In this paper, we analyze a performance model for the TCP Congestion Avoidance algorithm. The model predicts the bandwidth of a sustained TCP connection subjected to light to moderate packet losses, such as loss caused by network congestion. It assumes that TCP avoids retransmission timeouts and always has sufficient receiver window and sender data. The model predicts the Congestion Avoidance performance of nearly all TCP implementations under restricted conditions and of TCP with SelectiveAcknowledgements over a much wider range of Internet conditions. We verify

The quality of available network connections can often have a large impact on the performance of distributed applications. For example, document transfer applications such as FTP, Gopher and the World Wide Web suffer increased response times as a result of network congestion. For these applications, the document transfer time is directly related to the available bandwidth of the connection. Available bandwidth depends on two things: 1) the underlying capacity of the path from client to server, which is limited by the bottleneck link; and 2) the amount of other traffic competing for links on the path. If measurements of these quantities were available to the application, the current utilization of connections could be calculated. Network utilization could then be usedasabasis for selection from a set of alternative connections or servers, thus providing reduced response time. Such a dynamic server selection scheme would beespecially important in a mobile computing environment in which the set of available servers is frequently changing. In order to

Reliable multicast communication is important in large-scale distributed applications. For example, reliable multicast is used to transmit terrain and environmental updates in distributed simulations. To date, proposed protocols have not supported these applications' requirements, which include wide-area data distribution, low-latency packet loss detection and recovery, and minimal data and management overhead within fine-grained multicast groups, each containing a single data source.

ly, the remote procedure call problem, which an RPC protocol undertakes to solve, consists of emulating LPC using message passing. LPC has a number of "properties" -- a single procedure invocation results in exactly one execution of the procedure body, the result returned is reliably delivered to the invoker, and exceptions are raised if (and only if) an error occurs. Given a completely reliable communication environment, which never loses, duplicates, or reorders messages, and given client and server processes that never fail, RPC would be trivial to solve. The sender would merely package the invocation into one or more messages, and transmit these to the server. The server would unpack the data into local variables, perform the desired operation, and send back the result (or an indication of any exception that occurred) in a reply message. The challenge, then, is created by failures. Were it not for the possibility of process and machine crashes, an RPC protocol capable of overcomi...

The more information about current network conditions available to a transport protocol, the more efficiently it can use the network to transfer its data. In networks such as the Internet, the transport protocol must often form its own estimates of network properties based on measurements performed by the connection endpoints. We consider two basic transport estimation problems: determining the setting of the retransmission timer (RTO) for a reliable protocol, and estimating the bandwidth available to a connection as it begins. We look at both of these problems in the context of TCP, using a large TCP measurement set [Pax97b] for trace-driven simulations. For RTO estimation, we evaluate a number of different algorithms, finding that the performance of the estimators is dominated by their minimum values, and to a lesser extent, the timer granularity, while being virtually unaffected by how often round-trip time measurements are made or the settings of the parameters in the exponentially-weighted moving average estimators commonly used. For bandwidth estimation, we explore techniques previously sketched in the literature [Hoe96, AD98] and find that in practice they perform less well than anticipated. We then develop a receiver-side algorithm that performs significantly better. 1

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.

Measuring the bottleneck link bandwidth along a path is important for understanding the performance of many Internet applications. Existing tools to measure bottleneck bandwidth are relatively slow, can only measure bandwidth in one direction, and/or actively send probe packets. We present the nettimer bottleneck link bandwidth measurement tool, the libdpcap distributed packet capture library, and experiments quantifying their utility. We test nettimer across a variety of bottleneck network technologies ranging from 19.2Kb/s to 100Mb/s, wired and wireless, symmetric and asymmetric bandwidth, across local area and crosscountry paths, while using both one and two packet capture hosts. In most cases, nettimer has an error of less than 10%, but at worst has an error of 40%, even on cross-country paths of 17 or more hops. It converges within 10KB of the first large packet arrival while consuming less than 7% of the network traffic being measured.

The scenario is that a bulk data transfer is being performed over a TCP connection, from a host on a local area network (LAN) to a mobile host attached to the LAN by a radio link. In earlier work [10] we had assumed that packet losses in a TCP connection over a radio link are statistically independent. In this paper, we extend this analysis to a Rayleigh fading link, which we model by a two state Markov model. The bulk throughputs of TCP-OldTahoe and TCP-Tahoe are compared with and without fading, for various average signal-to-noise ratios. We also study the performance with a link protocol on the wireless link, and study the effect of varying the link packet size, the number of link packet attempts, and the vehicle speed. For the parameters of the BSD UNIX implementation, over a 1.5Mbps wireless link, we find that, with fading, a signal-to-noise ratio of at least 30dB is required to get reasonable throughput with TCP Tahoe or OldTahoe; this corresponds to at least 100 ti...

Multimedia communication can be supported in an integrated-services network in the general framework of realtime communication. The Tenet Group has devised an approach that provides some initial solutions to the realtime communication problem. This paper attempts to identify the principles behind these solutions. We also describe a suite of protocols, and their implementations in several environments, that embody these principles, and work in progress that will lead towards more complete solutions. This research was supported by the National Science Foundation and the Defense Advanced Research Projects Agency (DARPA) under CooperativeAgreement NCR-8919038 with the Corporation for National Research Initiatives, by AT&T Bell Laboratories, Digital Equipment Corporation, Hitachi, Ltd., Hitachi America, Ltd., Pacific Bell, the University of California under a MICRO grant, and the International Computer Science Institute. The views and conclusions contained in this document are those of th...

Replication is a commonly proposed solution to problems of scale associated with distributed services. However, when a service is replicated, each client must be assigned a server. Prior work has generally assumed that assignment to be static. In contrast, we propose dynamic server selection, and show that it enables application-level congestion avoidance. To make