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Internet Engineering Task Force                             G. Fairhurst
Internet-Draft                                    University of Aberdeen
Intended status: Standards Track                        November 3, 2019
Expires: May 6, 2020


        Guidelines for Internet Congestion Control at Endpoints
                      draft-fairhurst-tsvwg-cc-04

Abstract

   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.  This therefore seeks to gather and
   consolidate these recommendations and provide overall guidance.  It
   is intended to provide input to the design of new congestion control
   methods in protocols, such as the IETF Quick UDP Internet Connections
   (QUIC) transport.

   The present document is for discussion and comment by the IETF.  If
   published, it plans to 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 May 6, 2020.

Copyright Notice

   Copyright (c) 2019 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 Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Best Current Practice in the RFC-Series . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   5
   3.  Principles of Congestion Control  . . . . . . . . . . . . . .   5
     3.1.  A Diversity of Path Characteristics . . . . . . . . . . .   5
     3.2.  Flow Multiplexing and Congestion  . . . . . . . . . . . .   6
     3.3.  Avoiding Congestion Collapse and Flow Starvation  . . . .   9
   4.  Guidelines for Performing Congestion Control  . . . . . . . .  10
     4.1.  Connection Initialization . . . . . . . . . . . . . . . .  10
     4.2.  Using Path Capacity . . . . . . . . . . . . . . . . . . .  12
     4.3.  Timers and Retransmission . . . . . . . . . . . . . . . .  13
     4.4.  Responding to Potential Congestion  . . . . . . . . . . .  15
     4.5.  Using More Capacity . . . . . . . . . . . . . . . . . . .  16
     4.6.  Network Signals . . . . . . . . . . . . . . . . . . . . .  17
     4.7.  Protection of Protocol Mechanisms . . . . . . . . . . . .  18
   5.  IETF Guidelines on Evaluation of Congestion Control . . . . .  18
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  18
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  19
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  19
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  19
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  19
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  20
   Appendix A.  Revision Notes . . . . . . . . . . . . . . . . . . .  25
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  26

1.  Introduction

   The IETF has specified Internet transports (e.g., TCP
   [I-D.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 [RFC3550], QUIC
   [I-D.ietf-quic-transport], 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,



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   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.

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

1.1.  Best Current Practice in the RFC-Series

   Like RFC2119, this documents borrows heavily from earlier
   publications addressing the need for end-to-end congestion control,
   and this subsection provides an overview of key topics.

   [RFC2914] provides a general discussion of the principles of
   congestion control.  Section 3 discussed 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.

   Congestion collapse was first reported in the mid 1980s [RFC0896],
   and at that time was largely due to TCP connections unnecessarily
   retransmitting packets that were either in transit or had already
   been received at the receiver.  We call the congestion collapse that
   results from the unnecessary retransmission of packets classical
   congestion collapse.  Classical congestion collapse is a stable
   condition that can result in throughput that is a small fraction of
   normal [RFC0896].  Problems with classical congestion collapse have
   generally been corrected by improvements to timer and congestion
   control mechanisms, implemented in modern implementations of TCP
   [Jacobson88].  This classical congestion collapse was a key focus of
   [RFC2309].

   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.  This is probably the largest
   unresolved danger with respect to congestion collapse in the Internet



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   today.  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 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))."

   Section 3.3 of [RFC2914] notes: "In addition to the prevention of
   congestion collapse and concerns about fairness, a third reason for a
   flow to use end-to-end congestion control can be 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."

   In addition to the prevention of congestion collapse and concerns
   about fairness, a flow using end-to-end congestion control can
   optimize its own performance regarding throughput, delay, and loss
   [RFC2914].

   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 use of standard methods is therefore
   encouraged.

   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.  Some transport implementations
   fail to use standardised congestion avoidance mechanisms correctly
   because of poor implementation [RFC2525].  However, this is not the
   only reason fro 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



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   transports that have not presently been submitted for
   standardisation.

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" or
   source 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 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.

   [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.

   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 forSection 4.

3.1.  A Diversity of Path Characteristics

   Internet transports can reserve capacity at routers or on the links
   being used, but most uses do not rely upon prior reservation of
   capacity along the path 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.  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 that does not support



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   Active Queue Management (AQM) [RFC7567] typically uses a drop-tail
   policy to drop excess IP packets when its queue(s) becomes full.

   When a transport uses a path to send packets (i.e. a flow), this
   impacts any 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.  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).

   When choosing an appropriate sending rate, packet loss needs to be
   considered.  Although losses are not always due to congestion,
   endpoint congestion control needs to conservatively react to loss as
   a potential signal of reduced available capacity and reduce the
   sending rate.  Many designs place the responsibility of rate-adaption
   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]).

   Principles include:

   o  A transport design is REQUIRED be robust to a change in the set of
      devices forming the network path.  A reconfiguration, reset or
      other event could interrupt this path or trigger a change in the
      set of network devices forming the path.

   o  Transports are REQUIRED to operate safely over the wide range of
      path characteristics presented by Internet paths.

   o  The path characteristics can change over relatively short
      intervals of time (i.e., characteristics discovered do not
      necessarily remain valid for multiple Round Trip Times, RTTs).  In
      particular, the transport SHOULD measure and adapt to the
      characteristics of the path(s) being used.

3.2.  Flow Multiplexing and Congestion

   It is 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 other flows
   that share the resources along a path.  Congestion control methods
   satisfy this requirement and therefore can help avoid congestion
   collapse.



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   "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]

   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 traffic [RFC2914].  Additional mechanisms are, in
   some cases, needed in the upper layer protocol for an application
   that sends datagrams (e.g., using UDP) [RFC8085].

   Endpoints 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."  [RFC2140].  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."

   An endpoint can become aware of congestion by various means
   (including packet loss Section 3.1).  A signal that indicates
   congestion on the end-to-end network path, needs to result in a
   congestion control reaction by the transport to reduce the maximum
   rate permitted by the sending endpoint[RFC8087]).

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



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   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.

   Network devices can be configured 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.  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.

   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:

   o  Endpoints MUST perform congestion control [RFC1122] and SHOULD
      leverage existing congestion control techniques [RFC8085].

   o  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].

   o  Transports SHOULD control the aggregate traffic they send on a
      path [RFC8085].  They ought not to use multiple congestion-
      controlled flows between the same endpoints to gain a performance
      advantage.

   o  Transports that do not target Internet deployment need to be
      constrained to only operate in a controlled environment (e.g., see
      Section 3.6 of [RFC8085]) and provide appropriate mechanisms to
      prevent traffic accidentally leaving the controlled environment
      [RFC8084].

   o  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 constrained to operate in a controlled environment.






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3.3.  Avoiding Congestion Collapse and Flow Starvation

   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 [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."

   Transports need to be specifically designed with measures to avoid
   starving other flows of capacity (e.g., [RFC7567]).  [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 this, encouraging development of methods to prevent
   starvation.

   Principles include:

   o  Transports MUST avoid inducing flow starvation to other flows that
      share resources along the path they use.

   o  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).

   o  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 (Section 4.3).

   o  Protocol timers (e.g., used for retransmission or to detect
      persistent congestion) need to be appropriately initialised.  A
      transport SHOULD adapt its protocol timers to follow the measured
      the path Round Trip Rime (RTT) (e.g., Section 3.1.1 of [RFC8085]).

   o  A transport MUST employ exponential backoff each time persistent
      congestion is detected [RFC1122], until the path characteristics
      can again be confirmed.

   o  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 SHOULD
      mitigate the impact of non-conformant and malicious flows
      [RFC7567]).  These mechanisms complement, but do not replace, the
      endpoint congestion avoidance mechanisms.



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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.

   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 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.

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 they will use.  This section describes how a flow starts
   transmission over such a path.

   Flow Start:  A new flow between two endpoints needs to initialise a
      congestion controller for the path it will use.  It 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 TCP 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, 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 (e.g., determined by the IW) 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 will
      use.  An initial value needs to be used to initialise the
      principal retransmission timer, which 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



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      knowledge about the latency of a path (including the initial
      value), 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]).  (Note: Linux TCP has deployed a smaller
      initial RTO value).

      [[Author note: It could be useful to discuss cached values]].

   Initial RTO Expiry:  If the RTO timer expires while awaiting
      completion of a connection setup, or handshake (e.g., the three-
      way handshake 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.  [[Author note: It would
      be useful to discuss how the timer is managed to protect from
      multiple handshake failure]].

      The 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]).

   Current State:  A congestion controller MAY assume that recently used
      capacity between a pair of endpoints is an indication of future
      capacity available in the next RTT between the same endpoints.  It
      MUST react (reduce its rate) if this is not (later) confirmed to
      be true.  [[Author note: do we need to bound this]].

   Cached State:  A congestion controller that recently used a specific
      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) which it understands is using the same path and no
      will longer use the capacity it recently used.  In TCP, this
      mechanism is referred to as TCP Control Block (TCB) sharing
      [RFC2140] [I-D.ietf-tcpm-2140bis].  The capacity and other
      information can be used to suggest a faster initial sending rate,
      but this information 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



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      offered to it).  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 RTT, 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.  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.

   Transient Path:  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



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      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.

   Validating the 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
      [RFC5681].  An experimental specification [RFC7661] permits TCP
      senders to tentatively maintain a congestion window larger than
      the path supported in the last RTT when application-limited,
      provided that they appropriately and rapidly collapse the
      congestion window when potential congestion is detected.  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).

      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



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      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] [I-D.ietf-tcpm-rack]) or a combination of using a timer
      and ordering information to trigger retransmission of data.

   Retransmission:  Retransmission of lost packets or messages is a
      common reliability mechanism.  When loss is detected, the sender
      can choose to retransmit the lost data, ignore the loss, or send
      other data (e.g., [RFC8085] [I-D.ietf-quic-recovery]), depending
      on the reliability model provided by the transport service.  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 therefore responsible for
      congestion control of the retransmissions (and any other packets
      sent, including FEC information) as well as the original traffic.

   Measuring the RTT:  Once an endpoint has started communicating with
      its peer, the RTT be MUST adjusted by measuring the actual path
      RTT.  This adjustment MUST include adapting to the measured RTT
      variance (see equation 2.3 of [RFC6928]).

   Maintaining the RTO:  The RTO SHOULD be set based on recent RTT
      observations (including the RTT variance) [RFC8085].

   RTO Expiry:  Persistent lack of feedback (e.g., detected by an RTO
      timer, or other means) MUST be treated an indication of potential
      congestion collapse.  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 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: These recommendations should be re-evaluated in
      lite of the current chartered work in the TCPM WG.  ]]





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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 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 [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 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 an explicit congestion signal compared to the reaction
      to detected 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 RTT.  This style of response is appropriate when a flow uses
      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
      (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.

      Current work-in-progress [I-D.ietf-tsvwg-l4s-arch]defines a
      reaction for packets marked with ECT(1), building on the style of
      detailed feedback provided by [I-D.ietf-tcpm-accurate-ecn] and a
      modified marking system [I-D.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.



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      The only way to surely confirm that a sending endpoint has
      successfully communicated with a remote endpoint is to utilise a
      timer (seeSection 4.3) 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, 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 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.  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).

   TCP Reno [RFC5681] 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
   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.  The transport SHOULD stop increasing its congestion
      window as soon as it receives indication of congestion to avoid
      excessive "overshoot".



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      While the sender is increasing the congestion window, a sender
      will 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).

   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.

   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 RTT, that variability or changes in the path can
      have very significant constraints 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 transport
      messages or specific signalling messages to perform safely.  (See
      section 5.2 of [RFC8085] describing use of ICMP messages).  They
      need to be designed so that they safely operate when path
      characteristics change at any time.  Transport mechanisms MUST
      robust to potential black-holing of any signals (i.e., need to be
      robust to loss or modification of packets, noting that this can
      occur even after successful first use of a signal by a flow, as
      occurs when the path changes, see Section 3.1).

      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 [I-D.irtf-panrg-what-not-to-do]
      (e.g., ECN).  This can include context-sensitive treatment of
      "soft" signals provided to the endpoint [RFC5164].







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4.7.  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 it
   consumes (impacting other traffic that shared a bottleneck).

   Off Path Attack:   A design MUST protect from off-path attack to the
      protocol [RFC8085] (i.e., one by an attacker that is unable to see
      the contents of packets exchanged across the path).  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.

   Validation of Signals:  Network signalling and control messages
      (e.g., ICMP [RFC0792]) MUST be validated before they are used to
      protect from malicious abuse.  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 also 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.

6.  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.
   The University of Aberdeen received funding to support this work from
   the European Space Agency.








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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

   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 4.7 describes general requirements relating to the design of
   safe protocols and their protection from on and off path attack.

   Section 4.6 follows current best practice to validate ICMP messages
   prior to use.

9.  References

9.1.  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>.

   [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>.



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   [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>.

   [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>.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
              <https://www.rfc-editor.org/info/rfc5681>.

   [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>.

   [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>.

   [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>.

   [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>.

   [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>.

9.2.  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-quic-recovery]
              Iyengar, J. and I. Swett, "QUIC Loss Detection and
              Congestion Control", draft-ietf-quic-recovery-23 (work in
              progress), September 2019.



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   [I-D.ietf-quic-transport]
              Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
              and Secure Transport", draft-ietf-quic-transport-23 (work
              in progress), September 2019.

   [I-D.ietf-tcpm-2140bis]
              Touch, J., Welzl, M., and S. Islam, "TCP Control Block
              Interdependence", draft-ietf-tcpm-2140bis-00 (work in
              progress), April 2019.

   [I-D.ietf-tcpm-accurate-ecn]
              Briscoe, B., Kuehlewind, M., and R. Scheffenegger, "More
              Accurate ECN Feedback in TCP", draft-ietf-tcpm-accurate-
              ecn-09 (work in progress), July 2019.

   [I-D.ietf-tcpm-rack]
              Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "RACK:
              a time-based fast loss detection algorithm for TCP",
              draft-ietf-tcpm-rack-06 (work in progress), November 2019.

   [I-D.ietf-tcpm-rfc793bis]
              Eddy, W., "Transmission Control Protocol Specification",
              draft-ietf-tcpm-rfc793bis-14 (work in progress), July
              2019.

   [I-D.ietf-tsvwg-aqm-dualq-coupled]
              Schepper, K., Briscoe, B., and G. White, "DualQ Coupled
              AQMs for Low Latency, Low Loss and Scalable Throughput
              (L4S)", draft-ietf-tsvwg-aqm-dualq-coupled-10 (work in
              progress), July 2019.

   [I-D.ietf-tsvwg-l4s-arch]
              Briscoe, B., Schepper, K., Bagnulo, M., and G. White, "Low
              Latency, Low Loss, Scalable Throughput (L4S) Internet
              Service: Architecture", draft-ietf-tsvwg-l4s-arch-04 (work
              in progress), July 2019.

   [I-D.irtf-panrg-what-not-to-do]
              Dawkins, S., "Path Aware Networking: Obstacles to
              Deployment (A Bestiary of Roads Not Taken)", draft-irtf-
              panrg-what-not-to-do-03 (work in progress), May 2019.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              DOI 10.17487/RFC0768, August 1980,
              <https://www.rfc-editor.org/info/rfc768>.






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   [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>.

   [RFC2140]  Touch, J., "TCP Control Block Interdependence", RFC 2140,
              DOI 10.17487/RFC2140, April 1997,
              <https://www.rfc-editor.org/info/rfc2140>.

   [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>.

   [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>.






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   [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-A., Degermark, M., Pink, S., Jonsson, L-E., 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>.

   [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>.






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   [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>.

   [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>.

   [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>.

   [RFC8312]  Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
              R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
              RFC 8312, DOI 10.17487/RFC8312, February 2018,
              <https://www.rfc-editor.org/info/rfc8312>.

   [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>.







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Appendix A.  Revision Notes

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

   Individual draft -00:

   o  Comments and corrections are welcome directly to the authors or
      via the IETF TSVWG, working group mailing list.

   IndivRFC896 idual draft -01:

   o  This update is proposed for initial WG comments.

   o  If there is interest in progressing this document, the next
      version will include more complee referencing to citred material.

   Individual draft -02:

   o  Correction of typos.

   Individual draft -03:

   o  Added section 1.1 with text on current BCP status with additional
      alignment and updates to RFC2914 on Congestion Control Principles
      (after question from M.  Scharf).

   o  Edits to consolidate starvation text.

   o  Added text that multicast currently noting that this is out of
      scope.

   o  Revised sender-based CC text after comment from C.  Perkins
      (Section 3.1,3.3 and other places).

   o  Added more about FEC after comment from C.  Perkins.

   o  Added an explicit reference to RFC 5783 and updated this text
      (after question from M.  Scharf).

   o  To avoid doubt, added a para about "Each new transport needs to
      make its own design decisions about how to meet the
      recommendations and requirements for congestion control."

   o  Upated references.

   Individual draft -04:




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   o  Correction of NiTs.  Further clarifications.

   o  This draft does not attempt to address further alignment with
      draft-ietf-tcpm-rto-consider.  This will form part of a future
      revision.

Author's Address

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

   Email: gorry@erg.abdn.ac.uk



































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