such as in a mute operation. The reaction time between a load and then resuming a high load, such as in a mute/unmute
change in the bandwidth available from the algorithm and a operation. Note that the reaction time between a change in
change in the offered load is variable, and may be different the bandwidth available from the algorithm and a change in
when increasing versus decreasing. the offered load is variable, and may be different when
increasing versus decreasing.
F. The algorithm must not overreact to short-term bursts (such F. The algorithm requires to avoid building up queues when
as web-browsing) which can quickly saturate a local- competing with short-term bursts of traffic (for example,
bottleneck router or link, but also clear quickly, and traffic generated by web-browsing) which can quickly
should recover quickly when the burst ends. This is saturate a local-bottleneck router or link, but also clear
inherently at odds with the need to react quickly-enough to quickly. The algorithm should also attempt to regain its
avoid queue buildup. previous share of the bandwidth when the local-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. The algorithm must avoid too much delay buildup during TCP. Due to non-adaptiveness of the competing traffic as
those bursts, and quickly recover. Note that this competing such, the algorithm must not increase the already existing
delay buildup during those bursts. Note that this competing
traffic may on a shared access link, or the traffic burst traffic may on a shared access link, or the traffic burst
may cause a shift in the location of the bottleneck for the may cause a shift in the location of the bottleneck for the
duration of the burst. 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 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 roughly equal 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 quickly. up to a useful share of the bottleneck bandwidth as quickly
as possible. A useful share will depend on the media types
Note that relative RTTs may affect the rate new flows can involved and total bandwidth available. Note that relative
ramp up to a reasonable share. 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 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. An algorithm may be more successful at avoiding starvation A. The congestion control should prioritise achieving a useful
from short-lived TCP than long-lived/saturating TCP flows. share of the bandwidth depending on the media types and
total available bandwidth over achieving as low as possible
B. In order to avoid starvation other goals may need to be transit delay, when these two requirements are in conflict.
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 4. The algorithm should as quickly as possible adapt to initial
setup with dummy data may be useful in some applications; in network conditions at the start of a flow. This should occur
some cases there will be a considerable gap in time between both if the initial bandwidth is above or below the bottleneck
flow creation and the initial flow of data. bandwidth.
C. A flow may need to change adaptation rates due to network A. The algorithm should allow different modes of adaptation for
conditions or changes in the provided flows (such as un- example,the startup adaptation may be faster than adaptation
muting or sending data after a gap). 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. 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. Again, 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 5. The algorithm should be stable if the RTP streams are halted or
discontinuous (Voice Activity Detection/Discontinuous discontinuous (for example - Voice Activity Detection).
Transmission).
A. After a resumption of RTP data it may adapt more quickly A. After stream resumption, the algorithm should attempt to
(similar to the start of a flow), and previous bandwidth rapidly regain its previous share of the bandwidth; the
estimates may need to be aged or thrown away. aggressiveness with which this is done will decay with the
length of the pause.
6. The algorithm should where possible merge information across 6. The algorithm should where possible merge information across
multiple RTP streams between the same endpoints, whether or not multiple RTP streams sent between two endpoints, when those RTP
they're multiplexed on the same ports, in order to allow streams share a common bottleneck, whether or not those streams
are multiplexed onto 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
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 A. The algorithm 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], when
possible.
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 B. 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, the
should be possible for the application to control the algorithm should allow the application to control the
relative split of available bandwidth. relative split of available bandwidth.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). 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.
7. The algorithm should not require any special support from 7. The algorithm should not require any special support from
network elements (Explicit Congestion Notification (ECN) network elements to convey congestion related information to be
[RFC3168], etc). As much as possible, it should leverage functional. As much as possible, it should leverage available
available information about the incoming flow to provide information about the incoming flow to provide feedback to the
feedback to the sender. Examples of this information are the sender. Examples of this information are the packet arrival
ECN, packet arrival times, acknowledgments and feedback, packet times, acknowledgments and feedback, packet timestamps, and
timestamps, and packet losses; all of these can provide packet losses, Explicit Congestion Notification (ECN) [RFC3168];
information about the state of the path and any bottlenecks. all of these can provide information about the state of the path
and any bottlenecks. However, the use of available information
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.
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 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[RFC3550]; one
would be to include it instead in a reverse RTP channel using alternative would be to include it instead in a reverse RTP
header extensions. channel using 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. Note that in some cases, backchannel
messages may be delayed until the RTCP channel can be
B. Note that in some cases, backchannel messages may be delayed allocated enough bandwidth, even under AVPF rules. This may
until the RTCP channel can be allocated enough bandwidth, also imply negotiating a higher maximum percentage for RTCP
even under AVPF rules. This may also imply negotiating a data or allowing solutions to violate or modify the rules
higher maximum percentage for RTCP data or allowing RMCAT specified for AVPF.
solutions to violate or modify the rules specified for AVPF.
C. Bandwidth for the feedback messages should be minimized B. Bandwidth for the feedback messages should be minimized
(such as via RFC 5506 [RFC5506]to allow RTCP without Sender (such as via RFC 5506 [RFC5506]to allow RTCP without Sender
Reports/Receiver Reports) Reports/Receiver Reports)
D. Header extensions would avoid the RTCP timing rules issues, C. Backchannel data should be minimized to avoid taking too
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 much reverse-channel bandwidth (since this will often be
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.
skipping to change at page 7, line 44 skipping to change at page 7, line 47
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. The algorithm should attempt to not honor DSCP markings. The algorithm should attempt to
leverage DSCP markings when they're available. 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. 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 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
3. IANA Considerations 3. Deficiencies of existing mechanisms
Among the existing congestion control mechanisms TCP Friendly Rate
Control (TFRC) [RFC5348] is the one which claims to be suitable for
real-time interactive media. TFRC is, an equation based, congestion
control mechanism which provides reasonably fair share of the
bandwidth when competing with TCP flows and offers much lower
throughput variations than TCP. This is achieved by a slower
response to the available bandwidth change than TCP. TFRC is
designed to perform best with applications which has fixed packet
size and does not have fixed period between sending packets.
TFRC operates on detecting loss events and reacts to loss caused by
congestion by reducing its sending rate. It allows applications to
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
the network elements which introduces additional delay in the
communication, it becomes important to take all possible congestion
indications into considerations. TFRC's only consideration of loss
events as congestion indication can be considered as biggest lacking
looking at the current Internet deployment.
A typical real-time interactive communication includes live encoded
audio and video flow(s). In such a communication scenario an audio
source typically needs fixed interval between packets, needs to vary
their segment size instead of their packet rate in response to
congestion and sends smaller packets, a variance of TFRC , Small-
Packet TFRC (TFRC-SP) [RFC4828] addresses the issues related to such
kind of sources ; a video source generally varies video frame sizes,
can produce large frames which need to be further fragmented to fit
into path Maximum Transmission Unit (MTU) size, and have almost fixed
interval between producing frames under a certain frame rate, TFRC is
known to be less optimal when using with such video sources.
There are also some mismatches between TFRC's design assumptions and
how the media sources in a typical real-time interactive application
works. TFRC is design to maintain smooth sending rate however media
sources can change rates in steps for both rate increase and rate
decrease. TFRC can operate in two modes - i) Bytes per second and
ii) packets per second, where typical real-time interactive media
sources operates on bit per second. There are also limitations on
how quickly the media sources can adapt to specific sending rates.
The modern video encoders can operate on a mode where they can vary
the output bitrate a lot depending on the way there are configured,
the current scene it is encoding and more. Therefore, it is possible
that the video source does not always output at a bitrate they are
allowed to. TFRC tries to raise its sending rate when transmitting
at maximum allowed rate and increases only twice the current
transmission rate hence it may create issues when the video source
vary their bitrates.
Moreover, there are number of studies on TFRC which shows it's
limitations which includes TFRC's unfairness on low statistically
multiplexed links, oscillatory behavior, performance issue in highly
dynamic loss rates conditions and more [CH09].
Looking at all these deficiencies it can be concluded that the
requirements of congestion control mechanism for real-time
interactive media cannot be met by TFRC as defined in the standard.
4. 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 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 percieved available bandwidth
are concievable, and need to be evaluated. are concievable, 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.
5. 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.
6. References 7. References
6.1. Normative References 7.1. Normative References
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-11 Browser-based Applications", draft-ietf-rtcweb-overview-12
(work in progress), August 2014. (work in progress), October 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.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[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 [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 7.2. Informative References
[CH09] Choi, S. and M. Handley, "Designing TCP-Friendly Window-
based Congestion Control for Real-time Multimedia
Applications", PFLDNeT 2009 Workshop , May 2009.
[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.
[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] [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 [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001. 3168, September 2001.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828, April
2007.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, September 2008.
[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 [RFC7295] Tschofenig, H., Eggert, L., and Z. Sarker, "Report from
the IAB/IRTF Workshop on Congestion Control for
Interactive Real-Time Communication", RFC 7295, July 2014.
Authors' Addresses
Randell Jesup Randell Jesup
Mozilla Mozilla
USA USA
Email: randell-ietf@jesup.org Email: randell-ietf@jesup.org
Zaheduzzaman Sarker (editor)
Ericsson
Sweden
Email: zaheduzzaman.sarker@ericsson.com
 End of changes. 51 change blocks. 
160 lines changed or deleted 235 lines changed or added

This html diff was produced by rfcdiff 1.41. The latest version is available from http://tools.ietf.org/tools/rfcdiff/