congestion control

This document specifies a cautious method for Internet transports that enables fast startup of Congestion Control (CC) for a wide range of connections, known as "Careful Resume". It reuses a set of computed CC parameters that are based on previously observed path characteristics between the same pair of transport endpoints. These parameters are saved, allowing them to be later used to modify the CC behaviour of a subsequent connection.

This document describes the assumptions and defines the requirements for how a sender utilises these parameters to provide opportunities for a connection to more rapidly get up to speed and utilise available capacity. It discusses how the use of this method impacts the capacity at a shared network bottleneck and the safe response that is needed after any indication that the new rate is inappropriate.

Datagram Congestion Control Protocol (DCCP) communications, as defined in RFC 4340, are inherently restricted to a single path per connection, despite the availability of multiple network paths between peers. The ability to utilize multiple paths simultaneously for a DCCP session can enhance network resource utilization, improve throughput, and increase resilience to network failures, ultimately enhancing the user experience.

Use cases for Multipath DCCP (MP-DCCP) include mobile devices (e.g., handsets and vehicles) and residential home gateways that maintain simultaneous connections to distinct network types such as cellular and Wireless Local Area Networks (WLANs) or cellular and fixed access networks. Compared to existing multipath transport protocols, such as Multipath TCP (MPTCP), MP-DCCP is particularly suited for latency-sensitive applications with varying requirements for reliability and in-order delivery.

This document specifies a set of protocol extensions to DCCP that enable multipath operations. These extensions maintain the same service model as DCCP while introducing mechanisms to establish and utilize multiple concurrent DCCP flows across different network paths.

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).

RFC 5033 discusses the principles and guidelines for standardizing new congestion control algorithms. This document obsoletes RFC 5033 to reflect changes in the congestion control landscape by providing a framework for the development and assessment of congestion control mechanisms, promoting stability across diverse network paths. This document seeks to ensure that proposed congestion control algorithms operate efficiently and without harm when used in the global Internet. It emphasizes the need for comprehensive testing and validation to prevent adverse interactions with existing flows.

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 memo discusses the rate at which congestion control feedback can be sent using the RTP Control Protocol (RTCP) and the suitability of RTCP for implementing congestion control for unicast multimedia applications.

Forward Erasure Correction (FEC) is a reliability mechanism that is distinct and separate from the retransmission logic in reliable transfer protocols such as TCP. FEC coding can help deal with losses at the end of transfers or with networks having non-congestion losses. However, FEC coding mechanisms should not hide congestion signals. This memo offers a discussion of how FEC coding and congestion control can coexist. Another objective is to encourage the research community to also consider congestion control aspects when proposing and comparing FEC coding solutions in communication systems.

This document is the product of the Coding for Efficient Network Communications Research Group (NWCRG). The scope of the document is end-to-end communications; FEC coding for tunnels is out of the scope of the document.

This is a position paper. It documents the author's personal views on how Quality of Service (QoS) capabilities ought to be accommodated in Information-Centric Networking (ICN) protocols like Content-Centric Networking (CCNx) or Named Data Networking (NDN), which employ flow-balanced Interest/Data exchanges and hop-by-hop forwarding state as their fundamental machinery. It argues that such protocols demand a substantially different approach to QoS from that taken in TCP/IP and proposes specific design patterns to achieve both classification and differentiated QoS treatment on both a flow and aggregate basis. It also considers the effect of caches in addition to memory, CPU, and link bandwidth as resources that should be subject to explicitly unfair resource allocation. The proposed methods are intended to operate purely at the network layer, providing the primitives needed to achieve transport- and higher-layer QoS objectives. It explicitly excludes any discussion of Quality of Experience (QoE), which can only be assessed and controlled at the application layer or above.

This document is not a product of the IRTF Information-Centric Networking Research Group (ICNRG) but has been through formal Last Call and has the support of the participants in the research group for publication as an individual submission.

This document describes loss detection and congestion control mechanisms for QUIC.

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.

An effective RTP congestion control algorithm requires more fine-grained feedback on packet loss, timing, and Explicit Congestion Notification (ECN) marks than is provided by the standard RTP Control Protocol (RTCP) Sender Report (SR) and Receiver Report (RR) packets. This document describes an RTCP feedback message intended to enable congestion control for interactive real-time traffic using RTP. The feedback message is designed for use with a sender-based congestion control algorithm, in which the receiver of an RTP flow sends back to the sender RTCP feedback packets containing the information the sender needs to perform congestion control.

The Real-time Transport Protocol (RTP) is a common transport choice for interactive multimedia communication applications. The performance of these applications typically depends on a well-functioning congestion control algorithm. To ensure a seamless and robust user experience, a well-designed RTP-based congestion control algorithm should work well across all access network types. This document describes test cases for evaluating performances of candidate congestion control algorithms over cellular and Wi-Fi networks.

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.

RFC 6363 describes a framework for using Forward Error Correction (FEC) codes to provide protection against packet loss. The framework supports applying FEC to arbitrary packet flows over unreliable transport and is primarily intended for real-time, or streaming, media. However, FECFRAME as per RFC 6363 is restricted to block FEC codes. This document updates RFC 6363 to support FEC codes based on a sliding encoding window, in addition to block FEC codes, in a backward-compatible way. During multicast/broadcast real-time content delivery, the use of sliding window codes significantly improves robustness in harsh environments, with less repair traffic and lower FEC-related added latency.

This document describes two fully specified Forward Erasure Correction (FEC) Schemes for Sliding Window Random Linear Codes (RLC), one for RLC over the Galois Field (a.k.a., Finite Field) GF(2), a second one for RLC over the Galois Field GF(2^8), each time with the possibility of controlling the code density. They can protect arbitrary media streams along the lines defined by FECFRAME extended to Sliding Window FEC Codes. These Sliding Window FEC Codes rely on an encoding window that slides over the source symbols, generating new repair symbols whenever needed. Compared to block FEC codes, these Sliding Window FEC Codes offer key advantages with real-time flows in terms of reduced FEC-related latency while often providing improved packet erasure recovery capabilities.

This document describes two reference video traffic models for evaluating RTP congestion control algorithms. The first model statistically characterizes the behavior of a live video encoder in response to changing requests on the target video rate. The second model is trace-driven and emulates the output of actual encoded video frame sizes from a high-resolution test sequence. Both models are designed to strike a balance between simplicity, repeatability, and authenticity in modeling the interactions between a live video traffic source and the congestion control module. Finally, the document describes how both approaches can be combined into a hybrid model.

This document describes a mechanism to detect whether end-to-end data flows share a common bottleneck. This mechanism relies on summary statistics that are calculated based on continuous measurements and used as input to a grouping algorithm that runs wherever the knowledge is needed.

This memo describes a rate adaptation algorithm for conversational media services such as interactive video. The solution conforms to the packet conservation principle and uses a hybrid loss-and-delay- based congestion control algorithm. The algorithm is evaluated over both simulated Internet bottleneck scenarios as well as in a Long Term Evolution (LTE) system simulator and is shown to achieve both low latency and high video throughput in these scenarios.

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows.

This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms.

This document explains what is meant by the term "network transport Circuit Breaker". It describes the need for Circuit Breakers (CBs) for network tunnels and applications when using non-congestion- controlled traffic and explains where CBs are, and are not, needed. It also defines requirements for building a CB and the expected outcomes of using a CB within the Internet.

This document discusses both use cases and operational experience with Multipath TCP (MPTCP) in real networks. It lists several prominent use cases where Multipath TCP has been considered and is being used. It also gives insight to some heuristics and decisions that have helped to realize these use cases and suggests possible improvements.

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 summary of the IAB/IRTF Workshop on 'Congestion Control for Interactive Real-Time Communication', which took place in Vancouver, Canada, on July 28, 2012. The main goal of the workshop was to foster a discussion on congestion control mechanisms for interactive real-time communication. This report summarizes the discussions and lists recommendations to the Internet Engineering Task Force (IETF) community.

The views and positions in this report are those of the workshop participants and do not necessarily reflect the views and positions of the authors, the Internet Architecture Board (IAB), or the Internet Research Task Force (IRTF).

This document specifies an alternative encapsulation of the Datagram Congestion Control Protocol (DCCP), referred to as DCCP-UDP. This encapsulation allows DCCP to be carried through the current generation of Network Address Translation (NAT) middleboxes without modification of those middleboxes. This document also updates the Session Description Protocol (SDP) information for DCCP defined in RFC 5762. [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 document formally deprecates the use of ICMP Source Quench messages by transport protocols, formally updating RFC 792, RFC 1122, and RFC 1812. [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.

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.

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.

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 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.

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 document describes the congestion management system of Comcast Cable, a large cable broadband Internet Service Provider (ISP) in the U.S. Comcast completed deployment of this congestion management system on December 31, 2008. This document is not an Internet Standards Track specification; it is published for informational purposes.

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.

The Real-time Transport Protocol (RTP) is a widely used transport for real-time multimedia on IP networks. The Datagram Congestion Control Protocol (DCCP) is a transport protocol that provides desirable services for real-time applications. This memo specifies a mapping of RTP onto DCCP, along with associated signalling, such that real- time applications can make use of the services provided by DCCP. [STANDARDS-TRACK]

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 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 describes the usage of Service Codes by the Datagram Congestion Control Protocol, RFC 4340. It motivates the setting of a Service Code by applications. Service Codes provide a method to identify the intended service/application to process a DCCP connection request. This provides improved flexibility in the use and assignment of port numbers for connection multiplexing. The use of a DCCP Service Code can also enable more explicit coordination of services with middleboxes (e.g., network address translators and firewalls). This document updates the specification provided in RFC 4340. [STANDARDS-TRACK]

This document specifies an update to the Datagram Congestion Control Protocol (DCCP), a connection-oriented and datagram-based transport protocol. The update adds support for the DCCP-Listen packet. This assists DCCP applications to communicate through middleboxes (e.g., a Network Address Port Translator or a DCCP server behind a firewall), where peering endpoints need to initiate communication in a near- simultaneous manner to establish necessary middlebox state. [STANDARDS-TRACK]

This document defines a set of requirements for NATs handling the Datagram Congestion Control Protocol (DCCP). These requirements allow DCCP applications, such as streaming applications, to operate consistently, and they are very similar to the TCP requirements for NATs, which have already been published by the IETF. Ensuring that NATs meet this set of requirements will greatly increase the likelihood that applications using DCCP will function properly. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.

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 specifies the use of the Quick-Start mechanism by the Datagram Congestion Control Protocol (DCCP). DCCP is a transport protocol that allows the transmission of congestion-controlled, unreliable datagrams. DCCP is intended for applications such as streaming media, Internet telephony, and online games. In DCCP, an application has a choice of congestion control mechanisms, each specified by a Congestion Control Identifier (CCID). This document specifies general procedures applicable to all DCCP CCIDs and specific procedures for the use of Quick-Start with DCCP CCID 2, CCID 3, and CCID 4. Quick-Start enables a DCCP sender to cooperate with Quick-Start routers along the end-to-end path to determine an allowed sending rate at the start of a connection and, at times, in the middle of a DCCP connection (e.g., after an idle or application- limited period). The present specification is provided for 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.

This document specifies the use of Datagram Transport Layer Security (DTLS) over the Datagram Congestion Control Protocol (DCCP). DTLS provides communications privacy for applications that use datagram transport protocols and allows client/server applications to communicate in a way that is designed to prevent eavesdropping and detect tampering or message forgery. DCCP is a transport protocol that provides a congestion-controlled unreliable datagram service. [STANDARDS-TRACK]

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.

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.

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 document describes for the historical record the motivation behind the Datagram Congestion Control Protocol (DCCP), an unreliable transport protocol incorporating end-to-end congestion control. DCCP implements a congestion-controlled, unreliable flow of datagrams for use by applications such as streaming media or on-line games. This memo provides information for the Internet community.

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]

This document recommends methods that are intended to improve the scalability and stability of large networks using Open Shortest Path First (OSPF) Version 2 protocol. The methods include processing OSPF Hellos and Link State Advertisement (LSA) Acknowledgments at a higher priority compared to other OSPF packets, and other congestion avoidance procedures. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.

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 document specifies Wave and Equation Based Rate Control (WEBRC), which provides rate and congestion control for data delivery. WEBRC is specifically designed to support protocols using IP multicast. It provides multiple-rate, congestion-controlled delivery to receivers, i.e., different receivers joined to the same session may be receiving packets at different rates depending on the bandwidths of their individual connections to the sender and on competing traffic along these connections. WEBRC requires no feedback from receivers to the sender, i.e., it is a completely receiver-driven congestion control protocol. Thus, it is designed to scale to potentially massive numbers of receivers attached to a session from a single sender. Furthermore, because each individual receiver adjusts to the available bandwidth between the sender and that receiver, there is the potential to deliver data to each individual receiver at the fastest possible rate for that receiver, even in a highly heterogeneous network architecture, using a single sender. This memo defines an Experimental Protocol for the Internet 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 discusses IAB concerns about effective end-to-end congestion control for best-effort voice traffic in the Internet. These concerns have to do with fairness, user quality, and with the dangers of congestion collapse. The concerns are particularly relevant in light of the absence of a widespread Quality of Service (QoS) deployment in the Internet, and the likelihood that this situation will not change much in the near term. This document is not making any recommendations about deployment paths for Voice over Internet Protocol (VoIP) in terms of QoS support, and is not claiming that best-effort service can be relied upon to give acceptable performance for VoIP. We are merely observing that voice traffic is occasionally deployed as best-effort traffic over some links in the Internet, that we expect this occasional deployment to continue, and that we have concerns about the lack of effective end-to-end congestion control for this best-effort voice traffic. This memo provides information 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.

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.

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 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 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]

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.

The purpose of this paper is to present a review of the congestion control approaches, as a way of encouraging new discussion and experimentation. Included in the survey are Source Quench, Random Drop, Congestion Indication (DEC Bit), and Fair Queueing.

This memo is a discussion of some of the ideas expressed in RFC-1016 on Source Quench. This memo introduces the distinction of the cause of congestion in a gateway between the effects of "Funneling" and "Mismatch". It is offered in the same spirit as RFC-1016; to stimulate discussion. The opinions offered are personal, not corporate, opinions. Distribution of this memo is unlimited.

The memo is intended to explore the issue of what a host could do with a source quench. The proposal is for each source host IP module to introduce some delay between datagrams sent to the same destination host. This is a "crazy idea paper" and discussion is essential.