Congestion Control Requirements for Interactive Real-Time Media
RFC 8836
| Document | Type | RFC - Informational (January 2021; No errata) | |
|---|---|---|---|
| Authors | Randell Jesup , Zaheduzzaman Sarker | ||
| Last updated | 2021-01-18 | ||
| Replaces | draft-jesup-rmcat-reqs | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html xml pdf htmlized (tools) htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Mirja Kühlewind | ||
| Shepherd write-up | Show (last changed 2014-11-13) | ||
| IESG | IESG state | RFC 8836 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Yes | ||
| Telechat date | |||
| Responsible AD | Mirja Kühlewind | ||
| Send notices to | (None) | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | No IANA Actions | ||
Internet Engineering Task Force (IETF) R. Jesup
Request for Comments: 8836 Mozilla
Category: Informational Z. Sarker, Ed.
ISSN: 2070-1721 Ericsson AB
January 2021
Congestion Control Requirements for Interactive Real-Time Media
Abstract
Congestion control is needed for all data transported across the
Internet, in order to promote fair usage and prevent congestion
collapse. The requirements for interactive, point-to-point real-time
multimedia, which needs low-delay, semi-reliable data delivery, are
different from the requirements for bulk transfer like FTP or bursty
transfers like web pages. Due to an increasing amount of RTP-based
real-time media traffic on the Internet (e.g., with the introduction
of the Web Real-Time Communication (WebRTC)), it is especially
important to ensure that this kind of traffic is congestion
controlled.
This document describes a set of requirements that can be used to
evaluate other congestion control mechanisms in order to figure out
their fitness for this purpose, and in particular to provide a set of
possible requirements for a real-time media congestion avoidance
technique.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
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.
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 (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
1.1. Requirements Language
2. Requirements
3. Deficiencies of Existing Mechanisms
4. IANA Considerations
5. Security Considerations
6. References
6.1. Normative References
6.2. Informative References
Acknowledgements
Authors' Addresses
1. Introduction
Most of today's TCP congestion control schemes were developed with a
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
been used successfully to govern the reliable transfer of smaller
chunks of data in as short a time as possible, such as when fetching
web pages.
These algorithms have also been used for transfer of media streams
that are viewed in a non-interactive manner, such as "streaming"
video, where having the data ready when the viewer wants it is
important, but the exact timing of the delivery is not.
When handling real-time interactive media, the requirements are
different. One needs to provide the data continuously, within a very
limited time window (no more delay than hundreds of milliseconds end-
to-end). In addition, the sources of data may be able to adapt the
amount of data that needs sending within fairly wide margins, but
they can be rate limited by the application -- even not always having
data to send. They may tolerate some amount of packet loss, but
since the data is generated in real time, sending "future" data is
impossible, and since it's consumed in real time, data delivered late
is commonly useless.
While the requirements for real-time interactive media differ from
the requirements for the other flow types, these other flow types
will be present in the network. The congestion control algorithm for
real-time interactive media must work properly when these other flow
types are present as cross traffic on the network.
One particular protocol portfolio being developed for this use case
is WebRTC [RFC8825], where one envisions sending multiple flows using
the Real-time Transport Protocol (RTP) [RFC3550] between two peers,
in conjunction with data flows, all at the same time, without having
special arrangements with the intervening service providers. As RTP
does not provide any congestion control mechanism, a set of circuit
breakers, such as those described in [RFC8083], are required to
protect the network from excessive congestion caused by non-
congestion-controlled flows. When the real-time interactive media is
congestion controlled, it is recommended that the congestion control
mechanism operate within the constraints defined by these circuit
breakers when a circuit breaker is present and that it should not
cause congestion collapse when a circuit breaker is not implemented.
Given that this use case is the focus of this document, use cases
involving non-interactive media such as video streaming and those
using multicast/broadcast-type technologies, are out of scope.
The terminology defined in [RFC8825] is used in this 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
1. The congestion control algorithm MUST attempt to provide as-low-
as-possible-delay transit for interactive real-time traffic
while still providing a useful amount of bandwidth. There may
be lower limits on the amount of bandwidth that is useful, but
this is largely application specific, and the application may be
able to modify or remove flows in order to allow some useful
flows to get enough bandwidth. For example, although there
might not be enough bandwidth for low-latency video+audio, there
could be enough for audio only.
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
considered to track the short-term maximums of delay,
including jitter.
b. The algorithm should provide this as-low-as-possible-delay
transit and minimize self-induced latency even when faced
with intermediate bottlenecks and competing flows.
Competing flows may limit what's possible to achieve.
c. The algorithm should be resilient to the effects of 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 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.
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
available bandwidth at any given moment and may vary
dramatically over time, including dropping to no load and
then resuming a high load, such as in a mute/unmute
operation. Hence, the algorithm must be designed to handle
such behavior from a media source or application. Note that
the reaction time between a change in the bandwidth
available from the algorithm and a change in the offered
load is variable, and it may be different when increasing
versus decreasing.
f. The algorithm is required to avoid building up queues when
competing with short-term bursts of traffic (for example,
traffic generated by web browsing), which can quickly
saturate a local-bottleneck router or link but clear
quickly. The algorithm should also react quickly to regain
its previous share of the bandwidth when the local
bottleneck or link is cleared.
g. Similarly, periodic bursty flows such as MPEG DASH
[MPEG_DASH] or proprietary media streaming algorithms may
compete in bursts with the algorithm and may not be adaptive
within a burst. They are often layered on top of TCP but
use TCP in a bursty manner that can interact poorly with
competing flows during the bursts. The algorithm must not
increase the already existing delay buildup during those
bursts. Note that this competing traffic may be on a shared
access link, or the traffic burst may cause a shift in the
location of the bottleneck for the duration of the burst.
2. The algorithm MUST be fair to other flows, both real-time flows
(such as other instances of itself) and TCP flows, both long-
lived flows and bursts such as the traffic generated by a
typical web-browsing session. Note that "fair" is a rather
hard-to-define term. It SHOULD be fair with itself, giving a
fair share of the bandwidth to multiple flows with similar RTTs,
and if possible to multiple flows with different RTTs.
a. Existing flows at a bottleneck must also be fair to new
flows to that bottleneck and must allow new flows to ramp up
to a useful share of the bottleneck bandwidth as quickly as
possible. A useful share will depend on the media types
involved, total bandwidth available, and the user-experience
requirements of a particular service. Note that relative
RTTs may affect the rate at which new flows can ramp up to a
reasonable share.
3. The algorithm SHOULD NOT starve competing TCP flows and SHOULD,
as best as possible, avoid starvation by TCP flows.
a. The congestion control should prioritize achieving a useful
share of the bandwidth depending on the media types and
total available bandwidth over achieving as-low-as-possible
transit delay, when these two requirements are in conflict.
4. The algorithm SHOULD adapt as quickly as possible to initial
network conditions at the start of a flow. This SHOULD occur
whether the initial bandwidth is above or below the bottleneck
bandwidth.
a. The algorithm should allow different modes of adaptation;
for example, 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 too
high above the available bandwidth causes other problems for
user experience, so there's a tension here. Alternative
methods to help startup, such as 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 unmuting or sending
data after a gap).
5. The algorithm SHOULD be stable if the RTP streams are halted or
discontinuous (for example, when using Voice Activity
Detection).
a. After stream resumption, the algorithm should attempt to
rapidly regain its previous share of the bandwidth; the
aggressiveness with which this is done will decay with the
length of the pause.
6. Where possible, the algorithm SHOULD merge information across
multiple RTP streams sent between two endpoints when those RTP
streams share a common bottleneck, whether or not those streams
are multiplexed onto the same ports. This will allow congestion
control of the set of streams together instead of as multiple
independent streams. It will also allow better overall
bandwidth management, faster response to changing conditions,
and fairer sharing of bandwidth with other network users.
a. The algorithm should also share information and adaptation
with other non-RTP flows between the same endpoints, such as
a WebRTC data channel [RFC8831], when possible.
b. When there are multiple streams across the same 5-tuple
coordinating their bandwidth use and congestion control, the
algorithm should allow the application to control the
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 regarding startup
performance of a new flow.
7. The algorithm SHOULD NOT require any special support from
network elements to be able to convey congestion-related
information. As much as possible, it SHOULD leverage available
information about the incoming flow to provide feedback to the
sender. Examples of this information are the packet arrival
times, acknowledgements and feedback, packet timestamps, packet
losses, and Explicit Congestion Notification (ECN) [RFC3168];
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
more detailed information on actual send times (as opposed
to sampling times), but such information should not be
required.
8. Since the assumption here is a set of RTP streams, the
backchannel typically SHOULD be done via the RTP Control
Protocol (RTCP) [RFC3550]; instead, one alternative would be to
include it in a reverse-RTP channel using header extensions.
a. In order to react sufficiently quickly when using RTCP for a
backchannel, an RTP profile such as RTP/AVPF [RFC4585] or
RTP/SAVPF [RFC5124] that allows sufficiently frequent
feedback must be used. Note that in some cases, backchannel
messages may be delayed until the RTCP channel can be
allocated enough bandwidth, even under AVPF rules. This may
also imply negotiating a higher maximum percentage for RTCP
data or allowing solutions to violate or modify the rules
specified for AVPF.
b. Bandwidth for the feedback messages should be minimized
using techniques such as those in [RFC5506], to allow RTCP
without Sender/Receiver Reports.
c. Backchannel data should be minimized to avoid taking too
much reverse-channel bandwidth (since this will often be
used in a bidirectional set of flows). In areas of
stability, backchannel data may be sent more infrequently so
long as algorithm stability and fairness are maintained.
When the channel is unstable or has not yet reached
equilibrium after a change, backchannel feedback may be more
frequent and use more reverse-channel bandwidth. This is an
area with considerable flexibility of design, and different
approaches to backchannel messages and frequency are
expected to be evaluated.
9. Flows managed by this algorithm and flows competing against each
other at a bottleneck may have different Differentiated Services
Code Point (DSCP) [RFC5865] markings depending 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 not honor
DSCP markings. The algorithm SHOULD attempt to leverage DSCP
markings when they're available.
10. The algorithm SHOULD sense the unexpected lack of backchannel
information as a possible indication of a channel-overuse
problem and react accordingly to avoid burst events causing a
congestion collapse.
11. The algorithm SHOULD be stable and maintain low delay when faced
with Active Queue Management (AQM) algorithms. Also note that
these algorithms may apply across multiple queues in the
bottleneck or to a single queue.
3. Deficiencies of Existing Mechanisms
Among the existing congestion control mechanisms, TCP Friendly Rate
Control (TFRC) [RFC5348] is the one that claims to be suitable for
real-time interactive media. TFRC is an equation-based congestion
control mechanism that provides a reasonably fair share of 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 that have a fixed packet size and do not have
a fixed period between sending packets.
TFRC detects loss events and reacts to congestion-caused loss by
reducing its sending rate. It allows applications to increase the
sending rate until loss is observed in the flows. As noted in IAB/
IRTF report [RFC7295], large buffers are available in the network
elements, which introduce additional delay in the communication. It
becomes important to take all possible congestion indications into
consideration. Looking at the current Internet deployment, TFRC's
biggest deficiency is that it only considers loss events as a
congestion indication.
A typical real-time interactive communication includes live-encoded
audio and video flow(s). In such a communication scenario, an audio
source typically needs a fixed interval between packets and needs to
vary the segment size of the packets instead of the packet rate in
response to congestion; therefore, it sends smaller packets. A
variant 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 that need to be
further fragmented to fit into path Maximum Transmission Unit (MTU)
size, and has an almost fixed interval between producing frames under
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
how the media sources in a typical real-time interactive application
work. TFRC is designed to maintain a 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 operate on bit per second. There are also limitations on how
quickly the media sources can adapt to specific sending rates.
Modern video encoders can operate in a mode in which they can vary
the output bitrate a lot depending on the way they are configured,
the current scene they are encoding, and more. Therefore, it is
possible that the video source will not always output at an allowable
bitrate. TFRC tries to increase its sending rate when transmitting
at the maximum allowed rate, and it increases only twice the current
transmission rate; hence, it may create issues when the video sources
vary their bitrates.
Moreover, there are a number of studies on TFRC that show its
limitations, including TFRC's unfairness to low statistically
multiplexed links, oscillatory behavior, performance issues in highly
dynamic loss-rate conditions, and more [CH09].
Looking at all these deficiencies, it can be concluded that the
requirements for a congestion control mechanism for real-time
interactive media cannot be met by TFRC as defined in the standard.
4. IANA Considerations
This document has no IANA actions.
5. Security Considerations
An attacker with the ability to delete, delay, or insert messages
into the flow can fake congestion signals, unless they are passed on
a tamper-proof path. Since some possible algorithms depend on the
timing of packet arrival, even a traditional, protected channel does
not fully mitigate such attacks.
An attack that reduces bandwidth is not necessarily significant,
since an on-path attacker could break the connection by discarding
all packets. Attacks that increase the perceived available bandwidth
are conceivable and need to be evaluated. Such attacks could result
in starvation of competing flows and permit amplification attacks.
Algorithm designers should consider the possibility of malicious on-
path attackers.
6. References
6.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
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.
Jacobson, "RTP: A Transport Protocol for Real-Time
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,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<https://www.rfc-editor.org/info/rfc4585>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <https://www.rfc-editor.org/info/rfc5124>.
[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-
based Congestion Control for Real-time Multimedia
Applications", Proceedings of PFLDNeT, May 2009.
[MPEG_DASH]
ISO, "Information Technology -- Dynamic adaptive streaming
over HTTP (DASH) -- Part 1: Media presentation description
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
of Explicit Congestion Notification (ECN) to IP",
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
(TFRC): The Small-Packet (SP) Variant", RFC 4828,
DOI 10.17487/RFC4828, April 2007,
<https://www.rfc-editor.org/info/rfc4828>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
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
Real-Time Transport Control Protocol (RTCP): Opportunities
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
Services Code Point (DSCP) for Capacity-Admitted Traffic",
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
the IAB/IRTF Workshop on Congestion Control for
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
Randell Jesup
Mozilla
United States of America
Email: randell-ietf@jesup.org
Zaheduzzaman Sarker (editor)
Ericsson AB
Torshamnsgatan 23
SE-164 83 Stockholm
Sweden
Phone: +46 10 717 37 43
Email: zaheduzzaman.sarker@ericsson.com