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.

Abstract—Current Internet congestion control protocols operate independently on a per-flow basis. Recent work has demonstrated that cooperative congestion control strategies between flows can improve performance for a variety of applications, ranging from aggregated TCP transmissions to multiple-sender multicast applications. However, in order for this cooperation to be effective, one must first identify the flows that are congested at the same set of resources. In this paper, we present techniques based on loss or delay observations at end hosts to infer whether or not two flows experiencing congestion are congested at the same network resources. Our novel result is that such detection can be achieved for unicast flows, but the techniques can also be applied to multicast flows. We validate these techniques via queueing analysis, simulation, and experimentation within the Internet. In addition, we demonstrate preliminary simulation results that show that the delay-based technique can determine whether two TCP flows are congested at the same set of resources. We also propose metrics that can be used as a measure of the amount of congestion sharing between two flows. Index Terms—Hypothesis testing, inference, network congestion, queueing analysis. I.

rare event on some Internet paths. Reordering can cause performance problems for TCP's fast retransmission algorithm, which uses the arrival of duplicate acknowledgments to detect segment loss. Duplicate acknowledgments can be caused by the loss of a segment or by the reordering of segments by the network. In this paper we illustrate the impact of reordering on TCP performance. In addition, we show the performance of a conservative approach to "undo" the congestion control state changes made in conjunction with spurious retransmissions. Finally, we propose several alternatives to dynamically make the fast retransmission algorithm more tolerant of the reordering observed in the network and assess these algorithms.

Stream processing fits a large class of new applications for which conventional DBMSs fall short. Because many stream-oriented systems are inherently geographically distributed and because distribution offers scalable load management and higher availability, future stream processing systems will operate in a distributed fashion. They will run across the Internet on computers typically owned by multiple cooperating administrative domains. This paper describes the architectural challenges facing the design of large-scale distributed stream processing systems, and discusses novel approaches for addressing load management, high availability, and federated operation issues. We describe two stream processing systems, Aurora * and Medusa, which are being designed to explore complementary solutions to these challenges. 1

Although IP Multicast is an effective network primitive for best-effort, large-scale, multi-point communication, many multicast applications such as shared whiteboards, multi-player games and software distribution require reliable data delivery. Building services like reliable sequenced delivery on top of IP Multicast has proven to be a hard problem. The enormous extent of network and end-system heterogeneity in multipoint communication exacerbates the design of scalable end-to-end reliable multicast protocols. In this paper, we propose a radical departure from the traditional end-to-end model for reliable multicast and instead propose a hybrid approach that leverages the successes of unicast reliability protocols such as TCP while retaining the efficiency of IP multicast for multi-point data delivery. Our approach splits a large heterogeneous reliable multicast session into a number of multicast data groups of co-located homogeneous participants. A collection of application-aware agents--Reliable Multicast proxies (RMXs)--organizes these data groups into a spanning tree using an overlay network of TCP connections. Sources transmit data to their local group, and the RNLX in that group forwards the data towards the rest of the data groups. RMXs use detailed knowledge of application semantics to adapt to the effects of heterogeneity in the environment. To demonstrate the efficacy of our architecture, we have built a prototype implementation that can be customized for different kinds of applications.

Due to the availability of a wide variety of wireless access technologies, a mobile host can potentially have subscriptions and access to more than one wireless network at a given time. In this paper, we consider such a multi-homed mobile host, and address the problem of achieving bandwidth aggregation by striping data across the multiple interfaces of the mobile host. We show that both link layer striping approaches and application layer techniques that stripe data across multiple TCP sockets do not achieve the optimal bandwidth aggregation due to a variety of factors specific to wireless networks. We propose an end-to-end transport layer approach called pTCP that effectively performs bandwidth aggregation on multi-homed mobile hosts. We show through simulations that pTCP achieves the desired goals under a variety of network conditions.

\Soft state " is an often cited yet vague concept in network protocol design in which two or more network entities intercommunicate in a loosely coupled, often anonymous fashion. Researchers often de ne this concept operationally (if at all) rather than analytically: a source of soft state transmits periodic \refresh messages " over a (lossy) communication channel to one or more receivers that maintain a copy of that state, which in turn \expires " if the periodic updates cease. Though a number of crucial Internet protocol building blocks are rooted in soft state-based designs | e.g., RSVP refresh messages, PIM membership updates, various routing protocol updates, RTCP control messages, directory services like SAP, and so forth | controversy is building as to whether the performance overhead of soft state refresh messages justify their qualitative bene t of enhanced system \robustness". We believe that this controversy has risen not from fundamental performance tradeo s but rather from our lack of a comprehensive understanding of soft state. To better understand these tradeo s, we propose herein a formal model for soft state communication based on a probabilistic delivery model with relaxed reliability. Using this model, we conduct queueing analysis and simulation to characterize the data consistency and performance tradeo s under a range of workloads and network loss rates. We then extend our model with feedback and show, through simulation, that adding feedback dramatically improves data consistency (by up to 55%) without increasing network resource consumption. Our model not only provides a foundation for understanding soft state, but also induces a new fundamental transport protocol based on probabilistic delivery. Toward this end, we sketch our design of the \Soft State Transport Protocol " (SSTP), which enjoys the robustness of soft state while retaining the performance bene t of hard state protocols like TCP through its judicious use of feedback. This research was supported by DARPA contract N66001-96-C-8508, by the State of California under the MICRO program, and by

This document specifies TCP-Friendly Rate Control (TFRC). TFRC is a congestion control mechanism for unicast flows operating in a best-effort Internet environment. It is reasonably fair when competing for bandwidth with TCP flows, Handley/Padhye/Floyd/Widmer [Page 1] INTERNET-DRAFT Expires: September 2001 March 2001 but has a much lower variation of throughput over time compared with TCP, making it more suitable for applications such as telephony or streaming media where a relatively smooth sending rate is of importance. Handley/Padhye/Floyd/Widmer [Page 2] INTERNET-DRAFT Expires: September 2001 March 2001 Table of Contents 1. Introduction. . . . . . . . . . . . . . . . . . . . . . 4 2. Terminology . . . . . . . . . . . . . . . . . . . . . . 5 3. Protocol Mechanism. . . . . . . . . . . . . . . . . . . 5 3.1. TCP Throughput Equation. . . . . . . . . . . . . . . 5 3.2. Packet Contents. . . . . . . . . . . . . . . . . . . 7 3.2.1. Data Packets. . . . . . . . . . . . . . . . . . . 7 3....

IRON FILE SYSTEMSVijayan Prabhakaran Disk drives are widely used as a primary medium for storing information.While commodity file systems trust disks to either work or fail completely, modern disks exhibit complex failure modes such as latent sector faults and block corrup-tions, where only portions of a disk fail.

The delivery of multimedia content is a facet of Internet traffic that is rapidly growing in importance. The new generation of World Wide Web sites are relying heavily on extensive multimedia content such as graphics, sound, music and video to attract and retain visitors. While there have been extensive studies on the growth and effects of Hyper Text Transfer Protocol (HTTP) traffic used on the Web, little or no work has been performed in analyzing streaming multimedia traffic. We present the results of a brief study to examine the traffic emanating from a popular Internet audio service using the RealAudio program. We found protocol distributions that show a bias towards non-TCP friendly protocols. In addition, we observed consistencies in audio traffic packet sizes and data rate patterns may be useful as a tool for identifying audio data flows. Our results show that audio flows exhibit significant consistency in data rates and are considerably more persistent than HTTP connections. Index terms—multimedia, streaming audio, internet traffic 1.