draft-ietf-rmcat-cc-requirements-09.txt   rfc8836.txt 
Network Working Group R. Jesup Internet Engineering Task Force (IETF) R. Jesup
Internet-Draft Mozilla Request for Comments: 8836 Mozilla
Intended status: Informational Z. Sarker, Ed. Category: Informational Z. Sarker, Ed.
Expires: June 15, 2015 Ericsson ISSN: 2070-1721 Ericsson AB
December 12, 2014 January 2021
Congestion Control Requirements for Interactive Real-Time Media Congestion Control Requirements for Interactive Real-Time Media
draft-ietf-rmcat-cc-requirements-09
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 the Web Real-Time Communication (WebRTC)), it is especially of the Web Real-Time Communication (WebRTC)), it is especially
important to ensure that this kind of traffic is congestion important to ensure that this kind of traffic is congestion
controlled. 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 real-time media congestion avoidance possible requirements for a real-time media congestion avoidance
technique. technique.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
The terms are presented in many cases using lowercase for
readability.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This document is not an Internet Standards Track specification; it is
provisions of BCP 78 and BCP 79. published for informational purposes.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are candidates for any level of Internet
Standard; see Section 2 of RFC 7841.
This Internet-Draft will expire on June 15, 2015. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8836.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction
2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3 1.1. Requirements Language
3. Deficiencies of existing mechanisms . . . . . . . . . . . . . 8 2. Requirements
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 3. Deficiencies of Existing Mechanisms
5. Security Considerations . . . . . . . . . . . . . . . . . . . 9 4. IANA Considerations
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 10 5. Security Considerations
7. References . . . . . . . . . . . . . . . . . . . . . . . . . 10 6. References
7.1. Normative References . . . . . . . . . . . . . . . . . . 10 6.1. Normative References
7.2. Informative References . . . . . . . . . . . . . . . . . 10 6.2. Informative References
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11 Acknowledgements
Authors' Addresses
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 a 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 handling 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 delay than hundreds of milliseconds end-
delay), the sources of data may be able to adapt the amount of data to-end). In addition, the sources of data may be able to adapt the
that needs sending within fairly wide margins but can be rate limited amount of data that needs sending within fairly wide margins, but
by the application- even not always have data to send, and may they can be rate limited by the application -- even not always having
tolerate some amount of packet loss, but since the data is generated data to send. They may tolerate some amount of packet loss, but
in real-time, sending "future" data is impossible, and since it's since the data is generated in real time, sending "future" data is
consumed in real-time, data delivered late is commonly useless. impossible, and since it's consumed in real time, data delivered late
is commonly useless.
While the requirements for real-time interactive media differ from While the requirements for real-time interactive media differ from
the requirements for the other flow types, these other flow types the requirements for the other flow types, these other flow types
will be present in the network. The congestion control algorithm for will be present in the network. The congestion control algorithm for
real-time interactive media must work properly when these other flow real-time interactive media must work properly when these other flow
types are present as cross traffic on the network. types are present as cross traffic on the network.
One particular protocol portfolio 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 [RFC8825], where one envisions sending multiple flows using
multiple flows using the Real-time Transport Protocol (RTP) [RFC3550] the Real-time Transport Protocol (RTP) [RFC3550] between two peers,
between two peers, in conjunction with data flows, all at the same in conjunction with data flows, all at the same time, without having
time, without having special arrangements with the intervening special arrangements with the intervening service providers. As RTP
service providers. As RTP does not provide any congestion control does not provide any congestion control mechanism, a set of circuit
mechanism; a set of circuit breakers, such as breakers, such as those described in [RFC8083], are required to
[I-D.ietf-avtcore-rtp-circuit-breakers], are required to protect the protect the network from excessive congestion caused by non-
network from excessive congestion caused by the non-congestion congestion-controlled flows. When the real-time interactive media is
controlled flows. When the real-time interactive media is congestion congestion controlled, it is recommended that the congestion control
controlled, it is recommended that the congestion control mechanism mechanism operate within the constraints defined by these circuit
operates within the constraints defined by these circuit breakers breakers when a circuit breaker is present and that it should not
when circuit breaker is present and that it should not cause cause congestion collapse when a circuit breaker is not implemented.
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 those
cases using multicast/broadcast-type technologies, are out of scope. 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 [RFC8825] is used in this memo.
memo.
1.1. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14 [RFC2119].
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 to allow some useful
to get enough bandwidth. (Example: not enough bandwidth for flows to get enough bandwidth. For example, although there
low-latency video+audio, but enough for audio-only.) might not be enough bandwidth for low-latency video+audio, there
A. Jitter (variation in the bitrate over short time scales) could be enough for audio only.
also is relevant, though moderate amounts of jitter will be
a. Jitter (variation in the bitrate over short timescales) is
also 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 and b. The algorithm should provide this as-low-as-possible-delay
minimize self-induced latency even when faced with transit and minimize self-induced latency even when faced
intermediate bottlenecks and competing flows. Competing with intermediate bottlenecks and competing flows.
flows may limit what's possible to achieve. Competing flows may limit what's possible to achieve.
C. It should be resilience to the effects of the events, such
as routing changes, which may alter or remove bottlenecks or
change the bandwidth available especially if there is a
reduction in available bandwidth or increase in observed
delay. It is expected that the mechanism reacts quickly to
the such 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 react quickly to handle both local and remote c. The algorithm should be resilient to the effects of events,
interface changes (WLAN to 3G data, etc) which may radically such as routing changes, which may alter or remove
change the bandwidth available or bottlenecks, especially if bottlenecks or change the bandwidth available, especially if
there is a reduction in available bandwidth or increase in there is a reduction in available bandwidth or increase in
bottleneck delay. It is assumed that an interface change observed delay. It is expected that the mechanism reacts
can generate a notification to the algorithm. quickly to such 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 at the same time a
response cannot be arbitrarily delayed.
E. The real-time interactive media applications can be rate d. The algorithm should react quickly to handle both local and
remote interface changes (e.g., WLAN to 3G data) that may
radically change the bandwidth available or bottlenecks,
especially if there is a reduction in available bandwidth or
an increase in bottleneck delay. It is assumed that an
interface change can generate a notification to the
algorithm.
e. The real-time interactive media applications can be rate
limited. This means the offered loads can be less than the limited. This means the offered loads can be less than the
available bandwidth at any given moment, and may vary available bandwidth at any given moment and may vary
dramatically over time, including dropping to no load and dramatically over time, including dropping to no load and
then resuming a high load, such as in a mute/unmute then resuming a high load, such as in a mute/unmute
operation. Hence, the algorithm must be designed to handle operation. Hence, the algorithm must be designed to handle
such behavior from media source or application. Note that such behavior from a media source or application. Note that
the reaction time between a change in the bandwidth the reaction time between a change in the bandwidth
available from the algorithm and a change in the offered available from the algorithm and a change in the offered
load is variable, and may be different when increasing load is variable, and it may be different when increasing
versus decreasing. versus decreasing.
F. The algorithm requires to avoid building up queues when f. The algorithm is required 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 clear
quickly. The algorithm should also react quickly to regain quickly. The algorithm should also react quickly to regain
its previous share of the bandwidth when the local- its previous share of the bandwidth when the local
bottleneck or 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
adaptive within a burst. They are often layered on top of within a burst. They are often layered on top of TCP but
TCP but use TCP in a bursty manner that can interact poorly use TCP in a bursty manner that can interact poorly with
with competing flows during the bursts. The algorithm must competing flows during the bursts. The algorithm must not
not increase the already existing delay buildup during those increase the already existing delay buildup during those
bursts. Note that this competing traffic may be on a shared bursts. Note that this competing traffic may be on a shared
access link, or the traffic burst may cause a shift in the access link, or the traffic burst may cause a shift in the
location of the bottleneck for 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 flows and bursts such as the traffic generated by a
browsing session. Note that 'fair' is a rather hard-to-define typical web-browsing session. Note that "fair" is a rather
term. It should be fair with itself, giving fair share of the hard-to-define term. It SHOULD be fair with itself, giving a
bandwidth to multiple flows with similar RTTs, and if possible fair share of the bandwidth to multiple flows with similar RTTs,
to multiple flows with different 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
up to a useful share of the bottleneck bandwidth as quickly to a useful share of the bottleneck bandwidth as quickly as
as possible. A useful share will depend on the media types possible. A useful share will depend on the media types
involved, total bandwidth available and the user experience involved, total bandwidth available, and the user-experience
requirements of a particular service. Note that relative 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 at which 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 prioritize 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.
4. The algorithm should as quickly as possible adapt to initial 4. The algorithm SHOULD adapt as quickly as possible to initial
network conditions at the start of a flow. This should occur network conditions at the start of a flow. This SHOULD occur
both if the initial bandwidth is above or below the bottleneck whether the initial bandwidth is above or below the bottleneck
bandwidth. bandwidth.
A. The algorithm should allow different modes of adaptation for a. The algorithm should allow different modes of adaptation;
example,the startup adaptation may be faster than adaptation for example, the startup adaptation may be faster than
later in a flow. It should allow for both slow-start adaptation later in a flow. It should allow for both slow-
operation (adapt up) and history-based startup (start at a start operation (adapt up) and history-based startup (start
point expected to be at or below channel bandwidth from at a point expected to be at or below channel bandwidth from
historical information, which may need to adapt down quickly historical information, which may need to adapt down quickly
if the initial guess is wrong). Starting too low and/or if the initial guess is wrong). Starting too low and/or
adapting up too slowly can cause a critical point in a adapting up too slowly can cause a critical point in a
personal communication to be poor ("Hello!"). Starting personal communication to be poor ("Hello!"). Starting too
over-bandwidth causes other problems for user experience, so high above the available bandwidth causes other problems for
there's a tension here. Alternative methods to help startup user experience, so there's a tension here. Alternative
like probing during setup with dummy data may be useful in methods to help startup, such as probing during setup with
some applications; in some cases there will be a dummy data, may be useful in some applications; in some
considerable gap in time between flow creation and the cases, there will be a considerable gap in time between flow
initial flow of data. Again, A flow may need to change creation and the initial flow of data. Again, a flow may
adaptation rates due to network conditions or changes in the need to change adaptation rates due to network conditions or
provided flows (such as un-muting or sending data after a changes in the provided flows (such as unmuting or sending
gap). data after a gap).
5. The algorithm should be stable if the RTP streams are halted or 5. The algorithm SHOULD be stable if the RTP streams are halted or
discontinuous (for example - Voice Activity Detection). discontinuous (for example, when using Voice Activity
Detection).
A. After stream resumption, the algorithm should attempt to a. After stream resumption, the algorithm should attempt to
rapidly regain its previous share of the bandwidth; the rapidly regain its previous share of the bandwidth; the
aggressiveness with which this is done will decay with the aggressiveness with which this is done will decay with the
length of the pause. length of the pause.
6. The algorithm should where possible merge information across 6. Where possible, the algorithm SHOULD merge information across
multiple RTP streams sent between two endpoints, when those RTP multiple RTP streams sent between two endpoints when those RTP
streams share a common bottleneck, whether or not those streams streams share a common bottleneck, whether or not those streams
are multiplexed onto the same ports, in order to allow are multiplexed onto the same ports. This will allow congestion
congestion control of the set of streams together instead of as control of the set of streams together instead of as multiple
multiple independent streams. This allows better overall independent streams. It will also allow 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.
A. The algorithm 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], when a WebRTC data channel [RFC8831], when possible.
possible.
B. 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, the coordinating their bandwidth use and congestion control, the
algorithm should allow the application to control the algorithm should allow the application to control the
relative split of available bandwidth. The most correlated relative split of available bandwidth. The most correlated
bandwidth usage would be with other flows on the same bandwidth usage would be with other flows on the same
5-tuple, but there may be use in coordinating measurement 5-tuple, but there may be use in coordinating measurement
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 regarding 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 be able to convey congestion-related
functional. As much as possible, it should leverage available information. 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, acknowledgements and feedback, packet timestamps, and times, acknowledgements and feedback, packet timestamps, packet
packet losses, Explicit Congestion Notification (ECN) [RFC3168]; losses, and 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 such information 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
backchannel typically should be done via RTCP[RFC3550]; one backchannel typically SHOULD be done via the RTP Control
alternative would be to include it instead in a reverse RTP Protocol (RTCP) [RFC3550]; instead, one alternative would be to
channel using header extensions. include it in a reverse-RTP 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. Note that in some cases, backchannel feedback must be used. Note that in some cases, backchannel
messages may be delayed until the RTCP channel can be messages may be delayed until the RTCP channel can be
allocated enough bandwidth, even under AVPF rules. This may allocated enough bandwidth, even under AVPF rules. This may
also imply negotiating a higher maximum percentage for RTCP also imply negotiating a higher maximum percentage for RTCP
data or allowing solutions to violate or modify the rules data or allowing solutions to violate or modify the rules
specified for AVPF. specified for AVPF.
B. 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 using techniques such as those in [RFC5506], to allow RTCP
Reports/Receiver Reports) without Sender/Receiver Reports.
C. Backchannel data should be minimized to avoid taking too c. 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.
9. Flows managed by this algorithm and flows competing against at a 9. Flows managed by this algorithm and flows competing against each
bottleneck may have different DSCP[RFC5865] markings depending other at a bottleneck may have different Differentiated Services
on the type of traffic, or may be subject to flow-based QoS. A Code Point (DSCP) [RFC5865] markings depending on the type of
particular bottleneck or section of the network path may or may traffic or may be subject to flow-based QoS. A particular
not honor DSCP markings. The algorithm should attempt to bottleneck or section of the network path may or may not honor
leverage DSCP markings when they're available. DSCP markings. The algorithm SHOULD attempt to leverage DSCP
markings when they're available.
A. In WebRTC, a division of packets into 4 classes is
envisioned in order of priority: faster-than-audio, audio,
video, best-effort, and bulk-transfer. Typically the flows
managed by this algorithm would be audio or video in that
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 maintain low-delay when faced 11. The algorithm SHOULD be stable and maintain low delay when faced
with Active Queue Management (AQM) algorithms. Also note that with Active Queue Management (AQM) algorithms. Also note that
these algorithms may apply across multiple queues in the these algorithms may apply across multiple queues in the
bottleneck, 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 that 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 that provides a reasonably fair share of bandwidth
bandwidth when competing with TCP flows and offers much lower when competing with TCP flows and offers much lower throughput
throughput variations than TCP. This is achieved by a slower variations than TCP. This is achieved by a slower response to the
response to the available bandwidth change than TCP. TFRC is available bandwidth change than TCP. TFRC is designed to perform
designed to perform best with applications which has fixed packet best with applications that have a fixed packet size and do not have
size and does not have fixed period between sending packets. a fixed period between sending packets.
TFRC operates on detecting loss events and reacts to loss caused by TFRC detects loss events and reacts to congestion-caused loss by
congestion by reducing its sending rate. It allows applications to reducing its sending rate. It allows applications to increase the
increase the sending rate until loss is observed in the flows. As it sending rate until loss is observed in the flows. As noted in IAB/
is noted in IAB/IRTF report [RFC7295] large buffers are available in IRTF report [RFC7295], large buffers are available in the network
the network elements which introduces additional delay in the elements, which introduce additional delay in the communication. It
communication, it becomes important to take all possible congestion becomes important to take all possible congestion indications into
indications into considerations. Looking at the current Internet consideration. Looking at the current Internet deployment, TFRC's
deployment, TFRC's only consideration of loss events as congestion biggest deficiency is that it only considers loss events as a
indication can be considered as biggest lacking. congestion indication.
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 a fixed interval between packets and needs to
their segment size instead of their packet rate in response to vary the segment size of the packets instead of the packet rate in
congestion and sends smaller packets, a variant of TFRC , Small- response to congestion; therefore, it sends smaller packets. A
Packet TFRC (TFRC-SP) [RFC4828] addresses the issues related to such variant of TFRC, Small-Packet TFRC (TFRC-SP) [RFC4828], addresses the
kind of sources ; a video source generally varies video frame sizes, issues related to such kind of sources. A video source generally
can produce large frames which need to be further fragmented to fit varies video frame sizes, can produce large frames that need to be
into path Maximum Transmission Unit (MTU) size, and have almost fixed further fragmented to fit into path Maximum Transmission Unit (MTU)
interval between producing frames under a certain frame rate, TFRC is size, and has an almost fixed interval between producing frames under
known to be less optimal when using with such video sources. a certain frame rate. TFRC is known to be less optimal when using
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 work. TFRC is designed to maintain a smooth sending rate; however,
sources can change rates in steps for both rate increase and rate media sources can change rates in steps for both rate increase and
decrease. TFRC can operate in two modes - i) Bytes per second and rate decrease. TFRC can operate in two modes: i) bytes per second
ii) packets per second, where typical real-time interactive media and ii) packets per second, where typical real-time interactive media
sources operates on bit per second. There are also limitations on sources operate on bit per second. There are also limitations on how
how quickly the media sources can adapt to specific sending rates. quickly the media sources can adapt to specific sending rates.
The modern video encoders can operate on a mode where they can vary Modern video encoders can operate in a mode in which they can vary
the output bitrate a lot depending on the way there are configured, the output bitrate a lot depending on the way they are configured,
the current scene it is encoding and more. Therefore, it is possible the current scene they are encoding, and more. Therefore, it is
that the video source does not always output at a bitrate they are possible that the video source will not always output at an allowable
allowed to. TFRC tries to raise its sending rate when transmitting bitrate. TFRC tries to increase its sending rate when transmitting
at maximum allowed rate and increases only twice the current at the maximum allowed rate, and it increases only twice the current
transmission rate hence it may create issues when the video source transmission rate; hence, it may create issues when the video sources
vary their bitrates. vary their bitrates.
Moreover, there are number of studies on TFRC which shows it's Moreover, there are a number of studies on TFRC that show its
limitations which includes TFRC's unfairness on low statistically limitations, including TFRC's unfairness to low statistically
multiplexed links, oscillatory behavior, performance issue in highly multiplexed links, oscillatory behavior, performance issues in highly
dynamic loss rates conditions and more [CH09]. dynamic loss-rate conditions, and more [CH09].
Looking at all these deficiencies it can be concluded that the Looking at all these deficiencies, it can be concluded that the
requirements of congestion control mechanism for real-time requirements for a congestion control mechanism for real-time
interactive media cannot be met by TFRC as defined in the standard. interactive media cannot be met by TFRC as defined in the standard.
4. IANA Considerations 4. IANA Considerations
This document makes no request of IANA. This document has no IANA actions.
Note to RFC Editor: this section may be removed on publication as an
RFC.
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
the flow can fake congestion signals, unless they are passed on a into the flow can fake congestion signals, unless they are passed on
tamper-proof path. Since some possible algorithms depend on the a 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 perceived available bandwidth all packets. Attacks that increase the perceived available bandwidth
are conceivable, and need to be evaluated. Such attacks could result are conceivable and need to be evaluated. Such attacks could result
in starvation of competing flows and permit amplification attacks. in starvation of competing flows and permit amplification attacks.
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. References
This document is the result of discussions in various fora of the
WebRTC effort, in particular on the rtp-congestion@alvestrand.no
mailing list. Many people contributed their thoughts to this.
7. References
7.1. Normative References
[I-D.ietf-rtcweb-overview] 6.1. Normative References
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-13
(work in progress), November 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,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[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,
2006. DOI 10.17487/RFC4585, July 2006,
<https://www.rfc-editor.org/info/rfc4585>.
[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, DOI 10.17487/RFC5124, February
2008, <https://www.rfc-editor.org/info/rfc5124>.
7.2. Informative References [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
6.2. 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", Proceedings of PFLDNeT, 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-08 (work in progress),
December 2014.
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-12 (work in
progress), September 2014.
[MPEG_DASH] [MPEG_DASH]
"Dynamic adaptive streaming over HTTP (DASH) -- Part 1: ISO, "Information Technology -- Dynamic adaptive streaming
Media presentation description and segment formats", April over HTTP (DASH) -- Part 1: Media presentation description
2012. and segment formats", ISO/IEC 23009-1:2019, December 2019,
<https://www.iso.org/standard/79329.html>.
[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",
3168, September 2001. RFC 3168, DOI 10.17487/RFC3168, September 2001,
<https://www.rfc-editor.org/info/rfc3168>.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828, April (TFRC): The Small-Packet (SP) Variant", RFC 4828,
2007. DOI 10.17487/RFC4828, April 2007,
<https://www.rfc-editor.org/info/rfc4828>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC Friendly Rate Control (TFRC): Protocol Specification",
5348, September 2008. RFC 5348, DOI 10.17487/RFC5348, September 2008,
<https://www.rfc-editor.org/info/rfc5348>.
[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, DOI 10.17487/RFC5506, April
2009, <https://www.rfc-editor.org/info/rfc5506>.
[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, DOI 10.17487/RFC5865, May 2010,
<https://www.rfc-editor.org/info/rfc5865>.
[RFC7295] Tschofenig, H., Eggert, L., and Z. Sarker, "Report from [RFC7295] Tschofenig, H., Eggert, L., and Z. Sarker, "Report from
the IAB/IRTF Workshop on Congestion Control for the IAB/IRTF Workshop on Congestion Control for
Interactive Real-Time Communication", RFC 7295, July 2014. Interactive Real-Time Communication", RFC 7295,
DOI 10.17487/RFC7295, July 2014,
<https://www.rfc-editor.org/info/rfc7295>.
[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017,
<https://www.rfc-editor.org/info/rfc8083>.
[RFC8831] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
<https://www.rfc-editor.org/info/rfc8831>.
Acknowledgements
This document is the result of discussions in various fora of the
WebRTC effort, in particular on the <rtp-congestion@alvestrand.no>
mailing list. Many people contributed their thoughts to this.
Authors' Addresses Authors' Addresses
Randell Jesup Randell Jesup
Mozilla Mozilla
USA United States of America
Email: randell-ietf@jesup.org Email: randell-ietf@jesup.org
Zaheduzzaman Sarker (editor) Zaheduzzaman Sarker (editor)
Ericsson Ericsson AB
Torshamnsgatan 23
SE-164 83 Stockholm
Sweden Sweden
Phone: +46 10 717 37 43
Email: zaheduzzaman.sarker@ericsson.com Email: zaheduzzaman.sarker@ericsson.com
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