Transport Services (tsv) K. De Schepper
Internet-Draft Nokia Bell Labs
Intended status: Experimental B. Briscoe, Ed.
Expires: August 23, 2020 Independent
February 20, 2020

Identifying Modified Explicit Congestion Notification (ECN) Semantics for Ultra-Low Queuing Delay (L4S)


This specification defines the identifier to be used on IP packets for a new network service called low latency, low loss and scalable throughput (L4S). It is similar to the original (or 'Classic') Explicit Congestion Notification (ECN). 'Classic' ECN marking was required to be equivalent to a drop, both when applied in the network and when responded to by a transport. Unlike 'Classic' ECN marking, for packets carrying the L4S identifier, the network applies marking more immediately and more aggressively than drop, and the transport response to each mark is reduced and smoothed relative to that for drop. The two changes counterbalance each other so that the throughput of an L4S flow will be roughly the same as a non-L4S flow under the same conditions. Nonetheless, the much more frequent control signals and the finer responses to them result in ultra-low queuing delay (sub-millisecond) for L4S traffic without compromising link utilization, and this low delay can be maintained during high traffic load.

The L4S identifier defined in this document is the key piece that distinguishes L4S from TCP-Reno-Friendly (or 'Classic') traffic. It gives an incremental migration path so that suitably modified network bottlenecks can distinguish and isolate existing Classic traffic from L4S traffic to prevent it from degrading the ultra-low queuing delay and loss of the new scalable transports, without harming Classic performance. Examples of new active queue management (AQM) marking algorithms and examples of new transports (whether TCP-like or real-time) are specified separately.

Status of This Memo

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Table of Contents

1. Introduction

This specification defines the identifier to be used on IP packets for a new network service called low latency, low loss and scalable throughput (L4S). It is similar to the original (or 'Classic') Explicit Congestion Notification (ECN [RFC3168]). RFC 3168 required an ECN mark to be equivalent to a drop, both when applied in the network and when responded to by a transport. Unlike Classic ECN marking, the network applies L4S marking more immediately and more aggressively than drop, and the transport response to each mark is reduced and smoothed relative to that for drop. The two changes counterbalance each other so that the throughput of an L4S flow will be roughly the same as a non-L4S flow under the same conditions. Nonetheless, the much more frequent control signals and the finer responses to them result in ultra-low queuing delay without compromising link utilization, and this low delay can be maintained during high load. Ultra-low queuing delay means less than 1 millisecond (ms) on average and less than about 2 ms at the 99th percentile.

An example of a scalable congestion control that would enable the L4S service is Data Center TCP (DCTCP), which until now has been applicable solely to controlled environments like data centres [RFC8257], because it is too aggressive to co-exist with existing TCP-Reno-Friendly traffic. The DualQ Coupled AQM, which is defined in a complementary experimental specification [I-D.ietf-tsvwg-aqm-dualq-coupled], is an AQM framework that enables scalable congestion controls like DCTCP to co-exist with existing traffic, each getting roughly the same flow rate when they compete under similar conditions. Note that a transport such as DCTCP is still not safe to deploy on the Internet unless it satisfies the requirements listed in Section 4.

L4S is not only for elastic (TCP-like) traffic - there are scalable congestion controls for real-time media, such as the L4S variant of the SCReAM [RFC8298] real-time media congestion avoidance technique (RMCAT). The factor that distinguishes L4S from Classic traffic is its behaviour in response to congestion. The transport wire protocol, e.g. TCP, QUIC, SCTP, DCCP, RTP/RTCP, is orthogonal (and therefore not suitable for distinguishing L4S from Classic packets).

The L4S identifier defined in this document is the key piece that distinguishes L4S from TCP-Reno-Friendly (or 'Classic') traffic. It gives an incremental migration path so that suitably modified network bottlenecks can distinguish and isolate existing Classic traffic from L4S traffic to prevent it from degrading the ultra-low delay and loss of the new scalable transports, without harming Classic performance. Initial implementation of the separate parts of the system has been motivated by the performance benefits.

1.1. Latency, Loss and Scaling Problems

Latency is becoming the critical performance factor for many (most?) applications on the public Internet, e.g. interactive Web, Web services, voice, conversational video, interactive video, interactive remote presence, instant messaging, online gaming, remote desktop, cloud-based applications, and video-assisted remote control of machinery and industrial processes. In the 'developed' world, further increases in access network bit-rate offer diminishing returns, whereas latency is still a multi-faceted problem. In the last decade or so, much has been done to reduce propagation time by placing caches or servers closer to users. However, queuing remains a major intermittent component of latency.

The Diffserv architecture provides Expedited Forwarding [RFC3246], so that low latency traffic can jump the queue of other traffic. However, on access links dedicated to individual sites (homes, small enterprises or mobile devices), often all traffic at any one time will be latency-sensitive. Then, given nothing to differentiate from, Diffserv makes no difference. Instead, we need to remove the causes of any unnecessary delay.

The bufferbloat project has shown that excessively-large buffering ('bufferbloat') has been introducing significantly more delay than the underlying propagation time. These delays appear only intermittently—only when a capacity-seeking (e.g. TCP) flow is long enough for the queue to fill the buffer, making every packet in other flows sharing the buffer sit through the queue.

Active queue management (AQM) was originally developed to solve this problem (and others). Unlike Diffserv, which gives low latency to some traffic at the expense of others, AQM controls latency for all traffic in a class. In general, AQM methods introduce an increasing level of discard from the buffer the longer the queue persists above a shallow threshold. This gives sufficient signals to capacity-seeking (aka. greedy) flows to keep the buffer empty for its intended purpose: absorbing bursts. However, RED [RFC2309] and other algorithms from the 1990s were sensitive to their configuration and hard to set correctly. So, this form of AQM was not widely deployed.

More recent state-of-the-art AQM methods, e.g. fq_CoDel [RFC8290], PIE [RFC8033], Adaptive RED [ARED01], are easier to configure, because they define the queuing threshold in time not bytes, so it is invariant for different link rates. However, no matter how good the AQM, the sawtoothing sending window of a Classic congestion control will either cause queuing delay to vary or cause the link to be under-utilized. Even with a perfectly tuned AQM, the additional queuing delay will be of the same order as the underlying speed-of-light delay across the network.

If a sender's own behaviour is introducing queuing delay variation, no AQM in the network can "un-vary" the delay without significantly compromising link utilization. Even flow-queuing (e.g. [RFC8290]), which isolates one flow from another, cannot isolate a flow from the delay variations it inflicts on itself. Therefore those applications that need to seek out high bandwidth but also need low latency will have to migrate to scalable congestion control.

Altering host behaviour is not enough on its own though. Even if hosts adopt low latency behaviour (scalable congestion controls), they need to be isolated from the behaviour of existing Classic congestion controls that induce large queue variations. L4S enables that migration by providing latency isolation in the network and distinguishing the two types of packets that need to be isolated: L4S and Classic. L4S isolation can be achieved with a queue per flow (e.g. [RFC8290]) but a DualQ [I-D.ietf-tsvwg-aqm-dualq-coupled] is sufficient, and actually gives better tail latency. Both approaches are addressed in this document.

The DualQ solution was developed to make ultra-low latency available without requiring per-flow queues at every bottleneck. This was because FQ has well-known downsides - not least the need to inspect transport layer headers in the network, which makes it incompatible with privacy approaches such as IPSec VPN tunnels, and incompatible with link layer queue management, where transport layer headers can be hidden, e.g. 5G.

Latency is not the only concern addressed by L4S: It was known when TCP congestion avoidance was first developed that it would not scale to high bandwidth-delay products (footnote 6 of Jacobson and Karels [TCP-CA]). Given regular broadband bit-rates over WAN distances are already [RFC3649] beyond the scaling range of Reno TCP, 'less unscalable' Cubic [RFC8312] and Compound [I-D.sridharan-tcpm-ctcp] variants of TCP have been successfully deployed. However, these are now approaching their scaling limits. Unfortunately, fully scalable congestion controls such as DCTCP [RFC8257] cause Classic ECN congestion controls sharing the same queue to starve themselves, which is why they have been confined to private data centres or research testbeds (until now).

It turns out that a congestion control algorithm like DCTCP that solves the latency problem also solves the scalability problem of Classic congestion controls. The finer sawteeth in the congestion window have low amplitude, so they cause very little queuing delay variation and the average time to recover from one congestion signal to the next (the average duration of each sawtooth) remains invariant, which maintains constant tight control as flow-rate scales. A background paper [DCttH15] gives the full explanation of why the design solves both the latency and the scaling problems, both in plain English and in more precise mathematical form. The explanation is summarised without the maths in the L4S architecture document [I-D.ietf-tsvwg-l4s-arch].

1.2. Terminology

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119]. In this document, these words will appear with that interpretation only when in ALL CAPS. Lower case uses of these words are not to be interpreted as carrying RFC-2119 significance.

Classic Congestion Control:
A congestion control behaviour that can co-exist with standard TCP Reno [RFC5681] without causing flow rate starvation. With Classic congestion controls, as flow rate scales, the number of round trips between congestion signals (losses or ECN marks) rises with the flow rate. So it takes longer and longer to recover after each congestion event. Therefore control of queuing and utilization becomes very slack, and the slightest disturbance prevents a high rate from being attained [RFC3649].

For instance, with 1500 byte packets and an end-to-end round trip time (RTT) of 36 ms, over the years, as Reno flow rate scales from 2 to 100 Mb/s the number of round trips taken to recover from a congestion event rises proportionately, from 4 to 200. Cubic was developed to be less unscalable, but it is approaching its scaling limit; with the same RTT of 36ms, at 100Mb/s it takes over 300 round trips to recover, and at 800 Mb/s its recovery time doubles to over 600 round trips, or more than 20 seconds.
Scalable Congestion Control:
A congestion control where the average time from one congestion signal to the next (the recovery time) remains invariant as the flow rate scales, all other factors being equal. This maintains the same degree of control over queueing and utilization whatever the flow rate, as well as ensuring that high throughput is robust to disturbances. For instance, DCTCP averages 2 congestion signals per round-trip whatever the flow rate. See Section 4.3 for more explanation.
Classic service:
The Classic service is intended for all the congestion control behaviours that co-exist with Reno [RFC5681] (e.g. Reno itself, Cubic [RFC8312], Compound [I-D.sridharan-tcpm-ctcp], TFRC [RFC5348]).
Low-Latency, Low-Loss Scalable throughput (L4S) service:
The 'L4S' service is intended for traffic from scalable congestion control algorithms, such as Data Center TCP [RFC8257]. The L4S service is for more general traffic than just DCTCP—it allows the set of congestion controls with similar scaling properties to DCTCP to evolve (e.g. Relentless TCP [Mathis09], TCP Prague [LinuxPrague] and the L4S variant of SCREAM for real-time media [RFC8298]).

Both Classic and L4S services can cope with a proportion of unresponsive or less-responsive traffic as well, as long as it does not build a queue (e.g. DNS, VoIP, game sync datagrams, etc).

The terms Classic or L4S can also qualify other nouns, such 'queue', 'codepoint', 'identifier', 'classification', 'packet', 'flow'. For example: a 'Classic queue', means a queue providing the Classic service; an L4S packet means a packet with an L4S identifier sent from an L4S congestion control.
Classic ECN:
The original Explicit Congestion Notification (ECN) protocol [RFC3168], which requires ECN signals to be treated the same as drops, both when generated in the network and when responded to by the sender. The names used for the four codepoints of the 2-bit IP-ECN field are as defined in [RFC3168]: Not ECT, ECT(0), ECT(1) and CE, where ECT stands for ECN-Capable Transport and CE stands for Congestion Experienced.

1.3. Scope

The new L4S identifier defined in this specification is applicable for IPv4 and IPv6 packets (as for Classic ECN [RFC3168]). It is applicable for the unicast, multicast and anycast forwarding modes.

The L4S identifier is an orthogonal packet classification to the Differentiated Services Code Point (DSCP) [RFC2474]. Section 5.4 explains what this means in practice.

This document is intended for experimental status, so it does not update any standards track RFCs. Therefore it depends on [RFC8311], which is a standards track specification that:

This document is about identifiers that are used for interoperation between hosts and networks. So the audience is broad, covering developers of host transports and network AQMs, as well as covering how operators might wish to combine various identifiers, which would require flexibility from equipment developers.

2. Consensus Choice of L4S Packet Identifier: Requirements

This subsection briefly records the process that led to a consensus choice of L4S identifier, selected from all the alternatives in Appendix B.

The identifier for packets using the Low Latency, Low Loss, Scalable throughput (L4S) service needs to meet the following requirements:

Whether the identifier would be recoverable if the experiment failed is a factor that could be taken into account. However, this has not been made a requirement, because that would favour schemes that would be easier to fail, rather than those more likely to succeed.

It is recognised that the chosen identifier is unlikely to satisfy all these requirements, particularly given the limited space left in the IP header. Therefore a compromise will be necessary, which is why all the above requirements are expressed with the word 'SHOULD' not 'MUST'. Appendix B discusses the pros and cons of the compromises made in various competing identification schemes against the above requirements.

On the basis of this analysis, "ECT(1) and CE codepoints" is the best compromise. Therefore this scheme is defined in detail in the following sections, while Appendix B records the rationale for this decision.

3. L4S Packet Identification at Run-Time

The L4S treatment is an experimental track alternative packet marking treatment [RFC4774] to the Classic ECN treatment in [RFC3168], which has been updated by [RFC8311] to allow experiments such as the one defined in the present specification. Like Classic ECN, L4S ECN identifies both network and host behaviour: it identifies the marking treatment that network nodes are expected to apply to L4S packets, and it identifies packets that have been sent from hosts that are expected to comply with a broad type of sending behaviour.

For a packet to receive L4S treatment as it is forwarded, the sender sets the ECN field in the IP header to the ECT(1) codepoint. See Section 4 for full transport layer behaviour requirements, including feedback and congestion response.

A network node that implements the L4S service normally classifies arriving ECT(1) and CE packets for L4S treatment. See Section 5 for full network element behaviour requirements, including classification, ECN-marking and interaction of the L4S identifier with other identifiers and per-hop behaviours.

4. Prerequisite Transport Layer Behaviour

4.1. Prerequisite Codepoint Setting

A sender that wishes a packet to receive L4S treatment as it is forwarded, MUST set the ECN field in the IP header (v4 or v6) to the ECT(1) codepoint.

4.2. Prerequisite Transport Feedback

For a transport protocol to provide scalable congestion control it MUST provide feedback of the extent of CE marking on the forward path. When ECN was added to TCP [RFC3168], the feedback method reported no more than one CE mark per round trip. Some transport protocols derived from TCP mimic this behaviour while others report the accurate extent of ECN marking. This means that some transport protocols will need to be updated as a prerequisite for scalable congestion control. The position for a few well-known transport protocols is given below.

Support for the accurate ECN feedback requirements [RFC7560] (such as that provided by AccECN [I-D.ietf-tcpm-accurate-ecn]) by both ends is a prerequisite for scalable congestion control in TCP. Therefore, the presence of ECT(1) in the IP headers even in one direction of a TCP connection will imply that both ends must be supporting accurate ECN feedback. However, the converse does not apply. So even if both ends support AccECN, either of the two ends can choose not to use a scalable congestion control, whatever the other end's choice.
A suitable ECN feedback mechanism for SCTP could add a chunk to report the number of received CE marks (e.g. [I-D.stewart-tsvwg-sctpecn]), and update the ECN feedback protocol sketched out in Appendix A of the standards track specification of SCTP [RFC4960].
RTP over UDP:
A prerequisite for scalable congestion control is for both (all) ends of one media-level hop to signal ECN support [RFC6679] and use the new generic RTCP feedback format of [I-D.ietf-avtcore-cc-feedback-message]. The presence of ECT(1) implies that both (all) ends of that media-level hop support ECN. However, the converse does not apply. So each end of a media-level hop can independently choose not to use a scalable congestion control, even if both ends support ECN.
Support for sufficiently fine-grained ECN feedback is provided by the first IETF QUIC transport [I-D.ietf-quic-transport].
The ACK vector in DCCP [RFC4340] is already sufficient to report the extent of CE marking as needed by a scalable congestion control.

4.3. Prerequisite Congestion Response

As a condition for a host to send packets with the L4S identifier (ECT(1)), it SHOULD implement a congestion control behaviour that ensures that, in steady state, the average time from one ECN congestion signal to the next (the 'recovery time') does not increase as flow rate scales, all other factors being equal. This is termed a scalable congestion control. This is necessary to ensure that queue variations remain small as flow rate scales, without having to sacrifice utilization. For instance, for DCTCP, the average recovery time is always half a round trip, whatever the flow rate.

The condition 'all other factors being equal', allows the recovery time to be different for different round trip times, as long as it does not increase with flow rate for any particular RTT.

Saying that the recovery time remains roughly invariant is equivalent to saying that the number of ECN CE marks per round trip remains invariant as flow rate scales, all other factors being equal. For instance, DCTCP's average recovery time of half of 1 RTT is equivalent to 2 ECN marks per round trip. For those who understand steady-state congestion response functions, it is also equivalent to say that, the congestion window is inversely proportional to the proportion of bytes in packets marked with the CE codepoint (see section 2 of [PI2]).

As well as DCTCP, TCP Prague [LinuxPrague] and the L4S variant of SCReAM [RFC8298] are examples of scalable congestion controls.

As with all transport behaviours, a detailed specification (probably an experimental RFC) will need to be defined for each type of transport or application, including the timescale over which the proportionality is averaged, and control of burstiness. The recovery time requirement above is worded as a 'SHOULD' rather than a 'MUST' to allow reasonable flexibility when defining these specifications.

Each sender in a session can use a scalable congestion control independently of the congestion control used by the receiver(s) when they send data. Therefore there might be ECT(1) packets in one direction and ECT(0) or Not-ECT in the other.

In order to coexist safely with other Internet traffic, a scalable congestion control MUST NOT tag its packets with the ECT(1) codepoint unless it complies with the following bulleted requirements. The specification of a particular scalable congestion control MUST describe in detail how it satisfies each requirement and, for any non-mandatory requirements, it MUST justify why it does not comply:

As well as traffic controlled by a scalable congestion control, a reasonable level of smooth unresponsive traffic at a low rate relative to typical broadband capacities is likely to be acceptable (see "'Safe' Unresponsive Traffic" in Section

5. Prerequisite Network Node Behaviour

5.1. Prerequisite Classification and Re-Marking Behaviour

A network node that implements the L4S service MUST classify arriving ECT(1) packets for L4S treatment and, other than in the exceptional case referred to next, it MUST classify arriving CE packets for L4S treatment as well. CE packets might have originated as ECT(1) or ECT(0), but the above rule to classify them as if they originated as ECT(1) is the safe choice (see Appendix B.1 for rationale). The exception is where some flow-aware in-network mechanism happens to be available for distinguishing CE packets that originated as ECT(0), as described in Section 5.3, but there is no implication that such a mechanism is necessary.

An L4S AQM treatment follows similar codepoint transition rules to those in RFC 3168. Specifically, the ECT(1) codepoint MUST NOT be changed to any other codepoint than CE, and CE MUST NOT be changed to any other codepoint. An ECT(1) packet is classified as ECN-capable and, if congestion increases, an L4S AQM algorithm will increasingly mark the ECN field as CE, otherwise forwarding packets unchanged as ECT(1). Necessary conditions for an L4S marking treatment are defined in Section 5.2. Under persistent overload an L4S marking treatment SHOULD turn off ECN marking, using drop as a congestion signal until the overload episode has subsided, as recommended for all AQM methods in [RFC7567] (Section 4.2.1), which follows the similar advice in RFC 3168 (Section 7).

For backward compatibility in uncontrolled environments, a network node that implements the L4S treatment MUST also implement an AQM treatment for the Classic service as defined in Section 1.2. This Classic AQM treatment need not mark ECT(0) packets, but if it does, it will do so under the same conditions as it would drop Not-ECT packets [RFC3168]. It MUST classify arriving ECT(0) and Not-ECT packets for treatment by the Classic AQM (see the discussion of the classifier for the dual-queue coupled AQM in [I-D.ietf-tsvwg-aqm-dualq-coupled]).

5.2. The Meaning of L4S CE Relative to Drop

The likelihood that an AQM drops a Not-ECT Classic packet (p_C) MUST be roughly proportional to the square of the likelihood that it would have marked it if it had been an L4S packet (p_L). That is

The constant of proportionality (k) does not have to be standardised for interoperability, but a value of 2 is RECOMMENDED. The term 'likelihood' is used above to allow for marking and dropping to be either probabilistic or deterministic.

This formula ensures that Scalable and Classic flows will converge to roughly equal congestion windows, for the worst case of Reno congestion control. This is because the congestion windows of Scalable and Classic congestion controls are inversely proportional to p_L and sqrt(p_C) respectively. So squaring p_C in the above formula counterbalances the square root that characterizes TCP-Reno-friendly flows.

[I-D.ietf-tsvwg-aqm-dualq-coupled] specifies the essential aspects of an L4S AQM, as well as recommending other aspects. It gives example implementations in appendices.

Note that, contrary to RFC 3168, a Coupled Dual Queue AQM implementing the L4S and Classic treatments does not mark an ECT(1) packet under the same conditions that it would have dropped a Not-ECT packet, as allowed by [RFC8311], which updates RFC 3168. However, if it marks ECT(0) packets, it does so under the same conditions that it would have dropped a Not-ECT packet.

5.3. Exception for L4S Packet Identification by Network Nodes with Transport-Layer Awareness

To implement the L4S treatment, a network node does not need to identify transport-layer flows. Nonetheless, if an implementer is willing to identify transport-layer flows at a network node, and if the most recent ECT packet in the same flow was ECT(0), the node MAY classify CE packets for Classic ECN [RFC3168] treatment. In all other cases, a network node MUST classify all CE packets for L4S treatment. Examples of such other cases are: i) if no ECT packets have yet been identified in a flow; ii) if it is not desirable for a network node to identify transport-layer flows; or iii) if the most recent ECT packet in a flow was ECT(1).

If an implementer uses flow-awareness to classify CE packets, to determine whether the flow is using ECT(0) or ECT(1) it only uses the most recent ECT packet of a flow (this advice will need to be verified as part of L4S experiments). This is because a sender might switch from sending ECT(1) (L4S) packets to sending ECT(0) (Classic ECN) packets, or back again, in the middle of a transport-layer flow (e.g. it might manually switch its congestion control module mid-connection, or it might be deliberately attempting to confuse the network).

5.4. Interaction of the L4S Identifier with other Identifiers

The examples in this section concern how additional identifiers might complement the L4S identifier to classify packets between class-based queues. Firstly considering two queues, L4S and Classic, as in the Coupled DualQ AQM [I-D.ietf-tsvwg-aqm-dualq-coupled], then more complex structures within a larger queuing hierarchy.

5.4.1. DualQ Examples of Other Identifiers Complementing L4S Identifiers Inclusion of Additional Traffic with L4S

In a typical case for the public Internet a network element that implements L4S might want to classify some low-rate but unresponsive traffic (e.g. DNS, LDAP, NTP, voice, game sync packets) into the low latency queue to mix with L4S traffic. Such non-ECN-based packet types MUST be safe to mix with L4S traffic without harming the low latency service, where 'safe' is explained in Section below.

In this case it would not be appropriate to call the queue an L4S queue, because it is shared by L4S and non-L4S traffic. Instead it will be called the low latency or L queue. The L queue then offers two different treatments:

To identify packets for just the scheduling treatment, it would be inappropriate to use the L4S ECT(1) identifier, because such traffic is unresponsive to ECN marking. Therefore, a network element that implements L4S MAY classify additional packets into the L queue if they carry certain non-ECN identifiers. For instance:

Of course, a packet that carried both the ECT(1) codepoint and a non-ECN identifier associated with the L queue would be classified into the L queue.

For clarity, non-ECN identifiers, such as the examples itemized above, might be used by some network operators who believe they identify non-L4S traffic that would be safe to mix with L4S traffic. They are not alternative ways for a host to indicate that it is sending L4S packets. Only the ECT(1) ECN codepoint indicates to a network element that a host is sending L4S packets (and CE indicates that it could have originated as ECT(1)). Specifically ECT(1) indicates that the host claims its behaviour satisfies the prerequisite transport requirements in Section 4.

To include additional traffic with L4S, a network element only reads identifiers such as those itemized above. It MUST NOT alter these non-ECN identifiers, so that they survive for any potential use later on the network path. 'Safe' Unresponsive Traffic

The above section requires unresponsive traffic to be 'safe' to mix with L4S traffic. Ideally this means that the sender never sends any sequence of packets at a rate that exceeds the available capacity of the bottleneck link. However, typically an unresponsive transport does not even know the bottleneck capacity of the path, let alone its available capacity. Nonetheless, an application can be considered safe enough if it paces packets out (not necessarily completely regularly) such that its maximum instantaneous rate from packet to packet stays well below a typical broadband access rate.

This is a vague but useful definition, because many low latency applications of interest, such as DNS, voice, game sync packets, RPC, ACKs, keep-alives, could match this description. Exclusion of Traffic From L4S Treatment

To extend the above example, an operator might want to exclude some traffic from the L4S treatment for a policy reason, e.g. security (traffic from malicious sources) or commercial (e.g. initially the operator may wish to confine the benefits of L4S to business customers).

In this exclusion case, the operator MUST classify on the relevant locally-used identifiers (e.g. source addresses) before classifying the non-matching traffic on the end-to-end L4S ECN identifier.

The operator MUST NOT alter the end-to-end L4S ECN identifier from L4S to Classic, because its decision to exclude certain traffic from L4S treatment is local-only. The end-to-end L4S identifier then survives for other operators to use, or indeed, they can apply their own policy, independently based on their own choice of locally-used identifiers. This approach also allows any operator to remove its locally-applied exclusions in future, e.g. if it wishes to widen the benefit of the L4S treatment to all its customers. Generalized Combination of L4S and Other Identifiers

L4S concerns low latency, which it can provide for all traffic without differentiation and without affecting bandwidth allocation. Diffserv provides for differentiation of both bandwidth and low latency, but its control of latency depends on its control of bandwidth. The two can be combined if a network operator wants to control bandwidth allocation but it also wants to provide low latency - for any amount of traffic within one of these allocations of bandwidth (rather than only providing low latency by limiting bandwidth) [I-D.briscoe-tsvwg-l4s-diffserv].

The DualQ examples so far have been framed in the context of providing the default Best Efforts Per-Hop Behaviour (PHB) using two queues - a Low Latency (L) queue and a Classic (C) Queue. This single DualQ structure is expected to be the most common and useful arrangement. But, more generally, an operator might choose to control bandwidth allocation through a hierarchy of Diffserv PHBs at a node, and to offer one (or more) of these PHBs with a low latency and a Classic variant.

In the first case, if we assume that there are no other PHBs except the DualQ, if a packet carries ECT(1) or CE, a network element would classify it for the L4S treatment irrespective of its DSCP. And, if a packet carried (say) the EF DSCP, the network element could classify it into the L queue irrespective of its ECN codepoint. However, where the DualQ is in a hierarchy of other PHBs, the classifier would classify some traffic into other PHBs based on DSCP before classifying between the low latency and Classic queues (based on ECT(1), CE and perhaps also the EF DSCP or other identifiers as in the above example). [I-D.briscoe-tsvwg-l4s-diffserv] gives a number of examples of such arrangements to address various requirements.

[I-D.briscoe-tsvwg-l4s-diffserv] describes how an operator might use L4S to offer low latency for all L4S traffic as well as using Diffserv for bandwidth differentiation. It identifies two main types of approach, which can be combined: the operator might split certain Diffserv PHBs between L4S and a corresponding Classic service. Or it might split the L4S and/or the Classic service into multiple Diffserv PHBs. In either of these cases, a packet would have to be classified on its Diffserv and ECN codepoints.

In summary, there are numerous ways in which the L4S ECN identifier (ECT(1) and CE) could be combined with other identifiers to achieve particular objectives. The following categorization articulates those that are valid, but it is not necessarily exhaustive. Those tagged 'Recommended-standard-use' could be set by the sending host or a network. Those tagged 'Local-use' would only be set by a network:

  1. Identifiers Complementing the L4S Identifier
    1. Including More Traffic in the L Queue
      (Could use Recommended-standard-use or Local-use identifiers)
    2. Excluding Certain Traffic from the L Queue
      (Local-use only)
  2. Identifiers to place L4S classification in a PHB Hierarchy
    (Could use Recommended-standard-use or Local-use identifiers)
    1. PHBs Before L4S ECN Classification
    2. PHBs After L4S ECN Classification

5.4.2. Per-Flow Queuing Examples of Other Identifiers Complementing L4S Identifiers

At a node with per-flow queueing (e.g. FQ-CoDel [RFC8290]), the L4S identifier could complement the Layer-4 flow ID as a further level of flow granularity (i.e. Not-ECT and ECT(0) queued separately from ECT(1) and CE packets). "Risk of reordering Classic CE packets" in Appendix B.1 discusses the resulting ambiguity if packets originally marked ECT(0) are marked CE by an upstream AQM before they arrive at a node that classifies CE as L4S. It argues that the risk of re-ordering is vanishingly small and the consequence of such a low level of re-ordering is minimal.

Alternatively, it could be assumed that it is not in a flow's own interest to mix Classic and L4S identifiers. Then the AQM could use the ECN field to switch itself between a Classic and an L4S AQM behaviour within one per-flow queue. For instance, for ECN-capable packets, the AQM might consist of a simple marking threshold and an L4S ECN identifier might simply select a shallower threshold than a Classic ECN identifier would.

6. L4S Experiments

[I-D.ietf-tsvwg-aqm-dualq-coupled] sets operational and management requirements for experiments with DualQ Coupled AQMs. General operational and management requirements for experiments with L4S congestion controls are given in Section 4 and Section 5 above, e.g. co-existence and scaling requirements, incremental deployment arrangements.

The specification of each scalable congestion control will need to include protocol-specific requirements for configuration and monitoring performance during experiments. Appendix A of [RFC5706] provides a helpful checklist.

Monitoring for harm to other traffic, specifically bandwidth starvation or excess queuing delay, will need to be conducted alongside all early L4S experiments. It is hard, if not impossible, for an individual flow to measure its impact on other traffic. So such monitoring will need to be conducted using bespoke monitoring across flows and/or across classes of traffic.

7. IANA Considerations

This specification contains no IANA considerations.

8. Security Considerations

Approaches to assure the integrity of signals using the new identifer are introduced in Appendix C.1. See the security considerations in the L4S architecture [I-D.ietf-tsvwg-l4s-arch] for further discussion of mis-use of the identifier.

The recommendation to detect loss in time units prevents the ACK-splitting attacks described in [Savage-TCP].

9. Acknowledgements

Thanks to Richard Scheffenegger, John Leslie, David Täht, Jonathan Morton, Gorry Fairhurst, Michael Welzl, Mikael Abrahamsson and Andrew McGregor for the discussions that led to this specification. Ing-jyh (Inton) Tsang was a contributor to the early drafts of this document. And thanks to Mikael Abrahamsson, Lloyd Wood, Nicolas Kuhn, Greg White, Tom Henderson, David Black, Gorry Fairhurst, Brian Carpenter, Jake Holland, Rod Grimes and Richard Scheffenegger for providing help and reviewing this draft and to Ingemar Johansson for reviewing and providing substantial text. Appendix A listing the Prague L4S Requirements is based on text authored by Marcelo Bagnulo Braun that was originally an appendix to [I-D.ietf-tsvwg-l4s-arch]. That text was in turn based on the collective output of the attendees listed in the minutes of a 'bar BoF' on DCTCP Evolution during IETF-94 [TCPPrague].

The authors' contributions were part-funded by the European Community under its Seventh Framework Programme through the Reducing Internet Transport Latency (RITE) project (ICT-317700). Bob Briscoe was also funded partly by the Research Council of Norway through the TimeIn project, partly by CableLabs and partly by the Comcast Innovation Fund. The views expressed here are solely those of the authors.

10. References

10.1. Normative References

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997.
[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.
[RFC4774] Floyd, S., "Specifying Alternate Semantics for the Explicit Congestion Notification (ECN) Field", BCP 124, RFC 4774, DOI 10.17487/RFC4774, November 2006.
[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.

10.2. Informative References

[A2DTCP] Zhang, T., Wang, J., Huang, J., Huang, Y., Chen, J. and Y. Pan, "Adaptive-Acceleration Data Center TCP", IEEE Transactions on Computers 64(6):1522-1533, June 2015.
[Ahmed19] Ahmed, A., "Extending TCP for Low Round Trip Delay", Masters Thesis, Uni Oslo , August 2019.
[Alizadeh-stability] Alizadeh, M., Javanmard, A. and B. Prabhakar, "Analysis of DCTCP: Stability, Convergence, and Fairness", ACM SIGMETRICS 2011 , June 2011.
[ARED01] Floyd, S., Gummadi, R. and S. Shenker, "Adaptive RED: An Algorithm for Increasing the Robustness of RED's Active Queue Management", ACIRI Technical Report , August 2001.
[DCttH15] De Schepper, K., Bondarenko, O., Briscoe, B. and I. Tsang, "'Data Centre to the Home': Ultra-Low Latency for All", RITE Project Technical Report , 2015.
[I-D.briscoe-tsvwg-l4s-diffserv] Briscoe, B., "Interactions between Low Latency, Low Loss, Scalable Throughput (L4S) and Differentiated Services", Internet-Draft draft-briscoe-tsvwg-l4s-diffserv-02, November 2018.
[I-D.ietf-avtcore-cc-feedback-message] Sarker, Z., Perkins, C., Singh, V. and M. Ramalho, "RTP Control Protocol (RTCP) Feedback for Congestion Control", Internet-Draft draft-ietf-avtcore-cc-feedback-message-05, November 2019.
[I-D.ietf-quic-transport] Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed and Secure Transport", Internet-Draft draft-ietf-quic-transport-25, January 2020.
[I-D.ietf-tcpm-accurate-ecn] Briscoe, B., Kuehlewind, M. and R. Scheffenegger, "More Accurate ECN Feedback in TCP", Internet-Draft draft-ietf-tcpm-accurate-ecn-09, July 2019.
[I-D.ietf-tcpm-generalized-ecn] Bagnulo, M. and B. Briscoe, "ECN++: Adding Explicit Congestion Notification (ECN) to TCP Control Packets", Internet-Draft draft-ietf-tcpm-generalized-ecn-05, November 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", Internet-Draft draft-ietf-tcpm-rack-07, January 2020.
[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)", Internet-Draft draft-ietf-tsvwg-aqm-dualq-coupled-10, July 2019.
[I-D.ietf-tsvwg-ecn-encap-guidelines] Briscoe, B., Kaippallimalil, J. and P. Thaler, "Guidelines for Adding Congestion Notification to Protocols that Encapsulate IP", Internet-Draft draft-ietf-tsvwg-ecn-encap-guidelines-13, May 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", Internet-Draft draft-ietf-tsvwg-l4s-arch-04, July 2019.
[I-D.ietf-tsvwg-nqb] White, G. and T. Fossati, "A Non-Queue-Building Per-Hop Behavior (NQB PHB) for Differentiated Services", Internet-Draft draft-ietf-tsvwg-nqb-00, November 2019.
[I-D.sridharan-tcpm-ctcp] Sridharan, M., Tan, K., Bansal, D. and D. Thaler, "Compound TCP: A New TCP Congestion Control for High-Speed and Long Distance Networks", Internet-Draft draft-sridharan-tcpm-ctcp-02, November 2008.
[I-D.stewart-tsvwg-sctpecn] Stewart, R., Tuexen, M. and X. Dong, "ECN for Stream Control Transmission Protocol (SCTP)", Internet-Draft draft-stewart-tsvwg-sctpecn-05, January 2014.
[LinuxPacedChirping] Misund, J. and B. Briscoe, "Paced Chirping - Rethinking TCP start-up", Proc. Linux Netdev 0x13 , March 2019.
[LinuxPrague] Briscoe, B., De Schepper, K., Albisser, O., Misund, J., Tilmans, O., Kühlewind, M. and A. Ahmed, "Implementing the `TCP Prague' Requirements for Low Latency Low Loss Scalable Throughput (L4S)", Proc. Linux Netdev 0x13 , March 2019.
[Mathis09] Mathis, M., "Relentless Congestion Control", PFLDNeT'09 , May 2009.
[Paced-Chirping] Misund, J., "Rapid Acceleration in TCP Prague", Masters Thesis , May 2018.
[PI2] De Schepper, K., Bondarenko, O., Tsang, I. and B. Briscoe, "PI^2 : A Linearized AQM for both Classic and Scalable TCP", Proc. ACM CoNEXT 2016 pp.105-119, December 2016.
[QV] Briscoe, B. and P. Hurtig, "Up to Speed with Queue View", RITE Technical Report D2.3; Appendix C.2, August 2015.
[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.
[RFC2474] Nichols, K., Blake, S., Baker, F. and D. Black, "Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers", RFC 2474, DOI 10.17487/RFC2474, December 1998.
[RFC2983] Black, D., "Differentiated Services and Tunnels", RFC 2983, DOI 10.17487/RFC2983, October 2000.
[RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec, J., Courtney, W., Davari, S., Firoiu, V. and D. Stiliadis, "An Expedited Forwarding PHB (Per-Hop Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002.
[RFC3540] Spring, N., Wetherall, D. and D. Ely, "Robust Explicit Congestion Notification (ECN) Signaling with Nonces", RFC 3540, DOI 10.17487/RFC3540, June 2003.
[RFC3649] Floyd, S., "HighSpeed TCP for Large Congestion Windows", RFC 3649, DOI 10.17487/RFC3649, December 2003.
[RFC4340] Kohler, E., Handley, M. and S. Floyd, "Datagram Congestion Control Protocol (DCCP)", RFC 4340, DOI 10.17487/RFC4340, March 2006.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March 2006.
[RFC4342] Floyd, S., Kohler, E. and J. Padhye, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, DOI 10.17487/RFC4342, March 2006.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 4960, DOI 10.17487/RFC4960, September 2007.
[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.
[RFC5562] Kuzmanovic, A., Mondal, A., Floyd, S. and K. Ramakrishnan, "Adding Explicit Congestion Notification (ECN) Capability to TCP's SYN/ACK Packets", RFC 5562, DOI 10.17487/RFC5562, June 2009.
[RFC5622] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate Control for Small Packets (TFRC-SP)", RFC 5622, DOI 10.17487/RFC5622, August 2009.
[RFC5681] Allman, M., Paxson, V. and E. Blanton, "TCP Congestion Control", RFC 5681, DOI 10.17487/RFC5681, September 2009.
[RFC5706] Harrington, D., "Guidelines for Considering Operations and Management of New Protocols and Protocol Extensions", RFC 5706, DOI 10.17487/RFC5706, November 2009.
[RFC5865] Baker, F., Polk, J. and M. Dolly, "A Differentiated Services Code Point (DSCP) for Capacity-Admitted Traffic", RFC 5865, DOI 10.17487/RFC5865, May 2010.
[RFC5925] Touch, J., Mankin, A. and R. Bonica, "The TCP Authentication Option", RFC 5925, DOI 10.17487/RFC5925, June 2010.
[RFC6077] Papadimitriou, D., Welzl, M., Scharf, M. and B. Briscoe, "Open Research Issues in Internet Congestion Control", RFC 6077, DOI 10.17487/RFC6077, February 2011.
[RFC6660] Briscoe, B., Moncaster, T. and M. Menth, "Encoding Three Pre-Congestion Notification (PCN) States in the IP Header Using a Single Diffserv Codepoint (DSCP)", RFC 6660, DOI 10.17487/RFC6660, July 2012.
[RFC7560] Kuehlewind, M., Scheffenegger, R. and B. Briscoe, "Problem Statement and Requirements for Increased Accuracy in Explicit Congestion Notification (ECN) Feedback", RFC 7560, DOI 10.17487/RFC7560, August 2015.
[RFC7567] Baker, F. and G. Fairhurst, "IETF Recommendations Regarding Active Queue Management", BCP 197, RFC 7567, DOI 10.17487/RFC7567, July 2015.
[RFC7713] Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx) Concepts, Abstract Mechanism, and Requirements", RFC 7713, DOI 10.17487/RFC7713, December 2015.
[RFC8033] Pan, R., Natarajan, P., Baker, F. and G. White, "Proportional Integral Controller Enhanced (PIE): A Lightweight Control Scheme to Address the Bufferbloat Problem", RFC 8033, DOI 10.17487/RFC8033, February 2017.
[RFC8257] Bensley, S., Thaler, D., Balasubramanian, P., Eggert, L. and G. Judd, "Data Center TCP (DCTCP): TCP Congestion Control for Data Centers", RFC 8257, DOI 10.17487/RFC8257, October 2017.
[RFC8290] Hoeiland-Joergensen, T., McKenney, P., Taht, D., Gettys, J. and E. Dumazet, "The Flow Queue CoDel Packet Scheduler and Active Queue Management Algorithm", RFC 8290, DOI 10.17487/RFC8290, January 2018.
[RFC8298] Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation for Multimedia", RFC 8298, DOI 10.17487/RFC8298, December 2017.
[RFC8311] Black, D., "Relaxing Restrictions on Explicit Congestion Notification (ECN) Experimentation", RFC 8311, DOI 10.17487/RFC8311, January 2018.
[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.
[Savage-TCP] Savage, S., Cardwell, N., Wetherall, D. and T. Anderson, "TCP Congestion Control with a Misbehaving Receiver", ACM SIGCOMM Computer Communication Review 29(5):71--78, October 1999.
[sub-mss-prob] Briscoe, B. and K. De Schepper, "Scaling TCP's Congestion Window for Small Round Trip Times", BT Technical Report TR-TUB8-2015-002, May 2015.
[TCP-CA] Jacobson, V. and M. Karels, "Congestion Avoidance and Control", Laurence Berkeley Labs Technical Report , November 1988.
[TCPPrague] Briscoe, B., "Notes: DCTCP evolution 'bar BoF': Tue 21 Jul 2015, 17:40, Prague", tcpprague mailing list archive , July 2015.
[VCP] Xia, Y., Subramanian, L., Stoica, I. and S. Kalyanaraman, "One more bit is enough", Proc. SIGCOMM'05, ACM CCR 35(4)37--48, 2005.

Appendix A. The 'Prague L4S Requirements'

This appendix is informative, not normative. It gives a list of modifications to current scalable congestion controls so that they can be deployed over the public Internet and coexist safely with existing traffic. The list complements the normative requirements in Section 4 that a sender has to comply with before it can set the L4S identifier in packets it sends into the Internet. As well as necessary safety improvements (requirements) this appendix also includes preferable performance improvements (optimizations).

These recommendations have become know as the Prague L4S Requirements, because they were originally identified at an ad hoc meeting during IETF-94 in Prague [TCPPrague]. The wording has been generalized to apply to all scalable congestion controls, not just TCP congestion control specifically. They were originally called the 'TCP Prague Requirements', but they are not solely applicable to TCP, so the name has been generalized, and TCP Prague is now used for a specific implementation of the requirements.

At the time of writing, DCTCP [RFC8257] is the most widely used scalable transport protocol. In its current form, DCTCP is specified to be deployable only in controlled environments. Deploying it in the public Internet would lead to a number of issues, both from the safety and the performance perspective. The modifications and additional mechanisms listed in this section will be necessary for its deployment over the global Internet. Where an example is needed, DCTCP is used as a base, but it is likely that most of these requirements equally apply to other scalable congestion controls.

A.1. Requirements for Scalable Transport Protocols

A.1.1. Use of L4S Packet Identifier

Description: A scalable congestion control needs to distinguish the packets it sends from those sent by Classic congestion controls.

Motivation: It needs to be possible for a network node to classify L4S packets without flow state into a queue that applies an L4S ECN marking behaviour and isolates L4S packets from the queuing delay of Classic packets.

A.1.2. Accurate ECN Feedback

Description: The transport protocol for a scalable congestion control needs to provide timely, accurate feedback about the extent of ECN marking experienced by all packets.

Motivation: Classic congestion controls only need feedback about the existence of a congestion episode within a round trip, not precisely how many packets were marked with ECN or dropped. Therefore, in 2001, when ECN feedback was added to TCP [RFC3168], it could not inform the sender of more than one ECN mark per RTT. Since then, requirements for more accurate ECN feedback in TCP have been defined in [RFC7560] and [I-D.ietf-tcpm-accurate-ecn] specifies an experimental change to the TCP wire protocol to satisfy these requirements. Most other transport protocols already satisfy this requirement.

A.1.3. Fall back to Reno-friendly congestion control on packet loss

Description: As well as responding to ECN markings in a scalable way, a scalable congestion control needs to react to packet loss in a way that will coexist safely with a TCP Reno congestion control [RFC5681].

Motivation: Part of the safety conditions for deploying a scalable congestion control on the public Internet is to make sure that it behaves properly when it builds a queue at a network bottleneck that has not been upgraded to support L4S. Packet loss can have many causes, but it usually has to be conservatively assumed that it is a sign of congestion. Therefore, on detecting packet loss, a scalable congestion control will need to fall back to Classic congestion control behaviour. If it does not comply with this requirement it could starve Classic traffic.

A scalable congestion control can be used for different types of transport, e.g. for real-time media or for reliable transport like TCP. Therefore, the particular Classic congestion control behaviour to fall back on will need to be part of the congestion control specification of the relevant transport. In the particular case of DCTCP, the DCTCP specification [RFC8257] states that "It is RECOMMENDED that an implementation deal with loss episodes in the same way as conventional TCP." For safe deployment of a scalable congestion control in the public Internet, the above requirement would need to be defined as a "MUST".

Even though a bottleneck is L4S capable, it might still become overloaded and have to drop packets. In this case, the sender may receive a high proportion of packets marked with the CE bit set and also experience loss. Current DCTCP implementations react differently to this situation. At least one implementation reacts only to the drop signal (e.g. by halving the CWND) and at least another DCTCP implementation reacts to both signals (e.g. by halving the CWND due to the drop and also further reducing the CWND based on the proportion of marked packet). A third approach for the public Internet has been proposed that adjusts the loss response to result in a halving when combined with the ECN response. We believe that further experimentation is needed to understand what is the best behaviour for the public Internet, which may or not be one of these existing approaches.

A.1.4. Fall back to Reno-friendly congestion control on classic ECN bottlenecks

Description: A scalable congestion control needs to react to ECN marking from a non-L4S, but ECN-capable, bottleneck in a way that will coexist with a TCP Reno congestion control [RFC5681].

Motivation: Similarly to the requirement in Appendix A.1.3, this requirement is a safety condition to ensure a scalable congestion control behaves properly when it builds a queue at a network bottleneck that has not been upgraded to support L4S. On detecting Classic ECN marking (see below), a scalable congestion control will need to fall back to Classic congestion control behaviour. If it does not comply with this requirement it could starve Classic traffic.

It would take time for endpoints to distinguish Classic and L4S ECN marking. An increase in queuing delay or in delay variation would be a tell-tale sign, but it is not yet clear where a line would be drawn between the two behaviours. It might be possible to cache what was learned about the path to help subsequent attempts to detect the type of marking.

A.1.5. Reduce RTT dependence

Description: A scalable congestion control needs to reduce or eliminate RTT bias over as wide a range of RTTs as possible, or at least over the typical range of RTTs that will interact in the intended deployment scenario.

Motivation: The throughput of Classic congestion controls is known to be inversely proportional to RTT, so one would expect flows over very low RTT paths to nearly starve flows over larger RTTs. However, Classic congestion controls have never allowed a very low RTT path to exist because they induce a large queue. For instance, consider two paths with base RTT 1ms and 100ms. If a Classic congestion control induces a 100ms queue, it turns these RTTs into 101ms and 200ms leading to a throughput ratio of about 2:1. Whereas if a scalable congestion control induces only a 1ms queue, the ratio is 2:101, leading to a throughput ratio of about 50:1.

Therefore, with very small queues, long RTT flows will essentially starve, unless scalable congestion controls comply with this requirement.

A.1.6. Scaling down to fractional congestion windows

Description: A scalable congestion control needs to remain responsive to congestion when RTTs are significantly smaller than in the current public Internet.

Motivation: As currently specified, the minimum required congestion window of TCP (and its derivatives) is set to 2 sender maximum segment sizes (SMSS) (see equation (4) in [RFC5681]). Once the congestion window reaches this minimum, all known window-based congestion control algorithms become unresponsive to congestion signals. No matter how much drop or ECN marking, the congestion window no longer reduces. Instead, the sender's lack of any further congestion response forces the queue to grow, overriding any AQM and increasing queuing delay.

L4S mechanisms significantly reduce queueing delay so, over the same path, the RTT becomes lower. Then this problem becomes surprisingly common [sub-mss-prob]. This is because, for the same link capacity, smaller RTT implies a smaller window. For instance, consider a residential setting with an upstream broadband Internet access of 8 Mb/s, assuming a max segment size of 1500 B. Two upstream flows will each have the minimum window of 2 SMSS if the RTT is 6ms or less, which is quite common when accessing a nearby data centre. So, any more than two such parallel TCP flows will become unresponsive and increase queuing delay.

Unless scalable congestion controls are required to comply with this requirement from the start, they will frequently become unresponsive, negating the low latency benefit of L4S, for themselves and for others. One possible sub-MSS window mechanism is described in [Ahmed19], and other approaches are likely to be feasible.

A.1.7. Measuring Reordering Tolerance in Time Units

Description: A scalable congestion control needs to detect loss by counting in time-based units, which is scalable, rather than counting in units of packets, which is not.

Motivation: A primary purpose of L4S is scalable throughput (it's in the name). Scalability in all dimensions is, of course, also a goal of all IETF technology. The inverse linear congestion response in Section 4.3 is necessary, but not sufficient, to solve the congestion control scalability problem identified in [RFC3649]. As well as maintaining frequent ECN signals as rate scales, it is also important to ensure that a potentially false perception of loss does not limit throughput scaling.

End-systems cannot know whether a missing packet is due to loss or reordering, except in hindsight - if it appears later. So they can only deem that there has been a loss if a gap in the sequence space has not been filled, either after a certain number of subsequent packets has arrived (e.g. the 3 DupACK rule of standard TCP congestion control [RFC5681]) or after a certain amount of time (e.g. the experimental RACK approach [I-D.ietf-tcpm-rack]).

As we attempt to scale packet rate over the years:

Therefore hosts have an incentive to detect loss in time units (so as not to fool themselves too often into detecting losses when there are none). And for hosts that are changing their congestion control implementation to L4S, there is no downside to including time-based loss detection code in the change (loss recovery implemented in hardware is an exception, covered later). Therefore requiring L4S hosts to detect loss in time-based units would not be a burden.

If this requirement is not placed on L4S hosts, even though it would be no burden on them to do so, all networks will face unnecessary uncertainty over whether some L4S hosts might be detecting loss by counting packets. Then all link technologies will have to unnecessarily keep reducing the time within which reordering occurs. That is not a problem for some link technologies, but it becomes increasingly challenging for other link technologies to continue to scale, particularly those relying on channel bonding for scaling, such as LTE, 5G and DOCSIS.

Given Internet paths traverse many link technologies, any scaling limit for these more challenging access link technologies would become a scaling limit for the Internet as a whole.

It might be asked how it helps to place this loss detection requirement only on L4S hosts, because networks will still face uncertainty over whether non-L4S flows are detecting loss by counting DupACKs. The answer is that those link technologies for which it is challenging to keep squeezing the reordering time will only need to do so for non-L4S traffic (which they can do because the L4S identifier is visible at the IP layer). Therefore, they can focus their processing and memory resources into scaling non-L4S (Classic) traffic. Then, the higher the proportion of L4S traffic, the less of a scaling challenge they will have.

To summarize, there is no reason for L4S hosts not to be part of the solution instead of part of the problem.

Requirement ("MUST") or recommendation ("SHOULD")? As explained above, this is a subtle interoperability issue between hosts and networks, which seems to need a "MUST". Unless networks can be certain that all L4S hosts follow the time-based approach, they still have to cater for the worst case - continually squeeze reordering into a smaller and smaller duration - just for hosts that might be using the counting approach. However, it was decided to express this as a recommendation, using "SHOULD". The main justification was that networks can still be fairly certain that L4S hosts will follow this recommendation, because following it offers only gain and no pain.


The speed of loss recovery is much more significant for short flows than long, therefore a good compromise is to adapt the reordering window; from a small fraction of the RTT at the start of a flow, to a larger fraction of the RTT for flows that continue for many round trips.

This is broadly the approach adopted by TCP RACK (Recent ACKnowledgements) [I-D.ietf-tcpm-rack]. However, RACK starts with the 3 DupACK approach, because the RTT estimate is not necessarily stable. As long as the initial window is paced, such initial use of 3 DupACK counting would amount to time-based loss detection and therefore would satisfy the time-based loss detection recommendation of Section 4.3. This is because pacing of the initial window would ensure that 3 DupACKs early in the connection would be spread over a small fraction of the round trip.

As mentioned above, hardware implementations of loss recovery using DupACK counting exist (e.g. some implementations of RoCEv2 for RDMA). For low latency, these implementations can change their congestion control to implement L4S, because the congestion control (as distinct from loss recovery) is implemented in software. But they cannot easily satisfy this loss recovery requirement. However, it is believed they do not need to. It is believed that such implementations solely exist in controlled environments, where the network technology keeps reordering extremely low anyway. This is why the scope of the normative recommendation in Section 4.3 is limited to 'reordering-prone' networks.

Detecting loss in time units also prevents the ACK-splitting attacks described in [Savage-TCP].

A.2. Scalable Transport Protocol Optimizations

A.2.1. Setting ECT in TCP Control Packets and Retransmissions

Description: This item only concerns TCP and its derivatives (e.g. SCTP), because the original specification of ECN for TCP precluded the use of ECN on control packets and retransmissions. To improve performance, scalable transport protocols ought to enable ECN at the IP layer in TCP control packets (SYN, SYN-ACK, pure ACKs, etc.) and in retransmitted packets. The same is true for derivatives of TCP, e.g. SCTP.

Motivation: RFC 3168 prohibits the use of ECN on these types of TCP packet, based on a number of arguments. This means these packets are not protected from congestion loss by ECN, which considerably harms performance, particularly for short flows. [I-D.ietf-tcpm-generalized-ecn] counters each argument in RFC 3168 in turn, showing it was over-cautious. Instead it proposes experimental use of ECN on all types of TCP packet as long as AccECN feedback [I-D.ietf-tcpm-accurate-ecn] is available (which is itself a prerequisite for using a scalable congestion control).

A.2.2. Faster than Additive Increase

Description: It would improve performance if scalable congestion controls did not limit their congestion window increase to the standard additive increase of 1 SMSS per round trip [RFC5681] during congestion avoidance. The same is true for derivatives of TCP congestion control, including similar approaches used for real-time media.

Motivation: As currently defined [RFC8257], DCTCP uses the traditional TCP Reno additive increase in congestion avoidance phase. When the available capacity suddenly increases (e.g. when another flow finishes, or if radio capacity increases) it can take very many round trips to take advantage of the new capacity. TCP Cubic was designed to solve this problem, but as flow rates have continued to increase, the delay accelerating into available capacity has become prohibitive. For instance, with RTT=20 ms, to increase flow rate from 100Mb/s to 200Mb/s Cubic takes between 50 and 100 round trips. Every 8x increase in flow rate leads to 2x more acceleration delay.

In the steady state, DCTCP induces about 2 ECN marks per round trip, so it is possible to quickly detect when these signals have disappeared and seek available capacity more rapidly, while minimizing the impact on other flows (Classic and scalable) [LinuxPacedChirping]. Alternatively, approaches such as Adaptive Acceleration (A2DTCP [A2DTCP]) have been proposed to address this problem in data centres, which might be deployable over the public Internet.

A.2.3. Faster Convergence at Flow Start

Description: Particularly when a flow starts, scalable congestion controls need to converge (reach their steady-state share of the capacity) at least as fast as Classic congestion controls and preferably faster. This affects the flow start behaviour of any L4S congestion control derived from a Classic transport that uses TCP slow start, including those for real-time media.

Motivation: As an example, a new DCTCP flow takes longer than a Classic congestion control to obtain its share of the capacity of the bottleneck when there are already ongoing flows using the bottleneck capacity. In a data centre environment DCTCP takes about a factor of 1.5 to 2 longer to converge due to the much higher typical level of ECN marking that DCTCP background traffic induces, which causes new flows to exit slow start early [Alizadeh-stability]. In testing for use over the public Internet the convergence time of DCTCP relative to a regular loss-based TCP slow start is even less favourable [Paced-Chirping]) due to the shallow ECN marking threshold needed for L4S. It is exacerbated by the typically greater mismatch between the link rate of the sending host and typical Internet access bottlenecks. This problem is detrimental in general, but would particularly harm the performance of short flows relative to Classic congestion controls.

Appendix B. Alternative Identifiers

This appendix is informative, not normative. It records the pros and cons of various alternative ways to identify L4S packets to record the rationale for the choice of ECT(1) (Appendix B.1) as the L4S identifier. At the end, Appendix B.6 summarises the distinguishing features of the leading alternatives. It is intended to supplement, not replace the detailed text.

The leading solutions all use the ECN field, sometimes in combination with the Diffserv field. This is because L4S traffic has to indicate that it is ECN-capable anyway, because ECN is intrinsic to how L4S works. Both the ECN and Diffserv fields have the additional advantage that they are no different in either IPv4 or IPv6. A couple of alternatives that use other fields are mentioned at the end, but it is quickly explained why they are not serious contenders.

B.1. ECT(1) and CE codepoints



Consumes the last ECN codepoint:
The L4S service is intended to supersede the service provided by Classic ECN, therefore using ECT(1) to identify L4S packets could ultimately mean that the ECT(0) codepoint was 'wasted' purely to distinguish one form of ECN from its successor.
ECN hard in some lower layers:
It is not always possible to support ECN in an AQM acting in a buffer below the IP layer [I-D.ietf-tsvwg-ecn-encap-guidelines]. In such cases, the L4S service would have to drop rather than mark frames even though they might encapsulate an ECN-capable packet. However, such cases would be unusual.
Risk of reordering Classic CE packets:
Classifying all CE packets into the L4S queue risks any CE packets that were originally ECT(0) being incorrectly classified as L4S. If there were delay in the Classic queue, these incorrectly classified CE packets would arrive early, which is a form of reordering. Reordering can cause TCP senders (and senders of similar transports) to retransmit spuriously. However, the risk of spurious retransmissions would be extremely low for the following reasons:
  1. It is quite unusual to experience queuing at more than one bottleneck on the same path (the available capacities have to be identical).
  2. In only a subset of these unusual cases would the first bottleneck support Classic ECN marking while the second supported L4S ECN marking, which would be the only scenario where some ECT(0) packets could be CE marked by an AQM supporting Classic ECN then the remainder experienced further delay through the Classic side of a subsequent L4S DualQ AQM.
  3. Even then, when a few packets are delivered early, it takes very unusual conditions to cause a spurious retransmission, in contrast to when some packets are delivered late. The first bottleneck has to apply CE-marks to at least N contiguous packets and the second bottleneck has to inject an uninterrupted sequence of at least N of these packets between two packets earlier in the stream (where N is the reordering window that the transport protocol allows before it considers a packet is lost).
    • For example consider N=3, and consider the sequence of packets 100, 101, 102, 103,... and imagine that packets 150,151,152 from later in the flow are injected as follows: 100, 150, 151, 101, 152, 102, 103... If this were late reordering, even one packet arriving 50 out of sequence would trigger a spurious retransmission, but there is no spurious retransmission here, with early reordering, because packet 101 moves the cumulative ACK counter forward before 3 packets have arrived out of order. Later, when packets 148, 149, 153... arrive, even though there is a 3-packet hole, there will be no problem, because the packets to fill the hole are already in the receive buffer.
  4. Even with the current TCP recommendation of N=3 [RFC5681] spurious retransmissions will be unlikely for all the above reasons. As RACK [I-D.ietf-tcpm-rack] is becoming widely deployed, it tends to adapt its reordering window to a larger value of N, which will make the chance of a contiguous sequence of N early arrivals vanishingly small.
  5. Even a run of 2 CE marks within a Classic ECN flow is unlikely, given FQ-CoDel is the only known widely deployed AQM that supports Classic ECN marking and it takes great care to separate out flows and to space any markings evenly along each flow.

It is extremely unlikely that the above set of 5 eventualities that are each unusual in themselves would all happen simultaneously. But, even if they did, the consequences would hardly be dire: the odd spurious fast retransmission. Admittedly TCP (and similar transports) reduce their congestion window when they deem there has been a loss, but even this can be recovered once the sender detects that the retransmission was spurious.

Non-L4S service for control packets:
The Classic ECN RFCs [RFC3168] and [RFC5562] require a sender to clear the ECN field to Not-ECT for retransmissions and certain control packets specifically pure ACKs, window probes and SYNs. When L4S packets are classified by the ECN field alone, these control packets would not be classified into an L4S queue, and could therefore be delayed relative to the other packets in the flow. This would not cause re-ordering (because retransmissions are already out of order, and the control packets carry no data). However, it would make critical control packets more vulnerable to loss and delay. To address this problem, [I-D.ietf-tcpm-generalized-ecn] proposes an experiment in which all TCP control packets and retransmissions are ECN-capable as long as ECN feedback is available.


Should work e2e:
The ECN field generally works end-to-end across the Internet. Unlike the DSCP, the setting of the ECN field is at least forwarded unchanged by networks that do not support ECN, and networks rarely clear it to zero.
Should work in tunnels:
Unlike Diffserv, ECN is defined to always work across tunnels. However, tunnels do not always implement ECN processing as they should do, particularly because IPsec tunnels were defined differently for a few years.
Could migrate to one codepoint:
If all Classic ECN senders eventually evolve to use the L4S service, the ECT(0) codepoint could be reused for some future purpose, but only once use of ECT(0) packets had reduced to zero, or near-zero, which might never happen.

B.2. ECN Plus a Diffserv Codepoint (DSCP)


Use of a DSCP is the only approach for alternative ECN semantics given as an example in [RFC4774]. However, it was perhaps considered more for controlled environments than new end-to-end services.


Consumes DSCP pairs:
A DSCP is obviously not orthogonal to Diffserv. Therefore, wherever the L4S service is applied to multiple Diffserv scheduling behaviours, it would be necessary to replace each DSCP with a pair of DSCPs.
Uses critical lower-layer header space:
The resulting increased number of DSCPs might be hard to support for some lower layer technologies, e.g. 802.1p and MPLS both offer only 3-bits for a maximum of 8 traffic class identifiers. Although L4S should reduce and possibly remove the need for some DSCPs intended for differentiated queuing delay, it will not remove the need for Diffserv entirely, because Diffserv is also used to allocate bandwidth, e.g. by prioritising some classes of traffic over others when traffic exceeds available capacity.
Not end-to-end (host-network):
Very few networks honour a DSCP set by a host. Typically a network will zero (bleach) the Diffserv field from all hosts. Sometimes networks will attempt to identify applications by some form of packet inspection and, based on network policy, they will set the DSCP considered appropriate for the identified application. Network-based application identification might use some combination of protocol ID, port numbers(s), application layer protocol headers, IP address(es), VLAN ID(s) and even packet timing.
Not end-to-end (network-network):
Very few networks honour a DSCP received from a neighbouring network. Typically a network will zero (bleach) the Diffserv field from all neighbouring networks at an interconnection point. Sometimes bilateral arrangements are made between networks, such that the receiving network remarks some DSCPs to those it uses for roughly equivalent services. The likelihood that a DSCP will be bleached or ignored depends on the type of DSCP:
Local-use DSCP:
These tend to be used to implement application-specific network policies, but a bilateral arrangement to remark certain DSCPs is often applied to DSCPs in the local-use range simply because it is easier not to change all of a network's internal configurations when a new arrangement is made with a neighbour.
Recommended standard DSCP:
These do not tend to be honoured across network interconnections more than local-use DSCPs. However, if two networks decide to honour certain of each other's DSCPs, the reconfiguration is a little easier if both of their globally recognised services are already represented by the relevant recommended standard DSCPs.

Note that today a recommended standard DSCP gives little more assurance of end-to-end service than a local-use DSCP. In future the range recommended as standard might give more assurance of end-to-end service than local-use, but it is unlikely that either assurance will be high, particularly given the hosts are included in the end-to-end path.
Not all tunnels:
Diffserv codepoints are often not propagated to the outer header when a packet is encapsulated by a tunnel header. DSCPs are propagated to the outer of uniform mode tunnels, but not pipe mode [RFC2983], and pipe mode is fairly common.
ECN hard in some lower layers::
Because this approach uses both the Diffserv and ECN fields, an AQM wil only work at a lower layer if both can be supported. If individual network operators wished to deploy an AQM at a lower layer, they would usually propagate an IP Diffserv codepoint to the lower layer, using for example IEEE 802.1p. However, the ECN capability is harder to propagate down to lower layers because few lower layers support it.


Could migrate to e2e:
If all usage of Classic ECN migrates to usage of L4S, the DSCP would become redundant, and the ECN capability alone could eventually identify L4S packets without the interconnection problems of Diffserv detailed above, and without having permanently consumed more than one codepoint in the IP header. Although the DSCP does not generally function as an end-to-end identifier (see above), it could be used initially by individual ISPs to introduce the L4S service for their own locally generated traffic.

B.3. ECN capability alone

This approach uses ECN capability alone as the L4S identifier. It would only have been feasible if RFC 3168 ECN had not been widely deployed. This was the case when the choice of L4S identifier was being made and this appendix was first written. Since then, RFC 3168 ECN has been widely deployed and L4S did not take this approach anyway. So this approach is not discussed further, because it is no longer a feasible option.

B.4. Protocol ID

It has been suggested that a new ID in the IPv4 Protocol field or the IPv6 Next Header field could identify L4S packets. However this approach is ruled out by numerous problems:

B.5. Source or destination addressing

Locally, a network operator could arrange for L4S service to be applied based on source or destination addressing, e.g. packets from its own data centre and/or CDN hosts, packets to its business customers, etc. It could use addressing at any layer, e.g. IP addresses, MAC addresses, VLAN IDs, etc. Although addressing might be a useful tactical approach for a single ISP, it would not be a feasible approach to identify an end-to-end service like L4S. Even for a single ISP, it would require packet classifiers in buffers to be dependent on changing topology and address allocation decisions elsewhere in the network. Therefore this approach is not a feasible solution.

B.6. Summary: Merits of Alternative Identifiers

Table 1 provides a very high level summary of the pros and cons detailed against the schemes described respectively in Appendix B.2 and Appendix B.1, for six issues that set them apart.

Comparison of the Merits of Three Alternative Identifiers
Issue DSCP + ECN ECT(1) + CE
initial   eventual initial   eventual
end-to-end N . .      . ? . . . Y      . . Y
tunnels . O .      . O . . . ?      . . Y
lower layers N . .      . ? . . O .      . . ?
codepoints N . .      . . ? N . .      . . ?
reordering . . Y      . . Y . O .      . . ?
ctrl pkts . . Y      . . Y . O .      . . ?

The schemes are scored based on both their capabilities now ('initial') and in the long term ('eventual'). The scores are one of 'N, O, Y', meaning 'Poor', 'Ordinary', 'Good' respectively. The same scores are aligned vertically to aid the eye. A score of "?" in one of the positions means that this approach might optimistically become this good, given sufficient effort. The table summarises the text and is not meant to be understandable without having read the text.

Appendix C. Potential Competing Uses for the ECT(1) Codepoint

The ECT(1) codepoint of the ECN field has already been assigned once for the ECN nonce [RFC3540], which has now been categorized as historic [RFC8311]. ECN is probably the only remaining field in the Internet Protocol that is common to IPv4 and IPv6 and still has potential to work end-to-end, with tunnels and with lower layers. Therefore, ECT(1) should not be reassigned to a different experimental use (L4S) without carefully assessing competing potential uses. These fall into the following categories:

C.1. Integrity of Congestion Feedback

Receiving hosts can fool a sender into downloading faster by suppressing feedback of ECN marks (or of losses if retransmissions are not necessary or available otherwise).

The historic ECN nonce protocol [RFC3540] proposed that a TCP sender could set either of ECT(0) or ECT(1) in each packet of a flow and remember the sequence it had set. If any packet was lost or congestion marked, the receiver would miss that bit of the sequence. An ECN Nonce receiver had to feed back the least significant bit of the sum, so it could not suppress feedback of a loss or mark without a 50-50 chance of guessing the sum incorrectly.

It is highly unlikely that ECT(1) will be needed for integrity protection in future. The ECN Nonce RFC [RFC3540] as been reclassified as historic, partly because other ways have been developed to protect feedback integrity of TCP and other transports [RFC8311] that do not consume a codepoint in the IP header. For instance:

C.2. Notification of Less Severe Congestion than CE

Various researchers have proposed to use ECT(1) as a less severe congestion notification than CE, particularly to enable flows to fill available capacity more quickly after an idle period, when another flow departs or when a flow starts, e.g. VCP [VCP], Queue View (QV) [QV].

Before assigning ECT(1) as an identifer for L4S, we must carefully consider whether it might be better to hold ECT(1) in reserve for future standardisation of rapid flow acceleration, which is an important and enduring problem [RFC6077].

Pre-Congestion Notification (PCN) is another scheme that assigns alternative semantics to the ECN field. It uses ECT(1) to signify a less severe level of pre-congestion notification than CE [RFC6660]. However, the ECN field only takes on the PCN semantics if packets carry a Diffserv codepoint defined to indicate PCN marking within a controlled environment. PCN is required to be applied solely to the outer header of a tunnel across the controlled region in order not to interfere with any end-to-end use of the ECN field. Therefore a PCN region on the path would not interfere with any of the L4S service identifiers proposed in Appendix B.

Authors' Addresses

Koen De Schepper Nokia Bell Labs Antwerp, Belgium EMail: URI:
Bob Briscoe (editor) Independent UK EMail: URI: