| Rfc | 3366 |
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| Title | Advice to link designers on link Automatic Repeat reQuest (ARQ) |
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| Author | G.
Fairhurst, L. Wood |
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| Date | August 2002 |
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| Format: | TXT, HTML |
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| Also | BCP0062 |
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| Status: | BEST CURRENT PRACTICE |
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|
Network Working Group G. Fairhurst
Request for Comments: 3366 University of Aberdeen
BCP: 62 L. Wood
Category: Best Current Practice Cisco Systems Ltd
August 2002
Advice to link designers on link Automatic Repeat reQuest (ARQ)
Status of this Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2002). All Rights Reserved.
Abstract
This document provides advice to the designers of digital
communication equipment and link-layer protocols employing link-layer
Automatic Repeat reQuest (ARQ) techniques. This document presumes
that the designers wish to support Internet protocols, but may be
unfamiliar with the architecture of the Internet and with the
implications of their design choices for the performance and
efficiency of Internet traffic carried over their links.
ARQ is described in a general way that includes its use over a wide
range of underlying physical media, including cellular wireless,
wireless LANs, RF links, and other types of channel. This document
also describes issues relevant to supporting IP traffic over
physical-layer channels where performance varies, and where link ARQ
is likely to be used.
Table of Contents
1. Introduction. . . . . . . . . . . . . . . . . . . . . . . . .2
1.1 Link ARQ. . . . . . . . . . . . . . . . . . . . . . . . . . .4
1.2 Causes of Packet Loss on a Link . . . . . . . . . . . . . . .5
1.3 Why Use ARQ?. . . . . . . . . . . . . . . . . . . . . . . . .6
1.4 Commonly-used ARQ Techniques. . . . . . . . . . . . . . . . .7
1.4.1 Stop-and-wait ARQ . . . . . . . . . . . . . . . . . . . . . .7
1.4.2 Sliding-Window ARQ. . . . . . . . . . . . . . . . . . . . . .7
1.5 Causes of Delay Across a Link . . . . . . . . . . . . . . . .8
2. ARQ Persistence . . . . . . . . . . . . . . . . . . . . . . 10
2.1 Perfectly-Persistent (Reliable) ARQ Protocols . . . . . . . 10
2.2 High-Persistence (Highly-Reliable) ARQ Protocols. . . . . . 12
2.3 Low-Persistence (Partially-Reliable) ARQ Protocols. . . . . 13
2.4 Choosing Your Persistency . . . . . . . . . . . . . . . . . 13
2.5 Impact of Link Outages. . . . . . . . . . . . . . . . . . . 14
3. Treatment of Packets and Flows. . . . . . . . . . . . . . . 15
3.1 Packet Ordering . . . . . . . . . . . . . . . . . . . . . . 15
3.2 Using Link ARQ to Support Multiple Flows. . . . . . . . . . 16
3.3 Differentiation of Link Service Classes . . . . . . . . . . 17
4. Conclusions . . . . . . . . . . . . . . . . . . . . . . . . 19
5. Security Considerations . . . . . . . . . . . . . . . . . . 21
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . 21
7. Acknowledgements. . . . . . . . . . . . . . . . . . . . . . 22
8. References. . . . . . . . . . . . . . . . . . . . . . . . . 22
8.1 Normative References. . . . . . . . . . . . . . . . . . . . 22
8.2 Informative References. . . . . . . . . . . . . . . . . . . 23
9. Authors' Addresses. . . . . . . . . . . . . . . . . . . . . 26
10. Full Copyright Statement. . . . . . . . . . . . . . . . . . 27
1. Introduction
IP, the Internet Protocol [RFC791], forms the core protocol of the
global Internet and defines a simple "connectionless" packet-switched
network. Over the years, Internet traffic using IP has been carried
over a wide variety of links, of vastly different capacities, delays
and loss characteristics. In the future, IP traffic can be expected
to continue to be carried over a very wide variety of new and
existing link designs, again of varied characteristics.
A companion document [DRAFTKARN02] describes the general issues
associated with link design. This document should be read in
conjunction with that and with other documents produced by the
Performance Implications of Link Characteristics (PILC) IETF
workgroup [RFC3135, RFC3155].
This document is intended for three distinct groups of readers:
a. Link designers wishing to configure (or tune) a link for the IP
traffic that it will carry, using standard link-layer mechanisms
such as the ISO High-level Data Link Control (HDLC) [ISO4335a] or
its derivatives.
b. Link implementers who may wish to design new link mechanisms that
perform well for IP traffic.
c. The community of people using or developing TCP, UDP and related
protocols, who may wish to be aware of the ways in which links
can operate.
The primary audiences are intended to be groups (a) and (b). Group
(c) should not need to be aware of the exact details of an ARQ scheme
across a single link, and should not have to consider such details
for protocol implementations; much of the Internet runs across links
that do not use any form of ARQ. However, the TCP/IP community does
need to be aware that the IP protocol operates over a diverse range
of underlying subnetworks. This document may help to raise that
awareness.
Perfect reliability is not a requirement for IP networks, nor is it a
requirement for links [DRAFTKARN02]. IP networks may discard packets
due to a variety of reasons entirely unrelated to channel errors,
including lack of queuing space, congestion management, faults, and
route changes. It has long been widely understood that perfect
end-to-end reliability can be ensured only at, or above, the
transport layer [SALT81].
Some familiarity with TCP, the Transmission Control Protocol [RFC793,
STE94], is presumed here. TCP provides a reliable byte-stream
transport service, building upon the best-effort datagram delivery
service provided by the Internet Protocol. TCP achieves this by
dividing data into TCP segments, and transporting these segments in
IP packets. TCP guarantees that a TCP session will retransmit the
TCP segments contained in any data packets that are lost along the
Internet path between endhosts. TCP normally performs retransmission
using its Fast Retransmit procedure, but if the loss fails to be
detected (or retransmission is unsuccessful), TCP falls back to a
Retransmission Time Out (RTO) retransmission using a timer [RFC2581,
RFC2988]. (Link protocols also implement timers to verify integrity
of the link, and to assist link ARQ.) TCP also copes with any
duplication or reordering introduced by the IP network. There are a
number of variants of TCP, with differing levels of sophistication in
their procedures for handling loss recovery and congestion avoidance.
Far from being static, the TCP protocol is itself subject to ongoing
gradual refinement and evolution, e.g., [RFC2488, RFC2760].
Internet networks may reasonably be expected to carry traffic from a
wide and evolving range of applications. Not all applications
require or benefit from using the reliable service provided by TCP.
In the Internet, these applications are carried by alternate
transport protocols, such as the User Datagram Protocol (UDP)
[RFC768].
1.1 Link ARQ
At the link layer, ARQ operates on blocks of data, known as frames,
and attempts to deliver frames from the link sender to the link
receiver over a channel. The channel provides the physical-layer
connection over which the link protocol operates. In its simplest
form, a channel may be a direct physical-layer connection between the
two link nodes (e.g., across a length of cable or over a wireless
medium). ARQ may also be used edge-to-edge across a subnetwork,
where the path includes more than one physical-layer medium. Frames
often have a small fixed or maximum size for convenience of
processing by Medium-Access Control (MAC) and link protocols. This
contrasts with the variable lengths of IP datagrams, or 'packets'. A
link-layer frame may contain all, or part of, one or more IP packets.
A link ARQ mechanism relies on an integrity check for each frame
(e.g., strong link-layer CRC [DRAFTKARN02]) to detect channel errors,
and uses a retransmission process to retransmit lost (i.e., missing
or corrupted) frames.
Links may be full-duplex (allowing two-way communication over
separate forward and reverse channels), half-duplex (where two-way
communication uses a shared forward and reverse channel, e.g., IrDA,
IEEE 802.11) or simplex (where a single channel permits communication
in only one direction).
ARQ requires both a forward and return path, and therefore link ARQ
may be used over links that employ full- or half-duplex links. When
a channel is shared between two or more link nodes, a link MAC
protocol is required to ensure all nodes requiring transmission can
gain access to the shared channel. Such schemes may add to the delay
(jitter) associated with transmission of packet data and ARQ control
frames.
When using ARQ over a link where the probability of frame loss is
related to the frame size, there is an optimal frame size for any
specific target channel error rate. To allow for efficient use of
the channel, this maximum link frame size may be considerably lower
than the maximum IP datagram size specified by the IP Maximum
Transmission Unit (MTU). Each frame will then contain only a
fraction of an IP packet, and transparent implicit fragmentation of
the IP datagram is used [DRAFTKARN02]. A smaller frame size
introduces more frame header overhead per payload byte transported.
Explicit network-layer IP fragmentation is undesirable for a variety
of reasons, and should be avoided [KEN87, DRAFTKARN02]. Its use can
be minimized with use of Path MTU discovery [RFC1191, RFC1435,
RFC1981].
Another way to reduce the frame loss rate (or reduce transmit signal
power for the same rate of frame loss) is to use coding, e.g.,
Forward Error Correction (FEC) [LIN93].
FEC is commonly included in the physical-layer design of wireless
links and may be used simultaneously with link ARQ. FEC schemes
which combine modulation and coding also exist, and may also be
adaptive. Hybrid ARQ [LIN93] combines adaptive FEC with link ARQ
procedures to reduce the probability of loss of retransmitted frames.
Interleaving may also be used to reduce the probability of frame loss
by dispersing the occurrence of errors more widely in the channel to
improve error recovery; this adds further delay to the channel's
existing propagation delay.
The document does not consider the use of link ARQ to support a
broadcast or multicast service within a subnetwork, where a link may
send a single packet to more than one recipient using a single link
transmit operation. Although such schemes are supported in some
subnetworks, they raise a number of additional issues not examined
here.
Links supporting stateful reservation-based quality of service (QoS)
according to the Integrated Services (intserv) model are also not
explicitly discussed.
1.2 Causes of Packet Loss on a Link
Not all packets sent to a link are necessarily received successfully
by the receiver at the other end of the link. There are a number of
possible causes of packet loss. These may occur as frames travel
across a link, and include:
a. Loss due to channel noise, often characterised by random frame
loss. Channel noise may also result from other traffic degrading
channel conditions.
b. Frame loss due to channel interference. This interference can
be random, structured, and in some cases even periodic.
c. A link outage, a period during which the link loses all or
virtually all frames, until the link is restored. This is a
common characteristic of some types of link, e.g., mobile cellular
radio.
Other forms of packet loss are not related to channel conditions,
but include:
i. Loss of a frame transmitted in a shared channel where a
contention-aware MAC protocol is used (e.g., due to collision).
Here, many protocols require that retransmission is deferred to
promote stability of the shared channel (i.e., prevent excessive
channel contention). This is discussed further in section 1.5.
ii. Packet discards due to congestion. Queues will eventually
overflow as the arrival rate of new packets to send continues to
exceed the outgoing packet transmission rate over the link.
iii. Loss due to implementation errors, including hardware faults
and software errors. This is recognised as a common cause of
packet corruption detected in the endhosts [STONE00].
The rate of loss and patterns of loss experienced are functions of
the design of the physical and link layers. These vary significantly
across different link configurations. The performance of a specific
implementation may also vary considerably across the same link
configuration when operated over different types of channel.
1.3 Why Use ARQ?
Reasons that encourage considering the use of ARQ include:
a. ARQ across a single link has a faster control loop than TCP's
acknowledgement control loop, which takes place over the longer
end-to-end path over which TCP must operate. It is therefore
possible for ARQ to provide more rapid retransmission of TCP
segments lost on the link, at least for a reasonable number of
retries [RFC3155, SALT81].
b. Link ARQ can operate on individual frames, using implicit
transparent link fragmentation [DRAFTKARN02]. Frames may be
much smaller than IP packets, and repetition of smaller frames
containing lost or errored parts of an IP packet may improve the
efficiency of the ARQ process and the efficiency of the link.
A link ARQ procedure may be able to use local knowledge that is not
available to endhosts, to optimise delivery performance for the
current link conditions. This information can include information
about the state of the link and channel, e.g., knowledge of the
current available transmission rate, the prevailing error
environment, or available transmit power in wireless links.
1.4 Commonly-used ARQ Techniques
A link ARQ protocol uses a link protocol mechanism to allow the
sender to detect lost or corrupted frames and to schedule
retransmission. Detection of frame loss may be via a link protocol
timer, by detecting missing positive link acknowledgement frames, by
receiving explicit negative acknowledgement frames and/or by polling
the link receiver status.
Whatever mechanisms are chosen, there are two easily-described
categories of ARQ retransmission process that are widely used:
1.4.1 Stop-And-Wait ARQ
A sender using stop-and-wait ARQ (sometimes known as 'Idle ARQ'
[LIN93]) transmits a single frame and then waits for an
acknowledgement from the receiver for that frame. The sender then
either continues transmission with the next frame, or repeats
transmission of the same frame if the acknowledgement indicates that
the original frame was lost or corrupted.
Stop-and-wait ARQ is simple, if inefficient, for protocol designers
to implement, and therefore popular, e.g., tftp [RFC1350] at the
transport layer. However, when stop-and-wait ARQ is used in the link
layer, it is well-suited only to links with low bandwidth-delay
products. This technique is not discussed further in this document.
1.4.2 Sliding-Window ARQ
A protocol using sliding-window link ARQ [LIN93] numbers every frame
with a unique sequence number, according to a modulus. The modulus
defines the numbering base for frame sequence numbers, and the size
of the sequence space. The largest sequence number value is viewed
by the link protocol as contiguous with the first (0), since the
numbering space wraps around.
TCP is itself a sliding-window protocol at the transport layer
[STE94], so similarities between a link-interface-to-link-interface
protocol and end-to-end TCP may be recognisable. A sliding-window
link protocol is much more complex in implementation than the simpler
stop-and-wait protocol described in the previous section,
particularly if per-flow ordering is preserved.
At any time the link sender may have a number of frames outstanding
and awaiting acknowledgement, up to the space available in its
transmission window. A sufficiently-large link sender window
(equivalent to or greater than the number of frames sent, or larger
than the bandwidth*delay product capacity of the link) permits
continuous transmission of new frames. A smaller link sender window
causes the sender to pause transmission of new frames until a timeout
or a control frame, such as an acknowledgement, is received. When
frames are lost, a larger window, i.e., more than the link's
bandwidth*delay product, is needed to allow continuous operation
while frame retransmission takes place.
The modulus numbering space determines the size of the frame header
sequence number field. This sequence space needs to be larger than
the link window size and, if using selective repeat ARQ, larger than
twice the link window size. For continuous operation, the sequence
space should be larger than the product of the link capacity and the
link ARQ persistence (discussed in section 2), so that in-flight
frames can be identified uniquely.
As with TCP, which provides sliding-window delivery across an entire
end-to-end path rather than across a single link, there are a large
number of variations on the basic sliding-window implementation, with
increased complexity and sophistication to make them suitable for
various conditions. Selective Repeat (SR), also known as Selective
Reject (SREJ), and Go-Back-N, also known as Reject (REJ), are
examples of ARQ techniques using protocols implementing sliding
window ARQ.
1.5 Causes of Delay Across a Link
Links and link protocols contribute to the total path delay
experienced between communicating applications on endhosts. Delay
has a number of causes, including:
a. Input packet queuing and frame buffering at the link head before
transmission over the channel.
b. Retransmission back-off, an additional delay introduced for
retransmissions by some MAC schemes when operating over a shared
channel to prevent excessive contention. A high level of
contention may otherwise arise, if, for example, a set of link
receivers all retransmitted immediately after a collision on a
busy shared channel. Link ARQ protocols designed for shared
channels may select a backoff delay, which increases with the
number of attempts taken to retransmit a frame; analogies can be
drawn with end-to-end TCP congestion avoidance at the transport
layer [RFC2581]. In contrast, a link over a dedicated channel
(which has capacity pre-allocated to the link) may send a
retransmission at the earliest possible time.
c. Waiting for access to the allocated channel when the channel is
shared. There may be processing or protocol-induced delay
before transmission takes place [FER99, PAR00].
d. Frame serialisation and transmission processing. These are
functions of frame size and the transmission speed of the link.
e. Physical-layer propagation time, limited by the speed of
transmission of the signal in the physical medium of the
channel.
f. Per-frame processing, including the cost of QoS scheduling,
encryption, FEC and interleaving. FEC and interleaving also add
substantial delay and, in some cases, additional jitter. Hybrid
link ARQ schemes [LIN93], in particular, may incur significant
receiver processing delay.
g. Packet processing, including buffering frame contents at the
link receiver for packet reassembly, before onward transmission
of the packet.
When link ARQ is used, steps (b), (c), (d), (e), and (f) may be
repeated a number of times, every time that retransmission of a frame
occurs, increasing overall delay for the packet carried in part by
the frame. Adaptive ARQ schemes (e.g., hybrid ARQ using adaptive FEC
codes) may also incur extra per-frame processing for retransmitted
frames.
It is important to understand that applications and transport
protocols at the endhosts are unaware of the individual delays
contributed by each link in the path, and only see the overall path
delay. Application performance is therefore determined by the
cumulative delay of the entire end-to-end Internet path. This path
may include an arbitrary or even a widely-fluctuating number of
links, where any link may or may not use ARQ. As a result, it is not
possible to state fixed limits on the acceptable delay that a link
can add to a path; other links in the path will add an unknown delay.
2. ARQ Persistence
ARQ protocols may be characterised by their persistency. Persistence
is the willingness of the protocol to retransmit lost frames to
ensure reliable delivery of traffic across the link.
A link's retransmission persistency defines how long the link is
allowed to delay a packet, in an attempt to transmit all the frames
carrying the packet and its content over the link, before giving up
and discarding the packet. This persistency can normally be measured
in milliseconds, but may, if the link propagation delay is specified,
be expressed in terms of the maximum number of link retransmission
attempts permitted. The latter does not always map onto an exact
time limit, for the reasons discussed in section 1.5.
An example of a reliable link protocol that is perfectly persistent
is the ISO HDLC protocol in the Asynchronous Balanced Mode (ABM)
[ISO4335a].
A protocol that only retransmits a number of times before giving up
is less persistent, e.g., Ethernet [FER99], IEEE 802.11, or GSM RLP
[RFC2757]. Here, lower persistence also ensures stability and fair
sharing of a shared channel, even when many senders are attempting
retransmissions.
TCP, STCP [RFC2960] and a number of applications using UDP (e.g.,
tftp) implement their own end-to-end reliable delivery mechanisms.
Many TCP and UDP applications, e.g., streaming multimedia, benefit
from timely delivery from lower layers with sufficient reliability,
rather than perfect reliability with increased link delays.
2.1 Perfectly-Persistent (Reliable) ARQ Protocols
A perfectly-persistent ARQ protocol is one that attempts to provide a
reliable service, i.e., in-order delivery of packets to the other end
of the link, with no missing packets and no duplicate packets. The
perfectly-persistent ARQ protocol will repeat a lost or corrupted
frame an indefinite (and potentially infinite) number of times until
the frame is successfully received.
If traffic is going no further than across one link, and losses do
not occur within the endhosts, perfect persistence ensures
reliability between the two link ends without requiring any
higher-layer protocols. This reliability can become
counterproductive for traffic traversing multiple links, as it
duplicates and interacts with functionality in protocol mechanisms at
higher layers [SALT81].
Arguments against the use of perfect persistence for IP traffic
include:
a. Variable link delay; the impact of ARQ introduces a degree of
jitter, a function of the physical-layer delay and frame
serialisation and transmission times (discussed in section 1.5),
to all flows sharing a link performing frame retransmission.
b. Perfect persistence does not provide a clear upper bound on the
maximum retransmission delay for the link. Significant changes
in path delay caused by excessive link retransmissions may lead
to timeouts of TCP retransmission timers, although a high
variance in link delay and the resulting overall path delay may
also cause a large TCP RTO value to be selected [LUD99b, PAR00].
This will alter TCP throughput, decreasing overall performance,
but, in mitigation, it can also decrease the occurrence of
timeouts due to continued packet loss.
c. Applications needing perfectly-reliable delivery can implement a
form of perfectly-persistent ARQ themselves, or use a reliable
transport protocol within the endhosts. Implementing perfect
persistence at each link along the path between the endhosts is
redundant, but cannot ensure the same reliability as end-to-end
transport [SALT81].
d. Link ARQ should not adversely delay the flow of end-to-end
control information. As an example, the ARQ retransmission of
data for one or more flows should not excessively extend the
protocol control loops. Excessive delay of duplicate TCP
acknowledgements (dupacks [STE94, BAL97]), SACK, or Explicit
Congestion Notification (ECN) indicators will reduce the
responsiveness of TCP flows to congestion events. Similar
issues exist for TCP-Friendly Rate Control (TFRC), where
equation-based congestion control is used with UDP [DRAFTHAN01].
Perfectly-persistent link protocols that perform unlimited ARQ, i.e.,
that continue to retransmit frames indefinitely until the frames are
successfully received, are seldom found in real implementations.
Most practical link protocols give up retransmission at some point,
but do not necessarily do so with the intention of bounding the ARQ
retransmission persistence. A protocol may, for instance, terminate
retransmission after a link connection failure, e.g., after no frames
have been successfully received within a pre-configured timer period.
The number of times a protocol retransmits a specific frame (or the
maximum number of retransmissions) therefore becomes a function of
many different parameters (ARQ procedure, link timer values, and
procedure for link monitoring), rather than being pre-configured.
Another common feature of this type of behaviour is that some
protocol implementers presume that, after a link failure, packets
queued to be sent over the link are no longer significant and can be
discarded when giving up ARQ retransmission.
Examples of ARQ protocols that are perfectly persistent include
ISO/ITU-T LAP-B [ISO7776] and ISO HDLC in the Asynchronously Balanced
Mode (ABM) [ISO4335a], e.g., using Multiple Selective Reject (MSREJ
[ISO4335b]). These protocols will retransmit a frame an unlimited
number of times until receipt of the frame is acknowledged.
2.2 High-Persistence (Highly-Reliable) ARQ Protocols
High-persistence ARQ protocols limit the number of times (or number
of attempts) that ARQ may retransmit a particular frame before the
sender gives up on retransmission of the missing frame and moves on
to forwarding subsequent buffered in-sequence frames. Ceasing
retransmission of a frame does not imply a lack of link connectivity
and does not cause a link protocol state change.
It has been recommended that a single IP packet should never be
delayed by the network for more than the Maximum Segment Lifetime
(MSL) of 120 seconds defined for TCP [RFC1122]. It is, however,
difficult in practice to bound the maximum path delay of an Internet
path. One case where segment (packet) lifetime may be significant is
where alternate paths of different delays exist between endhosts and
route flapping or flow-unaware traffic engineering is used. Some TCP
packets may follow a short path, while others follow a much longer
path, e.g., using persistent ARQ over a link outage.
Failure to limit the maximum packet lifetime can result in TCP
sequence numbers wrapping at high transmission rates, where old data
segments may be confused with newer segments if the sequence number
space has been exhausted and reused in the interim. Some TCP
implementations use the Round Trip Timestamp Measurement (RTTM)
option in TCP packets to remove this ambiguity, using the Protection
Against Wrapped Sequence number (PAWS) algorithm [RFC1323].
In practice, the MSL is usually very large compared to the typical
TCP RTO. The calculation of TCP RTO is based on estimated round-trip
path delay [RFC2988]. If the number of link retransmissions causes a
path delay larger than the value of RTO, the TCP retransmission timer
can expire, leading to a timeout and retransmission of a segment
(packet) by the TCP sender.
Although high persistency may benefit bulk flows, the additional
delay (and variations in delay) that it introduces may be highly
undesirable for other types of flows. Being able to treat flows
separately, with different classes of link service, is useful, and is
discussed in section 3.
Examples of high-persistence ARQ protocols include [BHA97, ECK98,
LUD99a, MEY99].
2.3 Low-Persistence (Partially-Reliable) ARQ Protocols
The characteristics of a link using a low-persistence ARQ protocol
may be summarised as:
a. The link is not perfectly reliable and does not provide an
absolute guarantee of delivery, i.e., the transmitter will
discard some frames as it 'gives up' before receiving an
acknowledgement of successful transmission across the link.
b. There is a lowered limit on the maximum added delay that IP
packets will experience when travelling across the link
(typically lower than the TCP path RTO). This reduces
interaction with TCP timers or with UDP-based error-control
schemes.
c. The link offers a low bound for the time that retransmission for
any one frame can block completed transmission and assembly of
other correctly and completely-received IP packets whose
transmission was begun before the missing frame was sent.
Limiting delay avoids aggravating contention or interaction
between different packet flows (see also section 3.2).
Examples of low-persistence ARQ protocols include [SAM96, WARD95,
CHE00].
2.4 Choosing Your Persistency
The TCP Maximum RTO is an upper limit on the maximum time that TCP
will wait until it performs a retransmission. Most TCP
implementations will generally have a TCP RTO of at least several
times the path delay.
Setting a lower link persistency (e.g., of the order 2-5
retransmission attempts) reduces potential interaction with the TCP
RTO timer, and may therefore reduce the probability of duplicate
copies of the same packet being present in the link transmit buffer
under some patterns of loss.
A link using a physical layer with a low propagation delay may allow
tens of retransmission attempts to deliver a single frame, and still
satisfy a bound for (b) in section 2.3. In this case, a low delay is
defined as being where the total packet transmission time across the
link is much less than 100 ms (a common value for the granularity of
the internal TCP system timer).
A packet may traverse a number of successive links on its total end-
to-end path. This is therefore an argument for much lower
persistency on any individual link, as delay due to persistency is
accumulated along the path taken by each packet.
Some implementers have chosen a lower persistence, falling back on
the judgement of TCP or of a UDP application to retransmit any
packets that are not recovered by the link ARQ protocol.
2.5 Impact of Link Outages
Links experiencing persistent loss, where many consecutive frames are
corrupted over an extended time, may also need to be considered.
Examples of channel behaviour leading to link outages include fading,
roaming, and some forms of interference. During the loss event,
there is an increased probability that a retransmission request may
be corrupted, and/or an increased probability that a retransmitted
frame will also be lost. This type of loss event is often known as a
"transient outage".
If the transient outage extends for longer than the TCP RTO, the TCP
sender will also perform transport-layer retransmission. At the same
time, the TCP sender will reduce its congestion window (cwnd) to 1
segment (packet), recalculate its RTO, and wait for an ACK packet.
If no acknowledgement is received, TCP will retransmit again, up to a
retry limit. TCP only determines that the outage is over (i.e., that
path capacity is restored) by receipt of an ACK. If link ARQ
protocol persistency causes a link in the path to discard the ACK,
the TCP sender must wait for the next RTO retransmission and its ACK
to learn that the link is restored. This can be many seconds after
the end of the transient outage.
When a link layer is able to differentiate a set of link service
classes (see section 3.3), a link ARQ persistency longer than the
largest link loss event may benefit a TCP session. This would allow
TCP to rapidly restore transmission without the need to wait for a
retransmission time out, generally improving TCP performance in the
face of transient outages. Implementation of such schemes remains a
research issue.
When an outage occurs for a sender sharing a common channel with
other nodes, uncontrolled high persistence can continue to consume
transmission resources for the duration of the outage. This may be
undesirable, since it reduces the capacity available for other nodes
sharing the channel, which do not necessarily experience the same
outage. These nodes could otherwise use the channel for more
productive transfers. The persistence is often limited by another
controlling mechanism in such cases. To counter such contention
effects, ARQ protocols may delay retransmission requests, or defer
the retransmission of requested frames until the outage ends for the
sender.
An alternate suggested approach for a link layer that is able to
identify separate flows is to use low link persistency (section 2.3)
along with a higher-layer mechanism, for example one that attempts to
deliver one packet (or whole TCP segment) per TCP flow after a loss
event [DRAFTKARN02]. This is intended to ensure that TCP
transmission is restored rapidly. Algorithms to implement this
remain an area of research.
3. Treatment of Packets and Flows
3.1 Packet Ordering
A common debate is whether a link should be allowed to forward
packets in an order different from that in which they were originally
received at its transmit interface.
IP networks are not required to deliver all IP packets in order,
although in most cases networks do deliver IP packets in their
original transmission order. Routers supporting class-based queuing
do reorder received packets, by reordering packets in different
flows, but these usually retain per-flow ordering.
Policy-based queuing, allowing fairer access to the link, may also
reorder packets. There is still much debate on optimal algorithms,
and on optimal queue sizes for particular link speeds. This,
however, is not related to the use of link ARQ and applies to any
(potential) bottleneck router.
Although small amounts of reordering are common in IP networks
[BEN00], significant reordering within a flow is undesirable as it
can have a number of effects:
a. Reordering will increase packet jitter for real-time
applications. This may lead to application data loss if a small
play-out buffer is used by the receiving application.
b. Reordering will interleave arrival of TCP segments, leading to
generation of duplicate ACKs (dupacks), leading to assumptions
of loss. Reception of an ACK followed by a sequence of three
identical dupacks causes the TCP sender to trigger fast
retransmission and recovery, a form of congestion avoidance,
since TCP always presumes that packet loss is due to congestion
[RFC2581, STE94]. This reduces TCP throughput efficiency as far
as the application is concerned, although it should not impact
data integrity.
In addition, reordering may negatively impact processing by some
existing poorly-implemented TCP/IP stacks, by leading to unwanted
side-effects in poorly-implemented IP fragment reassembly code,
poorly-implemented IP demultiplexing (filter) code, or in
poorly-implemented UDP applications.
Ordering effects must also be considered when breaking the end-to-end
paradigm and evaluating transport-layer relays such as split-TCP
implementations or Protocol Enhancing Proxies [RFC3135].
As with total path delay, TCP and UDP flows are impacted by the
cumulative effect of reordering along the entire path. Link protocol
designers must not assume that their link is the only link
undertaking packet reordering, as some level of reordering may be
introduced by other links along the same path, or by router
processing within the network [BEN00]. Ideally, the link protocol
should not contribute to reordering within a flow, or at least ensure
that it does not significantly increase the level of reordering
within the flow. To achieve this, buffering is required at the link
receiver. The total amount of buffering required is a function of
the link's bandwidth*delay product and the level of ARQ persistency,
and is bounded by the link window size.
A number of experimental ARQ protocols have allowed out-of-order
delivery [BAL95, SAM96, WARD95].
3.2 Using Link ARQ to Support Multiple Flows
Most links can be expected to carry more than one IP flow at a time.
Some high-capacity links are expected to carry a very large number of
simultaneous flows, often from and to a large number of different
endhosts. With use of multiple applications at an endhost, multiple
flows can be considered the norm rather than the exception, even for
last-hop links.
When packets from several flows are simultaneously in transit within
a link ARQ protocol, ARQ may cause a number of additional effects:
a. ARQ introduces variable delay (jitter) to a TCP flow sharing a
link with another flow experiencing loss. This additional
delay, introduced by the need for a link to provide in-sequence
delivery of packets, may adversely impact other applications
sharing the link, and can increase the duration of the initial
slow-start period for TCP flows for these applications. This is
significant for short-lived TCP flows (e.g., those used by
HTTP/1.0 and earlier), which spend most of their lives in
slow-start.
b. ARQ introduces jitter to UDP flows that share a link with
another flow experiencing loss. An end-to-end protocol may not
require reliable delivery for its flows, particularly if it is
supporting a delay-sensitive application.
c. High-persistence ARQ may delay packets long enough to cause the
premature timeout of another TCP flow's RTO timer, although
modern TCP implementations should not experience this since
their computed RTO values should leave a sufficient margin over
path RTTs to cope with reasonable amounts of jitter.
Reordering of packets belonging to different flows may be desirable
[LUD99b, CHE00] to achieve fair sharing of the link between
established bulk-data transfer sessions and sessions that transmit
less data, but would benefit from lower link transit delay.
Preserving ordering within each individual flow, to avoid the effects
of reordering described earlier in section 3.1, is worthwhile.
3.3 Differentiation of Link Service Classes
High ARQ persistency is generally considered unsuitable for many
applications using UDP, where reliable delivery is not always
required and where it may introduce unacceptable jitter, but may
benefit bulk data transfers under certain link conditions. A scheme
that differentiates packet flows into two or more classes, to provide
a different service to each class, is therefore desirable.
Observation of flow behaviour can tell you which flows are controlled
and congestion-sensitive, or uncontrolled and not, so that you can
treat them differently and ensure fairness. However, this cannot
tell you whether a flow is intended as reliable or unreliable by its
application, or what the application requires for best operation.
Supporting different link services for different classes of flows
therefore requires that the link is able to distinguish the different
flows from each other. This generally needs an explicit indication
of the class associated with each flow.
Some potential schemes for indicating the class of a packet include:
a. Using the Type of Service (ToS) bits in the IP header [RFC791].
The IETF has replaced these globally-defined bits, which were
not widely used, with the differentiated services model
(diffserv [RFC2475, RFC3260]). In diffserv, each packet carries a
Differentiated Service Code Point (DSCP), which indicates which
one of a set of diffserv classes the flow belongs to. Each
router maps the DSCP value of a received IP packet to one of a
set of Per Hop Behaviours (PHBs) as the packet is processed
within the network. Diffserv uses include policy-based routing,
class-based queuing, and support for other QoS metrics,
including IP packet priority, delay, reliability, and cost.
b. Inspecting the network packet header and viewing the IP protocol
type [RFC791] to gain an idea of the transport protocol used and
thus guess its needs. This is not possible when carrying an
encrypted payload, e.g., using the IP security extensions (IPSec)
with Encapsulation Security Payload (ESP) [RFC2406] payload
encryption.
c. By inspecting the transport packet header information to view
the TCP or UDP headers and port numbers (e.g., [PAR00, BAL95]).
This is not possible when using payload encryption, e.g., IPSec
with ESP payload encryption [RFC2406], and incurs processing
overhead for each packet sent over the link.
There are, however, some drawbacks to these schemes:
i. The ToS/Differentiated Services Code Point (DSCP) values
[RFC2475] may not be set reliably, and may be overwritten by
intermediate routers along the packet's path. These values may
be set by an ISP, and do not necessarily indicate the level of
reliability required by the end application. The link must be
configured with knowledge of the local meaning of the values.
ii. Tunnelling of traffic (e.g., GRE, MPLS, L2TP, IP-in-IP
encapsulation) can aggregate flows of different transport
classes, complicating individual flow classification with
schemes (b) and (c) and incurring further header processing if
tunnel contents are inspected.
iii. Use of the TCP/UDP port number makes assumptions about
application behaviour and requirements. New applications or
protocols can invalidate these assumptions, as can the use of
e.g., Network Address Port Translation, where port numbers are
remapped [RFC3022].
iv. In IPv6, the entire IPv6 header must be parsed to locate the
transport layer protocol, adding complexity to header
inspection. Again, this assumes that IPSec payload encryption
is not used.
Despite the difficulties in providing a framework for accurate flow
identification, approach (a) may be beneficial, and is preferable to
adding optimisations that are triggered by inspecting the contents of
specific IP packets. Some such optimisations are discussed in detail
in [LUD99b].
Flow management is desirable; clear flow identification increases the
number of tools available for the link designer, and permits more
complex ARQ strategies that may otherwise make misassumptions about
traffic requirements and behaviour when flow identification is not
done.
Links that are unable to distinguish clearly and safely between
delay-sensitive flows, e.g., real-time multimedia, DNS queries or
telnet, and delay-insensitive flows, e.g., bulk ftp transfers or
reliable multicast file transfer, cannot separate link service
classes safely. All flows should therefore experience the same link
behaviour.
In general, if separation of flows according to class is not
practicable, a low persistency is best for link ARQ.
4. Conclusions
A number of techniques may be used by link protocol designers to
counter the effects of channel errors or loss. One of these
techniques is Automatic Repeat ReQuest, ARQ, which has been and
continues to be used on links that carry IP traffic. An ARQ protocol
retransmits link frames that have been corrupted during transmission
across a channel. Link ARQ may significantly improve the probability
of successful transmission of IP packets over links prone to
occasional frame loss.
A lower rate of packet loss generally benefits transport protocols
and endhost applications. Applications using TCP generally benefit
from Internet paths with little or no loss and low round trip path
delay. This reduces impact on applications, allows more rapid growth
of TCP's congestion window during slow start, and ensures prompt
reaction to end-to-end protocol exchanges (e.g., retransmission,
congestion indications). Applications using other transport
protocols, e.g., UDP or SCTP, also benefit from low loss and timely
delivery.
A side-effect of link ARQ is that link transit delay is increased
when frames are retransmitted. At low error rates, many of the
details of ARQ, such as degree of persistence or any resulting
out-of-order delivery, become unimportant. Most frame losses will be
resolved in one or two retransmission attempts, and this is generally
unlikely to cause significant impact to e.g., TCP. However, on
shared high-delay links, the impact of ARQ on other UDP or TCP flows
may lead to unwanted jitter.
Where error rates are highly variable, high link ARQ persistence may
provide good performance for a single TCP flow. However,
interactions between flows can arise when many flows share capacity
on the same link. A link ARQ procedure that provides flow management
will be beneficial. Lower ARQ persistence may also have merit, and
is preferable for applications using UDP. The reasoning here is that
the link can perform useful work forwarding some complete packets,
and that blocking all flows by retransmitting the frames of a single
packet with high persistence is undesirable.
During a link outage, interactions between ARQ and multiple flows are
less significant; the ARQ protocol is likely to be equally
unsuccessful in retransmitting frames for all flows. High
persistence may benefit TCP flows, by enabling prompt recovery once
the channel is restored.
Low ARQ persistence is particularly useful where it is difficult or
impossible to classify traffic flows, and hence treat each flow
independently, and where the link capacity can accommodate a large
number of simultaneous flows.
Link ARQ designers should consider the implications of their design
on the wider Internet. Effects such as increased transit delay,
jitter, and re-ordering are cumulative when performed on multiple
links along an Internet path. It is therefore very hard to say how
many ARQ links may exist in series along an arbitrary Internet path
between endhosts, especially as the path taken and its links may
change over time.
In summary, when links cannot classify traffic flows and treat them
separately, low persistence is generally desirable; preserving packet
ordering is generally desirable. Extremely high persistence and
perfect persistence are generally undesirable; highly-persistent ARQ
is a bad idea unless flow classification and detailed and accurate
knowledge of flow requirements make it possible to deploy high
persistency where it will be beneficial.
There is currently insufficient experience to recommend a specific
ARQ scheme for any class of link. It is also important to realize
that link ARQ is just one method of error recovery, and that other
complementary physical-layer techniques may be used instead of, or
together with, ARQ to improve overall link throughput for IP traffic.
The choice of potential schemes includes adapting the data rate,
adapting the signal bandwidth, adapting the transmission power,
adaptive modulation, adaptive information redundancy / forward error
control, and interleaving. All of these schemes can be used to
improve the received signal energy per bit, and hence reduce error,
frame loss and resulting packet loss rates given specific channel
conditions.
There is a need for more research to more clearly identify the
importance of and trade-offs between the above issues over various
types of link and over various types of channels. It would be useful
if researchers and implementers clearly indicated the loss model,
link capacity and characteristics, link and end-to-end path delays,
details of TCP, and the number (and details) of flows sharing a link
when describing their experiences. In each case, it is recommended
that specific details of the link characteristics and mechanisms also
be considered; solutions vary with conditions.
5. Security Considerations
No security implications have been identified as directly impacting
IP traffic. However, an unreliable link service may adversely impact
some existing link-layer key management distribution protocols if
link encryption is also used over the link.
Denial-of-service attacks exploiting the behaviour of the link
protocol, e.g., using knowledge of its retransmission behaviour and
propagation delay to cause a particular form of jamming, may be
specific to an individual link scenario.
6. IANA Considerations
No assignments from the IANA are required.
7. Acknowledgements
Much of what is described here has been developed from a summary of a
subset of the discussions on the archived IETF PILC mailing list. We
thank the contributors to PILC for vigorous debate.
In particular, the authors would like to thank Spencer Dawkins, Aaron
Falk, Dan Grossman, Merkourios Karaliopoulos, Gary Kenward, Reiner
Ludwig and Jean Tourrilhes for their detailed comments.
8. References
References of the form RFCnnnn are Internet Request for Comments
(RFC) documents available online at http://www.rfc-editor.org/.
8.1 Normative References
[RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC791] Postel, J., "Internet Protocol", STD 5, RFC 791,
September 1981.
[RFC793] Postel, J., "Transmission Control Protocol", RFC 793,
September 1981.
[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts --
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC2406] Kent, S. and R. Atkinson, "IP Encapsulating Security
Payload (ESP)", RFC 2406, November 1998.
[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.
and W. Weiss, "An Architecture for Differentiated
Services", RFC 2475, December 1998.
[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[RFC2988] Paxson, V. and M. Allman, "Computing TCP's
Retransmission Timer", RFC 2988, November 2000.
[RFC3135] Border, J., Kojo, M., Griner, J., Montenegro, G. and Z.
Shelby, "Performance Enhancing Proxies Intended to
Mitigate Link-Related Degradations", RFC 3135, June
2001.
[RFC3260] Grossman, D., "New Terminology and Clarifications for
Diffserv", RFC 3260, April 2002.
8.2 Informative References
[BAL95] Balakrishnan, H., Seshan, S. and R. H. Katz,
"Improving Reliable Transport and Handoff Performance
in Cellular Wireless Networks", ACM MOBICOM, Berkeley,
1995.
[BAL97] Balakrishnan, H., Padmanabhan, V. N., Seshan, S. and
R. H. Katz, "A Comparison of Mechanisms for Improving
TCP Performance over Wireless Links", IEEE/ACM
Transactions on Networking, 5(6), pp. 756-759, 1997.
[BEN00] Bennett, J. C., Partridge, C. and N. Schectman, "Packet
Reordering is Not Pathological Network Behaviour",
IEEE/ACM Transactions on Networking, 7(6), pp. 789-798,
2000.
[BHA97] Bhagwat, P., Bhattacharya, P., Krishna A. and S. K.
Tripathi, "Using channel state dependent packet
scheduling to improve TCP throughput over wireless
LANs", ACM/Baltzer Wireless Networks Journal, (3)1,
1997.
[CHE00] Cheng, H. S., G. Fairhurst et al., "An Efficient
Partial Retransmission ARQ Strategy with Error Codes
by Feedback Channel", IEE Proceedings - Communications,
(147)5, pp. 263-268, 2000.
[DRAFTKARN02] Karn, P., Ed., "Advice for Internet Subnetwork
Designers", Work in Progress.
[DRAFTHAN01] Handley, M., Floyd, S. and J. Widmer, "TCP Friendly
Rate Control (TFRC): Protocol Specification", Work in
Progress.
[ECK98] Eckhardt, D. A. and P. Steenkiste, "Improving Wireless
LAN Performance via Adaptive Local Error Control",
IEEE ICNP, 1998.
[FER99] Ferrero, A., "The Eternal Ethernet", Addison-Wesley,
1999.
[ISO4335a] HDLC Procedures: Specification for Consolidation of
Elements of Procedures, ISO 4335 and AD/1,
International Standardization Organization, 1985.
[ISO4335b] HDLC Procedures: Elements of Procedures, Amendment 4:
Multi-Selective Reject Option, ISO 4335/4,
International Standards Organization, 1991.
[ISO7776] Specification for X.25 LAPB-Compatible DTE Data Link
Procedures, ISO 4335/4, International Standards
Organization, 1985.
[KEN87] Kent, C. A. and J. C. Mogul, "Fragmentation
Considered Harmful", Proceedings of ACM SIGCOMM 1987,
ACM Computer Communications Review, 17(5), pp. 390-401,
1987.
[LIN93] Lin, S. and D. Costello, "Error Control Coding:
Fundamentals and Applications", Prentice Hall, 1993.
[LUD99a] Ludwig, R., Rathonyi, B., Konrad, A., Oden, K., and A.
Joseph, "Multi-Layer Tracing of TCP over a Reliable
Wireless Link", ACM SIGMETRICS, pp. 144-154, 1999.
[LUD99b] Ludwig, R., Konrad, A., Joseph, A. and R. H. Katz,
"Optimizing the End-to-End Performance of Reliable
Flows over Wireless Links", ACM MobiCOM, 1999.
[MEY99] Meyer, M., "TCP Performance over GPRS", IEEE Wireless
Communications and Networking Conference, 1999.
[PAR00] Parsa, C. and J. J. Garcia-Luna-Aceves, "Improving TCP
Performance over Wireless Networks at the Link Layer",
ACM Mobile Networks and Applications Journal, (5)1,
pp. 57-71, 2000.
[RFC1191] Mogul, J. and S. Deering, "Path MTU Discovery", RFC
1191, November 1990.
[RFC1323] Jacobson, V., Braden, R. and D. Borman, "TCP Extensions
for High Performance", RFC 1323, May 1992.
[RFC1350] Sollins, K., "The TFTP Protocol (Revision 2)", STD 33,
RFC 1350, July 1992.
[RFC1435] Knowles, S., "IESG Advice from Experience with Path MTU
Discovery", RFC 1435, March 1993.
[RFC1981] McCann, J., Deering, S. and J. Mogul, "Path MTU
Discovery for IP version 6", RFC 1981, August 1996.
[RFC2488] Allman, M., Glover, D. and L. Sanchez, "Enhancing TCP
Over Satellite Channels using Standard Mechanisms",
BCP 28, RFC 2488, January 1999.
[RFC2757] Montenegro, G., Dawkins, S., Kojo, M., Magret V. and
N. Vaidya, "Long Thin Networks", RFC 2757, January
2000.
[RFC2760] Allman, M., Dawkins, S., Glover, D., Griner, J.,
Tran, D., Henderson, T., Heidemann, J., Touch, J.,
Kruse, H., Ostermann, S., Scott K. and J. Semke
"Ongoing TCP Research Related to Satellites",
RFC 2760, February 2000.
[RFC2960] Stewart, R., Xie, Q., Morneault, K., Sharp, C.,
Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M.,
Zhang, L. and V. Paxson, "Stream Control Transmission
Protocol", RFC 2960, October 2000.
[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022,
January 2001.
[RFC3155] Dawkins, S., Montenegro, G., Kojo, M., Magret, V. and
N. Vaidya, "End-to-end Performance Implications of
Links with Errors", BCP 50, RFC 3155, August 2001.
[SALT81] Saltzer, J. H., Reed, D. P. and D. Clark, "End-to-End
Arguments in System Design", Second International
Conference on Distributed Computing Systems, pp.
509-512, 1981. Published with minor changes in ACM
Transactions in Computer Systems (2)4, pp. 277-288,
1984.
[SAM96] Samaraweera, N. and G. Fairhurst, "Robust Data Link
Protocols for Connection-less Service over Satellite
Links", International Journal of Satellite
Communications, 14(5), pp. 427-437, 1996.
[SAM98] Samaraweera, N. and G. Fairhurst, "Reinforcement of
TCP/IP Error Recovery for Wireless Communications",
ACM Computer Communications Review, 28(2), pp. 30-38,
1998.
[STE94] Stevens, W. R., "TCP/IP Illustrated, Volume 1",
Addison-Wesley, 1994.
[STONE00] Stone, J. and C. Partridge, "When the CRC and TCP
Checksum Disagree", Proceedings of SIGCOMM 2000, ACM
Computer Communications Review 30(4), pp. 309-321,
September 2000.
[WARD95] Ward, C., et al., "A Data Link Control Protocol for LEO
Satellite Networks Providing a Reliable Datagram
Service", IEEE/ACM Transactions on Networking, 3(1),
1995.
Authors' Addresses
Godred Fairhurst
Department of Engineering
University of Aberdeen
Aberdeen AB24 3UE
United Kingdom
EMail: gorry@erg.abdn.ac.uk
http://www.erg.abdn.ac.uk/users/gorry/
Lloyd Wood
Cisco Systems Ltd
4 The Square
Stockley Park
Uxbridge UB11 1BY
United Kingdom
EMail: lwood@cisco.com
http://www.ee.surrey.ac.uk/Personal/L.Wood/
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