A Survey on TCP-Friendly Congestion Control
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paper_title: Promoting the use of end-to-end congestion control in the Internet

paper_content:

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, "not TCP-friendly", or simply using disproportionate bandwidth. A flow that is not "TCP-friendly" is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion.
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paper_title: Modeling TCP Reno performance: a simple model and its empirical validation

paper_content:

The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization of the steady-state send rate as a function of loss rate and round trip time (RTT) for a bulk transfer TCP flow. Unlike the models of Lakshman and Madhow (see IEE/ACM Trans. Networking, vol.5, p.336-50, 1997), Mahdavi and Floyd (1997), Mathis, Semke, Mahdavi and Ott (see Comput. Commun. Rev., vol.27, no.3, 1997) and by by Ott et al., our model captures not only the behavior of the fast retransmit mechanism but also the effect of the time-out mechanism. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP send rate and is accurate over a wider range of loss rates. We also present a simple extension of our model to compute the throughput of a bulk transfer TCP flow, which is defined as the amount of data received by the receiver in unit time.
---
paper_title: Promoting the use of end-to-end congestion control in the Internet

paper_content:

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, "not TCP-friendly", or simply using disproportionate bandwidth. A flow that is not "TCP-friendly" is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion.
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paper_title: Global fairness of additive-increase and multiplicative-decrease with heterogeneous round-trip times

paper_content:

Consider a network with an arbitrary topology and arbitrary communication delays, in which congestion control is based on additive-increase and multiplicative-decrease. We show that the source rates tend to be distributed in order to maximize an objective function called F/sub A//sup h/ ("F/sub A//sup h/ fairness"). We derive this result under the assumption of rate proportional negative feedback and for the regime of rare negative feedback. This applies to TCP in moderately loaded networks, and to those TCP implementations that are designed to interpret multiple packet losses within one RTT as a single congestion indication and do not rely on re-transmission timeout. This result provides some insight into the distribution of rates, and hence of packet loss ratios, which can be expected in a given network with a number of competing TCP or TCP-friendly sources. We validate our findings by analyzing a multiple-bottleneck scenario, and comparing with previous results (Floyd, 1991, Mathis et al, 1997) and an extensive numerical simulation with realistic parameter settings. We apply F/sub A//sup h/ fairness to gain a more accurate understanding of the bias of TCP against long round-trip times.
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paper_title: Achieving bounded fairness for multicast and TCP traffic in the Internet

paper_content:

There is an urgent need for effective multicast congestion control algorithms which enable reasonably fair share of network resources between multicast and unicast TCP traffic under the current Internet infrastructure. In this paper, we propose a quantitative definition of a type of bounded fairness between multicast and unicast best-effort traffic, termed "essentially fair". We also propose a window-based Random Listening Algorithm (RLA) for multicast congestion control. The algorithm is proven to be essentially fair to TCP connections under a restricted topology with equal round-trip times and with phase effects eliminated. The algorithm is also fair to multiple multicast sessions. This paper provides the theoretical proofs and some simulation results to demonstrate that the RLA achieves good performance under various network topologies. These include the performance of a generalization of the RLA algorithm for topologies with different round-trip times.
---
paper_title: Promoting the use of end-to-end congestion control in the Internet

paper_content:

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, "not TCP-friendly", or simply using disproportionate bandwidth. A flow that is not "TCP-friendly" is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion.
---
paper_title: Modeling TCP Reno performance: a simple model and its empirical validation

paper_content:

The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization of the steady-state send rate as a function of loss rate and round trip time (RTT) for a bulk transfer TCP flow. Unlike the models of Lakshman and Madhow (see IEE/ACM Trans. Networking, vol.5, p.336-50, 1997), Mahdavi and Floyd (1997), Mathis, Semke, Mahdavi and Ott (see Comput. Commun. Rev., vol.27, no.3, 1997) and by by Ott et al., our model captures not only the behavior of the fast retransmit mechanism but also the effect of the time-out mechanism. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP send rate and is accurate over a wider range of loss rates. We also present a simple extension of our model to compute the throughput of a bulk transfer TCP flow, which is defined as the amount of data received by the receiver in unit time.
---
paper_title: Fundamental observations on multicast congestion control in the Internet

paper_content:

We study congestion control for one-to-many multicast applications in the Internet and establish a three-way relationship between the choice of regulation parameter (i.e., rate or window size), the requirement to estimate receiver round trip times, and the type of fairness that may be accomplished. In particular, we show that in order to provide TCP-compatible fairness in rate-based regulation, receiver round trip times must be known. However, such a requirement does not exist in window-based regulation. We further show that measurement of receiver round-trip times in multicast communication, is fundamentally different and more complex than unicast communication, in order to avoid implosion of acknowledgments at the source. A major part of the paper deals with extending window-based regulation to multicast communications. We show that window-based regulation using a common window-size for the whole session leads to unnecessary restrictions on the throughput. To alleviate this problem, we propose a multicast window scheme using a distinct window size for each receiver, and enforcing it as the limit on the number of outstanding packets to that receiver. The complexity of window-based regulation can be defused by a receiver-driven implementation and by consolidation of receiver feedback in successive stages, e.g., using a hierarchical architecture. This hierarchical approach is also useful for scalable consolidation of receiver feedback in the case of rate-based regulation, and for distributed estimation of receiver round trip times, when such estimation is necessary.
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paper_title: The loss path multiplicity problem in multicast congestion control

paper_content:

An important concern for source-based multicast congestion control algorithms is the loss path multiplicity (LPM) problem that arises because a transmitted packet can be lost on one or more of the many end-to-end paths in a multicast tree. Consequently, if a multicast source's transmission rate is regulated according to loss indications from receivers, the rate may be completely throttled as the number of loss paths increases. In this paper, we analyze a family of additive increase multiplicative decrease congestion control algorithms and show that, unless careful attention is paid to the LPM problem, the average session bandwidth of a multicast session may be reduced drastically as the size of the multicast group increases. This makes it impossible to share bandwidth in a max-min fair manner among unicast and multicast sessions. We show that max-min fairness can be achieved however if every multicast session regulates its rate according to the most congested end-to-end path in its multicast tree. We present an idealized protocol for tracking the most congested path under changing network conditions, and use simulations to illustrate that tracking the most congested path is indeed a promising approach.
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paper_title: Promoting the use of end-to-end congestion control in the Internet

paper_content:

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, "not TCP-friendly", or simply using disproportionate bandwidth. A flow that is not "TCP-friendly" is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion.
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paper_title: Tear: TCP emulation at receivers - flow control for multimedia streaming

paper_content:

Congestion and flow control is an integral part of any Internet data transport protocol. It is widely accepted that the congestion avoidance mechanisms of TCP have been one of the key contributors to the success of the Internet. However, TCP is ill-suited to real-time multimedia streaming applications. Its bursty transmission, and abrupt and frequent wide rate fluctuations cause high delay jitters and sudden quality degradation of multimedia applications. For asymmetric networks such as wireless networks, cable modems, ADSL, and satellite networks, transmitting feedback for (almost) every packet received as it is done in TCP causes congestion in the reverse path. In this environment, TCP may severely underutilize the forward path throughput. Use of multicast further complicates the problem; TCP-like frequent feedback from each receiver to the sender in a large scale multicast session cause well-known scalability limitations (e.g. acknowledgment implosion). We have developed a new flow control approach for multimedia streaming, called TCP emulation at receivers (TEAR). TEAR shifts most of flow control mechanisms to receivers. In TEAR, a receiver does not send to the sender the congestion signals detected in its forward path but rather processes them immediately to determine its own appropriate receiving rate. TEAR can determine this rate using congestion signals observed at the receiver. These signals are used to emulate the TCP sender’s flow control functions at receivers.The emulation allows receivers to estimate a TCP-friendly rate for the congestion conditions observed in their forward paths. TEAR also allows receivers to adjust their receiving rates to a TCP-friendly rate without actually modulating the rates to probe for spare bandwidth, or to react to packet losses directly. Thus, the perceived rate fluctuations at the application are much more smooth than in TCP. A unicast version of TEAR is implemented. This report describes the implementation of TEAR, examine the performance of this TEAR implementation from the NS simulation and Internet experiments, and compare it with that of other TCP-friendly flow control techniques. Our preliminary tests indicate that TEAR shows superior fairness to TCP with significantly lower rate fluctuations than TCP. TEAR’s sensitivity to feedback interval is very low, so that even under high feedback latency, TEAR flows exhibit acceptable performance in terms of fairness, TCP-friendliness, and rate fluctuations. Finally, I will discuss the future extension of TEAR for multicast environments. This work is supported in part by NSF CAREER ANI-9875651.
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paper_title: RAP: An end-to-end rate-based congestion control mechanism for realtime streams in the Internet

paper_content:

End-to-end congestion control mechanisms have been critical to the robustness and stability of the Internet. Most of today's Internet traffic is TCP, and we expect this to remain so in the future. Thus, having "TCP-friendly" behavior is crucial for new applications. However, the emergence of non-congestion-controlled realtime applications threatens unfairness to competing TCP traffic and possible congestion collapse. We present an end-to-end TCP-friendly rate adaptation protocol (RAP), which employs an additive-increase, multiplicative-decrease (AIMD) algorithm. It is well suited for unicast playback of realtime streams and other semi-reliable rate-based applications. Its primary goal is to be fair and TCP-friendly while separating network congestion control from application-level reliability. We evaluate RAP through extensive simulation, and conclude that bandwidth is usually evenly shared between TCP and RAP traffic. Unfairness to TCP traffic is directly determined by how TCP diverges from the AIMD algorithm. Basic RAP behaves in a TCP-friendly fashion in a wide range of likely conditions, but we also devised a fine-grain rate adaptation mechanism to extend this range further. Finally, we show that deploying RED queue management can result in an ideal fairness between TCP and RAP traffic.
---
paper_title: Promoting the use of end-to-end congestion control in the Internet

paper_content:

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, "not TCP-friendly", or simply using disproportionate bandwidth. A flow that is not "TCP-friendly" is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion.
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paper_title: Equation-based congestion control for unicast applications

paper_content:

This paper proposes a mechanism for equation-based congestion control for unicast traffic. Most best-effort traffic in the current Internet is well-served by the dominant transport protocol, TCP. However, traffic such as best-effort unicast streaming multimedia could find use for a TCP-friendly congestion control mechanism that refrains from reducing the sending rate in half in response to a single packet drop. With our mechanism, the sender explicitly adjusts its sending rate as a function of the measured rate of loss events, where a loss event consists of one or more packets dropped within a single round-trip time. We use both simulations and experiments over the Internet to explore performance.
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paper_title: Fundamental observations on multicast congestion control in the Internet

paper_content:

We study congestion control for one-to-many multicast applications in the Internet and establish a three-way relationship between the choice of regulation parameter (i.e., rate or window size), the requirement to estimate receiver round trip times, and the type of fairness that may be accomplished. In particular, we show that in order to provide TCP-compatible fairness in rate-based regulation, receiver round trip times must be known. However, such a requirement does not exist in window-based regulation. We further show that measurement of receiver round-trip times in multicast communication, is fundamentally different and more complex than unicast communication, in order to avoid implosion of acknowledgments at the source. A major part of the paper deals with extending window-based regulation to multicast communications. We show that window-based regulation using a common window-size for the whole session leads to unnecessary restrictions on the throughput. To alleviate this problem, we propose a multicast window scheme using a distinct window size for each receiver, and enforcing it as the limit on the number of outstanding packets to that receiver. The complexity of window-based regulation can be defused by a receiver-driven implementation and by consolidation of receiver feedback in successive stages, e.g., using a hierarchical architecture. This hierarchical approach is also useful for scalable consolidation of receiver feedback in the case of rate-based regulation, and for distributed estimation of receiver round trip times, when such estimation is necessary.
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paper_title: A novel loss indication filtering approach for multicast congestion control

paper_content:

A key issue in the design of source-based multicast congestion control schemes is how to aggregate loss indications from multiple receivers into a single rate control decision at the source. Such aggregation entails filtering out a portion of the loss indications received by the source, and then using the remaining for rate adjustments. In this paper, we first propose a set of goals guiding the design of loss indication filters. We then present a novel loss indication filtering approach, the linear proportional response (LPR) approach. Analysis and simulation is used to compare LPR to two well-known approaches - the random listening algorithm (RLA) [Proceedings of ACM SIGCOMM (1998)] and the worst estimate-based tracking (WET) [Proceedings of IEEE Infocom (1999)] approach. Our results indicate that LPR achieves a desirable tradeoff between stability and response, thereby making it more suitable than WET and RLA for deployment in an Internet-like environment.
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paper_title: Achieving bounded fairness for multicast and TCP traffic in the Internet

paper_content:

There is an urgent need for effective multicast congestion control algorithms which enable reasonably fair share of network resources between multicast and unicast TCP traffic under the current Internet infrastructure. In this paper, we propose a quantitative definition of a type of bounded fairness between multicast and unicast best-effort traffic, termed "essentially fair". We also propose a window-based Random Listening Algorithm (RLA) for multicast congestion control. The algorithm is proven to be essentially fair to TCP connections under a restricted topology with equal round-trip times and with phase effects eliminated. The algorithm is also fair to multiple multicast sessions. This paper provides the theoretical proofs and some simulation results to demonstrate that the RLA achieves good performance under various network topologies. These include the performance of a generalization of the RLA algorithm for topologies with different round-trip times.
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paper_title: MTCP: scalable TCP-like congestion control for reliable multicast

paper_content:

We present MTCT, a congestion control scheme for large-scale reliable multicast. Congestion control for reliable multicast is important because of its wide applications in multimedia and collaborative computing, yet nontrivial, because of the potentially large number of receivers involved. Many schemes have been proposed to handle the recovery of lost packets in a scalable manner; but there is little work on the design and implementation of congestion control schemes for reliable multicast. We propose new techniques that can effectively handle instances of congestion occurring simultaneously at various parts of a multicast tree. Our protocol incorporates several novel features: (1) hierarchical congestion status reports that distribute the load of processing feedback from all receivers across the multicast group, (2) the relative time delay (RTD) concept which overcomes the difficulty of estimating round-trip times in tree-based multicast environments, (3) window-based control that prevents the sender from transmitting faster than packets leave the bottleneck link an the multicast path through which the sender's traffic flows, (4) a retransmission window that regulates the flow of repair packets to prevent local recovery from causing congestion, and (5) a selective acknowledgment scheme that prevents independent (i.e., non-congestion-related) packet loss from reducing the sender's transmission rate. We have implemented MTCP both on UDP in SunOS 5.6 and on the simulator ns, and we have conducted extensive Internet experiments and simulation to test the scalability and inter-fairness properties of the protocol. The encouraging results we have obtained support our confidence that TCP-like congestion control for large-scale reliable multicast is within our grasp.
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paper_title: pgmcc: a TCP-friendly single-rate multicast congestion control scheme

paper_content:

We present a single rate multicast congestion control scheme(pgmcc) which is TCP-friendly and achieves scalability, stability and fast response to variations in network conditions. pgmcc is suitable for both non-reliable and reliable data transfers; it uses a window-based TCP-like controller based on positive ACKs and run between the sender and a group's representative, the acker . The innovative part of pgmcc is a fast and low-overhead procedure to select (and track changes of) the acker, which permits us to consider the acker as a moving receiver rather than a changing one. As such, the scheme is robust to measurement errors, and supports fast response to changes in the receiver set and/or network conditions. The scheme has been implemented in the PGM protocol, and the paper presents a number of experimental results on its performance.
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paper_title: TCP-like congestion control for layered multicast data transfer

paper_content:

We present a novel congestion control algorithm suitable for use with cumulative, layered data streams in the MBone. Our algorithm behaves similarly to TCP congestion control algorithms, and shares bandwidth fairly with other instances of the protocol and with TCP flows. It is entirely receiver driven and requires no per-receiver status at the sender, in order to scale to large numbers of receivers. It relies on standard functionalities of multicast routers, and is suitable for continuous stream and reliable bulk data transfer. In the paper we illustrate the algorithm, characterize its response to losses both analytically and by simulations, and analyse its behaviour using simulations and experiments in real networks. We also show how error recovery can be dealt with independently from congestion control by using FEC techniques, so as to provide reliable bulk data transfer.
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paper_title: A digital fountain approach to reliable distribution of bulk data

paper_content:

The proliferation of applications that must reliably distribute bulk data to a large number of autonomous clients motivates the design of new multicast and broadcast protocols. We describe an ideal, fully scalable protocol for these applications that we call a digital fountain. A digital fountain allows any number of heterogeneous clients to acquire bulk data with optimal efficiency at times of their choosing. Moreover, no feedback channels are needed to ensure reliable delivery, even in the face of high loss rates.We develop a protocol that closely approximates a digital fountain using a new class of erasure codes that for large block sizes are orders of magnitude faster than standard erasure codes. We provide performance measurements that demonstrate the feasibility of our approach and discuss the design, implementation and performance of an experimental system.
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paper_title: FLID-DL: congestion control for layered multicast

paper_content:

We describe Fair Layered Increase/Decrease with Dynamic Layering (FLID-DL), a new multi-rate congestion control algorithm for layered multicast sessions. FLID-DL generalizes the receiver-driven layered congestion (RLC) control protocol introduced by Vicisano, Rizzo, and Crowcroft, ameliorating the problems associated with large IGMP leave latencies and abrupt rate increases. Like RLC, FLID-DL is a scalable, receiver-driven congestion control mechanism in which receivers add layers at sender-initiated synchronization points and leave layers when they experience congestion. FLID-DL congestion control coexists with TCP flows as well as other FLID-DL sessions and supports general rates on the different multicast layers. We demonstrate via simulations that our congestion control scheme exhibits better fairness properties and provides better throughput than previous methods. A key contribution that enables FLID-DL and may be useful elsewhere is Dynamic Layering (DL), which mitigates the negative impact of long IGMP leave latencies and eliminates the need for probe intervals present in RLC. We use DL to respond to congestion much faster than IGMP leave operations, which have proven to be a bottleneck for prior work.
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paper_title: Error control for video multicast using hierarchical FEC

paper_content:

Bit-rate scalable video compression with layered multicast has been shown to be an effective method to achieve rate control in heterogeneous networks. We further propose the use of hierarchical FEC as an error control mechanism that allows receivers to individually trade-off latency for received video quality. The scheme is efficient since FEC packets are used to protect only the more important data layers and is multicast only to receivers that need them, thereby improving network utilization. Furthermore, there is no loss in error correcting capability by using hierarchical FEC when maximum distance separable codes are used. Actual MBONE experiments are performed to evaluate the performance of the proposed scheme.
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paper_title: MLDA: a TCP-friendly congestion control framework for heterogeneous multicast environments

paper_content:

To avoid overloading the Internet and starving TCP connections, multimedia flows using non-congestion controlled UDP need to be enhanced with congestion control mechanisms. In this paper, we present a general framework for achieving TCP-friendly congestion control called MLDA. Using MLDA, multimedia senders adjust their transmission rate in accordance with the network congestion state. Taking the heterogeneity of the Internet and the end systems into account, MLDA supports layered data transmission where the shape and number of the layers is determined dynamically based on feedback information generated by the receivers. Performance tests of MLDA as well as comparisons to other congestion control schemes suggest that MLDA achieves TCP-friendly congestion control on the one hand and is able to accommodate the needs of heterogeneous receivers on the other.
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paper_title: A window-based congestion control for reliable multicast based on TCP dynamics

paper_content:

The limitation of the current multicast model led to the development of embedded network assist and new service models for reliable transport. We show a viable solution for one of the hardest problems in reliable multicast — congestion control — through the deployment of a new forwarding service model, “Breadcrumb forwarding service” (BCFS). Our proposed reliable multicast transport, “Rainbow”, is built on top of this model. In our approach, each receiver maintains its own congestion window and individually runs window control modeled after TCP. To enhance Rainbow's scalability and support asynchronous receiver subscriptions, Rainbow utilizes Digital Fountain at the source. This allows receivers to exercise asynchronous and autonomous behavior while simultaneously enjoying the performance benefit of synchronous multicast communication with fast group establishment of BCFS. In this paper, we detail the congestion control of Rainbow and demonstrate its efficiency and scalability through simulation and analysis. According to simulation results, Rainbow shows more TCP-fair behavior than RLC, which is a TCP friendly congestion control scheme based on layered multicast.
---
paper_title: Modeling TCP Reno performance: a simple model and its empirical validation

paper_content:

The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization of the steady-state send rate as a function of loss rate and round trip time (RTT) for a bulk transfer TCP flow. Unlike the models of Lakshman and Madhow (see IEE/ACM Trans. Networking, vol.5, p.336-50, 1997), Mahdavi and Floyd (1997), Mathis, Semke, Mahdavi and Ott (see Comput. Commun. Rev., vol.27, no.3, 1997) and by by Ott et al., our model captures not only the behavior of the fast retransmit mechanism but also the effect of the time-out mechanism. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP send rate and is accurate over a wider range of loss rates. We also present a simple extension of our model to compute the throughput of a bulk transfer TCP flow, which is defined as the amount of data received by the receiver in unit time.
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paper_title: TCP congestion control with a misbehaving receiver

paper_content:

In this paper, we explore the operation of TCP congestion control when the receiver can misbehave, as might occur with a greedy Web client. We first demonstrate that there are simple attacks that allow a misbehaving receiver to drive a standard TCP sender arbitrarily fast, without losing end-to-end reliability. These attacks are widely applicable because they stem from the sender behavior specified in RFC 2581 rather than implementation bugs. We then show that it is possible to modify TCP to eliminate this undesirable behavior entirely, without requiring assumptions of any kind about receiver behavior. This is a strong result: with our solution a receiver can only reduce the data transfer rate by misbehaving, thereby eliminating the incentive to do so.
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Title: A Survey on TCP-Friendly Congestion Control

Section 1: Motivation and Outline
Description 1: Describe the importance of TCP-friendly congestion control, the unfairness of non-TCP applications, and an overview of the survey.

Section 2: TCP and TCP-Friendliness
Description 2: Explain TCP’s congestion control mechanism and provide a definition of TCP-friendliness for unicast and multicast flows.

Section 3: Classification of Congestion Control Schemes
Description 3: Discuss various ways to classify TCP-friendly congestion control schemes.

Section 4: Window-Based versus Rate-Based
Description 4: Classify and explain TCP-friendly schemes based on their use of congestion windows or transmission rates.

Section 5: Unicast versus Multicast
Description 5: Describe TCP-friendliness for unicast and multicast traffic and the challenges involved in multicast congestion control.

Section 6: Single-rate versus Multi-rate
Description 6: Explain the distinction between single-rate and multi-rate congestion control approaches, particularly for multicast.

Section 7: End-to-end versus Router-supported
Description 7: Discuss the differences between end-to-end congestion control and those that require router support.

Section 8: Single-Rate Congestion Control Protocols
Description 8: Present and describe a selection of single-rate congestion control protocols and their mechanisms.

Section 9: Multi-rate Congestion Control Protocols
Description 9: Present and describe a selection of multi-rate congestion control protocols and their mechanisms.

Section 10: Protocol Evaluation
Description 10: Evaluate the suitability of different congestion control mechanisms based on network characteristics and application requirements.

Section 11: Areas of Future Research
Description 11: Identify unresolved issues and areas requiring further research in TCP-friendly congestion control.

Section 12: Conclusions
Description 12: Summarize the findings of the survey and discuss the importance of TCP-friendly congestion control in maintaining Internet efficiency.