Robert E. Kinicki

WBest: A bandwidth estimation tool for IEEE 802.11 wireless networks

Bandwidth estimation techniques seek to provide an accurate estimation of available bandwidth such that network applications can adjust their behavior accordingly. However, most current techniques were designed for wired networks and produce relatively inaccurate results and long convergence times on wireless networks where capacity can vary dramatically. This paper presents a new wireless bandwidth estimation tool, WBest, designed for fast, non-intrusive, accurate estimation of available bandwidth in IEEE 802.11 networks. WBest is a two-stage algorithm: 1) a packet pair technique estimates the effective capacity over a flow path where the last hop is a wireless LAN (WLAN); and 2) a packet train technique estimates achievable throughput to infer the available bandwidth. WBest parameters are optimized given the tradeoffs of accuracy, intrusiveness and convergence time. The advantage of WBest stems from avoiding a search algorithm to detect the available bandwidth by statistically detecting the available fraction of the effective capacity to mitigate estimation delay and the impact of random wireless channel errors. WBest is implemented and evaluated on an 802.11 wireless testbed. Comparisons with other available bandwidth estimation tools shows WBest to have higher accuracy, lower intrusiveness and faster convergence times. Thus, WBest demonstrates the potential for improving the performance of applications that need bandwidth estimation, such as multimedia streaming, on wireless networks.

A computerized method for the rapid display of ventricular activation during the intraoperative study of arrhythmias.

Arrhythmias which last for only a few cycles or exhibit activation sequences that change from cycle to cycle may occur during the intraoperative study of ventricular tachyarrhythmias. To map these transient, varying arrhythmias requires recording from numerous sites simultaneously, which cannot be done with a single, hand-held electrode probe. Therefore, a computerized method was developed to record potentials simultaneously from up to 27 electrodes and to display epicardial and transmural activation sequences rapidly.Three arrays of electrodes are used. The electrodes of the first array are distributed over the epicardium of both ventricles and fixed within a flexible nylon mesh sock. The electrodes of the second array form a 3-cmsquare grid and are used to obtain a more detailed activation map of a small region of epicardium. The electrodes of the third array are spaced along the shafts of needles that are inserted through the myocardium to record transmural activation. The potentials recorded from an array of electrodes and the computer-selected activation times at the electrodes are displayed for visual inspection after which a map of activation is drawn. The activation sequence of a single cycle can be displayed 10 minutes after recording the potentials. The activation map obtained from the sock array determines the placement of the grid array; the map resulting from the grid array determines the placement of the needle array. Thus, the region responsible for initiating an arrhythmia can be determined if the arrhythmia can be induced three or more times.

Characteristics of streaming media stored on the Web

Despite the growth in multimedia, there have been few studies that focus on characterizing streaming audio and video stored on the Web. This investigation used a customized Web crawler to traverse 17 million Web pages from diverse geographic locations and identify nearly 30,000 streaming audio and video clips available for analysis. Using custom-built extraction tools, these streaming media objects were analyzed to determine attributes such as media type, encoding format, playout duration, bitrate, resolution, and codec. The streaming media content encountered is dominated by proprietary audio and video formats with the top four commercial products being RealPlayer, Windows Media Player, MP3 and QuickTime. The distribution of the stored playout durations of streaming audio and video clips are long-tailed. More than half of the streaming media clips encountered are video, encoded primarily for broadband connections and at resolutions considerably smaller than the resolutions of typical monitors.

MediaPlayer™ versus RealPlayer™: a comparison of network turbulence

The performance of currently available streaming media products will play an important role in the network impact of streaming media. However, there are few empirical studies that analyze the network traffic characteristics and lnternet impact of current streaming media products. This paper presents analysis from an empirical study of the two dominant streaming multimedia products, RealNetworks RealPlayer#8482; and Microsoft MediaPlayer#8482;. Utilizing two custom media player measurement tools, RealTracker and MediaTracker, we are able to gather application layer and network layer information about RcalPlayer and MediaPlayer for the same media under the same network conditions. Our analysis shows that RealPlayer and MediaPlayer have distinctly different behavior characteristics and exposes some of the impact of streaming media on the network and provides valuable information for building more realistic streaming media simulations.

Playout buffer and rate optimization for streaming over IEEE 802.11 wireless networks

Most streaming rate selection and buffer optimization algorithms are developed for wired networks and can perform poorly over wireless networks. Wireless MAC layer behavior, such as rate adaptation, retransmissions, and medium sharing, can significantly degrade the effectiveness of current streaming algorithms. This article presents the Buffer and Rate Optimization for Streaming (BROS) algorithm to improve streaming performance. BROS uses a bandwidth estimation tool designed specifically for wireless networks and models the relationship between buffer size, streaming data rate, and available bandwidth distribution. BROS optimizes the streaming data rate and initial buffer size, resulting in a high data rate but with few frame losses and buffer underflow events, while still keeping a small initial buffer delay. BROS is implemented in the Emulated Streaming (EmuS) client-server system and evaluated on an IEEE 802.11 wireless testbed with various wireless conditions. The evaluation shows that BROS can effectively optimize the streaming rate and initial buffer size based on wireless network bandwidth conditions, thus achieving better performance than static rate or buffer selection and jitter removal buffers.

A computer system for the intraoperative mapping of ventricular arrhythmias.

Abstract A computer system has been developed to measure the electrical activation sequence of the heart during ventricular tachyarrhythmias. The system incorporates 32 channels of parallel analog-to-digital conversion, rapid generation of results, a high degree of user interaction, and ability to capture events occurring at essentially random times. The system has been used extensively in the animal laboratory as a prototype for a system to be used in the surgical treatment of ventricular tachycardia. It has been shown to be an effective and practical tool in the location of the site of early activation during cardiac arrhythmias.

Adjusting forward error correction with quality scaling for streaming MPEG

Packet loss can severely impact streaming video quality. Repair techniques protect streaming video from packet loss but at the price of a reduced effective transmission rate when streaming a flow in a capacity constrained situation. This paper proposes an algorithm that optimizes the choice of Forward Error Correction (FEC) to repair packet loss for streaming MPEG videos under a capacity constraint with quality scaling. An analytic model is developed to estimate the video quality of streaming MPEG given a quality scaling level and a specific FEC strength. Given network conditions in terms of packet loss rate, the model searches the total variable space to find the combination of FEC and scaling that yields the optimal quality under the capacity constraint. Analysis over a range of network conditions indicates that adjusting FEC with quality scaling provides significant performance improvement.

Measurements of the congestion responsiveness of windows streaming media

While previous research has shown that streaming media can respond to network congestion, it is not known to what extent commercial products are responsive. This research characterizes the bitrate response of Windows Streaming Media (WSM) to changes in network capacity and loss rate. A streaming media test bed was built to systematically vary network and content encoding characteristics and measure WSM congestion responsiveness. The results demonstrate that WSM improve response to congestion when content is encoded into several bitrates by the content provider.

Inferring Queue Sizes in Access Networks by Active Measurement

Router queues can impact both round-trip times and throughput. Yet little is publicly known about queue provisioning employed by Internet services providers for the routers that control the access links to home computers. This paper proposes QFind, a black-box measurement technique, as a simple method to approximate the size of the access queue at last mile router. We evaluate QFind through simulation, emulation, and measurement. Although precise access queue results are limited by receiver window sizes and other system events, we find there are distinct difference between DSL and cable access queue sizes.

Adjusting forward error correction with temporal scaling for TCP-friendly streaming MPEG

New TCP-friendly constraints require multimedia flows to reduce their data rates under packet loss to that of a conformant TCP flow. To reduce data rates while preserving real-time playout, temporal scaling can be used to discard the encoded multimedia frames that have the least impact on perceived video quality. To limit the impact of lost packets, Forward Error Correction (FEC) can be used to repair frames damaged by packet loss. However, adding FEC requires further reduction of multimedia data, making the decision of how much FEC to use of critical importance. Current approaches use either inflexible FEC patterns or adapt to packet loss on the network without regard to TCP-friendly data rate constraints. In this article, we analytically model the playable frame rate of a TCP-friendly MPEG stream with FEC and temporal scaling, capturing the impact of distributing FEC within MPEG frame types with interframe dependencies. For a given network condition and MPEG video encoding, we use our model to exhaustively search for the optimal combination of FEC and temporal scaling that yields the highest playable frame rate within TCP-friendly constraints. Analytic experiments over a range of network and application conditions indicate that adjustable FEC with temporal scaling can provide a significant performance improvement over current approaches. Extensive simulation experiments based on Internet traces show that our model can be effective as part of a streaming protocol that chooses FEC and temporal scaling patterns that meet dynamically-changing application and network conditions.