As networked multimedia applications become widespread, it becomes increasingly important to ensure that these applications can coexist with current TCP-based applications. The TCP protocol is designed to reduce its sending rate when congestion is detected. Networked multimedia applications should exhibit similar behavior, if they wish to co-exist with TCP-based applications [9]. Using TCP for multimedia applications is not practical, since the protocol combines error control and congestion control, an appropriate combination for non-real time reliable data transfer, but inappropriate for loss-tolerant real time applications. In this paper we present a protocol that operates by measuring loss rates and round trip times and then uses them to set the transmission rate to that which TCP would achieve under similar conditions. The analysis in [13] is used to determine this "TCP-friendly" rate. This protocol represents a rst step towards developing a comprehensive protocol for congestion control for time-sensitive multimedia data streams. We evaluate the protocol under various tra c conditions, using simulations and implementation. The simulations are used to study the behavior of the protocol under controlled conditions. The implementation and experimentation involve over 300 experiments over the Internet, using several machines in the US and UK. Our experimental and simulation results show that the protocol is fair to TCP and to other sessions running TFRCP, and that the formula-based approach to achieving TCP-friendliness is indeed practical.

We address the question of how well end-to-end transport connections perform in a satellite environment composed of one or more satellites in geostationary orbit (GEO) or low-altitude earth orbit (LEO), in which the connection may traverse a portion of the wired Internet. We first summarize the various ways in which latency and asymmetry can impair the performance of the Internet's Transmission Control Protocol (TCP), and discuss extensions to standard TCP that alleviate some of these performance problems. Through analysis, simulation, and experiments, we quantify the performance of state-of-the-art TCP implementations in a satellite environment. A key part of the experimental method is the use of traffic models empirically derived from Internet traffic traces. We identify those TCP implementations that can be expected to perform reasonably well, and those that can suffer serious performance degradation. An important result is that, even with the best satellite-optimized TCP implementations, moderate levels of congestion in the wide-area Internet can seriously degrade

End-to-end congestion control mechanisms such as those in TCP are not enough to prevent congestion collapse in the Internet (for starters, not all applications might be willing to use them), and they must be supplemented by control mechanisms inside the network. The IRTF has singled out Random Early Detection (RED) as one queue management scheme recommended for rapid deployment throughout the Internet. However, RED is not a thoroughly understood scheme -- witness for example how the recommended parameter settings, or even the various benefits RED is claimed to provide, have changed over the past few years. In this paper, we describe simple analytic models for RED, and use these models to quantify the benefits (or lack thereof) brought about by RED. In particular, we examine the impact of RED on the loss and delay suffered by bursty and less bursty traffic (such as TCP and UDP traffic, respectively) . We find that (i) RED does eliminate the higher loss bias against bursty traffic obser...

Diagnosing faults in the Internet is arduous and time-consuming, in part because the network is composed of diverse components spread across many administrative domains. We consider an extreme form of this problem: can end users, with no special privileges, identify and pinpoint faults inside the network that degrade the performance of their applications? To answer this question, we present both an architecture for user-level Internet path diagnosis and a practical tool to diagnose paths in the current Internet. Our architecture requires only a small amount of network support, yet it is nearly as complete as analyzing a packet trace collected at all routers along the path. Our tool, tulip, diagnoses reordering, loss and significant queuing events by leveraging well deployed but little exploited router features that approximate our architecture. Tulip can locate points of reordering and loss to within three hops and queuing to within four hops on most paths that we measured. This granularity is comparable to that of a hypothetical network tomography tool that uses 65 diverse hosts to localize faults on a given path. We conclude by proposing several simple changes to the Internet to further improve its diagnostic capabilities.

Improving the performance of data transfers in the Internet (such as Web transfers) requires a detailed understanding of when and how delays are introduced. Unfortunately, the complexity of data transfers like those using HTTP is great enough that identifying the precise causes of delays is difficult. In this paper we describe a method for pinpointing where delays are introduced into applications like HTTP by using critical path analysis. By constructing and pro ling the critical path, it is possible to determine what fraction of total transfer latency is due to packet propagation, network variation (e.g., queuing at routers or route uctuation), packet losses, and delays at the server and at the client. We have implemented our technique in a tool called tcpeval that automates critical path analysis for Web transactions. We show that our analysis method is robust enough to analyze traces taken for two different TCP implementations (Linux and FreeBSD). To demonstrate the utility of our approach, we present the results of critical path analysis for a set of Web transactions taken over 14 days under a variety of server and network conditions. The results show that critical path analysis can shed considerable light on the causes of delays in Web transfers, and can expose subtleties in the behavior of the entire end-to-end system.

The packet losses imposed by IP networks can cause long and erratic recovery delays, since senders must often use conservative loss detection and retransmission mechanisms. This paper proposes a model to explain and predict loss rates for TCP traffic. Based on that model, the paper describes a new router buffering algorithm, Flow-Proportional Queuing (FPQ), that handles heavy TCP loads without imposing high loss rates. FPQ controls TCP by varying the router's queue length in proportion to the number of active TCP connections. Simulation results show that FPQ produces the same average transfer delays as existing schemes, but makes the delays more predictable and fairer.

This article discusses the problems arising when the TCP/IP protocol suite is used to provide Internet connectivity over existing and emerging wireless links. Due to the strong drive towards wireless Internet access through mobile terminals, these problems must be carefully studied in order to build improved systems. We review wireless link characteristics using Wireless LANs and Cellular Communications systems as examples. We then outline the performance problems of the TCP/IP protocol suite when employed over those links, such as degraded TCP performance due to mistaking wireless errors for congestion. We present various proposals for solving these problems and examine their benefits and limitations. Finally, we consider the future evolution of wireless systems and the challenges that emerging systems will impose on the Internet protocol suite.

We present a comprehensive set of measurements of a 2.4 GHz DSSS wireless LAN and analyze its behavior. We examine issues such as host and interface heterogeneity, bidirectional (TCP) traffic and error modeling, that have not been previously analyzed. We uncover multiple problems with TCP and UDP performance in this system. We investigate the causes of these problems (radio hardware, device drivers, network protocols) and discuss the effectiveness of proposed improvements.

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The space communication environment and mobile and wireless communication environments show many similarities when observed from the perspective of a transport protocol. Both types of environments exhibit loss caused by data corruption and link outage, in addition to congestion-related loss. The constraints imposed by the two environments are also similar --- power, weight, and physical volume of equipment are scarce resources. Finally, it is not uncommon for communication channel data rates to be severely limited and highly asymmetric. We are working on solutions to these types of problems for space communication environments, and we believe that these solutions may be applicable to the mobile and wireless community. As part of our work, we have defined and implemented the Space Communications Protocol Standards-Transport Protocol (SCPSTP) , a set of extensions to TCP that address the problems that we have identified. The results of our performance tests, both in the laboratory and on actual satellites, indicate that the SCPS-TP extensions yield significant improvements in throughput over unmodified TCP on error-prone links. Additionally, the SCPS modifications significantly improve performance over links with highly asymmetric data rates.