TCP Maintenance Working Group                                   Y. Cheng
Internet-Draft                                               N. Cardwell
Intended status: Experimental                               N. Dukkipati
Expires: September 14, 2017                                  Google, Inc
                                                          March 13, 2017

        RACK: a time-based fast loss detection algorithm for TCP


   This document presents a new TCP loss detection algorithm called RACK
   ("Recent ACKnowledgment").  RACK uses the notion of time, instead of
   packet or sequence counts, to detect losses, for modern TCP
   implementations that can support per-packet timestamps and the
   selective acknowledgment (SACK) option.  It is intended to replace
   the conventional DUPACK threshold approach and its variants, as well
   as other nonstandard approaches.

Status of This Memo

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   This Internet-Draft will expire on September 14, 2017.

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   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   include Simplified BSD License text as described in Section 4.e of

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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

1.  Introduction

   This document presents a new loss detection algorithm called RACK
   ("Recent ACKnowledgment").  RACK uses the notion of time instead of
   the conventional packet or sequence counting approaches for detecting
   losses.  RACK deems a packet lost if some packet sent sufficiently
   later has been delivered.  It does this by recording packet
   transmission times and inferring losses using cumulative
   acknowledgments or selective acknowledgment (SACK) TCP options.

   In the last couple of years we have been observing several
   increasingly common loss and reordering patterns in the Internet:

   1.  Lost retransmissions.  Traffic policers [POLICER16] and burst
       losses often cause retransmissions to be lost again, severely
       increasing TCP latency.

   2.  Tail drops.  Structured request-response traffic turns more
       losses into tail drops.  In such cases, TCP is application-
       limited, so it cannot send new data to probe losses and has to
       rely on retransmission timeouts (RTOs).

   3.  Reordering.  Link layer protocols (e.g., 802.11 block ACK) or
       routers' internal load-balancing can deliver TCP packets out of
       order.  The degree of such reordering is usually within the order
       of the path round trip time.

   Despite TCP stacks (e.g.  Linux) that implement many of the standard
   and proposed loss detection algorithms
   STREAM][TLP], we've found that together they do not perform well.
   The main reason is that many of them are based on the classic rule of
   counting duplicate acknowledgments [RFC5681].  They can either detect
   loss quickly or accurately, but not both, especially when the sender
   is application-limited or under reordering that is unpredictable.
   And under these conditions none of them can detect lost
   retransmissions well.

   Also, these algorithms, including RFCs, rarely address the
   interactions with other algorithms.  For example, FACK may consider a
   packet is lost while RFC3517 may not.  Implementing N algorithms
   while dealing with N^2 interactions is a daunting task and error-

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   The goal of RACK is to solve all the problems above by replacing many
   of the loss detection algorithms above with one simpler, and also
   more effective, algorithm.

2.  Overview

   The main idea behind RACK is that if a packet has been delivered out
   of order, then the packets sent chronologically before that were
   either lost or reordered.  This concept is not fundamentally
   different from [RFC5681][RFC3517][FACK].  But the key innovation in
   RACK is to use a per-packet transmission timestamp and widely
   deployed SACK options to conduct time-based inferences instead of
   inferring losses with packet or sequence counting approaches.

   Using a threshold for counting duplicate acknowledgments (i.e.,
   DupThresh) is no longer reliable because of today's prevalent
   reordering patterns.  A common type of reordering is that the last
   "runt" packet of a window's worth of packet bursts gets delivered
   first, then the rest arrive shortly after in order.  To handle this
   effectively, a sender would need to constantly adjust the DupThresh
   to the burst size; but this would risk increasing the frequency of
   RTOs on real losses.

   Today's prevalent lost retransmissions also cause problems with
   packet-counting approaches [RFC5681][RFC3517][FACK], since those
   approaches depend on reasoning in sequence number space.
   Retransmissions break the direct correspondence between ordering in
   sequence space and ordering in time.  So when retransmissions are
   lost, sequence-based approaches are often unable to infer and quickly
   repair losses that can be deduced with time-based approaches.

   Instead of counting packets, RACK uses the most recently delivered
   packet's transmission time to judge if some packets sent previous to
   that time have "expired" by passing a certain reordering settling
   window.  On each ACK, RACK marks any already-expired packets lost,
   and for any packets that have not yet expired it waits until the
   reordering window passes and then marks those lost as well.  In
   either case, RACK can repair the loss without waiting for a (long)
   RTO.  RACK can be applied to both fast recovery and timeout recovery,
   and can detect losses on both originally transmitted and
   retransmitted packets, making it a great all-weather loss detection

3.  Requirements

   The reader is expected to be familiar with the definitions given in
   the TCP congestion control [RFC5681] and selective acknowledgment

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   [RFC2018] RFCs.  Familiarity with the conservative SACK-based
   recovery for TCP [RFC6675] is not expected but helps.

   RACK has three requirements:

   1.  The connection MUST use selective acknowledgment (SACK) options

   2.  For each packet sent, the sender MUST store its most recent
       transmission time with (at least) millisecond granularity.  For
       round-trip times lower than a millisecond (e.g., intra-datacenter
       communications) microsecond granularity would significantly help
       the detection latency but is not required.

   3.  For each packet sent, the sender MUST remember whether the packet
       has been retransmitted or not.

   We assume that requirement 1 implies the sender keeps a SACK
   scoreboard, which is a data structure to store selective
   acknowledgment information on a per-connection basis ([RFC6675]
   section 3).  For the ease of explaining the algorithm, we use a
   pseudo-scoreboard that manages the data in sequence number ranges.
   But the specifics of the data structure are left to the implementor.

   RACK does not need any change on the receiver.

4.  Definitions of variables

   A sender needs to store these new RACK variables:

   "Packet.xmit_ts" is the time of the last transmission of a data
   packet, including retransmissions, if any.  The sender needs to
   record the transmission time for each packet sent and not yet
   acknowledged.  The time MUST be stored at millisecond granularity or

   "RACK.packet".  Among all the packets that have been either
   selectively or cumulatively acknowledged, RACK.packet is the one that
   was sent most recently (including retransmissions).

   "RACK.xmit_ts" is the latest transmission timestamp of RACK.packet.

   "RACK.end_seq" is the ending TCP sequence number of RACk.packet.

   "RACK.RTT" is the associated RTT measured when RACK.xmit_ts, above,
   was changed.  It is the RTT of the most recently transmitted packet
   that has been delivered (either cumulatively acknowledged or
   selectively acknowledged) on the connection.

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   "RACK.reo_wnd" is a reordering window for the connection, computed in
   the unit of time used for recording packet transmission times.  It is
   used to defer the moment at which RACK marks a packet lost.

   "RACK.min_RTT" is the estimated minimum round-trip time (RTT) of the

   "RACK.ack_ts" is the time when all the sequences in RACK.packet were
   selectively or cumulatively acknowledged.

   Note that the Packet.xmit_ts variable is per packet in flight.  The
   RACK.xmit_ts, RACK.end_seq, RACK.RTT, RACK.reo_wnd, and RACK.min_RTT
   variables are kept in the per-connection TCP control block.
   RACK.packet and RACK.ack_ts are used as local variables in the

5.  Algorithm Details

5.1.  Transmitting a data packet

   Upon transmitting a new packet or retransmitting an old packet,
   record the time in Packet.xmit_ts.  RACK does not care if the
   retransmission is triggered by an ACK, new application data, an RTO,
   or any other means.

5.2.  Upon receiving an ACK

   Step 1: Update RACK.min_RTT.

   Use the RTT measurements obtained in [RFC6298] or [RFC7323] to update
   the estimated minimum RTT in RACK.min_RTT.  The sender can track a
   simple global minimum of all RTT measurements from the connection, or
   a windowed min-filtered value of recent RTT measurements.  This
   document does not specify an exact approach.

   Step 2: Update RACK.reo_wnd.

   To handle the prevalent small degree of reordering, RACK.reo_wnd
   serves as an allowance for settling time before marking a packet
   lost.  By default it is 1 millisecond.  [REORDER-DETECT][RFC4737] MAY
   be implemented to dynamically adjust the reordering window upon
   detecting reordering.  When the sender detects packet reordering
   RACK.reo_wnd MAY be changed to RACK.min_RTT/4.  We discuss more about
   the reordering window in the next section.

   Step 3: Advance RACK.xmit_ts and update RACK.RTT and RACK.end_seq

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   Given the information provided in an ACK, each packet cumulatively
   ACKed or SACKed is marked as delivered in the scoreboard.  Among all
   the packets newly ACKed or SACKed in the connection, record the most
   recent Packet.xmit_ts in RACK.xmit_ts if it is ahead of RACK.xmit_ts.
   Ignore the packet if any of its TCP sequences have been retransmitted
   before and either of two conditions is true:

   1.  The Timestamp Echo Reply field (TSecr) of the ACK's timestamp
       option [RFC7323], if available, indicates the ACK was not
       acknowledging the last retransmission of the packet.

   2.  The packet was last retransmitted less than RACK.min_rtt ago.
       While it is still possible the packet is spuriously retransmitted
       because of a recent RTT decrease, we believe that our experience
       suggests this is a reasonable heuristic.

   If this ACK causes a change to RACK.xmit_ts then record the RTT and
   sequence implied by this ACK:

      RACK.RTT = Now() - RACK.xmit_ts
      RACK.end_seq = Packet.end_seq

   Exit here and omit the following steps if RACK.xmit_ts has not

   Step 4: Detect losses.

   For each packet that has not been fully SACKed, if RACK.xmit_ts is
   after Packet.xmit_ts + RACK.reo_wnd, then mark the packet (or its
   corresponding sequence range) lost in the scoreboard.  The rationale
   is that if another packet that was sent later has been delivered, and
   the reordering window or "reordering settling time" has already
   passed, then the packet was likely lost.

   If another packet that was sent later has been delivered, but the
   reordering window has not passed, then it is not yet safe to deem the
   unacked packet lost.  Using the basic algorithm above, the sender
   would wait for the next ACK to further advance RACK.xmit_ts; but this
   risks a timeout (RTO) if no more ACKs come back (e.g, due to losses
   or application limit).  For timely loss detection, the sender MAY
   install a "reordering settling" timer set to fire at the earliest
   moment at which it is safe to conclude that some packet is lost.  The
   earliest moment is the time it takes to expire the reordering window
   of the earliest unacked packet in flight.

   This timer expiration value can be derived as follows.  As a starting
   point, we consider that the reordering window has passed if the
   RACK.packet was sent sufficiently after the packet in question, or a

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   sufficient time has elapsed since the RACK.packet was S/ACKed, or
   some combination of the two.  More precisely, RACK marks a packet as
   lost if the reordering window for a packet has elapsed through the
   sum of:

   1.  delta in transmit time between a packet and the RACK.packet

   2.  delta in time between RACK.ack_ts and now

   So we mark a packet as lost if:

       RACK.xmit_ts > Packet.xmit_ts
       RACK.xmit_ts - Packet.xmit_ts + now - RACK.ack_ts > RACK.reo_wnd

   If we solve this second condition for "now", the moment at which we
   can declare a packet lost, then we get:

       now > Packet.xmit_ts + RACK.reo_wnd + RACK.ack_ts - RACK.xmit_ts

   Then (RACK.ack_ts - RACK.xmit_ts) is just the RTT of the packet we
   used to set RACK.xmit_ts, so this reduces to:

       now > Packet.xmit_ts + RACK.RTT + RACK.reo_wnd

   The following pseudocode implements the algorithm above.  When an ACK
   is received or the RACK timer expires, call RACK_detect_loss().  The
   algorithm includes an additional optimization to break timestamp ties
   by using the TCP sequence space.  The optimization is particularly
   useful to detect losses in a timely manner with TCP Segmentation
   Offload, where multiple packets in one TSO blob have identical
   timestamps.  It is also useful when the timestamp clock granularity
   is close to or longer than the actual round trip time.

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      min_timeout = 0

      For each packet, Packet, in the scoreboard:
          If Packet is already SACKed, ACKed,
             or marked lost and not yet retransmitted:
              Skip to the next packet

          If Packet.xmit_ts > RACK.xmit_ts:
              Skip to the next packet
          /*  Timestamp tie breaker */
          If Packet.xmit_ts == RACK.xmit_ts AND
             Packet.end_seq > RACK.end_seq:
              Skip to the next packet

          timeout = Packet.xmit_ts + RACK.RTT + RACK.reo_wnd + 1
          If Now() >= timeout:
              Mark Packet lost
          Else If (min_timeout == 0) or (timeout is before min_timeout):
              min_timeout = timeout

      If min_timeout != 0
          Arm a timer to call RACK_detect_loss() after min_timeout

6.  Tail Loss Probe: fast recovery on tail losses

   This section describes a supplemental algorithm, Tail Loss Probe
   (TLP), which leverages RACK to further reduce RTO recoveries.  TLP
   triggers fast recovery to quickly repair tail losses that can
   otherwise be recovered by RTOs only.  After an original data
   transmission, TLP sends a probe data segment within one to two RTTs.
   The probe data segment can either be new, previously unsent data, or
   a retransmission of previously sent data just below SND.NXT.  In
   either case the goal is to elicit more feedback from the receiver, in
   the form of an ACK (potentially with SACK blocks), to allow RACK to
   trigger fast recovery instead of an RTO.

   An RTO occurs when the first unacknowledged sequence number is not
   acknowledged after a conservative period of time has elapsed
   [RFC6298].  Common causes of RTOs include:

   1.  Tail losses at the end of an application transaction.

   2.  Lost retransmits, which can halt fast recovery based on [RFC6675]
       if the ACK stream completely dries up.  For example, consider a
       window of three data packets (P1, P2, P3) that are sent; P1 and
       P2 are dropped.  On receipt of a SACK for P3, RACK marks P1 and
       P2 as lost and retransmits them as R1 and R2.  Suppose R1 and R2

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       are lost as well, so there are no more returning ACKs to detect
       R1 and R2 as lost.  Recovery stalls.

   3.  Tail losses of ACKs.

   4.  An unexpectedly long round-trip time (RTT).  This can cause ACKs
       to arrive after the RTO timer expires.  The F-RTO algorithm
       [RFC5682] is designed to detect such spurious retransmission
       timeouts and at least partially undo the consequences of such
       events, but F-RTO cannot be used in many situations.

6.1.  Tail Loss Probe: An Example

   Following is an example of TLP.  All events listed are at a TCP

   (1) Sender transmits segments 1-10: 1, 2, 3, ..., 8, 9, 10.  There is
   no more new data to transmit.  A PTO is scheduled to fire in 2 RTTs,
   after the transmission of the 10th segment.  (2) Sender receives
   acknowledgements (ACKs) for segments 1-5; segments 6-10 are lost and
   no ACKs are received.  The sender reschedules its PTO timer relative
   to the last received ACK, which is the ACK for segment 5 in this
   case.  The sender sets the PTO interval using the calculation
   described in step (2) of the algorithm.  (3) When PTO fires, sender
   retransmits segment 10.  (4) After an RTT, a SACK for packet 10
   arrives.  The ACK also carries SACK holes for segments 6, 7, 8 and 9.
   This triggers RACK-based loss recovery.  (5) The connection enters
   fast recovery and retransmits the remaining lost segments.

6.2.  Tail Loss Probe Algorithm Details

   We define the terminology used in specifying the TLP algorithm:

   FlightSize: amount of outstanding data in the network, as defined in

   RTO: The transport's retransmission timeout (RTO) is based on
   measured round-trip times (RTT) between the sender and receiver, as
   specified in [RFC6298] for TCP.  PTO: Probe timeout (PTO) is a timer
   event indicating that an ACK is overdue.  Its value is constrained to
   be smaller than or equal to an RTO.

   SRTT: smoothed round-trip time, computed as specified in [RFC6298].

   Open state: the sender has so far received in-sequence ACKs with no
   SACK blocks, and no other indications (such as retransmission
   timeout) that a loss may have occurred.

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   The TLP algorithm has three phases, which we discuss in turn.

6.2.1.  Phase 1: Scheduling a loss probe

   Step 1: Check conditions for scheduling a PTO.

   A sender should schedule a PTO after transmitting new data or
   receiving an ACK if the following conditions are met:

   (a) The connection is in Open state
   (b) The connection is either cwnd-limited (the data in flight matches
       or exceeds the cwnd) or application-limited (there is no unsent
       data that the receiver window allows to be sent)
   (c) SACK is enabled for the connection
   (d) The most recently transmitted data was not itself a TLP probe
       (i.e. a sender MUST NOT send consecutive TLP probes)

   Step 2: Select the duration of the PTO.

   A sender SHOULD use the following logic to select the duration of a

       If an SRTT estimate is available:
           PTO = 2 * SRTT
           PTO = initial RTO of 1 sec
       If FlightSize = 1:
           PTO = max(PTO, 1.5 * SRTT + WCDelAckT)
       PTO = max(10ms, PTO)
       PTO = max(RTO, PTO)

   Aiming for a PTO value of 2*SRTT allows a sender to wait long enough
   to know that an ACK is overdue.  Under normal circumstances, i.e. no
   losses, an ACK typically arrives in one SRTT.  But choosing PTO to be
   exactly an SRTT is likely to generate spurious probes given that
   network delay variance and even end-system timings can easily push an
   ACK to be above an SRTT.  We chose PTO to be the next integral
   multiple of SRTT.  Similarly, current end-system processing latencies
   and timer granularities can easily push an ACK beyond 10ms, so
   senders SHOULD use a minimum PTO value of 10ms.  If RTO is smaller

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   than the computed value for PTO, then a probe is scheduled to be sent
   at the RTO time.

   WCDelAckT stands for worst case delayed ACK timer.  When FlightSize
   is 1, PTO is inflated additionally by WCDelAckT time to compensate
   for a potential long delayed ACK timer at the receiver.  The
   RECOMMENDED value for WCDelAckT is 200ms, or the delayed ACK interval
   value explicitly negotiated by the sender and receiver, if one is

6.2.2.  Phase 2: Sending a loss probe

   When the PTO fires, transmit a probe data segment:

       If a previously unsent segment exists AND
          the receive window allows new data to be sent:
           Transmit that new segment
           FlightSize += SMSS
           Retransmit the last segment
       The cwnd remains unchanged

6.2.3.  Phase 3: ACK processing

   On each incoming ACK, the sender should cancel any existing loss
   probe timer.  The sender should then reschedule the loss probe timer
   if the conditions in Step 1 of Phase 1 allow.

6.3.  TLP recovery detection

   If the only loss in an outstanding window of data was the last
   segment, then a TLP loss probe retransmission of that data segment
   might repair the loss.  TLP recovery detection examines ACKs to
   detect when the probe might have repaired a loss, and thus allows
   congestion control to properly reduce the congestion window (cwnd)

   Consider a TLP retransmission episode where a sender retransmits a
   tail packet in a flight.  The TLP retransmission episode ends when
   the sender receives an ACK with a SEG.ACK above the SND.NXT at the
   time the episode started.  During the TLP retransmission episode the
   sender checks for a duplicate ACK or D-SACK indicating that both the
   original segment and TLP retransmission arrived at the receiver,
   meaning there was no loss that needed repairing.  If the TLP sender
   does not receive such an indication before the end of the TLP
   retransmission episode, then it MUST estimate that either the
   original data segment or the TLP retransmission were lost, and

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   congestion control MUST react appropriately to that loss as it would
   any other loss.

   Since a significant fraction of the hosts that support SACK do not
   support duplicate selective acknowledgments (D-SACKs) [RFC2883] the
   TLP algorithm for detecting such lost segments relies only on basic
   SACK support [RFC2018].

   Definitions of variables

   TLPRxtOut: a boolean indicating whether there is an unacknowledged
   TLP retransmission.

   TLPHighRxt: the value of SND.NXT at the time of sending a TLP

6.3.1.  Initializing and resetting state

   When a connection is created, or suffers a retransmission timeout, or
   enters fast recovery, it executes the following:

       TLPRxtOut = false

6.3.2.  Recording loss probe states

   Senders must only send a TLP loss probe retransmission if TLPRxtOut
   is false.  This ensures that at any given time a connection has at
   most one outstanding TLP retransmission.  This allows the sender to
   use the algorithm described in this section to estimate whether any
   data segments were lost.

   Note that this condition only restricts TLP loss probes that are
   retransmissions.  There may be an arbitrary number of outstanding
   unacknowledged TLP loss probes that consist of new, previously-unsent
   data, since the retransmission timeout and fast recovery algorithms
   are sufficient to detect losses of such probe segments.

   Upon sending a TLP probe that is a retransmission, the sender sets
   TLPRxtOut to true and TLPHighRxt to SND.NXT.

   Detecting recoveries accomplished by loss probes

   Step 1: Track ACKs indicating receipt of original and retransmitted

   A sender considers both the original segment and TLP probe
   retransmission segment as acknowledged if either (i) or (ii) are

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   (i) This is a duplicate acknowledgment (as defined in [RFC5681],
   section 2), and all of the following conditions are met:

   (a) TLPRxtOut is true

   (b) SEG.ACK == TLPHighRxt

   (c) SEG.ACK == SND.UNA

   (d) the segment contains no SACK blocks for sequence ranges above

   (e) the segment contains no data

   (f) the segment is not a window update

   (ii) This is an ACK acknowledging a sequence number at or above
   TLPHighRxt and it contains a D-SACK; i.e. all of the following
   conditions are met:

   (a) TLPRxtOut is true

   (b) SEG.ACK >= TLPHighRxt and

   (c) the ACK contains a D-SACK block

   If either conditions (i) or (ii) are met, then the sender estimates
   that the receiver received both the original data segment and the TLP
   probe retransmission, and so the sender considers the TLP episode to
   be done, and records that fact by setting TLPRxtOut to false.

   Step 2: Mark the end of a TLP episode and detect losses

   If the sender receives a cumulative ACK for data beyond the TLP loss
   probe retransmission then, in the absence of reordering on the return
   path of ACKs, it should have received any ACKs for the original
   segment and TLP probe retransmission segment.  At that time, if the
   TLPRxtOut flag is still true and thus indicates that the TLP probe
   retransmission remains unacknowledged, then the sender should presume
   that at least one of its data segments was lost, so it SHOULD invoke
   a congestion control response equivalent to fast recovery.

   More precisely, on each ACK the sender executes the following:

       If TLPRxtOut and SEG.ACK >= TLPHighRxt:
          TLPRxtOut = false

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7.  RACK and TLP discussions

7.1.  Advantages

   The biggest advantage of RACK is that every data packet, whether it
   is an original data transmission or a retransmission, can be used to
   detect losses of the packets sent chronologically prior to it.

   Example: TAIL DROP.  Consider a sender that transmits a window of
   three data packets (P1, P2, P3), and P1 and P3 are lost.  Suppose the
   transmission of each packet is at least RACK.reo_wnd (1 millisecond
   by default) after the transmission of the previous packet.  RACK will
   mark P1 as lost when the SACK of P2 is received, and this will
   trigger the retransmission of P1 as R1.  When R1 is cumulatively
   acknowledged, RACK will mark P3 as lost and the sender will
   retransmit P3 as R3.  This example illustrates how RACK is able to
   repair certain drops at the tail of a transaction without any timer.
   Notice that neither the conventional duplicate ACK threshold
   [RFC5681], nor [RFC6675], nor the Forward Acknowledgment [FACK]
   algorithm can detect such losses, because of the required packet or
   sequence count.

   Example: LOST RETRANSMIT.  Consider a window of three data packets
   (P1, P2, P3) that are sent; P1 and P2 are dropped.  Suppose the
   transmission of each packet is at least RACK.reo_wnd (1 millisecond
   by default) after the transmission of the previous packet.  When P3
   is SACKed, RACK will mark P1 and P2 lost and they will be
   retransmitted as R1 and R2.  Suppose R1 is lost again but R2 is
   SACKed; RACK will mark R1 lost for retransmission again.  Again,
   neither the conventional three duplicate ACK threshold approach, nor
   [RFC6675], nor the Forward Acknowledgment [FACK] algorithm can detect
   such losses.  And such a lost retransmission is very common when TCP
   is being rate-limited, particularly by token bucket policers with
   large bucket depth and low rate limit.  Retransmissions are often
   lost repeatedly because standard congestion control requires multiple
   round trips to reduce the rate below the policed rate.

   Example: SMALL DEGREE OF REORDERING.  Consider a common reordering
   event: a window of packets are sent as (P1, P2, P3).  P1 and P2 carry
   a full payload of MSS octets, but P3 has only a 1-octet payload.

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   Suppose the sender has detected reordering previously (e.g., by
   implementing the algorithm in [REORDER-DETECT]) and thus RACK.reo_wnd
   is min_RTT/4.  Now P3 is reordered and delivered first, before P1 and
   P2.  As long as P1 and P2 are delivered within min_RTT/4, RACK will
   not consider P1 and P2 lost.  But if P1 and P2 are delivered outside
   the reordering window, then RACK will still falsely mark P1 and P2
   lost.  We discuss how to reduce false positives in the end of this

   The examples above show that RACK is particularly useful when the
   sender is limited by the application, which is common for
   interactive, request/response traffic.  Similarly, RACK still works
   when the sender is limited by the receive window, which is common for
   applications that use the receive window to throttle the sender.

   For some implementations (e.g., Linux), RACK works quite efficiently
   with TCP Segmentation Offload (TSO).  RACK always marks the entire
   TSO blob lost because the packets in the same TSO blob have the same
   transmission timestamp.  By contrast, the counting based algorithms
   (e.g., [RFC3517][RFC5681]) may mark only a subset of packets in the
   TSO blob lost, forcing the stack to perform expensive fragmentation
   of the TSO blob, or to selectively tag individual packets lost in the

7.2.  Disadvantages

   RACK requires the sender to record the transmission time of each
   packet sent at a clock granularity of one millisecond or finer.  TCP
   implementations that record this already for RTT estimation do not
   require any new per-packet state.  But implementations that are not
   yet recording packet transmission times will need to add per-packet
   internal state (commonly either 4 or 8 octets per packet or TSO blob)
   to track transmission times.  In contrast, the conventional [RFC6675]
   loss detection approach does not require any per-packet state beyond
   the SACK scoreboard.

7.3.  Adjusting the reordering window

   When the sender detects packet reordering, RACK uses a reordering
   window of min_rtt / 4.  It uses the minimum RTT to accommodate
   reordering introduced by packets traversing slightly different paths
   (e.g., router-based parallelism schemes) or out-of-order deliveries
   in the lower link layer (e.g., wireless links using link-layer
   retransmission).  RACK uses a quarter of minimum RTT because Linux
   TCP used the same factor in its implementation to delay Early
   Retransmit [RFC5827] to reduce spurious loss detections in the
   presence of reordering, and experience shows that this seems to work
   reasonably well.  We have evaluated using the smoothed RTT (SRTT from

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   [RFC6298] RTT estimation) or the most recently measured RTT
   (RACK.RTT) using an experiment similar to that in the Performance
   Evaluation section.  They do not make any significant difference in
   terms of total recovery latency.

   One potential improvement would be to further adapt the reordering
   window by measuring the degree of reordering in time, instead of
   packet distances.  But that requires storing the delivery timestamp
   of each packet.  Some scoreboard implementations currently merge
   SACKed packets together to support TSO (TCP Segmentation Offload) for
   faster scoreboard indexing.  Supporting per-packet delivery
   timestamps is difficult in such implementations.  However, we
   acknowledge that the current metric can be improved by further

7.4.  Relationships with other loss recovery algorithms

   The primary motivation of RACK is to ultimately provide a simple and
   general replacement for some of the standard loss recovery algorithms
   [RFC5681][RFC6675][RFC5827][RFC4653], as well as some nonstandard
   ones [FACK][THIN-STREAM].  While RACK can be a supplemental loss
   detection mechanism on top of these algorithms, this is not
   necessary, because RACK implicitly subsumes most of them.

   [RFC5827][RFC4653][THIN-STREAM] dynamically adjusts the duplicate ACK
   threshold based on the current or previous flight sizes.  RACK takes
   a different approach, by using only one ACK event and a reordering
   window.  RACK can be seen as an extended Early Retransmit [RFC5827]
   without a FlightSize limit but with an additional reordering window.
   [FACK] considers an original packet to be lost when its sequence
   range is sufficiently far below the highest SACKed sequence.  In some
   sense RACK can be seen as a generalized form of FACK that operates in
   time space instead of sequence space, enabling it to better handle
   reordering, application-limited traffic, and lost retransmissions.

   Nevertheless RACK is still an experimental algorithm.  Since the
   oldest loss detection algorithm, the 3 duplicate ACK threshold
   [RFC5681], has been standardized and widely deployed, we RECOMMEND
   TCP implementations use both RACK and the algorithm specified in
   Section 3.2 in [RFC5681] for compatibility.

   RACK is compatible with and does not interfere with the the standard
   RTO [RFC6298], RTO-restart [RFC7765], F-RTO [RFC5682] and Eifel
   algorithms [RFC3522].  This is because RACK only detects loss by
   using ACK events.  It neither changes the RTO timer calculation nor
   detects spurious timeouts.

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   Furthermore, RACK naturally works well with Tail Loss Probe [TLP]
   because a tail loss probe solicits either an ACK or SACK, which can
   be used by RACK to detect more losses.  RACK can be used to relax
   TLP's requirement for using FACK and retransmitting the the highest-
   sequenced packet, because RACK is agnostic to packet sequence
   numbers, and uses transmission time instead.  Thus TLP could be
   modified to retransmit the first unacknowledged packet, which could
   improve application latency.

7.5.  Interaction with congestion control

   RACK intentionally decouples loss detection from congestion control.
   RACK only detects losses; it does not modify the congestion control
   algorithm [RFC5681][RFC6937].  However, RACK may detect losses
   earlier or later than the conventional duplicate ACK threshold
   approach does.  A packet marked lost by RACK SHOULD NOT be
   retransmitted until congestion control deems this appropriate (e.g.
   using [RFC6937]).

   RACK is applicable for both fast recovery and recovery after a
   retransmission timeout (RTO) in [RFC5681].  RACK applies equally to
   fast recovery and RTO recovery because RACK is purely based on the
   transmission time order of packets.  When a packet retransmitted by
   RTO is acknowledged, RACK will mark any unacked packet sent
   sufficiently prior to the RTO as lost, because at least one RTT has
   elapsed since these packets were sent.

   The following simple example compares how RACK and non-RACK loss
   detection interacts with congestion control: suppose a TCP sender has
   a congestion window (cwnd) of 20 packets on a SACK-enabled
   connection.  It sends 10 data packets and all of them are lost.

   Without RACK, the sender would time out, reset cwnd to 1, and
   retransmit first packet.  It would take another four round trips (1 +
   2 + 4 + 3 = 10) to retransmit all the 10 lost packets using slow
   start.  The recovery latency would be RTO + 4*RTT, with an ending
   cwnd of 4 packets due to congestion window validation.

   With RACK, a sender would send the TLP after 2*RTT and get a DUPACK.
   If the sender implements Proportional Rate Reduction [RFC6937] it
   would slow start to retransmit the remaining 9 lost packets since the
   number of packets in flight (0) is lower than the slow start
   threshold (10).  The slow start would again take another four round
   trips (1 + 2 + 4 + 3 = 10).  The recovery latency would be 2*RTT +
   4*RTT, with an ending cwnd set to the slow start threshold of 10

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   In both cases, the sender after the recovery would be in congestion
   avoidance.  The difference in recovery latency (RTO + 4*RTT vs 6*RTT)
   can be significant if the RTT is much smaller than the minimum RTO (1
   second in RFC6298) or if the RTT is large.  The former case is common
   in local area networks, data-center networks, or content distribution
   networks with deep deployments.  The latter case is more common in
   developing regions with highly congested and/or high-latency
   networks.  The ending congestion window after recovery also impacts
   subsequent data transfer.

7.6.  TLP recovery detection with delayed ACKs

   Delayed ACKs complicate the detection of repairs done by TLP, since
   with a delayed ACK the sender receives one fewer ACK than would
   normally be expected.  To mitigate this complication, before sending
   a TLP loss probe retransmission, the sender should attempt to wait
   long enough that the receiver has sent any delayed ACKs that it is
   withholding.  The sender algorithm described above features such a
   delay, in the form of WCDelAckT.  Furthermore, if the receiver
   supports duplicate selective acknowledgments (D-SACKs) [RFC2883] then
   in the case of a delayed ACK the sender's TLP recovery detection
   algorithm (see above) can use the D-SACK information to infer that
   the original and TLP retransmission both arrived at the receiver.

   If there is ACK loss or a delayed ACK without a D-SACK, then this
   algorithm is conservative, because the sender will reduce cwnd when
   in fact there was no packet loss.  In practice this is acceptable,
   and potentially even desirable: if there is reverse path congestion
   then reducing cwnd can be prudent.

7.7.  RACK for other transport protocols

   RACK can be implemented in other transport protocols.  The algorithm
   can be simplified by skipping step 3 if the protocol can support a
   unique transmission or packet identifier (e.g.  TCP echo options).
   For example, the QUIC protocol implements RACK [QUIC-LR].

8.  Experiments and Performance Evaluations

   RACK and TLP have been deployed at Google, for both connections to
   users in the Internet and internally.  We conducted a performance
   evaluation experiment for RACK and TLP on a small set of Google Web
   servers in Western Europe that serve mostly European and some African
   countries.  The experiment lasted three days in March 2017.  The
   servers were divided evenly into four groups of roughly 5.3 million
   flows each:

   Group 1 (control): RACK off, TLP off, RFC 3517 on

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   Group 2: RACK on, TLP off, RFC 3517 on

   Group 3: RACK on, TLP on, RFC 3517 on

   Group 4: RACK on, TLP on, RFC 3517 off

   All groups used Linux with CUBIC congestion control, an initial
   congestion window of 10 packets, and the fq/pacing qdisc.  In terms
   of specific recovery features, all groups enabled RFC5682 (F-RTO) but
   disabled FACK because it is not an IETF RFC.  FACK was excluded
   because the goal of this setup is to compare RACK and TLP to RFC-
   based loss recoveries.  Since TLP depends on either FACK or RACK, we
   could not run another group that enables TLP only (with both RACK and
   FACK disabled).  Group 4 is to test whether RACK plus TLP can
   completely replace the DupThresh-based [RFC3517].

   The servers sit behind a load balancer that distributes the
   connections evenly across the four groups.

   Each group handles a similar number of connections and sends and
   receives similar amounts of data.  We compare total time spent in
   loss recovery across groups.  The recovery time is measured from when
   the recovery and retransmission starts, until the remote host has
   acknowledged the highest sequence (SND.NXT) at the time the recovery
   started.  Therefore the recovery includes both fast recoveries and
   timeout recoveries.

   Our data shows that Group 2 recovery latency is only 0.3% lower than
   the Group 1 recovery latency.  But Group 3 recovery latency is 25%
   lower than Group 1 due to a 40% reduction in RTO-triggered
   recoveries!  Therefore it is important to implement both TLP and RACK
   for performance.  Group 4's total recovery latency is 0.02% lower
   than Group 3's, indicating that RACK plus TLP can successfully
   replace RFC3517 as a standalone recovery mechanism.

   We want to emphasize that the current experiment is limited in terms
   of network coverage.  The connectivity in Western Europe is fairly
   good, therefore loss recovery is not a major performance bottleneck.
   We plan to expand our experiments to regions with worse connectivity,
   in particular on networks with strong traffic policing.

9.  Security Considerations

   RACK does not change the risk profile for TCP.

   An interesting scenario is ACK-splitting attacks [SCWA99]: for an
   MSS-size packet sent, the receiver or the attacker might send MSS
   ACKs that SACK or acknowledge one additional byte per ACK.  This

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   would not fool RACK.  RACK.xmit_ts would not advance because all the
   sequences of the packet are transmitted at the same time (carry the
   same transmission timestamp).  In other words, SACKing only one byte
   of a packet or SACKing the packet in entirety have the same effect on

10.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an

11.  Acknowledgments

   The authors thank Matt Mathis for his insights in FACK and Michael
   Welzl for his per-packet timer idea that inspired this work.  Eric
   Dumazet, Randy Stewart, Van Jacobson, Ian Swett, Rick Jones, and Jana
   Iyengar contributed to the draft and the implementations in Linux,
   FreeBSD and QUIC.

12.  References

12.1.  Normative References

   [RFC793]   Postel, J., "Transmission Control Protocol", September

   [RFC2018]  Mathis, M. and J. Mahdavi, "TCP Selective Acknowledgment
              Options", RFC 2018, October 1996.

   [RFC6937]  Mathis, M., Dukkipati, N., and Y. Cheng, "Proportional
              Rate Reduction for TCP", May 2013.

   [RFC4737]  Morton, A., Ciavattone, L., Ramachandran, G., Shalunov,
              S., and J. Perser, "Packet Reordering Metrics", RFC 4737,
              November 2006.

   [RFC6675]  Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M.,
              and Y. Nishida, "A Conservative Loss Recovery Algorithm
              Based on Selective Acknowledgment (SACK) for TCP",
              RFC 6675, August 2012.

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298, June

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   [RFC5827]  Allman, M., Ayesta, U., Wang, L., Blanton, J., and P.
              Hurtig, "Early Retransmit for TCP and Stream Control
              Transmission Protocol (SCTP)", RFC 5827, April 2010.

   [RFC5682]  Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
              "Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
              Spurious Retransmission Timeouts with TCP", RFC 5682,
              September 2009.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", RFC 2119, March 1997.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

   [RFC2883]  Floyd, S., Mahdavi, J., Mathis, M., and M. Podolsky, "An
              Extension to the Selective Acknowledgement (SACK) Option
              for TCP", RFC 2883, July 2000.

   [RFC7323]  Borman, D., Braden, B., Jacobson, V., and R.
              Scheffenegger, "TCP Extensions for High Performance",
              September 2014.

12.2.  Informative References

   [FACK]     Mathis, M. and M. Jamshid, "Forward acknowledgement:
              refining TCP congestion control", ACM SIGCOMM Computer
              Communication Review, Volume 26, Issue 4, Oct. 1996. ,

   [TLP]      Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
              "Tail Loss Probe (TLP): An Algorithm for Fast Recovery of
              Tail Drops", draft-dukkipati-tcpm-tcp-loss-probe-01 (work
              in progress), August 2013.

   [RFC7765]  Hurtig, P., Brunstrom, A., Petlund, A., and M. Welzl, "TCP
              and SCTP RTO Restart", February 2016.

              Zimmermann, A., Schulte, L., Wolff, C., and A. Hannemann,
              "Detection and Quantification of Packet Reordering with
              TCP", draft-zimmermann-tcpm-reordering-detection-02 (work
              in progress), November 2014.

   [QUIC-LR]  Iyengar, J. and I. Swett, "QUIC Loss Recovery And
              Congestion Control", draft-tsvwg-quic-loss-recovery-01
              (work in progress), June 2016.

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              Petlund, A., Evensen, K., Griwodz, C., and P. Halvorsen,
              "TCP enhancements for interactive thin-stream
              applications", NOSSDAV , 2008.

   [SCWA99]   Savage, S., Cardwell, N., Wetherall, D., and T. Anderson,
              "TCP Congestion Control With a Misbehaving Receiver", ACM
              Computer Communication Review, 29(5) , 1999.

              Flach, T., Papageorge, P., Terzis, A., Pedrosa, L., Cheng,
              Y., Karim, T., Katz-Bassett, E., and R. Govindan, "An
              Analysis of Traffic Policing in the Web", ACM SIGCOMM ,

Authors' Addresses

   Yuchung Cheng
   Google, Inc
   1600 Amphitheater Parkway
   Mountain View, California  94043


   Neal Cardwell
   Google, Inc
   76 Ninth Avenue
   New York, NY  10011


   Nandita Dukkipati
   Google, Inc
   1600 Amphitheater Parkway
   Mountain View, California  94043


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