Internet DRAFT - draft-tcpm-rack
draft-tcpm-rack
TCP Maintenance Working Group Y. Cheng
Internet-Draft N. Cardwell
Intended status: Experimental Google, Inc
Expires: March 2, 2017 September 2, 2016
RACK: a time-based fast loss detection algorithm for TCP
draft-tcpm-rack-00
Abstract
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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Copyright (c) 2016 IETF Trust and the persons identified as the
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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
[RFC3517][RFC4653][RFC5827][RFC5681][RFC6675][RFC7765][FACK][THIN-
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-
prone.
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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 recovery
mechanism.
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
[RFC2018].
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 store 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. 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 any 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
finer.
"RACK.xmit_ts" is the most recent Packet.xmit_ts among all the
packets that were delivered (either cumulatively acknowledged or
selectively acknowledged) on the connection.
"RACK.end_seq" is the ending TCP sequence number of the packet that
was used to record the RACK.xmit_ts above.
"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
connection.
Note that the Packet.xmit_ts variable is per packet in flight. The
RACK.xmit_ts, RACK.RTT, RACK.reo_wnd, and RACK.min_RTT variables are
per connection.
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. We RECOMMEND implementing the
reordering detection in [REORDER-DETECT][RFC4737] to dynamically
adjust the reordering window. 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
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 has been retransmitted
before and either of two condition is true:
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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
changed.
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, the packet was likely lost.
If a packet that was sent later has been delivered, but the
reordering window has not passed, then it is not yet safe to deem the
given 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
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
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2. delta in time between the S/ACK of the RACK packet (RACK.ack_ts)
and now
So we mark a packet as lost if:
RACK.xmit_ts > Packet.xmit_ts AND
(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.
RACK_detect_loss():
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
If Packet.xmit_ts == RACK.xmit_ts AND // Timestamp tie breaker 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
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6. Analysis and Discussion
6.1. 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 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 (as a tail
drop) 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 due
to application-limited behavior. 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,
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then RACK will still falsely mark P1 and P2 lost. We discuss how to
reduce the false positives in the end of this section.
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
scoreboard.
6.2. 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) to track
transmission times. In contrast, the conventional approach requires
one variable to track number of duplicate ACK threshold.
6.3. Adjusting the reordering window
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). Alternatively, RACK can use the
smoothed RTT used in RTT estimation [RFC6298]. However, smoothed RTT
can be significantly inflated by orders of magnitude due to
congestion and buffer-bloat, which would result in an overly
conservative reordering window and slow loss detection. Furthermore,
RACK uses a quarter of minimum RTT because Linux TCP uses 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.
One potential improvement is 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
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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 research.
6.4. 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] and nonstandard ones
[FACK][THIN-STREAM]. While RACK can be a supplemental loss detection
on top of these algorithms, this is not necessary, because the 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 timer calculation nor
detects spurious timeouts.
Furthermore, RACK naturally works well with Tail Loss Probe [TLP]
because a tail loss probe solicit seither 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 can be
modified to retransmit the first unacknowledged packet, which can
improve application latency.
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6.5. 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]. The distinction between
fast recovery or RTO recovery is not necessary 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.
6.6. RACK for other transport protocols
RACK can be implemented in other transport protocols. The algorithm
can skip step 3 and simplify if the protocol can support unique
transmission or packet identifier (e.g. TCP echo options). For
example, the QUIC protocol implements RACK [QUIC-LR] .
7. 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
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
RACK.
8. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
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9. Acknowledgments
The authors thank Matt Mathis for his insights in FACK and Michael
Welzl for his per-packet timer idea that inspired this work. Nandita
Dukkipati, Eric Dumazet, Randy Stewart, Van Jacobson, Ian Swett, and
Jana Iyengar contributed to the algorithm and the implementations in
Linux, FreeBSD and QUIC.
10. References
10.1. Normative References
[RFC793] Postel, J., "Transmission Control Protocol", September
1981.
[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
2011.
[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.
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[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.
10.2. 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. ,
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.
[REORDER-DETECT]
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.
[THIN-STREAM]
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.
[POLICER16]
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 ,
2016.
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Authors' Addresses
Yuchung Cheng
Google, Inc
1600 Amphitheater Parkway
Mountain View, California 94043
USA
Email: ycheng@google.com
Neal Cardwell
Google, Inc
76 9th Avenue
New York, NY 10011
USA
Email: ncardwell@google.com
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