TCP

Explicit Congestion Notification (ECN) is a mechanism by which network nodes can mark IP packets instead of dropping them to indicate incipient congestion to the endpoints. Receivers with an ECN-capable transport protocol feed back this information to the sender. ECN was originally specified for TCP in such a way that only one feedback signal can be transmitted per Round-Trip Time (RTT). More recently defined mechanisms like Congestion Exposure (ConEx), Data Center TCP (DCTCP), or Low Latency, Low Loss, and Scalable Throughput (L4S) need more accurate ECN feedback information whenever more than one marking is received in one RTT. This document updates the original ECN specification defined in RFC 3168 by specifying a scheme that provides more than one feedback signal per RTT in the TCP header, called More Accurate ECN (AccECN). Given TCP header space is scarce, it allocates a reserved header bit previously assigned to the ECN-nonce. It also overloads the two existing ECN flags in the TCP header. The resulting extra space is additionally exploited to feed back the IP-ECN field received during the TCP connection establishment. Supplementary feedback information can optionally be provided in two new TCP Option alternatives, which are never used on the TCP SYN. The document also specifies the treatment of this updated TCP wire protocol by middleboxes.

This document specifies a Standards Track version of the Proportional Rate Reduction (PRR) algorithm that obsoletes the Experimental version described in RFC 6937. PRR regulates the amount of data sent by TCP or other transport protocols during fast recovery. PRR accurately regulates the actual flight size through recovery such that at the end of recovery it will be as close as possible to the slow start threshold (ssthresh), as determined by the congestion control algorithm.

This document specifies receiver-driven Low Extra Delay Background Transport (rLEDBAT) -- a set of mechanisms that enable the execution of a less-than-best-effort congestion control algorithm for TCP at the receiver end. This document is a product of the Internet Congestion Control Research Group (ICCRG) of the Internet Research Task Force (IRTF).

This document specifies new IP Flow Information Export (IPFIX) Information Elements (IEs) to solve issues with existing ipv6ExtensionHeaders and tcpOptions IPFIX IEs, especially the ability to export any observed IPv6 extension headers or TCP options.

This document defines the SendHoldTimer, along with the SendHoldTimer_Expires event, for the Border Gateway Protocol (BGP) Finite State Machine (FSM). Implementation of the SendHoldTimer helps overcome situations where a BGP connection is not terminated after the local system detects that the remote system is not processing BGP messages. This document specifies that the local system should close the BGP connection and not solely rely on the remote system for connection closure when the SendHoldTimer expires. This document updates RFC 4271.

This document presents three YANG 1.1 modules to support the configuration of TCP clients and TCP servers. The modules include basic parameters of a TCP connection relevant for client or server applications, as well as client configuration required for traversing proxies. The data models defined by these modules may be used directly (e.g., to define a specific TCP client or TCP server) or in conjunction with the configuration defined for higher level protocols that depend on TCP (e.g., SSH, TLS, etc.). Examples of higher level protocol configuration designed to be used in conjunction with this configuration are in RFCs 9644 and 9645.

This document specifies a minimal YANG data model for TCP on devices that are configured and managed by network management protocols. The YANG data model defines a container for all TCP connections and groupings of authentication parameters that can be imported and used in TCP implementations or by other models that need to configure TCP parameters. The model also includes basic TCP statistics. The model is compliant with Network Management Datastore Architecture (NMDA) (RFC 8342).

RFC 7125 revised the tcpControlBits IP Flow Information Export (IPFIX) Information Element that was originally defined in RFC 5102 to reflect changes to the TCP header control bits since RFC 793. However, that update is still problematic for interoperability because some flag values have subsequently been deprecated.

This document removes stale information from the IANA "IPFIX Information Elements" registry and avoids future conflicts with the authoritative IANA "TCP Header Flags" registry.

This document obsoletes RFC 7125.

This document describes HyStart++, a simple modification to the slow start phase of congestion control algorithms. Slow start can overshoot the ideal send rate in many cases, causing high packet loss and poor performance. HyStart++ uses increase in round-trip delay as a heuristic to find an exit point before possible overshoot. It also adds a mitigation to prevent jitter from causing premature slow start exit.

This document describes a method to transport Internet Key Exchange Protocol (IKE) and IPsec packets over a TCP connection for traversing network middleboxes that may block IKE negotiation over UDP. This method, referred to as "TCP encapsulation", involves sending both IKE packets for Security Association (SA) establishment and Encapsulating Security Payload (ESP) packets over a TCP connection. This method is intended to be used as a fallback option when IKE cannot be negotiated over UDP.

TCP encapsulation for IKE and IPsec was defined in RFC 8229. This document clarifies the specification for TCP encapsulation by including additional clarifications obtained during implementation and deployment of this method. This documents obsoletes RFC 8229.

This document defines a YANG data model that can be used to configure and manage Bidirectional Forwarding Detection (BFD).

The YANG modules in this document conform to the Network Management Datastore Architecture (NMDA) (RFC 8342). This document updates "YANG Data Model for Bidirectional Forwarding Detection (BFD)" (RFC 9127).

This document specifies the Transmission Control Protocol (TCP). TCP is an important transport-layer protocol in the Internet protocol stack, and it has continuously evolved over decades of use and growth of the Internet. Over this time, a number of changes have been made to TCP as it was specified in RFC 793, though these have only been documented in a piecemeal fashion. This document collects and brings those changes together with the protocol specification from RFC 793. This document obsoletes RFC 793, as well as RFCs 879, 2873, 6093, 6429, 6528, and 6691 that updated parts of RFC 793. It updates RFCs 1011 and 1122, and it should be considered as a replacement for the portions of those documents dealing with TCP requirements. It also updates RFC 5961 by adding a small clarification in reset handling while in the SYN-RECEIVED state. The TCP header control bits from RFC 793 have also been updated based on RFC 3168.

This document provides test vectors to validate implementations of the two mandatory authentication algorithms specified for the TCP Authentication Option over both IPv4 and IPv6. This includes validation of the key derivation function (KDF) based on a set of test connection parameters as well as validation of the message authentication code (MAC). Vectors are provided for both currently required pairs of KDF and MAC algorithms: KDF_HMAC_SHA1 and HMAC- SHA-1-96, and KDF_AES_128_CMAC and AES-128-CMAC-96. The vectors also validate both whole TCP segments as well as segments whose options are excluded for middlebox traversal.

This document updates RFCs 1123 and 1536. This document requires the operational practice of permitting DNS messages to be carried over TCP on the Internet as a Best Current Practice. This operational requirement is aligned with the implementation requirements in RFC 7766. The use of TCP includes both DNS over unencrypted TCP as well as over an encrypted TLS session. The document also considers the consequences of this form of DNS communication and the potential operational issues that can arise when this Best Current Practice is not upheld.

This document describes a TCP convergence layer (TCPCL) for Delay-Tolerant Networking (DTN). This version of the TCPCL protocol resolves implementation issues in the earlier TCPCL version 3 as defined in RFC 7242 and provides updates to the Bundle Protocol (BP) contents, encodings, and convergence-layer requirements in BP version 7 (BPv7). Specifically, TCPCLv4 uses BPv7 bundles encoded by the Concise Binary Object Representation (CBOR) as its service data unit being transported and provides a reliable transport of such bundles. This TCPCL version also includes security and extensibility mechanisms.

This document defines a YANG data model that can be used to configure and manage Bidirectional Forwarding Detection (BFD).

The YANG modules in this document conform to the Network Management Datastore Architecture (NMDA) (RFC 8342).

This memo provides guidance to TCP implementers that is intended to help improve connection convergence to steady-state operation without affecting interoperability. It updates and replaces RFC 2140's description of sharing TCP state, as typically represented in TCP Control Blocks, among similar concurrent or consecutive connections.

This document provides guidance on how to implement and use the Transmission Control Protocol (TCP) in Constrained-Node Networks (CNNs), which are a characteristic of the Internet of Things (IoT). Such environments require a lightweight TCP implementation and may not make use of optional functionality. This document explains a number of known and deployed techniques to simplify a TCP stack as well as corresponding trade-offs. The objective is to help embedded developers with decisions on which TCP features to use.

This document presents the RACK-TLP loss detection algorithm for TCP. RACK-TLP uses per-segment transmit timestamps and selective acknowledgments (SACKs) and has two parts. Recent Acknowledgment (RACK) starts fast recovery quickly using time-based inferences derived from acknowledgment (ACK) feedback, and Tail Loss Probe (TLP) leverages RACK and sends a probe packet to trigger ACK feedback to avoid retransmission timeout (RTO) events. Compared to the widely used duplicate acknowledgment (DupAck) threshold approach, RACK-TLP detects losses more efficiently when there are application-limited flights of data, lost retransmissions, or data packet reordering events. It is intended to be an alternative to the DupAck threshold approach.

Many protocols must detect packet loss for various reasons (e.g., to ensure reliability using retransmissions or to understand the level of congestion along a network path). While many mechanisms have been designed to detect loss, ultimately, protocols can only count on the passage of time without delivery confirmation to declare a packet "lost". Each implementation of a time-based loss detection mechanism represents a balance between correctness and timeliness; therefore, no implementation suits all situations. This document provides high-level requirements for time-based loss detectors appropriate for general use in unicast communication across the Internet. Within the requirements, implementations have latitude to define particulars that best address each situation.

When multiple congestion-controlled Real-time Transport Protocol (RTP) sessions traverse the same network bottleneck, combining their controls can improve the total on-the-wire behavior in terms of delay, loss, and fairness. This document describes such a method for flows that have the same sender, in a way that is as flexible and simple as possible while minimizing the number of changes needed to existing RTP applications. This document also specifies how to apply the method for the Network-Assisted Dynamic Adaptation (NADA) congestion control algorithm and provides suggestions on how to apply it to other congestion control algorithms.

Active Queue Management (AQM) mechanisms allow for burst tolerance while enforcing short queues to minimise the time that packets spend enqueued at a bottleneck. This can cause noticeable performance degradation for TCP connections traversing such a bottleneck, especially if there are only a few flows or their bandwidth-delay product (BDP) is large. The reception of a Congestion Experienced (CE) Explicit Congestion Notification (ECN) mark indicates that an AQM mechanism is used at the bottleneck, and the bottleneck network queue is therefore likely to be short. Feedback of this signal allows the TCP sender-side ECN reaction in congestion avoidance to reduce the Congestion Window (cwnd) by a smaller amount than the congestion control algorithm's reaction to inferred packet loss. Therefore, this specification defines an experimental change to the TCP reaction specified in RFC 3168, as permitted by RFC 8311.

The Constrained Application Protocol (CoAP), although inspired by HTTP, was designed to use UDP instead of TCP. The message layer of CoAP over UDP includes support for reliable delivery, simple congestion control, and flow control.

Some environments benefit from the availability of CoAP carried over reliable transports such as TCP or Transport Layer Security (TLS). This document outlines the changes required to use CoAP over TCP, TLS, and WebSockets transports. It also formally updates RFC 7641 for use with these transports and RFC 7959 to enable the use of larger messages over a reliable transport.

Many communication protocols operating over the modern Internet use hostnames. These often resolve to multiple IP addresses, each of which may have different performance and connectivity characteristics. Since specific addresses or address families (IPv4 or IPv6) may be blocked, broken, or sub-optimal on a network, clients that attempt multiple connections in parallel have a chance of establishing a connection more quickly. This document specifies requirements for algorithms that reduce this user-visible delay and provides an example algorithm, referred to as "Happy Eyeballs". This document obsoletes the original algorithm description in RFC 6555.

This Informational RFC describes Data Center TCP (DCTCP): a TCP congestion control scheme for data-center traffic. DCTCP extends the Explicit Congestion Notification (ECN) processing to estimate the fraction of bytes that encounter congestion rather than simply detecting that some congestion has occurred. DCTCP then scales the TCP congestion window based on this estimate. This method achieves high-burst tolerance, low latency, and high throughput with shallow- buffered switches. This memo also discusses deployment issues related to the coexistence of DCTCP and conventional TCP, discusses the lack of a negotiating mechanism between sender and receiver, and presents some possible mitigations. This memo documents DCTCP as currently implemented by several major operating systems. DCTCP, as described in this specification, is applicable to deployments in controlled environments like data centers, but it must not be deployed over the public Internet without additional measures.

The goal of this document is to describe the potential benefits of applications using a transport that enables Explicit Congestion Notification (ECN). The document outlines the principal gains in terms of increased throughput, reduced delay, and other benefits when ECN is used over a network path that includes equipment that supports Congestion Experienced (CE) marking. It also discusses challenges for successful deployment of ECN. It does not propose new algorithms to use ECN nor does it describe the details of implementation of ECN in endpoint devices (Internet hosts), routers, or other network devices.

Recent RFCs have discussed issues with host identification in IP address-sharing systems, such as address/prefix-sharing devices and application-layer proxies. Potential solutions for revealing a host identifier in shared address deployments have also been discussed. This memo describes the design, deployment, and privacy considerations for one such solution in operational use on the Internet today that uses a TCP option to transmit a host identifier.

The RADIUS-over-TLS experiment described in RFC 6614 has opened RADIUS to new use cases where the 4096-octet maximum size limit of a RADIUS packet proves problematic. This specification extends the RADIUS-over-TCP experiment (RFC 6613) to permit larger RADIUS packets. This specification compliments other ongoing work to permit fragmentation of RADIUS authorization information. This document registers a new RADIUS code, an action that required IESG approval.

Pseudowires (PWs) have become a common mechanism for tunneling traffic and may be found in unmanaged scenarios competing for network resources both with other PWs and with non-PW traffic, such as TCP/IP flows. Thus, it is worthwhile specifying under what conditions such competition is acceptable, i.e., the PW traffic does not significantly harm other traffic or contribute more than it should to congestion. We conclude that PWs transporting responsive traffic behave as desired without the need for additional mechanisms. For inelastic PWs (such as Time Division Multiplexing (TDM) PWs), we derive a bound under which such PWs consume no more network capacity than a TCP flow. For TDM PWs, we find that the level of congestion at which the PW can no longer deliver acceptable TDM service is never significantly greater, and is typically much lower, than this bound. Therefore, as long as the PW is shut down when it can no longer deliver acceptable TDM service, it will never do significantly more harm than even a single TCP flow. If the TDM service does not automatically shut down, a mechanism to block persistently unacceptable TDM pseudowires is required.

Congestion Exposure (ConEx) is a mechanism by which senders inform the network about expected congestion based on congestion feedback from previous packets in the same flow. This document describes the necessary modifications to use ConEx with the Transmission Control Protocol (TCP).

The Real-time Transport Protocol (RTP) specification establishes a registry of profile names for use by higher-level control protocols, such as the Session Description Protocol (SDP), to refer to the transport methods. This specification describes the following new SDP transport protocol identifiers for transporting RTP Media over TCP: 'TCP/RTP/AVPF', 'TCP/RTP/SAVP', 'TCP/RTP/SAVPF', 'TCP/DTLS/RTP/SAVP', 'TCP/DTLS/RTP/SAVPF', 'TCP/TLS/RTP/AVP', and 'TCP/TLS/RTP/AVPF'.

This document reclassifies several TCP extensions and TCP-related documents that either have been superseded, have never seen widespread use, or are no longer recommended for use to "Historic" status. The affected documents are RFCs 675, 721, 761, 813, 816, 879, 896, 1078, and 6013. Additionally, this document reclassifies RFCs 700, 794, 814, 817, 872, 889, 964, and 1071 to "Informational" status.

DNS messages between clients and servers may be received over either UDP or TCP. UDP transport involves keeping less state on a busy server, but can cause truncation and retries over TCP. Additionally, UDP can be exploited for reflection attacks. Using TCP would reduce retransmits and amplification. However, clients commonly use TCP only for retries and servers typically use idle timeouts on the order of seconds.

This document defines an EDNS0 option ("edns-tcp-keepalive") that allows DNS servers to signal a variable idle timeout. This signalling encourages the use of long-lived TCP connections by allowing the state associated with TCP transport to be managed effectively with minimal impact on the DNS transaction time.

This document specifies the requirement for support of TCP as a transport protocol for DNS implementations and provides guidelines towards DNS-over-TCP performance on par with that of DNS-over-UDP. This document obsoletes RFC 5966 and therefore updates RFC 1035 and RFC 1123.

This document describes a modified sender-side algorithm for managing the TCP and Stream Control Transmission Protocol (SCTP) retransmission timers that provides faster loss recovery when there is a small amount of outstanding data for a connection. The modification, RTO Restart (RTOR), allows the transport to restart its retransmission timer using a smaller timeout duration, so that the effective retransmission timeout (RTO) becomes more aggressive in situations where fast retransmit cannot be used. This enables faster loss detection and recovery for connections that are short lived or application limited.

This document provides a mechanism to address issues that arise when TCP is used for traffic that exhibits periods where the sending rate is limited by the application rather than the congestion window. It provides an experimental update to TCP that allows a TCP sender to restart quickly following a rate-limited interval. This method is expected to benefit applications that send rate-limited traffic using TCP while also providing an appropriate response if congestion is experienced.

This document also evaluates the Experimental specification of TCP Congestion Window Validation (CWV) defined in RFC 2861 and concludes that RFC 2861 sought to address important issues but failed to deliver a widely used solution. This document therefore reclassifies the status of RFC 2861 from Experimental to Historic. This document obsoletes RFC 2861.

The Internet Architecture Board (IAB) through its IP Stack Evolution program, the Internet Society, and the Swiss Federal Institute of Technology (ETH) Zurich hosted the Stack Evolution in a Middlebox Internet (SEMI) workshop in Zurich on 26-27 January 2015 to explore the ability to evolve the transport layer in the presence of middlebox- and interface-related ossification of the stack. The goal of the workshop was to produce architectural and engineering guidance on future work to break the logjam, focusing on incrementally deployable approaches with clear incentives to deployment both on the endpoints (in new transport layers and applications) as well as on middleboxes (run by network operators). This document summarizes the contributions to the workshop and provides an overview of the discussion at the workshop, as well as the outcomes and next steps identified by the workshop. The views and positions documented in this report are those of the workshop participants and do not necessarily reflect IAB views and positions.

Explicit Congestion Notification (ECN) is a mechanism where network nodes can mark IP packets, instead of dropping them, to indicate congestion to the endpoints. An ECN-capable receiver will feed this information back to the sender. ECN is specified for TCP in such a way that it can only feed back one congestion signal per Round-Trip Time (RTT). In contrast, ECN for other transport protocols, such as RTP/UDP and SCTP, is specified with more accurate ECN feedback. Recent new TCP mechanisms (like Congestion Exposure (ConEx) or Data Center TCP (DCTCP)) need more accurate ECN feedback in the case where more than one marking is received in one RTT. This document specifies requirements for an update to the TCP protocol to provide more accurate ECN feedback.

This document provides recommendations to designers of application and service protocols on how to use the transport protocol port number space and when to request a port assignment from IANA. It provides designer guidance to requesters or users of port numbers on how to interact with IANA using the processes defined in RFC 6335; thus, this document complements (but does not update) that document.

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.

This document describes an experimental TCP mechanism called 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, as detailed in the Applicability section.

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.

This document obsoletes RFC 1323 and describes changes from it.

This document describes the protocol for the TCP-based convergence layer for Delay-Tolerant Networking (DTN). It is the product of the IRTF's DTN Research Group (DTNRG).

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.

This document describes an extension to the TCP Authentication Option (TCP-AO) to support its use over connections that pass through Network Address Translators and/or Network Address and Port Translators (NATs/NAPTs). This extension changes the data used to compute traffic keys, but it does not alter TCP-AO's packet processing or key generation algorithms.

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 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.

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.

This specification obsoletes RFC 4996. It fixes a technical issue with the SACK compression and clarifies other compression methods used. [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]

Consumers of IP network performance metrics have many different uses in mind. This memo provides "long-term" reporting considerations (e.g., hours, days, weeks, or months, as opposed to 10 seconds), based on analysis of the points of view of two key audiences. It describes how these audience categories affect the selection of metric parameters and options when seeking information that serves their needs. This document is not an Internet Standards Track specification; it is published for informational purposes.

The Remote Authentication Dial-In User Server (RADIUS) protocol has, until now, required the User Datagram Protocol (UDP) as the underlying transport layer. This document defines RADIUS over the Transmission Control Protocol (RADIUS/TCP), in order to address handling issues related to RADIUS over Transport Layer Security (RADIUS/TLS). It permits TCP to be used as a transport protocol for RADIUS only when a transport layer such as TLS or IPsec provides confidentiality and security. This document 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]

There have been many implementations and deployments of legacy syslog over TCP for many years. That protocol has evolved without being standardized and has proven to be quite interoperable in practice. This memo describes how TCP has been used as a transport for syslog messages. This document defines a Historic Document for the Internet community.

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]

Interactive Connectivity Establishment (ICE) defines a mechanism for NAT traversal for multimedia communication protocols based on the offer/answer model of session negotiation. ICE works by providing a set of candidate transport addresses for each media stream, which are then validated with peer-to-peer connectivity checks based on Session Traversal Utilities for NAT (STUN). ICE provides a general framework for describing candidates but only defines UDP-based media streams. This specification extends ICE to TCP-based media, including the ability to offer a mix of TCP and UDP-based candidates for a single stream. [STANDARDS-TRACK]

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.

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 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.

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.

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.

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.

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 specification defines an extension of Traversal Using Relays around NAT (TURN), a relay protocol for Network Address Translator (NAT) traversal. This extension allows a TURN client to request TCP allocations, and defines new requests and indications for the TURN server to open and accept TCP connections with the client\'s peers. TURN and this extension both purposefully restrict the ways in which the relayed address can be used. In particular, it prevents users from running general-purpose servers from ports obtained from the TURN server. [STANDARDS-TRACK]

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 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.

This document enables a pair of communicating proxies to reuse a congestion-controlled connection between themselves for sending requests in the forwards and backwards direction. Because the connection is essentially aliased for requests going in the backwards direction, reuse is predicated upon both the communicating endpoints authenticating themselves using X.509 certificates through Transport Layer Security (TLS). For this reason, we only consider connection reuse for TLS over TCP and TLS over Stream Control Transmission Protocol (SCTP). This document also provides guidelines on connection reuse and virtual SIP servers and the interaction of connection reuse and DNS SRV lookups in SIP. [STANDARDS-TRACK]

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 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]

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.

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.

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]

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 describes how an Extensible Provisioning Protocol (EPP) session is mapped onto a single Transmission Control Protocol (TCP) connection. This mapping requires use of the Transport Layer Security (TLS) protocol to protect information exchanged between an EPP client and an EPP server. This document obsoletes RFC 4934. [STANDARDS-TRACK]

This document specifies a profile for Congestion Control Identifier 4, the small-packet variant of TCP-Friendly Rate Control (TFRC), in the Datagram Congestion Control Protocol (DCCP). CCID 4 is for experimental use, and uses TFRC-SP (RFC 4828), a variant of TFRC designed for applications that send small packets. CCID 4 is considered experimental because TFRC-SP is itself experimental, and is not proposed for widespread deployment in the global Internet at this time. The goal for TFRC-SP is to achieve roughly the same bandwidth in bits per second (bps) as a TCP flow using packets of up to 1500 bytes but experiencing the same level of congestion. CCID 4 is for use for senders that send small packets and would like a TCP- friendly sending rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes. 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.

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.

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]

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 defines a set of requirements for NATs that handle TCP that would allow many applications, such as peer-to-peer applications and online games to work consistently. Developing NATs that meet this set of requirements will greatly increase the likelihood that these applications will function properly. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.

This document specifies TCP Friendly Rate Control (TFRC). TFRC is a congestion control mechanism for unicast flows operating in a best-effort Internet environment. It is reasonably fair when competing for bandwidth with TCP flows, but has a much lower variation of throughput over time compared with TCP, making it more suitable for applications such as streaming media where a relatively smooth sending rate is of importance.

This document obsoletes RFC 3448 and updates RFC 4342. [STANDARDS-TRACK]

Marker PDU Aligned Framing (MPA) is designed to work as an "adaptation layer" between TCP and the Direct Data Placement protocol (DDP) as described in RFC 5041. It preserves the reliable, in-order delivery of TCP, while adding the preservation of higher-level protocol record boundaries that DDP requires. MPA is fully compliant with applicable TCP RFCs and can be utilized with existing TCP implementations. MPA also supports integrated implementations that combine TCP, MPA and DDP to reduce buffering requirements in the implementation and improve performance at the system level. [STANDARDS-TRACK]

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 an extensibility mechanism for the Kerberos V5 protocol when used over TCP transports. The mechanism uses the reserved high-bit in the length field. It can be used to negotiate TCP-specific Kerberos extensions. [STANDARDS-TRACK]

This document describes a simple TCP transfer protocol for the Internet Registry Information Service (IRIS). Data is transferred between clients and servers using chunks to achieve pipelining. [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 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]

This document proposes a mechanism for further experimentation, but not for widespread deployment at this time in the global Internet.

TCP-Friendly Rate Control (TFRC) is a congestion control mechanism for unicast flows operating in a best-effort Internet environment (RFC 3448). TFRC was intended for applications that use a fixed packet size, and was designed to be reasonably fair when competing for bandwidth with TCP connections using the same packet size. This document proposes TFRC-SP, a Small-Packet (SP) variant of TFRC, that is designed for applications that send small packets. The design goal for TFRC-SP is to achieve the same bandwidth in bps (bits per second) as a TCP flow using packets of up to 1500 bytes. TFRC-SP enforces a minimum interval of 10 ms between data packets to prevent a single flow from sending small packets arbitrarily frequently.

Flows using TFRC-SP compete reasonably fairly with large-packet TCP and TFRC flows in environments where large-packet flows and small-packet flows experience similar packet drop rates. However, in environments where small-packet flows experience lower packet drop rates than large-packet flows (e.g., with Drop-Tail queues in units of bytes), TFRC-SP can receive considerably more than its share of the bandwidth. This memo defines an Experimental Protocol for the Internet community.

The TCP-MD5 option is most commonly used to secure BGP sessions between routers. However, changing the long-term key is difficult, since the change needs to be synchronized between different organizations. We describe single-ended strategies that will permit (mostly) unsynchronized key changes. This memo provides information 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.

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.

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 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.

This document specifies TCP-Friendly Multicast Congestion Control (TFMCC). TFMCC is a congestion control mechanism for multicast transmissions in a best-effort Internet environment. It is a single-rate congestion control scheme, where the sending rate is adapted to the receiver experiencing the worst network conditions. TFMCC is reasonably fair when competing for bandwidth with TCP flows and has a relatively low variation of throughput over time, making it suitable for applications where a relatively smooth sending rate is of importance, such as streaming media. This memo defines an Experimental Protocol for the Internet community.

This memo defines a method for framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) packets onto connection-oriented transport (such as TCP). The memo also defines how session descriptions may specify RTP streams that use the framing method. [STANDARDS-TRACK]

This document contains the profile for Congestion Control Identifier 3, TCP-Friendly Rate Control (TFRC), in the Datagram Congestion Control Protocol (DCCP). CCID 3 should be used by senders that want a TCP-friendly sending rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes. [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]

This memo describes TCP/IP field behavior in the context of header compression. Header compression is possible because most header fields do not vary randomly from packet to packet. Many of the fields exhibit static behavior or change in a more or less predictable way. When a header compression scheme is designed, it is of fundamental importance to understand the behavior of the fields in detail. An example of this analysis can be seen in RFC 3095. This memo performs a similar role for the compression of TCP/IP headers. This memo provides information for the Internet community.

The IETF Standards Process requires that all normative references for a document be at the same or higher level of standardization. RFC 2026 section 9.1 allows the IESG to grant a variance to the standard practices of the IETF. This document explains why the IESG is considering doing so for the revised version of the BGP-4 specification, which refers normatively to RFC 2385, "Protection of BGP Sessions via the TCP MD5 Signature Option". RFC 2385 will remain at the Proposed Standard level. This memo provides information for the Internet community.

This document specifies an architecture and a gateway-to-gateway protocol for the implementation of fibre channel fabric functionality over an IP network. This functionality is provided through TCP protocols for fibre channel frame transport and the distributed fabric services specified by the fibre channel standards. The architecture enables internetworking of fibre channel devices through gateway-accessed regions with the fault isolation properties of autonomous systems and the scalability of the IP network. [STANDARDS-TRACK]

Internet Small Computer System Interface (iSCSI) is a proposed transport protocol for Small Computer Systems Interface (SCSI) that operates on top of TCP. This memo describes a standard mechanism for enabling clients to bootstrap themselves using the iSCSI protocol. The goal of this standard is to enable iSCSI boot clients to obtain the information to open an iSCSI session with the iSCSI boot server. [STANDARDS-TRACK]

This document describes how to express media transport over TCP using the Session Description Protocol (SDP). It defines the SDP 'TCP' protocol identifier, the SDP 'setup' attribute, which describes the connection setup procedure, and the SDP 'connection' attribute, which handles connection reestablishment. [STANDARDS-TRACK]

This document contains requirements on the TCP/IP header compression scheme (profile) to be developed by the RObust Header Compression (ROHC) Working Group. The document discusses the scope of TCP compression, performance considerations, assumptions about the surrounding environment, as well as Intellectual Property Rights concerns. The structure of this document is inherited from RFC 3096, which defines IP/UDP/RTP requirements for ROHC. This memo provides information for the Internet community.

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. The F-RTO algorithm can also be applied to the Stream Control Transmission Protocol (SCTP). This memo defines an Experimental Protocol for the Internet community.

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]

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]

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.

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.

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 memo describes some considerations relating to the use of IP protocol number fields and payload protocol (e.g., TCP) port fields to identify particular services that may be associated with that port number or protocol number.

The TCP MD5 Signature Option (RFC 2385), used predominantly by BGP, has seen significant deployment in critical areas of Internet infrastructure. The security of this option relies heavily on the quality of the keying material used to compute the MD5 signature. This document addresses the security requirements of that keying material. This memo provides information 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 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.

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.

This memo defines a portion of the Management Information Base (MIB) for use with network management protocols in TCP/IP based internets. In particular, it defines objects for managing networks using Synchronous Optical Network (SONET) linear Automatic Protection Switching (APS) architectures. [STANDARDS-TRACK]

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 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 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 memo defines a transport mapping for using the Simple Network Management Protocol (SNMP) over TCP. The transport mapping can be used with any version of SNMP. This document extends the transport mappings defined in STD 62, RFC 3417. This memo defines an Experimental Protocol for the Internet community.

This document specifies an optional standard for TCP to increase the

permitted initial window from one segment to roughly 4K bytes,

replacing RFC 2414. This document discusses the advantages and

disadvantages of the higher initial window. The document includes

discussion of experiments and simulations showing that the higher

initial window does not lead to congestion collapse. Finally, the

document provides guidance on implementation issues.

This document is being written because there are a number of

firewalls in the Internet that inappropriately reset a TCP connection

upon receiving certain TCP SYN packets, in particular, packets with

flags set in the Reserved field of the TCP header. In this document

we argue that this practice is not conformant with TCP standards, and

is an inappropriate overloading of the semantics of the TCP reset.

We also consider the longer-term consequences of this and similar

actions as obstacles to the evolution of the Internet infrastructure.

The IP Storage Working group is chartered with developing

comprehensive technology to transport block storage data over IP

protocols. This effort includes a protocol to transport the Small

Computer Systems Interface (SCSI) protocol over the Internet

(iSCSI). The initial version of the iSCSI protocol will define a

mapping of SCSI transport protocol over TCP/IP so that SCSI storage

controllers (principally disk and tape arrays and libraries) can be

attached to IP networks, notably Gigabit Ethernet (GbE) and 10

Gigabit Ethernet (10 GbE).

This memo specifies the encapsulation of GSMP (General Switch Management Protocol) packets in ATM (Asynchronous Transfer Mode), Ethernet and TCP (Transmission Control Protocol). [STANDARDS-TRACK]