Internet DRAFT - draft-mdt-ippm-explicit-flow-measurements
draft-mdt-ippm-explicit-flow-measurements
IPPM M. Cociglio
Internet-Draft Telecom Italia - TIM
Intended status: Informational A. Ferrieux
Expires: January 13, 2022 Orange Labs
G. Fioccola
Huawei Technologies
I. Lubashev
Akamai Technologies
F. Bulgarella
Telecom Italia - TIM
I. Hamchaoui
Orange Labs
M. Nilo
Telecom Italia - TIM
R. Sisto
Politecnico di Torino
D. Tikhonov
LiteSpeed Technologies
July 12, 2021
Explicit Flow Measurements Techniques
draft-mdt-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/.
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This Internet-Draft will expire on January 13, 2022.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Notational Conventions . . . . . . . . . . . . . . . . . . . 4
3. Latency Bits . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Spin Bit . . . . . . . . . . . . . . . . . . . . . . . . 5
3.2. Delay Bit . . . . . . . . . . . . . . . . . . . . . . . . 6
3.2.1. Generation Phase . . . . . . . . . . . . . . . . . . 8
3.2.2. Reflection Phase . . . . . . . . . . . . . . . . . . 8
3.2.3. T_Max Selection . . . . . . . . . . . . . . . . . . . 9
3.2.4. Delay Measurement using Delay Bit . . . . . . . . . . 10
3.2.5. Observer's Algorithm . . . . . . . . . . . . . . . . 12
3.2.6. Two Bits Delay Measurement: Spin Bit + Delay Bit . . 13
3.2.7. Hidden Delay Bit - Delay Bit with Privacy Protection 13
4. Loss Bits . . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.1. T Bit - Round Trip Loss Bit . . . . . . . . . . . . . . . 14
4.1.1. Round Trip Packet Loss Measurement . . . . . . . . . 15
4.1.2. Setting the Round Trip Loss Bit on Outgoing Packets . 17
4.1.3. Observer's Logic for Round Trip Loss Signal . . . . . 18
4.1.4. Loss Coverage and Signal Timing . . . . . . . . . . . 19
4.2. Q Bit - Square Bit . . . . . . . . . . . . . . . . . . . 19
4.2.1. Q Block Length Selection . . . . . . . . . . . . . . 19
4.2.2. Upstream Loss . . . . . . . . . . . . . . . . . . . . 20
4.2.3. Identifying Q Block Boundaries . . . . . . . . . . . 21
4.3. L Bit - Loss Event Bit . . . . . . . . . . . . . . . . . 21
4.3.1. End-To-End Loss . . . . . . . . . . . . . . . . . . . 22
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4.3.2. Loss Profile Characterization . . . . . . . . . . . . 22
4.4. L+Q Bits - Upstream, Downstream, and End-to-End Loss
Measurements . . . . . . . . . . . . . . . . . . . . . . 22
4.4.1. Correlating End-to-End and Upstream Loss . . . . . . 23
4.5. R Bit - Reflection Square Bit . . . . . . . . . . . . . . 24
4.5.1. R+Q Bits - Using R and Q Bits for Passive Loss
Measurement . . . . . . . . . . . . . . . . . . . . . 25
4.5.2. Enhancement of R Block Length Computation . . . . . . 29
4.5.3. Improved Resilience to Packet Reordering . . . . . . 29
4.6. Improved Q and R Bits Resilience to Burst Losses . . . . 29
5. Summary of Delay and Loss Marking Methods . . . . . . . . . . 30
6. ECN-Echo Event Bit . . . . . . . . . . . . . . . . . . . . . 32
6.1. Setting the ECN-Echo Event Bit on Outgoing Packets . . . 32
6.2. Using E Bit for Passive ECN-Reported Congestion
Measurement . . . . . . . . . . . . . . . . . . . . . . . 32
7. Protocol Ossification Considerations . . . . . . . . . . . . 33
8. Examples of Application . . . . . . . . . . . . . . . . . . . 33
8.1. QUIC . . . . . . . . . . . . . . . . . . . . . . . . . . 33
8.2. TCP . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
9. Security Considerations . . . . . . . . . . . . . . . . . . . 34
9.1. Optimistic ACK Attack . . . . . . . . . . . . . . . . . . 35
10. Privacy Considerations . . . . . . . . . . . . . . . . . . . 35
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 36
12. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . 36
13. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 36
14. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 36
15. References . . . . . . . . . . . . . . . . . . . . . . . . . 36
15.1. Normative References . . . . . . . . . . . . . . . . . . 36
15.2. Informative References . . . . . . . . . . . . . . . . . 37
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 39
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
acknowledgment 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.
With encrypted protocols, the equivalent transport headers are
encrypted and passive packet loss and delay observations are not
possible, as described in [TRANSPORT-ENCRYPT].
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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. 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 encrypted 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
(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].
2. Notational Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
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3. 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.
3.1. 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 3.2.
3.2. 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.
Delay bit mechanism
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3.2.1. 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 3.2.2.
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 3.2.3 for details.
3.2.2. 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.
In both cases, if the outgoing delay sample is being transmitted with
a delay greater than a predetermined threshold after the reception of
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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.
3.2.3. 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 waits for two consecutive incoming samples and computes the
two related RTTs. Then it takes the largest of them as the basis of
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the "T_Max_c" formula. At this point, observers have already
measured a valid RTT and then computed their "T_Max_c".
3.2.4. 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.
3.2.4.1. 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
Round-trip time (both direction)
3.2.4.2. 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.
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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
Half Round-trip time (both direction)
3.2.4.3. 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
Intra-domain Round-trip time (client-observer: upstream)
3.2.5. 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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3.2.6. 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.
3.2.7. Hidden Delay Bit - Delay Bit with Privacy Protection
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). To
protect users privacy, the algorithm of the delay bit can be slightly
modified to mask the RTT of the connection to an intermediate
observer. This result can be achieved using a simple expedient which
consists in delaying the client-side reflection of the delay sample
by a predetermined time value. This would lead an intermediate
observer to inevitably measure a delay greater than the real one.
The 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 instance, a
new Additional Delay value could be generated whenever the client's
IP address changes.
Using this technique, despite the Additional Delay introduced, it is
still possible to correctly measure the right component of RTT
(observer-server) and all the intra-domain measurements used to
distribute the delay in the network. Furthermore, differently from
the Delay Bit, the hidden Delay Bit makes the use of the client
reflection threshold (1ms) redundant. Removing this threshold leads
to the further advantage of increasing the number of valid
measurements produced by the algorithm.
4. 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.
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- T: the "round Trip loss" bit is used in combination with the Spin
bit to measure round-trip loss. See Section 4.1.
- Q: the "sQuare signal" bit is used to measure upstream loss. See
Section 4.2.
- L: the "Loss event" bit is used to measure end-to-end loss. See
Section 4.3.
- R: the "Reflection square signal" bit is used in combination with
Q bit to measure end-to-end loss. See Section 4.1.
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.
4.1. 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;
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- 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 4.1.2 for details on packet
generation and for a mechanism to allow an observer to distinguish
between trains belonging to different phases (Generation and
Reflection).
4.1.1. Round Trip Packet Loss Measurement
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.
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=======================|======================>
= ********** -----Obs----> ********** =
= * Client * * Server * =
= ********** <------------ ********** =
<==============================================
(a) client-server RTPL
==============================================>
= ********** ------------> ********** =
= * Client * * Server * =
= ********** <----Obs----- ********** =
<======================|=======================
(b) server-client RTPL
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
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
Intra-domain Round-trip packet loss (observer-server)
4.1.2. 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 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; it maintains a "generation
token" count that is reset to zero at the beginning of the
algorithm phase and is incremented every time a packet arrives.
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;
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
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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.
4.1.3. 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.
In the following example, packets are represented by two bits (first
one is the spin bit, second one is the 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
Round Trip Loss signal example
Note that 5 marked packets have been generated of which 4 have been
reflected.
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4.1.4. 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.
4.2. Q Bit - Square Bit
The sQuare bit (Q bit) takes its name from the square wave generated
by its signal. Every outgoing packet contains the Q bit value, which
is initialized to the 0 and inverted after sending N packets (a
sQuare Block or simply Q Block). Hence, Q Period is 2*N. The Q bit
represents "packet color" as defined by [AltMark].
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 4.2.2.
4.2.1. 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 4.2.2).
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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 7).
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
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.
4.2.2. 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 4.2.3.
=====================>
********** -----Obs----> **********
* Client * * Server *
********** <------------ **********
(a) in client-server channel (uloss_up)
********** ------------> **********
* Client * * Server *
********** <----Obs----- **********
<=====================
(b) in server-client channel (uloss_down)
Upstream loss
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4.2.3. 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".
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 reordring, but it may erroneously coalesce packets from
multiple Q Blocks into fewer Q Blocks, if loss exceeds 50% for some Q
Blocks.
4.3. 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.
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
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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 4.3.1.
4.3.1. 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.
4.3.2. Loss Profile Characterization
In addition to measuring the end-to-end loss rate, 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 4.4.1). 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.
4.4. L+Q Bits - Upstream, Downstream, and End-to-End Loss Measurements
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 4.2.2)
- downstream loss: observer-to-receiver loss (see Section 4.4.1.1)
- end-to-end loss: sender-to-receiver loss on the observed path (see
Section 4.3.1) with loss profile characterization (see
Section 4.3.2)
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4.4.1. Correlating End-to-End and Upstream Loss
Upstream loss is calculated by observing packets that did not suffer
the upstream loss (Section 4.2.2). 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 acknowledgments) 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 3.1) or Delay bit (see Section 3.2).
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 4.3.1, then either the Q Period is too short
for the amount of packet reordering or there is observer loss,
described in Section 4.4.1.2. 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 like TCP and 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.
4.4.1.1. 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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4.4.1.2. 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 4.2.2). 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 4.3.1).
The need to adjust the upstream loss rate down to match end-to-end
loss rate as described in Section 4.4.1 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 4.2.2. 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.
4.5. R Bit - Reflection Square Bit
R bit requires a deployment alongside Q bit. Unlike the square
signal for which packets are transmitted into blocks of fixed size,
the Reflection square signal (being an alternate marking signal too)
produces blocks of packets whose size 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 the average size of the further received Q
Blocks. Implementation details follow.
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 opposite node
using the Q-bit). When this happens, the number of packets ("p"),
detected within this first Q Block, is used to generate a reflection
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square signal which 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)" is the average number of packets in a marking
period computed considering all the Q Blocks received since the
beginning of the current R Block.
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 {#endmarkingblock} should be used.
Looking at the R-bit, unidirectional observation points have an
indication of losses experienced by the entire unobserved channel
plus those occurred in the path from the sender up 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.
4.5.1. R+Q Bits - Using R and Q Bits for Passive Loss Measurement
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 4.2.2)
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- "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
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)
4.5.1.1. 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").
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=======================>
= ********** -----Obs----> **********
= * Client * * Server *
= ********** <------------ **********
<============================================
(a) in client-server channel (tqloss_up)
============================================>
********** ------------> ********** =
* Client * * Server * =
********** <----Obs----- ********** =
<=======================
(b) in server-client channel (tqloss_down)
Three-quarters connection loss
The following metrics derive from this last metric and the upstream
loss produced by the Q Bit.
4.5.1.2. 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)
End-To-End loss in the opposite direction
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4.5.1.3. 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)".
=======================>
= ********** ------|-----> **********
= * Client * Obs * Server *
= ********** <-----|------ **********
<=======================
(a) client-observer half round-trip loss (hrtloss_co)
=======================>
********** ------|-----> ********** =
* Client * Obs * Server * =
********** <-----|------ ********** =
<=======================
(b) observer-server half round-trip loss (hrtloss_os)
Half Round-trip loss (both direction)
4.5.1.4. 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 4.4.1.1 or using "half round-trip" loss and upstream loss in
the opposite direction.
For the latter, "dloss=(hrtloss-uloss_opposite)/(1-uloss_opposite)".
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=====================>
********** ------|-----> **********
* Client * Obs * Server *
********** <-----|------ **********
(a) in client-server channel (dloss_up)
********** ------|-----> **********
* Client * Obs * Server *
********** <-----|------ **********
<=====================
(b) in server-client channel (dloss_down)
Downstream loss
4.5.2. 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.
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.
4.5.3. 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 4.2.3.
This can be achieved by updating the size of the current R Block
while this 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.
4.6. Improved Q and R Bits Resilience to Burst Losses
Burst losses can affect Q and R measurements accuracy. Generally,
burst losses can be absorbed and correctly measured if smaller than
the established Q Block length. On the other hand, entire periods
might be wiped out if the burst sizes become too large thus making
the observer completely unaware of their loss.
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To improve burst loss resilience, an observer might consider a
received Q or R Block larger than the selected Q Block length as a
burst loss event. Then compute the loss as three times Q Block
length minus the measured block length. By doing so, an 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 equal or
greater than two consecutive periods would still remain unnoticed by
the observer (or underestimated if a period longer than Q Block
length were formed).
5. Summary of Delay and Loss Marking Methods
This section summarizes the marking methods described in this draft.
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 | | |
+---------------+----+------------+-----------+-------------+------+
|D^: Hidden | 1 | RTT^ | x2 | high | high |
| Delay Bit | | | Left Half^| | |
| | | | Right Half| | |
+---------------+----+------------+-----------+-------------+------+
|SD: Spin Bit & | 2 | RTT | x2 | high | very |
| Delay Bit *| | | Half RTT | | high |
+---------------+----+------------+-----------+-------------+------+
x2 Same metric for both directions
* Both algorithms work independtly; an observer could use
approximate spin bit measures when delay bit ones aren't available
^ Masked metric (real value can be calculated only by those who know
the Additional Delay)
Figure 1: Delay Comparison
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For the Loss measurement, each row in the table of Figure 2
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.
+-------------+-+-----------------------+-+------------------------+
| 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 (w/ or w/o 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 2: Loss Comparison
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6. ECN-Echo Event Bit
While the primary focus of the 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 6.1.
6.1. Setting the ECN-Echo Event Bit on Outgoing Packets
The Unreported ECN-Echo counter operates identically to Unreported
Loss counter (Section 4.3), 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].
6.2. 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.
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7. 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.
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.
8. Examples of Application
8.1. 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
experiments.
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|
+-+-+-+-+-+-+-+-+
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:
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0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0|1|S|Q|L|K|P|P|
+-+-+-+-+-+-+-+-+
Scheme 2A
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0|1|S|Q|R|K|P|P|
+-+-+-+-+-+-+-+-+
Scheme 2B
A further option would be to substitute the spin bit with the delay
bit (or hidden delay bit) leaving the two reserved bits for loss
detection. The proposed schemes are:
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+ +-+-+--+-+-+-+-+-+
|0|1|D|Q|L|K|P|P| OR |0|1|D^|Q|L|K|P|P|
+-+-+-+-+-+-+-+-+ +-+-+--+-+-+-+-+-+
Scheme 3A
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+ +-+-+--+-+-+-+-+-+
|0|1|D|Q|R|K|P|P| OR |0|1|D^|Q|R|K|P|P|
+-+-+-+-+-+-+-+-+ +-+-+--+-+-+-+-+-+
Scheme 3B
8.2. 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).
9. 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
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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.
9.1. 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.
10. 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
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incrementing Unreported Loss counter for loss of packets sent to a
different destination or with a different Connection ID.
11. IANA Considerations
This document makes no request of IANA.
12. Change Log
TBD
13. 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
14. Acknowledgements
TBD
15. References
15.1. 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>.
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[IP] Postel, J., "Internet Protocol", STD 5, RFC 791,
DOI 10.17487/RFC0791, September 1981,
<https://www.rfc-editor.org/info/rfc791>.
[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>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[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", STD 7,
RFC 793, DOI 10.17487/RFC0793, September 1981,
<https://www.rfc-editor.org/info/rfc793>.
15.2. Informative References
[ACCURATE]
Briscoe, B., Kuehlewind, M., and R. Scheffenegger, "More
Accurate ECN Feedback in TCP", draft-ietf-tcpm-accurate-
ecn-14 (work in progress), February 2021.
[AltMark] Fioccola, G., Ed., Capello, A., Cociglio, M., Castaldelli,
L., Chen, M., Zheng, L., Mirsky, G., and T. Mizrahi,
"Alternate-Marking Method for Passive and Hybrid
Performance Monitoring", RFC 8321, DOI 10.17487/RFC8321,
January 2018, <https://www.rfc-editor.org/info/rfc8321>.
[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.
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[I-D.trammell-ippm-spin]
Trammell, B., "An Explicit Transport-Layer Signal for
Hybrid RTT Measurement", draft-trammell-ippm-spin-00 (work
in progress), January 2019.
[I-D.trammell-tsvwg-spin]
Trammell, B., "A Transport-Independent Explicit Signal for
Hybrid RTT Measurement", draft-trammell-tsvwg-spin-00
(work in progress), July 2018.
[IPv6AltMark]
Fioccola, G., Zhou, T., Cociglio, M., Qin, F., and R.
Pang, "IPv6 Application of the Alternate Marking Method",
draft-ietf-6man-ipv6-alt-mark-04 (work in progress), March
2021.
[QUIC-TRANSPORT]
Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
and Secure Transport", draft-ietf-quic-transport-34 (work
in progress), January 2021.
[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>.
[SPIN-BIT]
Trammell, B., Vaere, P. D., Even, R., Fioccola, G.,
Fossati, T., Ihlar, M., Morton, A., and E. Stephan,
"Adding Explicit Passive Measurability of Two-Way Latency
to the QUIC Transport Protocol", draft-trammell-quic-
spin-03 (work in progress), May 2018.
[TRANSPORT-ENCRYPT]
Fairhurst, G. and C. Perkins, "Considerations around
Transport Header Confidentiality, Network Operations, and
the Evolution of Internet Transport Protocols", draft-
ietf-tsvwg-transport-encrypt-21 (work in progress), April
2021.
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[UDP-OPTIONS]
Touch, J., "Transport Options for UDP", draft-ietf-tsvwg-
udp-options-12 (work in progress), May 2021.
[UDP-SURPLUS]
Herbert, T., "UDP Surplus Header", draft-herbert-udp-
space-hdr-01 (work in progress), July 2019.
Authors' Addresses
Mauro Cociglio
Telecom Italia - TIM
Via Reiss Romoli, 274
Torino 10148
Italy
EMail: mauro.cociglio@telecomitalia.it
Alexandre Ferrieux
Orange Labs
EMail: alexandre.ferrieux@orange.com
Giuseppe Fioccola
Huawei Technologies
Riesstrasse, 25
Munich 80992
Germany
EMail: giuseppe.fioccola@huawei.com
Igor Lubashev
Akamai Technologies
EMail: ilubashe@akamai.com
Fabio Bulgarella
Telecom Italia - TIM
Via Reiss Romoli, 274
Torino 10148
Italy
EMail: fabio.bulgarella@guest.telecomitalia.it
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Isabelle Hamchaoui
Orange Labs
EMail: isabelle.hamchaoui@orange.com
Massimo Nilo
Telecom Italia - TIM
Via Reiss Romoli, 274
Torino 10148
Italy
EMail: massimo.nilo@telecomitalia.it
Riccardo Sisto
Politecnico di Torino
EMail: riccardo.sisto@polito.it
Dmitri Tikhonov
LiteSpeed Technologies
EMail: dtikhonov@litespeedtech.com
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