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Explicit Flow Measurements Techniques
draft-ietf-ippm-explicit-flow-measurements-02

Document Type Active Internet-Draft (ippm WG)
Authors Mauro Cociglio , Alexandre Ferrieux , Giuseppe Fioccola , Igor Lubashev , Fabio Bulgarella , Massimo Nilo , Isabelle Hamchaoui , Riccardo Sisto
Last updated 2022-11-17 (Latest revision 2022-10-13)
Replaces draft-mdt-ippm-explicit-flow-measurements
Stream Internet Engineering Task Force (IETF)
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Send notices to marcus.ihlar@ericsson.com
draft-ietf-ippm-explicit-flow-measurements-02
IPPM                                                         M. Cociglio
Internet-Draft                                      Telecom Italia - TIM
Intended status: Informational                               A. Ferrieux
Expires: 16 April 2023                                       Orange Labs
                                                             G. Fioccola
                                                     Huawei Technologies
                                                             I. Lubashev
                                                     Akamai Technologies
                                                           F. Bulgarella
                                                                 M. Nilo
                                                    Telecom Italia - TIM
                                                            I. Hamchaoui
                                                             Orange Labs
                                                                R. Sisto
                                                   Politecnico di Torino
                                                         13 October 2022

                 Explicit Flow Measurements Techniques
             draft-ietf-ippm-explicit-flow-measurements-02

Abstract

   This document describes protocol independent methods called Explicit
   Flow Measurement Techniques that employ few marking bits, inside the
   header of each packet, for loss and delay measurement.  The
   endpoints, marking the traffic, signal these metrics to intermediate
   observers allowing them to measure connection performance, and to
   locate the network segment where impairments happen.  Different
   alternatives are considered within this document.  These signaling
   methods apply to all protocols but they are especially valuable when
   applied to protocols that encrypt transport header and do not allow
   traditional methods for delay and loss detection.

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."

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   This Internet-Draft will expire on 16 April 2023.

Copyright Notice

   Copyright (c) 2022 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents (https://trustee.ietf.org/
   license-info) in effect on the date of publication of this document.
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.  Code Components
   extracted from this document must include Revised BSD License text as
   described in Section 4.e of the Trust Legal Provisions and are
   provided without warranty as described in the Revised BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Latency Bits  . . . . . . . . . . . . . . . . . . . . . . . .   5
     2.1.  Spin Bit  . . . . . . . . . . . . . . . . . . . . . . . .   5
     2.2.  Delay Bit . . . . . . . . . . . . . . . . . . . . . . . .   6
       2.2.1.  Generation Phase  . . . . . . . . . . . . . . . . . .   8
       2.2.2.  Reflection Phase  . . . . . . . . . . . . . . . . . .   8
       2.2.3.  T_Max Selection . . . . . . . . . . . . . . . . . . .   9
       2.2.4.  Delay Measurement Using Delay Bit . . . . . . . . . .  10
         2.2.4.1.  RTT Measurement . . . . . . . . . . . . . . . . .  10
         2.2.4.2.  Half-RTT Measurement  . . . . . . . . . . . . . .  11
         2.2.4.3.  Intra-Domain RTT Measurement  . . . . . . . . . .  11
       2.2.5.  Observer's Algorithm  . . . . . . . . . . . . . . . .  12
       2.2.6.  Two Bits Delay Measurement: Spin Bit + Delay Bit  . .  13
   3.  Loss Bits . . . . . . . . . . . . . . . . . . . . . . . . . .  13
     3.1.  T Bit -- Round Trip Loss Bit  . . . . . . . . . . . . . .  14
       3.1.1.  Round Trip Loss . . . . . . . . . . . . . . . . . . .  15
       3.1.2.  Setting the Round Trip Loss Bit on Outgoing
               Packets . . . . . . . . . . . . . . . . . . . . . . .  16
       3.1.3.  Observer's Logic for Round Trip Loss Signal . . . . .  17
       3.1.4.  Loss Coverage and Signal Timing . . . . . . . . . . .  18
     3.2.  Q Bit -- sQuare Bit . . . . . . . . . . . . . . . . . . .  18
       3.2.1.  Q Block Length Selection  . . . . . . . . . . . . . .  19
       3.2.2.  Upstream Loss . . . . . . . . . . . . . . . . . . . .  20
       3.2.3.  Identifying Q Block Boundaries  . . . . . . . . . . .  20
         3.2.3.1.  Improved Resilience to Burst Losses . . . . . . .  21
     3.3.  L Bit -- Loss Event Bit . . . . . . . . . . . . . . . . .  21
       3.3.1.  End-To-End Loss . . . . . . . . . . . . . . . . . . .  22
         3.3.1.1.  Loss Profile Characterization . . . . . . . . . .  22
       3.3.2.  L+Q Bits -- Loss Measurement Using L and Q Bits . . .  22
         3.3.2.1.  Correlating End-to-End and Upstream Loss  . . . .  23

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         3.3.2.2.  Downstream Loss . . . . . . . . . . . . . . . . .  23
         3.3.2.3.  Observer Loss . . . . . . . . . . . . . . . . . .  24
     3.4.  R Bit -- Reflection Square Bit  . . . . . . . . . . . . .  24
       3.4.1.  Enhancement of R Block Length Computation . . . . . .  25
       3.4.2.  Improved Resilience to Packet Reordering  . . . . . .  26
         3.4.2.1.  Improved Resilience to Burst Losses . . . . . . .  26
       3.4.3.  R+Q Bits -- Loss Measurement Using R and Q Bits . . .  26
         3.4.3.1.  Three-Quarters Connection Loss  . . . . . . . . .  27
         3.4.3.2.  End-To-End Loss in the Opposite Direction . . . .  28
         3.4.3.3.  Half Round-Trip Loss  . . . . . . . . . . . . . .  28
         3.4.3.4.  Downstream Loss . . . . . . . . . . . . . . . . .  29
     3.5.  E Bit -- ECN-Echo Event Bit . . . . . . . . . . . . . . .  30
       3.5.1.  Setting the ECN-Echo Event Bit on Outgoing Packets  .  30
       3.5.2.  Using E Bit for Passive ECN-Reported Congestion
               Measurement . . . . . . . . . . . . . . . . . . . . .  30
   4.  Summary of Delay and Loss Marking Methods . . . . . . . . . .  30
   5.  Protocol Ossification Considerations  . . . . . . . . . . . .  32
   6.  Examples of Application . . . . . . . . . . . . . . . . . . .  33
     6.1.  QUIC  . . . . . . . . . . . . . . . . . . . . . . . . . .  33
     6.2.  TCP . . . . . . . . . . . . . . . . . . . . . . . . . . .  34
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  34
     7.1.  Optimistic ACK Attack . . . . . . . . . . . . . . . . . .  35
     7.2.  Delay Bit with RTT Obfuscation  . . . . . . . . . . . . .  35
   8.  Privacy Considerations  . . . . . . . . . . . . . . . . . . .  36
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  36
   10. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  37
   11. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  37
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  37
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  37
     12.2.  Informative References . . . . . . . . . . . . . . . . .  38
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  40

1.  Introduction

   Packet loss and delay are hard and pervasive problems of day-to-day
   network operation.  Proactively detecting, measuring, and locating
   them is crucial to maintaining high QoS and timely resolution of
   crippling end-to-end throughput issues.  To this effect, in a TCP-
   dominated world, network operators have been heavily relying on
   information present in the clear in TCP headers: sequence and
   acknowledgement numbers and SACKs when enabled (see [RFC8517]).
   These allow for quantitative estimation of packet loss and delay by
   passive on-path observation.  Additionally, the problem can be
   quickly identified in the network path by moving the passive observer
   around.

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   With encrypted protocols, the equivalent transport headers are
   encrypted and passive packet loss and delay observations are not
   possible, as described in [RFC9065].

   Measuring TCP loss and delay between similar endpoints cannot be
   relied upon to evaluate encrypted protocol loss and delay.  Different
   protocols could be routed by the network differently, and the
   fraction of Internet traffic delivered using protocols other than TCP
   is increasing every year.  It is imperative to measure packet loss
   and delay experienced by encrypted protocol users directly.

   This document defines Explicit Flow Measurement Techniques.  These
   hybrid measurement path signals (see [IPM-Methods]) are to be
   embedded into a transport layer protocol and are explicitly intended
   for exposing RTT and loss rate information to on-path measurement
   devices.  They are designed to facilitate network operations and
   management and are "beneficial" for maintaining the quality of
   service (see [RFC9065]).  These measurement mechanisms are applicable
   to any transport-layer protocol, and, as an example, the document
   describes QUIC and TCP bindings.

   The Explicit Flow Measurement Techniques described in this document
   can be used alone or in combination with other Explicit Flow
   Measurement Techniques.  Each technique uses a small number of bits
   and exposes a specific measurement.

   Following the recommendation in [RFC8558] of making path signals
   explicit, this document proposes adding a small number of dedicated
   measurement bits to the clear portion of the protocol headers.  These
   bits can be added to an unencrypted portion of a header belonging to
   any protocol layer, e.g.  IP (see [IP]) and IPv6 (see [IPv6]) headers
   or extensions, such as [IPv6AltMark], UDP surplus space (see
   [UDP-OPTIONS] and [UDP-SURPLUS]), reserved bits in a QUIC v1 header,
   as already done with the latency Spin bit (see [QUIC-TRANSPORT]).

   The measurements are not designed for use in automated control of the
   network in environments where signal bits are set by untrusted hosts.
   Instead, the signal is to be used for troubleshooting individual
   flows as well as for monitoring the network by aggregating
   information from multiple flows and raising operator alarms if
   aggregate statistics indicate a potential problem.

   The Spin bit, Delay bit and loss bits explained in this document are
   inspired by [AltMark], [SPIN-BIT], [I-D.trammell-tsvwg-spin] and
   [I-D.trammell-ippm-spin].

   Additional details about the Performance Measurements for QUIC are
   described in the paper [ANRW19-PM-QUIC].

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2.  Latency Bits

   This section introduces bits that can be used for round trip latency
   measurements.  Whenever this section of the specification refers to
   packets, it is referring only to packets with protocol headers that
   include the latency bits.

   [QUIC-TRANSPORT] introduces an explicit per-flow transport-layer
   signal for hybrid measurement of RTT.  This signal consists of a Spin
   bit that toggles once per RTT.  [SPIN-BIT] discusses an additional
   two-bit Valid Edge Counter (VEC) to compensate for loss and
   reordering of the Spin bit and increase fidelity of the signal in
   less than ideal network conditions.

   This document introduces a stand-alone single-bit delay signal that
   can be used by passive observers to measure the RTT of a network
   flow, avoiding the Spin bit ambiguities that arise as soon as network
   conditions deteriorate.

2.1.  Spin Bit

   This section is a small recap of the Spin bit working mechanism.  For
   a comprehensive explanation of the algorithm, please see [SPIN-BIT].

   The Spin bit is an Alternate-Marking [AltMark] generated signal,
   where the size of the alternation changes with the flight size each
   RTT.

   The latency Spin bit is a single bit signal that toggles once per
   RTT, enabling latency monitoring of a connection-oriented
   communication from intermediate observation points.

   A "spin period" is a set of packets with the same Spin bit value sent
   during one RTT time interval.  A "spin period value" is the value of
   the Spin bit shared by all packets in a spin period.

   The client and server maintain an internal per-connection spin value
   (i.e. 0 or 1) used to set the Spin bit on outgoing packets.  Both
   endpoints initialize the spin value to 0 when a new connection
   starts.  Then:

   *  when the client receives a packet with the packet number larger
      than any number seen so far, it sets the connection spin value to
      the opposite value contained in the received packet;

   *  when the server receives a packet with the packet number larger
      than any number seen so far, it sets the connection spin value to
      the same value contained in the received packet.

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   The computed spin value is used by the endpoints for setting the spin
   bit on outgoing packets.  This mechanism allows the endpoints to
   generate a square wave such that, by measuring the distance in time
   between pairs of consecutive edges observed in the same direction, a
   passive on-path observer can compute the round trip delay of that
   network flow.

   Spin bit enables round trip latency measurement by observing a single
   direction of the traffic flow.

   Note that packet reordering can cause spurious edges that require
   heuristics to correct.  The Spin bit performance deteriorates as soon
   as network impairments arise as explained in Section 2.2.

2.2.  Delay Bit

   The Delay bit has been designed to overcome accuracy limitations
   experienced by the Spin bit under difficult network conditions:

   *  packet reordering leads to generation of spurious edges and errors
      in delay estimation;

   *  loss of edges causes wrong estimation of spin periods and
      therefore wrong RTT measurements;

   *  application-limited senders cause the Spin bit to measure the
      application delays instead of network delays.

   Unlike the Spin bit, which is set in every packet transmitted on the
   network, the Delay bit is set only once per round trip.

   When the Delay bit is used, a single packet with a marked bit (the
   Delay bit) bounces between a client and a server during the entire
   connection lifetime.  This single packet is called "delay sample".

   An observer placed at an intermediate point, observing a single
   direction of traffic, tracking the delay sample and the relative
   timestamp, can measure the round trip delay of the connection.

   The delay sample lifetime is comprised of two phases: initialization
   and reflection.  The initialization is the generation of the delay
   sample, while the reflection realizes the bounce behavior of this
   single packet between the two endpoints.

   The next figure describes the elementary Delay bit mechanism.

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         +--------+   -   -   -   -   -   +--------+
         |        |      ----------->     |        |
         | Client |                       | Server |
         |        |     <-----------      |        |
         +--------+   -   -   -   -   -   +--------+

         (a) No traffic at beginning.

         +--------+   0   0   1   -   -   +--------+
         |        |      ----------->     |        |
         | Client |                       | Server |
         |        |     <-----------      |        |
         +--------+   -   -   -   -   -   +--------+

          (b) The Client starts sending data and
           sets the first packet as Delay Sample.

         +--------+   0   0   0   0   0   +--------+
         |        |      ----------->     |        |
         | Client |                       | Server |
         |        |     <-----------      |        |
         +--------+   -   -   -   1   0   +--------+

          (c) The Server starts sending data
           and reflects the Delay Sample.

         +--------+   0   1   0   0   0   +--------+
         |        |      ----------->     |        |
         | Client |                       | Server |
         |        |     <-----------      |        |
         +--------+   0   0   0   0   0   +--------+

         (d) The Client reflects the Delay Sample.

         +--------+   0   0   0   0   0   +--------+
         |        |      ----------->     |        |
         | Client |                       | Server |
         |        |     <-----------      |        |
         +--------+   0   0   0   1   0   +--------+

         (e) The Server reflects the Delay Sample
          and so on.

                       Figure 1: Delay bit mechanism

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2.2.1.  Generation Phase

   Only client is actively involved in the generation phase.  It
   maintains an internal per-flow timestamp variable (ds_time) updated
   every time a delay sample is transmitted.

   When connection starts, the client generates a new delay sample
   initializing the Delay bit of the first outgoing packet to 1.  Then
   it updates the ds_time variable with the timestamp of its
   transmission.

   The server initializes the Delay bit to 0 at the beginning of the
   connection, and its only task during the connection is described in
   Section 2.2.2.

   In absence of network impairments, the delay sample should bounce
   between client and server continuously, for the entire duration of
   the connection.  That is highly unlikely for two reasons:

   1.  the packet carrying the Delay bit might be lost;

   2.  an endpoint could stop or delay sending packets because the
       application is limiting the amount of traffic transmitted.

   To deal with these problems, the client generates a new delay sample
   if more than a predetermined time (T_Max) has elapsed since the last
   delay sample transmission (including reflections).  Note that T_Max
   should be greater than the max measurable RTT on the network.  See
   Section 2.2.3 for details.

2.2.2.  Reflection Phase

   Reflection is the process that enables the bouncing of the delay
   sample between a client and a server.  The behavior of the two
   endpoints is almost the same.

   *  Server side reflection: when a delay sample arrives, the server
      marks the first packet in the opposite direction as the delay
      sample.

   *  Client side reflection: when a delay sample arrives, the client
      marks the first packet in the opposite direction as the delay
      sample.  It also updates the ds_time variable when the outgoing
      delay sample is actually forwarded.

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   In both cases, if the outgoing delay sample is being transmitted with
   a delay greater than a predetermined threshold after the reception of
   the incoming delay sample (1ms by default), the delay sample is not
   reflected, and the outgoing Delay bit is kept at 0.

   By doing so, the algorithm can reject measurements that would
   overestimate the delay due to lack of traffic on the endpoints.
   Hence, the maximum estimation error would amount to twice the
   threshold (e.g. 2ms) per measurement.

2.2.3.  T_Max Selection

   The internal ds_time variable allows a client to identify delay
   sample losses.  Considering that a lost delay sample is regenerated
   at the end of an explicit time (T_Max) since the last generation,
   this same value can be used by an observer to reject a measure and
   start a new one.

   In other words, if the difference in time between two delay samples
   is greater or equal than T_Max, then these cannot be used to produce
   a delay measure.  Therefore the value of T_Max must also be known to
   the on-path network probes.

   There are two alternatives to select the T_Max value so that both
   client and observers know it.  The first one requires that T_Max is
   known a priori (T_Max_p) and therefore set within the protocol
   specifications that implements the marking mechanism (e.g. 1 second
   which usually is greater than the max expectable RTT).  The second
   alternative requires a dynamic mechanism able to adapt the duration
   of the T_Max to the delay of the connection (T_Max_c).

   For instance, client and observers could use the connection RTT as a
   basis for calculating an effective T_Max.  They should use a
   predetermined initial value so that T_Max = T_Max_p (e.g. 1 second)
   and then, when a valid RTT is measured, change T_Max accordingly so
   that T_Max = T_Max_c.  In any case, the selected T_Max should be
   large enough to absorb any possible variations in the connection
   delay.

   T_Max_c could be computed as two times the measured RTT plus a fixed
   amount of time (100ms) to prevent low T_Max values in case of very
   small RTTs.  The resulting formula is: T_Max_c = 2RTT + 100ms.  If
   T_Max_c is greater than T_Max_p then T_Max_c is forced to T_Max_p
   value.

   Note that the observer's T_Max should always be less than or equal to
   the client's T_Max to avoid considering as a valid measurement what
   is actually the client's T_Max.  To obtain this result, the client

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   waits for two consecutive incoming samples and computes the two
   related RTTs.  Then it takes the largest of them as the basis of the
   T_Max_c formula.  At this point, observers have already measured a
   valid RTT and then computed their T_Max_c.

2.2.4.  Delay Measurement Using Delay Bit

   When the Delay bit is used, a passive observer can use delay samples
   directly and avoid inherent ambiguities in the calculation of the RTT
   as can be seen in Spin bit analysis.

2.2.4.1.  RTT Measurement

   The delay sample generation process ensures that only one packet
   marked with the Delay bit set to 1 runs back and forth between two
   endpoints per round trip time.  To determine the RTT measurement of a
   flow, an on-path passive observer computes the time difference
   between two delay samples observed in a single direction.

   To ensure a valid measurement, the observer must verify that the
   distance in time between the two samples taken into account is less
   than T_Max.

              =======================|======================>
              = **********     -----Obs---->     ********** =
              = * Client *                       * Server * =
              = **********     <------------     ********** =
              <==============================================

                        (a) client-server RTT

              ==============================================>
              = **********     ------------>     ********** =
              = * Client *                       * Server * =
              = **********     <----Obs-----     ********** =
              <======================|=======================

                        (b) server-client RTT

                 Figure 2: Round-trip time (both direction)

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2.2.4.2.  Half-RTT Measurement

   An observer that is able to observe both forward and return traffic
   directions can use the delay samples to measure "upstream" and
   "downstream" RTT components, also known as the half-RTT measurements.
   It does this by measuring the time between a delay sample observed in
   one direction and the delay sample previously observed in the
   opposite direction.

   As with RTT measurement, the observer must verify that the distance
   in time between the two samples taken into account is less than
   T_Max.

   Note that upstream and downstream sections of paths between the
   endpoints and the observer, i.e. observer-to-client vs client-to-
   observer and observer-to-server vs server-to-observer, may have
   different delay characteristics due to the difference in network
   congestion and other factors.

              =======================>
              = **********     ------|----->     **********
              = * Client *          Obs          * Server *
              = **********     <-----|------     **********
              <=======================

                     (a) client-observer half-RTT

                                     =======================>
                **********     ------|----->     ********** =
                * Client *          Obs          * Server * =
                **********     <-----|------     ********** =
                                     <=======================

                     (b) observer-server half-RTT

              Figure 3: Half Round-trip time (both direction)

2.2.4.3.  Intra-Domain RTT Measurement

   Intra-domain RTT is the portion of the entire RTT used by a flow to
   traverse the network of a provider.  To measure intra-domain RTT, two
   observers capable of observing traffic in both directions must be
   employed simultaneously at ingress and egress of the network to be
   measured.  Intra-domain RTT is difference between the two computed
   upstream (or downstream) RTT components.

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           =========================================>
           = =====================>
           = = **********      ---|-->           ---|-->      **********
           = = * Client *         Obs               Obs       * Server *
           = = **********      <--|---           <--|---      **********
           = <=====================
           <=========================================

                    (a) client-observer RTT components (half-RTTs)

                                  ==================>
               **********      ---|-->           ---|-->      **********
               * Client *         Obs               Obs       * Server *
               **********      <--|---           <--|---      **********
                                  <==================

                    (b) the intra-domain RTT resulting from the
                    subtraction of the above RTT components

     Figure 4: Intra-domain Round-trip time (client-observer: upstream)

2.2.5.  Observer's Algorithm

   An on-path observer maintains an internal per-flow variable to keep
   track of time at which the last delay sample has been observed.

   A unidirectional observer, upon detecting a delay sample:

   *  if a delay sample was also detected previously in the same
      direction and the distance in time between them is less than T_Max
      - K, then the two delay samples can be used to calculate RTT
      measurement.  K is a protection threshold to absorb differences in
      T_Max computation and delay variations between two consecutive
      delay samples (e.g.  K = 10% T_Max).

   If the observer can observe both forward and return traffic flows,
   and it is able to determine which direction contains the client and
   the server (e.g. by observing the connection handshake), upon
   detecting a delay sample:

   *  if a delay sample was also detected in the opposite direction and
      the distance in time between them is less than T_Max - K, then the
      two delay samples can be used to measure the observer-client half-
      RTT or the observer-server half-RTT, according to the direction of
      the last delay sample observed.

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2.2.6.  Two Bits Delay Measurement: Spin Bit + Delay Bit

   Spin and Delay bit algorithms work independently.  If both marking
   methods are used in the same connection, observers can choose the
   best measurement between the two available:

   *  when a precise measurement can be produced using the Delay bit,
      observers choose it;

   *  when a Delay bit measurement is not available, observers choose
      the approximate Spin bit one.

3.  Loss Bits

   This section introduces bits that can be used for loss measurements.
   Whenever this section of the specification refers to packets, it is
   referring only to packets with protocol headers that include the loss
   bits -- the only packets whose loss can be measured.

   *  T: the "round Trip loss" bit is used in combination with the Spin
      bit to measure round-trip loss.  See Section 3.1.

   *  Q: the "sQuare signal" bit is used to measure upstream loss.  See
      Section 3.2.

   *  L: the "Loss event" bit is used to measure end-to-end loss.  See
      Section 3.3.

   *  R: the "Reflection square signal" bit is used in combination with
      Q bit to measure end-to-end loss.  See Section 3.1.

   Loss measurements enabled by T, Q, and L bits can be implemented by
   those loss bits alone (T bit requires a working Spin bit).  Two-bit
   combinations Q+L and Q+R enable additional measurement opportunities
   discussed below.

   Each endpoint maintains appropriate counters independently and
   separately for each separately identifiable flow (each sub-flow for
   multipath connections).

   Since loss is reported independently for each flow, all bits (except
   for L bit) require a certain minimum number of packets to be
   exchanged per flow before any signal can be measured.  Therefore,
   loss measurements work best for flows that transfer more than a
   minimal amount of data.

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3.1.  T Bit -- Round Trip Loss Bit

   The round Trip loss bit is used to mark a variable number of packets
   exchanged twice between the endpoints realizing a two round-trip
   reflection.  A passive on-path observer, observing either direction,
   can count and compare the number of marked packets seen during the
   two reflections, estimating the loss rate experienced by the
   connection.  The overall exchange comprises:

   *  the client selects, generates and consequently transmits a first
      train of packets, by setting the T bit to 1;

   *  the server, upon receiving each packet included in the first
      train, reflects to the client a respective second train of packets
      of the same size as the first train received, by setting the T bit
      to 1;

   *  the client, upon receiving each packet included in the second
      train, reflects to the server a respective third train of packets
      of the same size as the second train received, by setting the T
      bit to 1;

   *  the server, upon receiving each packet included in the third
      train, finally reflects to the client a respective fourth train of
      packets of the same size as the third train received, by setting
      the T bit to 1.

   Packets belonging to the first round trip (first and second train)
   represent the Generation Phase, while those belonging to the second
   round trip (third and fourth train) represent the Reflection Phase.

   A passive on-path observer can count and compare the number of marked
   packets seen during the two round trips (i.e. the first and third or
   the second and the fourth trains of packets, depending on which
   direction is observed) and estimate the loss rate experienced by the
   connection.  This process is repeated continuously to obtain more
   measurements as long as the endpoints exchange traffic.  These
   measurements can be called Round Trip losses.

   Since packet rates in two directions may be different, the number of
   marked packets in the train is determined by the direction with the
   lowest packet rate.  See Section 3.1.2 for details on packet
   generation.

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3.1.1.  Round Trip Loss

   Since the measurements are performed on a portion of the traffic
   exchanged between the client and the server, the observer calculates
   the end-to-end Round Trip Packet Loss (RTPL) that, statistically,
   will correspond to the loss rate experienced by the connection along
   the entire network path.

              =======================|======================>
              = **********     -----Obs---->     ********** =
              = * Client *                       * Server * =
              = **********     <------------     ********** =
              <==============================================

                        (a) client-server RTPL

              ==============================================>
              = **********     ------------>     ********** =
              = * Client *                       * Server * =
              = **********     <----Obs-----     ********** =
              <======================|=======================

                        (b) server-client RTPL

             Figure 5: Round-trip packet loss (both direction)

   This methodology also allows the Half-RTPL measurement and the Intra-
   domain RTPL measurement in a way similar to RTT measurement.

              =======================>
              = **********     ------|----->     **********
              = * Client *          Obs          * Server *
              = **********     <-----|------     **********
              <=======================

                     (a) client-observer half-RTPL

                                     =======================>
                **********     ------|----->     ********** =
                * Client *          Obs          * Server * =
                **********     <-----|------     ********** =
                                     <=======================

                     (b) observer-server half-RTPL

           Figure 6: Half Round-trip packet loss (both direction)

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                              =========================================>
                                                =====================> =
           **********      ---|-->           ---|-->      ********** = =
           * Client *         Obs               Obs       * Server * = =
           **********      <--|---           <--|---      ********** = =
                                                <===================== =
                              <=========================================

                (a) observer-server RTPL components (half-RTPLs)

                              ==================>
           **********      ---|-->           ---|-->      **********
           * Client *         Obs               Obs       * Server *
           **********      <--|---           <--|---      **********
                              <==================

                (b) the intra-domain RTPL resulting from the
                subtraction of the above RTPL components

      Figure 7: Intra-domain Round-trip packet loss (observer-server)

3.1.2.  Setting the Round Trip Loss Bit on Outgoing Packets

   The round Trip loss signal requires a working Spin-bit signal to
   separate trains of marked packets (packets with T bit set to 1).  A
   "pause" of at least one empty spin-bit period between each phase of
   the algorithm serves as such separator for the on-path observer.

   The client maintains a "generation token" count that is set to zero
   at the beginning of the session and is incremented every time a
   packet is received (marked or unmarked).  The client also maintains a
   "reflection counter" that starts at zero at the beginning of the
   session.

   The client is in charge of launching trains of marked packets and
   does so according to the algorithm:

   1.  Generation Phase.  The client starts generating marked packets
       for two consecutive spin-bit periods; when the client transmits a
       packet and a "generation token" is available, the client marks
       the packet and retires a "generation token".  If no token is
       available, the outgoing packet is transmitted unmarked.  At the
       end of the first spin-bit period spent in generation, the
       reflection counter is unlocked to start counting incoming marked
       packets that will be reflected later;

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   2.  Pause Phase.  When the generation is completed, the client pauses
       till it has observed one entire Spin bit period with no marked
       packets.  That Spin bit period is used by the observer as a
       separator between generated and reflected packets.  During this
       marking pause, all the outgoing packets are transmitted with T
       bit set to 0.  The reflection counter is still incremented every
       time a marked packet arrives;

   3.  Reflection Phase.  The client starts transmitting marked packets,
       decrementing the reflection counter for each transmitted marked
       packet until the reflection counter reached zero.  The
       "generation token" method from the generation phase is used
       during this phase as well.  At the end of the first spin-period
       spent in reflection, the reflection counter is locked to avoid
       incoming reflected packets incrementing it;

   4.  Pause Phase 2.  The pause phase is repeated after the reflection
       phase and serves as a separator between the reflected packet
       train and a new packet train.

   The generation token counter should be capped to limit the effects of
   a subsequent sudden reduction in the other endpoint's packet rate
   that could prevent that endpoint from reflecting collected packets.
   The most conservative cap value is 1.

   A server maintains a "marking counter" that starts at zero and is
   incremented every time a marked packet arrives.  When the server
   transmits a packet and the "marking counter" is positive, the server
   marks the packet and decrements the "marking counter".  If the
   "marking counter" is zero, the outgoing packet is transmitted
   unmarked.

   Note that a choice of 2-RTT (two spin periods) for the generation
   phase is a tradeoff between the percentage of marked packets (i.e.
   the percentage of traffic monitored) and the measurement delay.
   Using this value the algorithm produces a measurement approximately
   every 6-RTT (2 generation, ~2 reflection, 2 pauses), marking ~1/3 of
   packets exchanged in the slower direction (see Section 3.1.4).
   Choosing a generation phase of 1-RTT, we would produce measurements
   every 4-RTT, monitoring just ~1/4 of packets in the slower direction.

3.1.3.  Observer's Logic for Round Trip Loss Signal

   The on-path observer counts marked packets and separates different
   trains by detecting spin-bit periods (at least one) with no marked
   packets.  The Round Trip Packet Loss (RTPL) is the difference between
   the size of the Generation train and the Reflection train.

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   In the following example, packets are represented by two bits (first
   one is the Spin bit, second one is the round Trip loss bit):

           Generation          Pause           Reflection       Pause
      ____________________ ______________ ____________________ ________
     |                    |              |                    |        |
      01 01 00 01 11 10 11 00 00 10 10 10 01 00 01 01 10 11 10 00 00 10

                  Figure 8: Round Trip Loss signal example

   Note that 5 marked packets have been generated of which 4 have been
   reflected.

3.1.4.  Loss Coverage and Signal Timing

   A cycle of the round Trip loss signaling algorithm contains 2 RTTs of
   Generation phase, 2 RTTs of Reflection phase, and two Pause phases at
   least 1 RTT in duration each.  Hence, the loss signal is delayed by
   about 6 RTTs since the loss events.

   The observer can only detect loss of marked packets that occurs after
   its initial observation of the Generation phase and before its
   subsequent observation of the Reflection phase.  Hence, if the loss
   occurs on the path that sends packets at a lower rate (typically ACKs
   in such asymmetric scenarios), 2/6 (1/3) of the packets will be
   sampled for loss detection.

   If the loss occurs on the path that sends packets at a higher rate,
   lowPacketRate/(3*highPacketRate) of the packets will be sampled for
   loss detection.  For protocols that use ACKs, the portion of packets
   sampled for loss in the higher rate direction during unidirectional
   data transfer is 1/(3*packetsPerAck), where the value of
   packetsPerAck can vary by protocol, by implementation, and by network
   conditions.

3.2.  Q Bit -- sQuare Bit

   The sQuare bit (Q bit) takes its name from the square wave generated
   by its signal.  This method is based on the Alternate-Marking method
   [AltMark] and the Q bit represents the "packet color" that allows to
   mark consecutive blocks of packets with different colors.

   [AltMark] introduces two variations of the Alternate-Marking method
   depending on whether the color is switched according to a fixed timer
   or after a fixed number of packets.  The method based on fixed timer
   can measure packet loss on a network segment by cooperating and
   synchronized observers on both ends of the segment comparing packets
   counters for the same packet blocks.  The time length of the blocks

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   can be chosen depending on the desired measurement frequency, but it
   must be long enough to guarantee the proper operation with respect to
   clock errors and network delay issues.

   The Q bit method described in this document chooses the color-
   switching method based on a fixed number of packets for each block.
   This approach has the advantage that it does not require cooperating
   or synchronized observers or network elements.  Each probe can
   measure packet loss autonomously without relying on an external
   Network Management System (NMS).  For the purpose of the packet loss
   measurement, all blocks have the same number of packets, and it is
   necessary to detect only the loss event and not to identify the exact
   block with losses.

   Following the method based on fixed number of packets, the square
   wave signal is generated by the switching of the Q bit: every
   outgoing packet contains the Q bit value, which is initialized to 0
   and inverted after sending N packets (a sQuare Block or simply Q
   Block).  Hence, Q Period is 2*N.

   Observation points can estimate upstream losses by watching a single
   direction of the traffic flow and counting the number of packets in
   each observed Q Block, as described in Section 3.2.2.

3.2.1.  Q Block Length Selection

   The length of the block must be known to the on-path network probes.
   There are two alternatives to selecting the Q Block length.  The
   first one requires that the length is known a priori and therefore
   set within the protocol specifications that implements the marking
   mechanism.  The second requires the sender to select it.

   In this latter scenario, the sender is expected to choose N (Q Block
   length) based on the expected amount of loss and reordering on the
   path.  The choice of N strikes a compromise -- the observation could
   become too unreliable in case of packet reordering and/or severe loss
   if N is too small, while short flows may not yield a useful upstream
   loss measurement if N is too large (see Section 3.2.2).

   The value of N should be at least 64 and be a power of 2.  This
   requirement allows an Observer to infer the Q Block length by
   observing one period of the square signal.  It also allows the
   Observer to identify flows that set the loss bits to arbitrary values
   (see Section 5).

   If the sender does not have sufficient information to make an
   informed decision about Q Block length, the sender should use N=64,
   since this value has been extensively tried in large-scale field

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   tests and yielded good results.  Alternatively, the sender may also
   choose a random power-of-2 N for each flow, increasing the chances of
   using a Q Block length that gives the best signal for some flows.

   The sender must keep the value of N constant for a given flow.

3.2.2.  Upstream Loss

   Blocks of N (Q Block length) consecutive packets are sent with the
   same value of the Q bit, followed by another block of N packets with
   an inverted value of the Q bit.  Hence, knowing the value of N, an
   on-path observer can estimate the amount of upstream loss after
   observing at least N packets.  The upstream loss rate (uloss) is one
   minus the average number of packets in a block of packets with the
   same Q value (p) divided by N (uloss=1-avg(p)/N).

   The observer needs to be able to tolerate packet reordering that can
   blur the edges of the square signal, as explained in Section 3.2.3.

             =====================>
             **********     -----Obs---->     **********
             * Client *                       * Server *
             **********     <------------     **********

               (a) in client-server channel (uloss_up)

             **********     ------------>     **********
             * Client *                       * Server *
             **********     <----Obs-----     **********
                                  <=====================

               (b) in server-client channel (uloss_down)

                          Figure 9: Upstream loss

3.2.3.  Identifying Q Block Boundaries

   Packet reordering can produce spurious edges in the square signal.
   To address this, the observer should look for packets with the
   current Q bit value up to X packets past the first packet with a
   reverse Q bit value.  The value of X, a "Marking Block Threshold",
   should be less than N/2.

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   The choice of X represents a trade-off between resiliency to
   reordering and resiliency to loss.  A very large Marking Block
   Threshold will be able to reconstruct Q Blocks despite a significant
   amount of reordering, but it may erroneously coalesce packets from
   multiple Q Blocks into fewer Q Blocks, if loss exceeds 50% for some Q
   Blocks.

3.2.3.1.  Improved Resilience to Burst Losses

   Burst losses can affect Q measurements accuracy.  Generally, burst
   losses can be absorbed and correctly measured if smaller than the
   established Q Block length.  If entire Q Block length of packets get
   lost in a burst, however, the observer may be left completely unaware
   of the loss.

   To improve burst loss resilience, an observer may consider a received
   Q Block larger than the selected Q Block length as an indication of a
   burst loss event.  The observer would then compute the loss as three
   times Q Block length minus the measured block length.  By doing so,
   the observer can detect burst losses of less than two blocks (e.g.,
   less than 128 packets for Q Block length of 64 packets).  A burst
   loss of two or more consecutive periods would still remain unnoticed
   by the observer (or underestimated if a period longer than Q Block
   length were formed).

3.3.  L Bit -- Loss Event Bit

   The Loss Event bit uses an Unreported Loss counter maintained by the
   protocol that implements the marking mechanism.  To use the Loss
   Event bit, the protocol must allow the sender to identify lost
   packets.  This is true of protocols such as QUIC, partially true for
   TCP and SCTP (losses of pure ACKs are not detected) and is not true
   of protocols such as UDP and IP/IPv6.

   The Unreported Loss counter is initialized to 0, and L bit of every
   outgoing packet indicates whether the Unreported Loss counter is
   positive (L=1 if the counter is positive, and L=0 otherwise).

   The value of the Unreported Loss counter is decremented every time a
   packet with L=1 is sent.

   The value of the Unreported Loss counter is incremented for every
   packet that the protocol declares lost, using whatever loss detection
   machinery the protocol employs.  If the protocol is able to rescind
   the loss determination later, a positive Unreported Loss counter may
   be decremented due to the rescission, but it should not become
   negative due to the rescission.

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   This loss signaling is similar to loss signaling in [ConEx], except
   the Loss Event bit is reporting the exact number of lost packets,
   whereas Echo Loss bit in [ConEx] is reporting an approximate number
   of lost bytes.

   For protocols, such as TCP ([TCP]), that allow network devices to
   change data segmentation, it is possible that only a part of the
   packet is lost.  In these cases, the sender must increment Unreported
   Loss counter by the fraction of the packet data lost (so Unreported
   Loss counter may become negative when a packet with L=1 is sent after
   a partial packet has been lost).

   Observation points can estimate the end-to-end loss, as determined by
   the upstream endpoint, by counting packets in this direction with the
   L bit equal to 1, as described in Section 3.3.1.

3.3.1.  End-To-End Loss

   The Loss Event bit allows an observer to estimate the end-to-end loss
   rate by counting packets with L bit value of 0 and 1 for a given
   flow.  The end-to-end loss rate is the fraction of packets with L=1.

   The assumption here is that upstream loss affects packets with L=0
   and L=1 equally.  If some loss is caused by tail-drop in a network
   device, this may be a simplification.  If the sender's congestion
   controller reduces the packet send rate after loss, there may be a
   sufficient delay before sending packets with L=1 that they have a
   greater chance of arriving at the observer.

3.3.1.1.  Loss Profile Characterization

   The Loss Event bit allows an observer to characterize loss profile,
   since the distribution of observed packets with L bit set to 1
   roughly corresponds to the distribution of packets lost between 1 RTT
   and 1 RTO before (see Section 3.3.2.1).  Hence, observing random
   single instances of L bit set to 1 indicates random single packet
   loss, while observing blocks of packets with L bit set to 1 indicates
   loss affecting entire blocks of packets.

3.3.2.  L+Q Bits -- Loss Measurement Using L and Q Bits

   Combining L and Q bits allows a passive observer watching a single
   direction of traffic to accurately measure:

   *  upstream loss: sender-to-observer loss (see Section 3.2.2)

   *  downstream loss: observer-to-receiver loss (see Section 3.3.2.2)

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   *  end-to-end loss: sender-to-receiver loss on the observed path (see
      Section 3.3.1) with loss profile characterization (see
      Section 3.3.1.1)

3.3.2.1.  Correlating End-to-End and Upstream Loss

   Upstream loss is calculated by observing packets that did not suffer
   the upstream loss (Section 3.2.2).  End-to-end loss, however, is
   calculated by observing subsequent packets after the sender's
   protocol detected the loss.  Hence, end-to-end loss is generally
   observed with a delay of between 1 RTT (loss declared due to multiple
   duplicate acknowledgements) and 1 RTO (loss declared due to a
   timeout) relative to the upstream loss.

   The flow RTT can sometimes be estimated by timing protocol handshake
   messages.  This RTT estimate can be greatly improved by observing a
   dedicated protocol mechanism for conveying RTT information, such as
   the Spin bit (see Section 2.1) or Delay bit (see Section 2.2).

   Whenever the observer needs to perform a computation that uses both
   upstream and end-to-end loss rate measurements, it should use
   upstream loss rate leading the end-to-end loss rate by approximately
   1 RTT.  If the observer is unable to estimate RTT of the flow, it
   should accumulate loss measurements over time periods of at least 4
   times the typical RTT for the observed flows.

   If the calculated upstream loss rate exceeds the end-to-end loss rate
   calculated in Section 3.3.1, then either the Q Period is too short
   for the amount of packet reordering or there is observer loss,
   described in Section 3.3.2.3.  If this happens, the observer should
   adjust the calculated upstream loss rate to match end-to-end loss
   rate, unless the following applies.

   In case of a protocol, such as TCP or SCTP, that does not track
   losses of pure ACK packets, observing a direction of traffic
   dominated by pure ACK packets could result in measured upstream loss
   that is higher than measured end-to-end loss, if said pure ACK
   packets are lost upstream.  Hence, if the measurement is applied to
   such protocols, and the observer can confirm that pure ACK packets
   dominate the observed traffic direction, the observer should adjust
   the calculated end-to-end loss rate to match upstream loss rate.

3.3.2.2.  Downstream Loss

   Because downstream loss affects only those packets that did not
   suffer upstream loss, the end-to-end loss rate (eloss) relates to the
   upstream loss rate (uloss) and downstream loss rate (dloss) as
   (1-uloss)(1-dloss)=1-eloss.  Hence, dloss=(eloss-uloss)/(1-uloss).

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3.3.2.3.  Observer Loss

   A typical deployment of a passive observation system includes a
   network tap device that mirrors network packets of interest to a
   device that performs analysis and measurement on the mirrored
   packets.  The observer loss is the loss that occurs on the mirror
   path.

   Observer loss affects upstream loss rate measurement, since it causes
   the observer to account for fewer packets in a block of identical Q
   bit values (see Section 3.2.2).  The end-to-end loss rate
   measurement, however, is unaffected by the observer loss, since it is
   a measurement of the fraction of packets with the L bit value of 1,
   and the observer loss would affect all packets equally (see
   Section 3.3.1).

   The need to adjust the upstream loss rate down to match end-to-end
   loss rate as described in Section 3.3.2.1 is an indication of the
   observer loss, whose magnitude is between the amount of such
   adjustment and the entirety of the upstream loss measured in
   Section 3.2.2.  Alternatively, a high apparent upstream loss rate
   could be an indication of significant packet reordering, possibly due
   to packets belonging to a single flow being multiplexed over several
   upstream paths with different latency characteristics.

3.4.  R Bit -- Reflection Square Bit

   R bit requires a deployment alongside Q bit.  Unlike the square
   signal for which packets are transmitted in blocks of fixed size, the
   number of packets in Reflection square signal blocks (also an
   Alternate-Marking signal) varies according to these rules:

   *  when the transmission of a new block starts, its size is set equal
      to the size of the last Q Block whose reception has been
      completed;

   *  if, before transmission of the block is terminated, the reception
      of at least one further Q Block is completed, the size of the
      block is updated to be the average size of the further received Q
      Blocks.

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   The Reflection square value is initialized to 0 and is applied to the
   R bit of every outgoing packet.  The Reflection square value is
   toggled for the first time when the completion of a Q Block is
   detected in the incoming square signal (produced by the other
   endpoint using the Q bit).  The number of packets detected within
   this first Q Block (p), is used to generate a reflection square
   signal that toggles every M=p packets (at first).  This new signal
   produces blocks of M packets (marked using the R bit) and each of
   them is called "Reflection Block" (R Block).

   The M value is then updated every time a completed Q Block in the
   incoming square signal is received, following this formula:
   M=round(avg(p)).

   The parameter avg(p), the average number of packets in a marking
   period, is computed based on all the Q Blocks received since the
   beginning of the current R Block.

   The transmission of an R Block is considered complete (and the signal
   toggled) when the number of packets transmitted in that block is at
   least the latest computed M value.

   To ensure a proper computation of the M value, endpoints implementing
   the R bit must identify the boundaries of incoming Q Blocks.  The
   same approach described in Section 3.2.3 should be used.

   Looking at the R bit, unidirectional observation points have an
   indication of loss experienced by the entire unobserved channel plus
   the loss on the path from the sender to them.

   Since the Q Block is sent in one direction, and the corresponding
   reflected R Block is sent in the opposite direction, the reflected R
   signal is transmitted with the packet rate of the slowest direction.
   Namely, if the observed direction is the slowest, there can be
   multiple Q Blocks transmitted in the unobserved direction before a
   complete R Block is transmitted in the observed direction.  If the
   unobserved direction is the slowest, the observed direction can be
   sending R Blocks of the same size repeatedly before it can update the
   signal to account for a newly-completed Q Block.

3.4.1.  Enhancement of R Block Length Computation

   The use of the rounding function used in the M computation introduces
   errors that can be minimized by storing the rounding applied each
   time M is computed, and using it during the computation of the M
   value in the following R Block.

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   This can be achieved introducing the new r_avg parameter in the
   computation of M.  The new formula is Mr=avg(p)+r_avg; M=round(Mr);
   r_avg=Mr-M where the initial value of r_avg is equal to 0.

3.4.2.  Improved Resilience to Packet Reordering

   When a protocol implementing the marking mechanism is able to detect
   when packets are received out of order, it can improve resilience to
   packet reordering beyond what is possible using methods described in
   Section 3.2.3.

   This can be achieved by updating the size of the current R Block
   while it is being transmitted.  The reflection block size is then
   updated every time an incoming reordered packet of the previous Q
   Block is detected.  This can be done if and only if the transmission
   of the current reflection block is in progress and no packets of the
   following Q Block have been received.

3.4.2.1.  Improved Resilience to Burst Losses

   Burst losses can affect R measurements accuracy similarly to how they
   affect Q measurements accuracy.  Therefore, recommendations in
   section Section 3.2.3.1 apply equally to improving burst loss
   resilience for R measurements.

3.4.3.  R+Q Bits -- Loss Measurement Using R and Q Bits

   Since both sQuare and Reflection square bits are toggled at most
   every N packets (except for the first transition of the R bit as
   explained before), an on-path observer can count the number of
   packets of each marking block and, knowing the value of N, can
   estimate the amount of loss experienced by the connection.  An
   observer can calculate different measurements depending on whether it
   is able to observe a single direction of the traffic or both
   directions.

   Single directional observer:

   *  upstream loss in the observed direction: the loss between the
      sender and the observation point (see Section 3.2.2)

   *  "three-quarters" connection loss: the loss between the receiver
      and the sender in the unobserved direction plus the loss between
      the sender and the observation point in the observed direction

   *  end-to-end loss in the unobserved direction: the loss between the
      receiver and the sender in the opposite direction

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   Two directions observer (same metrics seen previously applied to both
   direction, plus):

   *  client-observer half round-trip loss: the loss between the client
      and the observation point in both directions

   *  observer-server half round-trip loss: the loss between the
      observation point and the server in both directions

   *  downstream loss: the loss between the observation point and the
      receiver (applicable to both directions)

3.4.3.1.  Three-Quarters Connection Loss

   Except for the very first block in which there is nothing to reflect
   (a complete Q Block has not been yet received), packets are
   continuously R-bit marked into alternate blocks of size lower or
   equal than N.  Knowing the value of N, an on-path observer can
   estimate the amount of loss occurred in the whole opposite channel
   plus the loss from the sender up to it in the observation channel.
   As for the previous metric, the three-quarters connection loss rate
   (tqloss) is one minus the average number of packets in a block of
   packets with the same R value (t) divided by N (tqloss=1-avg(t)/N).

           =======================>
           = **********     -----Obs---->     **********
           = * Client *                       * Server *
           = **********     <------------     **********
           <============================================

               (a) in client-server channel (tqloss_up)

             ============================================>
             **********     ------------>     ********** =
             * Client *                       * Server * =
             **********     <----Obs-----     ********** =
                                  <=======================

               (b) in server-client channel (tqloss_down)

                 Figure 10: Three-quarters connection loss

   The following metrics derive from this last metric and the upstream
   loss produced by the Q bit.

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3.4.3.2.  End-To-End Loss in the Opposite Direction

   End-to-end loss in the unobserved direction (eloss_unobserved)
   relates to the "three-quarters" connection loss (tqloss) and upstream
   loss in the observed direction (uloss) as
   (1-eloss_unobserved)(1-uloss)=1-tqloss.  Hence,
   eloss_unobserved=(tqloss-uloss)/(1-uloss).

             **********     -----Obs---->     **********
             * Client *                       * Server *
             **********     <------------     **********
             <==========================================

               (a) in client-server channel (eloss_down)

             ==========================================>
             **********     ------------>     **********
             * Client *                       * Server *
             **********     <----Obs-----     **********

               (b) in server-client channel (eloss_up)

            Figure 11: End-To-End loss in the opposite direction

3.4.3.3.  Half Round-Trip Loss

   If the observer is able to observe both directions of traffic, it is
   able to calculate two "half round-trip" loss measurements -- loss
   from the observer to the receiver (in a given direction) and then
   back to the observer in the opposite direction.  For both directions,
   "half round-trip" loss (hrtloss) relates to "three-quarters"
   connection loss (tqloss_opposite) measured in the opposite direction
   and the upstream loss (uloss) measured in the given direction as
   (1-uloss)(1-hrtloss)=1-tqloss_opposite.  Hence,
   hrtloss=(tqloss_opposite-uloss)/(1-uloss).

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           =======================>
           = **********     ------|----->     **********
           = * Client *          Obs          * Server *
           = **********     <-----|------     **********
           <=======================

         (a) client-observer half round-trip loss (hrtloss_co)

                                  =======================>
             **********     ------|----->     ********** =
             * Client *          Obs          * Server * =
             **********     <-----|------     ********** =
                                  <=======================

         (b) observer-server half round-trip loss (hrtloss_os)

              Figure 12: Half Round-trip loss (both direction)

3.4.3.4.  Downstream Loss

   If the observer is able to observe both directions of traffic, it is
   able to calculate two downstream loss measurements using either end-
   to-end loss and upstream loss, similar to the calculation in
   Section 3.3.2.2 or using "half round-trip" loss and upstream loss in
   the opposite direction.

   For the latter, dloss=(hrtloss-uloss_opposite)/(1-uloss_opposite).

                                  =====================>
             **********     ------|----->     **********
             * Client *          Obs          * Server *
             **********     <-----|------     **********

                (a) in client-server channel (dloss_up)

             **********     ------|----->     **********
             * Client *          Obs          * Server *
             **********     <-----|------     **********
             <=====================

                (b) in server-client channel (dloss_down)

                         Figure 13: Downstream loss

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3.5.  E Bit -- ECN-Echo Event Bit

   While the primary focus of this draft is on exposing packet loss and
   delay, modern networks can report congestion before they are forced
   to drop packets, as described in [ECN].  When transport protocols
   keep ECN-Echo feedback under encryption, this signal cannot be
   observed by the network operators.  When tasked with diagnosing
   network performance problems, knowledge of a congestion downstream of
   an observation point can be instrumental.

   If downstream congestion information is desired, this information can
   be signaled with an additional bit.

   *  E: The "ECN-Echo Event" bit is set to 0 or 1 according to the
      Unreported ECN Echo counter, as explained below in Section 3.5.1.

3.5.1.  Setting the ECN-Echo Event Bit on Outgoing Packets

   The Unreported ECN-Echo counter operates identically to Unreported
   Loss counter (Section 3.3), except it counts packets delivered by the
   network with CE markings, according to the ECN-Echo feedback from the
   receiver.

   This ECN-Echo signaling is similar to ECN signaling in [ConEx].  ECN-
   Echo mechanism in QUIC provides the number of packets received with
   CE marks.  For protocols like TCP, the method described in
   [ConEx-TCP] can be employed.  As stated in [ConEx-TCP], such feedback
   can be further improved using a method described in [ACCURATE].

3.5.2.  Using E Bit for Passive ECN-Reported Congestion Measurement

   A network observer can count packets with CE codepoint and determine
   the upstream CE-marking rate directly.

   Observation points can also estimate ECN-reported end-to-end
   congestion by counting packets in this direction with a E bit equal
   to 1.

   The upstream CE-marking rate and end-to-end ECN-reported congestion
   can provide information about downstream CE-marking rate.  Presence
   of E bits along with L bits, however, can somewhat confound precise
   estimates of upstream and downstream CE-markings in case the flow
   contains packets that are not ECN-capable.

4.  Summary of Delay and Loss Marking Methods

   This section summarizes the marking methods described in this draft.

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   For the Delay measurement, it is possible to use the Spin bit and/or
   the delay bit.  A unidirectional or bidirectional observer can be
   used.

    +---------------+----+------------------------+--------------------+
    | Method        |# of|        Available       |             | # of |
    |               |bits|      Delay Metrics     | Impairments | meas.|
    |               |    +------------+-----------+ Resiliency  |      |
    |               |    |   UniDir   |   BiDir   |             |      |
    |               |    |  Observer  |  Observer |             |      |
    +---------------+----+------------+-----------+-------------+------+
    |S: Spin Bit    | 1  | RTT        | x2        | low         | very |
    |               |    |            | Half RTT  |             | high |
    +---------------+----+------------+-----------+-------------+------+
    |D: Delay Bit   | 1  | RTT        | x2        | high        |medium|
    |               |    |            | Half RTT  |             |      |
    +---------------+----+------------+-----------+-------------+------+
    |SD: Spin Bit & | 2  | RTT        | x2        | high        | very |
    |    Delay Bit *|    |            | Half RTT  |             | high |
    +---------------+----+------------+-----------+-------------+------+

    x2 Same metric for both directions
    *  Both bits work independently; an observer could use less accurate
       Spin bit measurements when Delay bit ones are unavailable

                        Figure 14: Delay Comparison

   For the Loss measurement, each row in the table of Figure 15
   represents a loss marking method.  For each method the table
   specifies the number of bits required in the header, the available
   metrics using an unidirectional or bidirectional observer, applicable
   protocols, measurement fidelity and delay.

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    +-------------+-+-----------------------+-+------------------------+
    | Method      |B|        Available      |P|  Measurement Aspects   |
    |             |i|      Loss Metrics     |r+------------+-----------+
    |             |t|   UniDir  |   BiDir   |t|  Fidelity  |   Delay   |
    |             |s|  Observer |  Observer |o|            |           |
    +-------------+-+-----------+-----------+-+------------+-----------+
    |T: Round Trip|$| RT        | x2        | | Rate by    | ~6 RTT    |
    |   Loss Bit  |1|           | Half RT   |*| sampling   +-----------+
    |             | |           |           | | 1/3 to 1/(3*ppa) of    |
    |             | |           |           | | pkts over 2 RTT        |
    +-------------+-+-----------+-----------+-+------------+-----------+
    |Q: sQuare Bit|1| Upstream  | x2        |*| Rate over  | N pkts    |
    |             | |           |           | | N pkts     | (e.g. 64) |
    |             | |           |           | | (e.g. 64)  |           |
    +-------------+-+-----------+-----------+-+------------+-----------+
    |L: Loss Event|1| E2E       | x2        |#| Loss shape | Min: RTT  |
    |   Bit       | |           |           | | (and rate) | Max: RTO  |
    +-------------+-+-----------+-----------+-+------------+-----------+
    |QL: sQuare + |2| Upstream  | x2        | | -> see Q   | Up: see Q |
    |    Loss Ev. | | Downstream| x2        |#| -> see Q|L | Others:   |
    |    Bits     | | E2E       | x2        | | -> see L   |     see L |
    +-------------+-+-----------+-----------+-+------------+-----------+
    |QR: sQuare + |2| Upstream  | x2        | | Rate over  | Up: see Q |
    |    Ref. Sq. | | 3/4 RT    | x2        | | N*ppa pkts | Others:   |
    |    Bits     | | !E2E      | E2E       |*| (see Q bit |  N*ppa pk |
    |             | |           | Downstream| |   for N)   |   (see Q  |
    |             | |           | Half RT   | |            |    for N) |
    +-------------+-+-----------+-----------+-+------------+-----------+

    *   All protocols
    #   Protocols employing loss detection
        (with or without pure ACK loss detection)
    $   Require a working Spin bit
    !   Metric relative to the opposite channel
    x2  Same metric for both directions
    ppa Packets-Per-Ack
    Q|L See Q if Upstream loss is significant; L otherwise

                         Figure 15: Loss Comparison

5.  Protocol Ossification Considerations

   Accurate loss and delay information is not critical to the operation
   of any protocol, though its presence for a sufficient number of flows
   is important for the operation of networks.

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   The delay and loss bits are amenable to "greasing" described in
   [RFC8701], if the protocol designers are not ready to dedicate (and
   ossify) bits used for loss reporting to this function.  The greasing
   could be accomplished similarly to the Latency Spin bit greasing in
   [QUIC-TRANSPORT].  Namely, implementations could decide that a
   fraction of flows should not encode loss and delay information and,
   instead, the bits would be set to arbitrary values.  The observers
   would need to be ready to ignore flows with delay and loss
   information more resembling noise than the expected signal.

6.  Examples of Application

6.1.  QUIC

   The binding of a delay signal to QUIC is partially described in
   [QUIC-TRANSPORT], which adds the Spin bit to the first byte of the
   short packet header, leaving two reserved bits for future use.

   To implement the additional signals discussed in this document, the
   first byte of the short packet header can be modified as follows:

   *  the Delay bit (D) can be placed in the first reserved bit (i.e.
      the fourth most significant bit _0x10_) while the round trip loss
      bit (T) in the second reserved bit (i.e. the fifth most
      significant bit _0x08_); the proposed scheme is:

             0 1 2 3 4 5 6 7
            +-+-+-+-+-+-+-+-+
            |0|1|S|D|T|K|P|P|
            +-+-+-+-+-+-+-+-+

                            Figure 16: Scheme 1

   *  alternatively, a two bits loss signal (QL or QR) can be placed in
      both reserved bits; the proposed schemes, in this case, are:

             0 1 2 3 4 5 6 7
            +-+-+-+-+-+-+-+-+
            |0|1|S|Q|L|K|P|P|
            +-+-+-+-+-+-+-+-+

                            Figure 17: Scheme 2A

             0 1 2 3 4 5 6 7
            +-+-+-+-+-+-+-+-+
            |0|1|S|Q|R|K|P|P|
            +-+-+-+-+-+-+-+-+

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                            Figure 18: Scheme 2B

   A further option would be to substitute the Spin bit with the Delay
   bit, leaving the two reserved bits for loss detection.  The proposed
   schemes are:

             0 1 2 3 4 5 6 7
            +-+-+-+-+-+-+-+-+
            |0|1|D|Q|L|K|P|P|
            +-+-+-+-+-+-+-+-+

                            Figure 19: Scheme 3A

             0 1 2 3 4 5 6 7
            +-+-+-+-+-+-+-+-+
            |0|1|D|Q|R|K|P|P|
            +-+-+-+-+-+-+-+-+

                            Figure 20: Scheme 3B

6.2.  TCP

   The signals can be added to TCP by defining bit 4 of byte 13 of the
   TCP header to carry the Spin bit or the Delay bit, and possibly bits
   5 and 6 to carry additional information, like the Delay bit and the
   round-trip loss bit (DT), or a two bits loss signal (QL or QR).

7.  Security Considerations

   Passive loss and delay observations have been a part of the network
   operations for a long time, so exposing loss and delay information to
   the network does not add new security concerns for protocols that are
   currently observable.

   In the absence of packet loss, Q and R bits signals do not provide
   any information that cannot be observed by simply counting packets
   transiting a network path.  In the presence of packet loss, Q and R
   bits will disclose the loss, but this is information about the
   environment and not the endpoint state.  The L bit signal discloses
   internal state of the protocol's loss detection machinery, but this
   state can often be gleamed by timing packets and observing congestion
   controller response.

   Hence, loss bits do not provide a viable new mechanism to attack data
   integrity and secrecy.

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   The described techniques can generally apply to different
   communication protocols operating in different security environments.
   An implementation of these techniques for a particular protocol must
   consider specifics of the protocol and its expected operating
   environment.  For example, security considerations for QUIC,
   discussed in [QUIC-TRANSPORT] and [QUIC-TLS], consider a possibility
   of active and passive attackers in the network as well as attacks on
   specific QUIC mechanisms.

7.1.  Optimistic ACK Attack

   A defense against an Optimistic ACK Attack, described in
   [QUIC-TRANSPORT], involves a sender randomly skipping packet numbers
   to detect a receiver acknowledging packet numbers that have never
   been received.  The Q bit signal may inform the attacker which packet
   numbers were skipped on purpose and which had been actually lost (and
   are, therefore, safe for the attacker to acknowledge).  To use the Q
   bit for this purpose, the attacker must first receive at least an
   entire Q Block of packets, which renders the attack ineffective
   against a delay-sensitive congestion controller.

   A protocol that is more susceptible to an Optimistic ACK Attack with
   the loss signal provided by Q bit and uses a loss-based congestion
   controller, should shorten the current Q Block by the number of
   skipped packets numbers.  For example, skipping a single packet
   number will invert the square signal one outgoing packet sooner.

   Similar considerations apply to the R bit, although a shortened R
   Block along with a matching skip in packet numbers does not
   necessarily imply a lost packet, since it could be due to a lost
   packet on the reverse path along with a deliberately skipped packet
   by the sender.

7.2.  Delay Bit with RTT Obfuscation

   Theoretically, delay measurements can be used to roughly evaluate the
   distance of the client from the server (using the RTT) or from any
   intermediate observer (using the client-observer half-RTT).  As
   described in [SPIN-RTT], connection RTT measurements for geolocating
   endpoints are usually inferior to even the most basic IP geolocation
   databases.  It is the variability within RTT measurements (the
   jitter) that is most informative, as it can provide insight into the
   operating environment of the endpoints as well as the state of the
   networks (queuing delays) used by the connection.

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   Nevertheless, to further mask the actual RTT of the connection, the
   Delay bit algorithm can be slightly modified by, for example,
   delaying the client-side reflection of the delay sample by a fixed
   randomly chosen time value.  This would lead an intermediate observer
   to measure a delay greater than the real one.

   This Additional Delay should be randomly selected by the client and
   kept constant for a certain amount of time across multiple
   connections.  This ensures that the client-server jitter remains the
   same as if no Additional Delay had been inserted.  For example, a new
   Additional Delay value could be generated whenever the client's IP
   address changes.

   Despite the Additional Delay, this Hidden Delay technique still
   allows an accurate measurement of the RTT components (observer-
   server) and all the intra-domain measurements used to distribute the
   delay in the network.  Furthermore, unlike the Delay bit, the Hidden
   Delay bit does not require the use of the client reflection threshold
   (1ms by default).  Removing this threshold may lead to increasing the
   number of valid measurements produced by the algorithm.

   Note that Hidden Delay bit does not affect an observer's ability to
   measure accurate RTT using other means, such as timing packets
   exchanged during the connection establishment.

8.  Privacy Considerations

   To minimize unintentional exposure of information, loss bits provide
   an explicit loss signal -- a preferred way to share information per
   [RFC8558].

   New protocols commonly have specific privacy goals, and loss
   reporting must ensure that loss information does not compromise those
   privacy goals.  For example, [QUIC-TRANSPORT] allows changing
   Connection IDs in the middle of a connection to reduce the likelihood
   of a passive observer linking old and new sub-flows to the same
   device.  A QUIC implementation would need to reset all counters when
   it changes the destination (IP address or UDP port) or the Connection
   ID used for outgoing packets.  It would also need to avoid
   incrementing Unreported Loss counter for loss of packets sent to a
   different destination or with a different Connection ID.

9.  IANA Considerations

   This document makes no request of IANA.

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10.  Contributors

   The following people provided valuable contributions to this
   document:

   *  Marcus Ihlar, Ericsson, marcus.ihlar@ericsson.com

   *  Jari Arkko, Ericsson, jari.arkko@ericsson.com

   *  Emile Stephan, Orange, emile.stephan@orange.com

   *  Dmitri Tikhonov, LiteSpeed Technologies,
      dtikhonov@litespeedtech.com

11.  Acknowledgements

   The authors would like to thank the QUIC WG for their contributions,
   Christian Huitema for implementing Q and L bits in his picoquic
   stack, and Ike Kunze for providing constructive reviews and helpful
   suggestions.

12.  References

12.1.  Normative References

   [ConEx]    Mathis, M. and B. Briscoe, "Congestion Exposure (ConEx)
              Concepts, Abstract Mechanism, and Requirements", RFC 7713,
              DOI 10.17487/RFC7713, December 2015,
              <https://www.rfc-editor.org/info/rfc7713>.

   [ConEx-TCP]
              Kuehlewind, M., Ed. and R. Scheffenegger, "TCP
              Modifications for Congestion Exposure (ConEx)", RFC 7786,
              DOI 10.17487/RFC7786, May 2016,
              <https://www.rfc-editor.org/info/rfc7786>.

   [ECN]      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>.

   [IP]       Postel, J., "Internet Protocol", STD 5, RFC 791,
              DOI 10.17487/RFC0791, September 1981,
              <https://www.rfc-editor.org/info/rfc791>.

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   [IPM-Methods]
              Morton, A., "Active and Passive Metrics and Methods (with
              Hybrid Types In-Between)", RFC 7799, DOI 10.17487/RFC7799,
              May 2016, <https://www.rfc-editor.org/info/rfc7799>.

   [IPv6]     Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", STD 86, RFC 8200,
              DOI 10.17487/RFC8200, July 2017,
              <https://www.rfc-editor.org/info/rfc8200>.

   [RFC8558]  Hardie, T., Ed., "Transport Protocol Path Signals",
              RFC 8558, DOI 10.17487/RFC8558, April 2019,
              <https://www.rfc-editor.org/info/rfc8558>.

   [TCP]      Postel, J., "Transmission Control Protocol", RFC 793,
              DOI 10.17487/RFC0793, September 1981,
              <https://www.rfc-editor.org/info/rfc793>.

12.2.  Informative References

   [ACCURATE] Briscoe, B., Kühlewind, M., and R. Scheffenegger, "More
              Accurate ECN Feedback in TCP", Work in Progress, Internet-
              Draft, draft-ietf-tcpm-accurate-ecn-20, 25 July 2022,
              <https://www.ietf.org/archive/id/draft-ietf-tcpm-accurate-
              ecn-20.txt>.

   [AltMark]  Fioccola, G., Cociglio, M., Mirsky, G., Mizrahi, T., and
              T. Zhou, "Alternate-Marking Method", Work in Progress,
              Internet-Draft, draft-ietf-ippm-rfc8321bis-03, 25 July
              2022, <https://www.ietf.org/archive/id/draft-ietf-ippm-
              rfc8321bis-03.txt>.

   [ANRW19-PM-QUIC]
              Bulgarella, F., Cociglio, M., Fioccola, G., Marchetto, G.,
              and R. Sisto, "Performance measurements of QUIC
              communications", Proceedings of the Applied Networking
              Research Workshop, DOI 10.1145/3340301.3341127, July 2019,
              <https://doi.org/10.1145/3340301.3341127>.

   [I-D.trammell-ippm-spin]
              Trammell, B., "An Explicit Transport-Layer Signal for
              Hybrid RTT Measurement", Work in Progress, Internet-Draft,
              draft-trammell-ippm-spin-00, 9 January 2019,
              <https://www.ietf.org/archive/id/draft-trammell-ippm-spin-
              00.txt>.

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   [I-D.trammell-tsvwg-spin]
              Trammell, B., "A Transport-Independent Explicit Signal for
              Hybrid RTT Measurement", Work in Progress, Internet-Draft,
              draft-trammell-tsvwg-spin-00, 2 July 2018,
              <https://www.ietf.org/archive/id/draft-trammell-tsvwg-
              spin-00.txt>.

   [IPv6AltMark]
              Fioccola, G., Zhou, T., Cociglio, M., Qin, F., and R.
              Pang, "IPv6 Application of the Alternate Marking Method",
              Work in Progress, Internet-Draft, draft-ietf-6man-ipv6-
              alt-mark-17, 27 September 2022,
              <https://www.ietf.org/archive/id/draft-ietf-6man-ipv6-alt-
              mark-17.txt>.

   [QUIC-TLS] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
              QUIC", RFC 9001, DOI 10.17487/RFC9001, May 2021,
              <https://www.rfc-editor.org/info/rfc9001>.

   [QUIC-TRANSPORT]
              Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
              Multiplexed and Secure Transport", RFC 9000,
              DOI 10.17487/RFC9000, May 2021,
              <https://www.rfc-editor.org/info/rfc9000>.

   [RFC8517]  Dolson, D., Ed., Snellman, J., Boucadair, M., Ed., and C.
              Jacquenet, "An Inventory of Transport-Centric Functions
              Provided by Middleboxes: An Operator Perspective",
              RFC 8517, DOI 10.17487/RFC8517, February 2019,
              <https://www.rfc-editor.org/info/rfc8517>.

   [RFC8701]  Benjamin, D., "Applying Generate Random Extensions And
              Sustain Extensibility (GREASE) to TLS Extensibility",
              RFC 8701, DOI 10.17487/RFC8701, January 2020,
              <https://www.rfc-editor.org/info/rfc8701>.

   [RFC9065]  Fairhurst, G. and C. Perkins, "Considerations around
              Transport Header Confidentiality, Network Operations, and
              the Evolution of Internet Transport Protocols", RFC 9065,
              DOI 10.17487/RFC9065, July 2021,
              <https://www.rfc-editor.org/info/rfc9065>.

   [SPIN-BIT] Kühlewind, M. and B. Trammell, "Manageability of the QUIC
              Transport Protocol", RFC 9312, DOI 10.17487/RFC9312,
              September 2022, <https://www.rfc-editor.org/info/rfc9312>.

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   [SPIN-RTT] Trammell, B. and M. Kühlewind, "Revisiting the Privacy
              Implications of Two-Way Internet Latency Data", Passive
              and Active Measurement pp. 73-84,
              DOI 10.1007/978-3-319-76481-8_6, 2018,
              <https://doi.org/10.1007/978-3-319-76481-8_6>.

   [UDP-OPTIONS]
              Touch, J. D., "Transport Options for UDP", Work in
              Progress, Internet-Draft, draft-ietf-tsvwg-udp-options-18,
              26 March 2022, <https://www.ietf.org/archive/id/draft-
              ietf-tsvwg-udp-options-18.txt>.

   [UDP-SURPLUS]
              Herbert, T., "UDP Surplus Header", Work in Progress,
              Internet-Draft, draft-herbert-udp-space-hdr-01, 8 July
              2019, <https://www.ietf.org/archive/id/draft-herbert-udp-
              space-hdr-01.txt>.

Authors' Addresses

   Mauro Cociglio
   Telecom Italia - TIM
   Via Reiss Romoli, 274
   10148 Torino
   Italy
   Email: mauro.cociglio@outlook.com

   Alexandre Ferrieux
   Orange Labs
   Email: alexandre.ferrieux@orange.com

   Giuseppe Fioccola
   Huawei Technologies
   Riesstrasse, 25
   80992 Munich
   Germany
   Email: giuseppe.fioccola@huawei.com

   Igor Lubashev
   Akamai Technologies
   Email: ilubashe@akamai.com

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   Fabio Bulgarella
   Telecom Italia - TIM
   Via Reiss Romoli, 274
   10148 Torino
   Italy
   Email: fabio.bulgarella@guest.telecomitalia.it

   Massimo Nilo
   Telecom Italia - TIM
   Via Reiss Romoli, 274
   10148 Torino
   Italy
   Email: massimo.nilo@telecomitalia.it

   Isabelle Hamchaoui
   Orange Labs
   Email: isabelle.hamchaoui@orange.com

   Riccardo Sisto
   Politecnico di Torino
   Email: riccardo.sisto@polito.it

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