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. We consider equation-based congestion control a promising avenue of development for congestion control of multicast traffic, and so an additional motivation for this work is to lay a sound basis for the further development of multicast congestion control.

Abstract-End-to-end congestion control mechanisms have been critical to the robustness and stability of the Internet. Most of today’s Internet trafftc 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, multiplicativedecrease (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 TCPfriendly 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. I.

We study the problem of media access control in the novel regime of sensor networks, where unique application behavior and tight constraints in computation power, storage, energy resources, and radio technology have shaped this design space to be very different from that found in traditional mobile computing regime. Media access control in sensor networks must not only be energy efficient but should also allow fair bandwidth allocation to the infrastructure for all nodes in a multihop network. We propose an adaptive rate control mechanism aiming to support these two goals and find that such a scheme is most effective in achieving our fairness goal while being energy effcient for both low and high duty cycle of network traffic.

Abstract—Several analytic models describe the steady-state throughput of bulk transfer TCP flows as a function of round trip time and packet loss rate. These models describe flows based on the assumption that they are long enough to sustain many packet losses. However, most TCP transfers across today’s Internet are short enough to see few, if any, losses and consequently their performance is dominated by startup effects such as connection establishment and slow start. This paper extends the steadystate model proposed in [34] in order to capture these startup effects. The extended model characterizes the expected value and distribution of TCP connection establishment and data transfer latency as a function of transfer size, round trip time, and packet loss rate. Using simulations, controlled measurements of TCP transfers, and live Web measurements we show that, unlike earlier steady-state models for TCP performance, our extended model describes connection establishment and data transfer latency under a range of packet loss conditions, including no loss. I.

This paper introduces and analyzes a class of nonlinear congestion control algorithms called binomial algorithms, motivated in part by the needs of streaming audio and video applications for which a drastic reduction in transmission rate upon each congestion indication (or loss) is problematic. Binomial algorithms generalize TCP-style additive-increase by increasing inversely proportional to a power of the current window (for TCP, ) ; they generalize TCP-style multiplicative-decrease by decreasing proportional to a power of the current window (for TCP, ). We show that there are an infinite number of deployable TCP-compatible binomial algorithms, those which satisfy , and that all binomial algorithms converge to fairness under a synchronized-feedback assumption provided . Our simulation results show that binomial algorithms interact well with TCP across a RED gateway. We focus on two particular algorithms, IIAD ( ) and SQRT ( !" ), showing that they are well-suited to applications that do not react well to large TCP-style window reductions. Keywords--- Congestion control, TCP-friendliness, TCP-compatibility, nonlinear algorithms, transport protocols, TCP, streaming media, Internet. I.

We present a framework for designing end-to-end congestion control schemes in a network where each user may have a different utility function and may experience non-congestion-related losses. We first show that there exists an additive increase-multiplicative decrease scheme using only end-to-end measurable losses such that a socially-optimal solution can be reached. We incorporate round-trip delay in this model, and show that one can generalize observations regarding TCP-type congestion avoidance to more general window flow control schemes. We then consider explicit congestion notification (ECN) as an alternate mechanism (instead of losses) for signaling congestion and show that ECN marking levels can be designed to nearly eliminate losses in the network by choosing the marking level independently for each node in the network. While the ECN marking level at each node may depend on the number of flows through the node, the appropriate marking level can be estimated using only aggregate flow measurements, i.e., per-flow measurements are not required. 1

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. 1

Streaming audio and video applications are becoming increasingly popular on the Internet, and the lack of effective congestion control in such applications is now a cause for significant concern. The problem is one of adapting the compression without requiring video-servers to re-encode the data, and fitting the resulting stream into the rapidly varying available bandwidth. At the same time, rapid fluctuations in quality will be disturbing to the users and so should be avoided. In this paper we present a mechanism for using layered video in the context of unicast congestion control. This quality adaptation mechanism adds and drops layers of the video stream to perform long-term coarse-grain adaptation, while using a TCP-friendly congestion control mechanism to react to congestion on very short timescales. The mismatches between the two timescales are absorbed using buffering at the receiver. We present a piecewiseoptimal scheme for the distribution of buffering among the active layers in order to maximize perceptual quality while minimizing rapid, disturbing changes in the quality. We discuss the issues involved in implementing and tuning such a mechanism, and present our simulation and experimental results.

Abstract—Due to the explosive growth of the Internet and increasing demand for multimedia information on the web, streaming video over the Internet has received tremendous attention from academia and industry. Transmission of real-time video typically has bandwidth, delay, and loss requirements. However, the current best-effort Internet does not offer any quality of service (QoS) guarantees to streaming video. Furthermore, for video multicast, it is difficult to achieve both efficiency and flexibility. Thus, Internet streaming video poses many challenges. To address these challenges, extensive research has been conducted. This special issue is aimed at dissemination of the contributions in the field of streaming video over the Internet. To introduce this special issue with the necessary background and provide an integral view on this field, we cover six key areas of streaming video. Specifically, we cover video compression, application-layer QoS control, continuous media distribution services, streaming servers, media synchronization mechanisms, and protocols for streaming media. For each area, we address the particular issues and review major approaches and mechanisms. We also discuss the tradeoffs of the approaches and point out future research directions. Index Terms—Application-layer QoS control, continuous media distribution services, Internet, protocol, streaming video,

FIFO queueing is simple but does not protect traffic from flows that send more than their share or flows that fail to use end-to-end congestion control. At the other extreme, per-flow scheduling mechanisms provide max-min fairness but are more complex, keeping state for all flows going through the router. This paper proposes RED-PD (RED with Preferential Dropping), a flow-based mechanism that combines simplicity and protection by keeping state for just the high-bandwidth flows. RED-PD uses the packet drop history at the router to detect high-bandwidth flows in times of congestion and preferentially drop packets from these flows. This paper discusses the design decisions underlying RED-PD, and presents simulations evaluating RED-PD in a range of environments.