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.

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.

Instead of the increase-by-one decrease-to-half strategy used in TCP Reno for congestion window adjustment, we consider the general case such that the increase value and decrease ratio are parameters. That is, in the congestion avoidance state, the window size is increased by ff per window of packets acknowledged and it is decreased to fi of the current value when there is congestion indication. We refer to this window adjustment strategy as general additive increase multiplicative decrease (GAIMD). We present the (mean) sending rate of a GAIMD flow as a function of ff, fi, loss rate, mean roundtrip time, mean timeout value, and the number of packets acknowledged by each ACK. We conducted extensive experiments to validate this sending rate formula. We found the formula to be quite accurate for a loss rate of up to 20%. We also present in this paper a simple relationship between ff and fi for a GAIMD flow to be TCP-friendly, that is, for the GAIMD flow to have approximately the same sending rate as a TCP flow under the same path conditions.

New trends in communication, in particular the deployment of multicast and real-time audio/video streaming applications, are likely to increase the percentage of non-TCP traffic in the Internet. These applications rarely perform congestion control in a TCP-friendly manner, i.e., they do not share the available bandwidth fairly with applications built on TCP, such as web browsers, FTP- or email-clients. The Internet community strongly fears that the current evolution could lead to a congestion collapse and starvation of TCP traffic. For this reason, TCP-friendly protocols are being developed that behave fairly with respect to co-existent TCP flows. In this article, we present a survey of current approaches to TCP-friendliness and discuss their characteristics. Both unicast and multicast congestion control protocols are examined, and an evaluation of the different approaches is presented.

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. For 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. Further, we discuss a measurement approach that allows receivers in large multicast sessions to estimate the round trip delay estimation to the sender in a scalable way. For exchanging control information between the sender and receivers we investigate the possibility of using the real time transport protocol (RTP) and discuss the required changes in order for RTP ...

This paper introduces Wave and Equation Based Rate Control (WEBRC), the first multiple rate multicast congestion control protocol to be equation based. The equation-based approach enforces fairness to TCP with the benefit that fluctuations in the flow rate are small in comparison to TCP.

This paper uses analysis and experiments to study the impact of various congestion control algorithms and receiver buffering strategies on the performance of streaming media delivery. While traditional congestion avoidance schemes such as TCP's additive-increase/- multiplicative-decrease (AIMD) achieve high utilization, they also cause large oscillations in transmission rates that degrade the smoothness and perceptual quality of the video stream. We focus on understanding the interactions of a family of congestion control algorithms that generalize AIMD, with buffer-based quality adaptation algorithms for hierarchically-encoded and simulcast video. Our work builds on and extends the results of Rejaie et al. [19]; we find that the combination of a non-AIMD algorithm that has smaller oscillations than AIMD and a suitable receiver buffer allocation and management strategy provides a good combination of low playout delay and TCP-friendly congestion control. The paper describes these mechanisms and the results of experiments conducted using a prototype video server for MPEG-4 video, showing that our approach can improve the interactivity and adaptivity of Internet video.

Abstract — In this paper, we present an end-to-end adaptation scheme, called the enhanced loss-delay based adaptation algorithm (LDA+) for regulating the transmission behavior of multimedia senders in accordance with the network congestion state. LDA+ uses the real-time transport protocol (RTP) for collecting loss and delay statistics which are then used for adjusting the transmission behavior of the senders in a manner similar to TCP connections suffering from equal losses and delays. The performance of LDA+ is then investigated by running several simulations as well as measurements over the Internet. Additionally, by conducting simulations, the performance of LDA+ is compared to that of other TCP-friendly congestion control schemes presented in the literature. I.

This paper presents a trace-driven simulation study of three classes of retransmission timeout (RTO) estimators in the context of low-bitrate real-time streaming over the Internet. We explore the viability of employing retransmission timeouts in NACK-based real-time streaming applications that support multiple retransmission attempts per lost packet. In such applications, real-time RTO estimation plays a major role (i.e., poor RTO estimation results in a larger number of duplicate packets and sometimes more frequent underflow events). Our study is based on trace data collected during a number of real-time streaming tests conducted between our dialup clients in all 50 states of the U.S. (including 653 major U.S. cities) and our backbone video server during a seven-month period. First, we define a generic performance measure for assessing the quality of hypothetical RTO estimators based on the samples of the roundtrip delay (RTT) recorded in the trace data. Second, using this performance measure, we evaluate the class of TCP-like estimators, find the most optimal estimator given our performance measure, and establish power laws that describe the tradeoff between the optimal number of duplicate packets and the optimal timeout waiting time. Third, we introduce a new class of RTO estimators based on delay jitter and show that they perform significantly better than TCP-like estimators in NACKbased applications. Finally, we gain a major insight into the RTT process by establishing which tuning parameters of an RTO estimator make it optimal given our performance measure and our experimental data, and give our explanation of the observed phenomena. I.

Video transmission over the current best-effort Internet should be made fair to com- peting TCP traffic. Because the available bandwidth to a TCP-friendly stream changes significantly over medium and long time scales, it is desirable to adapt the streaming rate of the video to the current bandwidth conditions. We consider two adaptive streaming schemes for stored video. The first scheme switches among multiple encoded versions, with each version encoded at a different rate. The second scheme adds and drops encoding layers. To compare the two schemes, we develop streaming control policies for each scheme and evaluate their performance using trace-driven simulation. Our results show that when analogous streaming policies are used, switching versions outperforms adding/dropping layers because of the overhead associated with layering. However, the enhanced flexibility of layering can compensate the performance degradation due to the layering overhead.