Applying cloud technology to 3D interactive multimedia applications is a promising way to provide flexible and cost-efficient online high-bandwidth immersive services to a large population of end users. One main reason cloud systems are popular among users is the fact that it relaxes the hardware requirements for high-end interactive visual applications. As most of the computational tasks are done on cloud servers, users no longer need to upgrade their hardware as frequently to keep up with the ever-increasing high-end computing requirements of the latest applications. Moreover, cloud systems make it easier for a user to enjoy applications on different platforms, including mobile devices that are usually not powerful enough to run high-end, memory-intensive services. In short, applying cloud technology to high-end immersive applications has advantages in cost-efficiency and flexibility both for the end users and the service providers. However, there are two main drawbacks to applying cloud technology to 3D interactive multimedia services: (1) high-bandwidth utilization and (2) latency. In this article, we propose a framework that addresses the two problems for single-player cloud gaming by using a combination of collaborative rendering, progressive meshes, and 3D image warping techniques. The experimental results show that the proposed system can reduce the bandwidth usage and improve the visual quality by utilizing local computing power on the client. The results also show that the interaction latency can be reduced somewhat by sacrificing some degree of visual quality in the end system.

Exergaming has drawn attention for years as a health benefactor that can contribute to public health improvement. Studies have shown that it can result in immediate health benefits. However, it has also been shown that current exergames do not have the staying power to bring players back for long-term use - the benefits tend to be short term with players initial excitement waning over time. The obvious conclusion is that current exergames are not designed with long-term challenges as their underlying game design criterion. To address the issue, we looked into adapting an existing popular game, with proven staying power, to an exercise bike. In a preliminary study that we conducted, we showed that children took very well to MineBike, an exergaming experience we developed using the popular game Minecraft. However, we also observed that to encourage the children to pursue a physical activity of consistent intensity we had to incorporate ongoing in-game challenges. In this paper, we specifically address the issue of embedding exergaming experiences in a popular game to enhance the staying power for sustained physical activity. We describe game design elements that can create the immersive exergaming experience that encourages mid- to high-level exercise intensity.

The benefits of regular physical activity are well known. In recent years, a lot of attention has been placed by the health industry on developing motivational techniques to get people to exercise - habit forming is key. In the iXercise project, we incorporate cloud computing, cloud networking, and physical computing solutions, in conjunction with innovative game design principles, to create the iXercise platform. iXercise (immersive exercise) is based on a virtual environment exergame platform that provides an adjusted intervention fitness program targeted to special needs populations. This is especially of import for the large population of special needs patients, e.g., leukemia, PTSD, obese/sedentary, and the elderly, that are unable to participate in fitness programs outside of special environments, such as medical clinics, special care residences or their homes. In this paper we describe our prototype which encourages adherence to a fitness program, by modding an existing popular video game that turns physical activity targets into in-game quests and rewards linked to individual fitness levels and performance.

The emergence of pico projectors as a part of future mobile devices presents unique opportunities for collaborative settings, especially in entertainment applications, such as video playback. By aggregating pico projectors from several users, it is possible to enhance resolution, brightness, or frame rate. In this paper we present a camera-based methodology for the alignment and synchronization of multiple projectors. The approach does not require any complicated ad hoc network setup among the mobile devices. A prototype system has been set up and used to test the proposed techniques.

Pico projectors are expected to become increasingly popular in the near future, in particular when embedded in mobile devices such as smart phones, portable media players and digital cameras. Consumers seeking more features and attracted to all in one portable devices, will find such integrated devices very appealing. However the resolution and the brightness provided by integrated mobile projectors are much lower than what standard projectors commonly offer. Our collaborative scheme based on our synchronization technique provides a means of increasing resolution and brightness for the video projected by such mobile devices, significantly enhancing the viewing experience for the user.

For a typical portable handheld device, the backlight accounts for a significant percentage of the total energy consumption (e.g., around 30% for a Compaq iPAQ 3650). Substantial energy savings can be achieved by dynamically adapting backlight intensity levels on such low- power portable devices. In this paper, we analyze the characteristics of video streaming services and propose an adaptive scheme called Quality Adapted Backlight Scaling (QABS), to achieve backlight energy savings for video playback applications on handheld devices. Specifically, we present a fast algorithm to optimize backlight dimming while keeping the degradation in image quality to a minimum so that the overall service quality is close to a specified threshold. Additionally, we propose two effective techniques to prevent frequent backlight switching, which negatively affects user perception of video. Our initial experimental results indicate that the energy used for backlight is significantly reduced, while the desired quality is satisfied. The proposed algorithms can be realized in real time. I. INTRODUCTION With the widespread availability of 3G cellular networks, mobile hand-held devices are increasingly being designed to support stream- ing video content. These devices have stringent power constraints because they use batteries with finite lifetime. On the other hand, multimedia services are known to be very resource intensive and tend to exhaust battery resources quickly. Therefore, conserving power to prolong battery life is an important research problem that needs to be addressed, specifically for video streaming applications on mobile handheld devices. Most hand-held devices are equipped with a TFT (Thin-Film Transistor) LCD (Liquid Crystal Display). For these devices, the display unit is driven by the illumination of backlight. The backlight consumes a considerable percentage of the total energy usage of the handheld device; it consumes 20%-40% of the total system power (for Compaq iPAQ) (?). Dynamically dimming the backlight is considered an effective method to save energy (?), (?), (?) with scaling up of the pixel luminance to compensate for the reduced fidelity. The luminance scaling, however, tends to saturate the bright part of the picture, thereby affecting the fidelity of the video quality. In (?), a dynamic backlight luminance scaling (DLS) scheme is proposed. Based on different scenarios, three compensation strategies are discussed, i.e., brightness compensation, image enhancement, and context processing. However, their calculation of the distortion does not consider the fact that the clipped pixel values do not contribute equally to the quality distortion. In (?), a similar method, named concurrent brightness and contrast scaling (CBCS), is proposed. CBCS aims at conserving power by reducing the backlight illumi- nation while retaining the image fidelity through preservation of the image contrast. Their distortion definition and proposed compensation technique may be good for static image based applications, such as the graphic user interface (GUI) and maps, but might not be suitable for streaming video scenarios, because their contrast compensation further compromises the fidelity of the images. In addition, neither (?) nor (?) solves the problem associated with frequent backlight switching which can be quite distracting to the end user. In this paper, we explicitly incorporate video quality into the backlight switching strategy and propose a quality adaptive back- light scaling (QABS) scheme. The backlight dimming affects the brightness of the video. Therefore, we only consider the luminance compensation such that the lost brightness can be restored. The lumi- nance compensation, however, inevitably results in quality distortion. For the video streaming application, the quality is normally defined as the resemblance between the original and processed video. Hence, for the sake of simplicity and without loss of generality, we define the quality distortion function as the mean square error (MSE)(see Equation (1)) and the quality function as the peak signal to noise ratio (PSNR)(see Equation (2)), both of which are well accepted objective video quality measurements.

Cross-Layer design has been the focus of several recent research efforts. Due to the highly variable nature of the links used in wireless communication systems and the resource-poor nature of the wireless mobile devices, there have been multiple research efforts to improve the performance of the protocol stack by allowing cross-layer interaction in wireless systems. Cross-layer interaction means allowing communication of a layer with any other possibly non-adjacent layer in the protocol stack. Several issues related to the cross-layer design paradigm need to be addressed before it can achieve its promises. One of these issues is to have a well defined framework that manages the interaction between the different layers of the protocol stack, such that the modularity of the stack is preserved while still achieving the flexibility and adaptability which cross-layer design promises. This paper addresses this issue by proposing a cross-layer coordination framework for next generation wireless systems. The proposed framework enables the interaction between non-adjacent layers in a systematic organized way while preserving the modularity of each layer. The proposed framework addresses some of the concerns stated in recent research about cross-layer design. We believe that the existence of such a framework will ease the development of cross-layer design schemes.

Supporting high-quality video transmission across wireless networks demands not only high but also constant bit rate (CBR). In order to provide a CBR channel, large modifications in transmitting power requirement would normally be necessary to inverse extremely high variation in channel fading. However, with a multiple input and multiple output (MIMO) system, sub-channel diversity can "smooth out" aggregate channel fluctuation, making it possible to support a CBR channel with efficient power control. We have devised an optimal antenna transmitting power and bits-per-symbol allocation algorithm that can reduce significantly, the total power consumption required to maintain CBR channels. We also utilize a secondary power control based on leaky bucket paradigm for smooth battery drain and to make peak power utilization to within system requirements. Simulation results reveal that under practical power fluctuation constraints, the scheme can achieve power savings of up to 10 dB with a 4/spl times/4 antenna system.

In this paper, we propose an encoder side error resilience technique, namely the progressive group of picture (PGOP), which is well suited for rate controlled real-time video transmission over wireless channels. The PGOP scheme can be seen as an alternative approach to the group of picture (GOP) that periodically uses intra-frames (I-frames) for low bit-rate channels. This is because: (1) the PGOP scheme is shown to completely stop error propagation in a timely manner; (2) the PGOP scheme avoids the bit rate fluctuation problem associated with I-frames by using a progressive intra-updating scheme. We use the PGOP scheme in a perceptual quality feedback framework, and apply a novel online algorithm that helps shape the bit rate and explores the maximum recovery capability of PGOP. The performance of our scheme is demonstrated in a wireless LAN environment.

The prevalence of the IEEE 802.11b technology leads to the emergence of Wi-Fi based AV conferencing applications. However, due to the "best-effort" transport service and other unpredictable factors such as user mobility, location and background traffic, the transport channel behavior often fluctuates drastically. It thus becomes rather difficult to configure an appropriate de-jitter buffer to maintain the temporal fidelity of the AV presentation. We propose in this paper an adaptive delay and synchronization control scheme for AV conferencing applications over campus-wide WLANs. Making use of a distributed timing mechanism, the scheme monitors the synchronization errors and estimates the delay jitters among adjacent MDUs in real-time. It piece-wisely controls the equalization delay to compensate for the delay jitters experienced by MDUs in a closed-loop manner. We investigate the performance of the proposed scheme through trace-driven simulations. Simulation results show that the scheme is capable of dynamically balancing between synchronization requirements and latency requirements in all scenarios. In particular, compared with solutions using a static setting, the proposed scheme is able to achieve a gain of around 100 ms in end-to-end delay with the same amount of MDU losses under some media-unfriendly situations.

This paper proposes an encoder side error resilience technique, namely the progressive group of picture (PGOP), which is well suited for rate controlled real-time video transmission over wireless channels. The PGOP scheme can be seen as an alternative approach, to the group of picture (GOP) that uses periodic frames (I-frames) for low bit-rate channels. This is because: (1) the PGOP scheme can completely stop error propagation in a timely manner; (2) the PGOP scheme avoids the bit rate unevenness problem of I-frames by a progressive intra-updating method and further applies the PGOP scheme in a novel video perceptual quality feedback framework, and design an on-line algorithm both to help shape the bit rate and to explore the maximum, recovery capability of PGOP. The performance of our scheme is demonstrated in a wireless LAN channel. It is to be noted that the PGOP am additionally provide the random access functionality and the technique is applicable to all inter frame based video standards, e.g.. MPEG-4.

The demands for multimedia and packet data services over wireless devices have increased over the past few years. The direct impact on performance makes scheduling for real-time traffic important. This paper presents a novel scheduling algorithm called fair channel-dependent scheduling which schedules packet delivery to mobile stations in a fair manner and at the same time takes into consideration the channel conditions for power efficiency. With added delay information for real-time traffic, this approach aims at delivering real-time traffic in a timely manner, while maintaining a balance between power conservation and fairness. Through comparative simulations with two conventional scheduling algorithms, we show that our scheme indeed achieves better overall performance than comparable scheduling schemes.

The increasing use of real-time applications over IP networks will require better use of available resources to meet the stringent requirements of real-time applications, especially over the wireless links. When performing data link/network layer scheduling, priorities of different real-time packets can be differentiated by exploiting the information available from a higher layer of the protocol stack (RTP/RTCP). Utilizing the fast power control feature available in CDMA-based wireless networks, a packet scheduling algorithm for real-time traffic over an IP-based cellular network was designed and evaluated. By utilizing the channel state and deadline information, much improved performance in transport of real-time traffic was achieved.

In this paper we address the important issue of providing QoS for VBR video communications in an efficient manner. We show that efficient transmission of VBR video with a high QoS is feasible when using a receiver-initiated resource renegotiation (RIR) scheme. The scheme for RIR is based on RTP and RSVP. RTP's media specific header is used to send video source information to receivers. Receivers utilize this information to estimate the traffic descriptors. Renegotiations are triggered based on the receiver's buffer status and RSVP is used to renegotiate flow parameters with the network. The performance of the proposed scheme is evaluated via simulations using several 20-minute-long MPEG-2 bit streams. Performance metrics considered are video quality and renegotiation overhead for different receiver buffer sizes and network delays. The results show that the proposed RIR scheme provides high video quality with an average renegotiation interval on the order of seconds, a 5-15 frames receiver buffer and network renegotiation delay below 300 msec. We also investigated call admission control (CAC) schemes for renegotiation VBR services. In particular, we studied the performance of the Central Limit Theorem based and the Chernoff bound based CAC algorithms, in terms of error in the calculation of the maximum number of admissible connections. Finally, to facilitate the scaling of the renegotiation scheme to large-scale networks, we studied how to reduce renegotiation overhead.

In this paper, we address the bandwidth reservation issue when MPEG-2 encoded VBR video streams are transmitted over wireless access networks using hybrid ARQ schemes to combat channel errors in VoD services. In addition to the original bandwidth which is necessary for the VBR traffic, a certain amount of excess bandwidth has to be reserved for retransmissions required by hybrid ARQ schemes. Different excess bandwidth reservation algorithms are proposed and investigated via simulations. The performance results are then presented and compared.

This paper presents a discussion of several middleware design issues related to the support of real-time multimedia communications over wireless personal communication services (PCS) networks. Specific interests are given to error recovery and synchronization mechanisms. A hybrid automatic repeat request (ARQ) scheme is employed for error control in the proposed system because it can efficiently adapt to nonstationary wireless channels and yield high throughput and reliability. In particular, delay and delay jitter control related to retransmissions in the error control module are addressed. An adaptive source rate control mechanism is used to handle the fluctuation of the effective channel data rate due to retransmissions. An adaptive synchronization scheme is developed to compensate for long-term delay variation caused by large-scale fading so that synchronization is preserved and end-to-end delay is kept low. Simulation results from the performance evaluation of the system are presented.

This paper proposes a hybrid ARQ error control scheme based on the concatenation of a Reed-Solomon (RS) code and a rate compatible punctured convolutional (RCPC) code for low-bit-rate video transmission over wireless channels. The concatenated hybrid ARQ scheme we propose combines the advantages of both type-I and type-II hybrid ARQ schemes. Certain error correction capability is provided in each (re)transmitted packet, and the information can be recovered from each transmission or retransmission alone if the errors are within the error correction capability (similar to type-I hybrid ARQ). The retransmitted packet contains redundancy bits which, when combined with the previous transmission, result in a more powerful RS/convolutional concatenated code to recover information if error correction fails for the individual transmissions (similar to type-II hybrid ARQ). Bit-error rate (BER) or signal-to-noise ratio (SNR) of a radio channel changes over time due to mobile movement and fading. The channel quality at any instant depends on the previous channel conditions. For the accurate analysis of the performance of the hybrid ARQ scheme, we use a multistate Markov chain (MSMC) to model the radio channel at the data packet level. We propose a method to partition the range of the received SNR into a set of states for constructing the model so that the difference between the error rate of the real radio channel and that of the MSMC model is minimized. Based on the model, we analyze the performance of the concatenated hybrid ARQ scheme. The results give valuable insight into the effects of the error protection capability in each packet, the mobile speed, and the number of retransmissions. Finally, the transmission of H.263 coded video over a wireless channel with error protection provided by the concatenated hybrid ARQ scheme is studied by means of simulations.

Uncontrolled variable-bit-rate (VBR) coded video yields consistent picture quality, but the traffic stream is very bursty. When sent over ATM networks, cell losses may be incurred due to limited buffer capacity at the switches; this could cause severe picture quality degradation. Source rate control can be implemented to generate a controlled VBR bit stream which conforms to specified bit rate bounds and buffer constraints. However, source rate control could result in picture quality degradation too. Hence, for real-time video services, an important issue to address is whether the picture quality degradation incurred by source rate control is within acceptable levels or how to choose the appropriate coding parameters to make it so. We establish quantitatively the relationship between picture quality and source rate control for the case of guaranteed service with different combinations of allocated bandwidth, buffer size, and other key video-coding parameters of MPEG-2. In addition, quality control in the context of two-layered scalable video service (basic and enhanced quality) is also considered. Our study reveals that, in order to maximize both the basic and the enhanced quality, source rate control should be implemented on both layers. The relationships between the two types of quality and different combinations of allocated bandwidths, buffer sizes, and some key coding parameters are also established quantitatively for MPEG-2 SNR scalability.

In a wireless packet (e.g., ATM) network that supports an integrated mix of multimedia traffic, a large variety of mobiles with different service rates will need to be accommodated efficiently in a bandwidth-on-demand fair-sharing manner. We propose and study multi-code CDMA (MC-CDMA) with distributed-queueing request update multiple access (DQRUMA) for multi-rate wireless packet networks. The network incorporates MC-CDMA (a flexible multiplexing scheme for providing multi-rate packet transmissions) and DQRUMA (an efficient demand-assignment multiple access protocol for wireless access and scheduling) to form a unified bandwidth-on-demand fair-sharing platform for multi-rate wireless services. A slot-by-slot packet transmit scheduling policy provides the possibility of slot-by-slot allocation of pseudo-noise (PN) codes (primary codes) and optimal power levels to the mobiles involved in packet transmissions. Also, a maximum capacity power allocation (MCPA) criterion is derived to exploit the subcode concatenation property of the MC-CDMA transmission. Simulation results show that the system provides close to ideal-access performance for multi-rate mobiles, both with homogeneous traffic characteristics and with a mix of heterogeneous traffic characteristics.

Multi-code direct-sequence code division multiple access (MC-CDMA) has been proposed as a flexible multiple access scheme for wireless packet networks that support a large variety of mobiles with different and even time-varying rates. However, the interference model typically used with conventional DS-CDMA cellular systems - a large number of low-power mobiles uniformly distributed in cells - may not hold. In MC-CDMA cellular systems, it is possible for a high-rate mobile, perhaps near the boundary of a cell, to have all the codes for high-rate transmission - therefore transmitting at high power and causing significant interference to the neighboring cell. In this study, we analyze the effects of MC-CDMA intercell interference on the reverse link (i.e., mobile to cell site) and investigate a maximum capacity power allocation (MCPA) criterion that exploits the sub-code concatenation property of MC-CDMA transmission. When the transmission rate is high, power control with MCPA criterion can significantly reduce reverse-link MC-CDMA intercell interference. However, with only four codes per mobile (e.g., as in IS-95), the reduction in intercell interference is less than 1 dB.

For high-end real-time video services, a layered source coding (LSC) scheme should generate a high priority (HP) layer which meets the following two criteria: 1) the HP data can be guaranteed to be delivered in a simple and efficient manner by a network usage parameter control (UPC) function such as the leaky bucket (LB) control; and 2) the HP layer should carry enough high visual quality. This paper investigates different data partitioning (DP) schemes for this purpose. Based upon the characteristics of the human visual system and the MPEG traffic characteristics, an adaptive DP scheme is implemented that achieves an acceptable compromise between high visual quality and low bit rate requirement during periods of long bursts.

This paper analyzes the transmission of voice and data over an 802.11 wireless local area network (WLAN). The data is transmitted in a contention based access period, while the voice samples are transmitted during a contention free period, based on a polling scheme. Because statistical multiplexing can be utilized, speech may be outdated when a poll arrives. The portion of outdated speech is then clipped to decrease the load on the channel. We analyze the quality of the voice conversations in terms of the percentage of bits clipped as well as the throughput of the data for various parameters. We show the boundary conditions involved in the transmission of voice over the WLAN and demonstrate the impact of a time-bounded service on the throughput during the contention period. The results show that the high overhead introduced by the 802.11 WLAN standard results in a low number of possible voice conversations. It can also be concluded that the cooperation of the contention based and contention free periods results in a poor performance. Further, variation of the maximum payload size reveals that the largest possible maximum payload size must be selected to minimize the percentage of clipped bits and maximize the throughput. Finally, we show that a larger superframe length provides the opportunity for more voice conversations or a higher data throughput, but requires increasing the time to live for the speech bits to retain an acceptable quality.

Successful exploitation of prioritized ATM networks requires matching the characteristics of the coded source with the unique properties of the network. This is particularly true for video which is very bandwidth intensive. The authors describe and investigate the performance of three prioritization schemes in terms of the traffic characteristics, SNR, and visual quality. These schemes are further designed such that the high priority data stream is MPEG-I compliant and only a single pass through the encoder is required. A technique for choosing parameter values which achieve a specified high priority traffic bandwidth is presented for two of these schemes. The technique is demonstrated to be remarkably accurate for different video sequences.>

The paper addresses the problem of designing a "virtual error free channel" for communication over unreliable wireless packet channels using hybrid error control schemes. For hybrid error control to be effective it is essential that an appropriate amount of forward error correction and automatic repeat requests be used. Optimal design methodologies for performance metrics of interest are developed. We also investigate the trade-offs involved when using interleaving to reduce the effect of burst errors. Asymptotic results are developed that can be used to develop simple design rules. In addition, several numerical examples are presented.

Presents a discrete Markov chain based analytic framework for the study of DS-CDMA with slotted ALOHA (DS-CDMA-S-ALOHA) random access protocols. The key feature of the work is that it distinguishes between the two stages in the transmission process, namely the access stage and the reception stage, which characterize the random access and the random errors associated with DS-CDMA-S-ALOHA respectively. Two DS-CDMA-S-ALOHA protocols are presented and analyzed. The performance of the protocols and the effects of the design parameters are evaluated numerically and compared with a bandwidth equivalent multi-channel slotted ALOHA system. The results show that with proper design, the DS-CDMA-S-ALOHA protocols can double the throughput with respect to that of a bandwidth equivalent multi-channel slotted ALOHA system.

In this paper we describe briefly a dynamic multi-path algorithm that has been considered for connection oriented asynchronous transfer mode (ATM) networks. Our scheme takes advantage of a cell multiplexing capability that has particular advantage in networks supporting variable bit rate (VBR) traffic. The fundamental objective of the scheme is to propose a congestion control based scheme that bridges the gap between routing and congestion control as the network becomes congested. The proposed routing scheme works as a shortest path first algorithm under light traffic conditions. However, as the shortest path becomes congested under unbalanced heavy traffic, the source uses multiple paths when and if available to distribute the calls and reduce cell loss. This mechanism will provide good Quality of Service for clients within the given constraints. We compare the performance of the proposed scheme with other competitive schemes. The throughput and cell loss performance are compared via simulations. These have been carried out concentrating on a five node network, each with varying traffic patterns, with the intention of gaining insight into the strengths and weaknesses of the various schemes.

Signal-to-interference ratio (SIR)-based call admission control (CAC) algorithms are proposed and studied in a DS-CDMA cellular system. Residual capacity is introduced as the additional number of initial calls a base station can accept such that system-wide outage probability will be guaranteed to remain below a certain level. The residual capacity at each cell is updated dynamically according to the reverse-link SIR measurements at the base station. A 2/sup k/ factorial experimental design and analysis via computer simulations is used to study the impact of the parameters used in the algorithms. The influence of these parameters on system performance, namely blocking probability and outage probability, is then examined via simulation. The performance of the algorithms is compared together with that of a fixed call admission control scheme (fixed CAC) under both homogeneous and hot spot traffic loading. The results show that SIR-based CAC always outperforms fixed CAC even under overload situations, which is not the case in FDMA/TDMA cellular systems. The primary benefit of SIR-based CAC in DS-CDMA cellular systems, however, lies in improving the system performance under hot spot traffic.>

An asynchronous transfer mode (ATM) switch mode at which cell arrive from a diverse set of source types is considered. To improve the utilization of network resources and facilitate management and control, source types are organized into traffic classes. Cells belonging to different traffic classes are transported by separate virtual paths though the network. Approximations for the quality of service that should be maintained for each traffic class are developed. Approximations are then developed for estimating the bandwidth and buffer requirements of each traffic class and also for the switch node. The problem of assigning weights, given a set of traffic classes, is addressed. The set of traffic classes that require the least amount of resources is determined. The closely related problem of minimizing the number of traffic classes given a finite bandwidth is also discussed. The structure of the problem at hand is then exploited to develop an efficient heuristic.>

The performance of variable bandwidth allocation schemes that estimate the required bandwidth for a variable-bit-rate (VBR) MPEG video source using a simple prediction algorithm is studied. The advantages of using a layered MPEG coder over the standard MPEG coder are discussed. A method for controlling a layered MPEG coder in order to reduce the effect of cell losses when compared to a standard MPEG coder is demonstrated.>

In this paper, we describe MPEGTool, an X window-based tool that can be used to generate a motion picture expert group (MPEG) encoded bit stream for video sequences and to study the statistical properties of the encoded data. The tool was designed to study the characteristics of variable bit rate video sources for transmission over asynchronous transfer mode (ATM) based broadband integrated services digital network (B-ISDN). The tool, which has a window-based graphic user interface, allows a user to specify several of the MPEG parameters such as the intraframe-to-interframe ratio and the quantizer scale. The tool also includes a statistical package that allows the user to plot graphs of various statistics including bit distributions, ATM cell distributions, time se ries, autocorrelation functions and cell interarrival times.

The complex traffic characteristics of variable-bit-rate (VBR) video sources makes them difficult to accommodate in asynchronous-transfer-mode (ATM) networks. To efficiently transport these services will require both an understanding of the source traffic and novel network control schemes. The performance of variable-bandwidth-allocation schemes that estimate the required bandwidth for a VBR Motion Picture Experts Group (MPEG) video source is studied using a simple prediction algorithm. The advantages of using a layered MPEG coder over the standard MPEG coder are discussed. A method for controlling a layered MPEG coder to reduce the effect of cell losses when compared to a standard MPEG coder is demonstrated.>

The performance of the Fiber Distributed Data Interface (FDDI) under overload is studied. It is well known that, under overload, FDDI can settle into one of many recurrent sets of states. The idea of desirable and undesirable states is introduced. A desirable state is one where users share the capacity more or less equally in each token rotation. It is shown that the response times of interactive users are adversely affected if the system operates in an undesirable set of states. The waiting time of heavy users can be reduced by forcing the system to settle into the desirable states. Simple control mechanisms are proposed to ensure that the system settle into the desirable states.>

The MPEG video coding standard for the transmission of variable-bit-rate video on asynchronous transfer mode (ATM)-based broadband ISDN is examined. The focus is on its use for real-time transmission of broadcast-quality video. The impact of two key parameters, the intraframe to interframe picture ratio and the quantization index that are defined in the standard, on the bit rates per frame was studied. These parameters can be used to control video sources depending on the state of the network. Also, as opposed to previous work which looks only at bit rates per frame, the bits generated per macroblock are studied. This is the basic MPEG coding unit. By packetizing these bits, insight was obtained into the cell arrival process to a network for a MPEG video source.>

In this paper we describe briefly a dynamic multi-path routing scheme that has been considered for connection oriented homogeneous high speed networks. The fundamental objective of the scheme is to bridge the gap between routing and congestion control as the network becomes congested. Because propagation delay far out shadows queueing and transmission delay in high speed networks, the proposed routing scheme works as a shortest path (minimum hop) first algorithm under light traffic conditions. However as the shortest path becomes congested, the source node uses multiple paths when and if available in order to distribute the load and reduce packet loss. The scheme is a cross between Alternate Path routing and Trunk Reservation.We compare the performance of the proposed scheme with the Shortest Path Only algorithm, the Alternate Path routing algorithm, the Random Routing algorithm, and the Trunk Reservation scheme. The throughput and packet loss performance are compared via simulations. These have been carried out concentrating on a 5 node network with varying traffic patterns, the intention being to gain insight into the strengths and weaknesses of the various schemes.

An analytical approach is provided for determining the performance of a virtual circuit connection for data transmission in high speed asynchronous transfer mode (ATM) network buffers at wide area network (WAN) nodes. The analysis assumes that the network operates using the best effort delivery strategy and that the end-to-end virtual circuit is responsible for guaranteeing the integrity of the connection. As the normal Markovian assumptions do not apply, a concise exact solution is impossible to obtain. A hybrid model incorporating finite buffers at the nodes was developed to study the effect on the performance of both link errors and buffer overflow in conjunction with an end-to-end packet loss recovery scheme.>

An investigation is conducted of a metropolitan area network that is an extension to the Manhattan street network (MSN) and features a three-connected, regular mesh topology. The design of a triangularly arranged connection (TAC) node is discussed, and the basic layout of the network is examined. The underlying routing approach of the network is explored and explained through a routing example. It is established through simulation that the increased interconnectivity of the TAC gives a significant boost in performance over the MSN at a cost of more physical links and node complexity.>

The routing and flow control techniques developed for wide-area, local-area, and metropolitan-area networks are surveyed. A classification that shows the characteristics that are desirable for high-speed wide-area networks is developed. On the basis of the classification, techniques that should and should not be considered for future high-speed networks are identified. In particular, an attempt is made to determine the effect of increasing transmission rates on the routing and flow control algorithms that are used in packet-switched networks.>

The authors discuss what they consider the fundamental issue of bandwidth allocation on an integrated local area network. An approach is introduced for dynamic bandwidth allocation which is based on traffic prediction concepts. It is especially well suited for real-time services such as video and voice. Using a control model two allocation schemes are proposed: the first is based on an analytical model of the traffic flow; the second is a simpler version that can be easily implemented on very high-speed systems. The results of simulation studies indicate a marked improvement in performance. The presented approach is especially effective when used in systems with large transmission path latencies as the network performance does not deteriorate with increasing latency. This is very useful if the network is to be used as a metropolitan area network.>

Fundamental issues arising in the design and implementation of architectures and protocols for integrated local area networks (ILAN's) supporting video, voice, graphics, data, and facsimile, are addressed, and open questions are identified. To characterize the performance of ILAN's, a set of performance evaluation guidelines is established. A class of conceptual solutions for ILAN's is proposed, and its expected performance is obtained by means of simulation. The class includes priority assignment schemes, parametrized sequential and simultaneous scheduling algorithms, and centralized and decentralized access control mechanisms. Our analysis, based upon simulations, shows that the bandwidth freed up by the video and voice calls during periods of little movement in the images and silence periods in the speech signals can be efficiently utilized for graphics and data transmission. The optimal priority mechanism among the different user classes depends on the traffic mix. Finally, it is argued that additional experimentation providing empirical data is imperative for understanding the real-time behavior of ILAN's. Consequently, a network testbed (NTB) called MAGNET is proposed for evaluating the performance of the proposed theoretical systems. It consists of an adaptive ILAN whose mediaaccess protocol is controlled by an Expert System.

MAGNET is a Network Testbed (NTB) designed to evaluate the performance of local area network architectures and protocols supporting integrated services such as televideo, video conferencing, telephone, computer communication, and facsimile. The integrated local area network (ILAN) presented in this paper has been designed to support the research activities intended for MAGNET. An active slotted ring topology was chosen to implement adaptive sequential-type access mechanisms that are a function of the network loading. The ILAN is based on fiber optic technology with wavelength-division multiplexing of two 100 Mbit/s channels. The media-access protocol provides higher levels with the capability of implementing two modes of operation referred to as circuit-switching, and packet-switching modes, respectively. Provisions have also been made for implementing simultaneous scheduling as well as centralized and decentralized access control schemes. Finally, the reconfigurable structure of the network testbed enables the study of the reliability and availability of ILAN's.