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Network Working Group                                       P. Karn, Ed.Request for Comments: 3819                                      QualcommBCP: 89                                                       C. BormannCategory: Best Current Practice                  Universitaet Bremen TZI                                                            G. Fairhurst                                                  University of Aberdeen                                                             D. Grossman                                                          Motorola, Inc.                                                               R. Ludwig                                                       Ericsson Research                                                              J. Mahdavi                                                                  Novell                                                           G. Montenegro                                   Sun Microsystems Laboratories, Europe                                                                J. Touch                                                                 USC/ISI                                                                 L. Wood                                                           Cisco Systems                                                               July 2004Advice for Internet Subnetwork DesignersStatus 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).Abstract   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 2004Table of Contents1.  Introduction and Overview. . . . . . . . . . . . . . . . . . .22.  Maximum Transmission Units (MTUs) and IP Fragmentation . . . .42.1.  Choosing the MTU in Slow Networks. . . . . . . . . . . .63.  Framing on Connection-Oriented Subnetworks . . . . . . . . . .74.  Connection-Oriented Subnetworks. . . . . . . . . . . . . . . .95.  Broadcasting and Discovery . . . . . . . . . . . . . . . . . .106.  Multicasting . . . . . . . . . . . . . . . . . . . . . . . . .117.  Bandwidth on Demand (BoD) Subnets. . . . . . . . . . . . . . .138.  Reliability and Error Control. . . . . . . . . . . . . . . . .148.1.  TCP vs Link-Layer Retransmission . . . . . . . . . . . .148.2.  Recovery from Subnetwork Outages . . . . . . . . . . . .178.3.  CRCs, Checksums and Error Detection. . . . . . . . . . .188.4.  How TCP Works. . . . . . . . . . . . . . . . . . . . . .208.5.  TCP Performance Characteristics. . . . . . . . . . . . .228.5.1.  The Formulae . . . . . . . . . . . . . . . . . .228.5.2.  Assumptions. . . . . . . . . . . . . . . . . . .23             8.5.3.  Analysis of Link-Layer Effects on TCP                     Performance. . . . . . . . . . . . . . . . . . .249.  Quality-of-Service (QoS) Considerations. . . . . . . . . . . .2610. Fairness vs Performance. . . . . . . . . . . . . . . . . . . .2911. Delay Characteristics. . . . . . . . . . . . . . . . . . . . .3012. Bandwidth Asymmetries. . . . . . . . . . . . . . . . . . . . .3113. Buffering, Flow and Congestion Control . . . . . . . . . . . .3114. Compression. . . . . . . . . . . . . . . . . . . . . . . . . .3415. Packet Reordering. . . . . . . . . . . . . . . . . . . . . . .3616. Mobility . . . . . . . . . . . . . . . . . . . . . . . . . . .3717. Routing. . . . . . . . . . . . . . . . . . . . . . . . . . . .3918. Security Considerations. . . . . . . . . . . . . . . . . . . .4119. Contributors . . . . . . . . . . . . . . . . . . . . . . . . .4420. Informative References . . . . . . . . . . . . . . . . . . . .4521. Contributors' Addresses. . . . . . . . . . . . . . . . . . . .5722. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . .5823. Full Copyright Statement . . . . . . . . . . . . . . . . . . .601.  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   service.   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   possible.   Optimizing a subnetwork for IP involves three complementary   considerations:   1.  Providing functionality sufficient to carry IP.   2.  Eliminating unnecessary functions that increase cost or       complexity.   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   support.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 lengthKarn, 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   necessary.   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   decrease.   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       and   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   boundaries.   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 implicitlyKarn, 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 thisKarn, 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   times.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   packet.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   channel.   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   data).   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].  BoDKarn, 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   prematurely.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, FECKarn, 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   layer.   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 (seeSection 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 itsKarn, 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   simulation.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 CRCKarn, 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 arbitraryKarn, 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   practiced.   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 theKarn, 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 20048.5.  TCP Performance Characteristics   Caveat   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:                                         MSS           BW = --------------------------------------------------------                RTT*sqrt(1.33*p) + RTO*p*[1+32*p^2]*min[1,3*sqrt(.75*p)]   where           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.   [MSMO97].                     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   formula).   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 20048.5.3.  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   disappointed.   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 theKarn, 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)   service.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       priorities.   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       subnets.   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       flows.   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   [ES00].Karn, et al.             Best Current Practice                 [Page 29]

RFC 3819        Advice for Internet Subnetwork Designers       July 200411.  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   [LS00].   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   compute.)   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, FECKarn, 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 BitKarn, 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   deployed.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 congestionKarn, 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   suffers.   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.Karn, et al.             Best Current Practice                 [Page 36]

RFC 3819        Advice for Internet Subnetwork Designers       July 2004   [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 200417.  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   two.   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   [RFC2328].   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 routingKarn, 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-endKarn, 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   function.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   numbers.   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       authentication.   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   group.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 www.rfc-editor.org.   [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                 15802-3.   [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                 99.   [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),                 2002.   [ATMFTM]      The ATM Forum, "Traffic Management Specification,                 Version 4.0", April 1996, document af-tm-0056.000.http://www.atmforum.com/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.http://www.research.att.com/~smb/papers/inet-crypto.pdf   [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.ftp://ftpeng.cisco.com/ipmulticast/specs/cgmp.txt   [Crypto9912]  Schneier, B., "European Cellular Encryption Algorithms"                 Crypto-Gram, December 15, 1999.http://www.counterpane.com   [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,                 Inc.   [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                 2000.   [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.http://standards.ieee.org/   [IEEE80211]   IEEE 802.11 Wireless LAN standard.http://standards.ieee.org/   [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-                 06-01.   [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                 Commission.   [ITU-I363]    ITU-T I.363.5 B-ISDN ATM Adaptation Layer Specification                 Type AAL5, International Standards Organisation (ISO),                 1996.   [Jac90]       Jacobson, V., Modified TCP Congestion Avoidance                 Algorithm.  Email to the end2end-interest mailing list,                 April 1990.ftp://ftp.ee.lbl.gov/email/vanj.90apr30.txtKarn, 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                 2000.   [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.http://www.aciri.org/floyd/papers/red/red.html   [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,RFC793, 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",RFC1191, 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                 1996.   [RFC2131]     Droms, R., "Dynamic Host Configuration Protocol",RFC2131, 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",RFC2212, 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                 1998.   [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                 1998.   [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                 1998.   [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                 1999.   [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                 1999.   [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                 1999.   [RFC2784]     Farinacci, D., Li, T., Hanks, S., Meyer, D. and P.                 Traina, "Generic Routing Encapsulation (GRE)",RFC2784, 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,RFC2914, September 2000.   [RFC2923]     Lahey, K., "TCP Problems with Path MTU Discovery",RFC2923, 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",RFC3168, 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 ofRFC 2598",RFC 3248,                 March 2002.   [RFC3344]     Perkins, C., Ed., "IP Mobility Support for IPv4",RFC3344, 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                 2002.   [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.   [RFC3501]     Crispin, M., "INTERNET MESSAGE ACCESS PROTOCOL -                 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.http://www.acm.org/sigcomm/sigcomm2000/conf/paper/sigcomm2000-9-1.pdfKarn, 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.http://www.eecs.harvard.edu/networking/papers/infocom-tcp-final-198.pdf   [WFBA2000]    Wagner, D., Foster, J., Brewer, E. and A. Aiken, "A                 First Step Toward Automated Detection of Buffer Overrun                 Vulnerabilities", Proceedings of NDSS2000.http://www.isoc.org/isoc/conferences/ndss/2000/proceedings/039.pdf   [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 200421. Contributors' Addresses   Aaron Falk   USC/Information Sciences Institute   4676 Admiralty Way   Marina Del Rey, CA 90292   Phone: 310-448-9327   EMail: falk@isi.edu   Saverio Mascolo   Dipartimento di Elettrotecnica ed Elettronica,   Politecnico di Bari Via Orabona 4, 70125 Bari, Italy   Phone: +39 080 596 3621   EMail: mascolo@poliba.it   URL:http://www-dee.poliba.it/dee-web/Personale/mascolo.html   Marie-Jose Montpetit   MJMontpetit.com   EMail: marie@mjmontpetit.comKarn, et al.             Best Current Practice                 [Page 57]

RFC 3819        Advice for Internet Subnetwork Designers       July 200422.  Authors' Addresses   Phil Karn, Editor   Qualcomm 5775 Morehouse Drive   San Diego CA 92121   Phone: 858 587 1121   EMail: karn@qualcomm.com   Carsten Bormann   Universitaet Bremen TZI   Postfach 330440   D-28334 Bremen, Germany   Phone: +49 421 218 7024   Fax:   +49 421 218 7000   EMail: cabo@tzi.org   Godred (Gorry) Fairhurst   Department of Engineering, University of Aberdeen,   Aberdeen, AB24 3UE, United Kingdom   EMail: gorry@erg.abdn.ac.uk   URL:http://www.erg.abdn.ac.uk/users/gorry   Dan Grossman   Motorola, Inc.   111 Locke Drive   Marlboro, MA 01752   EMail: Dan.Grossman@motorola.com   Reiner Ludwig   Ericsson Research   Ericsson Allee   1 52134 Herzogenrath, Germany   Phone: +49 2407 575 719   EMail: Reiner.Ludwig@ericsson.comKarn, et al.             Best Current Practice                 [Page 58]

RFC 3819        Advice for Internet Subnetwork Designers       July 2004   Jamshid Mahdavi   Novell, Inc.   EMail: jmahdavi@earthlink.net   Gabriel Montenegro   Sun Microsystems Laboratories, Europe   180, Avenue de l'Europe   38334 Saint Ismier CEDEX   France   EMail: gab@sun.com   Joe Touch   USC/Information Sciences Institute   4676 Admiralty Way   Marina del Rey CA 90292   Phone: 310 448 9151   EMail: touch@isi.edu   URL:http://www.isi.edu/touch   Lloyd Wood   Cisco Systems   9 New Square Park, Bedfont Lakes   Feltham TW14 8HA   United Kingdom   Phone: +44 (0)20 8824 4236   EMail: lwood@cisco.com   URL:http://www.ee.surrey.ac.uk/Personal/L.Wood/Karn, et al.             Best Current Practice                 [Page 59]

RFC 3819        Advice for Internet Subnetwork Designers       July 200423.  Full Copyright Statement   Copyright (C) The Internet Society (2004).  This document is subject   to the rights, licenses and restrictions contained inBCP 78, and   except as set forth therein, the authors retain all their rights.   This document and the information contained herein are provided on an   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE   REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE   INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR   IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF   THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.Intellectual Property   The IETF takes no position regarding the validity or scope of any   Intellectual Property Rights or other rights that might be claimed   to pertain to the implementation or use of the technology   described in this document or the extent to which any license   under such rights might or might not be available; nor does it   represent that it has made any independent effort to identify any   such rights.  Information on the procedures with respect to   rights in RFC documents can be found inBCP 78 andBCP 79.   Copies of IPR disclosures made to the IETF Secretariat and any   assurances of licenses to be made available, or the result of an   attempt made to obtain a general license or permission for the use   of such proprietary rights by implementers or users of this   specification can be obtained from the IETF on-line IPR repository   athttp://www.ietf.org/ipr.   The IETF invites any interested party to bring to its attention   any copyrights, patents or patent applications, or other   proprietary rights that may cover technology that may be required   to implement this standard.  Please address the information to the   IETF at ietf-ipr@ietf.org.Acknowledgement   Funding for the RFC Editor function is currently provided by the   Internet Society.Karn, et al.             Best Current Practice                 [Page 60]