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Network Working Group P. Karn, Ed. Request for Comments: 3819 Qualcomm BCP: 89 C. Bormann Category: Best Current Practice Universitaet Bremen TZI

                                                          G. Fairhurst
                                                University of Aberdeen
                                                           D. Grossman
                                                        Motorola, Inc.
                                                             R. Ludwig
                                                     Ericsson Research
                                                            J. Mahdavi
                                                         G. Montenegro
                                 Sun Microsystems Laboratories, Europe
                                                              J. Touch
                                                               L. Wood
                                                         Cisco Systems
                                                             July 2004
              Advice for Internet Subnetwork Designers

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 (2004).


 This document provides advice to the designers of digital
 communication equipment, link-layer protocols, and packet-switched
 local networks (collectively referred to as subnetworks), who wish to
 support the Internet protocols but may be unfamiliar with the
 Internet architecture and the implications of their design choices on
 the performance and efficiency of the Internet.

Karn, et al. Best Current Practice [Page 1] RFC 3819 Advice for Internet Subnetwork Designers July 2004

Table of Contents

 1.  Introduction and Overview. . . . . . . . . . . . . . . . . . .  2
 2.  Maximum Transmission Units (MTUs) and IP Fragmentation . . . .  4
     2.1.  Choosing the MTU in Slow Networks. . . . . . . . . . . .  6
 3.  Framing on Connection-Oriented Subnetworks . . . . . . . . . .  7
 4.  Connection-Oriented Subnetworks. . . . . . . . . . . . . . . .  9
 5.  Broadcasting and Discovery . . . . . . . . . . . . . . . . . . 10
 6.  Multicasting . . . . . . . . . . . . . . . . . . . . . . . . . 11
 7.  Bandwidth on Demand (BoD) Subnets. . . . . . . . . . . . . . . 13
 8.  Reliability and Error Control. . . . . . . . . . . . . . . . . 14
     8.1.  TCP vs Link-Layer Retransmission . . . . . . . . . . . . 14
     8.2.  Recovery from Subnetwork Outages . . . . . . . . . . . . 17
     8.3.  CRCs, Checksums and Error Detection. . . . . . . . . . . 18
     8.4.  How TCP Works. . . . . . . . . . . . . . . . . . . . . . 20
     8.5.  TCP Performance Characteristics. . . . . . . . . . . . . 22
           8.5.1.  The Formulae . . . . . . . . . . . . . . . . . . 22
           8.5.2.  Assumptions. . . . . . . . . . . . . . . . . . . 23
           8.5.3.  Analysis of Link-Layer Effects on TCP
                   Performance. . . . . . . . . . . . . . . . . . . 24
 9.  Quality-of-Service (QoS) Considerations. . . . . . . . . . . . 26
 10. Fairness vs Performance. . . . . . . . . . . . . . . . . . . . 29
 11. Delay Characteristics. . . . . . . . . . . . . . . . . . . . . 30
 12. Bandwidth Asymmetries. . . . . . . . . . . . . . . . . . . . . 31
 13. Buffering, Flow and Congestion Control . . . . . . . . . . . . 31
 14. Compression. . . . . . . . . . . . . . . . . . . . . . . . . . 34
 15. Packet Reordering. . . . . . . . . . . . . . . . . . . . . . . 36
 16. Mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
 17. Routing. . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
 18. Security Considerations. . . . . . . . . . . . . . . . . . . . 41
 19. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 44
 20. Informative References . . . . . . . . . . . . . . . . . . . . 45
 21. Contributors' Addresses. . . . . . . . . . . . . . . . . . . . 57
 22. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 58
 23. Full Copyright Statement . . . . . . . . . . . . . . . . . . . 60

1. Introduction and Overview

 IP, the Internet Protocol [RFC791] [RFC2460], is the core protocol of
 the Internet.  IP defines a simple "connectionless" packet-switched
 network.  The success of the Internet is largely attributed to IP's
 simplicity, the "end-to-end principle" [SRC81] on which the Internet
 is based, and the resulting ease of carrying IP on a wide variety of
 subnetworks, not necessarily designed with IP in mind.  A subnetwork
 refers to any network operating immediately below the IP layer to
 connect two or more systems using IP (i.e., end hosts or routers).
 In its simplest form, this may be a direct connection between the IP
 systems (e.g., using a length of cable or a wireless medium).

Karn, et al. Best Current Practice [Page 2] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 This document defines a subnetwork as a layer 2 network, which is a
 network that does not rely upon the services of IP routers to forward
 packets between parts of the subnetwork.  However, IP routers may
 bridge frames at Layer 2 between parts of a subnetwork.  Sometimes,
 it is convenient to aggregate a group of such subnetworks into a
 single logical subnetwork.  IP routing protocols (e.g., OSPF, IS-IS,
 and PIM) can be configured to support this aggregation, but typically
 present a layer-3 subnetwork rather than a layer-2 subnetwork.  This
 may also result in a specific packet passing several times over the
 same layer-2 subnetwork via an intermediate layer-3 gateway (router).
 Because that aggregation requires layer-3 components, issues thereof
 are beyond the scope of this document.
 However, while many subnetworks carry IP, they do not necessarily do
 so with maximum efficiency, minimum complexity, or cost, nor do they
 implement certain features to efficiently support newer Internet
 features of increasing importance, such as multicasting or quality of
 With the explosive growth of the Internet, IP packets comprise an
 increasingly large fraction of the traffic carried by the world's
 telecommunications networks.  It therefore makes sense to optimize
 both existing and new subnetwork technologies for IP as much as
 Optimizing a subnetwork for IP involves three complementary
 1.  Providing functionality sufficient to carry IP.
 2.  Eliminating unnecessary functions that increase cost or
 3.  Choosing subnetwork parameters that maximize the performance of
     the Internet protocols.
 Because IP is so simple, consideration 2 is more of an issue than
 consideration 1.  That is to say, subnetwork designers make many more
 errors of commission than errors of omission.  However, certain
 enhancements to Internet features, such as multicasting and quality-
 of-service, benefit significantly from support given by the
 underlying subnetworks beyond that necessary to carry "traditional"
 unicast, best-effort IP.

Karn, et al. Best Current Practice [Page 3] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 A major consideration in the efficient design of any layered
 communication network is the appropriate layer(s) in which to
 implement a given function.  This issue was first addressed in the
 seminal paper, "End-to-End Arguments in System Design" [SRC81].  That
 paper argued that many functions can be implemented properly *only*
 on an end-to-end basis, i.e., at the highest protocol layers, outside
 the subnetwork.  These functions include ensuring the reliable
 delivery of data and the use of cryptography to provide
 confidentiality and message integrity.
 Such functions cannot be provided solely by the concatenation of
 hop-by-hop services; duplicating these functions at the lower
 protocol layers (i.e., within the subnetwork) can be needlessly
 redundant or even harmful to cost and performance.
 However, partial duplication of functionality in a lower layer can
 *sometimes* be justified by performance, security, or availability
 considerations.  Examples include link-layer retransmission to
 improve the performance of an unusually lossy channel, e.g., mobile
 radio, link-level encryption intended to thwart traffic analysis, and
 redundant transmission links to improve availability, increase
 throughput, or to guarantee performance for certain classes of
 traffic.  Duplication of protocol functions should be done only with
 an understanding of system-level implications, including possible
 interactions with higher-layer mechanisms.
 The original architecture of the Internet was influenced by the
 end-to-end principle [SRC81], and has been, in our view, part of the
 reason for the Internet's success.
 The remainder of this document discusses the various subnetwork
 design issues that the authors consider relevant to efficient IP

2. Maximum Transmission Units (MTUs) and IP Fragmentation

 IPv4 packets (datagrams) vary in size, from 20 bytes (the size of the
 IPv4 header alone) to a maximum of 65535 bytes.  Subnetworks need not
 support maximum-sized (64KB) IP packets, as IP provides a scheme that
 breaks packets that are too large for a given subnetwork into
 fragments that travel as independent IP packets and are reassembled
 at the destination.  The maximum packet size supported by a
 subnetwork is known as its Maximum Transmission Unit (MTU).
 Subnetworks may, but are not required to, indicate the length of each
 packet they carry.  One example is Ethernet with the widely used DIX
 [DIX82] (not IEEE 802.3 [IEEE8023]) header, which lacks a length

Karn, et al. Best Current Practice [Page 4] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 field to indicate the true data length when the packet is padded to a
 minimum of 60 bytes.  This is not a problem for uncompressed IP
 because each IP packet carries its own length field.
 If optional header compression [RFC1144] [RFC2507] [RFC2508]
 [RFC3095] is used, however, it is required that the link framing
 indicate frame length because that is needed for the reconstruction
 of the original header.
 In IP version 4 (the version now in widespread use), fragmentation
 can occur at either the sending host or in an intermediate router,
 and fragments can be further fragmented at subsequent routers if
 In IP version 6 [RFC2460], fragmentation can occur only at the
 sending host; it cannot occur in a router (called "router
 fragmentation" in this document).
 Both IPv4 and IPv6 provide a "path MTU discovery" procedure [RFC1191]
 [RFC1435] [RFC1981] that allows the sending host to avoid
 fragmentation by discovering the minimum MTU along a given path and
 reduce its packet sizes accordingly.  This procedure is optional in
 IPv4 and IPv6.
 Path MTU discovery is widely deployed, but it sometimes encounters
 problems.  Some routers fail to generate the ICMP messages that
 convey path MTU information to the sender, and sometimes the ICMP
 messages are blocked by overly restrictive firewalls.  The result can
 be a "Path MTU Black Hole" [RFC2923] [RFC1435].
 The Path MTU Discovery procedure, the persistence of path MTU black
 holes, and the deletion of router fragmentation in IPv6 reflect a
 consensus of the Internet technical community that router
 fragmentation is best avoided.  This requires that subnetworks
 support MTUs that are "reasonably" large.  All IPv4 end hosts are
 required to accept and reassemble IP packets of size 576 bytes
 [RFC791], but such a small value would clearly be inefficient.
 Because IPv6 omits fragmentation by routers, [RFC2460] specifies a
 larger minimum MTU of 1280 bytes.  Any subnetwork with an internal
 packet payload smaller than 1280 bytes must implement a mechanism
 that performs fragmentation/reassembly of IP packets to/from
 subnetwork frames if it is to support IPv6.
 If a subnetwork cannot directly support a "reasonable" MTU with
 native framing mechanisms, it should internally fragment.  That is,
 it should transparently break IP packets into internal data elements
 and reassemble them at the other end of the subnetwork.

Karn, et al. Best Current Practice [Page 5] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 This leaves the question of what is a "reasonable" MTU.  Ethernet (10
 and 100 Mb/s) has an MTU of 1500 bytes, and because of the ubiquity
 of Ethernet few Internet paths currently have MTUs larger than this
 value.  This severely limits the utility of larger MTUs provided by
 other subnetworks.  Meanwhile, larger MTUs are increasingly desirable
 on high-speed subnetworks to reduce the per-packet processing
 overhead in host computers, and implementers are encouraged to
 provide them even though they may not be usable when Ethernet is also
 in the path.
 Various "tunneling" schemes, such as GRE [RFC2784] or IP Security in
 tunnel mode [RFC2406], treat IP as a subnetwork for IP.  Since
 tunneling adds header overhead, it can trigger fragmentation, even
 when the same physical subnetworks (e.g., Ethernet) are used on both
 sides of the host performing IPsec encapsulation.  Tunneling has made
 it more difficult to avoid router fragmentation and has increased the
 incidence of path MTU black holes [RFC2401] [RFC2923].  Larger
 subnetwork MTUs may help to alleviate this problem.

2.1. Choosing the MTU in Slow Networks

 In slow networks, the largest possible packet may take a considerable
 amount of time to send.  This is known as channelisation or
 serialisation delay.  Total end-to-end interactive response time
 should not exceed the well-known human factors limit of 100 to 200
 ms.  This includes all sources of delay: electromagnetic propagation
 delay, queuing delay, serialisation delay, and the store-and-forward
 time, i.e., the time to transmit a packet at link speed.
 At low link speeds, store-and-forward delays can dominate total
 end-to-end delay; these are in turn directly influenced by the
 maximum transmission unit (MTU) size.  Even when an interactive
 packet is given a higher queuing priority, it may have to wait for a
 large bulk transfer packet to finish transmission.  This worst-case
 wait can be set by an appropriate choice of MTU.
 For example, if the MTU is set to 1500 bytes, then an MTU-sized
 packet will take about 8 milliseconds to send on a T1 (1.536 Mb/s)
 link.  But if the link speed is 19.2kb/s, then the transmission time
 becomes 625 ms -- well above our 100-200ms limit.  A 256-byte MTU
 would lower this delay to a little over 100 ms.  However, care should
 be taken not to lower the MTU excessively, as this will increase
 header overhead and trigger frequent router fragmentation (if Path
 MTU discovery is not in use).  This is likely to be the case with
 multicast, where Path MTU discovery is ineffective.
 One way to limit delay for interactive traffic without imposing a
 small MTU is to give priority to this traffic and to preempt (abort)

Karn, et al. Best Current Practice [Page 6] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 the transmission of a lower-priority packet when a higher priority
 packet arrives in the queue.  However, the link resources used to
 send the aborted packet are lost, and overall throughput will
 Another way to limit delay is to implement a link-level multiplexing
 scheme that allows several packets to be in progress simultaneously,
 with transmission priority given to segments of higher-priority IP
 packets.  For links using the Point-To-Point Protocol (PPP)
 [RFC1661], multi-class multilink [RFC2686] [RFC2687] [RFC2689]
 provides such a facility.
 ATM (asynchronous transfer mode), where SNDUs are fragmented and
 interleaved across smaller 53-byte ATM cells, is another example of
 this technique.  However, ATM is generally used on high-speed links
 where the store-and-forward delays are already minimal, and it
 introduces significant (~9%) increases in overhead due to the
 addition of 5-byte cell overhead to each 48-byte ATM cell.
 A third example is the Data-Over-Cable Service Interface
 Specification (DOCSIS) with typical upstream bandwidths of 2.56 Mb/s
 or 5.12 Mb/s.  To reduce the impact of a 1500-byte MTU in DOCSIS 1.0
 [DOCSIS1], a data link layer fragmentation mechanism is specified in
 DOCSIS 1.1 [DOCSIS2].  To accommodate the installed base, DOCSIS 1.1
 must be backward compatible with DOCSIS 1.0 cable modems, which
 generally do not support fragmentation.  Under the co-existence of
 DOCSIS 1.0 and DOCSIS 1.1, the unfragmented large data packets from
 DOCSIS 1.0 cable modems may affect the quality of service for voice
 packets from DOCSIS 1.1 cable modems.  In this case, it has been
 shown in [DOCSIS3] that the use of bandwidth allocation algorithms
 can mitigate this effect.
 To summarize, there is a fundamental tradeoff between efficiency and
 latency in the design of a subnetwork, and the designer should keep
 this tradeoff in mind.

3. Framing on Connection-Oriented Subnetworks

 IP requires that subnetworks mark the beginning and end of each
 variable-length, asynchronous IP packet.  Some examples of links and
 subnetworks that do not provide this as an intrinsic feature include:
 1.  leased lines carrying a synchronous bit stream;
 2.  ISDN B-channels carrying a synchronous octet stream;
 3.  dialup telephone modems carrying an asynchronous octet stream;

Karn, et al. Best Current Practice [Page 7] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 4.  Asynchronous Transfer Mode (ATM) networks carrying an
     asynchronous stream of fixed-sized "cells".
 The Internet community has defined packet framing methods for all
 these subnetworks.  The Point-To-Point Protocol (PPP) [RFC1661],
 which uses a variant of HDLC, is applicable to bit synchronous,
 octet-synchronous, and octet asynchronous links (i.e., examples 1-3
 above).  PPP is one preferred framing method for IP, since a large
 number of systems interoperate with PPP.  ATM has its own framing
 methods, described in [RFC2684] [RFC2364].
 At high speeds, a subnetwork should provide a framed interface
 capable of carrying asynchronous, variable-length IP datagrams.  The
 maximum packet size supported by this interface is discussed above in
 the MTU/Fragmentation section.  The subnetwork may implement this
 facility in any convenient manner.
 IP packet boundaries need not coincide with any framing or
 synchronization mechanisms internal to the subnetwork.  When the
 subnetwork implements variable sized data units, the most
 straightforward approach is to place exactly one IP packet into each
 subnetwork data unit (SNDU), and to rely on the subnetwork's existing
 ability to delimit SNDUs to also delimit IP packets.  A good example
 is Ethernet.  However, some subnetworks have SNDUs of one or more
 fixed sizes, as dictated by switching, forward error correction
 and/or interleaving considerations.  Examples of such subnetworks
 include ATM, with a single cell payload size of 48 octets plus a 5-
 octet header, and IS-95 digital cellular, with two "rate sets" of
 four fixed frame sizes each that may be selected on 20 millisecond
 Because IP packets are of variable length, they may not necessarily
 fit into an integer multiple of fixed-sized SNDUs.  An "adaptation
 layer" is needed to convert IP packets into SNDUs while marking the
 boundary between each IP packet in some manner.
 There are several approaches to this problem.  The first is to encode
 each IP packet into one or more SNDUs with no SNDU containing pieces
 of more than one IP packet, and to pad out the last SNDU of the
 packet as needed.  Bits in a control header added to each SNDU
 indicate where the data segment belongs in the IP packet.  If the
 subnetwork provides in-order, at-most-once delivery, the header can
 be as simple as a pair of bits indicating whether the SNDU is the
 first and/or the last in the IP packet.  Alternatively, for
 subnetworks that do not reorder the fragments of an SNDU, only the
 last SNDU of the packet could be marked, as this would implicitly

Karn, et al. Best Current Practice [Page 8] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 indicate the next SNDU as the first in a new IP packet.  The AAL5
 (ATM Adaptation Layer 5) scheme used with ATM is an example of this
 approach, though it adds other features, including a payload length
 field and a payload CRC.
 In AAL5, the ATM User-User Indication, which is encoded in the
 Payload Type field of an ATM cell, indicates the last cell of a
 packet.  The packet trailer is located at the end of the SNDU and
 contains the packet length and a CRC.
 Another framing technique is to insert per-segment overhead to
 indicate the presence of a segment option.  When present, the option
 carries a pointer to the end of the packet.  This differs from AAL5
 in that it permits another packet to follow within the same segment.
 MPEG-2 Transport Streams [EN301192] [ISO13818] support this style of
 fragmentation, and may either use padding (limiting each MPEG
 transport stream packet to carry only part of one IP packet), or
 allow a second IP packet to start in the same Transport Stream packet
 (no padding).
 A third approach is to insert a special flag sequence into the data
 stream between each IP packet, and to pack the resulting data stream
 into SNDUs without regard to SNDU boundaries.  This may have
 implications when frames are lost.  The flag sequence can also pad
 unused space at the end of an SNDU.  If the special flag appears in
 the user data, it is escaped to an alternate sequence (usually larger
 than a flag) to avoid being misinterpreted as a flag.  The HDLC-based
 framing schemes used in PPP are all examples of this approach.
 All three adaptation schemes introduce overhead; how much depends on
 the distribution of IP packet sizes, the size(s) of the SNDUs, and in
 the HDLC-like approaches, the content of the IP packet (since flag-
 like sequences occurring in the packet must be escaped, which expands
 them).  The designer must also weigh implementation complexity and
 performance in the choice and design of an adaptation layer.

4. Connection-Oriented Subnetworks

 IP has no notion of a "connection"; it is a purely connectionless
 protocol.  When a connection is required by an application, it is
 usually provided by TCP [RFC793], the Transmission Control Protocol,
 running atop IP on an end-to-end basis.
 Connection-oriented subnetworks can be (and are widely) used to carry
 IP, but often with considerable complexity.  Subnetworks consisting
 of few nodes can simply open a permanent connection between each pair
 of nodes.  This is frequently done with ATM.  However, the number of
 connections increases as the square of the number of nodes, so this

Karn, et al. Best Current Practice [Page 9] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 is clearly impractical for large subnetworks.  A "shim" layer between
 IP and the subnetwork is therefore required to manage connections.
 This is one of the most common functions of a Subnetwork Dependent
 Convergence Function (SNDCF) sublayer between IP and a subnetwork.
 SNDCFs typically open subnetwork connections as needed when an IP
 packet is queued for transmission and close them after an idle
 timeout.  There is no relation between subnetwork connections and any
 connections that may exist at higher layers (e.g., TCP).
 Because Internet traffic is typically bursty and transaction-
 oriented, it is often difficult to pick an optimal idle timeout.  If
 the timeout is too short, subnetwork connections are opened and
 closed rapidly, possibly over-stressing the subnetwork connection
 management system (especially if it was designed for voice traffic
 call holding times).  If the timeout is too long, subnetwork
 connections are idle much of the time, wasting any resources
 dedicated to them by the subnetwork.
 Purely connectionless subnets (such as Ethernet), which have no state
 and dynamically share resources, are optimal for supporting best-
 effort IP, which is stateless and dynamically shares resources.
 Connection-oriented packet networks (such as ATM and Frame Relay),
 which have state and dynamically share resources, are less optimal,
 since best-effort IP does not benefit from the overhead of creating
 and maintaining state.  Connection-oriented circuit-switched networks
 (including the PSTN and ISDN) have state and statically allocate
 resources for a call, and thus require state creation and maintenance
 overhead, but do not benefit from the efficiencies of statistical
 multiplexing sharing of capacity inherent in IP.
 In any event, if an SNDCF that opens and closes subnet connections is
 used to support IP, care should be taken to make sure that connection
 processing in the subnet can keep up with relatively short holding

5. Broadcasting and Discovery

 Subnetworks fall into two categories: point-to-point and shared.  A
 point-to-point subnet has exactly two endpoint components (hosts or
 routers); a shared link has more than two endpoint components, using
 either an inherently broadcast medium (e.g., Ethernet, radio) or a
 switching layer hidden from the network layer (e.g., switched
 Ethernet, Myrinet [MYR95], ATM).  Switched subnetworks handle
 broadcast by copying broadcast packets, providing each interface that
 supports one, or more, systems (hosts or routers) with a copy of each

Karn, et al. Best Current Practice [Page 10] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Several Internet protocols for IPv4 make use of broadcast
 capabilities, including link-layer address lookup (ARP), auto-
 configuration (RARP, BOOTP, DHCP), and routing (RIP).
 A lack of broadcast capability can impede the performance of these
 protocols, or render them inoperable (e.g., DHCP).  ARP-like link
 address lookup can be provided by a centralized database, but at the
 expense of potentially higher response latency and the need for nodes
 to have explicit knowledge of the ARP server address.  Shared links
 should support native, link-layer subnet broadcast.
 A corresponding set of IPv6 protocols uses multicasting (see next
 section) instead of broadcasting to provide similar functions with
 improved scaling in large networks.

6. Multicasting

 The Internet model includes "multicasting", where IP packets are sent
 to all the members of a multicast group [RFC1112] [RFC3376]
 [RFC2710].  Multicast is an option in IPv4, but a standard feature of
 IPv6.  IPv4 multicast is currently used by multimedia,
 teleconferencing, gaming, and file distribution (web, peer-to-peer
 sharing) applications, as well as by some key network and host
 protocols (e.g., RIPv2, OSPF, NTP).  IPv6 additionally relies on
 multicast for network configuration (DHCP-like autoconfiguration) and
 link-layer address discovery [RFC2461] (replacing ARP).  In the case
 of IPv6, this can allow autoconfiguration and address discovery to
 span across routers, whereas the IPv4 broadcast-based services cannot
 without ad-hoc router support [RFC1812].
 Multicast-enabled IP routers organize each multicast group into a
 spanning tree, and route multicast packets by making copies of each
 multicast packet and forwarding the copies to each output interface
 that includes at least one downstream member of the multicast group.
 Multicasting is considerably more efficient when a subnetwork
 explicitly supports it.  For example, a router relaying a multicast
 packet onto an Ethernet segment need send only one copy of the
 packet, no matter how many members of the multicast group are
 connected to the segment.  Without native multicast support, routers
 and switches on shared links would need to use broadcast with
 software filters, such that every multicast packet sent incurs
 software overhead for every node on the subnetwork, even if a node is
 not a member of the multicast group.  Alternately, the router would
 transmit a separate copy to every member of the multicast group on
 the segment, as is done on multicast-incapable switched subnets.

Karn, et al. Best Current Practice [Page 11] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Subnetworks using shared channels (e.g., radio LANs, Ethernets) are
 especially suitable for native multicasting, and their designers
 should make every effort to support it.  This involves designating a
 section of the subnetwork's own address space for multicasting.  On
 these networks, multicast is basically broadcast on the medium, with
 Layer-2 receiver filters.
 Subnet interfaces also need to be designed to accept packets
 addressed to some number of multicast addresses, in addition to the
 unicast packets specifically addressed to them.  The number of
 multicast addresses that needs to be supported by a host depends on
 the requirements of the associated host; at least several dozen will
 meet most current needs.
 On low-speed networks, the multicast address recognition function may
 be readily implemented in host software, but on high-speed networks,
 it should be implemented in subnetwork hardware.  This hardware need
 not be complete; for example, many Ethernet interfaces implement a
 "hashing" function where the IP layer receives all of the multicast
 (and unicast) traffic to which the associated host subscribes, plus
 some small fraction of multicast traffic to which the host does not
 subscribe.  Host/router software then has to discard the unwanted
 packets that pass the Layer-2 multicast address filter [RFC1112].
 There does not need to be a one-to-one mapping between a Layer-2
 multicast address and an IP multicast address.  An address overlap
 may significantly degrade the filtering capability of a receiver's
 hardware multicast address filter.  A subnetwork supporting only
 broadcast should use this service for multicast and must rely on
 software filtering.
 Switched subnetworks must also provide a mechanism for copying
 multicast packets to ensure the packets reach at least all members of
 a multicast group.  One option is to "flood" multicast packets in the
 same manner as broadcast.  This can lead to unnecessary transmissions
 on some subnetwork links (notably non-multicast-aware Ethernet
 switches).  Some subnetworks therefore allow multicast filter tables
 to control which links receive packets belonging to a specific group.
 To configure this automatically requires access to Layer-3 group
 membership information (e.g., IGMP [RFC3376], or MLD [RFC2710]).
 Various implementation options currently exist to provide a subnet
 node with a list of mappings of multicast addresses to
 ports/interfaces.  These employ a range of approaches, including
 signaling from end hosts (e.g., IEEE 802 GARP/GMRP [802.1p]),
 signaling from switches (e.g., CGMP [CGMP] and RGMP [RFC3488]),
 interception and proxy of IP group membership packets (e.g., IGMP/MLD
 Proxy [MAGMA-PROXY]), and enabling Layer-2 devices to
 snoop/inspect/peek into forwarded Layer-3 protocol headers (e.g.,

Karn, et al. Best Current Practice [Page 12] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 IGMP, MLD, PIM) so that they may infer Layer-3 multicast group
 membership [MAGMA-SNOOP].  These approaches differ in their
 complexity, flexibility, and ability to support new protocols.

7. Bandwidth on Demand (BoD) Subnets

 Some subnets allow a number of subnet nodes to share a channel
 efficiently by assigning transmission opportunities dynamically.
 Transmission opportunities are requested by a subnet node when it has
 packets to send.  The subnet schedules and grants transmission
 opportunities sufficient to allow the transmitting subnet node to
 send one or more packets (or packet fragments).  We call these
 subnets Bandwidth on Demand (BoD) subnets.  Examples of BoD subnets
 include Demand Assignment Multiple Access (DAMA) satellite and
 terrestrial wireless networks, IEEE 802.11 point coordination
 function (PCF) mode, and DOCSIS.  A connection-oriented network (such
 as the PSTN, ATM or Frame Relay) reserves resources on a much longer
 timescale, and is therefore not a BoD subnet in our taxonomy.
 The design parameters for BoD are similar to those in connection-
 oriented subnetworks, although the implementations may vary
 significantly.  In BoD, the user typically requests access to the
 shared channel for some duration.  Access may be allocated for a
 period of time at a specific rate, for a certain number of packets,
 or until the user releases the channel.  Access may be coordinated
 through a central management entity or with a distributed algorithm
 amongst the users.  Examples of the resource that may be shared
 include a terrestrial wireless hop, an upstream channel in a cable
 television system, a satellite uplink, and an end-to-end satellite
 Long-delay BoD subnets pose problems similar to connection-oriented
 subnets in anticipating traffic.  While connection-oriented subnets
 hold idle channels open expecting new data to arrive, BoD subnets
 request channel access based on buffer occupancy (or expected buffer
 occupancy) on the sending port.  Poor performance will likely result
 if the sender does not anticipate additional traffic arriving at that
 port during the time it takes to grant a transmission request.  It is
 recommended that the algorithm have the capability to extend a hold
 on the channel for data that has arrived after the original request
 was generated (this may be done by piggybacking new requests on user
 There is a wide variety of BoD protocols available.  However, there
 has been relatively little comprehensive research on the interactions
 between BoD mechanisms and Internet protocol performance.  Research
 on some specific mechanisms is available (e.g., [AR02]).  One item
 that has been studied is TCP's retransmission timer [KY02].  BoD

Karn, et al. Best Current Practice [Page 13] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 systems can cause spurious timeouts when adjusting from a relatively
 high data rate, to a relatively low data rate.  In this case, TCP's
 transmitted data takes longer to get through the network than
 predicted by the TCP sender's computed retransmission timeout.
 Therefore, the TCP sender is prone to resending a segment

8. Reliability and Error Control

 In the Internet architecture, the ultimate responsibility for error
 recovery is at the end points [SRC81].  The Internet may occasionally
 drop, corrupt, duplicate, or reorder packets, and the transport
 protocol (e.g., TCP) or application (e.g., if UDP is used as the
 transport protocol) must recover from these errors on an end-to-end
 basis [RFC3155].  Error recovery in the subnetwork is therefore
 justifiable only to the extent that it can enhance overall
 performance.  It is important to recognize that a subnetwork can go
 too far in attempting to provide error recovery services in the
 Internet environment.  Subnet reliability should be "lightweight",
 i.e., it only has to be "good enough", *not* perfect.
 In this section, we discuss how to analyze characteristics of a
 subnetwork to determine what is "good enough".  The discussion below
 focuses on TCP, which is the most widely-used transport protocol in
 the Internet.  It is widely believed (and is a stated goal within the
 IETF) that non-TCP transport protocols should attempt to be "TCP-
 friendly" and have many of the same performance characteristics.
 Thus, the discussion below should be applicable, even to portions of
 the Internet where TCP may not be the predominant protocol.

8.1. TCP vs Link-Layer Retransmission

 Error recovery involves the generation and transmission of redundant
 information computed from user data.  Depending on how much redundant
 information is sent and how it is generated, the receiver can use it
 to reliably detect transmission errors, correct up to some maximum
 number of transmission errors, or both.  The general approach is
 known as Error Control Coding, or ECC.
 The use of ECC to detect transmission errors so that retransmissions
 (hopefully without errors) can be requested is widely known as "ARQ"
 (Automatic Repeat Request).
 When enough ECC information is available to permit the receiver to
 correct some transmission errors without a retransmission, the
 approach is known as Forward Error Correction (FEC).  Due to the
 greater complexity of the required ECC and the need to tailor its
 design to the characteristics of a specific modem and channel, FEC

Karn, et al. Best Current Practice [Page 14] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 has traditionally been implemented in special-purpose hardware
 integral to a modem.  This effectively makes it part of the physical
 Unlike ARQ, FEC was rarely used for telecommunications outside of
 space links prior to the 1990s.  It is now nearly universal in
 telephone, cable and DSL modems, digital satellite links, and digital
 mobile telephones.  FEC is also heavily used in optical and magnetic
 storage where "retransmissions" are not possible.
 Some systems use hybrid combinations of ARQ layered atop FEC; V.90
 dialup modems (in the upstream direction) with V.42 error control are
 one example.  Most errors are corrected by the trellis (FEC) code
 within the V.90 modem, and most remaining errors are detected and
 corrected by the ARQ mechanisms in V.42.
 Work is now underway to apply FEC above the physical layer, primarily
 in connection with reliable multicasting [RFC3048] [RFC3450-RFC3453]
 where conventional ARQ mechanisms are inefficient or difficult to
 implement.  However, in this discussion, we will assume that if FEC
 is present, it is implemented within the physical layer.
 Depending on the layer in which it is implemented, error control can
 operate on an end-to-end basis or over a shorter span, such as a
 single link.  TCP is the most important example of an end-to-end
 protocol that uses an ARQ strategy.
 Many link-layer protocols use ARQ, usually some flavor of HDLC
 [ISO3309].  Examples include the X.25 link layer, the AX.25 protocol
 used in amateur packet radio, 802.11 wireless LANs, and the reliable
 link layer specified in IEEE 802.2.
 Only end-to-end error recovery can ensure reliable service to the
 application (see Section 8).  However, some subnetworks (e.g., many
 wireless links) also have link-layer error recovery as a performance
 enhancement [RFC3366].  For example, many cellular links have small
 physical frame sizes (< 100 bytes) and relatively high frame loss
 rates.  Relying solely on end-to-end error recovery can clearly yield
 a performance degradation, as retransmissions across the end-to-end
 path take much longer to be received than when link layer
 retransmissions are used.  Thus, link-layer error recovery can often
 increase end-to-end performance.  As a result, link-layer and end-
 to-end recovery often co-exist; this can lead to the possibility of
 inefficient interactions between the two layers of ARQ protocols.
 This inter-layer "competition" might lead to the following wasteful
 situation.  When the link layer retransmits (parts of) a packet, the
 link latency momentarily increases.  Since TCP bases its

Karn, et al. Best Current Practice [Page 15] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 retransmission timeout on prior measurements of total end-to-end
 latency, including that of the link in question, this sudden increase
 in latency may trigger an unnecessary retransmission by TCP of a
 packet that the link layer is still retransmitting.  Such spurious
 end-to-end retransmissions generate unnecessary load and reduce end-
 to-end throughput.  As a result, the link layer may even have
 multiple copies of the same packet in the same link queue at the same
 time.  In general, one could say the competing error recovery is
 caused by an inner control loop (link-layer error recovery) reacting
 to the same signal as an outer control loop (end-to-end error
 recovery) without any coordination between the loops.  Note that this
 is solely an efficiency issue; TCP continues to provide reliable
 end-to-end delivery over such links.
 This raises the question of how persistent a link-layer sender should
 be in performing retransmission [RFC3366].  We define the link-layer
 (LL) ARQ persistency as the maximum time that a particular link will
 spend trying to transfer a packet before it can be discarded.  This
 deliberately simplified definition says nothing about the maximum
 number of retransmissions, retransmission strategies, queue sizes,
 queuing disciplines, transmission delays, or the like.  The reason we
 use the term LL ARQ persistency, instead of a term such as "maximum
 link-layer packet holding time," is that the definition closely
 relates to link-layer error recovery.  For example, on links that
 implement straightforward error recovery strategies, LL ARQ
 persistency will often correspond to a maximum number of
 retransmissions permitted per link-layer frame.
 For link layers that do not or cannot differentiate between flows
 (e.g., due to network layer encryption), the LL ARQ persistency
 should be small.  This avoids any harmful effects or performance
 degradation resulting from indiscriminate high persistence.  A
 detailed discussion of these issues is provided in [RFC3366].
 However, when a link layer can identify individual flows and apply
 ARQ selectively [LKJK02], then the link ARQ persistency should be
 high for a flow using reliable unicast transport protocols (e.g.,
 TCP) and must be low for all other flows.  Setting the link ARQ
 persistency larger than the largest link outage allows TCP to rapidly
 restore transmission without needing to wait for a retransmission
 time out.  This generally improves TCP performance in the face of
 transient outages.  However, excessively high persistence may be
 disadvantageous; a practical upper limit of 30-60 seconds may be
 desirable.  Implementation of such schemes remains a research issue.
 (See also the following section "Recovery from Subnetwork Outages").

Karn, et al. Best Current Practice [Page 16] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Many subnetwork designers have opportunities to reduce the
 probability of packet loss, e.g., with FEC, ARQ, and interleaving, at
 the cost of increased delay.  TCP performance improves with
 decreasing loss but worsens with increasing end-to-end delay, so it
 is important to find the proper balance through analysis and

8.2. Recovery from Subnetwork Outages

 Some types of subnetworks, particularly mobile radio, are subject to
 frequent temporary outages.  For example, an active cellular data
 user may drive or walk into an area (such as a tunnel) that is out of
 range of any base station.  No packets will be delivered successfully
 until the user returns to an area with coverage.
 The Internet protocols currently provide no standard way for a
 subnetwork to explicitly notify an upper layer protocol (e.g., TCP)
 that it is experiencing an outage rather than severe congestion.
 Under these circumstances TCP will, after each unsuccessful
 retransmission, wait even longer before trying again; this is its
 "exponential back-off" algorithm.  Furthermore, TCP will not discover
 that the subnetwork outage has ended until its next retransmission
 attempt.  If TCP has backed off, this may take some time.  This can
 lead to extremely poor TCP performance over such subnetworks.
 It is therefore highly desirable that a subnetwork subject to outages
 does not silently discard packets during an outage.  Ideally, the
 subnetwork should define an interface to the next higher layer (i.e.,
 IP) that allows it to refuse packets during an outage, and to
 automatically ask IP for new packets when it is again able to deliver
 them.  If it cannot do this, then the subnetwork should hold onto at
 least some of the packets it accepts during an outage and attempt to
 deliver them when the outage ends.  When packets are discarded, IP
 should be notified so that the appropriate ICMP messages can be sent.
 Note that it is *not* necessary to completely avoid dropping packets
 during an outage.  The purpose of holding onto a packet during an
 outage, either in the subnetwork or at the IP layer, is so that its
 eventual delivery will implicitly notify TCP that the subnetwork is
 again operational.  This is to enhance performance, not to ensure
 reliability -- reliability, as discussed earlier, can only be ensured
 on an end-to-end basis.
 Only a few packets per TCP connection, including ACKs, need be held
 in this way to cause the TCP sender to recover from the additional
 losses once the flow resumes [RFC3366].

Karn, et al. Best Current Practice [Page 17] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Because it would be a layering violation (and possibly a performance
 hit) for IP or a subnetwork layer to look at TCP headers (which would
 in any event be impossible if IPsec encryption [RFC2401] is in use),
 it would be reasonable for the IP or subnetwork layers to choose, as
 a design parameter, some small number of packets that will be
 retained during an outage.

8.3. CRCs, Checksums and Error Detection

 The TCP [RFC793], UDP [RFC768], ICMP, and IPv4 [RFC791] protocols all
 use the same simple 16-bit 1's complement checksum algorithm
 [RFC1071] to detect corrupted packets.  The IPv4 header checksum
 protects only the IPv4 header, while the TCP, ICMP, and UDP checksums
 provide end-to-end error detection for both the transport pseudo
 header (including network and transport layer information) and the
 transport payload data.  Protection of the data is optional for
 applications using UDP [RFC768] for IPv4, but is required for IPv6.
 The Internet checksum is not very strong from a coding theory
 standpoint, but it is easy to compute in software, and various
 proposals to replace the Internet checksums with stronger checksums
 have failed.  However, it is known that undetected errors can and do
 occur in packets received by end hosts [SP2000].
 To reduce processing costs, IPv6 has no IP header checksum.  The
 destination host detects "important" errors in the IP header, such as
 the delivery of the packet to the wrong destination.  This is done by
 including the IP source and destination addresses (pseudo header) in
 the computation of the checksum in the TCP or UDP header, a practice
 already performed in IPv4.  Errors in other IPv6 header fields may go
 undetected within the network; this was considered a reasonable price
 to pay for a considerable reduction in the processing required by
 each router, and it was assumed that subnetworks would use a strong
 link CRC.
 One way to provide additional protection for an IPv4 or IPv6 header
 is by the authentication and packet integrity services of the IP
 Security (IPsec) protocol [RFC2401].  However, this may not be a
 choice available to the subnetwork designer.
 Most subnetworks implement error detection just above the physical
 layer.  Packets corrupted in transmission are detected and discarded
 before delivery to the IP layer.  A 16-bit cyclic redundancy check
 (CRC) is usually the minimum for error detection.  This is
 significantly more robust against most patterns of errors than the
 16-bit Internet checksum.  Note that the error detection properties
 of a specific CRC code diminish with increasing frame size.  The
 Point-to-Point Protocol [RFC1662] requires support of a 16-bit CRC

Karn, et al. Best Current Practice [Page 18] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 for each link frame, with a 32-bit CRC as an option.  (PPP is often
 used in conjunction with a dialup modem, which provides its own error
 control).  Other subnetworks, including 802.3/Ethernet, AAL5/ATM,
 FDDI, Token Ring, and PPP over SONET/SDH all use a 32-bit CRC.  Many
 subnetworks can also use other mechanisms to enhance the error
 detection capability of the link CRC (e.g., FEC in dialup modems,
 mobile radio and satellite channels).
 Any new subnetwork designed to carry IP should therefore provide
 error detection for each IP packet that is at least as strong as the
 32-bit CRC specified in [ISO3309].  While this will achieve a very
 low undetected packet error rate due to transmission errors, it will
 not (and need not) achieve a very low packet loss rate as the
 Internet protocols are better suited to dealing with lost packets
 than to dealing with corrupted packets [SRC81].
 Packet corruption may be, and is, also caused by bugs in host and
 router hardware and software.  Even if every subnetwork implemented
 strong error detection, it is still essential that end-to-end
 checksums are used at the receiving end host [SP2000].
 Designers of complex subnetworks consisting of internal links and
 packet switches should consider implementing error detection on an
 edge-to-edge basis to cover an entire SNDU (or IP packet).  A CRC
 would be generated at the entry point to the subnetwork and checked
 at the exit endpoint.  This may be used instead of, or in combination
 with, error detection at the interface to each physical link.  An
 edge-to-edge check has the significant advantage of protecting
 against errors introduced anywhere within the subnetwork, not just
 within its transmission links.  Examples of this approach include the
 way in which the Ethernet CRC-32 is handled by LAN bridges [802.1D].
 ATM AAL5 [ITU-I363] also uses an edge-to-edge CRC-32.
 Some specific applications may be tolerant of residual errors in the
 data they exchange, but removal of the link CRC may expose the
 network to an undesirable increase in undetected errors in the IP and
 transport headers.  Applications may also require a high level of
 error protection for control information exchanged by protocols
 acting above the transport layer.  One example is a voice codec,
 which is robust against bit errors in the speech samples.  For such
 mechanisms to work, the receiving application must be able to
 tolerate receiving corrupted data.  This also requires that an
 application uses a mechanism to signal that payload corruption is
 permitted and to indicate the coverage (headers and data) required to
 be protected by the subnetwork CRC.  The UDP-Lite protocol [RFC3828]
 is the first Internet standards track transport protocol supporting
 partial payload protection.  Receipt of corrupt data by arbitrary

Karn, et al. Best Current Practice [Page 19] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 application protocols carries a serious danger that a subnet delivers
 data with errors that remain undetected by the application and hence
 corrupt the communicated data [SRC81].

8.4. How TCP Works

 One of TCP's functions is end-host based congestion control for the
 Internet.  This is a critical part of the overall stability of the
 Internet, so it is important that link-layer designers understand
 TCP's congestion control algorithms.
 TCP assumes that, at the most abstract level, the network consists of
 links and queues.  Queues provide output-buffering on links that are
 momentarily oversubscribed.  They smooth instantaneous traffic bursts
 to fit the link bandwidth.  When demand exceeds link capacity long
 enough to fill the queue, packets must be dropped.  The traditional
 action of dropping the most recent packet ("tail dropping") is no
 longer recommended [RFC2309] [RFC2914], but it is still widely
 TCP uses sequence numbering and acknowledgments (ACKs) on an
 end-to-end basis to provide reliable, sequenced delivery.  TCP ACKs
 are cumulative, i.e., each implicitly ACKs every segment received so
 far.  If a packet with an unexpected sequence number is received, the
 ACK field in the packets returned by the receiver will cease to
 advance.  Using an optional enhancement, TCP can send selective
 acknowledgments (SACKs) [RFC2018] to indicate which segments have
 arrived at the receiver.
 Since the most common cause of packet loss is congestion, TCP treats
 packet loss as an indication of potential Internet congestion along
 the path between TCP end hosts.  This happens automatically, and the
 subnetwork need not know anything about IP or TCP.  A subnetwork node
 simply drops packets whenever it must, though some packet-dropping
 strategies (e.g., RED) are more fair to competing flows than others.
 TCP recovers from packet losses in two different ways.  The most
 important mechanism is the retransmission timeout.  If an ACK fails
 to arrive after a certain period of time, TCP retransmits the oldest
 unacked packet.  Taking this as a hint that the network is congested,
 TCP waits for the retransmission to be ACKed before it continues, and
 it gradually increases the number of packets in flight as long as a
 timeout does not occur again.
 A retransmission timeout can impose a significant performance
 penalty, as the sender is idle during the timeout interval and
 restarts with a congestion window of one TCP segment following the

Karn, et al. Best Current Practice [Page 20] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 timeout.  To allow faster recovery from the occasional lost packet in
 a bulk transfer, an alternate scheme, known as "fast recovery", was
 introduced [RFC2581] [RFC2582] [RFC2914] [TCPF98].
 Fast recovery relies on the fact that when a single packet is lost in
 a bulk transfer, the receiver continues to return ACKs to subsequent
 data packets that do not actually acknowledge any newly-received
 data.  These are known as "duplicate acknowledgments" or "dupacks".
 The sending TCP can use dupacks as a hint that a packet has been lost
 and retransmit it without waiting for a timeout.  Dupacks effectively
 constitute a negative acknowledgment (NAK) for the packet sequence
 number in the acknowledgment field.  TCP waits until a certain number
 of dupacks (currently 3) are seen prior to assuming a loss has
 occurred; this helps avoid an unnecessary retransmission during
 out-of-sequence delivery.
 A technique called "Explicit Congestion Notification" (ECN) [RFC3168]
 allows routers to directly signal congestion to hosts without
 dropping packets.  This is done by setting a bit in the IP header.
 Since ECN support is likely to remain optional, the lack of an ECN
 bit must *never* be interpreted as a lack of congestion.  Thus, for
 the foreseeable future, TCP must interpret a lost packet as a signal
 of congestion.
 The TCP "congestion avoidance" [RFC2581] algorithm maintains a
 congestion window (cwnd) controlling the amount of data TCP may have
 in flight at any moment.  Reducing cwnd reduces the overall bandwidth
 obtained by the connection; similarly, raising cwnd increases
 performance, up to the limit of the available capacity.
 TCP probes for available network capacity by initially setting cwnd
 to one or two packets and then increasing cwnd by one packet for each
 ACK returned from the receiver.  This is TCP's "slow start"
 mechanism.  When a packet loss is detected (or congestion is signaled
 by other mechanisms), cwnd is reset to one and the slow start process
 is repeated until cwnd reaches one half of its previous setting
 before the reset.  Cwnd continues to increase past this point, but at
 a much slower rate than before.  If no further losses occur, cwnd
 will ultimately reach the window size advertised by the receiver.
 This is an "Additive Increase, Multiplicative Decrease" (AIMD)
 algorithm.  The steep decrease of cwnd in response to congestion
 provides for network stability; the AIMD algorithm also provides for
 fairness between long running TCP connections sharing the same path.

Karn, et al. Best Current Practice [Page 21] RFC 3819 Advice for Internet Subnetwork Designers July 2004

8.5. TCP Performance Characteristics

 Here we present a current "state-of-the-art" understanding of TCP
 performance.  This analysis attempts to characterize the performance
 of TCP connections over links of varying characteristics.
 Link designers may wish to use the techniques in this section to
 predict what performance TCP/IP may achieve over a new link-layer
 design.  Such analysis is encouraged.  Because this is a relatively
 new analysis, and the theory is based on single-stream TCP
 connections under "ideal" conditions, it should be recognized that
 the results of such analysis may differ from actual performance in
 the Internet.  That being said, we have done our best to provide the
 designers with helpful information to get an accurate picture of the
 capabilities and limitations of TCP under various conditions.

8.5.1. The Formulae

 The performance of TCP's AIMD Congestion Avoidance algorithm has been
 extensively analyzed.  The current best formula for the performance
 of the specific algorithms used by Reno TCP (i.e., the TCP specified
 in [RFC2581]) is given by Padhye, et al. [PFTK98].  This formula is:
         BW = --------------------------------------------------------
              RTT*sqrt(1.33*p) + RTO*p*[1+32*p^2]*min[1,3*sqrt(.75*p)]
         BW   is the maximum TCP throughout achievable by an
              individual TCP flow
         MSS  is the TCP segment size being used by the connection
         RTT  is the end-to-end round trip time of the TCP connection
         RTO  is the packet timeout (based on RTT)
         p    is the packet loss rate for the path
              (i.e., .01 if there is 1% packet loss)
 Note that the speed of the links making up the Internet path does not
 explicitly appear in this formula.  Attempting to send faster than
 the slowest link in the path causes the queue to grow at the
 transmitter driving the bottleneck.  This increases the RTT, which in
 turn reduces the achievable throughput.
 This is currently considered to be the best approximate formula for
 Reno TCP performance.  A further simplification of this formula is
 generally made by assuming that RTO is approximately 5*RTT.

Karn, et al. Best Current Practice [Page 22] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 TCP is constantly being improved.  A simpler formula, which gives an
 upper bound on the performance of any AIMD algorithm which is likely
 to be implemented in TCP in the future, was derived by Ott, et al.
                   MSS   1
         BW = C    --- -------
                   RTT sqrt(p)
 where C is 0.93.

8.5.2. Assumptions

 Both formulae assume that the TCP Receiver Window is not limiting the
 performance of the connection.  Because the receiver window is
 entirely determined by end-hosts, we assume that hosts will maximize
 the announced receiver window to maximize their network performance.
 Both of these formulae allow BW to become infinite if there is no
 loss.  However, an Internet path will drop packets at bottlenecked
 queues if the load is too high.  Thus, a completely lossless TCP/IP
 network can never occur (unless the network is being underutilized).
 The RTT used is the arithmetic average, including queuing delays.
 The formulae are for a single TCP connection.  If a path carries many
 TCP connections, each will follow the formulae above independently.
 The formulae assume long-running TCP connections.  For connections
 that are extremely short (<10 packets) and don't lose any packets,
 performance is driven by the TCP slow-start algorithm.  For
 connections of medium length, where on average only a few segments
 are lost, single connection performance will actually be slightly
 better than given by the formulae above.
 The difference between the simple and complex formulae above is that
 the complex formula includes the effects of TCP retransmission
 timeouts.  For very low levels of packet loss (significantly less
 than 1%), timeouts are unlikely to occur, and the formulae lead to
 very similar results.  At higher packet losses (1% and above), the
 complex formula gives a more accurate estimate of performance (which
 will always be significantly lower than the result from the simple
 Note that these formulae break down as p approaches 100%.

Karn, et al. Best Current Practice [Page 23] RFC 3819 Advice for Internet Subnetwork Designers July 2004

8.5.3. Analysis of Link-Layer Effects on TCP Performance

 Consider the following example:
 A designer invents a new wireless link layer which, on average, loses
 1% of IP packets.  The link layer supports packets of up to 1040
 bytes, and has a one-way delay of 20 msec.
 If this link were to be used on an Internet path with a round trip
 time greater than 80ms, the upper bound may be computed by:
 For MSS, use 1000 bytes to exclude the 40 bytes of minimum IPv4 and
 TCP headers.
 For RTT, use 120 msec (80 msec for the Internet part, plus 20 msec
 each way for the new wireless link).
 For p, use .01.  For C, assume 1.
 The simple formula gives:
    BW = (1000 * 8 bits) / (.120 sec * sqrt(.01)) = 666 kbit/sec
 The more complex formula gives:
    BW = 402.9 kbit/sec
 If this were a 2 Mb/s wireless LAN, the designers might be somewhat
 Some observations on performance:
 1.  We have assumed that the packet losses on the link layer are
     interpreted as congestion by TCP.  This is a "fact of life" that
     must be accepted.
 2.  The equations for TCP performance are all expressed in terms of
     packet loss, but many subnetwork designers think in terms of
     bit-error ratio.  *If* channel bit errors are independent, then
     the probability of a packet being corrupted is:
       p = 1 - ([1 - BER]^[FRAME_SIZE*8])
     Here we assume FRAME_SIZE is in bytes and "^" represents
     exponentiation.  It includes the user data and all headers
     (TCP,IP and subnetwork).  (Note: this analysis assumes the

Karn, et al. Best Current Practice [Page 24] RFC 3819 Advice for Internet Subnetwork Designers July 2004

     subnetwork does not perform ARQ or transparent fragmentation
     [RFC3366].)  If the inequality
       BER * [FRAME_SIZE*8] << 1
     holds, the packet loss probability p can be approximated by:
       p = BER * [FRAME_SIZE*8]
     These equations can be used to apply BER to the performance
     equations above.
     Note that FRAME_SIZE can vary from one packet to the next.  Small
     packets (such as TCP acks) generally have a smaller probability
     of packet error than, say, a TCP packet carrying one MSS (maximum
     segment size) of user data.  A flow of small TCP acks can be
     expected to be slightly more reliable than a stream of larger TCP
     data segments.
     It bears repeating that the above analysis assumes that bit
     errors are statistically independent.  Because this is not true
     for many real links, our computation of p is actually an upper
     bound, not the exact probability of packet loss.
     There are many reasons why bit errors are not independent on real
     links.  Many radio links are affected by propagation fading or by
     interference that lasts over many bit times.  Also, links with
     Forward Error Correction (FEC) generally have very non-uniform
     bit error distributions that depend on the type of FEC, but in
     general the uncorrected errors tend to occur in bursts even when
     channel symbol errors are independent.  In all such cases, our
     computation of p from BER can only place an upper limit on the
     packet loss rate.
     If the distribution of errors under the FEC scheme is known, one
     could apply the same type of analysis as above, using the correct
     distribution function for the BER.  It is more likely in these
     FEC cases, however, that empirical methods are needed to
     determine the actual packet loss rate.
 3.  Note that the packet size plays an important role.  If the
     subnetwork loss characteristics are such that large packets have
     the same probability of loss as smaller packets, then larger
     packets will yield improved performance.

Karn, et al. Best Current Practice [Page 25] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 4.  We have chosen a specific RTT that might occur on a wide-area
     Internet path within the USA.  It is important to recognize that
     a variety of RTT values are experienced in the Internet.
     For example, RTTs are typically less than 10 msec in a wired LAN
     environment when communicating with a local host.  International
     connections may have RTTs of 200 msec or more.  Modems and other
     low-capacity links can add considerable delay due to their long
     packet transmission (serialisation) times.
     Links over geostationary repeater satellites have one-way speed-
     of-light delays of around 250ms, a minimum of 125ms propagation
     delay up to the satellite and 125ms down.  The RTT of an end-to-
     end TCP connection that includes such a link can be expected to
     be greater than 250ms.
     Queues on heavily-congested links may back up, increasing RTTs.
     Finally, virtual private networks (VPNs) and other forms of
     encryption and tunneling can add significant end-to-end delay to
     network connections.

9. Quality-of-Service (QoS) considerations

 It is generally recognized that specific service guarantees are
 needed to support real-time multimedia, toll-quality telephony, and
 other performance-critical applications.  The provision of such
 Quality of Service guarantees in the Internet is an active area of
 research and standardization.  The IETF has not converged on a single
 service model, set of services, or single mechanism that will offer
 useful guarantees to applications and be scalable to the Internet.
 Indeed, the IETF does not have a single definition of Quality of
 Service.  [RFC2990] represents a current understanding of the
 challenges in architecting QoS for the Internet.
 There are presently two architectural approaches to providing
 mechanisms for QoS support in the Internet.
 IP Integrated Services (Intserv) [RFC1633] provides fine-grained
 service guarantees to individual flows.  Flows are identified by a
 flow specification (flowspec), which creates a stateful association
 between individual packets by matching fields in the packet header.
 Capacity is reserved for the flow, and appropriate traffic
 conditioning and scheduling is installed in routers along the path.
 The ReSerVation Protocol (RSVP) [RFC2205] [RFC2210] is usually, but
 need not necessarily be, used to install the flow QoS state.  Intserv
 defines two services, in addition to the Default (best effort)

Karn, et al. Best Current Practice [Page 26] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 1.  Guaranteed Service (GS) [RFC2212] offers hard upper bounds on
     delay to flows that conform to a traffic specification (TSpec).
     It uses a fluid-flow model to relate the TSpec and reserved
     bandwidth (RSpec) to variable delay.  Non-conforming packets are
     forwarded on a best-effort basis.
 2.  Controlled Load Service (CLS) [RFC2211] offers delay and packet
     loss equivalent to that of an unloaded network to flows that
     conform to a TSpec, but no hard bounds.  Non-conforming packets
     are forwarded on a best-effort basis.
 Intserv requires installation of state information in every
 participating router.  Performance guarantees cannot be made unless
 this state is present in every router along the path.  This, along
 with RSVP processing and the need for usage-based accounting, is
 believed to have scalability problems, particularly in the core of
 the Internet [RFC2208].
 IP Differentiated Services (Diffserv) [RFC2475] provides a "toolkit"
 offering coarse-grained controls to aggregates of flows.  Diffserv in
 itself does *not* provide QoS guarantees, but can be used to
 construct services with QoS guarantees across a Diffserv domain.
 Diffserv attempts to address the scaling issues associated with
 Intserv by requiring state awareness only at the edge of a Diffserv
 domain.  At the edge, packets are classified into flows, and the
 flows are conditioned (marked, policed, or shaped) to a traffic
 conditioning specification (TCS).  A Diffserv Codepoint (DSCP),
 identifying a per-hop behavior (PHB), is set in each packet header.
 The DSCP is carried in the DS-field, subsuming six bits of the former
 Type-of-Service (ToS) byte [RFC791] of the IP header [RFC2474].   The
 PHB denotes the forwarding behavior to be applied to the packet in
 each node in the Diffserv domain.  Although there is a "recommended"
 DSCP associated with each PHB, the mappings from DSCPs to PHBs are
 defined by the DS-domain.  In fact, there can be several DSCPs
 associated with the same PHB.  Diffserv presently defines three PHBs.
 1.  The class selector PHB [RFC2474] replaces the IP precedence field
     of the former ToS byte.  It offers relative forwarding
 2.  The Expedited Forwarding (EF) PHB [RFC3246] [RFC3248] guarantees
     that packets will have a well-defined minimum departure rate
     which, if not exceeded, ensures that the associated queues are
     short or empty.  EF is intended to support services that offer
     tightly-bounded loss, delay, and delay jitter.

Karn, et al. Best Current Practice [Page 27] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 3.  The Assured Forwarding (AF) PHB group [RFC2597] offers different
     levels of forwarding assurance for each aggregated flow of
     packets.  Each AF group is independently allocated forwarding
     resources.  Packets are marked with one of three drop
     precedences; those with the highest drop precedence are dropped
     with lower probability than those marked with the lowest drop
     precedence.  DSCPs are recommended for four independent AF
     groups, although a DS domain can have more or fewer AF groups.
 Ongoing work in the IETF is addressing ways to support Intserv with
 Diffserv.  There is some belief (e.g., as expressed in [RFC2990])
 that such an approach will allow individual flows to receive service
 guarantees and scale to the global Internet.
 The QoS guarantees that can be offered by the IP layer are a product
 of two factors:
 1.  the concatenation of the QoS guarantees offered by the subnets
     along the path of a flow.  This implies that a subnet may wish to
     offer multiple services (with different QoS guarantees) to the IP
     layer, which can then determine which flows use which subnet
     service.  To put it another way, forwarding behavior in the
     subnet needs to be "clued" by the forwarding behavior (service or
     PHB) at the IP layer, and
 2.  the operation of a set of cooperating mechanisms, such as
     bandwidth reservation and admission control, policy management,
     traffic classification, traffic conditioning (marking, policing
     and/or shaping), selective discard, queuing, and scheduling.
     Note that support for QoS in subnets may require similar
     mechanisms, especially when these subnets are general topology
     subnets (e.g., ATM, frame relay, or MPLS) or shared media
 Many subnetwork designers face inherent tradeoffs between delay,
 throughput, reliability, and cost.  Other subnetworks have parameters
 that manage bandwidth, internal connection state, and the like.
 Therefore, the following subnetwork capabilities may be desirable,
 although some might be trivial or moot if the subnet is a dedicated
 point-to-point link.
 1.  The subnetwork should have the ability to reserve bandwidth for a
     connection or flow and schedule packets accordingly.
 2.  Bandwidth reservations should be based on a one- or two-token
     bucket model, depending on whether the service is intended to
     support constant-rate or bursty traffic.

Karn, et al. Best Current Practice [Page 28] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 3.  If a connection or flow does not use its reserved bandwidth at a
     given time, the unused bandwidth should be available for other
 4.  Packets in excess of a connection or flow's agreed rate should be
     forwarded as best-effort or discarded, depending on the service
     offered by the subnet to the IP layer.
 5.  If a subnet contains error control mechanisms (retransmission
     and/or FEC), it should be possible for the IP layer to influence
     the inherent tradeoffs between uncorrected errors, packet losses,
     and delay.  These capabilities at the subnet/IP layer service
     boundary correspond to selection of more or less error control
     and/or to selection of particular error control mechanisms within
     the subnetwork.
 6.  The subnet layer should know, and be able to inform the IP layer,
     how much fixed delay and delay jitter it offers for a flow or
     connection.  If the Intserv model is used, the delay jitter
     component may be best expressed in terms of the TSpec/RSpec model
     described in [RFC2212].
 7.  Support of the Diffserv class selectors [RFC2474] suggests that
     the subnet might consider mechanisms that support priorities.

10. Fairness vs Performance

 Subnetwork designers should be aware of the tradeoffs between
 fairness and efficiency inherent in many transmission scheduling
 algorithms.  For example, many local area networks use contention
 protocols to resolve access to a shared transmission channel.  These
 protocols represent overhead.  While limiting the amount of data that
 a subnet node may transmit per contention cycle helps assure timely
 access to the channel for each subnet node, it also increases
 contention overhead per unit of data sent.
 In some mobile radio networks, capacity is limited by interference,
 which in turn depends on average transmitter power.  Some receivers
 may require considerably more transmitter power (generating more
 interference and consuming more channel capacity) than others.
 In each case, the scheduling algorithm designer must balance
 competing objectives: providing a fair share of capacity to each
 subnet node while maximizing the total capacity of the network.  One
 approach for balancing performance and fairness is outlined in

Karn, et al. Best Current Practice [Page 29] RFC 3819 Advice for Internet Subnetwork Designers July 2004

11. Delay Characteristics

 The TCP sender bases its retransmission timeout (RTO) on measurements
 of the round trip delay experienced by previous packets.  This allows
 TCP to adapt automatically to the very wide range of delays found on
 the Internet.  The recommended algorithms are described in [RFC2988].
 Evaluations of TCP's retransmission timer can be found in [AP99] and
 These algorithms model the delay along an Internet path as a
 normally-distributed random variable with a slowly-varying mean and
 standard deviation.  TCP estimates these two parameters by
 exponentially smoothing individual delay measurements, and it sets
 the RTO to the estimated mean delay plus some fixed number of
 standard deviations.  (The algorithm actually uses mean deviation as
 an approximation to standard deviation, because it is easier to
 The goal is to compute an RTO that is small enough to detect and
 recover from packet losses while minimizing unnecessary ("spurious")
 retransmissions when packets are unexpectedly delayed but not lost.
 Although these goals conflict, the algorithm works well when the
 delay variance along the Internet path is low, or the packet loss
 rate is low.
 If the path delay variance is high, TCP sets an RTO that is much
 larger than the mean of the measured delays.  If the packet loss rate
 is low, the large RTO is of little consequence, as timeouts occur
 only rarely.  Conversely, if the path delay variance is low, then TCP
 recovers quickly from lost packets; again, the algorithm works well.
 However, when delay variance and the packet loss rate are both high,
 these algorithms perform poorly, especially when the mean delay is
 also high.
 Because TCP uses returning acknowledgments as a "clock" to time the
 transmission of additional data, excessively high delays (even if the
 delay variance is low) also affect TCP's ability to fully utilize a
 high-speed transmission pipe.  It also slows the recovery of lost
 packets, even when delay variance is small.
 Subnetwork designers should therefore minimize all three parameters
 (delay, delay variance, and packet loss) as much as possible.
 In many subnetworks, these parameters are inherently in conflict.
 For example, on a mobile radio channel, the subnetwork designer can
 use retransmission (ARQ) and/or forward error correction (FEC) to
 trade off delay, delay variance, and packet loss in an effort to
 improve TCP performance.  While ARQ increases delay variance, FEC

Karn, et al. Best Current Practice [Page 30] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 does not.  However, FEC (especially when combined with interleaving)
 often increases mean delay, even on good channels where ARQ
 retransmissions are not needed and ARQ would not increase either the
 delay or the delay variance.
 The tradeoffs among these error control mechanisms and their
 interactions with TCP can be quite complex, and are the subject of
 much ongoing research.  We therefore recommend that subnetwork
 designers provide as much flexibility as possible in the
 implementation of these mechanisms, and provide access to them as
 discussed above in the section on Quality of Service.

12. Bandwidth Asymmetries

 Some subnetworks may provide asymmetric bandwidth (or may cause TCP
 packet flows to experience asymmetry in the capacity) and the
 Internet protocol suite will generally still work fine.  However,
 there is a case when such a scenario reduces TCP performance.  Since
 TCP data segments are "clocked" out by returning acknowledgments, TCP
 senders are limited by the rate at which ACKs can be returned
 [BPK98].  Therefore, when the ratio of the available capacity of the
 Internet path carrying the data to the bandwidth of the return path
 of the acknowledgments is too large, the slow return of the ACKs
 directly impacts performance.  Since ACKs are generally smaller than
 data segments, TCP can tolerate some asymmetry, but as a general
 rule, designers of subnetworks should be aware that subnetworks with
 significant asymmetry can result in reduced performance, unless
 issues are taken to mitigate this [RFC3449].
 Several strategies have been identified for reducing the impact of
 asymmetry of the network path between two TCP end hosts, e.g.,
 [RFC3449].  These techniques attempt to reduce the number of ACKs
 transmitted over the return path (low bandwidth channel) by changes
 at the end host(s), and/or by modification of subnetwork packet
 forwarding.  While these solutions may mitigate the performance
 issues caused by asymmetric subnetworks, they do have associated cost
 and may have other implications.  A fuller discussion of strategies
 and their implications is provided in [RFC3449].

13. Buffering, flow and congestion control

 Many subnets include multiple links with varying traffic demands and
 possibly different transmission speeds.  At each link there must be a
 queuing system, including buffering, scheduling, and a capability to
 discard excess subnet packets.  These queues may also be part of a
 subnet flow control or congestion control scheme.

Karn, et al. Best Current Practice [Page 31] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 For the purpose of this discussion, we talk about packets without
 regard to whether they refer to a complete IP packet or a subnetwork
 frame.  At each queue, a packet experiences a delay that depends on
 competing traffic and the scheduling discipline, and is subjected to
 a local discarding policy.
 Some subnets may have flow or congestion control mechanisms in
 addition to packet dropping.  Such mechanisms can operate on
 components in the subnet layer, such as schedulers, shapers, or
 discarders, and can affect the operation of IP forwarders at the
 edges of the subnet.  However, with the exception of Explicit
 Congestion Notification [RFC3168] (discussed below), IP has no way to
 pass explicit congestion or flow control signals to TCP.
 TCP traffic, especially aggregated TCP traffic, is bursty.  As a
 result, instantaneous queue depths can vary dramatically, even in
 nominally stable networks.  For optimal performance, packets should
 be dropped in a controlled fashion, not just when buffer space is
 unavailable.  How much buffer space should be supplied is still a
 matter of debate, but as a rule of thumb, each node should have
 enough buffering to hold one link_bandwidth*link_delay product's
 worth of data for each TCP connection sharing the link.
 This is often difficult to estimate, since it depends on parameters
 beyond the subnetwork's control or knowledge.  Internet nodes
 generally do not implement admission control policies, and cannot
 limit the number of TCP connections that use them.  In general, it is
 wise to err in favor of too much buffering rather than too little.
 It may also be useful for subnets to incorporate mechanisms that
 measure propagation delays to assist in buffer sizing calculations.
 There is a rough consensus in the research community that active
 queue management is important to improving fairness, link
 utilization, and throughput [RFC2309].  Although there are questions
 and concerns about the effectiveness of active queue management
 (e.g., [MBDL99]), it is widely considered an improvement over tail-
 drop discard policies.
 One form of active queue management is the Random Early Detection
 (RED) algorithm [RED93], a family of related algorithms.  In one
 version of RED, an exponentially-weighted moving average of the queue
 depth is maintained:
    When this average queue depth is between a maximum threshold
    max_th and a minimum threshold min_th, the probability of packets
    that are dropped is proportional to the amount by which the
    average queue depth exceeds min_th.

Karn, et al. Best Current Practice [Page 32] RFC 3819 Advice for Internet Subnetwork Designers July 2004

    When this average queue depth is equal to max_th, the drop
    probability is equal to a configurable parameter max_p.
    When this average queue depth is greater than max_th, packets are
    always dropped.
 Numerous variants on RED appear in the literature, and there are
 other active queue management algorithms which claim various
 advantages over RED [GM02].
 With an active queue management algorithm, dropped packets become a
 feedback signal to trigger more appropriate congestion behavior by
 the TCPs in the end hosts.  Randomization of dropping tends to break
 up the observed tendency of TCP windows belonging to different TCP
 connections to become synchronized by correlated drops, and it also
 imposes a degree of fairness on those connections that implement TCP
 congestion avoidance properly.  Another important property of active
 queue management algorithms is that they attempt to keep average
 queue depths short while accommodating large short-term bursts.
 Since TCP neither knows nor cares whether congestive packet loss
 occurs at the IP layer or in a subnet, it may be advisable for
 subnets that perform queuing and discarding to consider implementing
 some form of active queue management.  This is especially true if
 large aggregates of TCP connections are likely to share the same
 queue.  However, active queue management may be less effective in the
 case of many queues carrying smaller aggregates of TCP connections,
 e.g., in an ATM switch that implements per-VC queuing.
 Note that the performance of active queue management algorithms is
 highly sensitive to settings of configurable parameters, and also to
 factors such as RTT [MBB00] [FB00].
 Some subnets, most notably ATM, perform segmentation and reassembly
 at the subnetwork edges.  Care should be taken here in designing
 discard policies.  If the subnet discards a fragment of an IP packet,
 then the remaining fragments become an unproductive load on the
 subnet that can markedly degrade end-to-end performance [RF95].
 Subnetworks should therefore attempt to discard these extra fragments
 whenever one of them must be discarded.  If the IP packet has already
 been partially forwarded when discarding becomes necessary, then
 every remaining fragment except the one marking the end of the IP
 packet should also be discarded.  For ATM subnets, this specifically
 means using Early Packet Discard and Partial Packet Discard [ATMFTM].
 Some subnets include flow control mechanisms that effectively require
 that the rate of traffic flows be shaped upon entry to the subnet.
 One example of such a subnet mechanism is in the ATM Available Bit

Karn, et al. Best Current Practice [Page 33] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 rate (ABR) service category [ATMFTM].  Such flow control mechanisms
 have the effect of making the subnet nearly lossless by pushing
 congestion into the IP routers at the edges of the subnet.  In such a
 case, adequate buffering and discard policies are needed in these
 routers to deal with a subnet that appears to have varying bandwidth.
 Whether there is a benefit in this kind of flow control is
 controversial; there are numerous simulation and analytical studies
 that go both ways.  It appears that some of the issues leading to
 such different results include sensitivity to ABR parameters, use of
 binary rather than explicit rate feedback, use (or not) of per-VC
 queuing, and the specific ATM switch algorithms selected for the
 study.  Anecdotally, some large networks that used IP over ABR to
 carry TCP traffic have claimed it to be successful, but have
 published no results.
 Another possible approach to flow control in the subnet would be to
 work with TCP Explicit Congestion Notification (ECN) semantics
 [RFC3168] through utilizing explicit congestion indicators in subnet
 frames.  Routers at the edges of the subnet, rather than shaping,
 would set the explicit congestion bit in those IP packets that are
 received in subnet frames that have an ECN indication.  Nodes in the
 subnet would need to implement an active queue management protocol
 that marks subnet frames instead of dropping them.
 ECN is currently a proposed standard, but it is not yet widely

14. Compression

 Application data compression is a function that can usually be
 omitted in the subnetwork.  The endpoints typically have more CPU and
 memory resources to run a compression algorithm and a better
 understanding of what is being compressed.  End-to-end compression
 benefits every network element in the path, while subnetwork-layer
 compression, by definition, benefits only a single subnetwork.
 Data presented to the subnetwork layer may already be in a compressed
 format (e.g., a JPEG file), compressed at the application layer
 (e.g., the optional "gzip", "compress", and "deflate" compression in
 HTTP/1.1 [RFC2616]), or compressed at the IP layer (the IP Payload
 Compression Protocol [RFC3173] supports DEFLATE [RFC2394] and LZS
 [RFC2395]).  Compression at the subnetwork edges is of no benefit for
 any of these cases.
 The subnetwork may also process data that has been encrypted by the
 application (OpenPGP [RFC2440] or S/MIME [RFC2633]), just above TCP
 (SSL, TLS [RFC2246]), or just above IP (IPsec ESP [RFC2406]).

Karn, et al. Best Current Practice [Page 34] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Ciphers generate high-entropy bit streams lacking any patterns that
 can be exploited by a compression algorithm.
 However, much data is still transmitted uncompressed over the
 Internet, so subnetwork compression may be beneficial.  Any
 subnetwork compression algorithm must not expand uncompressible data,
 e.g., data that has already been compressed or encrypted.
 We make a strong recommendation that subnetworks operating at low
 speed or with small MTUs compress IP and transport-level headers (TCP
 and UDP) using several header compression schemes developed within
 the IETF [RFC3150].  An uncompressed 40-byte TCP/IP header takes
 about 33 milliseconds to send at 9600 bps.  "VJ" TCP/IP header
 compression [RFC1144] compresses most headers to 3-5 bytes, reducing
 transmission time to several milliseconds on dialup modem links.
 This is especially beneficial for small, latency-sensitive packets in
 interactive sessions.
 Similarly, RTP compression schemes, such as CRTP [RFC2508] and ROHC
 [RFC3095], compress most IP/UDP/RTP headers to 1-4 bytes.  The
 resulting savings are especially significant when audio packets are
 kept small to minimize store-and-forward latency.
 Designers should consider the effect of the subnetwork error rate on
 the performance of header compression.  TCP ordinarily recovers from
 lost packets by retransmitting only those packets that were actually
 lost; packets arriving correctly after a packet loss are kept on a
 resequencing queue and do not need to be retransmitted.  In VJ TCP/IP
 [RFC1144] header compression, however, the receiver cannot explicitly
 notify a sender of data corruption and subsequent loss of
 synchronization between compressor and decompressor.  It relies
 instead on TCP retransmission to re-synchronize the decompressor.
 After a packet is lost, the decompressor must discard every
 subsequent packet, even if the subnetwork makes no further errors,
 until the sending TCP retransmits to re-synchronize the decompressor.
 This effect can substantially magnify the effect of subnetwork packet
 losses if the sending TCP window is large, as it will often be on a
 path with a large bandwidth*delay product [LRKOJ99].
 Alternate header compression schemes, such as those described in
 [RFC2507], include an explicit request for retransmission of an
 uncompressed packet to allow decompressor resynchronization without
 waiting for a TCP retransmission.  However, these schemes are not yet
 in widespread use.
 Both TCP header compression schemes do not compress widely-used TCP
 options such as selective acknowledgements (SACK).  Both fail to
 compress TCP traffic that makes use of explicit congestion

Karn, et al. Best Current Practice [Page 35] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 notification (ECN).  Work is under way in the IETF ROHC WG to address
 these shortcomings in a ROHC header compression scheme for TCP
 [RFC3095] [RFC3096].
 The subnetwork error rate also is important for RTP header
 compression.  CRTP uses delta encoding, so a packet loss on the link
 causes uncertainty about the subsequent packets, which often must be
 discarded until the decompressor has notified the compressor and the
 compressor has sent re-synchronizing information.  This typically
 takes slightly more than the end-to-end path round-trip time.  For
 links that combine significant error rates with latencies that
 require multiple packets to be in flight at a time, this leads to
 significant error propagation, i.e., subsequent losses caused by an
 initial loss.
 For links that are both high-latency (multiple packets in flight from
 a typical RTP stream) and error-prone, RTP ROHC provides a more
 robust way of RTP header compression, at a cost of higher complexity
 at the compressor and decompressor.  For example, within a talk
 spurt, only extended losses of (depending on the mode chosen) 12-64
 packets typically cause error propagation.

15. Packet Reordering

 The Internet architecture does not guarantee that packets will arrive
 in the same order in which they were originally transmitted;
 transport protocols like TCP must take this into account.
 However, reordering does come at a cost with TCP as it is currently
 defined.  Because TCP returns a cumulative acknowledgment (ACK)
 indicating the last in-order segment that has arrived, out-of-order
 segments cause a TCP receiver to transmit a duplicate acknowledgment.
 When the TCP sender notices three duplicate acknowledgments, it
 assumes that a segment was dropped by the network and uses the fast
 retransmit algorithm [Jac90] [RFC2581] to resend the segment.  In
 addition, the congestion window is reduced by half, effectively
 halving TCP's sending rate.  If a subnetwork reorders segments
 significantly such that three duplicate ACKs are generated, the TCP
 sender needlessly reduces the congestion window and performance
 Packet reordering frequently occurs in parts of the Internet, and it
 seems to be difficult or impossible to eliminate [BPS99].  For this
 reason, research on improving TCP's behavior in the face of packet
 reordering [LK00] [BA02] has begun.

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 [BPS99] cites reasons why it may even be undesirable to eliminate
 reordering.  There are situations where average packet latency can be
 reduced, link efficiency can be increased, and/or reliability can be
 improved if reordering is permitted.  Examples include certain high
 speed switches within the Internet backbone and the parallel links
 used over many Internet paths for load splitting and redundancy.
 This suggests that subnetwork implementers should try to avoid packet
 reordering whenever possible, but not if doing so compromises
 efficiency, impairs reliability, or increases average packet delay.
 Note that every header compression scheme currently standardized for
 the Internet requires in-order packet delivery on the link between
 compressor and decompressor.  PPP is frequently used to carry
 compressed TCP/IP packets; since it was originally designed for
 point-to-point and dialup links, it is assumed to provide in-order
 delivery.  For this reason, subnetwork implementers who provide PPP
 interfaces to VPNs and other more complex subnetworks, must also
 maintain in-order delivery of PPP frames.

16. Mobility

 Internet users are increasingly mobile.  Not only are many Internet
 nodes laptop computers, but pocket organizers and mobile embedded
 systems are also becoming nodes on the Internet.  These nodes may
 connect to many different access points on the Internet over time,
 and they expect this to be largely transparent to their activities.
 Except when they are not connected to the Internet at all, and for
 performance differences when they are connected, they expect that
 everything will "just work" regardless of their current Internet
 attachment point or local subnetwork technology.
 Changing a host's Internet attachment point involves one or more of
 the following steps.
 First, if use of the local subnetwork is restricted, the user's
 credentials must be verified and access granted.  There are many ways
 to do this.  A trivial example would be an "Internet cafe" that
 grants physical access to the subnetwork for a fee.  Subnetworks may
 implement technical access controls of their own; one example is IEEE
 802.11 Wireless Equivalent Privacy [IEEE80211].  It is common
 practice for both cellular telephone and Internet service providers
 (ISPs) to agree to serve one anothers' users; RADIUS [RFC2865] is the
 standard method for ISPs to exchange authorization information.
 Second, the host may have to be reconfigured with IP parameters
 appropriate for the local subnetwork.  This usually includes setting
 an IP address, default router, and domain name system (DNS) servers.

Karn, et al. Best Current Practice [Page 37] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 On multiple-access networks, the Dynamic Host Configuration Protocol
 (DHCP) [RFC2131] is almost universally used for this purpose.  On PPP
 links, these functions are performed by the IP Control Protocol
 (IPCP) [RFC1332].
 Third, traffic destined for the mobile host must be routed to its
 current location.  This roaming function is the most common meaning
 of the term "Internet mobility".
 Internet mobility can be provided at any of several layers in the
 Internet protocol stack, and there is ongoing debate as to which is
 the most appropriate and efficient.  Mobility is already a feature of
 certain application layer protocols; the Post Office Protocol (POP)
 [RFC1939] and the Internet Message Access Protocol (IMAP) [RFC3501]
 were created specifically to provide mobility in the receipt of
 electronic mail.
 Mobility can also be provided at the IP layer [RFC3344].  This
 mechanism provides greater transparency, viz., IP addresses that
 remain fixed as the nodes move, but at the cost of potentially
 significant network overhead and increased delay because of the sub-
 optimal network routing and tunneling involved.
 Some subnetworks may provide internal mobility, transparent to IP, as
 a feature of their own internal routing mechanisms.  To the extent
 that these simplify routing at the IP layer, reduce the need for
 mechanisms like Mobile IP, or exploit mechanisms unique to the
 subnetwork, this is generally desirable.  This is especially true
 when the subnetwork covers a relatively small geographic area and the
 users move rapidly between the attachment points within that area.
 Examples of internal mobility schemes include Ethernet switching and
 intra-system handoff in cellular telephony.
 However, if the subnetwork is physically large and connects to other
 parts of the Internet at multiple geographic points, care should be
 taken to optimize the wide-area routing of packets between nodes on
 the external Internet and nodes on the subnet.  This is generally
 done with "nearest exit" routing strategies.  Because a given
 subnetwork may be unaware of the actual physical location of a
 destination on another subnetwork, it simply routes packets bound for
 the other subnetwork to the nearest router between the two.  This
 implies some awareness of IP addressing and routing within the
 subnetwork.  The subnetwork may wish to use IP routing internally for
 wide area routing and restrict subnetwork-specific routing to
 constrained geographic areas where the effects of suboptimal routing
 are minimized.

Karn, et al. Best Current Practice [Page 38] RFC 3819 Advice for Internet Subnetwork Designers July 2004

17. Routing

 Subnetworks connecting more than two systems must provide their own
 internal Layer-2 forwarding mechanisms, either implicitly (e.g.,
 broadcast) or explicitly (e.g., switched).  Since routing is the
 major function of the Internet layer, the question naturally arises
 as to the interaction between routing at the Internet layer and
 routing in the subnet, and proper division of function between the
 Layer-2 subnetworks can be point-to-point, connecting two systems, or
 multipoint.  Multipoint subnetworks can be broadcast (e.g., shared
 media or emulated) or non-broadcast.  Generally, IP considers
 multipoint subnetworks as broadcast, with shared-medium Ethernet as
 the canonical (and historical) example, and point-to-point
 subnetworks as a degenerate case.  Non-broadcast subnetworks may
 require additional mechanisms, e.g., above IP at the routing layer
 IP is ignorant of the topology of the subnetwork layer.  In
 particular, reconfiguration of subnetwork paths is not tracked by the
 IP layer.  IP is only affected by whether it can send/receive packets
 sent to the remotely connected systems via the subnetwork interface
 (i.e., the reachability from one router to another).  IP further
 considers that subnetworks are largely static -- that both their
 membership and existence are stable at routing timescales (tens of
 seconds); changes to these are considered re-provisioning, rather
 than routing.
 Routing functionality in a subnetwork is related to addressing in
 that subnetwork.  Resolution of addresses on subnetwork links is
 required for forwarding IP packets across links (e.g., ARP for IPv4,
 or ND for IPv6).  There is unlikely to be direct interaction between
 subnetwork routing and IP routing.  Where broadcast is provided or
 explicitly emulated, address resolution can be used directly; where
 not provided, the link layer routing may interface to a protocol for
 resolution, e.g., to the Next-Hop Resolution Protocol [RFC2322] to
 provide context-dependent address resolution capabilities.
 Subnetwork routing can either complement or compete with IP routing.
 It complements IP when a subnetwork encapsulates its internal
 routing, and where the effects of that routing are not visible at the
 IP layer.  However, if different paths in the subnetwork have
 characteristics that affect IP routing, it can affect or even inhibit
 the convergence of IP routing.

Karn, et al. Best Current Practice [Page 39] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Routing protocols generally consider Layer-2 subnetworks, i.e., with
 subnet masks and no intermediate IP hops, to have uniform routing
 metrics to all members.  Routing can break when a link's
 characteristics do not match the routing metric, in this case, e.g.,
 when some member pairs have different path characteristics.  Consider
 a virtual Ethernet subnetwork that includes both nearby (sub-
 millisecond latency) and remote (100's of milliseconds away) systems.
 Presenting that group as a single subnetwork means that some routing
 protocols will assume that all pairs have the same delay, and that
 that delay is small.  Because this is not the case, the routing
 tables constructed may be suboptimal or may even fail to converge.
 When a subnetwork is used for transit between a set of routers, it
 conventionally provides the equivalent of a full mesh of point-to-
 point links.  Simplicity of the internal subnet structure can be used
 (e.g., via NHRP [RFC2332]) to reduce the size of address resolution
 tables, but routing exchanges will continue to reflect the full mesh
 they emulate.  In general, subnetworks should not be used as a
 transit among a set of routers where routing protocols would break if
 a full mesh of equivalent point-to-point links were used.
 Some subnetworks have special features that allow the use of more
 effective or responsive routing mechanisms that cannot be implemented
 in IP because of its need for generality.  One example is the self-
 learning bridge algorithm widely used in Ethernet networks.  Learning
 bridges perform Layer-2 subnetwork forwarding, avoiding the need for
 dynamic routing at each subnetwork hop.  Another is the "handoff"
 mechanism in cellular telephone networks, particularly the "soft
 handoff" scheme in IS-95 CDMA.
 Subnetworks that cover large geographic areas or include links of
 widely-varying capabilities should be avoided.  IP routing generally
 considers all multipoint subnets equivalent to a local, shared-medium
 link with uniform metrics between any pair of systems, and ignores
 internal subnetwork topology.  Where a subnetwork diverges from that
 assumption, it is the obligation of subnetwork designers to provide
 compensating mechanisms.  Not doing so can affect the scalability and
 convergence of IP routing, as noted above.
 The subnetwork designer who decides to implement internal routing
 should consider whether a custom routing algorithm is warranted, or
 if an existing Internet routing algorithm or protocol may suffice.
 The designer should consider whether this decision is to reduce the
 address resolution table size (possible, but with additional protocol
 support required), or is trying to reduce routing table complexity.
 The latter may be better achieved by partitioning the subnetwork,
 either physically or logically, and using network-layer protocols to
 support partitioning (e.g., AS's in BGP).  Protocols and routing

Karn, et al. Best Current Practice [Page 40] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 algorithms can be notoriously subtle, complex, and difficult to
 implement correctly.  Much work can be avoided if existing protocols
 or implementations can be readily reused.

18. Security Considerations

 Security has become a high priority in the design and operation of
 the Internet.  The Internet is vast, and countless organizations and
 individuals own and operate its various components.  A consensus has
 emerged for what might be called a "security placement principle": a
 security mechanism is most effective when it is placed as close as
 possible to, and under the direct control of the owner of the asset
 that it protects.
 A corollary of this principle is that end-to-end security (e.g.,
 confidentiality, authentication, integrity, and access control)
 cannot be ensured with subnetwork security mechanisms.  Not only are
 end-to-end security mechanisms much more closely associated with the
 end-user assets they protect, they are also much more comprehensive.
 For example, end-to-end security mechanisms cover gaps that can
 appear when otherwise good subnetwork mechanisms are concatenated.
 This is an important application of the end-to-end principle [SRC81].
 Several security mechanisms that can be used end-to-end have already
 been deployed in the Internet and are enjoying increasing use.  The
 most important are the Secure Sockets Layer (SSL) [SSL2] [SSL3] and
 TLS [RFC2246] primarily used to protect web commerce, Pretty Good
 Privacy (PGP) [RFC1991] and S/MIME [RFCs-2630-2634], primarily used
 to protect and authenticate email and software distributions, the
 Secure Shell (SSH), used for secure remote access and file transfer,
 and IPsec [RFC2401], a general purpose encryption and authentication
 mechanism that sits just above IP and can be used by any IP
 application.  (IPsec can actually be used either on an end-to-end
 basis or between security gateways that do not include either or both
 end systems.)
 Nonetheless, end-to-end security mechanisms are not used as widely as
 might be desired.  However, the group could not reach consensus on
 whether subnetwork designers should be actively encouraged to
 implement mechanisms to protect user data.
 The clear consensus of the working group held that subnetwork
 security mechanisms, especially when weak or incorrectly implemented
 [BGW01], may actually be counterproductive.  The argument is that
 subnetwork security mechanisms can lull end users into a false sense
 of security, diminish the incentive to deploy effective end-to-end

Karn, et al. Best Current Practice [Page 41] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 mechanisms, and encourage "risky" uses of the Internet that would not
 be made if users understood the inherent limits of subnetwork
 security mechanisms.
 The other point of view encourages subnetwork security on the
 principle that it is better than the default situation, which all too
 often is no security at all.  Users of especially vulnerable subnets
 (such as consumers who have wireless home networks and/or shared
 media Internet access) often have control over at most one endpoint
 -- usually a client -- and therefore cannot enforce the use of end-
 to-end mechanisms.  However, subnet security can be entirely adequate
 for protecting low-valued assets against the most likely threats.  In
 any event, subnet mechanisms do not preclude the use of end-to-end
 mechanisms, which are typically used to protect highly-valued assets.
 This viewpoint recognizes that many security policies implicitly
 assume that the entire end-to-end path is composed of a series of
 concatenated links that are nominally physically secured.  That is,
 these policies assume that all endpoints of all links are trusted,
 and that access to the physical medium by attackers is difficult.  To
 meet the assumptions of such policies, explicit mechanisms are needed
 for links (especially shared medium links) that lack physical
 protection.  This, for example, is the rationale that underlies Wired
 Equivalent Privacy (WEP) in the IEEE 802.11 [IEEE80211] wireless LAN
 standard, and the Baseline Privacy Interface in the DOCSIS [DOCSIS1]
 [DOCSIS2] data over cable television networks standards.
 We therefore recommend that subnetwork designers who choose to
 implement security mechanisms to protect user data be as candid as
 possible with the details of such security mechanisms and the
 inherent limits of even the most secure mechanisms when implemented
 in a subnetwork rather than on an end-to-end basis.
 In keeping with the "placement principle", a clear consensus exists
 for another subnetwork security role: the protection of the
 subnetwork itself.  Possible threats to subnetwork assets include
 theft of service and denial of service; shared media subnets tend to
 be especially vulnerable to such attacks.  In some cases, mechanisms
 that protect subnet assets can also improve (but cannot ensure) end-
 to-end security.
 One security service can be provided by the subnetwork that will aid
 in the solution of an overall Internet problem: subnetwork security
 should provide a mechanism to authenticate the source of a subnetwork
 frame.  This function is missing in some current protocols, e.g., the
 use of ARP [RFC826] to associate an IPv4 address with a MAC address.
 The IPv6 Neighbor Discovery (ND) [RFC2461] performs a similar

Karn, et al. Best Current Practice [Page 42] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 There are well-known security flaws with this address resolution
 mechanism [Wilbur89].  However, the inclusion of subnetwork frame
 source authentication will permit a secure subnetwork address.
 Another potential role for subnetwork security is to protect users
 against traffic analysis, i.e., identifying the communicating parties
 and determining their communication patterns and volumes even when
 their actual contents are protected by strong end-to-end security
 mechanisms.  Lower-layer security can be more effective against
 traffic analysis due to its inherent ability to aggregate the
 communications of multiple parties sharing the same physical
 facilities while obscuring higher-layer protocol information that
 indicates specific end points, such as IP addresses and TCP/UDP port
 However, traffic analysis is a notoriously subtle and difficult
 threat to understand and defeat, far more so than threats to
 confidentiality and integrity.  We therefore urge extreme care in the
 design of subnetwork security mechanisms specifically intended to
 thwart traffic analysis.
 Subnetwork designers must keep in mind that design and implementation
 for security is difficult [Schneier00].  [Schneier95] describes
 protocols and algorithms which are considered well-understood and
 believed to be sound.
 Poor design process, subtle design errors and flawed implementation
 can result in gaping vulnerabilities.  In recent years, a number of
 subnet standards have had problems exposed.  The following are
 examples of mistakes that have been made:
 1.  Use of weak and untested algorithms [Crypto9912] [BGW01].  For a
     variety of reasons, algorithms were chosen which had subtle
     flaws, making them vulnerable to a variety of attacks.
 2.  Use of "security by obscurity" [Schneier00] [Crypto9912].  One
     common mistake is to assume that keeping cryptographic algorithms
     secret makes them more secure.  This is intuitive, but wrong.
     Full public disclosure early in the design process attracts peer
     review by knowledgeable cryptographers.  Exposure of flaws by
     this review far outweighs any imagined benefit from forcing
     attackers to reverse engineer security algorithms.
 3.  Inclusion of trapdoors [Schneier00] [Crypto9912].  Trapdoors are
     flaws surreptitiously left in an algorithm to allow it to be
     broken.  This might be done to recover lost keys or to permit
     surreptitious access by governmental agencies.  Trapdoors can be
     discovered and exploited by malicious attackers.

Karn, et al. Best Current Practice [Page 43] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 4.  Sending passwords or other identifying information as clear text.
     For many years, analog cellular telephones could be cloned and
     used to steal service.  The cloners merely eavesdropped on the
     registration protocols that exchanged everything in clear text.
 5.  Keys which are common to all systems on a subnet [BGW01].
 6.  Incorrect use of a sound mechanism.  For example [BGW01], one
     subnet standard includes an initialization vector which is poorly
     designed and poorly specified.  A determined attacker can easily
     recover multiple ciphertexts encrypted with the same key stream
     and perform statistical attacks to decipher them.
 7.  Identifying information sent in clear text that can be resolved
     to an individual, identifiable device.  This creates a
     vulnerability to attacks targeted to that device (or its owner).
 8.  Inability to renew and revoke shared secret information.
 9.  Insufficient key length.
 10. Failure to address "man-in-the-middle" attacks, e.g., with mutual
 11. Failure to provide a form of replay detection, e.g., to prevent a
     receiver from accepting packets from an attacker that simply
     resends previously captured network traffic.
 12. Failure to provide integrity mechanisms when providing
     confidentiality schemes [Bel98].
 This list is by no means comprehensive.  Design problems are
 difficult to avoid, but expert review is generally invaluable in
 avoiding problems.
 In addition, well-designed security protocols can be compromised by
 implementation defects.  Examples of such defects include use of
 predictable pseudo-random numbers [RFC1750], vulnerability to buffer
 overflow attacks due to unsafe use of certain I/O system calls
 [WFBA2000], and inadvertent exposure of secret data.

19. Contributors

 This document represents a consensus of the members of the IETF
 Performance Implications of Link Characteristics (PILC) working

Karn, et al. Best Current Practice [Page 44] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 This document would not have been possible without the contributions
 of a great number of people in the Performance Implications of Link
 Characteristics Working Group.  In particular, the following people
 provided major contributions of text, editing, and advice on this
 document: Mark Allman provided the final editing to complete this
 document.  Carsten Bormann provided text on robust header
 compression.  Gorry Fairhurst provided text on broadcast and
 multicast issues, routing,  and many valuable comments on the entire
 document.  Aaron Falk provided text on bandwidth on demand.  Dan
 Grossman provided text on many facets of the document.  Reiner Ludwig
 provided thorough document review and text on TCP vs. Link-Layer
 Retransmission.  Jamshid Mahdavi provided text on TCP performance
 calculations.  Saverio Mascolo provided feedback on the document.
 Gabriel Montenegro provided feedback on the document.  Marie-Jose
 Montpetit provided text on bandwidth on demand.  Joe Touch provided
 text on multicast, broadcast, and routing, and Lloyd Wood provided
 many valuable comments on versions of the document.

20. Informative References

 References of the form RFCnnnn are Internet Request for Comments
 (RFC) documents available online at
 [802.1D]      Information Technology Telecommunications and
               information exchange between systems Local and
               metropolitan area networks, Common specifications Media
               access control (MAC) bridges, IEEE 802.1D, 1998.  ISO
 [802.1p]      IEEE, 802.1p, Standard for Local and Metropolitan Area
               Networks - Supplement to Media Access Control (MAC)
               Bridges: Traffic Class Expediting and Multicast.
 [AP99]        Allman, M. and V. Paxson, On Estimating End-to-End
               Network Path Properties, In Proceedings of ACM SIGCOMM
 [AR02]        Acar, G. and C. Rosenberg, Weighted Fair Bandwidth-on-
               Demand (WFBoD) for Geo-Stationary Satellite Networks
               with On-Board Processing, Computer Networks, 39(1),
 [ATMFTM]      The ATM Forum, "Traffic Management Specification,
               Version 4.0", April 1996, document af-tm-0056.000.

Karn, et al. Best Current Practice [Page 45] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [BA02]        Blanton, E. and M. Allman, On Making TCP More Robust to
               Packet Reordering. ACM Computer Communication Review,
               32(1), January 2002.
 [Bel98]       Bellovin, S., "Cryptography and the Internet", in
               Proceedings of CRYPTO '98, August 1998.
 [BGW01]       Borisov, N., Goldberg, I. and D. Wagner, "Intercepting
               Mobile Communications: The Insecurity of 802.11," In
               Proceedings of ACM MobiCom, July 2001.
 [BPK98]       Balakrishnan, H., Padmanabhan, V. and R. Katz.  "The
               Effects of Asymmetry on TCP Performance."  ACM Mobile
               Networks and Applications (MONET), 1998.
 [BPS99]       Bennet,, J.C.R., Partridge, C. and N. Shectman, "Packet
               Reordering is Not Pathological Network Behavior",
               IEEE/ACM Transactions on Networking, Vol. 7, No. 6,
               December 1999.
 [CGMP]        Farinacci D., Tweedly A. and T. Speakman, "Cisco Group
               Management Protocol (CGMP)", 1996/1997.
 [Crypto9912]  Schneier, B., "European Cellular Encryption Algorithms"
               Crypto-Gram, December 15, 1999.
 [DIX82]       Digital Equipment Corp, Intel Corp, Xerox Corp,
               Ethernet Local Area Network Specification Version 2.0,
               November 1982.
 [DOCSIS1]     Data-Over-Cable Service Interface Specifications, Radio
               Frequency Interface Specification 1.0, SP-RFI-I05-
               991105, November 1999, Cable Television Laboratories,
 [DOCSIS2]     Data-Over-Cable Service Interface Specifications, Radio
               Frequency Interface Specification 1.1, SP-RFIv1.1-I05-
               000714, July 2000, Cable Television Laboratories, Inc.
 [DOCSIS3]     Lai, W.S., "DOCSIS-Based Cable Networks: Impact of
               Large Data Packets on Upstream Capacity", 14th ITC
               Specialists Seminar on Access Networks and Systems,
               Barcelona, Spain, April 25-27, 2001.

Karn, et al. Best Current Practice [Page 46] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [EN301192]    ETSI, European Broadcasting Union, Digital Video
               Broadcasting (DVB); DVB Specification for Data
               Broadcasting, European Standard (Telecommunications
               Series)  EN 301 192 v1.2.1(1999-06).
 [ES00]        Eckhardt, D. and P. Steenkiste, "Effort-limited Fair
               (ELF) Scheduling for Wireless Networks, Proceedings of
               IEEE Infocom 2000.
 [FB00]        Firoiu V. and M. Borden, "A Study of Active Queue
               Management for Congestion Control" to appear in Infocom
 [GM02]        Grieco1, L. and S. Mascolo, "TCP Westwood and Easy RED
               to Improve Fairness in High-Speed Networks",
               Proceedings of the 7th International Workshop on
               Protocols for High-Speed Networks, April 2002.
 [IEEE8023]    IEEE 802.3 CSMA/CD Access Method.
 [IEEE80211]   IEEE 802.11 Wireless LAN standard.
 [ISO3309]     ISO/IEC 3309:1991(E), "Information Technology -
               Telecommunications and information exchange between
               systems - High-level data link control (HDLC)
               procedures - Frame structure", International
               Organization For Standardization, Fourth edition 1991-
 [ISO13818]    ISO/IEC, ISO/IEC 13818-1:2000(E)  Information
               Technology - Generic coding of moving pictures and
               associated audio information:  Systems, Second edition,
               2000-12-01 International Organization for
               Standardization and International Electrotechnical
 [ITU-I363]    ITU-T I.363.5 B-ISDN ATM Adaptation Layer Specification
               Type AAL5, International Standards Organisation (ISO),
 [Jac90]       Jacobson, V., Modified TCP Congestion Avoidance
               Algorithm.  Email to the end2end-interest mailing list,
               April 1990.

Karn, et al. Best Current Practice [Page 47] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [KY02]        Khafizov, F. and M. Yavuz, Running TCP Over IS-2000,
               Proceedings of IEEE ICC, 2002.
 [LK00]        Ludwig, R. and R. H. Katz, "The Eifel Algorithm: Making
               TCP Robust Against Spurious Retransmissions", ACM
               Computer Communication Review, Vol. 30, No. 1, January
 [LKJK02]      Ludwig, R., Konrad, A., Joseph, A. D. and R. H. Katz,
               "Optimizing the End-to-End Performance of Reliable
               Flows over Wireless Links", Kluwer/ACM Wireless
               Networks Journal, Vol. 8, Nos. 2/3, pp. 289-299,
               March-May 2002.
 [LRKOJ99]     Ludwig, R., Rathonyi, B., Konrad, A., Oden, K. and A.
               Joseph, Multi-Layer Tracing of TCP over a Reliable
               Wireless Link, pp. 144-154, In Proceedings of ACM
               SIGMETRICS 99.
 [LS00]        Ludwig, R. and K. Sklower, The Eifel Retransmission
               Timer, ACM Computer Communication Review, Vol. 30, No.
               3, July 2000.
 [MAGMA-PROXY] Fenner, B., He, H., Haberman, B. and H. Sandick,
               "IGMP/MLD-based Multicast Forwarding ("IGMP/MLD
               Proxying")", Work in Progress.
 [MAGMA-SNOOP] Christensen, M., Kimball, K. and F. Solensky,
               "Considerations for IGMP and MLD Snooping Switches",
               Work in Progress.
 [MBB00]       May, M., Bonald, T. and J-C. Bolot, "Analytic
               Evaluation of RED Performance", INFOCOM 2000.
 [MBDL99]      May, M., Bolot, J., Diot, C. and B. Lyles, "Reasons not
               to deploy RED", Proc. of 7th. International Workshop on
               Quality of Service (IWQoS'99), June 1999.
 [MSMO97]      Mathis, M., Semke, J., Mahdavi, J. and T. Ott, "The
               Macroscopic Behavior of the TCP Congestion Avoidance
               Algorithm", Computer Communication Review, Vol. 27,
               number 3, July 1997.
 [MYR95]       Boden, N., Cohen, D., Felderman, R., Kulawik, A.,
               Seitz, C., et al.  MYRINET: A Gigabit per Second Local
               Area Network, IEEE-Micro, Vol. 15, No.1, February 1995,
               pp. 29-36.

Karn, et al. Best Current Practice [Page 48] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [PFTK98]      Padhye, J., Firoiu, V., Towsley, D. and J. Kurose,
               "Modeling TCP Throughput: a Simple Model and its
               Empirical Validation", UMASS CMPSCI Tech Report TR98-
               008, Feb. 1998.
 [RED93]       Floyd, S. and V. Jacobson, "Random Early Detection
               gateways for Congestion Avoidance", IEEE/ACM
               Transactions in Networking, Vol. 1 No. 4, August 1993.
 [RF95]        Romanow, A. and S. Floyd, "Dynamics of TCP Traffic over
               ATM Networks".  IEEE Journal of Selected Areas in
               Communication, Vol.13 No.  4, May 1995, p. 633-641.
 [RFC791]      Postel, J., "Internet Protocol", STD 5, RFC 791,
               September 1981.
 [RFC793]      Postel, J., "Transmission Control Protocol", STD 7, RFC
               793, September 1981.
 [RFC768]      Postel, J., "User Datagram Protocol", STD 6, RFC 768,
               August 1980.
 [RFC826]      Plummer, D.C., "Ethernet Address Resolution Protocol:
               Or converting network protocol addresses to 48-bit
               Ethernet address for transmission on Ethernet
               hardware", STD 37, RFC 826, November 1982.
 [RFC1071]     Braden, R., Borman, D. and C. Partridge, "Computing the
               Internet checksum", RFC 1071, September 1988.
 [RFC1112]     Deering, S., "Host Extensions for IP Multicasting", STD
               5, RFC 1112, August 1989.
 [RFC1144]     Jacobson, V., "Compressing TCP/IP Headers for Low-Speed
               Serial Links", RFC 1144, February 1990.
 [RFC1191]     Mogul, J. and S. Deering, "Path MTU Discovery", RFC
               1191, November 1990.
 [RFC1332]     McGregor, C., "The PPP Internet Protocol Control
               Protocol (IPCP)", RFC 1332, May 1992.
 [RFC1435]     Knowles, S., "IESG Advice from Experience with Path MTU
               Discovery", RFC 1435, March 1993.

Karn, et al. Best Current Practice [Page 49] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [RFC1633]     Braden, R., Clark, D. and S. Shenker, "Integrated
               Services in the Internet Architecture: an Overview",
               RFC 1633, June 1994.
 [RFC1661]     Simpson, W., "The Point-to-Point Protocol (PPP)", STD
               51, RFC 1661, July 1994.
 [RFC1662]     Simpson, W., Ed., "PPP in HDLC-like Framing", STD 51,
               RFC 1662, July 1994.
 [RFC1750]     Eastlake 3rd, D., Crocker, S. and J. Schiller,
               "Randomness Recommendations for Security", RFC 1750,
               December 1994.
 [RFC1812]     Baker, F., Ed., "Requirements for IP Version 4
               Routers", RFC 1812, June 1995.
 [RFC1939]     Myers, J. and M. Rose, "Post Office Protocol - Version
               3", STD 53, RFC 1939, May 1996.
 [RFC1981]     McCann, J., Deering, S. and J. Mogul, "Path MTU
               Discovery for IP version 6", RFC 1981, August 1996.
 [RFC1991]     Atkins, D., Stallings, W. and P. Zimmermann, "PGP
               Message Exchange Formats", RFC 1991, August 1996.
 [RFC2018]     Mathis, M., Mahdavi, J., Floyd, S. and A. Romanow, "TCP
               Selective Acknowledgement Options", RFC 2018, October
 [RFC2131]     Droms, R., "Dynamic Host Configuration Protocol", RFC
               2131, March 1997.
 [RFC2205]     Braden, R., Ed., Zhang, L., Berson, S., Herzog, S. and
               S. Jamin, "Resource ReSerVation Protocol (RSVP) --
               Version 1 Functional Specification", RFC 2205,
               September 1997.
 [RFC2208]     Mankin, A., Baker, F., Braden, B., Bradner, S., O`Dell,
               M., Romanow, A., Weinrib, A. and L. Zhang, "Resource
               ReSerVation Protocol (RSVP) -- Version 1 Applicability
               Statement Some Guidelines on Deployment", RFC 2208,
               September 1997.
 [RFC2210]     Wroclawski, J., "The Use of RSVP with IETF Integrated
               Services", RFC 2210, September 1997.

Karn, et al. Best Current Practice [Page 50] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [RFC2211]     Wroclawski, J., "Specification of the Controlled-Load
               Network Element Service", RFC 2211, September 1997.
 [RFC2212]     Shenker, S., Partridge, C. and R. Guerin,
               "Specification of Guaranteed Quality of Service", RFC
               2212, September 1997.
 [RFC2246]     Dierks, T. and C. Allen, "The TLS Protocol Version
               1.0", RFC 2246, January 1999.
 [RFC2309]     Braden, B., Clark, D., Crowcroft, J., Davie, B.,
               Deering, S., Estrin, D., Floyd, S., Jacobson, V.,
               Minshall, G., Partridge, C., Peterson, L.,
               Ramakrishnan, K., Shenker, S., Wroclawski, J. and L.
               Zhang, "Recommendations on Queue Management and
               Congestion Avoidance in the Internet", RFC 2309, April
 [RFC2322]     van den Hout, K., Koopal, A. and R. van Mook,
               "Management of IP numbers by peg-dhcp", RFC 2322, 1
               April 1998.
 [RFC2328]     Moy, J., "OSPF Version 2", STD 54, RFC 2328, April
 [RFC2332]     Luciani, J., Katz, D., Piscitello, D., Cole, B. and N.
               Doraswamy, "NBMA Next Hop Resolution Protocol (NHRP)",
               RFC 2332, April 1998.
 [RFC2364]     Gross, G., Kaycee, M., Li, A., Malis, A. and J.
               Stephens, "PPP Over AAL5", RFC 2364, July 1998.
 [RFC2394]     Pereira, R., "IP Payload Compression Using DEFLATE",
               RFC 2394, December 1998.
 [RFC2395]     Friend, R. and R. Monsour, "IP Payload Compression
               Using LZS", RFC 2395, December 1998.
 [RFC2401]     Kent, S. and R. Atkinson, "Security Architecture for
               the Internet Protocol", RFC 2401, November 1998.
 [RFC2406]     Kent, S. and R. Atkinson, "IP Encapsulating Security
               Payload (ESP)", RFC 2406, November 1998.
 [RFC2440]     Callas, J., Donnerhacke, L., Finney, H. and R. Thayer,
               "OpenPGP Message Format", RFC 2440, November 1998.

Karn, et al. Best Current Practice [Page 51] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [RFC2460]     Deering, S. and R. Hinden, "Internet Protocol, Version
               6 (IPv6) Specification", RFC 2460, December 1998.
 [RFC2461]     Narten, T., Nordmark, E. and W. Simpson, "Neighbor
               Discovery for IP Version 6 (IPv6)", RFC 2461, December
 [RFC2474]     Nichols, K., Blake, S., Baker, F. and D. Black,
               "Definition of the Differentiated Services Field (DS
               Field) in the IPv4 and IPv6 Headers", RFC 2474,
               December 1998.
 [RFC2475]     Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.
               and W. Weiss, "An Architecture for Differentiated
               Services", RFC 2475, December 1998.
 [RFC2507]     Degermark, M., Nordgren, B. and S. Pink, "IP Header
               Compression", RFC 2507, February 1999.
 [RFC2508]     Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP
               Headers for Low-Speed Serial Links", RFC 2508, February
 [RFC2581]     Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
               Control", RFC 2581, April 1999.
 [RFC2582]     Floyd, S. and T. Henderson, "The NewReno Modification
               to TCP's Fast Recovery Algorithm", RFC 2582, April
 [RFC2597]     Heinanen, J., Baker, F., Weiss, W. and J. Wroclawski,
               "Assured Forwarding PHB Group", RFC 2597, June 1999.
 [RFC2616]     Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
               Masinter, L., Leach, P. and T. Berners-Lee, "Hypertext
               Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
 [RFC2630]     Housley, R., "Cryptographic Message Syntax", RFC 2630,
               June 1999.
 [RFC2631]     Rescorla, E., "Diffie-Hellman Key Agreement Method",
               RFC 2631, June 1999.
 [RFC2632]     Ramsdell, B., Ed., "S/MIME Version 3 Certificate
               Handling", RFC 2632, June 1999.

Karn, et al. Best Current Practice [Page 52] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [RFC2633]     Ramsdell, B., "S/MIME Version 3 Message Specification",
               RFC 2633, June 1999.
 [RFC2634]     Hoffman, P., "Enhanced Security Services for S/MIME",
               RFC 2634, June 1999.
 [RFC2684]     Grossman, D. and J. Heinanen, "Multiprotocol
               Encapsulation over ATM Adaptation Layer 5", RFC 2684,
               September 1999.
 [RFC2686]     Bormann, C., "The Multi-Class Extension to Multi-Link
               PPP", RFC 2686, September 1999.
 [RFC2687]     Bormann, C., "PPP in a Real-time Oriented HDLC-like
               Framing", RFC 2687, September 1999.
 [RFC2689]     Bormann, C., "Providing Integrated Services over Low-
               bitrate Links", RFC 2689, September 1999.
 [RFC2710]     Deering, S., Fenner, W. and B. Haberman, "Multicast
               Listener Discovery (MLD) for IPv6", RFC 2710, October
 [RFC2784]     Farinacci, D., Li, T., Hanks, S., Meyer, D. and P.
               Traina, "Generic Routing Encapsulation (GRE)", RFC
               2784, March 2000.
 [RFC2865]     Rigney, C., Willens, S., Rubens, A. and W. Simpson,
               "Remote Authentication Dial In User Service (RADIUS)",
               RFC 2865, June 2000.
 [RFC2914]     Floyd, S., "Congestion Control Principles", BCP 41, RFC
               2914, September 2000.
 [RFC2923]     Lahey, K., "TCP Problems with Path MTU Discovery", RFC
               2923, September 2000.
 [RFC2988]     Paxson, V. and M. Allman, "Computing TCP's
               Retransmission Timer", RFC 2988, November 2000.
 [RFC2990]     Huston, G., "Next Steps for the IP QoS Architecture",
               RFC 2990, November 2000.
 [RFC3048]     Whetten, B., Vicisano, L., Kermode, R., Handley, M.,
               Floyd, S. and M. Luby, "Reliable Multicast Transport
               Building Blocks for One-to-Many Bulk-Data Transfer",
               RFC 3048, January 2001.

Karn, et al. Best Current Practice [Page 53] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [RFC3095]     Bormann, C., Ed., Burmeister, C., Degermark, M.,
               Fukushima, H., Hannu, H., Jonsson, L-E., Hakenberg, R.,
               Koren, T., Le, K., Liu, Z., Martensson, A., Miyazaki,
               A., Svanbro, K., Wiebke, T., Yoshimura, T. and H.
               Zheng, "RObust Header Compression (ROHC):  Framework
               and four profiles: RTP, UDP, ESP, and uncompressed",
               RFC 3095, July 2001.
 [RFC3096]     Degermark, M., Ed., "Requirements for robust IP/UDP/RTP
               header compression", RFC 3096, July 2001.
 [RFC3150]     Dawkins, S., Montenegro, G., Kojo, M. and V. Magret,
               "End-to-end Performance Implications of Slow Links",
               BCP 48, RFC 3150, July 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.
 [RFC3168]     Ramakrishnan, K., Floyd, S. and D. Black, "The Addition
               of Explicit Congestion Notification (ECN) to IP", RFC
               3168, September 2001.
 [RFC3173]     Shacham, A., Monsour, B., Pereira, R. and M. Thomas,
               "IP Payload Compression Protocol (IPComp)", RFC 3173,
               September 2001.
 [RFC3246]     Davie, B., Charny, A., Bennet, J.C.R., Benson, K., Le
               Boudec, J.Y., Courtney, W., Davari, S., Firoiu, V. and
               D. Stiliadis, "An Expedited Forwarding PHB (Per-Hop
               Behavior)", RFC 3246, March 2002.
 [RFC3248]     Armitage, G., Carpenter, B., Casati, A., Crowcroft, J.,
               Halpern, J., Kumar, B. and J. Schnizlein, "A Delay
               Bound alternative revision of RFC 2598", RFC 3248,
               March 2002.
 [RFC3344]     Perkins, C., Ed., "IP Mobility Support for IPv4", RFC
               3344, August 2002.
 [RFC3366]     Fairhurst, G. and L. Wood, "Advice to link designers on
               link Automatic Repeat reQuest (ARQ)", BCP 62, RFC 3366,
               August 2002.

Karn, et al. Best Current Practice [Page 54] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [RFC3376]     Cain, B., Deering, S., Kouvelas, I., Fenner, B. and A.
               Thyagarajan, "Internet Group Management Protocol,
               Version 3", RFC 3376, October 2002.
 [RFC3449]     Balakrishnan, H., Padmanabhan, V., Fairhurst, G. and M.
               Sooriyabandara, "TCP Performance Implications of
               Network Path Asymmetry", BCP 69, RFC 3449, December
 [RFC3450]     Luby, M., Gemmell, J., Vicisano, L., Rizzo, L. and J.
               Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
               Instantiation", RFC 3450, December 2002.
 [RFC3451]     Luby, M., Gemmell, J., Vicisano, L., Rizzo, L.,
               Handley, M. and J. Crowcroft, "Layered Coding Transport
               (LCT) Building Block", RFC 3451, December 2002.
 [RFC3452]     Luby, M., Vicisano, L., Gemmell, J., Rizzo, L.,
               Handley, M. and J. Crowcroft, "Forward Error Correction
               (FEC) Building Block", RFC 3452, December 2002.
 [RFC3453]     Luby, M., Vicisano, L., Gemmell, J., Rizzo, L.,
               Handley, M. and J. Crowcroft, "The Use of Forward Error
               Correction (FEC) in Reliable Multicast", RFC 3453,
               December 2002.
 [RFC3488]     Wu, I. and T. Eckert, "Cisco Systems Router-port Group
               Management Protocol (RGMP)", RFC 3488, February 2003.
               VERSION 4rev1", RFC 3501, March 2003.
 [RFC3828]     Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E.,
               Ed. and G. Fairhurst, Ed., "The User Datagram Protocol
               (UDP)-Lite Protocol", RFC 3828, June 2004.
 [Schneier95]  Schneier, B., Applied Cryptography: Protocols,
               Algorithms and Source Code in C (John Wiley and Sons,
               October 1995).
 [Schneier00]  Schneier, B., Secrets and Lies: Digital Security in a
               Networked World (John Wiley and Sons, August 2000).
 [SP2000]      Stone, J. and C. Partridge, "When the CRC and TCP
               Checksum Disagree", ACM SIGCOMM, September 2000.

Karn, et al. Best Current Practice [Page 55] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 [SRC81]       Saltzer, J., Reed D. and D. Clark, "End-to-End
               Arguments in System Design".  Second International
               Conference on Distributed Computing Systems (April,
               1981) pages 509-512. Published with minor changes in
               ACM Transactions in Computer Systems 2, 4, November,
               1984, pages 277-288. Reprinted in Craig Partridge,
               editor Innovations in internetworking. Artech House,
               Norwood, MA, 1988, pages 195-206. ISBN 0-89006-337-0.
 [SSL2]        Hickman, K., "The SSL Protocol", Netscape
               Communications Corp., Feb 9, 1995.
 [SSL3]        Frier, A., Karlton, P. and P. Kocher, "The SSL 3.0
               Protocol", Netscape Communications Corp., Nov 18, 1996.
 [TCPF98]      Lin, D. and H.T. Kung, "TCP Fast Recovery Strategies:
               Analysis and Improvements", IEEE Infocom, March 1998.
 [WFBA2000]    Wagner, D., Foster, J., Brewer, E. and A. Aiken, "A
               First Step Toward Automated Detection of Buffer Overrun
               Vulnerabilities", Proceedings of NDSS2000.
 [Wilbur89]    Wilbur, Steve R., Jon Crowcroft, and Yuko Murayama.
               "MAC layer Security Measures in Local Area Networks",
               Local Area Network Security, Workshop LANSEC '89
               Proceedings, Springer-Verlag, April 1989, pp. 53-64.

Karn, et al. Best Current Practice [Page 56] RFC 3819 Advice for Internet Subnetwork Designers July 2004

21. Contributors' Addresses

 Aaron Falk
 USC/Information Sciences Institute
 4676 Admiralty Way
 Marina Del Rey, CA 90292
 Phone: 310-448-9327
 Saverio Mascolo
 Dipartimento di Elettrotecnica ed Elettronica,
 Politecnico di Bari Via Orabona 4, 70125 Bari, Italy
 Phone: +39 080 596 3621
 Marie-Jose Montpetit

Karn, et al. Best Current Practice [Page 57] RFC 3819 Advice for Internet Subnetwork Designers July 2004

22. Authors' Addresses

 Phil Karn, Editor
 Qualcomm 5775 Morehouse Drive
 San Diego CA 92121
 Phone: 858 587 1121
 Carsten Bormann
 Universitaet Bremen TZI
 Postfach 330440
 D-28334 Bremen, Germany
 Phone: +49 421 218 7024
 Fax:   +49 421 218 7000
 Godred (Gorry) Fairhurst
 Department of Engineering, University of Aberdeen,
 Aberdeen, AB24 3UE, United Kingdom
 Dan Grossman
 Motorola, Inc.
 111 Locke Drive
 Marlboro, MA 01752
 Reiner Ludwig
 Ericsson Research
 Ericsson Allee
 1 52134 Herzogenrath, Germany
 Phone: +49 2407 575 719

Karn, et al. Best Current Practice [Page 58] RFC 3819 Advice for Internet Subnetwork Designers July 2004

 Jamshid Mahdavi
 Novell, Inc.
 Gabriel Montenegro
 Sun Microsystems Laboratories, Europe
 180, Avenue de l'Europe
 38334 Saint Ismier CEDEX
 Joe Touch
 USC/Information Sciences Institute
 4676 Admiralty Way
 Marina del Rey CA 90292
 Phone: 310 448 9151
 Lloyd Wood
 Cisco Systems
 9 New Square Park, Bedfont Lakes
 Feltham TW14 8HA
 United Kingdom
 Phone: +44 (0)20 8824 4236

Karn, et al. Best Current Practice [Page 59] RFC 3819 Advice for Internet Subnetwork Designers July 2004

23. Full Copyright Statement

 Copyright (C) The Internet Society (2004).  This document is subject
 to the rights, licenses and restrictions contained in BCP 78, and
 except as set forth therein, the authors retain all their rights.
 This document and the information contained herein are provided on an

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 under such rights might or might not be available; nor does it
 represent that it has made any independent effort to identify any
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 rights in RFC documents can be found in BCP 78 and BCP 79.
 Copies of IPR disclosures made to the IETF Secretariat and any
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 The IETF invites any interested party to bring to its attention
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 to implement this standard.  Please address the information to the
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 Funding for the RFC Editor function is currently provided by the
 Internet Society.

Karn, et al. Best Current Practice [Page 60]

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