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Transport TCP Maintenance and Minor Extensions (TCPM) WG TCP Roadmap This document contains a roadmap to the Request for Comments (RFC) documents relating to the Internet's Transmission Control Protocol (TCP). This roadmap provides a brief summary of the documents defining TCP and various TCP extensions that have accumulated in the RFC series. This serves as a guide and quick reference for both TCP implementers and other parties who desire information contained in the TCP-related RFCs. This document obsoletes RFC 4614.

A correct and efficient implementation of the Transmission Control Protocol (TCP) is a critical part of the software of most Internet hosts. As TCP has evolved over the years, many distinct documents have become part of the accepted standard for TCP. At the same time, a large number of experimental modifications to TCP have also been published in the RFC series, along with informational notes, case studies, and other advice. As an introduction to newcomers and an attempt to organize the plethora of information for old hands, this document contains a roadmap to the TCP-related RFCs. It provides a brief summary of the RFC documents that define TCP. This should provide guidance to implementers on the relevance and significance of the standards-track extensions, informational notes, and best current practices that relate to TCP. This document is not an update of RFC 1122 and is not a rigorous standard for what needs to be implemented in TCP. This document is merely an informational roadmap that captures, organizes, and summarizes most of the RFC documents that a TCP implementer, experimenter, or student should be aware of. Particular comments or broad categorizations that this document makes about individual mechanisms and behaviors are not to be taken as definitive, nor should the content of this document alone influence implementation decisions. This roadmap includes a brief description of the contents of each TCP-related RFC. In some cases, we simply supply the abstract or a key summary sentence from the text as a terse description. In addition, a letter code after an RFC number indicates its category in the RFC series (see BCP 9 for explanation of these categories): Standards Track (Proposed Standard, Draft Standard, or Internet Standard) Experimental Informational Historic Best Current Practice Unknown (not formally defined) Note that the category of an RFC does not necessarily reflect its current relevance. For instance, RFC 5681 is considered part of the required core functionality of TCP, although the RFC is only a Draft Standard. Similarly, some Informational RFCs contain significant technical proposals for changing TCP. Finally, if an error in the technical content has been found after publication of an RFC (at the time of this writing), this fact is indicated by the term "(Errata)" in the headline of the RFC's description. The contents of the errata can be found through the RFC Errata page . This roadmap is divided into three main sections. lists the RFCs that describe absolutely required TCP behaviors for proper functioning and interoperability. Further RFCs that describe strongly encouraged, but nonessential, behaviors are listed in . Experimental extensions that are not yet standard practices, but that potentially could be in the future, are described in . The reader will probably notice that these three sections are broadly equivalent to MUST/SHOULD/MAY specifications (per RFC 2119 ), and although the authors support this intuition, this document is merely descriptive; it does not represent a binding Standards Track position. Individual implementers still need to examine the Standards Track RFCs themselves to evaluate specific requirement levels. describes both the procedures that the Internet Assigned Numbers Authority (IANA) uses and an RFC author should follow when new TCP parameters are requested and finally assigned. A small number of older experimental extensions that have not been widely implemented, deployed, and used are noted in . Many other supporting documents that are relevant to the development, implementation, and deployment of TCP are described in . A small number of fairly ubiquitous important implementation practices that are not currently documented in the RFC series are listed in . Within each section, RFCs are listed in the chronological order of their publication dates.

A small number of documents compose the core specification of TCP. These define the required core functionalities of TCP's header parsing, state machine, congestion control, and retransmission timeout computation. These base specifications must be correctly followed for interoperability. This is the fundamental TCP specification document . Written by Jon Postel as part of the Internet protocol suite's core, it describes the TCP packet format, the TCP state machine and event processing, and TCP's semantics for data transmission, reliability, flow control, multiplexing, and acknowledgment. Section 3.6 of RFC 793, describing TCP's handling of the IP precedence and security compartment, is mostly irrelevant today. RFC 2873 (discussed later in below) changed the IP precedence handling, and the security compartment portion of the API is no longer implemented or used. In addition, RFC 793 did not describe any congestion control mechanism. Otherwise, however, the majority of this document still accurately describes modern TCPs. RFC 793 is the last of a series of developmental TCP specifications, starting in the Internet Experimental Notes (IENs) and continuing in the RFC series. This document updates and clarifies RFC 793 (see above in ), fixing some specification bugs and oversights. It also explains some features such as keep-alives and Karn's and Jacobson's RTO estimation algorithms . ICMP interactions are mentioned, and some tips are given for efficient implementation. RFC 1122 is an Applicability Statement, listing the various features that MUST, SHOULD, MAY, SHOULD NOT, and MUST NOT be present in standards-conforming TCP implementations. Unlike a purely informational roadmap, this Applicability Statement is a standards document and gives formal rules for implementation. This document is of relevance to TCP because it defines how the pseudo-header for TCP's checksum computation is derived when 128-bit IPv6 addresses are used instead of 32-bit IPv4 addresses. Additionally, RFC 2675 (see of this document) describes TCP changes required to support IPv6 jumbograms. This document removes from the TCP specification all processing of the precedence bits of the TOS byte of the IP header. This resolves a conflict over the use of these bits between RFC 793 (see above in ) and Differentiated Services . Although RFC 793 (see above in ) did not contain any congestion control mechanisms, today congestion control is a required component of TCP implementations. This document defines congestion avoidance and control mechanism for TCP, based on Van Jacobson's 1988 SIGCOMM paper . A number of behaviors that together constitute what the community refers to as "Reno TCP" is described in RFC 5681. The name "Reno" comes from the Net/2 release of the 4.3 BSD operating system. This is generally regarded as the least common denominator among TCP flavors currently found running on Internet hosts. Reno TCP includes the congestion control features of slow start, congestion avoidance, fast retransmit, and fast recovery. RFC 5681 details the currently accepted congestion control mechanism, while RFC 1122, (see above in ) mandates that such a congestion control mechanism must be implemented. RFC 5681 differs slightly from the other documents listed in this section, as it does not affect the ability of two TCP endpoints to communicate; however, congestion control remains a critical component of any widely deployed TCP implementation and is required for the avoidance of congestion collapse and to ensure fairness among competing flows. RFCs 2001 and 2581 are the conceptual precursors of RFC 5681. The most important changes relative to RFC 2581 are: The initial window requirements were changed to allow larger Initial Windows as standardized in (see of this document). During slow start and congestion avoidance, the usage of Appropriate Byte Counting (see of this document) is explicitly recommended. The use of Limited Transmit (see of this document) is now recommended. This document analyzes how current TCP stacks process TCP urgent indications, and how the behavior of widely deployed middleboxes affects the urgent indications processing. The document updates the relevant specifications such that it accommodates current practice in processing TCP urgent indications. Finally, the document raises awareness about the reliability of TCP urgent indications in the Internet, and recommends against the use of urgent mechanism. Abstract of RFC 6298 : "This document defines the standard algorithm that Transmission Control Protocol (TCP) senders are required to use to compute and manage their retransmission timer. It expands on the discussion in Section 4.2.3.1 of RFC 1122 and upgrades the requirement of supporting the algorithm from a SHOULD to a MUST." RFC 6298 updates RFC 2988 by changing the initial RTO from 3s to 1s. This document clarifies what value to use with the TCP Maximum Segment Size (MSS) option when IP and TCP options are in use.

This section describes recommended TCP modifications that improve performance and security. represents fundamental changes to the protocol. Sections and list improvements over the congestion control and loss recovery mechanisms as specified in RFC 5681 (see ). describes algorithms that allow a TCP sender to detect whether it has entered loss recovery spuriously. comprises Path MTU Discovery mechanisms. Schemes for TCP/IP header compression are listed in . Finally, deals with the problem of preventing acceptance of forged segments and flooding attacks.

RFCs 2675 and 7323 represent fundamental changes to TCP by redefining how parts of the basic TCP header and options are interpreted. RFC 7323 defines the Window Scale option, which reinterprets the advertised receive window. RFC 2675 specifies that MSS option and urgent pointer fields with a value of 65,535 are to be treated specially. IPv6 supports longer datagrams than were allowed in IPv4. These are known as jumbograms, and use with TCP has necessitated changes to the handling of TCP's MSS and Urgent fields (both 16 bits). This document explains those changes. Although it describes changes to basic header semantics, these changes should only affect the use of very large segments, such as IPv6 jumbograms, which are currently rarely used in the general Internet. Supporting the behavior described in this document does not affect interoperability with other TCP implementations when IPv4 or non-jumbogram IPv6 is used. This document states that jumbograms are to only be used when it can be guaranteed that all receiving nodes, including each router in the end-to-end path, will support jumbograms. If even a single node that does not support jumbograms is attached to a local network, then no host on that network may use jumbograms. This explains why jumbogram use has been rare, and why this document is considered a performance optimization and not part of TCP over IPv6's basic functionality. This document defines TCP extensions for window scaling, timestamps, and protection against wrapped sequence numbers, for efficient and safe operation over paths with large bandwidth-delay products. These extensions are commonly found in currently used systems. The predecessor of this document, RFC 1323, was published in 1992, and is deployed in most TCP implementations. This document includes fixes and clarifications based on the gained deployment experience. One specific issued addressed in this specification is a recommendation how to modify the algorithm for estimating the mean RTT when timestamps are used. RFCs 1072, 1185, and 1323 are the conceptual precursors of RFC 7323.

Two of the most important aspects of TCP are its congestion control and loss recovery features. TCP treats lost packets as indicating congestion-related loss and cannot distinguish between congestion-related loss and loss due to transmission errors. Even when ECN is in use, there is a rather intimate coupling between congestion control and loss recovery mechanisms. There are several extensions to both features, and more often than not, a particular extension applies to both. In these two subsections, we group enhancements to TCP's congestion control, while the next subsection focus on TCP's loss recovery. This document defines a means for end hosts to detect congestion before congested routers are forced to discard packets. Although congestion notification takes place at the IP level, ECN requires support at the transport level (e.g., in TCP) to echo the bits and adapt the sending rate. This document updates RFC 793 (see of this document) to define two previously unused flag bits in the TCP header for ECN support. RFC 3540 (see of this document) provides a supplementary (experimental) means for more secure use of ECN, and RFC 2884 (see of this document) provides some sample results from using ECN. This document specifies an increase in the permitted initial window for TCP from one segment to three or four segments during the slow start phase, depending on the segment size. This document suggests that congestion control use the number of bytes acknowledged instead of the number of acknowledgments received. This change improves the performance of TCP in situations where there is no one-to-one relationship between data segments and acknowledgments (e.g., delayed ACKs or ACK loss) and closes a security hole TCP receivers can use to induce the sender into increasing the sending rate too rapidly (ACK-division ). ABC is recommended by RFC 5681 (see of this document). This document formally deprecates the use of ICMP Source Quench messages by transport protocols and recommends against the implementation of .

For the typical implementation of the TCP fast recovery algorithm described in RFC 5681 (see of this document), a TCP sender only retransmits a segment after a retransmit timeout has occurred, or after three duplicate ACKs have arrived triggering the fast retransmit. A single RTO might result in the retransmission of several segments, while the fast retransmit algorithm in RFC 5681 leads only to a single retransmission. Hence, multiple losses from a single window of data can lead to a performance degradation. Documents listed in this section aim to improve the overall performance of TCP's standard loss recovery algorithms. In particular, some of them allow TCP senders to recover more effectively when multiple segments are lost from a single flight of data. When more than one packet is lost during one RTT, TCP may experience poor performance since a TCP sender can only learn about a single lost packet per RTT from cumulative acknowledgments. This document defines the basic selective acknowledgment (SACK) mechanism for TCP, which can help to overcome these limitations. The receiving TCP returns SACK blocks to inform the sender which data has been received. The sender can then retransmit only the missing data segments. Abstract of RFC 3042 : "This document proposes a new Transmission Control Protocol (TCP) mechanism that can be used to more effectively recover lost segments when a connection's congestion window is small, or when a large number of segments are lost in a single transmission window." This algorithm described in RFC 3042 is called "Limited Transmit". Tests from 2004 showed that Limited Transmit was deployed in roughly one third of the web servers tested . Limited Transmit is recommended by RFC 5681 (see of this document). This document specifies a modification to the standard Reno fast recovery algorithm, whereby a TCP sender can use partial acknowledgments to make inferences determining the next segment to send in situations where SACK would be helpful but isn't available. Although it is only a slight modification, the NewReno behavior can make a significant difference in performance when multiple segments are lost from a single window of data. RFCs 2582 and 3782 are the conceptual precursors of RFC 6582. The main change in RFC 3782 relative to RFC 2582 was to specify the Careful variant of NewReno's Fast Retransmit and Fast Recovery algorithms and advance those two algorithms from Experimental to Standards Track status. The main change in RFC 6582 relative to RFC 3782 was to solve a performance degradation that could occur if FlightSize on Full ACK reception is zero. This document describes a conservative loss recovery algorithm for TCP that is based on the use of the selective acknowledgment (SACK) TCP option (see above in ). The algorithm conforms to the spirit of the congestion control specification in RFC 5681 (see of this document), but allows TCP senders to recover more effectively when multiple segments are lost from a single flight of data. RFC 6675 is a revision of RFC 3517 to address several situations that are not handled explicitly before. In particular, it improves the loss detection in the event that the sender has outstanding segments that are smaller than Sender Maximum Segment Size (SMSS). it modifies the definition of a "duplicate acknowledgment" to utilize the SACK information in detecting loss. it maintains the ACK clock under certain circumstances involving loss at the end of the window.

Spurious retransmission timeouts are harmful to TCP performance and multiple algorithms have been defined for detecting when spurious retransmissions have occurred, but they respond differently with regard to their manners of recovering performance. The IETF defined multiple algorithms because there are trade-offs in whether or not certain TCP options need to be implemented and concerns about IPR status. The Standards Track RFCs in this section are closely related to the Experimental RFCs in also addressing this topic. This document extends RFC 2018 (see of this document). It enables use of the SACK option to acknowledge duplicate packets. With this extension, called DSACK, the sender is able to infer the order of packets received at the receiver and, therefore, to infer when it has unnecessarily retransmitted a packet. A TCP sender could then use this information to detect spurious retransmissions (see ). This document describes the response portion of the Eifel algorithm, which can be used in conjunction with one of several methods of detecting when loss recovery has been spuriously entered, such as the Eifel detection algorithm in RFC 3522 (see ), the algorithm in RFC 3708 (see of this document), or F-RTO in RFC 5682 (see below in ). Abstract of RFC 4015 : "Based on an appropriate detection algorithm, the Eifel response algorithm provides a way for a TCP sender to respond to a detected spurious timeout. It adapts the retransmission timer to avoid further spurious timeouts and (depending on the detection algorithm) can avoid the often unnecessary go-back-N retransmits that would otherwise be sent. In addition, the Eifel response algorithm restores the congestion control state in such a way that packet bursts are avoided." The F-RTO detection algorithm , originally described in RFC 4138, provides an option for inferring spurious retransmission timeouts. Unlike some similar detection methods (e.g., RFCs 3522 and 3708, both listed in of this document), F-RTO does not rely on the use of any TCP options. The basic idea is to send previously unsent data after the first retransmission after a RTO. If the ACKs advance the window, the RTO may be declared spurious.

The MTUs supported by different links and tunnels within the Internet can vary widely. Fragmentation of packets larger than the supported MTU on a hop is undesirable. As TCP is the segmentation layer for dividing an application's byte stream into IP packet payloads, TCP implementations generally include Path MTU Discovery (PMTUD) mechanisms in order to maximize the size of segments they send, without causing fragmentation within the network. Some algorithms may utilize signaling from routers on the path to determine that the MTU on some part of the path has been exceeded. Abstract of RFC 1191 : "This memo describes a technique for dynamically discovering the maximum transmission unit (MTU) of an arbitrary internet path. It specifies a small change to the way routers generate one type of ICMP message. For a path that passes through a router that has not been so changed, this technique might not discover the correct Path MTU, but it will always choose a Path MTU as accurate as, and in many cases more accurate than, the Path MTU that would be chosen by current practice." Abstract of RFC 1981 : "This document describes Path MTU Discovery for IP version 6. It is largely derived from RFC 1191, which describes Path MTU Discovery for IP version 4."> Abstract of RFC 4821 : "This document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets. This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively."

Especially in streaming applications, the overhead of TCP/IP headers could correspond to more than 50% of the total amount of data sent. Such large overheads may be tolerable in wired LANs where capacity is often not an issue, but are excessive for WANs and wireless systems where bandwidth is scarce. Header compression schemes for TCP/IP like RObust Header Compression (ROHC) can significantly compress this overhead. It performs well over links with significant error rates and long round-trip times. This document describes a method for compressing the headers of TCP/IP datagrams to improve performance over low-speed serial links. The method described in this document is limited in its handling of TCP options and cannot compress the headers of SYNs and FINs. From the Abstract of RFC 6846 : "This document specifies a RObust Header Compression (ROHC) profile for compression of TCP/IP packets. The profile, called ROHC-TCP, provides efficient and robust compression of TCP headers, including frequently used TCP options such as selective acknowledgments (SACKs) and Timestamps." RFC 6846 is the successor of RFC 4996. It fixes a technical issue with the SACK compression and clarifies other compression methods used.

By default, TCP lacks any cryptographic structures to differentiate legitimate segments from those spoofed from malicious hosts. Spoofing valid segments requires correctly guessing a number of fields. The documents in this subsection describe ways to make that guessing harder or to prevent it from being able to affect a connection negatively. This document discusses the recently increased vulnerability of long-lived TCP connections, such as BGP connections, to reset (send RST) spoofing attacks. The document analyzes the vulnerability, discussing proposed solutions at the transport level and their inherent challenges, as well as existing network level solutions and the feasibility of their deployment. This document describes a nonstandard but widely implemented modification to TCP's handling of ICMP soft error messages that rejects pending connection-requests when such error messages are received. This behavior reduces the likelihood of long delays between connection-establishment attempts that may arise in some scenarios. This document describes the well-known TCP SYN flooding attack. It analyzes and discusses various countermeasures against these attacks, including their use and trade-offs. This document describes the TCP Authentication Option (TCP-AO), which is used to authenticate TCP segments. TCP-AO obsoletes the TCP MD5 Signature option of RFC 2385. It supports the use of stronger hash functions, protects against replays for long-lived TCP connections (as used, e.g., in BGP and LDP), coordinates key exchanges between endpoints, and provides a more explicit recommendation for external key management. Cryptographic algorithms for TCP-AO are defined in (see below in ). This document specifies the algorithms and attributes that can be used in TCP Authentication Option's (TCP-AO) (see above in ) current manual keying mechanism and provides the interface for future message authentication codes (MACs). Abstract of RFC 5927 : "This document discusses the use of the Internet Control Message Protocol (ICMP) to perform a variety of attacks against the Transmission Control Protocol (TCP). Additionally, this document describes a number of widely implemented modifications to TCP's handling of ICMP error messages that help to mitigate these issues." This document describes minor modifications to how TCP handles inbound segments. This renders TCP connections, especially long-lived connections such as H-323 or BGP, less vulnerable to spoofed packet injection attacks where the 4-tuple (the source and destination IP addresses and the source and destination ports) has been guessed. Abstract of RFC 6528 : "This document specifies an algorithm for the generation of TCP Initial Sequence Numbers (ISNs), such that the chances of an off-path attacker guessing the sequence numbers in use by a target connection are reduced. This document revises (and formally obsoletes) RFC 1948, and takes the ISN generation algorithm originally proposed in that document to Standards Track, formally updating RFC 793"

The RFCs in this section are either Experimental and may become Proposed Standards in the future or are Proposed Standards (or Informational), but can be considered experimental due to lack of wide deployment. At least part of the reason that they are still experimental is to gain more wide-scale experience with them before a standards track decision is made. If the Experimental RFC is a proposal for a new protocol capability or service, i.e., it requires a new TCP option code point, the implementation and experimentation should follow (see Section 5 of this document), which describes how the experimental TCP option code points can concurrently support multiple TCP extensions. By their publication as Experimental RFCs, it is hoped that the community of TCP researchers will analyze and test the contents of these RFCs. Although experimentation is encouraged, there is not yet formal consensus that these are fully logical and safe behaviors. Wide-scale deployment of implementations that use these features should be well thought out in terms of consequences.

As multiple flows may share the same paths, sections of paths, or other resources, the TCP implementation may benefit from sharing information across TCP connections or other flows. Some experimental proposals have been documented and some implementations have included the concepts. This document suggests how TCP connections between the same endpoints might share information, such as their congestion control state. To some degree, this is done in practice by a few operating systems; for example, Linux currently has a destination cache. Although this RFC is technically Informational, the concepts it describes are in experimental use, so we include it in this section. This document is a related proposal to RFC 2140 (see above in ). The idea behind the Congestion Manager, moving congestion control outside of individual TCP connections, represents a modification to the core of TCP, which supports sharing information among TCP connections. Although a Proposed Standard, some pieces of the Congestion Manager support architecture have not been specified yet, and it has not achieved use or implementation beyond experimental stacks, so it is not listed among the standard TCP enhancements in this roadmap.

Like the Standards Track documents listed in , there also exist new Experimental RFCs that specify fundamental changes to TCP. At the time of writing, the only example so far is TCP Fast Open that deviates from the standard TCP semantics of . This document describes TCP Fast Open that allows data to be carried in the SYN and SYN-ACK packets and consumed by the receiver during the initial connection handshake. It saves up to one RTT compared to the standard TCP, which requires a three-way handshake to complete before data can be exchanged.

TCP congestion control has been an extremely active research area for many years (see RFC 5783 discussed in of this document), as it determines the performance of many applications that use TCP. A number of Experimental RFCs address issues with flow start up, overshoot, and steady-state behavior in the basic algorithms of RFC 5681 (see of this document). In these subsections, enhancements to TCP's congestion control are listed. The next subsection focuses on TCP's loss recovery. This document suggests reducing the congestion window over time when no packets are flowing. This behavior is more aggressive than that specified in RFC 5681 (see of this document), which says that a TCP sender SHOULD set its congestion window to the initial window after an idle period of an RTO or greater. This document describes an optional addition to ECN that protects against accidental or malicious concealment of marked packets from the TCP sender. This document proposes a modification to TCP's congestion control mechanism for use with TCP connections with large congestion windows, to allow TCP to achieve a higher throughput in high-bandwidth environments. This document describes a more conservative slow-start behavior to prevent massive packet losses when a connection uses a very large congestion window. This document specifies the optional Quick-Start mechanism for TCP. This mechanism allows connections to use higher sending rates at the beginning of the data transfer or after an idle period, provided that there is significant unused bandwidth along the path, and the sender and all of the routers along the path approve this higher rate. This document describes an experimental modification to ECN (see of this document) for the use of ECN in TCP SYN/ACK packets. This would allow to ECN-mark rather than drop the TCP SYN/ACK packet at an ECN-capable router, and to avoid the severe penalty of a retransmission timeout for a connection when the SYN/ACK packet is dropped. This document describes a congestion control mechanism for acknowledgment (ACKs) traffic in TCP. The mechanism is based on the acknowledgment congestion control of the Datagram Congestion Control Protocol's (DCCP's) Congestion Control Identifier (CCID) 2 . This document proposes to increase the TCP initial window from between 2 and 4 segments, as specified in RFC 3390 (see of this document), to 10 segments with a fallback to the existing recommendation when performance issues are detected.

This document proposes the "Early Retransmit" mechanism for TCP (and SCTP) that can be used to recover lost segments when a connection's congestion window is small. In certain special circumstances, Early Retransmit reduces the number of duplicate acknowledgments required to trigger fast retransmit to recover segment losses without waiting for a lengthy retransmission timeout. This document describes how standard ICMP messages can be used to disambiguate true congestion loss from non-congestion loss caused by connectivity disruptions. It proposes a reversion strategy of TCP's retransmission timer that enables a more prompt detection of whether or not the connectivity has been restored. This document describes an experimental Proportional Rate Reduction (PRR) algorithm as an alternative to the widely deployed Fast Recovery algorithm, to improve the accuracy of the amount of data sent by TCP during loss recovery.

In addition to the Standards Track extensions to deal with spurious retransmissions in , Experimental proposals have also been documented. The Eifel detection algorithm allows a TCP sender to detect a posteriori whether it has entered loss recovery unnecessarily by using the TCP timestamp option to solve the ACK ambiguity. Abstract: "TCP and Stream Control Transmission Protocol (SCTP) provide notification of duplicate segment receipt through Duplicate Selective Acknowledgement (DSACKs) and Duplicate Transmission Sequence Number (TSN) notification, respectively. This document presents conservative methods of using this information to identify unnecessary retransmissions for various applications." In the presence of non-congestion events, such as packet reordering, an out-of-order segment does not necessarily indicate a lost segment and congestion. This document proposes to increase the threshold used to trigger a fast retransmission from the fixed value of three duplicate ACKs to about one congestion window of data in order to disambiguate true segment loss from segment reordering.

Besides the well-known retransmission timeout the TCP standard defines other timeouts. This section lists documents that deal with TCP's various timeouts. As a local per-connection parameter, the TCP user timeout controls how long transmitted data may remain unacknowledged before a connection is forcefully closed. This document specifies the TCP User Timeout Option that allows one end of a TCP connection to advertise its current user timeout value. This information provides advice to the other end of the TCP connection to adapt its user timeout accordingly.

MultiPath TCP (MPTCP) is an ongoing effort within the IETF that allows a TCP connection to simultaneously use multiple IP addresses / interfaces to spread their data across several subflows, while presenting a regular TCP interface to applications. Benefits of this include better resource utilization, better throughput and smoother reaction to failures. The documents listed in this section specify the Multipath TCP scheme, while the documents in Sections , , and provide some additional background information. This document presents a congestion control algorithm for multipath transport protocols such as Multipath TCP. It couples the congestion control algorithms running on different subflows by linking their increase functions, and dynamically controls the overall aggressiveness of the multipath flow. The result is an algorithm that is fair to TCP at bottlenecks while moving traffic away from congested links. This document presents protocol changes required to add multipath capability to TCP; specifically, those for signaling and setting up multiple paths ("subflows"), managing these subflows, reassembly of data, and termination of sessions.

RFCs listed here describes both the procedures that the Internet Assigned Numbers Authority (IANA) uses when handling assignments and the procedures an RFC author should follow when requesting new TCP option code points. Abstract of RFC 2780 : "This memo provides guidance for the IANA to use in assigning parameters for fields in the IPv4, IPv6, ICMP, UDP and TCP protocol headers." This document reserves both TCP options 253 and 254 for experimentation purposes. When such experiments are deployed in the Internet, they should follow the additional requirements in RFC 6994 (see below in ). From the Abstract of RFC 6335 : "This document defines the procedures that the Internet Assigned Numbers Authority (IANA) uses when handling assignment and other requests related to the Service Name and Transport Protocol Port Number registry." This document describes how the experimental TCP option code points can concurrently support multiple TCP extensions, even within the same connection. It creates an IANA registry for extensions to the experimental code points.

The RFCs listed here define extensions that have thus far failed to arouse substantial interest from implementers and have never seen widespread deployment or were found to be defective for general use. Most of them were reclassified by to Historic status. RFC 721 addresses the problem of implementing a reliable out-of-band signal (interrupts) for use in a host-to-host protocol. The proposal was not included in the final TCP specification. This document proposes a protocol to contact multiple services on a single well-known TCP port using a service name instead of a well-known number. This RFC defined an alternative to the Window Scale option for using large windows and described the "negative acknowledgment" or NAK option. There is a comparison of NAK and SACK methods and early discussion of TCP over satellite issues. RFC 1110 (see below in ) explains some problems with the approaches described in RFC 1106. The options described in this document have not been adopted by the larger community, although NAKs are used in the SCPS-TP adaptation of TCP for satellite and spacecraft use, developed by the Consultative Committee for Space Data Systems (CCSDS). Abstract of RFC 1110 : "The TCP Big Window option discussed in RFC 1106 will not work properly in an Internet environment which has both a high bandwidth * delay product and the possibility of disordering and duplicating packets. In such networks, the window size must not be increased without a similar increase in the sequence number space. Therefore, a different approach to big windows should be taken in the Internet." This document defined more robust TCP checksums than the 16-bit ones-complement in use today. A typographical error in RFC 1145 is fixed in RFC 1146; otherwise, the documents are the same. This document argues against "backwards compatible" TCP extensions. Specifically mentioned are several TCP enhancements that have been successful, including timestamps, window scaling, PAWS, and SACK. RFC 1263 presents an alternative approach called "protocol evolution", whereby several evolutionary versions of TCP would exist on hosts. These distinct TCP versions would represent upgrades to each other and could be header incompatible. Interoperability would be provided by having a virtualization layer select the right TCP version for a particular connection. This idea did not catch on with the community, while the type of extensions RFC 1263 specifically targeted as harmful did become popular. See RFC 1644, in below. The inventors of TCP believed that cached connection state could have been used to eliminate TCP's three-way handshake, to support two-packet request/response exchanges. RFC 1379 (see above in ) and RFC 1644 show that this is far from simple. Furthermore, T/TCP floundered on the ease of denial-of-service attacks that can result. One idea pioneered by T/TCP lives on in RFC 2140 (see of this document), in the sharing of state across connections. This document defines a TCP extension for applications that do not care about the order in which application-layer objects are received. Examples are multimedia and database applications. In practice, these applications either accept the possible performance loss because of TCP's strict ordering or use specialized transport protocols other than TCP, such as PR-SCTP . To overcome the exhaustion of the IP class B address space, this document suggests that a new version of TCP (TCPng) needs to be developed and deployed. It proposes that a globally unique address be assigned to the transport layer to uniquely identify an Internet host without specifying any routing information. Later work on splitting locator and identifier values is summarized well in , but no resulting changes to TCP have occurred. This document describes a method to exchange a cookie (nonce) during the connection establishment to negotiate elimination of receiver state. These cookies are later used to inhibit premature closing of connections and reduce retention of state after the connection has terminated. Since the cookie pair is too large to fit with the other TCP options in the 40 bytes of TCP option space, the document further describes a method to extent the option space after the connection establishment. Although RFC 6013 was published in 2011, the authors of this document places it in this section of the roadmap document due to two factors. The authors are not aware of any wide deployment and use of RFC 6013. RFC 6013 uses experimental TCP option code points, which prohibits a large-scale deployment.

This section contains several classes of documents that do not necessarily define current protocol behaviors but that are nevertheless of interest to TCP implementers. describes several foundational RFCs that give modern readers a better understanding of the principles underlying TCP's behaviors and development over the years. contains architectural guidelines and principles for TCP architects and designers. The documents listed in provide advice on using TCP in various types of network situations that pose challenges above those of typical wired links. Guidance for developing, analyzing, and evaluating TCP is given in . Some implementation notes and implementation advice can be found in . RFCs that describe tools for testing and debugging TCP implementations or that contain high-level tutorials on the protocol are listed . The TCP Management Information Bases are described in , and lists a number of case studies that have explored TCP performance.

The documents listed in this section contain information that is largely duplicated by the standards documents previously discussed. However, some of them contain a greater depth of problem statement explanation or other context. Particularly, RFCs 813 - 817 (known as the "Dave Clark Five") describe some early problems and solutions (RFC 815 only describes the reassembly of IP fragments and is not included in this TCP roadmap). This document is a very early precursor of the fundamental RFC 793 (see of this document), which already contained the three-way handshake in its final form and the concept of sliding windows for reliable data transmission. Apart from that, the segment layout is totally different and the specified API differs from the latter RFC 793 (see of this document). This document is the immediate precursor of RFC 793 (see of this document). The header format, the connection establishment (including the different connection states), and the overall API correspond mostly to the final Standard RFC 793 (see of this document). This document contains an early discussion of Silly Window Syndrome and its avoidance and motivates and describes the use of delayed acknowledgments. Suggestions and guidance for the design of tables and algorithms to keep track of various identifiers within a TCP/IP implementation are provided by this document . In this document , TCP's response to indications of network error conditions such as timeouts or received ICMP messages is discussed. This document contains implementation suggestions that are general and not TCP specific. However, they have been used to develop TCP implementations and describe some performance implications of the interactions between various layers in the Internet stack. Conclusion of RFC 872 : "The sometimes-expressed fear that using TCP on a local net is a bad idea is unfounded." This document contains some early experiences with congestion collapse and some initial thoughts on how to avoid it using congestion control in TCP. Furthermore, it defined an algorithm for efficient transmission of small packets that is today known as the Nagle algorithm. This document points out several specification bugs in the US Military's MIL-STD-1778 document, which was intended as a successor to RFC 793 (see of this document). This serves to remind us of the difficulty in specification writing (even when we work from existing documents!).

Some documents in this section contain architectural guidance and concerns, while others specify TCP- and congestion-control-related mechanisms that are broadly applicable and have impacts on TCP's congestion control techniques. Some of these documents are direct products of the Internet Architecture Board (IAB) giving their guidance on specific aspects of congestion control in the Internet. This document describes the underlying principles of the Internet architecture. It provides guidelines for network systems designs that have proven useful in the evolution of the Internet. This document motivates the use of end-to-end congestion control for preventing congestion collapse and providing fairness to TCP. Later work on TCP has included several more aggressive mechanisms than Reno TCP includes, and RFC 5033 (see of this document) provides additional guidance on use of such algorithms. The fundamental architectural discussion in RFC 2914 remains valid, regarding the standards process role in defining protocol aspects that are critical to performance and avoiding congestion collapse scenarios. This document is a plea that firewall vendors not send gratuitous TCP RST (Reset) packets when unassigned TCP header bits are used. This practice prevents desirable extension and evolution of the protocol and thus is potentially harmful to the future of the Internet. This document updates RFC 1958 (see above in ) by outlining some philosophical guidelines for architects and designers of Internet backbone networks. The document describes the Simplicity Principle, which states that complexity is the primary impediment to efficient scaling. This document discusses some of the issues in defining alternate semantics for the ECN field and specifies requirements for a safe coexistence with routers that do not understand the defined alternate semantics. Abstract of RFC 6182 : "This document outlines architectural guidelines for the development of a Multipath Transport Protocol, with references to how these architectural components come together in the development of a Multipath TCP (MPTCP) (see of this document). This document lists certain high-level design decisions that provide foundations for the design of the MPTCP protocol, based upon these architectural requirements"

As the internetworking field has explored wireless, satellite, cellular telephone, and other kinds of link-layer technologies, a large body of work has built up on enhancing TCP performance for such links. The RFCs listed in this section describe some of these more challenging network environments and how TCP interacts with them. From the Abstract of RFC 2488 : "While TCP works over satellite channels there are several IETF standardized mechanisms that enable TCP to more effectively utilize the available capacity of the network path. This document outlines some of these TCP mitigations. At this time, all mitigations discussed in this document are IETF standards track mechanisms (or are compliant with IETF standards)." Several methods of improving TCP performance over long thin networks (i.e., networks with low bandwidth and high delay), such as geosynchronous satellite links, are discussed in this document . A particular set of TCP options is developed that should work well in such environments and be safe to use in the global Internet. The implications of such environments have been further discussed in RFCs 3150 and 3155 (see below in ), and these documents should be preferred where there is overlap between them and RFC 2757 (see of this document). This document discusses the advantages and disadvantages of several different experimental means of improving TCP performance over long-delay or error-prone paths. These include T/TCP, larger initial windows, byte counting, delayed acknowledgments, slow start thresholds, NewReno and SACK-based loss recovery, FACK , ECN, various corruption-detection mechanisms, congestion avoidance changes for fairness, use of multiple parallel flows, pacing, header compression, state sharing, and ACK congestion control, filtering, and reconstruction. Although RFC 2488 (see above in ) looks at standard extensions, this document focuses on more experimental means of performance enhancement. From the Abstract of RFC 3135 : "This document is a survey of Performance Enhancing Proxies (PEPs) often employed to improve degraded TCP performance caused by characteristics of specific link environments, for example, in satellite, wireless WAN, and wireless LAN environments. Different types of Performance Enhancing Proxies are described as well as the mechanisms used to improve performance." From the Abstract of RFC 3150 : "This document makes performance-related recommendations for users of network paths that traverse "very low bit-rate" links....This recommendation may be useful in any network where hosts can saturate available bandwidth, but the design space for this recommendation explicitly includes connections that traverse 56 Kb/second modem links or 4.8 Kb/second wireless access links - both of which are widely deployed." From the Abstract of RFC 3155 : "This document discusses the specific TCP mechanisms that are problematic in environments with high uncorrected error rates, and discusses what can be done to mitigate the problems without introducing intermediate devices into the connection." From the Abstract of RFC 3366 : "This document provides advice to the designers of digital communication equipment and link-layer protocols employing link-layer Automatic Repeat reQuest (ARQ) techniques. This document presumes that the designers wish to support Internet protocols, but may be unfamiliar with the architecture of the Internet and with the implications of their design choices for the performance and efficiency of Internet traffic carried over their links." From the Abstract of RFC 3449 : "This document describes TCP performance problems that arise because of asymmetric effects. These problems arise in several access networks, including bandwidth-asymmetric networks and packet radio subnetworks, for different underlying reasons. However, the end result on TCP performance is the same in both cases: performance often degrades significantly because of imperfection and variability in the ACK feedback from the receiver to the sender. The document details several mitigations to these effects, which have either been proposed or evaluated in the literature, or are currently deployed in networks. From the Abstract of RFC 3481 : "This document describes a profile for optimizing TCP to adapt so that it handles paths including second (2.5G) and third (3G) generation wireless networks." This document describes how TCP performance can be negatively affected by some particular lower-layer behaviors and provides guidance in designing lower-layer networks and protocols to be amicable to TCP. RFC 3366 (see above in ) specifically focuses on ARQ mechanisms, while RFC 3819 more widely covers additional aspects of the underlying layers

Documents in this section give general guidance for developing, analyzing, and evaluating TCP. Some of the documents discuss, for example, the properties of congestion control protocols that are "safe" for Internet deployment as well as how to measure the properties of congestion control mechanisms and transport protocols. This document considers the evaluation of suggested congestion control algorithms that differ from the principles outlined in RFC 2914 (see of this document). It is useful for authors of such algorithms as well as for IETF members reviewing the associated documents. This document discusses metrics that need to be considered when evaluating new or modified congestion control mechanisms for the Internet. Among other topics, the document discusses throughput, delay, loss rates, response times, fairness, and robustness for challenging environments. This document summarizes the main open problems in the domain of Internet congestion control. As a good starting point for newcomers, the document describes several new challenges that are becoming important as the network grows, as well as some issues that have been known for many years. This document describes a threat analysis for Multipath TCP (MPTCP) (see of this document). The document discusses several types of attacks and provides recommendations for MPTCP designers how to create an MPTCP specification that is as secure as the current (single-path) TCP. From the Abstract of RFC 6349 : "This framework describes a practical methodology for measuring end-to-end TCP Throughput in a managed IP network. The goal is to provide a better indication in regard to user experience. In this framework, TCP and IP parameters are specified to optimize TCP Throughput."

This document clarifies that operating systems need to manage their limited resources, which may include TCP connection state, and that these decisions can be made with application input, but they do not need to be part of the TCP protocol specification itself. Abstract of RFC 879 : "This memo discusses the TCP Maximum Segment Size Option and related topics. The purposes [sic] is to clarify some aspects of TCP and its interaction with IP. This memo is a clarification to the TCP specification, and contains information that may be considered as 'advice to implementers'." This document lists a number of implementation techniques for efficiently computing the Internet checksum (used by TCP). Incrementally updating the Internet checksum is useful to routers in updating IP checksums. Some middleboxes that alter TCP headers may also be able to update the TCP checksum incrementally. This document expands upon the explanation of the incremental update procedure in RFC 1071 (see above in ). This document describes the motivation for implementing the Internet checksum in hardware, rather than in software, and provides an implementation example. From the Abstract of RFC 2525 : "This memo catalogs a number of known TCP implementation problems. The goal in doing so is to improve conditions in the existing Internet by enhancing the quality of current TCP/IP implementations." From abstract: "This memo catalogs several known Transmission Control Protocol (TCP) implementation problems dealing with Path Maximum Transmission Unit Discovery (PMTUD), including the long-standing black hole problem, stretch acknowledgments (ACKs) due to confusion between Maximum Segment Size (MSS) and segment size, and MSS advertisement based on PMTU." This document describes the de facto standard sockets API for programming with TCP. This API is implemented nearly ubiquitously in modern operating systems and programming languages. This document describes a number of simple and efficient methods for the selection of the client port number. It reduces the possibility of an attacker guessing the correct five-tuple (Protocol, Source/Destination Address, Source/Destination Port). This document describes the usage of the TCP Timestamps option (RFC 7323, see of this document) to perform heuristics to determine whether or not to allow the creation of a new incarnation of a connection that is in the TIME-WAIT state. This document clarifies the actions that a TCP can take on connections that are experiencing the Zero Window Probe (ZWP) condition. This document characterizes the impact that Multipath TCP (MPTCP) (see of this document) may have on applications. It further discusses compatibility issues of MPTCP in combination with non-MPTCP-aware applications. Finally, it describes a basic API that is a simple extension of TCP's interface for MPTCP-aware applications.

This document is an extremely brief overview of the TCP/IP protocol suite as a whole. It gives some explanation as to how and where TCP fits in. A few of the tools that this document describes are still maintained and in use today, for example, ttcp and tcpdump. However, many of the tools described do not relate specifically to TCP and are no longer used or easily available. This document describes a number of TCP packet generation and analysis tools. Although some of these tools are no longer readily available or widely used, for the most part they are still relevant and usable. This document provides an overview of RFCs related to congestion control that had been published at the time. The focus of the document is on end-host-based congestion control.

The first MIB module defined for use with Simple Network Management Protocol (SNMP) was a single monolithic MIB module, called MIB-I, defined in RFC 1156. This evolved over time to the MIB-II specification in RFC 1213, which obsoletes RFC 1156. It then became apparent that having a single monolithic MIB module was not scalable, given the number and breadth of MIB data definitions that needed to be included. Thus, additional MIB modules were defined, and those parts of MIB-II that needed to evolve were split off. Eventually, the remaining parts of MIB-II were also split off, the TCP-specific part being documented in RFC 2012. RFC 2012 was obsoleted by RFC 4022, which is the primary TCP MIB document at the time of writing. For current TCP implementers, RFC 4022 should be supported. This document describes the required MIB fields for TCP implementations with minor corrections and no technical changes from RFC 1066, which it obsoletes. This is the Standards Track RFC for MIB-I. This document describes the second version of the MIB in a monolithic form. It is the immediate successor of RFC 1158, with minor modifications. It obsoletes the MIB-I, defined in RFC 1156 (see above in ). In an update to RFC 1213 (see of this document), this document defines the TCP MIB by splitting out the TCP-specific portions. It is now obsoleted by RFC 4022 (see below in ). This document augments RFC 2012 (see of this document) by adding an IPv6-specific connection table. The rest of RFC 2012 holds for any IP version. RFC 2452 is now obsoleted by RFC 4022 (see below in ). Although it is a Standards Track RFC, RFC 2452 is considered a historic mistake by the MIB community, as it is based on the idea of parallel IPv4 and IPv6 structures. Although IPv6 requires new structures, the community has decided to define a single generic structure for both IPv4 and IPv6. This will aid in definition, implementation, and transition between IPv4 and IPv6. This document obsoletes RFCs 2012 and 2452 (see above in ) and specifies the current standard for the TCP MIB that should be deployed. This document describes extended performance statistics for TCP. They are designed to use TCP's ideal vantage point to diagnose performance problems in both the network and the application.

This document presents a field report about the deployment of a very early version of TCP, the so-called INWN #39 protocol, which is originally described by Cerf and Kahn in INWG Note #39 to use a PDP-11 line printer via the ARPANET. This document is a status report about experiments concerning the TCP retransmission timeout calculation and also provides advice for implementers. This document points out a problem with acting on received reset segments while one is in the TIME-WAIT state. The main recommendation is that hosts in TIME-WAIT ignore resets. This recommendation might not currently be widely implemented. This document presents results of some simulations using TCP initial windows greater than 1 segment. The analysis indicates that user-perceived performance can be improved by increasing the initial window to 3 segments. This document uses simulation results to clear up some concerns about using an initial window of 4 segments when the network path has less provisioning. This document describes experimental results that show some improvements to the performance of both short- and long-lived connections due to ECN.

There are a few important implementation tactics for the TCP that have not yet been described in any RFC. Although this roadmap is primarily concerned with mapping the TCP RFCs, this section is included because an implementer needs to be aware of these important issues. Header prediction is a trick to speed up the processing of segments. Van Jacobson and Mike Karels developed the technique in the late 1980s. The basic idea is that some processing time can be saved when most of a segment's fields can be predicted from previous segments. A good description of this was sent to the TCP-IP mailing list by Van Jacobson on March 9, 1988 (see for the full message): Quite a bit of the speedup comes from an algorithm that we ('we' refers to collaborator Mike Karels and myself) are calling "header prediction". The idea is that if you're in the middle of a bulk data transfer and have just seen a packet, you know what the next packet is going to look like: It will look just like the current packet with either the sequence number or ack number updated (depending on whether you're the sender or receiver). Combining this with the "Use hints" epigram from Butler Lampson's classic "Epigrams for System Designers", you start to think of the tcp state (rcv.nxt, snd.una, etc.) as "hints" about what the next packet should look like. If you arrange those "hints" so they match the layout of a tcp packet header, it takes a single 14-byte compare to see if your prediction is correct (3 longword compares to pick up the send & ack sequence numbers, header length, flags and window, plus a short compare on the length). If the prediction is correct, there's a single test on the length to see if you're the sender or receiver followed by the appropriate processing. E.g., if the length is non-zero (you're the receiver), checksum and append the data to the socket buffer then wake any process that's sleeping on the buffer. Update rcv.nxt by the length of this packet (this updates your "prediction" of the next packet). Check if you can handle another packet the same size as the current one. If not, set one of the unused flag bits in your header prediction to guarantee that the prediction will fail on the next packet and force you to go through full protocol processing. Otherwise, you're done with this packet. So, the *total* tcp protocol processing, exclusive of checksumming, is on the order of 6 compares and an add. FACK includes an alternate algorithm for triggering fast retransmit , based on the extent of the SACK scoreboard. Its goal is to trigger fast retransmit as soon as the receiver's reassembly queue is larger than the duplicate ACK threshold, as indicated by the difference between the forward most SACK block edge and SND.UNA. This algorithm quickly and reliably triggers fast retransmit in the presence of burst losses -- often on the first SACK following such a loss. Such a threshold-based algorithm also triggers fast retransmit immediately in the presence of any reordering with extent greater than the duplicate ACK threshold. FACK is implemented in Linux and turned on per default. In the last decade significant research effort has been put into experimental TCP congestion control modifications for obtaining high throughput with reduced startup and recovery times. Only a few RFCs have been published on some of these modifications, including HighSpeed TCP , Limited Slow-Start , and Quick-Start (see of this document for more information on each), but high-rate congestion control mechanisms are still considered an open issue in congestion control research. Some other schemes have been published as Internet-Drafts, e.g. CUBIC (the standard TCP congestion control algorithm in Linux), Compound TCP , and H-TCP or have been discussed a little by the IETF, but much of the work in this area has not been adopted within the IETF yet, so the majority of this work is outside the RFC series and may be discussed in other products of the IRTF Internet Congestion Control Research Group (ICCRG).

This document introduces no new security considerations. Each RFC listed in this document attempts to address the security considerations of the specification it contains.

University of Southern California (USC)/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90291 US

University of Southern California (USC)/ Information Sciences Institute (ISI)

4676 Admiralty Way Marina del Rey CA 90292-6695 US +1 213 822 1511 Braden@ISI.EDU

Cisco Systems, Inc.

170 West Tasman Drive San Jose CA 95134-1706 USA +1 408 527 8213 +1 408 527 8254 deering@cisco.com

Nokia

232 Java Drive Sunnyvale CA 94089 USA +1 408 990 2004 +1 408 743 5677 hinden@iprg.nokia.com

Internet internet protocol version 6 IPv6 This document specifies version 6 of the Internet Protocol (IPv6), also sometimes referred to as IP Next Generation or IPng. This memo describes a conflict between TCP and DiffServ on the use of the three leftmost bits in the TOS octet of an IPv4 header. [STANDARDS-TRACK] This document defines TCP's four intertwined congestion control algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. In addition, the document specifies how TCP should begin transmission after a relatively long idle period, as well as discussing various acknowledgment generation methods. This document obsoletes RFC 2581. [STANDARDS-TRACK] This document analyzes how current TCP implementations process TCP urgent indications and how the behavior of some widely deployed middleboxes affects how end systems process urgent indications. This document updates the relevant specifications such that they accommodate current practice in processing TCP urgent indications, raises awareness about the reliability of TCP urgent indications in the Internet, and recommends against the use of urgent indications (but provides advice to applications that do). [STANDARDS-TRACK] This document defines the standard algorithm that Transmission Control Protocol (TCP) senders are required to use to compute and manage their retransmission timer. It expands on the discussion in Section 4.2.3.1 of RFC 1122 and upgrades the requirement of supporting the algorithm from a SHOULD to a MUST. This document obsoletes RFC 2988. [STANDARDS-TRACK] This memo discusses what value to use with the TCP Maximum Segment Size (MSS) option, and updates RFC 879 and RFC 2385. This document is not an Internet Standards Track specification; it is published for informational purposes. Berkeley Software Design, Inc.

4719 Weston Hills Drive Eagan MN 55123 US +1 612 405 8194 dab@bsdi.com

Cisco Systems, Inc.

170 West Tasman Drive San Jose CA 95134-1706 US +1 408 527 8213 deering@cisco.com

Nokia

313 Fairchild Drive Mountain View CA 94043 US +1 650 625 2004 hinden@iprg.nokia.com

A "jumbogram" is an IPv6 packet containing a payload longer than 65,535 octets. This document describes the IPv6 Jumbo Payload option, which provides the means of specifying such large payload lengths. It also describes the changes needed to TCP and UDP to make use of jumbograms. Jumbograms are relevant only to IPv6 nodes that may be attached to links with a link MTU greater than 65,575 octets, and need not be implemented or understood by IPv6 nodes that do not support attachment to links with such large MTUs. This document specifies a set of TCP extensions to improve performance over paths with a large bandwidth * delay product and to provide reliable operation over very high-speed paths. It defines the TCP Window Scale (WS) option and the TCP Timestamps (TS) option and their semantics. The Window Scale option is used to support larger receive windows, while the Timestamps option can be used for at least two distinct mechanisms, Protection Against Wrapped Sequences (PAWS) and Round-Trip Time Measurement (RTTM), that are also described herein.</t><t> This document obsoletes RFC 1323 and describes changes from it. This memo specifies the incorporation of ECN (Explicit Congestion Notification) to TCP and IP, including ECN's use of two bits in the IP header. [STANDARDS-TRACK] This document proposes a small modification to the way TCP increases its congestion window. Rather than the traditional method of increasing the congestion window by a constant amount for each arriving acknowledgment, the document suggests basing the increase on the number of previously unacknowledged bytes each ACK covers. This change improves the performance of TCP, as well as closes a security hole TCP receivers can use to induce the sender into increasing the sending rate too rapidly. This memo defines an Experimental Protocol for the Internet community. This document formally deprecates the use of ICMP Source Quench messages by transport protocols, formally updating RFC 792, RFC 1122, and RFC 1812. [STANDARDS-TRACK] Pittsburgh Supercomputing Center

4400 Fifth Ave Pittsburgh PA 15213 mathis@psc.edu

Pittsburgh Supercomputing Center

4400 Fifth Ave Pittsburgh PA 15213 mahdavi@psc.edu

Lawrence Berkeley National Laboratory

One Cyclotron Road Berkeley CA 94720 floyd@ee.lbl.gov

Sun Microsystems, Inc.

2550 Garcia Ave. MPK17-202 Mountain View CA 94043 allyn@eng.sun.com

Transport TCP acknowledgement transmission control protocol TCP may experience poor performance when multiple packets are lost from one window of data. With the limited information available from cumulative acknowledgments, a TCP sender can only learn about a single lost packet per round trip time. An aggressive sender could choose to retransmit packets early, but such retransmitted segments may have already been successfully received. A Selective Acknowledgment (SACK) mechanism, combined with a selective repeat retransmission policy, can help to overcome these limitations. The receiving TCP sends back SACK packets to the sender informing the sender of data that has been received. The sender can then retransmit only the missing data segments. This memo proposes an implementation of SACK and discusses its performance and related issues. This document proposes a new Transmission Control Protocol (TCP) mechanism that can be used to more effectively recover lost segments when a connection's congestion window is small, or when a large number of segments are lost in a single transmission window. [STANDARDS-TRACK] RFC 5681 documents the following four intertwined TCP congestion control algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. RFC 5681 explicitly allows certain modifications of these algorithms, including modifications that use the TCP Selective Acknowledgment (SACK) option (RFC 2883), and modifications that respond to "partial acknowledgments" (ACKs that cover new data, but not all the data outstanding when loss was detected) in the absence of SACK. This document describes a specific algorithm for responding to partial acknowledgments, referred to as "NewReno". This response to partial acknowledgments was first proposed by Janey Hoe. This document obsoletes RFC 3782. [STANDARDS-TRACK] This document presents a conservative loss recovery algorithm for TCP that is based on the use of the selective acknowledgment (SACK) TCP option. The algorithm presented in this document conforms to the spirit of the current congestion control specification (RFC 5681), but allows TCP senders to recover more effectively when multiple segments are lost from a single flight of data. This document obsoletes RFC 3517 and describes changes from it. [STANDARDS-TRACK] This note defines an extension of the Selective Acknowledgement (SACK) Option for TCP. [STANDARDS-TRACK] Based on an appropriate detection algorithm, the Eifel response algorithm provides a way for a TCP sender to respond to a detected spurious timeout. It adapts the retransmission timer to avoid further spurious timeouts and (depending on the detection algorithm) can avoid the often unnecessary go-back-N retransmits that would otherwise be sent. In addition, the Eifel response algorithm restores the congestion control state in such a way that packet bursts are avoided. [STANDARDS-TRACK] The purpose of this document is to move the F-RTO (Forward RTO-Recovery) functionality for TCP in RFC 4138 from Experimental to Standards Track status. The F-RTO support for Stream Control Transmission Protocol (SCTP) in RFC 4138 remains with Experimental status. See Appendix B for the differences between this document and RFC 4138.</t><t> Spurious retransmission timeouts cause suboptimal TCP performance because they often result in unnecessary retransmission of the last window of data. This document describes the F-RTO detection algorithm for detecting spurious TCP retransmission timeouts. F-RTO is a TCP sender-only algorithm that does not require any TCP options to operate. After retransmitting the first unacknowledged segment triggered by a timeout, the F-RTO algorithm of the TCP sender monitors the incoming acknowledgments to determine whether the timeout was spurious. It then decides whether to send new segments or retransmit unacknowledged segments. The algorithm effectively helps to avoid additional unnecessary retransmissions and thereby improves TCP performance in the case of a spurious timeout. [STANDARDS-TRACK] Digital Equipment Corporation (DEC) , Western Research Laboratory

100 Hamilton Avenue Palo Alto CA 94301 US +1 415 853 6643 mogul@decwrl.dec.com

Xerox Palo Alto Research Center

3333 Coyote Hill Road Palo Alto CA 94304 US +1 415 494 4839 deering@xerox.com

This memo describes a technique for dynamically discovering the maximum transmission unit (MTU) of an arbitrary internet path. It specifies a small change to the way routers generate one type of ICMP message. For a path that passes through a router that has not been so changed, this technique might not discover the correct Path MTU, but it will always choose a Path MTU as accurate as, and in many cases more accurate than, the Path MTU that would be chosen by current practice. Digital Equipment Corporation

110 Spitbrook Road ZKO3-3/U14 Nashua NH 03062 US +1 603 881 2608 +1 603 881 0120 mccann@zk3.dec.com

Xerox Palo Alto Research Center

3333 Coyote Hill Road Palo Alto CA 94304 US +1 415 812 4839 +1 415 812 4471 deering@parc.xerox.com

Digital Equipment Corporation, Western Research Laboratory

250 University Avenue Palo Alto CA 94301 US +1 415 617 3304 mogul@pa.dec.com

This document describes Path MTU Discovery for IP version 6. It is largely derived from RFC 1191, which describes Path MTU Discovery for IP version 4. This document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets. This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively. [STANDARDS-TRACK] Lawrence Berkeley Laboratory, Real Time Systems Group

Mail Stop 46A Berkeley CA 94720 US van@helios.ee.lbl.gov

This document specifies a RObust Header Compression (ROHC) profile for compression of TCP/IP packets. The profile, called ROHC-TCP, provides efficient and robust compression of TCP headers, including frequently used TCP options such as selective acknowledgments (SACKs) and Timestamps.</t><t> ROHC-TCP works well when used over links with significant error rates and long round-trip times. For many bandwidth-limited links where header compression is essential, such characteristics are common.</t><t> This specification obsoletes RFC 4996. It fixes a technical issue with the SACK compression and clarifies other compression methods used. [STANDARDS-TRACK] Recent analysis of potential attacks on core Internet infrastructure indicates an increased vulnerability of TCP connections to spurious resets (RSTs), sent with forged IP source addresses (spoofing). TCP has always been susceptible to such RST spoofing attacks, which were indirectly protected by checking that the RST sequence number was inside the current receive window, as well as via the obfuscation of TCP endpoint and port numbers. For pairs of well-known endpoints often over predictable port pairs, such as BGP or between web servers and well-known large-scale caches, increases in the path bandwidth-delay product of a connection have sufficiently increased the receive window space that off-path third parties can brute-force generate a viable RST sequence number. The susceptibility to attack increases with the square of the bandwidth, and thus presents a significant vulnerability for recent high-speed networks. This document addresses this vulnerability, discussing proposed solutions at the transport level and their inherent challenges, as well as existing network level solutions and the feasibility of their deployment. This document focuses on vulnerabilities due to spoofed TCP segments, and includes a discussion of related ICMP spoofing attacks on TCP connections. This memo provides information for the Internet community. This document describes TCP SYN flooding attacks, which have been well-known to the community for several years. Various countermeasures against these attacks, and the trade-offs of each, are described. This document archives explanations of the attack and common defense techniques for the benefit of TCP implementers and administrators of TCP servers or networks, but does not make any standards-level recommendations. This memo provides information for the Internet community. This document describes a non-standard, but widely implemented, modification to TCP's handling of ICMP soft error messages that rejects pending connection-requests when those error messages are received. This behavior reduces the likelihood of long delays between connection-establishment attempts that may arise in a number of scenarios, including one in which dual-stack nodes that have IPv6 enabled by default are deployed in IPv4 or mixed IPv4 and IPv6 environments. This memo provides information for the Internet community. This document specifies the TCP Authentication Option (TCP-AO), which obsoletes the TCP MD5 Signature option of RFC 2385 (TCP MD5). TCP-AO specifies the use of stronger Message Authentication Codes (MACs), protects against replays even for long-lived TCP connections, and provides more details on the association of security with TCP connections than TCP MD5. TCP-AO is compatible with either a static Master Key Tuple (MKT) configuration or an external, out-of-band MKT management mechanism; in either case, TCP-AO also protects connections when using the same MKT across repeated instances of a connection, using traffic keys derived from the MKT, and coordinates MKT changes between endpoints. The result is intended to support current infrastructure uses of TCP MD5, such as to protect long-lived connections (as used, e.g., in BGP and LDP), and to support a larger set of MACs with minimal other system and operational changes. TCP-AO uses a different option identifier than TCP MD5, even though TCP-AO and TCP MD5 are never permitted to be used simultaneously. TCP-AO supports IPv6, and is fully compatible with the proposed requirements for the replacement of TCP MD5. [STANDARDS-TRACK] The TCP Authentication Option (TCP-AO) relies on security algorithms to provide authentication between two end-points. There are many such algorithms available, and two TCP-AO systems cannot interoperate unless they are using the same algorithms. This document specifies the algorithms and attributes that can be used in TCP-AO's current manual keying mechanism and provides the interface for future message authentication codes (MACs). [STANDARDS-TRACK] This document discusses the use of the Internet Control Message Protocol (ICMP) to perform a variety of attacks against the Transmission Control Protocol (TCP). Additionally, this document describes a number of widely implemented modifications to TCP's handling of ICMP error messages that help to mitigate these issues. This document is not an Internet Standards Track specification; it is published for informational purposes. TCP has historically been considered to be protected against spoofed off-path packet injection attacks by relying on the fact that it is difficult to guess the 4-tuple (the source and destination IP addresses and the source and destination ports) in combination with the 32-bit sequence number(s). A combination of increasing window sizes and applications using longer-term connections (e.g., H-323 or Border Gateway Protocol (BGP) [STANDARDS-TRACK] This document specifies an algorithm for the generation of TCP Initial Sequence Numbers (ISNs), such that the chances of an off-path attacker guessing the sequence numbers in use by a target connection are reduced. This document revises (and formally obsoletes) RFC 1948, and takes the ISN generation algorithm originally proposed in that document to Standards Track, formally updating RFC 793. [STANDARDS-TRACK] University of Southern California/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292-6695 USA +1 310-822-1511 x151 +1 310-823-6714 touch@isi.edu http://www.isi.edu/~touch

Transport TCP congestion transmission control protocol This memo makes the case for interdependent TCP control blocks, where part of the TCP state is shared among similar concurrent connections, or across similar connection instances. TCP state includes a combination of parameters, such as connection state, current round- trip time estimates, congestion control information, and process information. This state is currently maintained on a per-connection basis in the TCP control block, but should be shared across connections to the same host. The goal is to improve transient transport performance, while maintaining backward-compatibility with existing implementations. This document is a product of the LSAM project at ISI. This document describes the Congestion Manager (CM), an end-system module that enables an ensemble of multiple concurrent streams from a sender destined to the same receiver and sharing the same congestion properties to perform proper congestion avoidance and control, and allows applications to easily adapt to network congestion. [STANDARDS-TRACK] This document describes an experimental TCP mechanism TCP Fast Open (TFO). TFO allows data to be carried in the SYN and SYN-ACK packets and consumed by the receiving end during the initial connection handshake, and saves up to one full round trip time (RTT) compared to the standard TCP, which requires a three-way handshake (3WHS) to complete before data can be exchanged. However TFO deviates from the standard TCP semantics since the data in the SYN could be replayed to an application in some rare circumstances. Applications should not use TFO unless they can tolerate this issue detailed in the Applicability section. This document describes a simple modification to TCP's congestion control algorithms to decay the congestion window cwnd after the transition from a sufficiently-long application-limited period, while using the slow-start threshold ssthresh to save information about the previous value of the congestion window. This memo defines an Experimental Protocol for the Internet community. This note describes the Explicit Congestion Notification (ECN)-nonce, an optional addition to ECN that protects against accidental or malicious concealment of marked packets from the TCP sender. It improves the robustness of congestion control by preventing receivers from exploiting ECN to gain an unfair share of network bandwidth. The ECN-nonce uses the two ECN-Capable Transport (ECT)codepoints in the ECN field of the IP header, and requires a flag in the TCP header. It is computationally efficient for both routers and hosts. This memo defines an Experimental Protocol for the Internet community. The proposals in this document are experimental. While they may be deployed in the current Internet, they do not represent a consensus that this is the best method for high-speed congestion control. In particular, we note that alternative experimental proposals are likely to be forthcoming, and it is not well understood how the proposals in this document will interact with such alternative proposals. This document proposes HighSpeed TCP, a modification to TCP's congestion control mechanism for use with TCP connections with large congestion windows. The congestion control mechanisms of the current Standard TCP constrains the congestion windows that can be achieved by TCP in realistic environments. For example, for a Standard TCP connection with 1500-byte packets and a 100 ms round-trip time, achieving a steady-state throughput of 10 Gbps would require an average congestion window of 83,333 segments, and a packet drop rate of at most one congestion event every 5,000,000,000 packets (or equivalently, at most one congestion event every 1 2/3 hours). This is widely acknowledged as an unrealistic constraint. To address his limitation of TCP, this document proposes HighSpeed TCP, and solicits experimentation and feedback from the wider community. This document describes an optional modification for TCP's slow-start for use with TCP connections with large congestion windows. For TCP connections that are able to use congestion windows of thousands (or tens of thousands) of MSS-sized segments (for MSS the sender's MAXIMUM SEGMENT SIZE), the current slow-start procedure can result in increasing the congestion window by thousands of segments in a single round-trip time. Such an increase can easily result in thousands of packets being dropped in one round-trip time. This is often counter-productive for the TCP flow itself, and is also hard on the rest of the traffic sharing the congested link. This note describes Limited Slow-Start as an optional mechanism for limiting the number of segments by which the congestion window is increased for one window of data during slow-start, in order to improve performance for TCP connections with large congestion windows. This memo defines an Experimental Protocol for the Internet community. This document specifies an optional Quick-Start mechanism for transport protocols, in cooperation with routers, to determine an allowed sending rate at the start and, at times, in the middle of a data transfer (e.g., after an idle period). While Quick-Start is designed to be used by a range of transport protocols, in this document we only specify its use with TCP. Quick-Start is designed to allow connections to use higher sending rates when there is significant unused bandwidth along the path, and the sender and all of the routers along the path approve the Quick-Start Request.</t><t> This document describes many paths where Quick-Start Requests would not be approved. These paths include all paths containing routers, IP tunnels, MPLS paths, and the like that do not support Quick- Start. These paths also include paths with routers or middleboxes that drop packets containing IP options. Quick-Start Requests could be difficult to approve over paths that include multi-access layer- two networks. This document also describes environments where the Quick-Start process could fail with false positives, with the sender incorrectly assuming that the Quick-Start Request had been approved by all of the routers along the path. As a result of these concerns, and as a result of the difficulties and seeming absence of motivation for routers, such as core routers to deploy Quick-Start, Quick-Start is being proposed as a mechanism that could be of use in controlled environments, and not as a mechanism that would be intended or appropriate for ubiquitous deployment in the global Internet. This memo defines an Experimental Protocol for the Internet community. The proposal in this document is Experimental. While it may be deployed in the current Internet, it does not represent a consensus that this is the best possible mechanism for the use of Explicit Congestion Notification (ECN) in TCP SYN/ACK packets.</t><t> This document describes an optional, experimental modification to RFC 3168 to allow TCP SYN/ACK packets to be ECN-Capable. For TCP, RFC 3168 specifies setting an ECN-Capable codepoint on data packets, but not on SYN and SYN/ACK packets. However, because of the high cost to the TCP transfer of having a SYN/ACK packet dropped, with the resulting retransmission timeout, this document describes the use of ECN for the SYN/ACK packet itself, when sent in response to a SYN packet with the two ECN flags set in the TCP header, indicating a willingness to use ECN. Setting the initial TCP SYN/ACK packet as ECN-Capable can be of great benefit to the TCP connection, avoiding the severe penalty of a retransmission timeout for a connection that has not yet started placing a load on the network. The TCP responder (the sender of the SYN/ACK packet) must reply to a report of an ECN-marked SYN/ACK packet by resending a SYN/ACK packet that is not ECN-Capable. If the resent SYN/ACK packet is acknowledged, then the TCP responder reduces its initial congestion window from two, three, or four segments to one segment, thereby reducing the subsequent load from that connection on the network. If instead the SYN/ACK packet is dropped, or for some other reason the TCP responder does not receive an acknowledgement in the specified time, the TCP responder follows TCP standards for a dropped SYN/ACK packet (setting the retransmission timer). This memo defines an Experimental Protocol for the Internet community. This document describes a possible congestion control mechanism for acknowledgement (ACKs) traffic in TCP. The document specifies an end-to-end acknowledgement congestion control mechanism for TCP that uses participation from both TCP hosts: the TCP data sender and the TCP data receiver. The TCP data sender detects lost or Explicit Congestion Notification (ECN)-marked ACK packets, and tells the TCP data receiver the ACK Ratio R to use to respond to the congestion on the reverse path from the data receiver to the data sender. The TCP data receiver sends roughly one ACK packet for every R data packets received. This mechanism is based on the acknowledgement congestion control in the Datagram Congestion Control Protocol's (DCCP's) Congestion Control Identifier (CCID) 2. This acknowledgement congestion control mechanism is being specified for further evaluation by the network community. This document is not an Internet Standards Track specification; it is published for informational purposes. This document proposes an experiment to increase the permitted TCP initial window (IW) from between 2 and 4 segments, as specified in RFC 3390, to 10 segments with a fallback to the existing recommendation when performance issues are detected. It discusses the motivation behind the increase, the advantages and disadvantages of the higher initial window, and presents results from several large-scale experiments showing that the higher initial window improves the overall performance of many web services without resulting in a congestion collapse. The document closes with a discussion of usage and deployment for further experimental purposes recommended by the IETF TCP Maintenance and Minor Extensions (TCPM) working group. This document proposes a new mechanism for TCP and Stream Control Transmission Protocol (SCTP) that can be used to recover lost segments when a connection's congestion window is small. The "Early Retransmit" mechanism allows the transport to reduce, in certain special circumstances, the number of duplicate acknowledgments required to trigger a fast retransmission. This allows the transport to use fast retransmit to recover segment losses that would otherwise require a lengthy retransmission timeout. [STANDARDS-TRACK] Disruptions in end-to-end path connectivity, which last longer than one retransmission timeout, cause suboptimal TCP performance. The reason for this performance degradation is that TCP interprets segment loss induced by long connectivity disruptions as a sign of congestion, resulting in repeated retransmission timer backoffs. This, in turn, leads to a delayed detection of the re-establishment of the connection since TCP waits for the next retransmission timeout before it attempts a retransmission.</t><t> This document proposes an algorithm to make TCP more robust to long connectivity disruptions (TCP-LCD). It describes how standard ICMP messages can be exploited during timeout-based loss recovery to disambiguate true congestion loss from non-congestion loss caused by connectivity disruptions. Moreover, a reversion strategy of the retransmission timer is specified that enables a more prompt detection of whether or not the connectivity to a previously disconnected peer node has been restored. TCP-LCD is a TCP sender- only modification that effectively improves TCP performance in the case of connectivity disruptions. This document defines an Experimental Protocol for the Internet community. This document describes an experimental Proportional Rate Reduction (PRR) algorithm as an alternative to the widely deployed Fast Recovery and Rate-Halving algorithms. These algorithms determine the amount of data sent by TCP during loss recovery. PRR minimizes excess window adjustments, and the actual window size at the end of recovery will be as close as possible to the ssthresh, as determined by the congestion control algorithm. The Eifel detection algorithm allows a TCP sender to detect a posteriori whether it has entered loss recovery unnecessarily. It requires that the TCP Timestamps option defined in RFC 1323 be enabled for a connection. The Eifel detection algorithm makes use of the fact that the TCP Timestamps option eliminates the retransmission ambiguity in TCP. Based on the timestamp of the first acceptable ACK that arrives during loss recovery, it decides whether loss recovery was entered unnecessarily. The Eifel detection algorithm provides a basis for future TCP enhancements. This includes response algorithms to back out of loss recovery by restoring a TCP sender's congestion control state. This memo defines an Experimental Protocol for the Internet community. TCP and Stream Control Transmission Protocol (SCTP) provide notification of duplicate segment receipt through Duplicate Selective Acknowledgement (DSACKs) and Duplicate Transmission Sequence Number (TSN) notification, respectively. This document presents conservative methods of using this information to identify unnecessary retransmissions for various applications. This memo defines an Experimental Protocol for the Internet community. This document specifies Non-Congestion Robustness (NCR) for TCP. In the absence of explicit congestion notification from the network, TCP uses loss as an indication of congestion. One of the ways TCP detects loss is using the arrival of three duplicate acknowledgments. However, this heuristic is not always correct, notably in the case when network paths reorder segments (for whatever reason), resulting in degraded performance. TCP-NCR is designed to mitigate this degraded performance by increasing the number of duplicate acknowledgments required to trigger loss recovery, based on the current state of the connection, in an effort to better disambiguate true segment loss from segment reordering. This document specifies the changes to TCP, as well as the costs and benefits of these modifications. This memo defines an Experimental Protocol for the Internet community. The TCP user timeout controls how long transmitted data may remain unacknowledged before a connection is forcefully closed. It is a local, per-connection parameter. This document specifies a new TCP option -- the TCP User Timeout Option -- that allows one end of a TCP connection to advertise its current user timeout value. This information provides advice to the other end of the TCP connection to adapt its user timeout accordingly. Increasing the user timeouts on both ends of a TCP connection allows it to survive extended periods without end-to-end connectivity. Decreasing the user timeouts allows busy servers to explicitly notify their clients that they will maintain the connection state only for a short time without connectivity. [STANDARDS-TRACK] Often endpoints are connected by multiple paths, but communications are usually restricted to a single path per connection. Resource usage within the network would be more efficient were it possible for these multiple paths to be used concurrently. Multipath TCP is a proposal to achieve multipath transport in TCP.</t><t> New congestion control algorithms are needed for multipath transport protocols such as Multipath TCP, as single path algorithms have a series of issues in the multipath context. One of the prominent problems is that running existing algorithms such as standard TCP independently on each path would give the multipath flow more than its fair share at a bottleneck link traversed by more than one of its subflows. Further, it is desirable that a source with multiple paths available will transfer more traffic using the least congested of the paths, achieving a property called "resource pooling" where a bundle of links effectively behaves like one shared link with bigger capacity. This would increase the overall efficiency of the network and also its robustness to failure.</t><t> This document presents a congestion control algorithm that couples the congestion control algorithms running on different subflows by linking their increase functions, and dynamically controls the overall aggressiveness of the multipath flow. The result is a practical algorithm that is fair to TCP at bottlenecks while moving traffic away from congested links. This document defines an Experimental Protocol for the Internet community. TCP/IP communication is currently restricted to a single path per connection, yet multiple paths often exist between peers. The simultaneous use of these multiple paths for a TCP/IP session would improve resource usage within the network and, thus, improve user experience through higher throughput and improved resilience to network failure.</t><t> Multipath TCP provides the ability to simultaneously use multiple paths between peers. This document presents a set of extensions to traditional TCP to support multipath operation. The protocol offers the same type of service to applications as TCP (i.e., reliable bytestream), and it provides the components necessary to establish and use multiple TCP flows across potentially disjoint paths. This document defines an Experimental Protocol for the Internet community. Harvard University

Cambridge MA 02138 US +1 617 495 3864 sob@harvard.edu

ACIRI / ICSI

1947 Center Street Suite 600 Berkeley CA 94704-1198 US +1 510 666 2882 vern@aciri.org

This memo provides guidance for the IANA to use in assigning parameters for fields in the IPv4, IPv6, ICMP, UDP and TCP protocol headers. When experimenting with or extending protocols, it is often necessary to use some sort of protocol number or constant in order to actually test or experiment with the new function, even when testing in a closed environment. This document reserves some ranges of numbers for experimentation purposes in specific protocols where the need to support experimentation has been identified, and it describes the numbers that have already been reserved by other documents. [STANDARDS-TRACK] This document defines the procedures that the Internet Assigned Numbers Authority (IANA) uses when handling assignment and other requests related to the Service Name and Transport Protocol Port Number registry. It also discusses the rationale and principles behind these procedures and how they facilitate the long-term sustainability of the registry.</t><t> This document updates IANA's procedures by obsoleting the previous UDP and TCP port assignment procedures defined in Sections 8 and 9.1 of the IANA Allocation Guidelines, and it updates the IANA service name and port assignment procedures for UDP-Lite, the Datagram Congestion Control Protocol (DCCP), and the Stream Control Transmission Protocol (SCTP). It also updates the DNS SRV specification to clarify what a service name is and how it is registered. This memo documents an Internet Best Current Practice. This document describes how the experimental TCP option codepoints can concurrently support multiple TCP extensions, even within the same connection, using a new IANA TCP experiment identifier. This approach is robust to experiments that are not registered and to those that do not use this sharing mechanism. It is recommended for all new TCP options that use these codepoints. Stanford Research Institute (SRI) International, Augmentation Research Center (ARC) This note addresses the problem of implementing a reliable out-of-band signal for use in a host-to-host protocol. It is motivated by the fact that such a satisfactory mechanism does not exist in the Transmission Control Protocol (TCP) of Cerf et. al.,. In addition to discussing some requirements for such an out-of-band signal (interrupts) and the implications for the implementation of the requirements, a discussion of the problem for the TCP case will be presented. While the ARPANET host-to-host protocol does not support reliable transmission of either data or controls, it does meet the other requirements we have for an out-of-band control signal and will be drawn upon to provide a solution for the TCP case. SRI-NIC Tandem

950 Linden #208 Sunnyvale CA 94086 US rfox@tandem.com

Two extensions to the TCP protocol are described in this RFC in order to provide a more efficient operation over a network with a high bandwidth*delay product. The main issue that still needs to be solved is congestion versus noise. This issue is touched on in this memo, but further research is still needed on the applicability of the extensions in the Internet as a whole infrastructure and not just high bandwidth*delay product networks. Even with this outstanding issue, this document does describe the use of these options in the isolated satellite network environment to help facilitate more efficient use of this special medium to help off load bulk data transfers from links needed for interactive use. Bolt Beranek and Newman Inc. (BBN)

10 Moulton Street Cambridge MA 02238 US +1 617 873 2962 MCKENZIE@BBN.COM

The TCP Big Window option discussed in RFC 1106 will not work properly in an Internet environment which has both a high bandwidth * delay product and the possibility of disordering and duplicating packets. In such networks, the window size must not be increased without a similar increase in the sequence number space. Therefore, a different approach to big windows should be taken in the Internet. University of Illinois, Digital Computer Lab

1304 West Springfield Avenue CAMPUS MC 258 Urbana IL 61801 US +1 217 333 7937 zweig@CS.UIUC.EDU

Bolt Beranek and Newman Inc.

50 Moulton Street Cambridge MA 02138 US +1 617 873 2459 craig@BBN.COM

University of Arizona, Department of Computer Sciences

Tucson AZ 85721 US +1 602 621 8373 sean@cs.arizona.edu

University of Arizona, Department of Computer Sciences

Tucson AZ 85721 US +1 602 621 4231 llp@cs.arizona.edu

This RFC comments on recent proposals to extend TCP. It argues that the backward compatible extensions proposed in RFC's 1072 and 1185 should not be pursued, and proposes an alternative way to evolve the Internet protocol suite. Its purpose is to stimulate discussion in the Internet community. University of Southern California, Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292 US +1 310 822 1511 Braden@ISI.EDU

This memo discusses extension of TCP to provide transaction-oriented service, without altering its virtual-circuit operation. This extension would fill the large gap between connection-oriented TCP and datagram-based UDP, allowing TCP to efficiently perform many applications for which UDP is currently used. A separate memo contains a detailed functional specification for this proposed extension. This work was supported in part by the National Science Foundation under Grant Number NCR-8922231. University of Southern California, Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292 US +1 310 822 1511 Braden@ISI.EDU

This memo specifies T/TCP, an experimental TCP extension for efficient transaction-oriented (request/response) service. This backwards-compatible extension could fill the gap between the current connection-oriented TCP and the datagram-based UDP. This work was supported in part by the National Science Foundation under Grant Number NCR-8922231. University of Delaware, Department of Computer & Information

101C Smith Hall Newark DE 19716-2586 US connolly@udel.edu

University of Delaware, Department of Computer & Information

101C Smith Hall Newark DE 19716-2586 US amer@udel.edu

University of Delaware, Department of Computer & Information

101C Smith Hall Newark DE 19716-2586 US pconrad@udel.edu

This RFC introduces a new transport mechanism for TCP based upon partial ordering. The aim is to present the concepts of partial ordering and promote discussions on its usefulness in network communications. Distribution of this memo is unlimited. Argonne National Laboratory, Electronics and Computing Technologies

Argonne IL 60439 US +1 708 252 7289 RACarlson@anl.gov

Motorola

+1 708 632 4029 ficarell@cpdmfg.cig.mot.com

This document was submitted to the IETF IPng area in response to RFC 1550. Publication of this document does not imply acceptance by the IPng area of any ideas expressed within. Comments should be submitted to the big-internet@munnari.oz.au mailing list. TCP Cookie Transactions (TCPCT) deter spoofing of connections and prevent resource exhaustion, eliminating Responder (server) state during the initial handshake. The Initiator (client) has sole responsibility for ensuring required delays between connections. The cookie exchange may carry data, limited to inhibit amplification and reflection denial of service attacks. This document defines an Experimental Protocol for the Internet community. This document reclassifies several TCP extensions that have never seen widespread use to Historic status. The affected RFCs are RFC 1072, RFC 1106, RFC 1110, RFC 1145, RFC 1146, RFC 1379, RFC 1644, and RFC 1693. This document is not an Internet Standards Track specification; it is published for informational purposes. University of Southern California (USC)/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90291 US

Massachusetts Institute of Technology (MIT), Laboratory for Computer Science, Computer Systems and Communications Group Massachusetts Institute of Technology (MIT), Laboratory for Computer Science, Computer Systems and Communications Group Massachusetts Institute of Technology (MIT), Laboratory for Computer Science, Computer Systems and Communications Group Massachusetts Institute of Technology (MIT), Laboratory for Computer Science, Computer Systems and Communications Group The Mitre Corporation

P.O. Box 208 Bedford MA 01730 US Padlipsky@USC-ISIA

The sometimes-held position that the DoD Standard Transmission Control Protocol (TCP) and Internet Protocol (IP) are inappropriate for use "on" a Local Area Network (LAN) is shown to be fallacious. The paper is a companion piece to M82-47, M82-49, M82-50, and M82-51. Ford Aerospace and Communications Corporation This memo discusses some aspects of congestion control in IP/TCP Internetworks. It is intended to stimulate thought and further discussion of this topic. While some specific suggestions are made for improved congestion control implementation, this memo does not specify any standards. SDC - A Burroughs Company SDC - A Burroughs Company This note points out three errors with the specification of the Military Standard Transmission Control Protocol (MIL-STD-1778, dated August 1983). These results are based on an initial investigation of this protocol standard. The first problem is that data accompanying a SYN can not be accepted because of errors in the acceptance policy. The second problem is that no retransmission timer is set for a SYN packet, and therefore the SYN will not be retransmitted if it is lost. The third problem is that when the connection has been established, neither entity takes the proper steps to accept incoming data. This note also proposes solutions to these problems. CERN, European Laboratory for Particle Physics

1211 Geneva 23 CH +41 22 7674967 +41 22 7677155 brian@dxcoms.cern.ch

The Internet and its architecture have grown in evolutionary fashion from modest beginnings, rather than from a Grand Plan. While this process of evolution is one of the main reasons for the technology's success, it nevertheless seems useful to record a snapshot of the current principles of the Internet architecture. This is intended for general guidance and general interest, and is in no way intended to be a formal or invariant reference model. The goal of this document is to explain the need for congestion control in the Internet, and to discuss what constitutes correct congestion control. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document extends RFC 1958 by outlining some of the philosophical guidelines to which architects and designers of Internet backbone networks should adhere. We describe the Simplicity Principle, which states that complexity is the primary mechanism that impedes efficient scaling, and discuss its implications on the architecture, design and engineering issues found in large scale Internet backbones. This memo provides information for the Internet community. There have been a number of proposals for alternate semantics for the Explicit Congestion Notification (ECN) field in the IP header RFC 3168. This document discusses some of the issues in defining alternate semantics for the ECN field, and specifies requirements for a safe coexistence in an Internet that could include routers that do not understand the defined alternate semantics. This document evolved as a result of discussions with the authors of one recent proposal for such alternate semantics. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Hosts are often connected by multiple paths, but TCP restricts communications to a single path per transport connection. Resource usage within the network would be more efficient were these multiple paths able to be used concurrently. This should enhance user experience through improved resilience to network failure and higher throughput.</t><t> This document outlines architectural guidelines for the development of a Multipath Transport Protocol, with references to how these architectural components come together in the development of a Multipath TCP (MPTCP). This document lists certain high-level design decisions that provide foundations for the design of the MPTCP protocol, based upon these architectural requirements. This document is not an Internet Standards Track specification; it is published for informational purposes. NASA Lewis Research Center/Sterling Software

21000 Brookpark Rd., MS 54-2 Cleveland OH 44135 USA +1 216 433 6586 mallman@lerc.nasa.gov http://roland.lerc.nasa.gov/~mallman

NASA Lewis Research Center

Cleveland OH 44135 USA +1 216 433 2847 Daniel.R.Glover@lerc.nasa.gov

BBN Technologies, GTE Internetworking

10 Moulton Street Cambridge MA 02140 USA +1 617 873 3351 lsanchez@ir.bbn.com

Transport transmission control protocol satellite TCP The Transmission Control Protocol (TCP) provides reliable delivery of data across any network path, including network paths containing satellite channels. While TCP works over satellite channels there are several IETF standardized mechanisms that enable TCP to more effectively utilize the available capacity of the network path. This document outlines some of these TCP mitigations. At this time, all mitigations discussed in this document are IETF standards track mechanisms (or are compliant with IETF standards). Sun Microsystems, Inc., Sun Labs Networking and Security Group

901 San Antonio Road Mailstop UMPK 15-214 Palo Alto CA 94303 US +1 650 786 6288 +1 650 786 6445 gab@sun.com

Nortel Networks

P.O. Box 833805 Richardson TX 75083-3805 US +1 972 684 4827 +1 972 685 3292 sdawkins@nortel.com

University of Helsinki, Department of Computer Science

P.O. Box 26 (Teollisuuskatu 23) Helsinki FIN-00014 FI +358 9 19144179 +358 9 19144441 kojo@cs.helsinki.fi

Alcatel Network Systems, Inc, Corporate Research Center

1201 Campbell M/S 446-310 Richardson TX 75081 US +1 972 996 2625 +1 972 996 5902 vincent.magret@aud.alcatel.com

Texas A&M University, Dept. of Computer Science

College Station TX 77843-3112 US +1 979 845 0512 +1 979 847 8578 vaidya@cs.tamu.edu

In view of the unpredictable and problematic nature of long thin networks (for example, wireless WANs), arriving at an optimized transport is a daunting task. We have reviewed the existing proposals along with future research items. Based on this overview, we also recommend mechanisms for implementation in long thin networks. Our goal is to identify a TCP that works for all users, including users of long thin networks. We started from the working recommendations of the IETF TCP Over Satellite Links (tcpsat) working group with this end in mind. We recognize that not every tcpsat recommendation will be required for long thin networks as well, and work toward a set of TCP recommendations that are 'benign' in environments that do not require them. NASA Glenn Research Center/BBN Technologies, Lewis Field

21000 Brookpark Rd. MS 54-2 Cleveland OH 44135 US mallman@grc.nasa.gov

Nortel

P.O.Box 833805 Richardson TX 75083-3805 US Spencer.Dawkins.sdawkins@nt.com

NASA Glenn Research Center, Lewis Field

21000 Brookpark Rd. MS 3-6 Cleveland OH 44135 US Daniel.R.Glover@grc.nasa.gov

NASA Glenn Research Center, Lewis Field

21000 Brookpark Rd. MS 54-2 Cleveland OH 44135 US jgriner@grc.nasa.gov

NASA Glenn Research Center, Lewis Field

21000 Brookpark Rd. MS 54-2 Cleveland OH 44135 US dtran@grc.nasa.gov

University of California at Berkeley

Berkeley CA 94720 US +1 510 642 8919 tomh@cs.berkeley.edu

University of Southern California/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292-6695 US johnh@isi.edu

University of Southern California/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292-6601 US +1 310 448 9151 +1 310 823 6714 touch@isi.edu

Ohio University, J. Warren McClure School of Communication Systems Management

9 S. College Street Athens OH 45701 US +1 740 593 4891 +1 740 593 4889 hkruse1@ohiou.edu

Ohio University, School of Electrical Engineering and Computer Science

416 Morton Hall Athens OH 45701 US +1 740 593 1234 ostermann@cs.ohiou.edu

The MITRE Corporation

1820 Dolley Madison Blvd. M/S W650 McLean VA 22102-3481 US kscott@mitre.org

Pittsburgh Supercomputing Center

4400 Fifth Ave. Pittsburgh PA 15213 US semke@psc.edu

This document outlines possible TCP enhancements that may allow TCP to better utilize the available bandwidth provided by networks containing satellite links. The algorithms and mechanisms outlined have not been judged to be mature enough to be recommended by the IETF. The goal of this document is to educate researchers as to the current work and progress being done in TCP research related to satellite networks. This document is a survey of Performance Enhancing Proxies (PEPs) often employed to improve degraded TCP performance caused by characteristics of specific link environments, for example, in satellite, wireless WAN, and wireless LAN environments. This memo provides information for the Internet community. This document makes performance-related recommendations for users of network paths that traverse "very low bit-rate" links. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document discusses the specific TCP mechanisms that are problematic in environments with high uncorrected error rates, and discusses what can be done to mitigate the problems without introducing intermediate devices into the connection. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document describes TCP performance problems that arise because of asymmetric effects. These problems arise in several access networks, including bandwidth-asymmetric networks and packet radio subnetworks, for different underlying reasons. However, the end result on TCP performance is the same in both cases: performance often degrades significantly because of imperfection and variability in the ACK feedback from the receiver to the sender. The document details several mitigations to these effects, which have either been proposed or evaluated in the literature, or are currently deployed in networks. These solutions use a combination of local link- layer techniques, subnetwork, and end-to-end mechanisms, consisting of: (i) techniques to manage the channel used for the upstream bottleneck link carrying the ACKs, typically using header compression or reducing the frequency of TCP ACKs, (ii) techniques to handle this reduced ACK frequency to retain the TCP sender's acknowledgment-triggered self- clocking and (iii) techniques to schedule the data and ACK packets in the reverse direction to improve performance in the presence of two-way traffic. Each technique is described, together with known issues, and recommendations for use. A summary of the recommendations is provided at the end of the document. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document describes a profile for optimizing TCP to adapt so that it handles paths including second (2.5G) and third (3G) generation wireless networks. It describes the relevant characteristics of 2.5G and 3G networks, and specific features of example deployments of such networks. It then recommends TCP algorithm choices for nodes known to be starting or ending on such paths, and it also discusses open issues. The configuration options recommended in this document are commonly found in modern TCP stacks, and are widely available standards-track mechanisms that the community considers safe for use on the general Internet. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. 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. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. The IETF's standard congestion control schemes have been widely shown to be inadequate for various environments (e.g., high-speed networks). Recent research has yielded many alternate congestion control schemes that significantly differ from the IETF's congestion control principles. Using these new congestion control schemes in the global Internet has possible ramifications to both the traffic using the new congestion control and to traffic using the currently standardized congestion control. Therefore, the IETF must proceed with caution when dealing with alternate congestion control proposals. The goal of this document is to provide guidance for considering alternate congestion control algorithms within the IETF. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document discusses the metrics to be considered in an evaluation of new or modified congestion control mechanisms for the Internet. These include metrics for the evaluation of new transport protocols, of proposed modifications to TCP, of application-level congestion control, and of Active Queue Management (AQM) mechanisms in the router. This document is the first in a series of documents aimed at improving the models that we use in the evaluation of transport protocols.</t><t> This document is a product of the Transport Modeling Research Group (TMRG), and has received detailed feedback from many members of the Research Group (RG). As the document tries to make clear, there is not necessarily a consensus within the research community (or the IETF community, the vendor community, the operations community, or any other community) about the metrics that congestion control mechanisms should be designed to optimize, in terms of trade-offs between throughput and delay, fairness between competing flows, and the like. However, we believe that there is a clear consensus that congestion control mechanisms should be evaluated in terms of trade-offs between a range of metrics, rather than in terms of optimizing for a single metric. This memo provides information for the Internet community. Multipath TCP (MPTCP for short) describes the extensions proposed for TCP so that endpoints of a given TCP connection can use multiple paths to exchange data. Such extensions enable the exchange of segments using different source-destination address pairs, resulting in the capability of using multiple paths in a significant number of scenarios. Some level of multihoming and mobility support can be achieved through these extensions. However, the support for multiple IP addresses per endpoint may have implications on the security of the resulting MPTCP. This note includes a threat analysis for MPTCP. This document is not an Internet Standards Track specification; it is published for informational purposes. This framework describes a practical methodology for measuring end- to-end TCP Throughput in a managed IP network. The goal is to provide a better indication in regard to user experience. In this framework, TCP and IP parameters are specified to optimize TCP Throughput. This document is not an Internet Standards Track specification; it is published for informational purposes. Advanced Research Projects Agency (ARPA) In general, precedence is a tool for deciding how to allocate resources when systems are saturated. In circuit switched systems, the resource is circuits; in message switched systems the resource is the message switch processor; and in packet switching the resource is the packet switching system itself. This capability can be realized in AUTODIN II without adding any new mechanisms to TCP (except to make precedence of incoming connection requests visible to the processes which use TCP). To allow pre-emptive access to a particular terminal, the software (i.e., THP) which supports terminal access to the TAC can be configured so as to always have a LISTEN posted for that terminal, even if the terminal has a connection in operation. For example in the ARPANET TENEX systems, the user TELNET permits a user to have many connections open at one time - the user can switch among them at will. To the extent that this can be done without violating security requirements, one could imagine a multi-connection THP which always leaves a LISTEN pending for incoming connection requests. If a connection is established, the THP can decide, based on its precedence, whether to pre-empt any existing connection and to switch the user to the high precedence one. University of Southern California (USC)/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90291 US +1 213 822 1511

This memo discusses the TCP Maximum Segment Size Option and related topics. The purposes is to clarify some aspects of TCP and its interaction with IP. This memo is a clarification to the TCP specification, and contains information that may be considered as "advice to implementers". USC/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90291 US +1 213 822 1511

Cray Research Bolt Baranek and Newman (BBN) Laboratories Bolt Beranek and Newman, Inc. (BBN)

50 Moulton Street Cambridge MA 02138 US

Digital Equipment Corporation

550 King St Littleton MA 01460 US +1 508 486 6786 anil@levers.enet.dec.com

This memo describes an updated technique for incremental computation of the standard Internet checksum. It updates the method described in RFC 1141. University of Southern California, Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292-6695 US +1 310 822 1511 x151 +1 310 823 6714 touch@isi.edu http://www.isi.edu/~touch

University of Southern California, Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292-6695 US +1 310 822 1511 x101 +1 310 823 6714 bparham@isi.edu

This memo presents a techniques for efficiently implementing the Internet checksum in hardware. It includes PLD code for programming a single, low cost part to perform checksumming at 1.26 Gbps. ACIRI / ICSI

1947 Center Street Suite 600 Berkeley CA 94704-1198 US +1 510 642 4274 x302 vern@aciri.org

University of Michigan

Real-Time Computing Laboratory EECS Building Ann Arbor MI 48109-2122 US +1 313 763 5363 sdawson@eecs.umich.edu

Xerox PARC

3333 Coyote Hill Road Palo Alto CA 94304 US +1 650 812 4816 fenner@parc.xerox.com

NASA Glenn Research Center

Lewis Field 21000 Brookpark Road MS 54-2 Cleveland OH 44135 US +1 216 433 5787 jgriner@grc.nasa.gov

Spider Software Ltd.

8 John's Place Leith Edinburgh Scotland EH6 7EL UK +44 131 4757015 ian@spider.com

NASA Ames Research Center/MRJ

MS 258-6 Moffett Field CA 94035 US +1 650 604 4334 kml@nas.nasa.gov

Pittsburgh Supercomputing Center

4400 Fifth Ave Pittsburgh PA 15213 US +1 412 268 4960 semke@psc.edu

Process Software Corporation

959 Concord Street Framingham MA 01701 US +1 508 879 6994 volz@process.com

This memo catalogs several known Transmission Control Protocol (TCP) implementation problems dealing with Path Maximum Transmission Unit Discovery (PMTUD), including the long-standing black hole problem, stretch acknowlegements (ACKs) due to confusion between Maximum Segment Size (MSS) and segment size, and MSS advertisement based on PMTU. This memo provides information for the Internet community. The de facto standard Application Program Interface (API) for TCP/IP applications is the "sockets" interface. Although this API was developed for Unix in the early 1980s it has also been implemented on a wide variety of non-Unix systems. TCP/IP applications written using the sockets API have in the past enjoyed a high degree of portability and we would like the same portability with IPv6 applications. But changes are required to the sockets API to support IPv6 and this memo describes these changes. These include a new socket address structure to carry IPv6 addresses, new address conversion functions, and some new socket options. These extensions are designed to provide access to the basic IPv6 features required by TCP and UDP applications, including multicasting, while introducing a minimum of change into the system and providing complete compatibility for existing IPv4 applications. Additional extensions for advanced IPv6 features (raw sockets and access to the IPv6 extension headers) are defined in another document. This memo provides information for the Internet community. During the last few years, awareness has been raised about a number of "blind" attacks that can be performed against the Transmission Control Protocol (TCP) and similar protocols. The consequences of these attacks range from throughput reduction to broken connections or data corruption. These attacks rely on the attacker's ability to guess or know the five-tuple (Protocol, Source Address, Destination Address, Source Port, Destination Port) that identifies the transport protocol instance to be attacked. This document describes a number of simple and efficient methods for the selection of the client port number, such that the possibility of an attacker guessing the exact value is reduced. While this is not a replacement for cryptographic methods for protecting the transport-protocol instance, the aforementioned port selection algorithms provide improved security with very little effort and without any key management overhead. The algorithms described in this document are local policies that may be incrementally deployed and that do not violate the specifications of any of the transport protocols that may benefit from them, such as TCP, UDP, UDP-lite, Stream Control Transmission Protocol (SCTP), Datagram Congestion Control Protocol (DCCP), and RTP (provided that the RTP application explicitly signals the RTP and RTCP port numbers). This memo documents an Internet Best Current Practice. This document describes an algorithm for processing incoming SYN segments that allows higher connection-establishment rates between any two TCP endpoints when a TCP Timestamps option is present in the incoming SYN segment. This document only modifies processing of SYN segments received for connections in the TIME-WAIT state; processing in all other states is unchanged. This memo documents an Internet Best Current Practice. This document clarifies the Zero Window Probes (ZWPs) described in RFC 1122 ("Requirements for Internet Hosts -- Communication Layers"). In particular, it clarifies the actions that can be taken on connections that are experiencing the ZWP condition. Rather than making a change to the standard, this document clarifies what has been until now a misinterpretation of the standard as specified in RFC 1122. This document is not an Internet Standards Track specification; it is published for informational purposes. Multipath TCP (MPTCP) adds the capability of using multiple paths to a regular TCP session. Even though it is designed to be totally backward compatible to applications, the data transport differs compared to regular TCP, and there are several additional degrees of freedom that applications may wish to exploit. This document summarizes the impact that MPTCP may have on applications, such as changes in performance. Furthermore, it discusses compatibility issues of MPTCP in combination with non-MPTCP-aware applications. Finally, the document describes a basic application interface that is a simple extension of TCP's interface for MPTCP-aware applications. Hughes LAN Systems, The Wollongong Group

1129 San Antonio Road Palo Alto CA 94303 US +1 415 962 7160 sytek!kzm@HPLABS.HP.COM

Performance Systems International, Inc. (PSI), PSI California Office

P.O. Box 391776 Mountain View CA 94039 US +1 415 961 3380 mrose@PSI.COM

Hughes, LAN Systems

1225 Charleston Road Mountain View CA 94043 US +1 415 966 7934 kzm@hls.com

Performance Systems International

5201 Great America Parkway Suite 3106 Santa Clara CA 95054 US +1 408 562 6222 mrose@psi.com

This memo defines the second version of the Management Information Base (MIB-II) for use with network management protocols in TCP/IP-based internets. In particular, together with its companion memos which describe the structure of management information (RFC 1155) along with the network management protocol (RFC 1157) for TCP/IP-based internets, these documents provide a simple, workable architecture and system for managing TCP/IP-based internets and in particular the Internet community. Cisco Systems, Inc.

170 West Tasman Drive San Jose CA 95134-1706 US +1 408 526 5260 kzm@cisco.com

Transport Management TCP Management Information Base control protocol simple network management protocol version 2 structure of management information version 2 transmission control protocol The IP, UDP, and TCP MIB modules currently support only IPv4. These three modules use the IpAddress type defined as an OCTET STRING of length 4 to represent the IPv4 32-bit internet addresses. (See RFC 1902, SMI for SNMPv2.) They do not support the new 128-bit IPv6 internet addresses. Compaq Computer Corporation

110 Spit Brook Rd Nashua NH 03062 USA +1 603 884 1423 daniele@zk3.dec.com

Transport Management Management Information Base control protocol internet protocol version 6 transmission control protocol This document is one in the series of documents that define various MIB objects for IPv6. Specifically, this document is the MIB module which defines managed objects for implementations of the Transmission Control Protocol (TCP) over IP Version 6 (IPv6). This document also recommends a specific policy with respect to the applicability of RFC 2012 for implementations of IPv6. Namely, that most of managed objects defined in RFC 2012 are independent of which IP versions underlie TCP, and only the TCP connection information is IP version-specific. This memo defines an experimental portion of the Management Information Base (MIB) for use with network management protocols in IPv6-based internets. This memo defines a portion of the Management Information Base (MIB) for use with network management protocols in the Internet community. In particular, it describes managed objects used for implementations of the Transmission Control Protocol (TCP) in an IP version independent manner. This memo obsoletes RFCs 2452 and 2012. [STANDARDS-TRACK] This document describes extended performance statistics for TCP. They are designed to use TCP's ideal vantage point to diagnose performance problems in both the network and the application. If a network-based application is performing poorly, TCP can determine if the bottleneck is in the sender, the receiver, or the network itself. If the bottleneck is in the network, TCP can provide specific information about its nature. [STANDARDS-TRACK] Spider Systems Limited

Stanwell Street Spider Park Edinburgh Scotland EH6 5NG UK +44 31 5549424 +44 31 5540649 TEDS@SPIDER.CO.UK

12 Gosford Place Edinburgh England EH6 4BJ UK +44 31 5547432 CLAUDIAK@SPIDER.CO.UK

Advanced Network and Services

1875 Campus Commons Drive Suite 220 Reston VA 22091-1552 US +1 703 758 7722 enger@reston.ans.net

University of Southern California, Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292 US +1 310 822 1511 JKREY@ISI.EDU

The goal of this FYI memo is to provide an update to FYI 2, RFC 1147, which provided practical information to site administrators and network managers. New and/or updated tools are listed in this RFC. Additonal descriptions are welcome, and should be sent to: noctools- entries@merit.edu. Sun Microsystems, Inc.

901 San Antonio Road UMPK17-202 Palo Alto CA 94043 USA (650) 786-5176 sparker@eng.sun.com

Sun Microsystems, Inc.

901 San Antonio Road UMPK17-202 Palo Alto 94043 USA (650) 786-4053 cschmec@eng.sun.com

Transport tool transmission control protocol This document is an informational snapshot taken by the IRTF\'s Internet Congestion Control Research Group (ICCRG) in October 2008. It provides a survey of congestion control topics described by documents in the RFC series. This does not modify or update the specifications or status of the RFC documents that are discussed. It may be used as a reference or starting point for the future work of the research group, especially in noting gaps or open issues in the current IETF standards. This document is not an Internet Standards Track specification; it is published for informational purposes. This document describes some of the open problems in Internet congestion control that are known today. This includes several new challenges that are becoming important as the network grows, as well as some issues that have been known for many years. These challenges are generally considered to be open research topics that may require more study or application of innovative techniques before Internet-scale solutions can be confidently engineered and deployed. This document is not an Internet Standards Track specification; it is published for informational purposes. This memo reports on some measurement experiments and suggests some possible improvements to the TCP retransmission timeout calculation. This memo is both a status report on the measurements and advice to implementers of TCP. University of Southern California (USC), Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90292 US +1 213 822 1511 EMail: Braden@ISI.EDU

This note describes some theoretically-possible failure modes for TCP connections and discusses possible remedies. In particular, one very simple fix is identified. Bay Networks

4401 Great America Parkway SC01-04 Santa Clara CA 95052-8185 +1-408-495-2463 +1-408-495-1299 knichols@baynetworks.com @Baynetworks.com

Transport file transfer transmission control protocol TCP An increase in the permissible initial window size of a TCP connection, from one segment to three or four segments, has been under discussion in the tcp-impl working group. This document covers some simulation studies of the effects of increasing the initial window size of TCP. Both long-lived TCP connections (file transfers) and short-lived web-browsing style connections were modeled. The simulations were performed using the publicly available ns-2 simulator and our custom models and files are also available. BBN Technologies

10 Moulton Street Cambridge MA 02138 USA shep@alum.mit.edu

BBN Technologies

10 Moulton Street Cambridge MA 02138 USA craig@bbn.com

Transport TCP transmission control protocol This memo is to document a simple experiment. The experiment showed that in the case of a TCP receiver behind a 9600 bps modem link at the edge of a fast Internet where there are only 3 buffers before the modem (and the fourth packet of a four-packet start will surely be dropped), no significant degradation in performance is experienced by a TCP sending with a four-packet start when compared with a normal slow start (which starts with just one packet). This memo presents a performance study of the Explicit Congestion Notification (ECN) mechanism in the TCP/IP protocol using our implementation on the Linux Operating System. This memo provides information for the Internet community. USC/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90291 US +1 213 822 1511

USC/Information Sciences Institute

4676 Admiralty Way Marina del Rey CA 90291 US +1 213 822 1511

Harvard University

1350 Mass. Ave. Cambridge MA 02138 US +1 617 495 3864 sob@harvard.edu

This memo documents the process used by the Internet community for the standardization of protocols and procedures. It defines the stages in the standardization process, the requirements for moving a document between stages and the types of documents used during this process. It also addresses the intellectual property rights and copyright issues associated with the standards process. Harvard University

1350 Mass. Ave. Cambridge MA 02138 - +1 617 495 3864 sob@harvard.edu

General keyword In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. Authors who follow these guidelines should incorporate this phrase near the beginning of their document: The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119. Note that the force of these words is modified by the requirement level of the document in which they are used. Cisco Systems

170 West Tasman Drive San Jose CA 95134-1706 USA +1 408 525 4857 kmn@cisco.com

Torrent Networking Technologies

3000 Aerial Center Morrisville NC 27560 USA +1 919 468 8466 x232 slblake@torrentnet.com

Cisco Systems

519 Lado Drive Santa Barbara CA 93111 USA +1 408 526 4257 fred@cisco.com

EMC Corporation

35 Parkwood Drive Hopkinton MA 01748 USA +1 508 435 1000 x76140 black_david@emc.com

Internet internet protocol version 4 IPv6 IPv4 internet protocol version 6 type of service Differentiated services enhancements to the Internet protocol are intended to enable scalable service discrimination in the Internet without the need for per-flow state and signaling at every hop. A variety of services may be built from a small, well-defined set of building blocks which are deployed in network nodes. The services may be either end-to-end or intra-domain; they include both those that can satisfy quantitative performance requirements (e.g., peak bandwidth) and those based on relative performance (e.g., "class" differentiation). Services can be constructed by a combination of: - setting bits in an IP header field at network boundaries (autonomous system boundaries, internal administrative boundaries, or hosts), - using those bits to determine how packets are forwarded by the nodes inside the network, and - conditioning the marked packets at network boundaries in accordance with the requirements or rules of each service. The requirements or rules of each service must be set through administrative policy mechanisms which are outside the scope of this document. A differentiated services-compliant network node includes a classifier that selects packets based on the value of the DS field, along with buffer management and packet scheduling mechanisms capable of delivering the specific packet forwarding treatment indicated by the DS field value. Setting of the DS field and conditioning of the temporal behavior of marked packets need only be performed at network boundaries and may vary in complexity. This document defines the IP header field, called the DS (for differentiated services) field. In IPv4, it defines the layout of the TOS octet; in IPv6, the Traffic Class octet. In addition, a base set of packet forwarding treatments, or per-hop behaviors, is defined. For a more complete understanding of differentiated services, see also the differentiated services architecture . This memo describes an extension to the Stream Control Transmission Protocol (SCTP) that allows an SCTP endpoint to signal to its peer that it should move the cumulative ack point forward. When both sides of an SCTP association support this extension, it can be used by an SCTP implementation to provide partially reliable data transmission service to an upper layer protocol. This memo describes the protocol extensions, which consist of a new parameter for INIT and INIT ACK, and a new FORWARD TSN chunk type, and provides one example of a partially reliable service that can be provided to the upper layer via this mechanism. [STANDARDS-TRACK] The Datagram Congestion Control Protocol (DCCP) is a transport protocol that provides bidirectional unicast connections of congestion-controlled unreliable datagrams. DCCP is suitable for applications that transfer fairly large amounts of data and that can benefit from control over the tradeoff between timeliness and reliability. [STANDARDS-TRACK] This document contains the profile for Congestion Control Identifier 2 (CCID 2), TCP-like Congestion Control, in the Datagram Congestion Control Protocol (DCCP). CCID 2 should be used by senders who would like to take advantage of the available bandwidth in an environment with rapidly changing conditions, and who are able to adapt to the abrupt changes in the congestion window typical of TCP's Additive Increase Multiplicative Decrease (AIMD) congestion control. [STANDARDS-TRACK] It is commonly recognized that the Internet routing and addressing architecture is facing challenges in scalability, multihoming, and inter-domain traffic engineering. This document presents, as a recommendation of future directions for the IETF, solutions that could aid the future scalability of the Internet. To this end, this document surveys many of the proposals that were brought forward for discussion in this activity, as well as some of the subsequent analysis and the architectural recommendation of the chairs. This document is a product of the Routing Research Group. This document is not an Internet Standards Track specification; it is published for informational purposes. CUBIC is an extension to the current TCP standards. The protocol differs from the current TCP standards only in the congestion window adjustment function in the sender side. In particular, it uses a cubic function instead of a linear window increase of the current TCP standards to improve scalability and stability under fast and long distance networks. BIC-TCP, a predecessor of CUBIC, has been a default TCP adopted by Linux since year 2005 and has already been deployed globally and in use for several years by the Internet community at large. CUBIC is using a similar window growth function as BIC-TCP and is designed to be less aggressive and fairer to TCP in bandwidth usage than BIC-TCP while maintaining the strengths of BIC- TCP such as stability, window scalability and RTT fairness. Through extensive testing in various Internet scenarios, we believe that CUBIC is safe for deployment and testing in the global Internet. The intent of this document is to provide the protocol specification of CUBIC for a third party implementation and solicit the community feedback through experimentation on the performance of CUBIC. We expect this document to be eventually published as an experimental RFC. Compound TCP (CTCP) is a modification to TCP's congestion control mechanism for use with TCP connections with large congestion windows. This document describes the Compound TCP algorithm in detail, and solicits experimentation and feedback from the wider community. The key idea behind CTCP is to add a scalable delay-based component to the standard TCP's loss-based congestion control. The sending rate of CTCP is controlled by both loss and delay components. The delay-based component has a scalable window increasing rule that not only efficiently uses the link capacity, but on sensing queue build up, proactively reduces the sending rate. This document describes a number of changes to the TCP congestion control algorithm to to improve performance in high bandwidth-delay product paths. We focus on changes to the congestion avoidance mode, rather than slow-start. RFC Editor

This document grew out of a discussion on the end2end-interest mailing list, the public list of the End-to-End Research Group of the IRTF, and continued development under the IETF's TCP Maintenance and Minor Extensions (TCPM) working group. We thank Mark Allman, Yuchung Cheng, Ted Faber, Fairhurst, Sally Floyd, Janardhan Iyengar, Reiner Ludwig, Pekka Savola, and Joe Touch for their contributions, in particular. Keith McCloghrie provided some useful notes and clarification on the various MIB-related RFCs.