Congestion Control Requirements for
Interactive Real-Time Media
Mozilla
United States of America
randell-ietf@jesup.org
Ericsson AB
Torshamnsgatan 23
Stockholm
164 83
Sweden
+46 10 717 37 43
zaheduzzaman.sarker@ericsson.com
Interactive multimedia
webrtc
video communication
RTP/RTCP
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.
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 , where one
envisions sending multiple flows using the Real-time Transport Protocol
(RTP) 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 ,
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
is used in this memo.
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
.
Requirements
-
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.
- 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.
- 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.
- 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.
- 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.
- 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.
- 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.
- Similarly, periodic bursty flows such as 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.
-
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.
- 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.
-
The algorithm SHOULD NOT starve competing TCP flows and SHOULD,
as best as possible, avoid starvation by TCP flows.
- 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.
-
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.
- 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).
-
The algorithm SHOULD be stable if the RTP streams are halted or
discontinuous (for example, when using Voice Activity Detection).
- 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.
-
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.
- The algorithm should also share information and adaptation
with other non-RTP flows between the same endpoints, such as a
WebRTC data channel , when
possible.
- 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.
-
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) ; all of these can
provide information about the state of the path and any bottlenecks.
However, the use of available information is algorithm
dependent.
- 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.
-
Since the assumption here is a set of RTP streams, the
backchannel typically SHOULD be done via the RTP Control Protocol
(RTCP) ; instead, one alternative
would be to include it
in a reverse-RTP channel using header extensions.
- In order to react sufficiently quickly when using RTCP for a
backchannel, an RTP profile such as RTP/AVPF or RTP/SAVPF 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.
- Bandwidth for the feedback messages should be minimized
using techniques such as those in , to allow RTCP
without Sender/Receiver Reports.
- 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.
-
Flows managed by this algorithm and flows competing against each
other at a
bottleneck may have different Differentiated Services Code Point
(DSCP)
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.
- 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.
- 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.
Deficiencies of Existing Mechanisms
Among the existing congestion control mechanisms, TCP Friendly Rate
Control (TFRC) 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 , 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) , 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 .
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.
IANA Considerations
This document has no IANA actions.
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.
References
Normative References
Overview: Real-Time Protocols for Browser-Based Applications
Informative References
WebRTC Data Channels
Information Technology -- Dynamic adaptive streaming over
HTTP (DASH) -- Part 1: Media presentation description and segment
formats
ISO
Designing TCP-Friendly Window-based Congestion Control for
Real-time Multimedia Applications
Proceedings of PFLDNeT
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.