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271
Modeling TCP Reno Performance: A Simple Model and Its Empirical Validation
- IEEE/ACM Transactions on Networking
, 2000
"... Abstract—The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization ..."
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Cited by 371 (4 self)
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Abstract—The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization of the steady-state send rate as a function of loss rate and round trip time (RTT) for a bulk transfer TCP flow. Unlike the models in [7]–[9], and [12], our model captures not only the behavior of the fast retransmit mechanism but also the effect of the time-out mechanism. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP send rate and is accurate over a wider range of loss rates. We also present a simple extension of our model to compute the throughput of a bulk transfer TCP flow, which is defined as the amount of data received by the receiver in unit time. Index Terms—Empirical validation, modeling, retransmission timeouts, TCP.
On the constancy of Internet path properties
- In Proceedings of ACM SIGCOMM Internet Measurement Workshop
, 2001
"... Abstract — Many Internet protocols and operational procedures use measurements to guide future actions. This is an effective strategy if the quantities being measured exhibit a degree of constancy: that is, in some fundamental sense, they are not changing. In this paper we explore three different no ..."
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Cited by 294 (15 self)
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Abstract — Many Internet protocols and operational procedures use measurements to guide future actions. This is an effective strategy if the quantities being measured exhibit a degree of constancy: that is, in some fundamental sense, they are not changing. In this paper we explore three different notions of constancy: mathematical, operational, and predictive. Using a large measurement dataset gathered from the NIMI infrastructure, we then apply these notions to three Internet path properties: loss, delay, and throughput. Our aim is to provide guidance as to when assumptions of various forms of constancy are sound, versus when they might prove misleading. I.
Modeling TCP latency
- in IEEE INFOCOM
, 2000
"... 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 toda ..."
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Cited by 235 (8 self)
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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.
PROMISE: Peer-to-Peer Media Streaming Using CollectCast
, 2003
"... We present the design, implementation, and evaluation of PROMISE, a novel peer-to-peer media streaming system encompassing the key functions of peer lookup, peer-based aggregated streaming, and dynamic adaptations to network and peer conditions. Particularly, PROMISE is based on a new application l ..."
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Cited by 172 (12 self)
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We present the design, implementation, and evaluation of PROMISE, a novel peer-to-peer media streaming system encompassing the key functions of peer lookup, peer-based aggregated streaming, and dynamic adaptations to network and peer conditions. Particularly, PROMISE is based on a new application level P2P service called CollectCast. CollectCast performs three main functions: (1) inferring and leveraging the underlying network topology and performance information for the selection of senders; (2) monitoring the status of peers and connections and reacting to peer/connection failure or degradation with low overhead; (3) dynamically switching active senders and standby senders, so that the collective network performance out of the active senders remains satisfactory. Based on both real-world measurement and simulation, we evaluate the performance of PROMISE, and discuss lessons learned from our experience with respect to the practicality and further optimization of PROMISE.
Detecting shared congestion of flows via end-to-end measurement
- IEEE/ACM Transactions on Networking
, 2000
"... 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-sen ..."
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Cited by 165 (6 self)
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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.
On Multiple Description Streaming with Content Delivery Networks
- In Proceedings of IEEE INFOCOM
, 2002
"... CDNs have been widely used to provide low latency, scalability, fault tolerance, and load balancing for the delivery of web content and more recently streaming media. We propose a system that improves the performance of streaming media CDNs by exploiting the path diversity provided by existing C ..."
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Cited by 151 (4 self)
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CDNs have been widely used to provide low latency, scalability, fault tolerance, and load balancing for the delivery of web content and more recently streaming media. We propose a system that improves the performance of streaming media CDNs by exploiting the path diversity provided by existing CDN infrastructure. Path diversity is provided by the different network paths that exist between a client and its nearby edge servers; and multiple description (MD) coding is coupled with this path diversity to provide resilience to losses. In our system, MD coding is used to code a media stream into multiple complementary descriptions, which are distributed across the edge servers in the CDN. When a client requests a media stream, it is directed to multiple nearby servers which host complementary descriptions. These servers simultaneously stream these complementary descriptions to the client over different network paths.
Modeling of Packet Loss and Delay and their Effect on Real-Time Multimedia Service Quality
- PROCEEDINGS OF NOSSDAV '2000
, 2000
"... Internet packet loss and delay exhibits temporal dependency. If packet n is lost, packet n + 1 is also likely to be lost. It leads to bursty network losses and late losses in real-time multimedia services such as Voice over IP (VoIP). This may degrade perceptual quality and the effectiveness of Forw ..."
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Cited by 117 (3 self)
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Internet packet loss and delay exhibits temporal dependency. If packet n is lost, packet n + 1 is also likely to be lost. It leads to bursty network losses and late losses in real-time multimedia services such as Voice over IP (VoIP). This may degrade perceptual quality and the effectiveness of Forward Error Correction (FEC). To characterize this burstiness, we first discuss the modeling of packet loss and delay. We propose the joint use of the extended Gilbert model and the inter-loss distance (ILD) metric to characterize temporal loss dependency. For delay, we introduce a metric called the conditional cumulative distribution function. We have applied these models to some Internet packet traces to validate the necessity and effectiveness of these models. We then evaluate the effect of these dependencies on VoIP by investigating the final loss pattern (FLP) after applying playout delay adjustment and FEC. Our results through a set of simulations confirmed that the FLP is still bursty.
Multi-path selection for multiple description encoded video streaming
- in IEEE Int. Conf. Communications (ICC
, 2003
"... Abstract — This paper presents a new framework for multimedia streaming that integrates the application and network layer functionalities to meet such stringent application requirements as delay and loss. The coordination between these two layers provides more robust media transmission even under se ..."
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Cited by 86 (3 self)
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Abstract — This paper presents a new framework for multimedia streaming that integrates the application and network layer functionalities to meet such stringent application requirements as delay and loss. The coordination between these two layers provides more robust media transmission even under severe network conditions. In this framework, a multiple description source coder is used to produce multiple independently-decodable streams that are routed over partially link-disjoint (non-shared) paths to combat bursty packet losses. We model multi-path streaming and propose a multi-path selection method that chooses a set of paths maximizing the overall quality at the client. Overlay infrastructure is then used to achieve multi-path routing over these selected paths. The simulation results show that the average peak signal-to-noise ratio (PSNR) improves by up to 8.1 dB, if the same source video is routed over intelligently selected multiple paths instead of the shortest path or maximally link-disjoint paths. In addition to PSNR improvement in quality, the enduser experiences a more continual streaming quality. I.
Path Diversity with Forward Error Correction (PDF) System for Packet Switched Networks
- in Proceedings of IEEE INFOCOM
, 2003
"... Packet loss and end-to-end delay limit delay sensitive applications over the best effort packet switched networks such as the Internet. In our previous work, we have shown that substantial reduction in packet loss can be achieved by sending packets at appropriate sending rates to a receiver from mul ..."
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Cited by 84 (0 self)
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Packet loss and end-to-end delay limit delay sensitive applications over the best effort packet switched networks such as the Internet. In our previous work, we have shown that substantial reduction in packet loss can be achieved by sending packets at appropriate sending rates to a receiver from multiple senders, using disjoint paths, and by protecting packets with forward error correction. In this paper, we propose a Path Diversity with Forward error correction (PDF) system for delay sensitive applications over the Internet in which, disjoint paths from a sender to a receiver are created using a collection of relay nodes. We propose a scalable, heuristic scheme for selecting a redundant path between a sender and a receiver, and show that substantial reduction in packet loss can be achieved by dividing packets between the default path and the redundant path. NS simulations are used to verify the effectiveness of PDF system.
The Stationarity of Internet Path Properties: Routing, Loss, and Throughput
- In ACIRI Technical Report
, 2000
"... There is much interest in using network measurements for both modeling and operational purposes. In this paper we focus on the fundamental question of the stationarity of such measurements. That is, to what extent are past measurements a good predictor of the future? We used the NIMI infrastructure ..."
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Cited by 83 (0 self)
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There is much interest in using network measurements for both modeling and operational purposes. In this paper we focus on the fundamental question of the stationarity of such measurements. That is, to what extent are past measurements a good predictor of the future? We used the NIMI infrastructure and a set of public traceroute servers to capture large measurement datasets of three quantities: routing, packet loss, and TCP throughput. We apply statistical tests to attempt to develop sound characterizations of the stationarity of these data sets, and discuss several types of nonstationarity. 1 Introduction In recent years there has been a surge of interest in network measurements. These measurements have deepened our understanding of network behavior and led to more accurate and qualitatively different models of network traffic. Network measurements are also used operationally by various protocols to guide network usage. For instance, RLM [MJV96] and equation-based congestion control...
