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Versions: 00 01 02 03 RFC 3366

   Internet Engineering Task Force              Gorry Fairhurst
   INTERNET DRAFT                               University of Aberdeen
                                                Lloyd Wood
                                                Cisco Systems Ltd

                                                August 2001
   draft-ietf-pilc-link-arq-issues-03.txt       Expires: February 2002

     Advice to link designers on link Automatic Repeat reQuest (ARQ)


   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as

   Internet-Drafts are draft documents valid for a maximum of six
   months and may be updated, replaced, or obsoleted by other documents
   at any time.  It is inappropriate to use Internet-Drafts as
   reference material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at


   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 on 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 bearer channel.  This
   document also describes issues relevant to supporting IP traffic
   over physical channels where performance varies, and link ARQ is
   likely to be used.

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   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 . . . . . . . . . . . . . . 6
          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. . . . . . . . . . . . . . . . . . . . . . . 9
      2.1 Perfectly-persistent (reliable) ARQ protocols           10
      2.2 High-persistence (highly-reliable) ARQ protocols. . . . 11
      2.3 Low-persistence (partially-reliable) ARQ protocols      12
      2.4 Choosing your persistency. . . . . . . . . . . . . . . .13
      2.5 Impact of channel outages                               13
   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 implications                                       21
   6. IANA considerations. . . . . . . . . . . . . . . . . . . . .21
   7. Acknowledgements                                            21
   8. References. . . . . . . . . . . . . . . . . . . . . . . . . 21
   Authors' addresses                                             24
   Full copyright statement. . . . . . . . . . . . . . . . . . . .25


   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

   A companion document [DRAFTKARN01] 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 [DRAFTDAW01, RFC3135].

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   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 it in
   their 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

   Perfect reliability is not a requirement for IP networks or for
   links [DRAFTKARN01].  IP networks may discard packets due to a
   variety of reasons entirely unrelated to link 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 the transport layer [SALT81].

   Some familiarity with TCP, the Transmission Control Protocol
   [STE94], is presumed here.  TCP provides a reliable byte-stream
   transport service, building upon the best-effort 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]. (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].

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   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)


   At the link layer, ARQ operates on blocks of data, known as frames,
   and attempts to deliver frames from the sender to the receiver over
   a physical link.  A link frame may contain one or more complete IP

   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'.
   An ARQ link mechanism relies on an integrity check for each frame
   (e.g. strong link-layer CRC [DRAFTKARN01]) and a retransmission
   process to retransmit lost or errored frames.

   Links may be full-duplex (allowing two-way communication over
   separate forward and reverse physical channels), half-duplex (where
   two-way communication uses a shared forward and reverse physical
   channel, e.g. IrDA, IEEE 802.11) or simplex (where one channel
   allows communication in only one direction).  ARQ requires a forward
   and return path, and therefore link ARQ may be used over links that
   employ full- or half-duplex links.  When the channel is shared
   between two or more link nodes, a link MAC protocol is required to
   ensure all nodes requiring transmission get access to the shared
   physical 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 error rate.  To allow for efficient use of the
   channel, the maximum link frame size may therefore be considerably
   lower than the maximum IP datagram size expressed in 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 [DRAFTKARN01].  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 [KEN88, DRAFTKARN01].  Its use can
   be minimised with use of Path MTU discovery [RFC1191, RFC1435].

   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].

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   FEC is commonly included in the design of wireless links and may be
   used simultaneously with link ARQ.  FEC schemes also exist which
   combine modulation and coding 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.

   Links supporting stateful reservation-based Quality of Service (QoS)
   according to the Integrated Services (intserv) model are also not
   explicitly discussed.


   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, causing loss due to unavailability of the
       physical link for a period of time.  This is a common
       characteristic of some types of link, e.g. mobile cellular
       radio.  During the outage, the link loses all or virtually all
       frames, until the link is restored.

   Other forms of packet loss are not related to channel conditions,
   but include:

   i.  Loss of a frame transmitted in a shared physical 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 link
       congestion collapse). This is discussed further in section 1.5.

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   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 levels of loss and patterns of loss experienced are functions of
   the design of the link's 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
   physical channel.


   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 [DRAFTDAW01, SALT81].

   b.  Link ARQ can operate on individual frames, using implicit
       transparent link fragmentation [DRAFTKARN01].  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, in order to optimise delivery performance for
   the current link conditions.  This information can include
   information about the state of the link and physical layer channel,
   e.g. knowledge of the current available transmission rate, the
   prevailing error environment, or available transmit power in
   wireless links.


   A link ARQ protocol uses a link protocol mechanism to allow the
   sender to detect lost or errored frames and to schedule
   retransmission.  Detection of frame loss may be via a link protocol
   timer, by detecting missing positive link acknowledgment frames,

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   receiving explicit negative acknowledgment frames and/or by polling
   the link receiver status.

   Whatever the mechanisms that are chosen, there are two easily-
   described categories of ARQ retransmission process that are widely


   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 original frame was lost or

   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.


   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.

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   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
   channel 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.


   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 link.

   b.  Retransmission back-off, an additional delay introduced for
       retransmissions by some MAC schemes when operating over a shared
       physical channel to prevent link congestion collapse.  This
       collapse may otherwise arise, if, for example, a set of link
       receivers all retransmitted immediately after a collision on a
       busy channel.  Many protocols 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].  Retransmission
       backoff is not required by protocols used on point-to-point
       links, where retransmissions can be sent at the earliest
       possible time.

   c.  Waiting for access to the allocated channel when the physical
       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.

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   e.  Physical link propagation time, limited by the speed of
       transmission of the signal in the physical medium of the

   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
       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, for each time retransmission of a frame
   occurs, increasing overall delay for the packet that the frame
   partially carries.  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 each 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


   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 an IP 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.

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   An example of a reliable link protocol that is perfectly persistent
   is the ISO HDLC protocol in the Asynchronous Balanced Mode (ABM)

   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 subnetwork, even when many senders are
   attempting retransmissions.

   TCP, STCP [RFC2960] and a number of applications using UDP (e.g.
   tftp [RFC1350]) 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.


   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 errored
   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 perfect persistence for IP traffic include:

   a.  Variable link delay; the impact of ARQ introduces a degree of
       jitter, a function of the link's physical delay and frame
       serialisation and transmission times (see 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 timeout 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.

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   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
       acknowledgments (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 reality.

   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 time 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 assume that after a link failure, queued data
   is no longer significant and discard packets when giving up ARQ

   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


   High-persistence ARQ protocols limit the time (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

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   to forwarding subsequent buffered in-sequence frames.  Giving up
   frame retransmission 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,
   practically difficult 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.  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 3.3.

   Examples of high-persistence ARQ protocols include [BHA97, ECK98,
   LUD99a, MEY99].


   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.

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   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

   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-received IP packets originally sent before the
       missing frame.  Limiting delay avoids aggravating contention or
       interaction between different packet flows (see also section

   Examples of low-persistence ARQ protocols include [SAM96, WARD95,


   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 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.

   Links 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 link is defined as
   one where the total packet transmission time is much less than 100
   ms (this is 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
   accumulates along the path for each packet.

   Some implementers have chosen a lower persistence, falling back on
   the judgement of TCP or a UDP application to retransmit any packets
   that are not recovered by the link ARQ protocol.

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   Channels experiencing persistent loss, where many consecutive frames
   are corrupted over an extended time, may also need to be considered.
   Examples of this type of channel behaviour 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
   an '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 cwnd to 1 segment
   (packet), recalculate its RTO, and wait for an ACK packet.  If none
   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 the link ARQ protocol
   persistency causes the link to discard the ACK, the TCP sender must
   wait for the next RTO retransmission to discover the link is
   restored.  This can be many seconds after the end of the channel

   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 physical 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 this case.  To counter such
   effects, ARQ protocols may delay retransmission requests, or defer
   retransmission of requested frames until the outage ends for the

   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-level mechanism, for example one that attempts
   to deliver one TCP packet (whole TCP segment) per TCP flow after a
   loss event [DRAFTKARN01].  This is intended to ensure that TCP
   transmission is restored rapidly.  Algorithms to implement this
   remain an area of future research.

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   A common debate is whether a link should be allowed to forward
   packets in an order different to that in which they were originally
   received at its transmit interface.

   IP networks are not required to deliver all IP packets in order,
   although generally networks do deliver most 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 use of link ARQ and applies to any
   (potential) bottleneck router.

   Although small amounts of reordering are common in IP networks
   [BEN00], significant re-ordering 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.  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 loss due to
       congestion [RFC2581, STE94].  This reduces TCP throughput
       efficiency as far as the application is concerned, but 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 poorly-
   implemented UDP applications.

   Ordering effects must also be considered when breaking the end-to-
   end paradigm and evaluating transport-level 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

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   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].


   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-

   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, particularly if it is supporting a
       delay-sensitive application.

   c.  High-persistence ARQ may delay packets long enough to cause
       premature timeout of another TCP flow's RTO timer, although
       modern TCP implementations should not experience this since
       their computed RTO values should leave sufficient margin over
       path RTTs to cope with reasonable amounts of jitter.

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   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


   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 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]).  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) [RFC1827] payload

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   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 [RFC1827] payload encryption, 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, locating the transport layer protocol type requires
       parsing the entire IPv6 header, 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

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   In general, if separation of flows according to class is not
   practicable, a low persistency is best for link ARQ.


   A number of techniques may be used by link protocol designers to
   counter the effects of errors / loss on links. One of these 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 on the physical link.  Link ARQ
   may significantly improve the probability of successful transmission
   of IP packets over links prone to occasional 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 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.

   For links where error rates are highly variable, high 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.

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   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 realise
   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

   The choice of potential schemes includes adapting the data rate,
   adapting the signal bandwidth, adapting the transmission power,
   adaptive modulation, and 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
   physical 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.  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 are also considered; solutions vary
   with conditions.

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   No security implications have been identified as directly impacting
   IP traffic.  However, an unreliable link service may adversely
   impact some link-layer key management distribution protocols if
   encryption is used over such a 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.


   No assignments from the IANA are required.


   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 Kenwood,
   Reiner Ludwig and Jean Tourrilhes for their detailed comments.


   References of the form RFCnnnn are Internet Request for Comments
   (RFC) documents available online at http://www.rfc-editor.org/.

   [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.

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   [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.

   [DRAFTDAW01] Dawkins, S., Montenegro, G., Kojo, M., Magret, V. and
   N. Vaidya, End-to-end Performance Implications of Links with Errors,
   draft-ietf-pilc-error-08.txt, to be published as a BCP RFC, 2001.

   [DRAFTKARN01] Karn, P. (editor) et al., Advice for Internet
   Subnetwork Designers, draft-ietf-pilc-link-design-nn.txt, work in
   progress as internet-draft, 2001.

   [DRAFTHAN01] Handley, M., Floyd, S. and J. Widmer, TCP Friendly Rate
   Control (TFRC): Protocol Specification, draft-ietf-tsvwg-tfrc-nn
   .txt, work in progress as internet-draft, 2001.

   [ECK98] Eckhardt, D. A., and P. Steenkiste, Improving Wireless LAN
   Performance via Adaptive Local Error Control, IEEE ICNP, 1998.

   [FER99] A. Ferrero, 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.

   [KEN88] Kent, C. A. and J. C. Mogul, Fragmentation Considered
   Harmful, Proceedings of ACM SIGCOMM, pp. 390-401, 1988.

   [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. Katz, Optimizing
   the End-to-End Performance of Reliable Flows over Wireless Links,
   ACM MobiCOM, 1999.

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   [MEY99] M. Meyer, TCP Performance over GPRS, IEEE WCNC, 1999.

   [PAR00] Parsa, C. and J. J. Garcia-Luna-Aceves, Improving TCP
   Performance over Wireless Networks at the Link Layer, Mobile
   Networks and Applications, (1)5, pp. 57-71, 2000.

   [RFC768] J. Postel, User Datagram Protocol, 1980.

   [RFC791] J. Postel, Internet Protocol, RFC 791, 1981.

   [RFC1122] R. Braden et al., Requirements for Internet Hosts --
   Communication Layers, RFC 1122, 1989.

   [RFC1191] Mogul, J. and S. Deering, Path MTU Discovery, RFC 1191,

   [RFC1323] Jacobson, V. and R. Braden, TCP Extensions for High
   Performance, RFC 1323, 1992.

   [RFC1350] K. Solins, The TFTP Protocol (Revision 2), RFC 1350, 1992.

   [RFC1435] S. Knowles, IESG Advice from Experience with Path MTU
   Discovery, RFC 1435, 1993.

   [RFC1827] R. Atkinson, IP Encapsulating Security Payload (ESP),
   RFC 1827, 1995.

   [RFC2475] Blake, S., Black, D., M. Carlson et al., An Architecture
   for Differentiated Services, RFC 2475, 1998.

   [RFC2488] Allman, M., Glover, D. and L. Sanchez, Enhancing TCP Over
   Satellite Channels using Standard Mechanisms, RFC 2488, 1999.

   [RFC2581] Allman, M., Paxson, V. and W. Stevens, TCP Congestion
   Control, RFC 2581, 1999.

   [RFC2757] Dawkins, S., Kojo, M., Magret V. and N. Vaidya, Long Thin
   Networks, RFC 2757, 2000.

   [RFC2760] Allman, M., Dawkins, S., Glover, D., J. Griner et al.,
   Ongoing TCP Research Related to Satellites, RFC 2760, 1999.

   [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, 2000.

   [RFC3022] Srisuresh, P., and K. Egevang, Traditional IP Network
   Address Translator (Traditional NAT), RFC 3022, 2001.

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   [RFC3135] Border, J., Kojo, M., Griner, J., Montenegro, G. and Z.
   Shelby, Performance Enhancing Proxies Intended to Mitigate Link-
   Related Degradations, RFC 3135, 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 CCR, 28(2), pp. 30-
   38, 1998.

   [STE94] W. R. Stevens, TCP/IP Illustrated, Volume 1, Addison-Wesley,

   [STONE00] Stone, J. and C. Partridge, When the CRC and TCP Checksum
   Disagree, Proceedings of SIGCOMM 2000, ACM Computer Communications
   Review pp. 309-321, September 2000.

   [WARD95] C. Ward et al., A Data Link Control Protocol for LEO
   Satellite Networks Providing a Reliable Datagram Service, IEEE/ACM
   Transactions on Networking, 3(1), 1995.


   Gorry Fairhurst (gorry@erg.abdn.ac.uk)

   Department of Engineering, University of Aberdeen,
   Aberdeen AB24 3UE, United Kingdom.

   Lloyd Wood (lwood@cisco.com)

   Cisco Systems UK Ltd, 4 The Square, Stockley Park,
   Uxbridge UB11 1BY, United Kingdom.

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   Copyright (C) The Internet Society (2000-2001).
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   This document and translations of it may be copied and furnished to
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   This document and the information contained herein is provided on an

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