Guidelines for Internet Congestion Control at Endpoints
draft-fairhurst-ccwg-cc-00
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draft-fairhurst-ccwg-cc-00
Network Working Group G. Fairhurst
Internet-Draft University of Aberdeen
Intended status: Best Current Practice 3 July 2023
Expires: 4 January 2024
Guidelines for Internet Congestion Control at Endpoints
draft-fairhurst-ccwg-cc-00
Abstract
When published as an RFC, this document provides guidance on the
design of methods to avoid congestion collapse and how an endpoint
needs to react to congestion. The IETF provides recommendations and
requirements on this topic that is distributed across many documents
in the RFC series. This document therefore gathers and consolidates
these recommendations. Based on these, and Internet engineering
experience, the document provides best current practice for the
design of new congestion control methods in Internet protocols.
When published, the document will update or replace the Best Current
Practice in BCP 41, which currently includes "Congestion Control
Principles" provided in RFC2914.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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 https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 4 January 2024.
Copyright Notice
Copyright (c) 2023 IETF Trust and the persons identified as the
document authors. All rights reserved.
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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 publication of this document.
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document. Code Components
extracted from this document must include Revised BSD License text as
described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Revised BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Incipient and Persistent Congestion . . . . . . . . . . . 4
1.2. The Need to Mitigate the Effects of Incipient
Congestion . . . . . . . . . . . . . . . . . . . . . . . 4
1.3. The Need to Avoid the Effects of Persistent Congestion . 5
1.4. Current IETF Guidelines on Evaluation of Congestion
Control . . . . . . . . . . . . . . . . . . . . . . . . . 6
1.5. Current Challenges . . . . . . . . . . . . . . . . . . . 7
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 8
3. Requirements from the RFC Series . . . . . . . . . . . . . . 9
3.1. The need to React to Congestion . . . . . . . . . . . . . 9
3.2. Tolerance to a Diversity of Path Characteristics . . . . 9
3.3. Protection of Protocol Mechanisms . . . . . . . . . . . . 10
4. Principles of Congestion Control . . . . . . . . . . . . . . 11
4.1. Robustness: Timers and Retransmission . . . . . . . . . . 11
4.2. Detecting and Reacting to Incipient Congestion . . . . . 13
4.2.1. Congestion Initialization . . . . . . . . . . . . . . 13
4.2.2. Using Path Capacity . . . . . . . . . . . . . . . . . 13
4.2.3. Congestion Detection and Retransmission . . . . . . . 14
4.2.4. Responding to Incipient Congestion . . . . . . . . . 15
4.2.5. Using More Capacity . . . . . . . . . . . . . . . . . 16
4.2.6. Utilising Additional Path Information . . . . . . . . 17
4.3. Avoiding Persistent Congestion . . . . . . . . . . . . . 18
4.3.1. Avoiding Congestion Collapse and Flow Starvation . . 18
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 19
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
7. Security Considerations . . . . . . . . . . . . . . . . . . . 19
8. Normative References . . . . . . . . . . . . . . . . . . . . 19
9. Informative References . . . . . . . . . . . . . . . . . . . 20
Appendix A. Internet Congestion Control . . . . . . . . . . . . 25
A.1. Flow Multiplexing and Congestion . . . . . . . . . . . . 25
A.2. Adjusting the Rate . . . . . . . . . . . . . . . . . . . 26
Appendix B. Revision Notes . . . . . . . . . . . . . . . . . . . 27
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 28
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1. Introduction
The IETF has specified Internet transports (e.g., TCP [RFC9293], UDP
[RFC0768], UDP-Lite [RFC3828], SCTP [RFC4960], and DCCP [RFC4340]) as
well as protocols layered on top of these transports (e.g., RTP
[RFC3550], QUIC [RFC9000] [RFC9002], SCTP/UDP [RFC6951], DCCP/UDP
[RFC6773]) and transports that work directly over the IP network
layer. These transports are implemented in endpoints (either
Internet hosts or routers acting as endpoints), and are designed to
detect and react to network congestion. TCP was the first transport
to provide this, although the TCP specifications found in RFC 793
predates the inclusion of congestion control and did not contain any
discussion of using or managing a congestion window. RFC 9293
[RFC9293] seek to address this.
Internet transports need to react to avoid congestion that impacts
other flows sharing a path. The Requirements for Internet Hosts
[RFC1122] formally mandates that endpoints perform congestion
control. "Because congestion control is critical to the stable
operation of the Internet, applications and other protocols that
choose to use UDP as an Internet transport must employ mechanisms to
prevent congestion collapse and to establish some degree of fairness
with concurrent flows [RFC2914].
The popularity of the Internet has led to a proliferation in the
number of TCP implementations [RFC2914]. A variety of non-TCP
transports have also being deployed, including QUIC [RFC9000]. Some
transport implementations fail to use standardised congestion
avoidance mechanisms correctly, e.g., because of poor implementation
[RFC2525]. However, this is not the only reason for not using
standard methods. Some transports have chosen mechanisms that are
not presently standardised, or have adopted approaches to their
design that differ from present standards. Guidance is needed
therefore not only for future standardisation, but to ensure safe and
appropriate evolution of transports that have not presently been
submitted for standardisation.
Experience has shown that successful protocols developed in a
specific context or for a particular application tend to also become
used in a wider range of contexts. Therefore, IETF specifications by
default target deployment on the general Internet, or need to be
defined for use only within a controlled environment, and restricted
to specific controlled environments.
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1.1. Incipient and Persistent Congestion
Paths through the Internet can experience congestion (loss or delay)
that is a result of excess load at a bottleneck(s) along the path.
This document differentiates two levels of congestion:
* Incipient congestion: This is a consequential side effect of the
statistical multiplexing of packet flows. There will be time
where packets need to be buffered or dropped at the bottleneck(s)
on the path, and flows need to react when they encounter this
congestion to reduce their contribution to the load.
* Persistent congestion: This occurs when the pattern of arriving
traffic results in over consumption of the path resources.
Typically this results in packet loss. The effects of persistent
congestion might impact the flow that induces the congestion, but
could adversely impact other flows, e.g., starving them of
resources; or reducing the efficiency of the path (e.g.,
congestion collapse).
1.2. The Need to Mitigate the Effects of Incipient Congestion
Incipient congestion results in normal operation of the Internet.
Buffering (an increase in latency) or congestion loss (discard of a
packet) arises when the traffic arriving at a link or network exceeds
the resources available. Loss can also occur for other reasons, but
it is usually not possible for an endpoint to reliably disambiguate
the cause of packet loss (e.g., loss could be due to link corruption,
receiver overrun, etc. [RFC3819]). A network device typically uses
a drop-tail policy to drop excess IP packets when its queue(s)
becomes full. The filling of a buffer can be managed using Active
Queue Management (AQM) [RFC7567]. This can be combined with Explicit
Congestion Notification (ECN) signalling [RFC3168] to mitigate
incipient congestion [RFC8087].
Buffers can be divided into pools and traffic can be associated with
specific buffer pools (e.g., using a local configuration, an
automated method, or coordinated using the Differentiated Services
[RFC2475] architecture.) Network devices can configure a schedular
[RFC7806] to isolate the queuing of packets for different flows, or
aggregates of flows, and thereby assist in reducing the impact of
flow multiplexing on other flows (e.g., flow scheduling [RFC7567]).
This could include methods seeking to equally distribute resources
between sharing flows, but this is explicitly not a requirement for a
network device [Flow-Rate-Fairness]. Endpoints can not rely on the
presence and correct configuration of these methods, and therefore
even when a path is expected to support such methods, also need to
employ methods that work end-to-end.
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In some controlled environments, Internet transports can use
mechanisms to reserve capacity at routers or on the links/paths being
used in a controlled environment. Most Internet paths do not support
this, and endpoints are typically unable to rely upon prior
reservation of capacity along the path(s) they use. In the absence
of such a reservation, endpoints are unable to determine a safe rate
at which to start or continue their transmission. The use of an
Internet path therefore requires a combination of end-to-end
transport mechanisms to detect and then respond to changes in the
capacity that it discovers is available across the network path.
Section 3.3 of [RFC2914] notes that a flow that use congestion
control to "optimize its own performance regarding throughput, delay,
and loss. In some circumstances, for example in environments with
high statistical multiplexing, the delay and loss rate experienced by
a flow are largely independent of its own sending rate. However, in
environments with lower levels of statistical multiplexing or with
per-flow scheduling, the delay and loss rate experienced by a flow is
in part a function of the flow's own sending rate. Thus, a flow can
use end-to-end congestion control to limit the delay or loss
experienced by its own packets. We would note, however, that in an
environment like the current best-effort Internet, concerns regarding
congestion collapse and fairness with competing flows limit the range
of congestion control behaviors available to a flow."
1.3. The Need to Avoid the Effects of Persistent Congestion
Early RFCs recognised that a significant pathology can arise when a
poorly designed transport creates significant congestion. This can
result in severe service degradation or "Internet meltdown". This
phenomenon was first observed during the early growth phase of the
Internet in the mid 1980s [RFC0896] [RFC0970]. It is technically
called "Congestion Collapse". [RFC2914] notes that informally,
"congestion collapse occurs when an increase in the network load
results in a decrease in the useful work done by the network." The
historical problem of congetsion collapse was largely due to TCP
connections unnecessarily retransmitting packets that were either in
transit or had already been received at the receiver. This became a
stable condition that can result in throughput that is a small
fraction of normal [RFC0896].
A second form of congestion collapse occurs due to undelivered
packets, where Section 5 of [RFC2914] notes: "Congestion collapse
from undelivered packets arises when bandwidth is wasted by
delivering packets through the network that are dropped before
reaching their ultimate destination. Different scenarios can result
in different degrees of congestion collapse, in terms of the fraction
of the congested links' bandwidth used for productive work. The
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danger of congestion collapse from undelivered packets is due
primarily to the increasing deployment of open-loop applications not
using end-to-end congestion control. Even more destructive would be
best-effort applications that increase their sending rate in response
to an increased packet drop rate (e.g., automatically using an
increased level of FEC (Forward Error Correction))."
The problems of congestion collapse have generally been corrected by
improvements to timer and congestion control mechanisms, that were
implemented in modern implementations of TCP [Jac88]. Transports
need to be specifically designed with measures to avoid starving
other flows of capacity (e.g., [RFC7567]). Section 3 discusses
Fairness, stating "The equitable sharing of bandwidth among flows
depends on the fact that all flows are running compatible congestion
control algorithms". Section 3.1 describes preventing congestion
collapse. [RFC2309] also discussed the dangers of congestion-
unresponsive flows, and states that "all UDP-based streaming
applications should incorporate effective congestion avoidance
mechanisms." [RFC7567] and [RFC8085] both reaffirm the continued
need to provide methods to prevent starvation.
1.4. Current IETF Guidelines on Evaluation of Congestion Control
Congestion control is an evolving subject, responding to changes in
protocol design, operation of applications using the network and
understanding of the network operation under load. The IETF has
provided guidance [RFC5033] for considering and evaluating alternate
congestion control algorithms.
The IRTF has described a set of metrics and related trade-off between
metrics that can be used to compare, contrast, and evaluate
congestion control techniques [RFC5166]. [RFC5783] provides a
snapshot of congestion-control research in 2008.
In contrast to fairness, a different approach is needed to analyse
persistent congestion effects (the collateral impact on loss,
starvation, collapse, etc). Such an analysis of the suitability of a
new mechanism needs to consider the impact on the flows that have
outliers in performance, (e.g., the last 5%, 1%) and specifically
needs to understand how changes impact other flows sharing a
bottleneck. For example, the flow performance often does not provide
any indication that a new method could starve other applications that
share the bottleneck capacity, or when patterns of packets (e.g.,
bursts) are sent that disrupt the packet timing needed by another
application flow (see Appendix A.1).
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1.5. Current Challenges
Recommendations and requirements on congestion control are
distributed across many documents in the RFC series. This document
therefore gathers and consolidates these recommendations. These, and
Internet engineering experience are used as a basis for the best
current practice in the design of congestion control methods for
Internet protocols.
The standardization of congestion control in new transports can avoid
a congestion control "arms race" among competing protocols [RFC2914].
That is, avoid designs of transports that could compete for Internet
resource in a way that significantly reduces the ability of other
flows to use the Internet.
The general recommendation in the UDP Guidelines [RFC8085] is that
applications SHOULD leverage existing congestion control techniques,
such as those defined for TCP [RFC9293], TCP-Friendly Rate Control
(TFRC) [RFC5348], SCTP [RFC4960], and other IETF-defined transports.
This is because there are many trade offs and details that can have a
serious impact on the performance of congestion control for the
application they support and other traffic that seeks to share the
resources along the path over which they communicate (see
Appendix A.1).
There are several reasons to think that things may have changed since
the original best current practice was published: At one time, it was
common that the serialisation delay of a packet at the bottleneck
formed a large proportion of the round time of a path, motivating a
need for conservative loss recovery. This is not often the case for
today's higher capacity links. This general increase in the link
speed often means that for many users, current traffic often does not
normally experience persistent congestion.
There also have been changes over time in the way that protocol
mechanisms are deployed in Internet endpoints:
On the one hand, techniques have evolved that now allow incremental
deployment and testing of new methods. This can enable more rapid
development of methods to detect and react to incipient congestion.
This allows new mechanisms can be tested to ensure that 95%, 99%, etc
of users see benefit in the networks they use. there has been
considerable progress in developing new loss recovery and congestion
responses that have been evaluated in this way.
On the other hand, the Internet continues to be heterogenous, some
endpoints experience very different network path characteristics and
some endpoints generate very different patterns of traffic. The IETF
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seeks to avoid congestion collapse, and also avoid prejudicing the
performance (e.g., throughput, latency) experienced when the Internet
is shared. The equitable or reasonable share of the bottleneck
capacity is often judged using a fairness metric.
The focus of the present document is upon unicast point-to-point
transports, this includes migration from using one path to another
path. Some recommendations [RFC5783] and requirements in this
document apply to point-to-multipoint transports (e.g., multicast),
however this topic extends beyond the current document's scope.
[RFC2914] provides additional guidance on the use of multicast.
2. Terminology
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 [RFC2119].
The path between endpoints (sometimes called "Internet Hosts" for
IPv4 and called "source nodes" and "destination nodes" in IPv6)
consists of the endpoint protocol stack at the sender and the
receiver (which together implement the transport service), and a
succession of links and network devices (routers or middleboxes) that
provide connectivity across a network path. The set of network
devices forming the path is not usually fixed, and it should
generally be assumed that this set can change over arbitrary lengths
of time.
[RFC5783] defines congestion control as "the feedback-based
adjustment of the rate at which data is sent into the network.
Congestion control is an indispensable set of principles and
mechanisms for maintaining the stability of the Internet." [RFC5783]
also provides an informational snapshot taken by the IRTF's Internet
Congestion Control Research Group (ICCRG) from October 2008.
The text draws on language used in the specifications of TCP and
other IETF transports. For example, a protocol timer is generally
needed to detect persistent congestion, and this document uses the
term Retransmission Timeout (RTO) to refer to the operation of this
timer. Similarly, the document refers to a congestion window (cwnd)
as the variable that controls the rate of transmission by the
congestion controller. The use of these terms does not imply that
endpoints need to implement functions in the way that TCP currently
does. Each new transport needs to make its own design decisions
about how to meet the recommendations and requirements for congestion
control.
Other terminology is directly copied from the cited RFCs.
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3. Requirements from the RFC Series
3.1. The need to React to Congestion
This includes:
* Endpoints MUST perform congestion control [RFC1122] and SHOULD
leverage existing congestion control techniques [RFC8085].
* If an application or protocol chooses not to use a congestion-
controlled transport protocol, it SHOULD control the rate at which
it sends datagrams to a destination host, in order to fulfil the
requirements of [RFC2914], as stated in [RFC8085].
* Endpoints ought not to use multiple congestion-controlled flows
between the same endpoints solely to gain a performance advantage.
Transports SHOULD control the aggregate traffic that the endpoint
sends on a path [RFC8085]. An endpoint can become aware of
congestion by various means (including, delay variation, timeout,
ECN, packet loss). A signal that indicates congestion on the end-
to-end network path, SHOULD result in a congestion control
reaction by the transport that reduces the current rate of the
sending endpoint for that path[RFC8087]).
* Although network devices can be configured to reduce the impact of
flow multiplexing on other flows, endpoints MUST NOT rely solely
on the presence and correct configuration of these methods, except
when they are constrained to operate in a controlled environment.
* A transport that does not target Internet deployment needs to be
constrained to only operate in a controlled environment (e.g., see
Section 3.6 of [RFC8085]) and provide appropriate mechanisms to
prevent this traffic from accidentally leaving the controlled
environment [RFC8084].
3.2. Tolerance to a Diversity of Path Characteristics
Internet transports need to use a congestion control method designed
for Internet paths.
* Path Capacity: The forward path can be congested in terms of the
number of packets it can support and/or the number of rate of
bytes it can transfer. The return path (towards the sender) can
be congested in terms of the number of packets it can support and/
or the number of rate of bytes it can transfer. Methods need to
operate over paths where capacity in the forward and return
directions are significantly different.
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* Path Loss: Paths can experience packet loss for various reasons
besides experiencing congestion. Whilst links and other
mechanisms below the transport layer can mitigate this loss, the
only way to surely confirm that a sending endpoint has
successfully communicated with a remote endpoint is to utilise a
timer (see Section 4.1) to detect a lack of response that could
result from a change in the path or the path characteristics. The
detection of congestion and the resulting reduction MUST NOT
solely depend upon reception of a signal from the remote endpoint,
because congestion indications could themselves be lost due to
congestion.
* Path Round Trip Time (RTT): The RTT from an endpoint cannot be
determined a-priori, and must be measured dynamically (see
Section 4.1).
* Path Change: An endpoint MUST assume that path characteristics can
change over time (i.e. path characteristics and sharing traffic
once discovered do not necessarily remain valid in the future).
* Network devices MAY provide mechanisms to mitigate the impact of
congestion collapse by transport flows (e.g., priority forwarding
of control information, and starvation detection), and ought to
mitigate the impact of non-conformant and malicious flows
[RFC7567]). These mechanisms complement, but do not replace, the
endpoint congestion avoidance mechanisms.
* Security: Internet endpoints need to be protected from intentional
disruption of the service they provide, and from the exploitation
of methods to attack other endpoints or services (see
Section 3.3).
3.3. Protection of Protocol Mechanisms
An endpoint needs to provide protection from attacks on the traffic
it generates, or attacks that seek to increase the capacity that is
consumed (impacting other traffic that share a bottleneck).
The following guidance is provided on protection from attack:
* Off-Path Attack: A design MUST protect from off-path attack to the
protocol [RFC8085] (i.e., an attack where the attacker is unable
to observe the packets exchanged across the path). Such an attack
on the congestion control can lead to a Denial of Service (DoS)
vulnerability for the flow being controlled and/or other flows
that share network resources along the path.
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* On-Path Attack: A protocol can be designed to protect from on-path
attacks (i.e., where the attacker can observe the packets
exchanged across the path). Protecting from on path attacks can
require more complexity and typically utilises encryption and/or
authentication mechanisms (e.g., IPsec [RFC4301], QUIC [RFC9000]).
* Validation of Signals: Network signals and control messages (e.g.,
ICMP [RFC0792]) MUST be validated before they are used to protect
from malicious abuse (see Section 3.3). This MUST at least
include protection from off-path attack [RFC8085] (e.g., by
validating the quoted information in an ICMP message corresponds
to the flow, before utilising the signalling it contains).
4. Principles of Congestion Control
This section summarises the principles for providing congestion
control. The section describes principles associated with preventing
persistent congestion; reacting to incipient congestion and utilising
additional path information.
4.1. Robustness: Timers and Retransmission
An endpoint can utilise timers to implement transport mechanisms,
these timers can be used to recover from loss; to trigger pre-emptive
retransmission and other protocol functions. Whilst parts of the
guidance below apply to these uses, the focus here is on detecting
and reacting to the congestion state of the path. An endpoint that
does utilise timers needs to follow the rules section 3.3 of
[RFC8085].
Principles include:
* Initial RTO Interval: When a flow sends the first packet(s), it
has no way to know the actual RTT of the path it will use. An
initial timer value is needed that will be used to detect lack of
responsiveness from the remote endpoint. In TCP, this is the
starting value of the RTO. The selection of a safe initial value
is a trade off that has important consequences on the overall
Internet stability [RFC6928] [RFC8085]. In the absence of any
knowledge about the latency of a path (including the initial
value), TCP SHOULD conservatively set the RTO to no less than 1
second. (Although Linux TCP has deployed a smaller initial RTO
value,the appendix of [RFC6298]) states that values shorter than 1
second can be problematic.)
* Initial RTO Expiry: If the RTO timer expires while awaiting
completion of a connection setup, or handshake (e.g., the ACK of a
SYN segment in the three-way handshake in TCP), and the
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implementation is using an RTO of less than 3 seconds, the local
endpoint can resend the connection setup. [[Author note: It would
be useful to discuss how the timer is managed to protect from
multiple handshake failure]]. This RTO MUST then be re-
initialized to increase it to 3 seconds when data transmission
begins (i.e., after the handshake completes) [RFC6298] [RFC8085].
This conservative increase is necessary to avoid congestion
collapse when many flows retransmit across a shared bottleneck
with restricted capacity.
* Initial Measured RTO: Once an RTT measurement is available (e.g.,
through reception of an acknowledgement), the timeout value must
be adjusted. This adjustment MUST take into account the RTT
variance. For the first sample, this variance cannot be
determined, and a local endpoint MUST therefore initialise the
variance to RTT/2 (see equation 2.2 of [RFC6928] and related text
for UDP in section 3.1.1 of [RFC8085]).
* Updating the Path RTT: Once an endpoint has started communicating
with its peer, the RTT MUST be adjusted by measuring the actual
path RTT. This adjustment MUST include adapting to the measured
RTT variance (see equation 2.3 of [RFC6928]). An RTO interval
SHOULD be set based on recent RTT observations (including the RTT
variance) (e.g., Section 3.1.1 of [RFC8085]).
* Persistent Lack of Feedback: Persistent lack of feedback (e.g.,
detected by an RTO timer expiry, or other means) MUST be treated
as an indication of persistent congestion. A failure to receive
any specific response within an RTO interval could be potentially
a result of a RTT change, change of path, excessive loss, or even
congestion collapse. If there is no response within the RTO
interval, TCP collapses the congestion window to one segment
[RFC9293]. Other transports MUST similarly respond when they fail
to receive confirmation of feedback. An endpoint also needs to
exponentially backoff the RTO interval [RFC8085] each time
persistent congestion is detected [RFC1122], until the path
characteristics can again be confirmed That is, the RTO interval
MUST be set to at least the RTO * 2 [RFC6298] [RFC8085].
* Maximum RTO: A maximum value MAY be placed on the RTO interval.
This maximum limit to the RTO interval MUST NOT be less than 60
seconds [RFC6298].
* [[Author Note: Re-check RTO-Consider. ]]
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4.2. Detecting and Reacting to Incipient Congestion
This section describes the principles related to mitigation of
incipient congestion Section 1.2.
4.2.1. Congestion Initialization
When a connection or flow to a new destination is first established,
the endpoints have little information about the characteristics of
the network path they will use.
* Flow Start: A new flow between two endpoints needs to initialise a
congestion controller for the path it will use. It MUST NOT
assume that capacity is available at the start of the flow, unless
it uses a mechanism to explicitly reserve capacity. In the
absence of a capacity signal, a flow can be expected to start
slowly. The TCP slow-start algorithm is an accepted standard for
flow startup [RFC9293], which uses the notion of an Initial Window
(IW) [RFC3390], updated by [RFC6928]) to define the initial volume
of data that can be sent on a path. This is not the smallest
burst size, nor the smallest window, but it is considered a safe
starting point for a path that is not suffering persistent
congestion, and is applicable until feedback about the path is
received. The initial sending rate needs to be viewed as
tentative, until capacity is confirmed to be available.
* Cached State: A congestion controller MAY assume that the recently
used capacity between a pair of endpoints is an indication of
future capacity that might be available in the next RTT between
the same endpoints (Section 4.2.6). The congestion controller
MUST reduce its rate if this is not subsequently confirmed to be
true. [[Author note: we likely need to bound this reaction in
time or size]].
4.2.2. Using Path Capacity
This section describes how a sender needs to regulate the maximum
volume of data in flight over the interval of the current RTT, and
how it manages the use of the capacity that it perceives is
available, and reacts to incipient congestion.
* Setting a Congestion Window: Standard TCP states that a TCP sender
"SHOULD set the congestion window to no more than the Restart
Window (R)" before beginning transmission, if the sender has not
sent data in an interval that exceeds the current retransmission
timeout, i.e., when an application becomes idle [RFC9293].
Congestion Window Validation (CWV) [RFC7661] describes how a TCP
sender can tentatively maintain a congestion window that is larger
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than the path has supported in the last RTT when it is
application-limited, provided that the endpoint appropriately and
rapidly reduces the congestion window when potential congestion is
detected.
* Using the Congestion Window: Unless managed by a resource
reservation protocol, path capacity information is transient. A
sender that does not use capacity has no understanding whether
previously used capacity remains available to use, or whether that
capacity has disappeared (e.g., a change in the path that causes a
flow to experience a smaller bottleneck, or when more traffic
emerges that consumes previously available capacity resulting in a
new bottleneck). For this reason, a transport that is limited by
the volume of data available to send MUST NOT continue to grow its
congestion window when the current congestion window is more than
twice the volume of data acknowledged in the last RTT.
* Collateral Damage: Even in the absence of congestion, statistical
multiplexing of flows can result in transient effects for flows
sharing common resources. A sender therefore SHOULD avoid
inducing excessive congestion to other flows (collateral damage
that could result in flow starvation).
* Burst Mitigation: While a congestion controller ought to limit
sending at the granularity of the current RTT, this can be
insufficient to satisfy the goals of mitigating collateral damage.
This requires moderating the burst rate of the sender to avoid
significant periods where a flow(s) consume all buffer capacity at
the path bottleneck, which would otherwise prevent other flows
from gaining a reasonable share. Endpoints SHOULD provide
mechanisms to regulate the bursts of transmission that the
application/protocol sends to the network (section 3.1.6 of
[RFC8085]). ACK-Clocking [RFC9293] can help mitigate bursts for
protocols that receive continuous feedback of reception (such as
TCP). Sender pacing can also mitigate this [RFC8085], (described
in Section 4.6 of [RFC3449]), and has been recommended for TCP in
conditions where ACK-Clocking is not effective, (e.g., [RFC3742],
[RFC7661]). SCTP [RFC4960] defines a maximum burst length
(Max.Burst) with a recommended value of 4 segments to limit the
SCTP burst size. QUIC recommends that a sender paces sending of
all in-flight packets based on input from the congestion
controller[RFC9002].
4.2.3. Congestion Detection and Retransmission
This section describes mechanisms to detect and provide
retransmission, and to protect the network in the absence of timely
feedback.
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* Congestion Detection: Loss is typically detected when a sender
determines there is no delivery confirmation within an expected
period of time (e.g., by observing the time-ordering of the
reception of ACKs, as in TCP DupACK) or by utilising a timer to
detect loss (e.g., a transmission timer with a period less than
the RTO, [RFC8085] [RFC8985]) or a combination of using a timer
and ordering information to trigger retransmission of data. A
transport is usually unable to reliably detect whether a loss is a
result of congestion or another reason. For this reason, loss
needs to be treated as incipient congestion, at least until the
cause of loss can be reliably determined.
* Retransmission: When loss is detected, the sender can choose to
retransmit the lost data, ignore the loss, or send other data
(e.g., [RFC8085] [RFC9002]), depending on the reliability model
provided by the transport service. All transmissions consumes
network capacity, therefore retransmissions MUST NOT increase the
network load in response to congestion loss (which worsens that
congestion) [RFC8085]. Any method that sends additional data
following loss is therefore responsible for congestion control of
the retransmissions (and any other packets sent, including FEC
information) as well as the original traffic.
4.2.4. Responding to Incipient Congestion
The safety and responsiveness of new proposals need to be evaluated
[RFC5166]. In determining an appropriate congestion response to
incipient congestion, designs could take into consideration the size
of the packets that experience congestion [RFC4828].
* Congestion Response: An endpoint MUST promptly reduce the rate of
transmission when it receive or detects an indication of
congestion (e.g., loss) [RFC2914]. TCP Reno established a method
that relies on multiplicative-decrease to halve the sending rate
while congestion is detected. This response to congestion
indications is considered sufficient for safe Internet operation,
but other decrease factors have also been published in the RFC
Series [I-D.ietf-tcpm-rfc8312bis].
* ECN Detection: ECN can help determine an appropriate congestion
window to enable early indication of incipient congestion when it
is supported by routers on the path [RFC7567]. An early detection
of incipient congestion allows a different reaction to an explicit
congestion signal compared to the reaction to a detected packet
loss [RFC8311] [RFC8087]. congestion control design should
provide the necessary mechanisms to support ECN [RFC3168]
[RFC6679], as described in section 3.1.7 of [RFC8085].
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* Response to ECN Congestion Marking: Simple feedback of received
Congestion Experienced (CE) marks [RFC3168], relies only on an
indication that congestion has been experienced within the last
RTT. This style of response is appropriate when a flow uses
ECT(0) [RFC3168]. ABE included a modification to the reaction to
ECN [RFC8511]. Further detail about the received CE-marking can
be obtained by using more accurate receiver feedback (e.g.,
[I-D.ietf-tcpm-accurate-ecn] and extended RTP feedback). The more
detailed feedback provides an opportunity for a finer-granularity
of congestion response. The L4S architecture [RFC9330] allows
routers to use a different marking system that can provide early
reaction to incipient congestion [RFC9332] and defines a reaction
for this feedback when packets marked with ECT(1).
* [RFC8085] provides guidelines for a sender that does not, or is
unable to, adapt the congestion window.
4.2.5. Using More Capacity
In the phase where a sender is increasing the congestion window, it
will transmit faster than the last confirmed safe rate. Such an
increase above the last confirmed rate needs to be regarded as
tentative and a sender needs to reduce its rate below the last
confirmed safe rate when congestion is detected (see also
Appendix A.2).
* Increasing Congestion Window: In the absence of congestion, an
endpoint MAY increase its congestion window and hence the sending
rate. An increase should only occur when there is additional data
available to send across the path (i.e., the sender will utilise
the additional capacity in the next RTT). A sender MUST NOT
increase the sending rate for more than one RTT after congestion
is first detected. This helps manage incipient congestion.
* Avoiding Overshoot: Overshoot of the congestion window beyond the
point of congestion can significantly impact other flows sharing
resources along a path, and can impact the performance of the flow
itself. As endpoints experience more paths with a large Bandwidth
Delay Product (BDP) and a wider range of potential path RTT,
variability or changes in the path can significantly impact the
appropriate dynamics for increasing a congestion window (see also
burst mitigation, Section 4.2.2). Methods such as HyStart are
designed to avoid overshoot [RFC9406].
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* Reponse to Detected Congestion: An endpoint MUST utilise a method
that assures the sender will keep the rate below the previously
confirmed safe rate for multiple RTT periods after an observed
congestion event. In TCP, this is performed by using a linear
increase from a slow start threshold that is re-initialised when
congestion is experienced.
4.2.6. Utilising Additional Path Information
An endpoint is permitted to cache path information. This could be
used to inform parameter selection for a new or on-going flow. An
endpoint might also utilise signals from the network to help
determine how to regulate the traffic it sends.
Any information that is used to accelerate the growth of the
congestion window MUST be viewed as tentative until the path capacity
is confirmed by receiving a confirmation that actual traffic has been
sent across the path. (i.e., the new flow needs to either use or
loose the capacity that has been tentatively offered to it). A
sender MUST reduce its rate if this capacity is not confirmed within
the current RTO interval.
* Utilising Cached Path Information: A congestion controller that
recently used a specific path could allow a flow to take-over the
capacity that was previously consumed by another flow (e.g., in
the last RTT) which it understands is using the same path and no
will longer use the capacity it recently used. In TCP, this
mechanism was called TCP Control Block (TCB) sharing and is now
called TCP Control Block Interdependence, and is described in
[RFC9040]. The capacity and other information can be used to
suggest a faster initial sending rate.
* Receiving Network Signals: Mechanisms MUST NOT solely rely on
transport messages or specific signalling messages to perform
safely. (Section 5.2 of [RFC8085] describes use of ICMP
messages). Mechanisms need to be designed to safely operate when
path characteristics can change at any time. Transport mechanisms
MUST be robust to potential loss of any signals. Loss or
modification of packets can occur after a path changes, even when
a signal was successfully first used by a flow, see Section 3.2).
* Utilising Network Signals: A mechanism that utilises signals
originating in the network (e.g., RSVP, NSIS, Quick-Start, ECN),
MUST assume that the set of network devices on the path can
change. This motivates a design that uses soft-state for
protocols that interact with signals originating from network
devices [RFC9049] (e.g., ECN) and includes context-sensitive
treatment of "soft" signals provided to the endpoint [RFC5164].
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Endpoit CC MUST assume the set of routers and links the forming
the forward path can change and that network devices can be
reconfigured, reset, or the set of devices on-path can change,
therefore information about which packets traverse a packet (e.g.
whether IP options are supported, or a specific treatment
applies.)
4.3. Avoiding Persistent Congestion
All endpoints are required to implement mechanisms that avoid the
persistent congestion and can demonstrate that they do not induce
starvation and congestion collapseSection 1.3.
Principles include:
* Persistent congestion can result in congestion collapse, which
MUST be aggressively avoided [RFC2914]. Endpoints that experience
persistent congestion and have already exponentially reduced their
congestion window to the restart window (e.g., one packet), MUST
further reduce the rate if the RTO timer continues to expire. For
example, TFRC [RFC5348] continues to reduce its sending rate under
persistent congestion to one packet per RTT, and then
exponentially backs-off the time between single packet
transmissions if a congestion event continues to persist
[RFC2914]. QUIC [RFC9002] does not directly specify a period, but
does specify a probe to detect tail loss. The Tail Loss Probe
(TLP) mechanism [RFC8985] determines that persisent congestion is
experienced after a loss for a duration of 2 TLP probes plus the
RTO.
4.3.1. Avoiding Congestion Collapse and Flow Starvation
Principles include:
* Transports MUST avoid inducing flow starvation to the other flows
that share resources along the path they use.
* Endpoints MUST treat a loss of all feedback (e.g., expiry of a
retransmission time out, RTO) as an indication of persistent
congestion (i.e., an indication of potential congestion collapse).
* When an endpoint detects persistent congestion, it MUST reduce the
maximum rate (e.g., reduce its congestion window). This normally
involves the use of protocol timers to detect a lack of
acknowledgment for transmitted data (see Section 4.1).
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5. Acknowledgements
This document owes much to the insight offered by Sally Floyd, both
at the time of writing of RFC2914 and her help and review in the many
years that followed this.
Nicholas Kuhn helped develop the first draft of these guidelines.
Tom Jones and Ana Custura reviewed the first version of this draft.
Many discussions with Michael Welzl and others have provided
imeasurale help to get this far. The University of Aberdeen received
funding to support this work from the European Space Agency.
6. IANA Considerations
This memo includes no request to IANA.
RFC Editor Note: If there are no requirements for IANA, the section
will be removed during conversion into an RFC by the RFC Editor.
7. Security Considerations
This document introduces no new security considerations. Each RFC
listed in this document discusses the security considerations of the
specification it contains. The security considerations for the use
of transports are provided in the references section of the cited
RFCs. Security guidance for applications using UDP is provided in
the UDP Usage Guidelines [RFC8085].
Section 3.3 describes general requirements relating to the design of
safe protocols and their protection from on and off path attack.
Section 4.2.6 follows current best practice to validate ICMP messages
prior to use.
8. Normative References
[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122,
DOI 10.17487/RFC1122, October 1989,
<https://www.rfc-editor.org/info/rfc1122>.
[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>.
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[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41,
RFC 2914, DOI 10.17487/RFC2914, September 2000,
<https://www.rfc-editor.org/info/rfc2914>.
[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>.
[RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's
Initial Window", RFC 3390, DOI 10.17487/RFC3390, October
2002, <https://www.rfc-editor.org/info/rfc3390>.
[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>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<https://www.rfc-editor.org/info/rfc6298>.
[RFC7567] Baker, F., Ed. and G. Fairhurst, Ed., "IETF
Recommendations Regarding Active Queue Management",
BCP 197, RFC 7567, DOI 10.17487/RFC7567, July 2015,
<https://www.rfc-editor.org/info/rfc7567>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/info/rfc8085>.
9. Informative References
[Flow-Rate-Fairness]
Briscoe, Bob., "Flow Rate Fairness: Dismantling a
Religion, ACM Computer Communication Review 37(2):63-74",
April 2007.
[I-D.ietf-tcpm-accurate-ecn]
Briscoe, B., Kühlewind, M., and R. Scheffenegger, "More
Accurate Explicit Congestion Notification (ECN) Feedback
in TCP", Work in Progress, Internet-Draft, draft-ietf-
tcpm-accurate-ecn-24, 30 March 2023,
<https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-
accurate-ecn-24>.
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[I-D.ietf-tcpm-rfc8312bis]
Xu, L., Ha, S., Rhee, I., Goel, V., and L. Eggert, "CUBIC
for Fast and Long-Distance Networks", Work in Progress,
Internet-Draft, draft-ietf-tcpm-rfc8312bis-15, 31 January
2023, <https://datatracker.ietf.org/doc/html/draft-ietf-
tcpm-rfc8312bis-15>.
[I-D.nishida-tcpm-standard-cc-analysis]
Nishida, Y., "Analysis for the Differences Between
Standard Congestion Control Schemes", Work in Progress,
Internet-Draft, draft-nishida-tcpm-standard-cc-analysis-
01, 9 November 2022,
<https://datatracker.ietf.org/doc/html/draft-nishida-tcpm-
standard-cc-analysis-01>.
[Jac88] Jacobson, V., "Congestion Avoidance and Control", Computer
Communication Review, vol. 18, no. 4, pp. 314-329 , August
1988, <ftp://ftp.ee.lbl.gov/papers/congavoid.ps.Z.>.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980,
<https://www.rfc-editor.org/info/rfc768>.
[RFC0792] Postel, J., "Internet Control Message Protocol", STD 5,
RFC 792, DOI 10.17487/RFC0792, September 1981,
<https://www.rfc-editor.org/info/rfc792>.
[RFC0896] Nagle, J., "Congestion Control in IP/TCP Internetworks",
RFC 896, DOI 10.17487/RFC0896, January 1984,
<https://www.rfc-editor.org/info/rfc896>.
[RFC0970] Nagle, J., "On Packet Switches With Infinite Storage",
RFC 970, DOI 10.17487/RFC0970, December 1985,
<https://www.rfc-editor.org/info/rfc970>.
[RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
S., Wroclawski, J., and L. Zhang, "Recommendations on
Queue Management and Congestion Avoidance in the
Internet", RFC 2309, DOI 10.17487/RFC2309, April 1998,
<https://www.rfc-editor.org/info/rfc2309>.
[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
and W. Weiss, "An Architecture for Differentiated
Services", RFC 2475, DOI 10.17487/RFC2475, December 1998,
<https://www.rfc-editor.org/info/rfc2475>.
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[RFC2525] Paxson, V., Allman, M., Dawson, S., Fenner, W., Griner,
J., Heavens, I., Lahey, K., Semke, J., and B. Volz, "Known
TCP Implementation Problems", RFC 2525,
DOI 10.17487/RFC2525, March 1999,
<https://www.rfc-editor.org/info/rfc2525>.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616,
DOI 10.17487/RFC2616, June 1999,
<https://www.rfc-editor.org/info/rfc2616>.
[RFC3449] Balakrishnan, H., Padmanabhan, V., Fairhurst, G., and M.
Sooriyabandara, "TCP Performance Implications of Network
Path Asymmetry", BCP 69, RFC 3449, DOI 10.17487/RFC3449,
December 2002, <https://www.rfc-editor.org/info/rfc3449>.
[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>.
[RFC3742] Floyd, S., "Limited Slow-Start for TCP with Large
Congestion Windows", RFC 3742, DOI 10.17487/RFC3742, March
2004, <https://www.rfc-editor.org/info/rfc3742>.
[RFC3819] Karn, P., Ed., Bormann, C., Fairhurst, G., Grossman, D.,
Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and L.
Wood, "Advice for Internet Subnetwork Designers", BCP 89,
RFC 3819, DOI 10.17487/RFC3819, July 2004,
<https://www.rfc-editor.org/info/rfc3819>.
[RFC3828] Larzon, L., Degermark, M., Pink, S., Jonsson, L., Ed., and
G. Fairhurst, Ed., "The Lightweight User Datagram Protocol
(UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July 2004,
<https://www.rfc-editor.org/info/rfc3828>.
[RFC4301] Kent, S. and K. Seo, "Security Architecture for the
Internet Protocol", RFC 4301, DOI 10.17487/RFC4301,
December 2005, <https://www.rfc-editor.org/info/rfc4301>.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340,
DOI 10.17487/RFC4340, March 2006,
<https://www.rfc-editor.org/info/rfc4340>.
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[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>.
[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007,
<https://www.rfc-editor.org/info/rfc4960>.
[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion
Control Algorithms", BCP 133, RFC 5033,
DOI 10.17487/RFC5033, August 2007,
<https://www.rfc-editor.org/info/rfc5033>.
[RFC5164] Melia, T., Ed., "Mobility Services Transport: Problem
Statement", RFC 5164, DOI 10.17487/RFC5164, March 2008,
<https://www.rfc-editor.org/info/rfc5164>.
[RFC5166] Floyd, S., Ed., "Metrics for the Evaluation of Congestion
Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March
2008, <https://www.rfc-editor.org/info/rfc5166>.
[RFC5783] Welzl, M. and W. Eddy, "Congestion Control in the RFC
Series", RFC 5783, DOI 10.17487/RFC5783, February 2010,
<https://www.rfc-editor.org/info/rfc5783>.
[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework", RFC 6363,
DOI 10.17487/RFC6363, October 2011,
<https://www.rfc-editor.org/info/rfc6363>.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
2012, <https://www.rfc-editor.org/info/rfc6679>.
[RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A
Datagram Congestion Control Protocol UDP Encapsulation for
NAT Traversal", RFC 6773, DOI 10.17487/RFC6773, November
2012, <https://www.rfc-editor.org/info/rfc6773>.
[RFC6928] Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis,
"Increasing TCP's Initial Window", RFC 6928,
DOI 10.17487/RFC6928, April 2013,
<https://www.rfc-editor.org/info/rfc6928>.
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[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
Control Transmission Protocol (SCTP) Packets for End-Host
to End-Host Communication", RFC 6951,
DOI 10.17487/RFC6951, May 2013,
<https://www.rfc-editor.org/info/rfc6951>.
[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015,
<https://www.rfc-editor.org/info/rfc7661>.
[RFC7806] Baker, F. and R. Pan, "On Queuing, Marking, and Dropping",
RFC 7806, DOI 10.17487/RFC7806, April 2016,
<https://www.rfc-editor.org/info/rfc7806>.
[RFC8084] Fairhurst, G., "Network Transport Circuit Breakers",
BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017,
<https://www.rfc-editor.org/info/rfc8084>.
[RFC8087] Fairhurst, G. and M. Welzl, "The Benefits of Using
Explicit Congestion Notification (ECN)", RFC 8087,
DOI 10.17487/RFC8087, March 2017,
<https://www.rfc-editor.org/info/rfc8087>.
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion
Notification (ECN) Experimentation", RFC 8311,
DOI 10.17487/RFC8311, January 2018,
<https://www.rfc-editor.org/info/rfc8311>.
[RFC8511] Khademi, N., Welzl, M., Armitage, G., and G. Fairhurst,
"TCP Alternative Backoff with ECN (ABE)", RFC 8511,
DOI 10.17487/RFC8511, December 2018,
<https://www.rfc-editor.org/info/rfc8511>.
[RFC8985] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The
RACK-TLP Loss Detection Algorithm for TCP", RFC 8985,
DOI 10.17487/RFC8985, February 2021,
<https://www.rfc-editor.org/info/rfc8985>.
[RFC9000] Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", RFC 9000,
DOI 10.17487/RFC9000, May 2021,
<https://www.rfc-editor.org/info/rfc9000>.
[RFC9002] Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
and Congestion Control", RFC 9002, DOI 10.17487/RFC9002,
May 2021, <https://www.rfc-editor.org/info/rfc9002>.
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[RFC9040] Touch, J., Welzl, M., and S. Islam, "TCP Control Block
Interdependence", RFC 9040, DOI 10.17487/RFC9040, July
2021, <https://www.rfc-editor.org/info/rfc9040>.
[RFC9049] Dawkins, S., Ed., "Path Aware Networking: Obstacles to
Deployment (A Bestiary of Roads Not Taken)", RFC 9049,
DOI 10.17487/RFC9049, June 2021,
<https://www.rfc-editor.org/info/rfc9049>.
[RFC9293] Eddy, W., Ed., "Transmission Control Protocol (TCP)",
STD 7, RFC 9293, DOI 10.17487/RFC9293, August 2022,
<https://www.rfc-editor.org/info/rfc9293>.
[RFC9330] Briscoe, B., Ed., De Schepper, K., Bagnulo, M., and G.
White, "Low Latency, Low Loss, and Scalable Throughput
(L4S) Internet Service: Architecture", RFC 9330,
DOI 10.17487/RFC9330, January 2023,
<https://www.rfc-editor.org/info/rfc9330>.
[RFC9332] De Schepper, K., Briscoe, B., Ed., and G. White, "Dual-
Queue Coupled Active Queue Management (AQM) for Low
Latency, Low Loss, and Scalable Throughput (L4S)",
RFC 9332, DOI 10.17487/RFC9332, January 2023,
<https://www.rfc-editor.org/info/rfc9332>.
[RFC9406] Balasubramanian, P., Huang, Y., and M. Olson, "HyStart++:
Modified Slow Start for TCP", RFC 9406,
DOI 10.17487/RFC9406, May 2023,
<https://www.rfc-editor.org/info/rfc9406>.
Appendix A. Internet Congestion Control
A.1. Flow Multiplexing and Congestion
When a transport uses a path to send packets (i.e. a flow), this
often impacts other Internet flows (possibly from or to other
endpoints) that share the capacity of any common network device or
link (i.e., are multiplexed) along the path. This can result in
variation in capacity, loss, or additonal latency. It can result
from various interactions: [RFC3819] (Quality of Service link
scheduling, link radio resource management/bandwidth on demand,
transient outages, link retransmission, and connection/resource setup
below the IP layer, etc).
It is therefore normal to observe some perturbation in latency and/or
loss when flows shares a common network bottleneck with other
traffic. This impact needs to be considered and Internet flows ought
to implement appropriate safeguards to avoid inappropriate impact on
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other flows that share the resources along a path. Congestion
control methods satisfy this requirement and therefore can help avoid
congestion collapse.
"This raises the issue of the appropriate granularity of a 'flow',
where we define a 'flow' as the level of granularity appropriate for
the application of both fairness and congestion control. [RFC2309]
states: "There are a few `natural' answers: 1) a TCP or UDP
connection (source address/port, destination address/port); 2) a
source/destination host pair; 3) a given source host or a given
destination host. We would guess that the source/destination host
pair gives the most appropriate granularity in many circumstances.
The granularity of flows for congestion management is, at least in
part, a policy question that needs to be addressed in the wider IETF
community." [RFC2914]
Many designs place the responsibility of rate-adaption for congestion
control at the sender (source) endpoint, utilising feedback
information provided by the remote endpoint (receiver). Congestion
control can also be implemented by determining an appropriate rate
limit at the receiver and using this limit to control the maximum
transport rate (e.g., using methods such as [RFC5348] and [RFC4828]).
Applications at an endpoint can send more than one flow. "The
specific issue of a browser opening multiple connections to the same
destination has been addressed by [RFC2616]. Section 8.1.4 states
that "Clients that use persistent connections SHOULD limit the number
of simultaneous connections that they maintain to a given server. A
single-user client SHOULD NOT maintain more than 2 connections with
any server or proxy." [RFC9040].
This suggests that there are opportunities for transport connections
between the same endpoints (from the same or differing applications)
might share some information, including their congestion control
state, if they are known to share the same path. [RFC8085] adds "An
application that forks multiple worker processes or otherwise uses
multiple sockets to generate UDP datagrams SHOULD perform congestion
control over the aggregate traffic."
A.2. Adjusting the Rate
TCP Reno [RFC9293] defines an algorithm, known as the Additive-
Increase/ Multiplicative-Decrease (AIMD) algorithm, which allows a
sender to exponentially increase the congestion window each RTT from
the initial window to the first detected congestion event. This is
designed to allow new flows to rapidly acquire a suitable congestion
window. Where the bandwidth delay product (BDP) is large, it can
take many RTT periods to determine a suitable share of the path
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capacity. Such high BDP paths benefit from methods that more rapidly
increase the congestion window, but in compensation these need to be
designed to also react rapidly to any detected congestion (e.g., TCP
Cubic [I-D.ietf-tcpm-rfc8312bis]).
* The capacity available to a flow could be expressed as the number
of bytes in flight, the sending rate or a limit on the number of
unacknowledged segments. When determining the capacity used, all
data sent by a sender needs to be accounted, this includes any
additional overhead or data generated by the transport. A
transport performing congestion management will usually optimise
performance for its application by avoiding excessive loss or
delay and maintain a congestion window. In steady-state this
congestion window reflects a safe limit to the sending rate that
has not resulted in persistent congestion. A congestion
controller for a flow that uses packet Forward Error Correction
(FEC) encoding (e.g., [RFC6363]) needs to consider all additional
overhead introduced by packet FEC when setting and managing its
congestion window.
* One common model views the path between two endpoints as a "pipe".
New packets enter the pipe at the sending endpoint, older ones
leave the pipe at the receiving endpoint. Congestion and other
forms of loss result in "leakage" from this pipe. Received data
(leaving the network path at the remote endpoint) is usually
acknowledged to the congestion controller.
* The rate that data leaves the pipe indicates the share of the
capacity that has been utilised by the flow. If, on average (over
an RTT), the sending rate equals the receiving rate, this
indicates the path capacity. This capacity can be safely used
again in the next RTT. If the average receiving rate is less than
the sending rate, then the path is either queuing packets, the
RTT/path has changed, or there is packet loss.
Appendix B. Revision Notes
Note to RFC-Editor: please remove this entire section prior to
publication.
Previous versions of the document were presented and discsussed in
tsvwg, and eveolved through several versions. This version is a
refocus towards the newly formed CC Working Group where it is offered
as a candidate for progression.
Individual draft -00:
* First draft contributed to CC WG targeting publication as BCP.
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Author's Address
Godred Fairhurst
University of Aberdeen
School of Engineering
Fraser Noble Building
Aberdeen
AB24 3UE
United Kingdom
Email: gorry@erg.abdn.ac.uk
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