Internet Engineering Task Force G. Fairhurst
Internet-Draft University of Aberdeen
Intended status: Standards Track July 05, 2019
Expires: January 6, 2020

Guidelines for Internet Congestion Control at Endpoints


This document provides guidance on the design of methods to avoid congestion collapse and to provide congestion control. Recommendations and requirements on this topic are distributed across many documents in the RFC series. It seeks to gather and consolidate these recommendations. This is intended to provide input to the design of new congestion control methods in protocols, such as IETF QUIC.

The present document is for discussion and comment by the IETF.

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

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This Internet-Draft will expire on January 6, 2020.

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Table of Contents

1. Introduction

The IETF has specified Internet transports (e.g., TCP [ID.ietf-tcpm-rfc793bis], UDP [RFC0768], UDP-Lite [RFC3828], SCTP [RFC4960], and DCCP [RFC4340]) as well as protocols layered on top of these transports (e.g., RTP, QUIC [I-D.ietf-quic-transport], SCTP/UDP [RFC6951], DCCP/UDP) and transports that work directly over the IP network layer. These transports are implemented in endpoints (Internet hosts or routers acting as endpoints) and are designed to detect and react to network congestion.

Recommendations and requirements on this topic are distributed across many documents in the RFC series. This document seeks to gather and consolidate these recommendations. This is intended to provide input to the design of new congestion control methods in protocols.

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") consists of the endpoint protocol stack at the sender and receiver (which implements the transport service), and a succession of links and network devices (routers or middleboxes) that provide connectivity across the network. 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.

Other terminology is directly copied from the cited RFCs.

3. Principles of Congestion Control

This section summarises the principles for providing congestion control, and provides the background for section Section 4.

3.1. A Diversity of Path Characteristics

Internet transports do not usually rely upon prior reservation of capacity along the path they use. In the absence of such a resource 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 respond to changes in the capacity available across the network path. Buffering (an increase in latency) or loss (discard of a packet) arises when the traffic arriving at a link or network exceeds the resources available.

A network device that does not support Active Queue Management (AQM) [RFC7567] typically uses a drop-tail policy to drop excess IP packets when its queue becomes full. Although losses are not always due to congestion (loss may be due to link corruption, receiver overrun, etc. [RFC3819]), endpoint congestion control has to conservatively assume that any loss is potentially due to congestion and then reduce the sending rate of their flows to reflect the available capacity.

The use of a path to send packets impacts any flows (possibly from or to other endpoints) that share the capacity (i.e., multiplex packets) using a common network device or link. Even when a path is not congested, flows can still experience an increased latency when the path multiplexes traffic belonging to multiple flows. As with loss, latency can also be incurred for other reasons [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).

Principles include:

3.2. Flow Multiplexing and Congestion

It is normal to observe some perturbation in latency or loss to traffic when it 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 other flows that share the resources along a path. Congestion control methods satisfy this requirement and therefore avoid congestion collapse [@ARTICLE{author = {Bob Briscoe}, title = {Flow Rate Fairness: Dismantling a Religion}, journal = {ACM CCR}, year = {2007} }].

An endpoint can become aware of congestion by various means. A signal that indicates congestion on the end-to-end network path, must result in a congestion control reaction by the transport to reduce the maximum rate permitted by the sending endpoint [RFC8087]]. Internet transports should react to avoid congestion that impacts other flows sharing a path, and need to be designed to avoid starving other flows of capacity. This could include methods seeking to equally distribute resources between sharing flows, but this is explicitly not a requirement for a network device.

The Requirements for Internet Hosts 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 traffic [RFC2914]. They may also need to implement additional mechanisms, depending on how they use UDP" [RFC8085]. [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] reaffirm this.

The general recommendation in the UDP Guidelines [RFC8085] is that applications SHOULD leverage existing congestion control techniques, such as those defined for TCP [RFC5681], 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.

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.

Principles include:

3.3. Avoiding Congestion Collapse

A significant pathology can arise when a poorly designed transport creates 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 [RFC896] [RFC970]; it is technically called "congestion collapse" and was a key focus of [RFC2309].

4. Guidelines for Performing Congestion Control

This section provides guidance for designers of a new transport protocol that decide to implement congestion control and its associated mechanisms.

4.1. Connection Initialization

When a connection or flow to a new destination is established, the endpoints have little information about the characteristics of the network path. This section describes how a flow starts transmission over such a path.

Flow Start:
A new flow between two endpoints cannot 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 MUST therefore start slowly.
The slow-start algorithm is the accepted standard for flow startup [RFC5681]. TCP 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, or the smallest window - it is considered a safe starting point for a network that is not suffering persistent congestion, and applicable until feedback about the path is received. This initial sending rate needs to be viewed as tentative until the capacity is confirmed to be available.
Initial RTO Interval:
When a flow sends the first packet it typically has no way to know the actual RTT of the path it uses. The initial value used to the Retransmission Timeout (RTO) is therefore 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, the RTO MUST be conservatively set to no less than 1 second. Values shorter than 1 second can be problematic (see the appendix of [RFC6298]).
Initial RTO Expiry:
If the RTO timer expires while awaiting completion of the connection setup (in TCP, the ACK of a SYN segment), and the implementation is using an RTO less than 3 seconds, the local endpoint can resend the connection setup. The RTO MUST then be re-initialized to increase it to 3 seconds when data transmission begins (i.e., after the three-way 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), this value must be adjusted, and 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]).
Current State:
A congestion controller MAY assume that recently used capacity between a pair of endpoint addresses is an indication of capacity available in the next RTT between the same endpoints (and react accordingly if this is not confirmed to be true).
Cached State:
A congestion controller that recently used a path could use additional state that lets a flow take-over the capacity that was previously consumed by another flow (e.g., in the last RTT). In TCP, this mechanism is referred to as TCP Control Block (CB) sharing [RFC2140] [ID.ietf-tcpm-2140bis]. This and other information can be used to suggest a faster initial sending rate, but MUST be viewed as tentative until the capacity is confirmed to be available. A sender MUST reduce its rate if this capacity is not confirmed within the current RTO interval.

4.2. 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 RT, and how it manages transmission of the capacity that it perceives is available.

Congestion Management:
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. In steady-state this congestion window reflects a safe limit to the sending rate that has not resulted in persistent congestion. A sender performing congestion management will usually optimise performance for its application by avoiding excessive loss or delay.
One common model views the path between two endpoints as a pipe. New packets enter the pipe at the sending endpoint, older ones leave at the receiving endpoint, and are usually acknowledged to the sender. 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 RT), the sending rate equals the receiving rate, this indicates that this capacity can be safely used again in the next RT. 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.
Transient Path:
Path capacity information is transient. A sender that fails to use capacity has no understanding whether that capacity remains available to use - or whether it has disappeared (e.g., to a change to a path with a smaller bottleneck, or more traffic has emerged that has consumed the previously available capacity). For this reason, a sender that is limited by the volume of application data available to send MUST NOT continue to grow the congestion window [RFC5681].
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 TCP sender has not sent data in an interval that exceeds the current retransmission timeout, i.e., when an application becomes idle [RFC5681]. Experimental specifications permit TCP senders to tentatively maintain a congestion window when application-limited, provided that they appropriately and rapidly collapse the window when potential congestion is detected [RFC7661]. This mechanism is called Congestion Window Validation (CWV).
Burst Mitigation:
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), or patterns of loss that result in denying a reasonable access to the available capacity (sometimes called flow starvation).
While a congestion controller ought to limit sending at the granularity of the current RTT, this can be insufficient to satisfy the goals of preventing starvation and 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 [RFC5681] can help mitigate bursts for protocols that receive continuous feedback of reception (such as TCP). Sender pacing can mitigate this [RFC8085], (See 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.

4.3. Timers and Retransmission

This section describes mechanisms to detect and provide retransmission, and to protect the network in the absence of timely feedback.

Loss Detection:
Loss detection occurs after a sender determines there is no delivery confirmation within an expected period of time. Loss detection can be performed observing the time-ordering of the reception of ACKs (as in TCP DupACK) or can utilise a timer to detect loss before the expiry of the RTO [RFC8085] [ID.ietf-tcpm-rack] or a combination of using a timer and ordering information to trigger retransmission of data.
Retransmission of lost packets or messages is a common reliability mechanism. When a loss is detected, the sender can choose to retransmit the lost data, ignore the loss, or send other data. Any transmission 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 responsible for congestion control of the retransmissions (and any other packets sent) as well as the original traffic.
Measuring the RTT:
Once an endpoint has started communicating with its peer, the RTT MUST adjusted by measuring the actual path RTT and its variance (see equation 2.3 of [RFC6928]).
Maintaining the RTO:
The RTO SHOULD be set based on recent RTT observations [RFC8530].
RTO Expiry:
Persistent lack of feedback detected by the RTO timer (or other means) MUST be treated an indication of potential congestion. A failure to receive any specific response within a RTO interval could potentially be 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 [RFC5681]. Other transports must similarly respond when they detect loss of feedback.
An endpoint needs to exponentially backoff the RTO interval [RFC8085] each time the RTO expires. That is the RTO interval MUST be set to the RTO * 2[RFC6298] [RFC8085].
Maximum RTO:
A maximum value MAY be placed on the RTO interval. The maximum limit to the RTO interval MUST NOT be less than 60 seconds [RFC6298].

4.4. Responding to Potential Congestion

Internet flows SHOULD implement appropriate safeguards to avoid inappropriate impact on other flows that share the resources along a path. The safety and responsiveness of new proposals need to be evaluated [RFC5166]. In determining an appropriate congestion response, designs could take into consideration the size of the packets that experience congestion [RFC4828].

Congestion Response:
An endpoint MUST reduce the rate of transmission when it detects loss (or some other indicator of congestion) [RFC2914]. Prompt reaction should follow a signal from the remote endpoint indicating congestion (or inference of that, e.g., through detecting packet loss).
TCP Reno established a method that relies on multiplicative-decrease to halve the sending rate while congestion is detected. This response to loss is considered sufficient for safe Internet operation, but other decrease factors have also been published in the RFC Series [RFC8312].
ECN Response:
A congestion control design should provide the necessary mechanisms to support Explicit Congestion Notification (ECN) [RFC3168] [RFC6679], as described in section 3.1.7. of [RFC8085]. This can provide help determine an appropriate congestion window when supported by routers on the path [RFC7567] to enable rapid early indication of incipient congestion.
The early detection of incipient congestion justifies a different reaction to the reaction to packet loss [RFC8311] [RFC8087]]. Simple feedback of received Congestion Experienced (CE) marks [RFC3168], relies only on an indication that congestion has been experienced within the last RT, appropriate for using ECT(0). The reaction to reception of this indication was modified in TCP ABE [RFC8511]. Further detail about the received CE-marking can be obtained by using more accurate receiver feedback [ID.-ietf-tcpm-accurate-ecn]. This more detailed feedback provides an opportunity for a finer-granularity of congestion response.
Current work-in-progress [ID.ietf-tsvwg-l4s-arch] defines a reaction for packets marked with ECT(1), building on the style of detailed feedback provided by [ID.-ietf-tcpm-accurate-ecn] and a modified marking system [ID.ietf-tsvwg-aqm-dualq-coupled].
Robustness to Path Change:
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 under persistent congestion.
The only way to surely confirm that a sending endpoint has successfully communicated with a remote endpoint is to utilise a timer (see) to detect a lack of response that could result from a change in the path or the path characteristics (usually called the RTO). Congestion controllers that are unable to react after one (or at most a few) RTTs after receiving a congestion indication should observe the guidance in section 3.3 of the UDP Guidelines [RFC8085].
Persistent Congestion:
Persistent congestion can result in congestion collapse. This MUST be aggressively avoided [RFC2914]. Endpoints that experience persistent congestion and have already exponentially reduced their congestion window to the restart window (e.g., 1 packet), MUST further reduce the rate if the RTO timer continues to expire. For example, TCP-Friendly Rate Control (TFRC) [RFC5348] continues to reduce its sending rate under persistent congestion to one packet per RT, and then exponentially backs off the time between single packet transmissions if the congestion continues to persist [RFC2914].
[RFC8085] provides guidelines for a sender that does not, or is unable to, adapt the congestion window.

4.5. Using More Capacity

In the absence of persistent congestion, an endpoint is permitted to increase its congestion window and hence the sending rate. This 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 RT).

TCP Reno [RFC5681] defines an algorithm, known as the AIMD (additive-increase/ multiplicative-decrease) that 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 RTTs to determine a suitable share of the path 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 [RFC8312]).

Increasing Congestion Window:
A sender MUST NOT continue to increase its rate for more than an RTT after a congestion indication is received. It SHOULD stop increasing its congestion window as soon as it receives indication of congestion to avoid excessive "overshoot".
While the sender is increasing the congestion window, a sender can transmit faster than the last known safe rate. Any increase above the last confirmed rate needs to be regarded as tentative and the sender reduce their rate below the last confirmed safe rate when congestion is experienced (a congestion event).
An endpoint MUST utilise a method that assures the sender will keep the rate below the previously confirmed safe rate for multiple RTTs 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.
Avoiding Overshoot:
Overshoot of the congestion window beyond the point of congestion can significantly impact other flows sharing resources along a path. It is important to note that as endpoints experience more paths with a large BDP and a wider range of potential path RT, that variability or changes in the path can have very significant impacts on appropriate dynamics for increasing the congestion window (see also burst mitigation Section 4.2).

4.6. Network Signals

An endpoint can utilise signals from the network to help determine how to regulate the traffic it sends.

Network Signals:
Mechanisms MUST NOT solely rely on messages or other specific signalling messages to perform safely (See section 5.2 of [RFC8085] describing use of ICMP messages). The path characteristics can change at any time. Transport mechanisms need to be robust to potential black-holing of any signals (i.e., it needs to be robust to loss or modification of packets).
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 the use of soft-state when designing protocols that interact with signals originating from network devices (e.g., ECN). This can include context-sensitive treatment of "soft" signals provided to the endpoint [RFC5461].

4.7. Protection of Protocol Mechanisms

An endpoint needs to provide protection from attacks on the traffic it generates, or atatcks that increase the capacity it consumes (impacting other traffic that shared a bottleneck).

Off Path Attack:
A design MUST protect from off-path attack to the protocol [RFC8085]. An attack on the congestion control can lead to a DoS vulnerability for the flow being controlled and/or other flows that share network resources along the path.
Validation of Signals:
Network signalling and control messages (e.g., ICMP [RFC0792]) MUST to be validated before they are used to protect from malicious use. This MUST at least include protection from off-path attack [RFC8085].
On Path Attack:
A protocol can be designed to protect from on-path attacks, but this requires more complexity and the use of encryption/authentication mechanisms (e.g., IPsec [RFC4301], QUIC [I-D.ietf-quic-transport]).

5. IETF Guidelines on Evaluation of Congestion Control

The IETF has provided guidance [RFC5033] for considering 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].

6. Acknowledgements

Nicholas Kuhn helped develop the first draft of these guidelines. Tom Jones reviewed the first version of this draft. Gorry Fairhurst and Tom Jones were funded at the University of Aberdeen by the European Space Agency. Ana Custura helped review the text.

The views expressed are solely those of the author(s).

7. 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.

8. Security Considerations

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 Section 4.6 supports current best practice to validate ICMP messages prior to use. Section Section 4.7 describes general requirements relating to the design of safe protocols and their protection from on and off path attack.

9. References

9.1. Normative References

[RFC1122] Braden, R., "Requirements for Internet Hosts - Communication Layers", STD 3, RFC 1122, DOI 10.17487/RFC1122, October 1989.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914, DOI 10.17487/RFC2914, September 2000.
[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.
[RFC3742] Floyd, S., "Limited Slow-Start for TCP with Large Congestion Windows", RFC 3742, DOI 10.17487/RFC3742, March 2004.
[RFC5681] Allman, M., Paxson, V. and E. Blanton, "TCP Congestion Control", RFC 5681, DOI 10.17487/RFC5681, September 2009.
[RFC6298] Paxson, V., Allman, M., Chu, J. and M. Sargent, "Computing TCP's Retransmission Timer", RFC 6298, DOI 10.17487/RFC6298, June 2011.
[RFC6928] Chu, J., Dukkipati, N., Cheng, Y. and M. Mathis, "Increasing TCP's Initial Window", RFC 6928, DOI 10.17487/RFC6928, April 2013.
[RFC7567] Baker, F. and G. Fairhurst, "IETF Recommendations Regarding Active Queue Management", BCP 197, RFC 7567, DOI 10.17487/RFC7567, July 2015.
[RFC7661] Fairhurst, G., Sathiaseelan, A. and R. Secchi, "Updating TCP to Support Rate-Limited Traffic", RFC 7661, DOI 10.17487/RFC7661, October 2015.
[RFC8085] Eggert, L., Fairhurst, G. and G. Shepherd, "UDP Usage Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, March 2017.

9.2. Informative References

[I-D.ietf-quic-transport] Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport-20, April 2019.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI 10.17487/RFC0768, August 1980.
[RFC0792] Postel, J., "Internet Control Message Protocol", STD 5, RFC 792, DOI 10.17487/RFC0792, September 1981.
[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.
[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.
[RFC3819] Karn, P., 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.
[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E. and G. Fairhurst, "The Lightweight User Datagram Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July 2004.
[RFC4340] Kohler, E., Handley, M. and S. Floyd, "Datagram Congestion Control Protocol (DCCP)", RFC 4340, DOI 10.17487/RFC4340, March 2006.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control (TFRC): The Small-Packet (SP) Variant", RFC 4828, DOI 10.17487/RFC4828, April 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 4960, DOI 10.17487/RFC4960, September 2007.
[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion Control Algorithms", BCP 133, RFC 5033, DOI 10.17487/RFC5033, August 2007.
[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.
[RFC8087] Fairhurst, G. and M. Welzl, "The Benefits of Using Explicit Congestion Notification (ECN)", RFC 8087, DOI 10.17487/RFC8087, March 2017.
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018.
[RFC8511] Khademi, N., Welzl, M., Armitage, G. and G. Fairhurst, "TCP Alternative Backoff with ECN (ABE)", RFC 8511, DOI 10.17487/RFC8511, December 2018.

Appendix A. Revision Notes

Note to RFC-Editor: please remove this entire section prior to publication.

Individual draft -00:

Individual draft -01:

Author's Address

Godred Fairhurst University of Aberdeen School of Engineering Fraser Noble Building Aberdeen, AB24 3U UK EMail: