Practitioners of a growing number of scientific and artificial-intelligence (AI) applications use High-Performance Wide-Area Networks (HP-WANs) for moving massive data sets between remote facilities. Accurate prediction of the flow completion time (FCT) is essential in these data-transfer workflows because compute and storage resources are tightly scheduled and expensive. We assess the viability of three TCP congestion control algorithms (CUBIC, BBRv1, and BBRv3) for massive data transfers over public HP-WANs, where limited control of critical data-path parameters precludes the use of Remote Direct Memory Access (RDMA) over Converged Ethernet (RoCEv2), which is known to outperform TCP in private HP-WANs. Extensive experiments on the FABRIC testbed indicate that the configuration control limitations can also hinder TCP, especially through microburst-induced packet losses. Under these challenging conditions, we show that the highest FCT predictability is achieved by combination of BBRv1 with the application of traffic shaping before the HP-WAN entry points.

Machine learning is increasingly used across wireless systems, with research papers reporting highly promising results, yet most proposed solutions are never deployed in practice. One major reason is the generalization gap between research evaluation and real world wireless settings. In this paper, we reproduce studies that apply machine learning to wireless systems across three application areas: throughput prediction, channel estimation, and sensing from wireless signals. We examine how data collection, processing, and splitting decisions affect reported performance. Across these case studies, we find that common evaluation practices, such as random assignment of samples to training and test sets, can introduce relationships between training and test sets that would not exist in the proposed deployment setting. These arise from temporal, environmental, or subject correlations in wireless measurements and can lead to overly optimistic estimates of model performance. When we redesign the evaluation to better reflect intended deployment settings, the estimated model performance is much worse, revealing substantial generalization gaps. We conclude with practical recommendations for designing and reporting machine learning evaluations for wireless systems, emphasizing data partitioning strategies that reflect the intended deployment setting.

A potential use case for the sixth-generation wireless networks is the widespread adoption of eXtended Reality (XR) on mobile devices. These applications require high data rates, which may exhaust the limited available bandwidth ofmid-band(MB) frequencies, comprising frequency range 1 (FR1), i.e., sub-6 GHz, and the lower part of frequency range 3 (FR3), spanning 7-24 GHz, especially in dense urban clusters. In contrast, thehigh-band(HB) frequencies, including the frequency range 2 (FR2), i.e., millimeter-wave frequencies, and the upper part of FR3, offer more bandwidth, but are prone to reliability issues due to obstructions that block line-of-sight links between user equipment (UEs) and base stations (BSs). Therefore, we propose offloading high data rate UEs experiencing link outages in HB to MB. Using a cell-free massive multiple-input multiple-output architecture in MB, we ensure stable throughput for both low data rate UEs and offloaded high data rate UEs, while distributing the traffic load across more BSs. Our results demonstrate significant reduction in the mean outage duration and a hundred-fold reduction in the mean outage probability, with minimal impact on service quality for low data rate UEs, offering a cost-effective, practical solution for reliable HB deployments.

As virtualization of Radio Access Networks (RAN) gains momentum, understanding the impact of hardware and software disaggregation on resiliency becomes critical to meet the high availability requirements of mobile networks. Our paper presents an analytical model, using continuous time Markov chains, to study the impact of virtualization and disaggregation on RAN availability. Our evaluation, assuming typical parameter value ranges for failure and recovery rates, points to container-ized platform reliability as a constraint on vRAN availability. We also find that with active-passive replication, increasing hardware replication factor beyond 2 may not bring any benefits unless failover times are reduced. We also compare the reliability of centralized and distributed virtualized central units.

We study the data packet transmission problem (mmDPT) in dense cell-free millimeter wave (mmWave) networks, i.e., users sending data packet requests to access points (APs) via uplinks and APs transmitting requested data packets to users via downlinks. Our objective is to minimize the average delay in the system due to APs’ limited service capacity and unreliable wireless channels between APs and users. This problem can be formulated as a restless multi-armed bandits problem with fairness constraint (RMAB-F). Since finding the optimal policy forRMAB-Fis intractable, existing learning algorithms are computationally expensive and not suitable for practical dynamic dense mmWave networks. In this paper, we propose a structured reinforcement learning (RL) solution formmDPTby exploiting the inherent structure encoded inRMAB-F. To achieve this, we first design a low-complexity and provably asymptotically optimal index policy forRMAB-F. Then, we leverage this structure information to develop a structured RL algorithm calledmmDPT-TS, which provably achieves an$\tilde {\mathcal {O}}(\sqrt {T})$Bayesian regret. More importantly,mmDPT-TSis computation-efficient and thus amenable to practical implementation, as it fully exploits the structure of index policy for making decisions. Extensive emulation based on data collected in realistic mmWave networks demonstrate significant gains ofmmDPT-TSover existing approaches.

Smart glasses that support augmented reality (AR) have the potential to become the consumer's primary medium of connecting to the future internet. For the best quality of user experience, AR glasses must have a small form factor and long battery life, while satisfying the data rate and latency requirements of AR applications. To extend the AR glasses' battery life, the computation and processing involved in AR may be offloaded to a companion device, such as a smartphone, through a wireless connection. Sidelink (SL), i.e., the D2D communication interface of 5G NR, is a potential candidate for this wireless link. In this paper, we use system-level simulations to analyze the feasibility of NR SL for supporting AR. Our simulator incorporates the PHY layer structure and MAC layer resource scheduling of 3GPP SL, standard 3GPP channel models, and MCS configurations. Our results suggest that the current SL standard specifications are insufficient for high-end AR use cases with heavy interaction but can support simpler previews and file transfers. We further propose two enhancements to SL resource allocation, which have the potential to offer significant performance improvements for AR applications.

Oblivious routing of network traffic uses predetermined paths that do not change with changing traffic patterns. It has the benefit of using a fixed network configuration while robustly handling a range of varying and unpredictable traffic. Theoretical advances have shown that the benefits of oblivious routing are achievable without compromising much capacity efficiency. For oblivious routing, we only assume knowledge of the ingress/egress capacities of the edge nodes through which traffic enters or leaves the network. All traffic patterns possible subject to the ingress/egress capacity constraints (also known as the hose constraints) are permissible and are to be handled using oblivious routing. We use the widely deployed segment routing method for route control. Furthermore, for ease of deployment and to not deviate too much from conventional shortest path routing, we restrict paths to be 2-segment paths (the composition of two shortest path routed segments). We solve the 2-segment oblivious routing problem for all permissible traffic matrices (which can be infinitely-many). We develop a new adversarial and machine-learning driven approach that uses an iterative gradient descent method to solve the routing problem with worst-case performance guarantees. Additionally, the parallelism involved in descent methods allows this method to scale well with the network size making it amenable for use in practice.

Internet switches become harder and costlier to build for higher line rates and switch capacities. In-sequence delivery of packets has traditionally been a constraint on switch designs because TCP loss detection was considered vulnerable to out-of-sequence arrivals. For this reason, extremely efficient and simple designs, such as the Load Balanced Birkhoff-von Neumann Switch, were considered impractical. However, we reevaluate this constraint considering modern TCP implementations with loss detection algorithms like Recent Acknowledgment (RACK) that are more resilient to out-of-order arrivals. In a set of testbed experiments representative of wide area core networks, we evaluated the performance of TCP flows traversing a load balanced switch that reorders some packets within a flow. We show that widely deployed and standard TCP implementations of the last decade achieve similar performance when traversing a load balanced switch as they do when there is no reordering. Furthermore, we also verified that an increase in the line rate leads to favorable conditions for time based loss detection methods, such as the one used in RACK. Our results, if further validated, suggest that switch designs that were previously thought to be unsuitable can potentially be utilized, thanks to the relaxation of the in-sequence delivery constraint.

Next generation cellular networks will rely heavily on mmWave and THz spectrum for the abundantly available bandwidth. However, these frequencies suffer from high path and penetration losses. One way to improve the performance is network densification which comes with higher operational cost. To reduce the operational cost of Base Station (BS) deployment, the 3GPP has proposed the Integrated Access and Backhaul (IAB) architecture. Nonetheless, when a handover does occur in IAB networks, even with minimal handover time, the traffic that is already en route to the UE is delayed, as the new serving BS must retrieve the packets from either the previous serving BS or the core. To address these challenges, we propose Fast Wireless Backhaul (FWB), a new wireless backhaul solution. FWB takes advantage of the multi-connectivity of UEs and the high-capacity low-cost wireless backhaul promised by IAB to reduce latency and increase reliability in case of unexpected blockages on the wireless signal path. In FWB, the BSs serving the UE participate in a multicast tree, and receive all packets designated for the UE, but only one BS transmits packets to the UE. In the event of a link failure between the UE and the serving BS, another BS takes over to maintain the data plane connection, without first having to retrieve undelivered downlink packets. We believe our architecture can enable mission critical applications with stringent latency and reliability requirements.

In 5G systems, a predefined codebook with a limited number of beams is used during the initial access and beam management procedures to establish and maintain the connection between the users and the network. At 5G mmimeter wave (mmWave) frequencies, due to the very narrow and directional beams obtained by beamforming, intelligently designing a codebook with a limited number of beams is crucial to avoid coverage holes. We formulate an optimization problem for the beam-codebook design to maximize the coverage probability, which is a quadratically-constrained mixed-integer problem. We propose a set of data-driven codebook design algorithms to solve the optimization problem, which, for a given codebook size constraint, adapts the codebook to the deployment scenario using the provided input channel data. For a sample deployment scenario, we show that as the codebook size increases, the proposed algorithms converge to the upper bound in terms of the coverage probability much faster than several benchmark algorithms. Hence, the proposed algorithms can achieve the coverage levels of benchmark algorithms with a much smaller codebook size. This can significantly reduce the initial access, beam management, and handover delays, which in turn provide higher data rates, lower latency, and lower interruption times.

Device-to-device (D2D) communication using 5G New Radio (NR) sidelink (SL) is envisioned to be a key enabler of high speed, low latency applications, with automated driving being the prime use case. To meet the high data rate requirements, it is essential for SL devices to be able to operate in mmWave/sub-THz frequencies where bandwidth is abundant. Consequently, several enhancements will be needed to the current version of NR SL, which is mainly designed for sub-6 GHz frequencies. Beamforming based highly directional transmission/reception used at high carrier frequencies result in directional deafness in other directions. For SL UE autonomous resource allocation, termed as Mode 2 of NR SL in 3GPP, this results in a UE’s inability to detect transmissions not aligned to its primary direction of reception, which leads to high packet errors. In this paper, we focus on system wide performance of 3GPP based NR SL Mode 2 resource allocation for directional systems at mmWave/sub-THz frequencies. We propose a composite strategy that comprises paired SL control transmission and sensing, whereby SL UEs transmit and receive the SL control information in an additional “paired” direction, directly opposite to their intended direction of transmission. This helps eliminate hidden node interference while avoiding too many exposed nodes. System level simulations in NR V2X highway scenarios show significant performance improvement of the proposed scheme over conventional solutions like omnidirectional or directional transmission/reception of SL control information.

We consider a full-duplex cell-free massive multiple-input-multiple-output system with limited capacity fronthaul links. We derive its downlink/uplink closed-form spectral efficiency (SE) lower bounds with maximum-ratio transmission/maximum-ratio combining and optimal uniform quantization. To reduce carbon footprint, this paper maximizes the non-convex weighted sum energy efficiency (WSEE) via downlink and uplink power control, and successive convex approximation framework. We show that with low fronthaul capacity, the system requires a higher number of fronthaul quantization bits to achieve high SE and WSEE. For high fronthaul capacity, higher number of bits, however, achieves high SE but a reduced WSEE.

Motivated by important industrial automation use cases, such as closed loop motion control and autonomous mobile robots, we study wait-free scheduling of periodic flows with stringent delay and jitter requirements in time sensitive networks. The goal is to assign initial transmission time-slots to periodic flows so that network queuing delays are eliminated or are very small. We make use of Bézour's Identity to develop simple and fast scheduling algorithms for this NP-hard problem. Operating in an online mode, our algorithms can quickly allocate contention free start time-slots to new flows, without changing allocations of already scheduled flows. Our main results are greedy and random scheduling algorithms that can trade speed for solution quality. Our simulations on different network topologies show that these algorithms are computationally efficient and can easily schedule a large number of flows, thus meeting the requirements of many industrial automation use cases.

Wireless communication over terahertz (THz) frequency bands is envisioned as the key enabler of many applications and services offered in 6G networks. The abundantly available bandwidth in THz frequencies can satisfy the ultra-high user throughput requirements and accommodate a massive number of connected devices. However, poor propagation characteristics, shadowing, and blockages may result in sudden outages and necessitate frequent handovers. Therefore, an inefficient handover procedure will impose severe challenges in meeting the ultra-high reliability and low latency requirements of emerging applications. In blockage driven mmWave and THz networks, a higher multi-connectivity degree and efficient handover procedures are needed to reduce the data plane interruptions and to achieve high reliability. We present an analytical model to study the impact of handover procedures and multi-connectivity degree on the latency and reliability of blockage driven wireless networks. From the network protocol design perspective, our study offers a quick and accurate way to envisage how network architecture and protocols should evolve in terms of multi-connectivity degrees and handover procedural efficiency. Our results suggest that, for THz systems, coverage range should be increased even if it comes at the cost of increased initial access and base station discovery times.

In this paper, we propose a general framework for combining deep neural networks (DNNs) with dynamic programming to solve combinatorial optimization problems. For problems that can be broken into smaller subproblems and solved by dynamic programming, we train a set of neural networks to replace value or policy functions at each decision step. Two variants of the neural network approximated dynamic programming (NDP) methods are proposed; in the value-based NDP method, the networks learn to estimate the value of each choice at the corresponding step, while in the policy-based NDP method the DNNs only estimate the best decision at each step. The training procedure of the NDP starts from the smallest problem size and a new DNN for the next size is trained to cooperate with previous DNNs. After all the DNNs are trained, the networks are fine-tuned together to further improve overall performance. We test NDP on the linear sum assignment problem, the traveling salesman problem and the talent scheduling problem. Experimental results show that NDP can achieve considerable computation time reduction on hard problems with reasonable performance loss. In general, NDP can be applied to reducible combinatorial optimization problems for the purpose of computation time reduction.

Millimeter wave wireless systems rely heavily on directional communication in narrow steerable beams. Tools to measure the spatial and temporal nature of the channel are necessary to evaluate beamforming and related algorithms. This paper presents a novel 60 GHz phased-array based directional channel sounder and data analysis procedure that can accurately extract paths and their transmit and receive directions. The gains along each path can also be measured for analyzing blocking scenarios. The sounder is validated in an indoor office environment.

Channel measurements at millimeter wave (mmWave) frequencies are typically carried out using directional antennas to overcome high path loss at mmWave frequencies. Unlike traditional directional channel measurements using narrow beam horn antennas where ray paths can be uniquely mapped to a direction with sufficient high accuracy, phased arrays with irregular beam pattern offers additional complexity in determining the arrival statistics of ray paths. In this work, we propose a systematic method to infer the ray path characteristics with the knowledge of beam patterns used by the phased array. We leverage on the channel measurement data obtained from extensive 60 GHz measurement campaign performed for indoor living room scenario and extract the ray path information from the measured data with high reliability. We also verify our findings through ray tracing simulation.

We consider vehicle-to-infrastructure (V2I) type communication over millimeter wave (mmWave) frequencies and evaluate its performance limitations in terms of coverage. Specifically, we analyze the outage probability and average outage duration for a communicating vehicle exchanging data with roadside units (RSUs), where the system performance is limited by the path-loss that the intermediate links experience. The path-loss depends on the length of the link, as well as the number of vehicles blocking the line-of-sight (LOS) path. By using a Markov chain formulation and its steady-state solution, we obtain analytical expressions for the outage probability and average outage duration. After demonstrating that the durations of outage events constitute a greater limitation on the system performance than outage probability, we examine two approaches to mitigate this limitation: increasing the RSU density and increasing the path-loss threshold for communication.

5G will transform the cellular network from a generic supplier of high-speed data to a flexible, super-fast, and highly-reliable communications medium that supports diverse sets of services and applications. Low packet latency is a critical enabler for the interactivity features of many of these applications, but maybe compromised in saturated packets buffers if the buffers are oversized, as is often the case in the RLC layer of the 3GPP RAN stack. To protect all interactive applications from excessive buffering, we completely revise the flow-control component of a prior solution of ours that splits the per-bearer buffer between the RLC and PDCP layers and regulates the packet flow between the two buffers so that queue build-up in the RLC layer is constrained. On a cellular link shared with a greedy TCP flow, our new enhanced bearer buffer (EBB) design with continuous flow control (CFC) reduces the latency of packets from low-bandwidth interactive applications by at least 80% compared to previous EBB versions, and by orders of magnitude compared to the default bearer configuration with a large RLC buffer. The CFC also benefits throughput-intensive interactive applications like augmented and virtual reality because it confines the buffering within the PDCP layer, where it can be effectively controlled with active queue management techniques.

With the new emerging throughput-intensive ultralow latency applications, there is a need for a transport layer protocol that can achieve high throughput with low latency. One promising candidate is TCP BBR, a protocol developed by Google, with the aim of achieving high throughput and low latency by operating around the Bandwidth Delay Product (BDP) of the bottleneck link. Google reported significant throughput gains and much lower latency relative to TCP Cubic following the deployment of BBR in their high-speed wide area wired network. As most of these emerging applications will be supported by Millimeter Wave (mmWave) wireless networks, BBR should achieve both high throughput and ultra-low latency in these settings. However, in our preliminary experiments with BBR over a mmWave wireless link operating at 60 GHz, we observed a severe degradation in throughput that we were able to attribute to high delay variation on the link. In this paper, we show that “throughput collapse” occurs when BBR's estimate of minimum RTT is less than half of the average RTT of the uncongested link (as when delay jitter is large). We demonstrate this phenomenon and explain the underlying reasons for it using a series of controlled experiments on the CloudLab testbed. We also present a mathematical analysis of BBR, which matches our experimental results closely. Based on our analysis, we propose and experimentally evaluate potential solutions that can overcome the throughput collapse without addina sianificant latency.

Millimeter-wave (mmWave) is a promising network access technology to enable reliable communications in future vehicular networks, which is expected to be dominated by autonomous vehicles. In this study, we consider line-of- sight (LOS) communications between mmWave-enabled vehicles and roadside units (RSUs) in a highway scenario where the LOS links can potentially get blocked by blocking vehicles. We focus on two performance metrics, namely the blockage probability and the average blockage duration. We establish a continuous-time Markov chain formulation of the problem in which transition rates depend on the speed, length and intervehicle distance of blocking vehicles. By using the steady-state solution of this model, we explicitly derive the blockage probability and average blockage duration. After validating the accuracy of the analytical results by comparing with simulations, we demonstrate that the main performance limitation of mmWave systems is due to the blockage duration, not the probability of blockage events. We also discuss the implications of our results in terms of the deployment of RSUs to provide ultra-reliable and low-latency connectivity.

With the increasing number of base stations (BSs) and network densification in 5G, interference management using link scheduling and power control are vital for better utilization of radio resources. However, the complexity of solving link scheduling and the power control problem grows exponentially with the number of BS. Due to high computation time, previous methods are useful for research purposes but impractical for real time usage. In this paper we propose to use deep neural networks (DNNs) to approximate optimal link scheduling and power control for the case with multiple small cells. A deep Q-network (DQN) estimates a suitable schedule, then a DNN allocates power for the corresponding schedule. Simulation results show that compared with Geometric Programming based power allocation and exhaustive search based scheduling, the proposed method achieves over five orders of magnitude speed-up with less than nine percent performance loss, making real time usage practical.

Millimeter wave communication systems can provide high data rates, but the system performance may degrade significantly due to interruptions by mobile blockers, such as humans or vehicles. High-frequency interruptions and lengthy blockage durations will degrade the quality of the user's experience. A promising solution is to employ the macrodiversity of base stations (BSs), where the user equipment (UE) can handover to other available BSs if the current serving BS gets blocked. However, an analytical model to evaluate the system performance of dynamic blockage events in this setting is unknown. In this paper, we develop a line-of-sight (LOS) dynamic blockage model and evaluate the probability, duration, and frequency of blockage events considering all the links to the UE which are not blocked by buildings or the user's own body. For a dense urban area, we also analyze the impact of non-LOS links on blockage events. Our results indicate that the minimum density of the BS required to satisfy the quality of service requirements of ultra-reliable low-latency communication applications will be driven mainly by blockage and latency constraints, rather than coverage or capacity requirements.

Fifth Generation (5G) Millimeter Wave (mmWave) cellular networks are expected to serve a large set of throughput-intensive, ultra-reliable, and ultra-low latency applications. To meet these stringent requirements, while minimizing the network cost, the 3rdGeneration Partnership Project has proposed a new transport architecture, where certain functional blocks can be placed closer to the network edge. In this architecture, however, blockages and shadowing in 5G mmWave cellular networks may lead to frequent handovers (HOs) causing significant performance degradation. To meet the ultra-reliable and low-latency requirements of applications and services in an environment with frequent HOs, we propose the Fast Inter-Base Station Ring (FIBR) architecture, where Base Stations (BSs) that are in close proximity are grouped together, interconnected by a bi-directional counter-rotating buffer insertion ring network. FIBR enables high-speed control signaling and fast-switching among BSs during HOs, while allowing the user equipment to maintain a high degree of connectivity. We demonstrate that the FIBR architecture efficiently handles frequent HO events in mmWave cellular systems, and thus more effectively satisfies the QoS requirements of 5G applications.

5G is expected to expand the set of interactive applications enabled by the mobile network. Low end-to-end packet latency is a fundamental requirement for proper operation of those applications, but may be compromised in packet buffers shared with elastic applications that use greedy TCP sources for end-to-end transport. The accumulation of queuing delay by the elastic application degrades the latency for the interactive application, however light the throughput of the latter may be (as is the case with online gaming and over-the top voice). Buffer sharing is unavoidable in the RLC layer of the 3GPP RAN stack. To minimize its negative effect on interactive applications, we split the buffering between the RLC and PDCP layers. Then we equip the PDCP buffer with per-flow queues, and apply to the RLC buffer a new dynamic sizing mechanism that enforces the lowest queuing delay that is compatible with the existing configuration of the RLC connection. On a cellular link shared with a greedy TCP flow, our dynamic sizing solution can reduce the queuing delay of PING packets by up to two orders of magnitude compared to the default configuration with an RLC-only buffer of fixed size.

The bandwidth required for TV content distribution is rapidly increasing due to the evolution of high definition TV (HDTV) and ultra HDTV. Service providers are constantly trying to differentiate themselves by innovating new ways of distributing content more efficiently with lower cost and higher penetration. We propose a cost-efficient wireless architecture [wireless live TV (WiLiTV)], consisting of a mix of wireless access technologies [satellite, Wi-Fi, and LTE/5G millimeter wave (mmWave) overlay links], for delivering live TV services. In the proposed architecture, live TV content is injected into the network at selected locations, consisting of some homes and/or cellular base stations, using satellite antennas. The content is then further distributed to other homes using a house-to-house Wi-Fi network or an LTE/5G mmWave overlay. We construct an optimal content distribution network with the minimum number of satellite injection points, while preserving the highest quality of experience, for different neighborhood densities. We evaluate the framework using time-varying demand patterns and a diverse set of home location data provided from an operational content distribution network. Our study demonstrates that this architecture reduces the overall cost by 60% compared with the traditional architecture. We have also shown that the WiLiTV is robust in its support for several TV formats.

With the increasing interest in the use of millimeter wave bands for 5G cellular systems comes renewed interest in resource sharing. Properties of millimeter wave bands such as massive bandwidth, highly directional antennas, high penetration loss, and susceptibility to shadowing, suggest technical advantages to spectrum and infrastructure sharing in millimeter wave cellular networks. However, technical advantages do not necessarily translate to increased profit for service providers, or increased consumer surplus. In this paper, detailed network simulations are used to better understand the economic implications of resource sharing in a vertically differentiated duopoly market for cellular service. The results suggest that resource sharing is less often profitable for millimeter wave service providers compared to microwave cellular service providers, and does not necessarily increase consumer surplus.

Full duplex (FD) communications, which increases spectral efficiency through simultaneous transmission and reception on the same frequency band, is a promising technology to meet the demand of next generation wireless networks. In this paper, we consider the application of such FD communication to self-backhauled small cells. We consider a FD capable small cell base station (BS) being wirelessly backhauled by a FD capable macro-cell BS. FD communication enables simultaneous back-haul and access transmissions at small cell BSs, which reduces the need to orthogonalize allocated spectrum between access and backhaul. However, in such simultaneous operations, all the links experience higher interference, which significantly suppresses the gains of FD operations. We propose an interference-aware scheduling method to maximize the FD gain across multiple UEs in both uplink and downlink directions, while maintaining a level of fairness between all UEs. It jointly schedules the appropriate links and traffic based on the back-pressure algorithm, and allocates appropriate transmission powers to the scheduled links using Geometric Programming. Our simulation results show that the proposed scheduler nearly doubles the throughput of small cells compared to traditional half-duplex self-backhauling, even when the uplink/downlink traffic is asymmetric.

As latency sensitive applications, such as online video chatting and virtual reality become popular, end-to-end latency prediction is becoming an important problem. Traditional methods for latency prediction are static, and are unsuitable to predict time-varying round-trip times between the servers and edge devices. Though distance-feature decomposition is able to predict time-varying round-trip times with time sampled information, it fails to utilize network similarity for better prediction. When the time correlation is weak between different sample matrices, it performs no better than static prediction algorithms. But in most cases, as long as network structure remains the same, network similarity still exists even if round-trip times change greatly over time. In this paper we show that similar patterns of round-trip time sequences exist both across time and among different pairs of devices. By exploiting both time correlation and network similarity, we can achieve much lower prediction error even if time correlation of 3D sampled data is weak.

The approaching 5G era of cellular communications is posing stringent performance requirements. New groundbreaking applications can be enabled only by means of multi-Gbps data rates and ultra-low latencies. The spectrum scarcity at frequencies below 6 GHz stimulated a new wave of wireless research that focuses on higher bands, namely mmWave. Directionality and high penetration loss represent the key challenges when operating with such carriers. The resulting intermittent connectivity makes the user association problem even more complex and critical than in previous generations of cellular systems, where the channel was better behaved. In this paper, we aim at deriving an optimal and fair cell selection policy that encapsulates the reallocation cost of potential handovers, and captures the erratic nature of the mmWave channel. An important conclusion is that (i) if there is no, or minimal, reallocation cost, each user associates with a single BS, while (ii) for higher handover cost values, users tend to connect to multiple base stations simultaneously.

In the past decade, mobile devices and applications have experienced an explosive growth, and users are expecting higher data rates and better quality services every year. In this paper, we propose several ideas to increase the functionality and capacity of wireless networks using software-defined networking (SDN) and cloud computing technologies. Connections between users and services in mobile networks typically have to pass through a required set of middleboxes. The complex routing is one of the major impetus for the SDN paradigm, which enables flexible policy-aware routing in the next generation mobile networks. In addition, the high costs of middleboxes and limited capabilities of mobile devices call for revolutionary virtualization technologies enabled by cloud computing. Based on these, we consider an online routing problem for mobile networks with SDN and cloud computing. In this problem, connection requests are given one at a time (as in a real mobile system), and the objective is to steer traffic flows to maximize the total amount of traffic accepted over time, subject to capacity, budget, policy, and quality of service constraints. A fast log-competitive approximation algorithm is developed based on time-dependent duals.

Recent advances in self-interference cancellation enable radios to transmit and receive on the same frequency at the same time. Such a full duplex radio is being considered as a potential candidate for the next generation of wireless networks due to its ability to increase the spectral efficiency of wireless systems. In this paper, the performance of full duplex radio in small cellular systems is analyzed by assuming full duplex capable base stations and half duplex user equipment. However, using only full duplex base stations increases interference leading to outage. We therefore propose a mixed multi-cell system, composed of full duplex and half duplex cells. A stochastic geometry based model of the proposed mixed system is provided, which allows us to derive the outage and area spectral efficiency of such a system. The effect of full duplex cells on the performance of the mixed system is presented under different network parameter settings. We show that the fraction of cells that have full duplex base stations can be used as a design parameter by the network operator to target an optimal tradeoff between area spectral efficiency and outage in a mixed system.

The resource usage model for millimeter wave bands has been the subject of considerable debate. The massive bandwidth, highly directional antennas, high penetration loss and susceptibility to shadowing in these bands suggest certain advantages to spectrum and infrastructure sharing. In particular, resources that are "open", such as unlicensed spectrum or a deployment of base stations open to all service providers, may offer greater gains in mmWave bands than at conventional cellular frequencies. However, even when sharing is technically beneficial (as recent research in this area suggests that it is), it may not be profitable. In this paper, both the technical and economic implications of resource sharing in millimeter wave networks are studied. Millimeter wave service is considered in the economic framework of a network good, and detailed network simulations are used to understand data rates, profit, and demand for millimeter wave service, with and without open resources. The results suggest that "open" deployments of neutral small cells that serve subscribers of any service provider encourage market entry by making it easier for networks to reach critical mass, more than "open" (unlicensed) spectrum would.

A common simplifying assumption made in wireless simulation and modeling is that the world is flat, i.e. to ignore the effect of the terrain in which the wireless signal propagates. In this paper, we show with empirical measurements from an urban wireless network testbed how the terrain affects the spatial and temporal correlation of the wireless signal, and in turn, the distance or duration over which the wireless signal remains consistent. Furthermore, we suggest that this effect has practical implications for systems that make assumptions about the duration over which wireless signal quality stays roughly the same, such as adaptive transmission schemes or applications that buffer data to smooth over variations in signal quality.

In this paper, we present an approach for joint rate allocation and quality selection for a novel video streaming scheme called streamloading. Streamloading is a recently developed method for delivering high-quality video without violating copyright enforced restrictions on content access for video streaming. In regular streaming services, content providers restrict the amount of viewable video that users can download prior to playback. This approach can cause inferior user experience due to bandwidth variations, especially in mobile networks with varying capacity. In streamloading, the video is encoded using scalable video coding, and users are allowed to pre-fetch enhancement layers and store them on the device, while base layers are streamed in a near real-time fashion ensuring that buffering constraints on viewable content are met. We begin by formulating the offline problem of jointly optimizing rate allocation and quality selection for streamloading in a wireless network. This motivates our proposed online algorithms for joint scheduling at the base station and segment quality selection at receivers. The results indicate that streamloading outperforms the state-of-the-art streaming schemes in terms of the number of additional streams we can admit for a given video quality. Furthermore, the quality adaptation mechanism of our proposed algorithm achieves a higher performance than baseline algorithms with no (or limited) video-centric optimization of the base station's allocation of resources, e.g., proportional fairness.

Millimeter wave (mmWave) bands have attracted considerable recent interest for next-generation cellular systems due to the massive available spectrum at these frequencies. However, a key challenge in designing mmWave cellular systems is initial access-the procedure by which a mobile device establishes an initial link-layer connection to a cell. MmWave communication relies on highly directional transmissions and the initial access procedure must thus provide a mechanism by which initial transmission directions can be searched in a potentially large angular space. Design options are compared considering different scanning and signaling procedures to evaluate access delay and system overhead. The channel structure and multiple access issues are also considered. The results of our analysis demonstrate significant benefits of low-resolution fully digital architectures in comparison with single stream analog beamforming.

In a cellular network, connections between the Base Station (BS) and Mobile Stations (MS) may fail when the channel is in a deep fade. Shadow fading is large-scale fading which can cause significant received power loss for a wide area. This will lead to lost connections and/or packet loss which is harmful to mobile users, especially to those who are using real-time applications such as video conferencing. Cooperative communication is an efficient way to reduce outage and provide better Quality of Service (QoS) support for delay sensitive applications. A third station, which is often referred as a relay, can be used to forward signals between the BS and the MS. This paper focuses on a study of the performance of relay deployments under correlated shadow fading. We consider the downlink direction in a single cell deployment, for which the shadowing effect is modeled as an angle and distance based correlated shadowing. The received signal-to- noise ratio (SNR) is then calculated by assuming jointly Gaussian shadow fading at the MS. Simulation results show channel variations over time with fixed user speed under different relay deployments. These results demonstrate that a modest number of relays can improve the performance of real-time applications significantly.

This paper introduces a novel and efficient approach for user cooperation in wireless video multicast using randomized distributed space time codes (R-DSTC), in which the sender first transmits the source packets, and the sender and receivers that have received all source packets then generate and send the parity packets simultaneously using R-DSTC. As more parity packets are delivered, more receivers can recover all source packets and join the parity packet transmission. Four variations of the proposed systems are considered. The first one requires complete channel information between the sender and all receivers and between all receivers to derive the optimal transmission rates for sending source and parity packets, and employs receiver feedback to determine when to terminate parity transmission. The other three suboptimal systems do not require full channel information and/or receiver feedback, and hence are more feasible in practice. All four versions can support significantly higher video rates and correspondingly higher quality of decoded video, than prior approaches in the literature, which require full channel information but not feedback.

The acute disparity between increasing bandwidth demand and available spectrum has brought millimeter wave (mmWave) bands to the forefront of candidate solutions for the next-generation cellular networks. Highly directional transmissions are essential for cellular communication in these frequencies to compensate for higher isotropic path loss. This reliance on directional beamforming, however, complicates initial cell search since mobiles and base stations must jointly search over a potentially large angular directional space to locate a suitable path to initiate communication. To address this problem, this paper proposes a directional cell discovery procedure where base stations periodically transmit synchronization signals, potentially in time-varying random directions, to scan the angular space. Detectors for these signals are derived based on a Generalized Likelihood Ratio Test (GLRT) under various signal and receiver assumptions. The detectors are then simulated under realistic design parameters and channels based on actual experimental measurements at 28 GHz in New York City. The study reveals two key findings: 1) digital beamforming can significantly outperform analog beamforming even when digital beamforming uses very low quantization to compensate for the additional power requirements and 2) omnidirectional transmissions of the synchronization signals from the base station generally outperform random directional scanning.

Recent progress in establishing the capability of radios to operate in full duplex mode on a single channel has been attracting growing attention from many researchers. We extend this work by considering the application to small cells, in particular resource-managed cellular systems similar to the TDD variant of LTE. We derive conditions where full duplex operation provides improved throughput compared to half duplex for a single cell scenario. We present a hybrid scheduler that defaults to half duplex operation but can assign full duplex timeslots when it is advantageous to do so. We compare the performance of such a scheduler with a traditional half duplex scheduler in terms of throughput and energy efficiency. Our simulation results show that we achieve as much as 81% of the capacity doubling promised by full duplex, with limitations deriving from interference effects specific to full duplex operation.

Recent progress on self-interference cancellation technology in a full-duplex system makes self-interference no longer an obstacle that restricts the system from capacity gain. In this paper, assuming both relays and access point (AP) are capable of working in full- duplex mode, we focus on to what extent a wireless relaying network can benefit from full-duplex technology. We first evaluate a conventional full-duplex relaying scheme for IEEE 802.11 networks, where each source picks its optimum relay. Then we propose a novel relaying strategy, where the downlink and uplink traffic can share a common relay/link simultaneously. Using this strategy, we design a hybrid scheduling algorithm that opportunistically switches between direct link, half-duplex and full-duplex relaying. Simulation results shows our hybrid scheme can provide up to 80% more throughput than a traditional half-duplex relaying scheme. These results can be extended to cellular networks.

The single-chip crosspoint-queued (CQ) switch is a compact switching architecture that has all its buffers placed at the crosspoints of input and output lines. Scheduling is also performed inside the switching core and does not rely on latency-limited communications with input or output line-cards. Compared with other legacy switching architectures, the CQ switch has the advantages of high throughput, minimal delay, low scheduling complexity, and no speedup requirement. However, the crosspoint buffers are small and segregated; thus, how to efficiently use the buffers and avoid packet drops remains a major problem that needs to be addressed. In this paper, we consider load balancing, deflection routing, and buffer pooling for efficient buffer sharing in the CQ switch. We also design scheduling algorithms to maintain the correct packet order even while employing multi-path switching and resolve contentions caused by multiplexing. All these techniques require modest hardware modifications and memory speedup in the switching core but can greatly boost the buffer utilizations by up to 10 times and reduce the packet drop rates by one to three orders of magnitude. Extensive simulations and analyses have been done to demonstrate the advantages of the proposed buffering and scheduling techniques. By pushing the on-chip memory to the limit of current ASIC technology, we show that a cell drop rate of 10-8, which is low enough for practical uses, can be achieved under real Internet traffic traces corresponding to a load of 0.9.

As cellular operators are suffering from a data explosion problem, and users are consequently experiencing poor data services, the introduction of femtocells offers a cost-effective way to mitigate this problem. Femtocells enable larger network capacity by increasing spatial reuse of the spectrum and shortening the distance to the users. Existing work has shown that open access femtocells, which allow unregistered macro users to connect, are efficient in reducing inter-cell interference and offloading traffic. However, a major obstacle constraining the potential capability of femtocells and open access is the lack of incentives for privately-owned femtocells to serve unregistered users. Hence in this paper, we propose a Vickrey-Clarke-Groves (VCG) auction based incentive framework for accessing such selfish femtocells. We consider two scenarios: One scenario involves a single macro user and another scenario has multiple macro users. We design auction schemes for both scenarios and show analytically that our schemes are truthful and have low computational complexity. Extensive simulations validate these properties and show huge performance improvement to the macro users.

The single-chip crosspoint-queued (CQ) switch is a self-sufficient switching architecture enabled by state-of-art ASIC technology. Unlike the legacy input-queued or output-queued switches, this kind of switch has all its buffers placed at the crosspoints of input and output lines. Scheduling is also performed inside the switching core, and does not rely on instantaneous communications with input or output line-cards. Compared with other legacy switching architectures, the CQ switch has the advantages of high throughput, minimal delay, low scheduling complexity, and no speedup requirement. However, since the crosspoint buffers are small and segregated, packets may be dropped as soon as one of them becomes full. Thus how to efficiently use the crosspoint buffers and decrease the packet drop rate remains a major problem that needs to be addressed. In this paper, we propose a novel chained structure for the CQ switch, which supports load balancing and deflection routing. We also design scheduling algorithms to maintain the correct packet order caused by multi-path switching. All these techniques require modest hardware modifications and memory speedup in the switching core, but can greatly boost the overall buffer utilization and reduce the packet drop rate, especially for large switches with small crosspoint buffers under bursty and non-uniform traffic.

As cellular networks are turning into a platform for ubiquitous data access, cellular operators are facing a severe data capacity crisis due to the exponential growth of traffic generated by mobile users. In this work, we investigate the benefits of sharing infrastructure and spectrum among two cellular operators. Specifically, we provide a multi-cell analytical model using stochastic geometry to identify the performance gain under different sharing strategies, which gives tractable and accurate results. To validate the performance using a realistic setting, we conduct extensive simulations for a multi-cell OFDMA system using real base station locations. Both analytical and simulation results show that even a simple cooperation strategy between two similar operators, where they share spectrum and base stations, roughly quadruples capacity as compared to the capacity of a single operator. This is equivalent to doubling the capacity per customer, providing a strong incentive for operators to cooperate, if not actually merge.

With the advent of femtocells in cellular networks, the inter-cell handover process will become more complex, frequent and time-sensitive. The legacy handover procedures in 3G and LTE systems were originally designed for handovers between macrocells whose base stations are part of the mobile core networks. The use of the public internet to connect the femtocell base station with the mobile core network will contribute to higher latency if such legacy handover procedures are employed over femtocells. This makes the legacy handover procedures slower, and in most cases inefficient. In the absence of a specific standardized procedure that involves femtocells, rather than using existing procedures designed to handle handovers between macrocells, we propose a new Prefetch-based Fast Handover procedure that is designed to overcome the drawbacks introduced by the usage of the public internet for message paths between femtocell base stations and the mobile core network. The new procedure can be introduced into the existing LTE infrastructure with few modifications to the femtocell base station architecture, and the core network packet flows. The proposed procedure simplifies and speeds the handover procedure at the cost of consuming more network resources at the femtocell base station.

Two-way real-time video communication in wireless networks requires high bandwidth, low delay and error resiliency. This paper addresses these demands by proposing a system with the integration of Network Coding (NC), user cooperation using Randomized Distributed Space-time Coding (R-DSTC) and packet level Forward Error Correction (FEC) under a one-way delay constraint. Simulation results show that the proposed scheme significantly outperforms both conventional direct transmission as well as R-DSTC based two-way cooperative transmission, and is most effective when the distance between the users is large.

Relay-assisted cooperative wireless communication has been shown to have significant performance gains over the legacy direct transmission scheme. Compared with single relay based cooperation schemes, utilizing multiple relays further improves the reliability and rate of transmissions. Distributed space-time coding (DSTC), as one of the schemes to utilize multiple relays, requires tight coordination between relays and does not perform well in a distributed environment with mobility. In this paper, a cooperative medium access control (MAC) layer protocol, called STiCMAC, is designed to allow multiple relays to transmit at the same time in an IEEE 802.11 network. The transmission is based on a novel DSTC scheme called randomized distributed space-time coding (R-DSTC), which requires minimum coordination. Unlike conventional cooperation schemes that pick nodes with good links, STiCMAC picks a transmission mode that could most improve the end-to-end data rate. Any station that correctly receives from the source can act as a relay and participate in forwarding. The MAC protocol is implemented in a fully decentralized manner and is able to opportunistically recruit relays on the fly, thus making it robust to channel variations and user mobility. Simulation results show that the network capacity and delay performance are greatly improved, especially in a mobile environment.

The rapid increase in Internet traffic is forcing packet routers to grow in capacity to meet the demand. Optical packet routers with less buffering and a greater degree of optical transparency are actively being researched as a way to improve energy efficiency and capacity scaling over traditional electronic routers. Since it is difficult to buffer packets in the optical domain, in this paper we analyze the performance of a hybrid optoelectronic packet router. The router architecture has multiple optical switch planes and a shared electronic buffer to resolve output-port contention. By using multiple ports on the switch planes for each input and output fiber, and by using some switch-plane ports to inter-connect the planes, we can achieve a relatively low packet loss ratio in a router with no buffer. In this case, most traffic can be switched using only the through optical paths of the router without entering the shared buffer. The shared electronic buffer is primarily used to reduce the packet drop ratio under periods of heavy loads and occasionally for optical regeneration of a packet.We run extensive simulations to evaluate the performance of the router with varying number of switch plane ports, number of connections to the electronic buffer, and number of interconnections between the switch planes. We show that the router can provide good throughput, with realistic on-off bursty traffic and asynchronous packet arrivals.

Recently, a new paradigm for cognitive radio networks has been advocated, where primary users (PUs) recruit some secondary users (SUs) to cooperatively relay the primary traffic. However, all existing work on such cooperative cognitive radio networks (CCRNs) operate in the temporal domain. The PU needs to give out a dedicated portion of channel access time to the SUs for transmitting the secondary data in exchange for the SUs' cooperation, which limits the performance of both PUs and SUs. On the other hand, Multiple Input Multiple Output (MIMO) enables transmission of multiple independent data streams and suppression of interference via beam-forming in the spatial domain over MIMO antenna elements to provide significant performance gains. Researches have not yet explored how to take advantage of the MIMO technique in CCRNs. In this paper, we propose a novel MIMO-CCRN framework, which enables the SUs to utilize the capability provided by the MIMO to cooperatively relay the traffic for the PUs while concurrently accessing the same channel to transmit their own traffic. We design the MIMO-CCRN architecture by considering both the temporal and spatial domains to improve spectrum efficiency. Further we provide theoretical analysis for the primary and secondary transmission rate under MIMO cooperation and then formulate an optimization model based on a Stackelberg game to maximize the utilities of PUs and SUs. Evaluation results show that both primary and secondary users achieve higher utility by leveraging MIMO spatial cooperation in MIMO-CCRN than with conventional schemes.

In next generation broadband wireless networks, the orthogonal frequency division multiple access (OFDMA) technique provides a flexible physical interface for mobile users (MS) to share the radio resources. Since OFDMA allows MSs in different cells to reuse the same frequency subchannels to increase network spectrum utilization, MSs at the edge of a cell suffer from severe inter-cell interference from co-channel MSs in the neighboring cells. In WiMAX and LTE networks, fractional frequency reuse (FFR) is employed to mitigate the interference for edge users, but may incur a cost in terms of a lower overall spectrum utilization. In this paper, we propose a novel frequency planning scheme to improve the signal-to-interference (SIR) ratio of edge MSs in a multi-cell OFDMA system. A cell is partitioned in a way that minimizes the interference to the edge areas while sustaining a full frequency reuse factor (FRF) for each cell. Simulation results show that our scheme substantially outperforms FFR methods in terms of SIR under the condition of full frequency reuse and high traffic load.

With the increased popularity of mobile multimedia services, efficient and robust video multicast strategies are of critical importance. Cooperative communications has been shown to improve the robustness and the data rates for point-to-point transmission. In this paper, a two-hop cooperative transmission scheme for multicast in infrastructure-based networks is used, where multiple relays forward the data simultaneously using randomized distributed space time codes (RDSTC). This randomized cooperative transmission is further integrated with layered video coding and packet level forward error correction (FEC) to enable efficient and robust video multicast. Three different schemes are proposed to find the system operating parameters based on the availability of the channel information at the source station: RDSTC with full channel information, RDSTC with limited channel information, and RDSTC with node count. The performance of these three schemes are compared with rate adaptive direct transmission and conventional multicast that does not use rate adaptation. The results show that while rate-adaptive direct transmission provides better video quality than conventional multicast, all three proposed randomized cooperative schemes outperform both strategies significantly as long as the network has enough nodes. Furthermore, the performance gap between RDSTC with full channel information and RDSTC with limited channel information or node count is relatively small, indicating the robustness of the proposed cooperative multicast system using RDSTC.

Mobile video broadcasting service, or mobile TV, is expected to become a popular application for 3G wireless network operators. Most existing solutions for video Broadcast Multicast Services (BCMCS) in 3G networks employ a single transmission rate to cover all viewers. The system-wide video quality of the cell is therefore throttled by a few viewers close to the boundary, and is far from reaching the social-optimum allowed by the radio resources available at the base station. In this paper, we propose a novel scalable video broadcast/multicast solution, SV-BCMCS, that efficiently integrates scalable video coding, 3G broadcast, and ad-hoc forwarding to balance the system-wide and worst-case video quality of all viewers at 3G cell. We solve the optimal resource allocation problem in SV-BCMCS and develop practical helper discovery and relay routing algorithms. Moreover, we analytically study the gain of using ad-hoc relay, in terms of users' effective distance to the base station. Through extensive real video sequence driven simulations, we show that SV-BCMCS significantly improves the system-wide perceived video quality. The users' average PSNR increases by as much as 1.70 dB with slight quality degradation for the few users close to the 3G cell boundary.

Fast growing traffic for both the Internet and within data centers has lead to an increasing demand for high-speed switching systems. In this paper, we propose a fully distributed scheduling algorithm with an O(1) complexity, for a switch with an optical switching fabric. The inputs only use local queue information to make their scheduling decisions, and the switch consumes much less power than an electronic switch. Therefore, we call the switch HELIOS: High Energy-efficiency Locally-scheduled Input-queued Optical Switch. HELIOS can achieve 100% throughput for any admissible Bernoulli i.i.d traffic. To our knowledge, this is the first distributed scheduling algorithm to guarantee 100% throughput for an input-queued optical switch.

The promise of a buffered crossbar switch - a crossbar switch with a packet buffer at each crosspoint - is that it can provide good delay performance with much less complex, practical scheduling algorithms. With today's technology, it is now possible to implement it in a single chip. Thus it has attracted great attention recently. Though simple distributed algorithms can achieve 100% throughput under uniform traffic, so far there are no distributed algorithms which can achieve 100% throughput under general admissible arrival patterns. In this paper, we propose a distributed scheduling algorithm which achieves 100% throughput for any admissible Bernoulli arrival traffic. To the best of our knowledge, this is the first distributed algorithm which can achieve this. The algorithm is called DISQUO: DIStributed QUeue input-Output scheduler. Our simulation results also show that DISQUO can provide good delay performance for different traffic patterns.

In this paper we study a two-hop cooperative transmission scheme where multiple relays forward the data simultaneously using Randomized Distributed Space Time Codes (R-DSTC). We propose to integrate this randomized cooperative transmission with layered video coding and packet level Forward Error Correction (FEC) to enable error resilient video multicast. Data rates in both hops as well as the FEC rate are adopted to maximize the video quality. Our results show that while rate-adaptive direct transmission provides better video quality than conventional multicast, randomized cooperative scheme outperforms both strategies significantly.

The performance unfairness problem in a single cell IEEE 802.11 wireless local area network (WLAN) is considered. While existing research is based on the assumption that all nodes have the same transmission success probability and per-node throughput, this fairness exists only if all nodes within range of the access point can sense each other. Recent measurements suggest that this is not necessarily true and terminals can be hidden from each other. In this paper, the impact of hidden terminals on the performance unfairness among individual nodes is investigated via analysis, simulation and experimental measurements in a real network. In the presence of hidden terminals, it is observed that the widely accepted conclusion of equal performance among nodes does not hold any more. Instead, nodes far from the access point (AP) see more hidden terminals than those close to the AP, so they get more packet losses and lower throughput. This phenomenon is not due to inter-cell interference, or channel disparities among nodes, and is significant even when the RTS/CTS mechanism designed to mitigate the impact of hidden terminals is turned on. The simulation results show that for a 16-node WLAN with a fixed data rate of 6 Mbps, the throughput of a node close to the AP is more than twice that of an edge node, due to hidden terminals.

In this paper, a cooperative multicast scheme that uses Randomized Distributed Space Time Codes (R-DSTC), along with packet level Forward Error Correction (FEC), is studied. For the source packets, two-hop transmission is considered, where a packet is transmitted first by the access point (AP), and then forwarded using R-DSTC by the nodes that receive the packet. On the other hand, parity packets are generated by the nodes that receive all the source packets correctly and are transmitted using R-DSTC. The optimum transmission rates for source and parity packets, as well as the number of parity packets required, are determined such that the video quality at all nodes is maximized. It is shown that this scheme can support a higher video rate than a previously developed R-DSTC based scheme where both source and parity packets go through a two-hop transmission, as well as non-cooperative direct transmission.

Wireless video multicast enables delivery of popular events to many mobile users in a bandwidth efficient manner. However, providing good and stable video quality to a large number of users with varying channel conditions remains elusive. In this paper, an integration of layered video coding, packet level forward error correction, and two-hop relaying is proposed to enable efficient and robust video multicast in infrastructure-based wireless networks. First, transmission with conventional omni-directional antennas is considered where relays have to transmit in non-overlapping time slots in order to avoid collision. In order to improve system efficiency, we next investigate a system in which relays transmit simultaneously using directional antennas. In both systems, we consider a non-layered configuration, where the relays forward all received video packets and all users receive the same video quality, as well as a layered setup, where the relays forward only the base-layer video. For each system setup, we consider optimization of the relay placement, user partition, transmission rates of each hop, and time scheduling between source and relay transmissions. Our analysis shows that the non-layered system can provide better video quality to all users than the conventional direct transmission system, and the layered system enables some users to enjoy significantly better quality, while guaranteeing other users the same or better quality than direct transmission. The directional relay system can provide substantial improvements over the omni-directional relay system. To support our results, a prototype is implemented using open source drivers and socket programming, and the system performance is validated with real-world experiments.

Crosspoint buffered switches are emerging as the focus of research in high-speed routers. They have simpler scheduling algorithms and achieve better performance than bufferless crossbar switches. Crosspoint buffered switches have a buffer at each crosspoint. A cell is first delivered to a crosspoint buffer, and then transferred to the output port. With a speedup of 2, a crosspoint buffered switch has previously been proved to provide 100% throughput. In this paper, we propose two 100% throughput scheduling algorithms without speedup for crosspoint buffered switches, called SQUISH and SQUID. We prove that both schemes can achieve 100% throughput for any admissible Bernoulli traffic, with the minimum required crosspoint buffer size being as small as a single cell buffer. Both schemes have a low time complexity ofO(logN), whereNis the switch size. Simulation results show a delay performance comparable to output-queued switches. We also present a novel queuing model that models crosspoint buffered switches under uniform traffic.

Video multicast over wireless local area networks (WLANs) faces many challenges due to varying channel conditions and limited bandwidth. A promising solution to this problem is the use of packet level forward error correction (FEC) mechanisms. However, the adjustment of the FEC rate is not a trivial issue due to the dynamic wireless environment. This decision becomes more complicated if we consider the multi-rate capability of the existing wireless LAN technology that adjusts the transmission rates based on the channel conditions and the coverage range. In order to explore the above issues we conducted an experimental study of the packet loss behavior of the IEEE 802.11b protocol. In our experiments we considered different transmission rates under the broadcast mode in indoor and outdoor environments. We further explored the effectiveness of packet level FEC for video multicast over wireless networks with multi-rate capability. In order to evaluate the system quantitatively, we implemented a prototype using open source drivers and socket programming. Based on the experimental results, we provide guidelines on how to efficiently use FEC for wireless video multicast in order to improve the overall system performance. We show that the Packet Error Rate (PER) increases exponentially with distance and using a higher transmission rate together with stronger FEC is more efficient than using a lower transmission rate with weaker FEC for video multicast.

Wireless video multicast enables delivery of popular events to many mobile users in a bandwidth efficient manner. However, providing good and stable video quality to a large number of users with varying channel conditions remains elusive. A promising solution to this problem is the use of packet level (FEC) mechanisms. However, the adjustment of the FEC rate is not a trivial issue due to the dynamic wireless environment. This decision becomes more complicated if we consider the multi-rate capability of the existing wireless LAN technology that adjusts the transmission rates based on the channel conditions and the coverage range. In this paper, we explore the dynamics of Forward Error Correction (FEC) schemes in multi-rate wireless local area networks. We study the fundamental behavior of a 802.11g network which already has embedded error correction in physical layer, under unicast and broadcast modes in a real outdoor environment. We then explore the effectiveness of packet level FEC over wireless networks with multi-rate capability. In order to evaluate the system quantitatively, we implemented a prototype using open source drivers, and ran experiments. Based on the experimental results, we provide guidelines on how to efficiently use FEC for wireless multicast services in order to improve the overall system performance. We argue that even there is a physical layer error correction, using a higher transmission rate together with stronger FEC is more efficient than using a lower transmission rate with weaker FEC for multicast.

Mobile video broadcasting service, or mobile TV, is a promising application for 3G wireless network operators. Most existing solutions for video broadcast/multicast services in 3G networks employ a single transmission rate to cover all viewers. The system-wide video quality of the cell is therefore throttled by a few viewers close to the boundary, and is far from reaching the social-optimum allowed by the radio resources available at the base station. In this paper, we propose a novel scalable video broadcast/multicast solution, SV-BCMCS, that efficiently integrates scalable video coding, 3G broadcast and adhoc forwarding to balance the system-wide and worst-case video quality of all viewers in a 3G cell. We study the optimal resource allocation problem in SV-BCMCS and develop practical helper discovery and relay routing algorithms. Through analysis and extensive OPNET simulations, we demonstrate that SV-BCMCS can significantly improve the system-wide video quality at the price of slight quality degradation of a few viewers close to the boundary.

A key technology in cooperative communications is distributed space-time coding (DSTC) which achieves spatial diversity gain from multiple relays. A novel DSTC, called randomized distributed space-time coding (R-DSTC), shows considerable advantages over a regular DSTC in terms of system complexity. In this paper, we exploit the benefits of R-DSTC physical (PHY) layer and develop a distributed and opportunistic medium access control (MAC) layer protocol for R-DSTC deployment in an IEEE 802.11 wireless local area network (WLAN). Unlike other cooperative MAC designs, in our proposed PHY-MAC cross-layer framework, there is no need to decide which stations will serve as relays before each packet transmission. Instead, the MAC layer opportunistically recruits relay stations on the fly; any station that receives a packet from the source correctly forwards it to the destination. Through extensive simulations, we validate the efficiency of our MAC layer protocol and demonstrate that network capacity and delay performance is considerably improved with respect to legacy IEEE 802.11g network.

Cooperative communication is a technique that can be employed to meet the increased throughput needs of next-generation WiMAX systems. In a cooperative scenario, multiple stations can jointly emulate the antenna elements of a multi-input multi-output (MIMO) system in a distributed fashion. Although distributed space-time coding (DSTC) is being considered by the IEEE 802.16j/16 m standards for spatial diversity gain, it has several inherent drawbacks. These are addressed in the recently invented randomized distributed space-time coding, called R-DSTC. In this paper, we present the framework for the R-DSTC technique in the emerging relay-assisted WiMAX network, and develop a cooperative medium access control (MAC) layer protocol, called CoopMAX, for R-DSTC deployment in an IEEE 802.16 system. Our scheme couples the MAC layer with the physical (PHY) layer for performance optimization. The PHY layer yields significant diversity gain, while the MAC layer achieves a substantial end-to-end throughput gain. Through extensive simulations, we evaluate the performance of CoopMAX and show that it can generate capacity gains of up to about 77% for an IEEE 802.16 network.

Wireless video multicast enables delivery of popular events to many wireless users in a bandwidth efficient manner. However, providing good and stable video quality to a large number of users with varying channel conditions remains elusive. In our previous work, we integrated layered video coding with cooperative communication to enable efficient and robust video multicast in infrastructure-based wireless networks. Through simulation and analysis, we showed that cooperative multicast improves the multicast system performance and the coverage area. In this work, we integrate the proposed system with packet level forward error correction (FEC) and evaluate the viability of the system in a realistic environment. We implement the system at the MAC layer and report the experimental results in a medium size (i.e., 8 stations) testbed. The experimental results confirm that the new cooperative MAC protocol for multicast, delivers superior performance.

In this paper, we design a simple and robust cooperation scheme that allows multiple relay stations, with only one antenna each, multiplex transmissions to a BS with multiple antennas. The data transmission is over two hops, where the source first sends information to relaying nodes. In the second hop, the data packet is split into multiple parallel streams in a deterministic manner. Each participating relaying node transmits a random and independent linear combination of the signal streams that would have been transmitted by all the elements of a multi-antenna system. This random processing eliminates the need to index and allocate a code for each participating relay, and thus makes the system more efficient and robust. We use the information-theoretic capacity as an upper bound for what can be achieved using this scheme. Since this scheme can achieve higher spatial multiplexing gain for the relay-destination link, it greatly improves the effective data rate for stations at the edge of the cell. This scheme requires multiple antennas at the receiver and is therefore more suitable for uplink transmission in wireless LANs and cellular networks.

Medium access control (MAC) is a key issue for the efficiency of IEEE 802.11 wireless local area network. Legacy 802.11 DCF, using CSMA/CA and a random back-off mechanism, provides a simple-to-implement solution, however it achieves very low throughput. Exploiting the existence of a central entity in the infrastructure mode of an 802.11 wireless network, we propose a new centralized scheduling-based MAC protocol which is based on the popular DCF mechanism. This simple yet efficient scheme illustrates a new paradigm for realistic medium access control protocol design for next generation wireless networks. We have evaluated the performance of the proposed protocol by extensive simulations in different scenarios of wireless networks. Simulation results show that the new protocol significantly improves the network performance in terms of throughput and medium access delay.

Carrier sensing has been used as an effective way to reduce collisions and exploit spatial reuse in wireless networks. Previous research has attempted to tune the carrier sensing range to maximize the network throughput. However, the impact of carrier sensing threshold on the probability of successful transmission has been ignored. In this paper, we derive an analytical model to calculate the successful transmission probability. We then calculate the throughput of routing protocols using different link metrics. To the best of our knowledge, this perhaps is the first attempt to derive the throughput of routing protocols like expected transmission count (ETX) [1] and expected transmission time (ETT) [2] in wireless networks. We also investigate the impact of some other important factors, such as node density, average contention window size and packet length. Our results show that optimal routing protocols that are using ETT as the path metric can achieve around 30% more throughput than those using ETX and End-to-end delay. Compared to the minimum hop count protocols such as DSR and AODV, the optimal routing protocol can improve the throughput by up to 100%.

The load-balanced (LB) switch proposed by C.S. Chang et al. consists of two stages. First, a load-balancing stage converts arriving packets into uniform traffic. Then, a forwarding stage transfers packets from the line-cards to their final output destination. Load-balanced switches do not need a centralized scheduler and can achieve 100% throughput for a broad class of traffic distributions. However, load-balanced switches may cause packets at the output port to be out of sequence. Several schemes have been proposed to tackle the out of- sequence problem of the load-balanced switch. They are either too complex to implement, or introduce a large additional delay. In this paper, we present a practical load-balanced switch, called the Byte-Focal switch, which uses packet-by-packet scheduling to significantly improve the delay performance over switches of comparable complexity. We prove that the queues at the input need only finite buffering, and that the overall switch is stable under any traffic matrix. Our analysis shows that the average queuing delay is roughly linear with the switch size N, and although the worst case resequencing delay is N2, the average resequencing delay is much smaller. This means that we can reduce the required resequencing buffer size significantly.

Although there are several successful commercial deployments of live P2P streaming systems, the current designs 1) lack incentives for users to contribute bandwidth resources, 2) lack adaptation to aggregate bandwidth availability, and 3) exhibit poor video quality when bandwidth availability falls below bandwidth supply. In this paper, we propose, prototype, deploy, and validateLayerP2P, a P2P live streaming system that addresses all three of these problems. LayerP2P combines layered video, mesh P2P distribution, and a tit-for-tat-like algorithm, in a manner such that a peer contributing more upload bandwidth receives more layers and consequently better video quality. We implement LayerP2P (including seeds, clients, trackers, and layered codecs), deploy the prototype in PlanetLab, and perform extensive experiments. We also examine a wide range of scenarios using trace-driven simulations. The results show that LayerP2P has high efficiency, provides differentiated service, adapts to bandwidth deficient scenarios, and provides protection against free-riders.

In the next-generation WiMAX system, cooperative communication is being considered as an advanced technique to increase the throughput and improve the signal quality. In a cooperative scenario, multiple stations can jointly emulate the antenna elements of a multi-input multi-output (MIMO) system in a distributed fashion. Unlike conventional space-time coding (STC) mechanisms used by a IEEE 802.16e antenna array, distributed space-time coding (DSTC) is employed across the cooperating stations to achieve a higher spatial diversity gain. In this paper, we present the framework for DSTC in the emerging relay-assisted WiMAX network, and develop a cooperative MAC layer protocol, called CoopMAX, for DSTC deployment in a WiMAX system. Through extensive simulations, we evaluate the performance of CoopMAX and show that DSTC can yield capacity gains of up to about 50% for the uplink of an IEEE 802.16 network.

In a distributed wireless network, it is possible to employ several relays and mimic a multiple antenna transmission system. In this paper we propose a MAC layer solution that allows multiple relays to send information to the receiver at unison, using a randomized distributed space time code. The randomized space-time coding can recruit relays on the fly, thus significantly reducing signaling overhead. The cross-layer design between physical layer and MAC layer involves relay discovery and rate adaptation, and results in improvements in throughput and delay performance. The design is dynamic and can be adapted to changing network conditions. The proposed MAC scheme can be integrated into various wireless technologies such as distributed contention based networks (e.g., IEEE 802.11 BSS and ad hoc mode) as well as centralized multiple access networks (e.g., IEEE 802.16).

Wireless video multicast enables delivery of popular events to many wireless users in a bandwidth efficient manner. However, providing good and stable video quality to a large number of users with varying channel conditions remains elusive. We propose to integrate layered video coding with cooperative communication to enable efficient and robust video multicast in infrastructure-based wireless networks. We determine the user partition and transmission time scheduling that can optimize a multicast performance criterion.

The IEEE 802.16/WiMAX standard has fully embraced multi-antenna technology and can, thus, deliver robust and high transmission rates and higher system capacity. Nevertheless, due to its inherent form-factor constraints and cost concerns, a WiMAX mobile station (MS) should preferably contain fewer radio frequency (RF) chains than antenna elements. This is because RF chains are often substantially more expensive than antenna elements. Thus, antenna selection, wherein a subset of antennas is dynamically selected to connect to the limited RF chains for transceiving, is a highly appealing performance enhancement technique for multi-antenna WiMAX terminals. In this paper, a novel antenna selection protocol tailored for next-generation IEEE 802.16 mobile stations is proposed. As demonstrated by the extensive OPNET simulations, the proposed protocol delivers a significant performance improvement over conventional 802.16 terminals that lack the antenna selection capability. Moreover, the new protocol leverages the existing signaling methods defined in 802.16, thereby incurring a negligible signaling overhead and requiring only diminutive modifications of the standard. To the best of our knowledge, this paper represents the first effort to support antenna selection capability in IEEE 802.16 mobile stations.

As the two key technologies that have the potential to reshape the landscape of next-generation wireless network, cooperative communications and directional antenna system so far have been developed in parallel, if not in isolation from each other. In order to establish a thorough comparison between the relative system performance of cooperative diversity and directional transmission in an adhoc environment, we design and quantitatively evaluate three medium access control protocols, namely O-CoopMAC, D-NoopMAC and D-CoopMAC. The study yields the unexpected yet crucial observation that cooperative forwarding significantly limits the spatial reuse created by transmission directionality, and therefore can appreciably degrade the performance of a directional system with a sufficiently narrow antenna beam. To the best knowledge of the authors, this is the first paper to systematically compare the performance of these two technologies in an adhoc environment and reveal the key fact that cooperation and directionality can be rather foes than friends!

In this paper, we explore the use of directional antennas in relay transmission to improve the performance of video multicast with omni-directional relays in infrastructure- based wireless networks. We describe the system setup with directional relays and determine the user partition along with transmission time scheduling that can optimize a multicast performance criterion. We demonstrate that directional relays significantly improve the multicast system performance compared to omni-directional relays. Furthermore, it also provides larger coverage area.

We consider the design of an open P2P live-video streaming system. When designing a live video system that is both open and P2P, the system must include mechanisms that incentivize peers to contribute upload capacity. We advocate an incentive principle for live P2P streaming: a peer’s video quality is commensurate with its upload rate. We propose Substream Trading, a new P2P streaming design which not only enables differentiated video quality commensurate with a peer’s upload contribution but can also accommodate different video coding schemes, including single-layer coding, layered coding, and multiple description coding. Extensive trace-driven simulations show that substream trading has high efficiency, provides differentiated service, low start-up latency, synergies among peers with different Internet access rates, and protection against free-riders.

Cooperation in wireless networks has shown significant performance gains in comparison to legacy wireless networks. Cooperative wireless protocols achieve such efficiency by enabling cooperation among nodes to exploit spatial diversity. CoopMAC is a medium access control (MAC) protocol that enables cooperation by using an intermediate node as a helper to a direct communication under poor channel conditions. The helper is typically located in a position where it experiences a good channel with both the source and destination. Therefore, it increases the efficiency of the communication by forwarding a packet from the source to the destination using high transmission rates. In an earlier attempt, we demonstrated the benefits of cooperation at the MAC layer by implementing the CoopMAC protocol using an open source wireless driver platform. However, due to some limitations posed by the hardware, the full potential of the protocol could not be explored. In this paper, we proceed with a complete implementation of the cooperative MAC protocol using an OFDM based software defined radio (SDR) platform. We investigate the benefits of the SDR approach, describe the details of the implementation, as well as the experiments we run in order to evaluate the protocol. Experimental results show that CoopMAC can easily be implemented and can lead to a significant improvement in the performance of wireless networks.

In multicast/broadcast services over infrastructure- based/cellular wireless networks (e.g. 3G cellular networks, WiMax, DVB), data is transmitted to multiple recipients from an access point/base station. Multicast greatly improves the network efficiency to distribute data to multiple recipients as compared to multiple unicast sessions of the same data to each receiver individually, by taking advantage of the shared nature of the wireless medium. However it is difficult to guarantee the reception reliability of multiple multicast/broadcast recipients because the wireless medium is error prone and each receiver experiences different channel conditions. An additional difficulty is that multicast/broadcast services in many networks such as 3G multimedia multicast services do not provide a reverse communications channel for the receivers to request the retransmission of lost data packets. This research proposes a novel method to provide QoS support by using an assistant network to recover the loss of multicast data in the principal network. Wireless devices are connected to the principal network to receive the multicast data. A wireless device may lose some of the multicast data sent over the principal network. The wireless devices form an assistant network to recover the lost multicast data cooperatively from their peers. The performance of this recovery mechanism has been investigated using extensive simulation experiments.

The IEEE 802.16J protocol for a multi-hop relay (MMR) WiMAX network is being developed to increase data rates and extend service coverage as an enhancement of existing WiMAX standards. The IEEE 802.16J protocol supports transparent and non-transparent modes. In the transparent mode, only data traffic is relayed by an intermediate relay station (RS) between a mobile station (MS) and the base station (BS), while in the non-transparent mode, both signaling and data traffic are forwarded by RSs. Furthermore, non-transparent mode is either distributed or centralized with regard to scheduling. The difference between them resides in that distributed scheduling enables RSs to participate in bandwidth allocation (BWAlloc), while centralized scheduling leaves all BWAlloc coordinated by the BS. In this paper, we propose a novel multi-hop polling service (mPS) for non-transparent centralized scheduling in a multi- hop 802.16J environment. Our model is adaptive to the traffic pattern so as to provide bandwidth efficiency over access and relay links. Besides, aggregation of bandwidth requests (BWReq) from MSs is conducted at the RS to further save bandwidth. The performances of mPS with BWReq aggregation is evaluated via simulations which demonstrate our approach outperforms the current multi-hop bandwidth request mechanism in terms of overall spectrum efficiency.

Cooperative communications is a promising technology that tends to change the conventional access and transmission schemes in wireless networks. By enabling additional collaboration from nodes that otherwise will not directly participate in the transmission, it enables spatial diversity and dramatically improves the performance of the network. In this paper we propose a cross-layer cooperative protocol based on a MAC protocol called CoopMAC (Liu et al., 2005; Korakis et al., 2007; and Liu et al., 2006) for ad-hoc wireless networks in order to leverage cooperation in both MAC and PHY layer. Exploiting physical layer combining at the receiver, this simple yet efficient scheme illustrates a new paradigm for realistic cross-layer cooperative protocol design for next generation wireless ad-hoc networks. We have evaluated the performance of the proposed protocol by extensive simulations in a large scale wireless ad-hoc network. Simulation results show that the new protocol significantly improves the network performance in terms of throughput and delay.

It has been shown in the literature that many MAC protocols for wireless networks have a considerable control overhead, which limits their achievable throughput and delay performance. In this paper, we study the problem of improving the efficiency of MAC protocols. We first analyze the popular p- Persistent CSMA scheme and show that it does not achieve 100% throughput.Motivated by insights from polling system theory, we then present three polling service-based MAC schemes, termed PSMACs, for improved performance. The main idea is to serve multiple data frames after a successful contention resolution, thus amortizing the high control overhead and making the protocols more efficient. We present analysis and simulation studies of the proposed schemes. Our results show that PSMAC can effectively improve the throughput and delay performance of p-Persistent CSMA, as well as providing energy savings. We also observe that PSMAC is more efficient for handling the more general and challenging bursty traffic and outperforms p-Persistent CSMA with respect to fairness.

In this paper, we consider applying multiple description coding in mesh-pull P2P live streaming networks to provide incentives for redistribution. In our system, a video is encoded into multiple descriptions with each description having equal importance. We consider a heterogeneous system with peers having different uplink bandwidths. We design a distributed protocol in which a peer contributing more uplink bandwidth receives more descriptions and consequently better video quality. Previous approaches consider single-layer video, where each peer receives the same video quality no matter how much bandwidth it contributes to the system. The simulation results show that our approach can provide differentiated video quality commensurate with a peer's contribution to other peers.

Due to the broadcast nature of wireless signals, a wireless transmission intended for a particular destination station can be overheard by other neighboring stations. A focus of recent research activities in cooperative communications is to achieve spatial diversity gains by requiring these neighboring stations to retransmit the overheard information to the final destination. In this paper we demonstrate that such cooperation among stations in a wireless LAN (WLAN) can achieve both higher throughput and lower interference. We present the design for a medium access control protocol called CoopMAC, in which high data rate stations assist low data rate stations in their transmission by forwarding their traffic. In our proposed protocol, using the overheard transmissions, each low data rate node maintains a table, called a CoopTable, of potential helper nodes that can assist in its transmissions. During transmission, each low data rate node selects either direct transmission or transmission through a helper node in order to minimize the total transmission time. Using analysis, simulation and testbed experimentation, we quantify the increase in the total network throughput, and the reduction in delay, if such cooperative transmissions are utilized. The CoopMAC protocol is simple and backward compatible with the legacy 802.11 system. In this paper, we also demonstrate a reduction in the signal-to-interference ratio in a dense deployment of 802.11 access points, which in some cases is a more important consequence of cooperation

authors calculate the maximum number of VoIP calls that a WiFi (VoFi) network can support. In this paper we extend their analysis to calculate the maximum number of VoFi calls when two-hop forwarding is used in order to avoid rate adaptation at nodes with reduced received signal strength. We calculate this number for different combinations of data rates for slow node transmissions and potential two-hop transmissions. These calculations demonstrate that the use of two-hop forwarding increases the maximum number of VoIP calls in a multi-rate 802.11b network. We validate the analysis by means of simulation of G711 codec sources in a 802.11b network. Even though the earlier discussions focus on specific combinations of high data rate two-hop and low data rate one-hop transmissions in an 802.11b network, we conclude the paper by calculating the maximum number of VoIP calls that can be supported with uniform node distri-bution in 802.11b and 802.11g networks. These calculations show a significant increase in the number of VoIP calls when two-hop forwarding is used in an 802.11g network. This sig-nificant increase is due to the higher data rates and the low PHY overhead of the 802.11g MAC.

Virtual Output Queuing is widely used by highspeed packet switches to overcome head-of-line blocking. This is done by means of matching algorithms. In fixed-length VOQ switches, variable-length IP packets are segmented into fixed- length cells at the inputs. When a cell is transferred to its destination output, it will stay in the reassembly buffer and wait for the other cells of the same packet before the entire packet can depart the system. The delay a packet suffers in the system includes the waiting time in the VOQ, the widely studied cell delay, and the waiting time at the output reassembly buffer, the reassembly delay often ignored in many papers. Among all existing matching algorithms, Maximum Weight Matching (MWM) has the lowest average cell delay. In this paper, we investigate the average packet delay, one of the key performance measure for an input buffered packet switch. A new class of matching algorithms, PDA-MWM, is defined and proved to be stable under all admissible traffic. Three PDA-MWM matching algorithms are studied by simulation. We show that, in order to achieve low packet delay, there is a tradeoff between the cell delay performance and the reassembly delay performance. If both of them are carefully considered, a matching scheme can greatly reduce the packet delay as compared to MWM.

In wireless LANs that provide multi-rate support (IEEE 802.11a, 802.11b), stations that experience poor channel quality tend to use low transmission rates to reduce the bit-error-rate (BER) of each transmission. This phenomenon usually leads to a throughput fairness problem between the stations with good channel quality and those without. This fairness problem has been shown to result in throughput degradation for the whole network [8]. The MAC protocol proposed in [5] addresses this issue using an efficient cooperative scheme. Under this scheme, low rate stations are assisted by a high rate station, referred to as helper stations, in its transmissions. With such assistance, the low rate station will be able to transmit data at a higher rate in a two-hop manner using the helper station. We implemented this new protocol in a Linux testbed. This paper describes the assumptions, the implementation process and the challenges we were presented with. We evaluated the protocol using our testbed through experiments. The implementation of the protocol shows that it performs efficiently in supporting TCP applications.

In this paper, we examine the prefetching strategies in a peer-driven video on-demand system. In our design, each video is encoded into multiple low bit-rate substreams and copies of the substreams are distributed to the participating peers. When a peer streams in a substream of rate r, it instead streams at rate rcirc, where r>rcirc. In this manner, if one of the peer's suppliers disconnects, the client peer can tap the reservoir of prefetched bits while searching for a replacement server, thereby avoiding any glitches or reduced visual quality. We examine how to assign prefetching rates to each of substreams as a function of their importance. Our studies show that appropriate prefetching strategies can bring significant performance improvements for both multiple description and layered videos

IEEE 802.15.4-2003 (2003) was introduced to address the market need for connecting low-rate devices in a wireless personal area network (WPAN). A slotted CSMA/CA medium access control (MAC) protocol is defined in the standard to coordinate the channel access of a large number of wireless devices. In this paper, we propose a novel Markov chain for IEEE 802.15.4 MAC, which faithfully captures all the essential features of the protocol, and thus can provide valuable insight into the strengths and weaknesses of this multiple access scheme. The evaluation reveals that the double carrier sensing mechanism specified in 802.15.4 MAC is not an optimal design, and a slight modification in the protocol can result in further performance improvement in terms of throughput, delay and energy efficiency

In this letter, we propose a three-dimensional Markov chain model for the 802.11e enhanced distributed channel access (EDCA) mode. This model can be used to compute the maximum sustainable throughput and service delay distribution for each priority class when under saturation load. The new framework models the performance impact of major quality-of-service (QoS)-specific features (e.g., CWMin, CWMax, AIFS, internal collision resolution) of the 802.11e EDCA mode, and hence can provide an analytical approach to pick the parameter values associated with EDCA to meet the QoS requirements of each priority.

Real-time multimedia transport has stringent quality of service requirements, which are generally not supported by current network architectures. In emerging mobile ad hoc networks, frequent topology changes and link failures cause severe packet losses, which degrade the quality of received media. However, in such mesh networks, there usually exist multiple paths between any source and destination nodes. Such path diversity has been demonstrated to be effective in combating congestion and link failures for improved media quality. In this paper, we present a new protocol to facilitate multipath transport of real-time multimedia data. The proposed protocol, the multiflow real-time transport protocol (MRTP), provides a convenient vehicle for real-time applications to partition and transmit data using multiple flows. We demonstrate through analysis that data partitioning, which is an essential function of MRTP, can effectively reduce the short-range dependence of multimedia data, thus improving its queueing performance in underlying networks. Furthermore, we show that a few flows are sufficient for MRTP to exploit most of the benefits of multipath transport. Finally, we present a comprehensive simulation study on the performance of MRTP under a mobile ad hoc network. We show that with one additional path, MRTP outperformed single-flow RTP by a significant margin.

Multimedia traffic is becoming an increasing portion of today's Internet traffic due to the flourishing of multimedia applications such as music/video streaming, video teleconferencing, IP telephony, and distance learning. In this paper, we study the problem of supporting multimedia traffic using a generalized processor sharing (GPS) server. By examining the sample path behavior and exploring the inherent feasible ordering of the classes, we derive tight performance bounds on backlog and delay for regulated multimedia traffic classes in a GPS system. Our approach is quite general since we do not assume any arriving traffic model or any specific traffic regulator, other than that each traffic flow is deterministically regulated. Such deterministic regulators, as well as approximations of the GPS server, are widely implemented in commercial routers. In addition, our analysis is very accurate and achieves a high utilization of the server capacity, since we exploit the independence among the traffic flows for higher statistical multiplexing gains. Numerical examples and simulation results are presented to demonstrate the accuracy and merits of our approach, which is practical and well suited for supporting multimedia applications in the Internet

Recent research shows that animals can be guided remotely by stimulating regions of the brain. Therefore, it is possible to set up an animal mobile sensor network for search and rescue operations. Applications of such an animal sensor network have great importance to society, including natural disaster recovery, homeland security and military operations. In this paper, the system architecture and operation is introduced, and major challenges and issues are discussed. Because of its unique challenge, a simple and efficient routing scheme is devised for this special ad hoc network. Each animal needs to carry a backpack to perform sensing and network communications. The implementation of a backpack prototype is presented, and a simple network is set up to capture and transfer video sensor data.

In this paper, a novel idea of user cooperation in wireless networks has been exploited to improve the performance of the IEEE 802.11 medium access control (MAC) protocol. The new MAC protocol leverages the multi-rate capability of IEEE 802.11b and allows the mobile stations (STA) far away from the access point (AP) to transmit at a higher rate by using an intermediate station as a relay. Two specific variations of the new MAC protocol, namely CoopMAC I and CoopMAC II, are introduced in the paper. Both are able to increase the throughput of the whole network and reduce the average packet delay. Moreover, CoopMAC II also maintains backward compatibility with the legacy 802.11 protocol. The performance improvement is further evaluated by analysis and extensive simulations.

Multimedia traffic is becoming an increasing portion of today's Internet traffic due to the flourish of multimedia applications such as music/video streaming, video teleconferencing, IP telephony, and distance learning. In this paper, we study the problem of supporting multimedia traffic using a generalized processor sharing (GPS) server. We derive tight performance bounds on the backlog and delay for regulated multimedia traffic classes in a GPS system. Our approach is quite general since we do not assume any arriving traffic model or any specific traffic regulator, other than that each traffic flow is deterministically regulated. Such deterministic regulators, as well as approximations of the GPS server, are amenable to implementation and are widely implemented in commercial routers. In addition, our analysis is very accurate and has a much high utilization of the server capacity, since we exploit the independence among the traffic flows for higher statistical multiplexing gains. Numerical examples and simulation results are presented to demonstrate the accuracy and merits of our approach, which is practical and well suited for supporting multimedia applications in the Internet.

Peer-to-peer video streaming has emerged as an important means to transport stored video. The peers are less costly and more scalable than an infrastructure-based video streaming network which deploys a dedicated set of servers to store and distribute videos to clients. In this paper, we investigate streaming layered encoded video using peers. Each video is encoded into hierarchical layers which are stored on different peers. The system serves a client request by streaming multiple layers of the requested video from separate peers. The system provides unequal error protection for different layers by varying the number of copies stored for each layer according to its importance. We evaluate the performance of our proposed system with different copy number allocation schemes through extensive simulations. Finally, we compare the performance of layered coding with multiple description coding.

Multipath transport provides higher usable bandwidth for a session. It has also been shown to provide load balancing and error resilience for end-to-end multimedia sessions. Two key issues in the use of multiple paths are (1) how to minimize the end-to-end delay, which now includes the delay along the paths and the resequencing delay at the receiver, and (2) how to select paths. In this paper, we present an analytical framework for the optimal partitioning of realtime multimedia traffic that minimizes the total end-to-end delay. Specifically, we formulate optimal traffic partitioning as a constrained optimization problem using deterministic network calculus, and derive its closed form solution. Compared with previous work, our scheme is simpler to implement and enforce. This analysis also greatly simplifies the solution to the path selection problem as compared to previous efforts. Analytical results show that for a given flow and a set of paths, we can choose a minimal subset to achieve the minimum end-to-end delay with O(N) time, where N is the number of available paths. The selected path set is optimal in the sense that adding any rejected path to the set will only increase the end-to-end delay.

IEEE 802.11 specifies two modes of operation, an infrastructure mode where nodes communicate to/through an access point, and an ad-hoc mode, where nodes communicate with each other directly. Neither mode supports multiple hop transmissions between these nodes. In this paper we present two advantages in extending 802.11 MAC to support multiple hops in the infrastructure mode. One advantage is higher available bandwidth in a multi-rate 802.11 network. IEEE 802.11 allows hosts to select different transmission rates based on the quality of the signal received by the host. Based on performance results both from analytical modeling and simulations, we demonstrate that the total available bandwidth can be improved by using multiple hops instead of reducing the transmission rates of nodes. We present the results in terms of both the increase in total throughput of the network and the available throughput for the forwarding node. The second advantage presented is that by using multi-hop transmissions, the power of transmission at the edges of 802.11 cells can be reduced resulting in lower interference with nodes at the edges of other 802.11 cells. This leads to a more uniform coverage, with increased throughput experienced by nodes at the cell edges.

Real-time multimedia transport has stringent bandwidth, delay, and loss requirements. Supporting this application in current wireless ad hoc networks is a challenge. Such networks are characterized with frequent link failures, as well as congestion. Consequently, data packets are dropped when a link fails or congestion occurs, resulting in low received quality. In addition, a realtime multimedia service may be unavailable when a particular server is unreachable. In this article, we make the case for using multipath transport for realtime multimedia services in wireless ad hoc networks, which provides a unified solution to the above problems. We review existing work on multipath multimedia transport, and discuss the advantages, as well as related issues, of using multipath transport for realtime multimedia transport.

Recent research has demonstrated that it is possible to use wireless communication to deliver brain stimulation to guide the movements of rats through a variety of terrains by stimulating multiple brain regions to produce stimulus cues for various commanded movements. The rats can be controlled not only while in direct line of sight of a human controller, but also by teleoperation, using signals transmitted from a wireless video camera mounted on the animal's backpack. Remotely guided rats or other animals are ideal for search and rescue operations because they are highly adept at negotiating difficult 3D terrain, in both light and dark. The animals autonomously choose their own methods for traversing particular obstacles. Primarily for these reasons, they can be more effective than mechanical robots in search and rescue applications. They can also be trained to detect and home in on specific sensory targets, allowing them to be used as biosensors. To meet these challenges, a cluster of interrelated research studies, including network topology design, routing algorithms, media access control techniques, radio resource management, radio propagation modeling, and mobility modeling are comprised. In this paper, the authors have designed and implemented the hardware of backpacks and set up a simple wireless ad hoc network by using static routing.

The IEEE 802.11e protocol is designed to enhance the QoS capability of wireless local area networks (WLAN). In this paper, we propose a three dimensional Markov chain model for the 802.11e enhanced distributed coordination function (EDCF) mode and compute the maximum sustainable throughput and service delay distribution for each priority class when under heavy load. This provides an analytical approach to pick the parameter values associated with EDCF to meet the QoS requirements of each priority. This is accomplished by modeling the performance impact of all the major QoS-specific features (i.e. CWMin, CWMax, AIFS, internal collision resolution) of the 802.11e EDCF mode.

The IEEE 802.11e protocol is designed to enhance the QoS capability of wireless local area networks (WLAN). In this paper, we propose a multidimensional Markov model for the 802.11e enhanced distributed coordination function (EDCF) mode and compute the maximum sustainable throughput and service delay distribution for each priority class when under heavy load. Since the QoS mechanisms and their associated parameters in IEEE 802.11e interact with each other in a complex way, it is important to model the aggregate effect of all the mechanisms. The approach we present does so and therefore provides an analytical approach to pick the parameter values associated with EDCF to meet the QoS requirements of each priority.

Caching is an important means to scale up the growth of the Internet. Weak consistency is a major approach used in Web caching and has been deployed in various forms. The paper investigates some fundamental properties and performance issues associated with an expiration-based caching system. We focus on a hierarchical caching system based on the time-to-live expiration mechanism and present a basic model for such system. By analyzing the intrinsic timing behavior of the basic model, we derive important performance metrics from the perspectives of the caching system and end users, respectively. Based on the results for the basic model, we introduce threshold-based and randomization-based techniques to enhance and generalize the basic model further. Our results offer some important insights into a hierarchical caching system based on the weak consistency paradigm.

We consider the fundamental problem of bandwidth allocation among flows in a packet-switched network. The classical max-min rate allocation has been widely regarded as a fair rate allocation policy. But, for a flow with a minimum rate requirement and a peak rate constraint, the classical max-min policy no longer suffices to determine rate allocation since it is not capable of supporting either the minimum rate or the peak rate constraint from a flow. We generalize the theory of the classical max-min rate allocation with the support of both the minimum rate and peak rate constraints for each flow. Additionally, to achieve generalized max-min rate allocation in a fully distributed packet network, we present a distributed algorithm that uses a feedback-based flow control mechanism. Our design not only offers a fresh perspective on flow marking technique, but also advances the state-of-the-art flow marking technique favored by other researchers. We provide proof that such a distributed algorithm, through asynchronous iterations, will always converge to the generalized max-min rate allocation under any network configuration and any set of link distances. We use simulation results to demonstrate the fast convergence property of the distributed algorithm.

Enabling video transport over ad hoc networks is more challenging than over other wireless networks. The wireless links in an ad hoc network are highly error prone and can go down frequently because of node mobility, interference, channel fading, and the lack of infrastructure. However, the mesh topology of ad hoc networks implies that it is possible to establish multiple paths between a source and a destination. Indeed, multipath transport provides an extra degree of freedom in designing error resilient video coding and transport schemes. In this paper, we propose to combine multistream coding with multipath transport, to show that, in addition to traditional error control techniques, path diversity provides an effective means to combat transmission error in ad hoc networks. The schemes that we have examined are: 1) feedback based reference picture selection; 2) layered coding with selective automatic repeat request; and 3) multiple description motion compensation coding. All these techniques are based on the motion compensated prediction technique found in modern video coding standards. We studied the performance of these three schemes via extensive simulations using both Markov channel models and OPNET Modeler. To further validate the viability and performance advantages of these schemes, we implemented an ad hoc multiple path video streaming testbed using notebook computers and IEEE 802.11b cards. The results show that great improvement in video quality can be achieved over the standard schemes with limited additional cost. Each of these three video coding/transport techniques is best suited for a particular environment, depending on the availability of a feedback channel, the end-to-end delay constraint, and the error characteristics of the paths.

We investigate transport layer processing intended to improve the communication of multimedia data over wireless and wireline networks. Specifically, we consider data striping and thinning techniques, which are applicable to the multipath/multiflow transmission of multimedia (e.g., MPEG-4). We also introduce the shuffling procedure, which reorders the data at the network edges. We show that all three techniques break up short-term correlations in data traffic thus improving its queueing performance. We demonstrate that while both long range dependence (LRD) and short range dependence (SRD) influence the queueing performance for some timescale of the queueing system, it is the short-range statistical properties of multimedia traffic within the critical time scale that are dominant in determining the buffer efficiency of the queue. We also show that estimation of LRD for thinned and striped data may lead to a misleading notion of LRD reduction, when none is warranted. We further outline the ideas for a new transport layer protocol that explores the combination of thinning, striping and shuffling approaches to multimedia data transmission.

Caching is an important means to scale up the growth of the Internet. Weak consistency is a major approach used in Web caching and has been deployed in various forms. The paper investigates some properties and performance issues of an expiration-based caching system. We focus on a hierarchical caching system based on the time-to-live (TTL) expiration mechanism and present a basic model for such a system. By analyzing the intrinsic TTL timing behavior in the basic model, we derive several important performance metrics from the perspective of the caching system and end users, respectively. Our results offer some basic understanding of a hierarchical caching system based on the weak consistency paradigm.

Typical video applications may need a higher bandwidth and/or higher reliability connection than that provided by a single link in current or emerging wireless networks. We propose to employ path diversity to provide higher bandwidth and more robust end-to-end connections than that affordable by a single path. Under this transport environment, two viable strategies for video coding are multiple description coding (MDC) and layered coding (LC). MDC is more effective when the underlying application has a very stringent delay constraint and the round trip time on each path is relatively long. LC can be a good alternative when limited retransmission of the base layer is acceptable and when it is feasible to apply unequal error protection over different paths. The paper describes the general issues involved in integrating MDC/LC with multiple path transport, and compares the performances of MDC and LC, under different path conditions.

This paper examines the effectiveness of combining multiple description coding (MDC) and multiple path transport (MPT) for video and image transmission in a multihop mobile radio network. The video and image information is encoded nonhierarchically into multiple descriptions with the following objectives. The received picture quality should be acceptable, even if only one description is received and every additional received description contributes to enhanced picture quality. Typical applications will need a higher bandwidth/higher reliability connection than that provided by a single link in current mobile networks. To support these applications, a mobile node may need to set up and use multiple paths to the desired destination, either simply because of the lack of raw bandwidth on a single channel or because of its poor error characteristics, which reduce its effective throughput. The principal reason for considering such an architecture is to provide high bandwidth and more robust end-to-end connections. We describe a protocol architecture that addresses this need and, with the help of simulations, we demonstrate the feasibility of this system and compare the performance of the MDC-MPT scheme to a system using layered coding and asymmetrical paths for the base and enhancement layers.

Generalized processor sharing (GPS) is an important scheduling discipline because it enables bandwidth sharing with work conservation and traffic isolation properties. While Markov modulated fluid processes (MMFP) capture the dynamics of the sources, the analysis of such sources with a GPS server is difficult because of the large state space. We study a multi-queue GPS system with MMFP classes and propose a scalable, low complexity algorithm for the tail distributions of the logical queues. The effective bandwidth of the classes and a simple connection admission control (CAC) scheme are derived. Numerical results illustrate the efficiency and accuracy of the technique. The application to an example system of classes consisting of voice and variable bit rate (VBR) video traffic is included.

We introduce a statistical method to analyze multiplexing of multiple sessions sharing link bandwidth using generalized processor sharing (GPS) scheduling. Our method is shown to substantially improve previous upper bounds for GPS scheduling of Markov modulated fluid processes (MMFP) sessions, especially as the number of sessions increases. Application of analytical results to admission control indicate that by sharing bandwidth using GPS among traffic classes there are significant gains over systems that statically segregate the link bandwidth. This effect is quantified in several experiments where various combinations of source types are used. The gains are pronounced when bursty sources with stricter QoS requirements are used.

We develop an analytical fluid model for TCP Reno based on delay differential equations. Using this model, we have developed a prototype for a fluid based simulation tool for TCP. We empirically validate this model by matching the window, throughput and queue evolution curves obtained from the model to that offered by a simulation output. We claim that this analytical model provides the basis for an analyzer that can complement simulators that perform a packet by packet simulation.

Enabling video transmission over ad-hoc networks is more challenging than over conventional mobile networks because a connection path in an ad-hoc network is highly error-prone and the path can go down frequently. On the other hand, it is possible to establish multiple paths between a source and a destination, which provides an extra degree of freedom in coding algorithm design. This paper presents a feedback-based reference picture selection scheme for video transmission over ad-hoc networks. Encoded video streams are transmitted over multiple paths and the reference frames for motion compensated prediction are selected according to the feedback information about the paths' condition. Simulations under the two paths scenario have shown significant improvement over two standard techniques, layered coding and video redundancy coding, which do not use feedback. A novel statistical model for the ad-hoc multi-path environment is also proposed and used in our simulation of transmission loss.

The dual round-robin matching (DRRM) switch has a scalable, low complexity architecture which allows for an aggregate bandwidth exceeding 1 Tb/s using current CMOS technology. In this paper we prove that the DRRM switch can achieve 100% throughput under i.i.d. and uniform traffic. The DRRM is the first practical matching scheme for which this property has been proved. The performance of the DRRM switch is then studied and compared with the iSLIP switch. The delay performance under uniform traffic and the hot-spot throughput of DRRM is better than that of iSLIP, while the throughput of iSLIP under some nonuniform traffic scenarios is slightly higher than that of DRRM. Since throughput drops below 100%, under nonuniform traffic, we also examine some variations of the DRRM matching scheme for nonuniform traffic.

The increase in the bandwidth of the wireless channels and the computing power of the mobile devices makes it possible to offer video service for wireless networks in the near future. In an ad-hoc network, strong error protection is required because of the lack of a fixed infrastructure. On the other hand, the mesh structure of an ad-hoc network implies that there may be multiple paths existing between a source and destination, which can be used to enhance video transmissions. We propose a simple but robust scheme for reliable transmission of video in bandwidth limited ad-hoc networks. In our scheme, a video stream is layer coded. The base layer (BL) packets and the enhancement layer (EL) packets are transmitted separately on two disjoint paths using multipath transport (MPT). BL packets are protected by automatic repeat request (ARQ), and a lost BL packet is retransmitted through the path where EL packets are transmitted. An EL packet has lower priority than a retransmitted BL packet and may be dropped at the sender when congestion occurs. Simulation results show that this scheme can guarantee a graceful video quality in adverse channel conditions. It is effective for video transmission over the high loss environment found in ad-hoc networks.

Abstract In this paper, we consider a problem in networks with storage servers for providing multimedia service. The design involves assigning communication link capacity, sizing the multimedia servers, and distributing different types of content at each server, while guaranteeing an upper limit on the individual end‐to‐end blocking probability. We consider alternative methods for obtaining the end‐to‐end blocking probability with low computation time and present optimization procedures to obtain an optimal solution. Under a linear cost structure, our numerical investigations consider different scenarios that might be helpful in understanding how to distribute multimedia content for a cost‐optimized solution. © 2001 John Wiley & Sons, Inc.

This paper investigates the need for multiple path transport (MPT) of video and image information in a multihop mobile radio network. The video and image information is encoded non-hierarchically into a multiple description coding (MDC) with the following objectives. The received picture quality should be acceptable even if only one description is received and every additional received description contributes to enhanced picture quality. Typical applications will need a higher bandwidth/higher reliability connection than that provided by current mobile networks. For supporting these applications a mobile node may need to set up and use multiple paths to the desired destination either simply because of the lack of raw bandwidth on a single channel or because of its poor error characteristics which reduces its effective throughput. In the context of this work, the principal reasons describe a protocol architecture that addresses this need and with the help of simulations, we demonstrate the feasibility of this system and compare the performance of the MDC-MPT scheme to that of a hierarchical coding scheme.

We consider an optimization problem in networks with storage servers for providing multimedia service. The design involves assigning communication link capacity, sizing the multimedia servers and distributing different types of content at each server, while guaranteeing an upper limit on the individual end-to-end blocking probability. We present optimization algorithms to obtain an optimal solution. Under a linear cost structure, our numerical investigations consider different scenarios that might be helpful in understanding how to distribute multimedia content for a cost-optimized solution.

We investigate the performance of the early packet discard (EPD) and the early selective packet discard (ESPD) scheme that we have previously proposed. In an effort to reduce the complexity while maintaining a good performance, we pick two special cases of the ESPD and compare their performance with that of the EPD. For performance evaluation purposes, the effective throughput and fairness index of these schemes were determined through a simulation study. We observed that the ESPD scheme improves the effective throughput over the EPD by up to 16% with our network model. We also found that ESPD is more effective in alleviating the TCP's unfairness among connections which have different roundtrip times. A major factor causing throughput degradation of the EPD was determined to be the synchronization of TCP windows.

This paper motivates the need for the multiple path transport (MPT) of information in a multihop mobile radio network for supporting non-real time applications. Typical applications will need a higher bandwidth/higher reliability connection than that provided by current mobile networks. For supporting these applications a mobile node may need to set up and use multiple paths to the desired destination, either simply because of the lack of raw bandwidth on a single channel or because of its poor error characteristics, which reduces its effective throughput. In the context of this work, the principal reasons for considering such an architecture are providing high bandwidth and a more robust end-to-end connection. We describe a protocol architecture that addresses this need and, with the help of simple simulation models, we show that the delay and throughput performance of multiple path schemes is significantly better than that of a conventional scheme in which one session makes use of one path. For the data applications considered in this work, there is an additional advantage of security because tapping any one path does not give access to the complete information.

This paper presents a novel generic weight-based network bandwidth sharing policy and an available bit rate (ABR) algorithm that achieves this policy. Our policy supports the minimum cell rate (MCR) requirement and peak cell rate (PCR) constraint of each connection and allocates network bandwidth among all connections based on a weight associated with each connection. To achieve this policy for ABR connections, we design an ABR algorithm which employs per virtual connection (VC) accounting to keep track of the state information of each VC. Our ABR algorithm is proven to provide guaranteed convergence to our generic weight-based rate allocation policy under any network configuration and any set of link distances. Simulation results show that our ABR algorithm has a fast convergence property.

We generalize the classical max-min rate allocation policy with the support of the minimum rate requirement and peak rate constraint for each connection. Since a centralized algorithm for the generalized max-min (GMM) rate allocation requires global information, which is difficult to maintain and manage in a large network, we develop a distributed protocol to achieve the GMM policy using the available bit rate (ABR) flow control mechanism. We give a proof that our distributed protocol converges to the GMM rate allocation through distributed and asynchronous iterations under any network configuration and any set of link distances.

A novel concept in available bit rate (ABR) service model as defined by the ATM Forum is the minimum cell rate (MCR) bandwidth guarantee for each connection. In this paper, we present a network bandwidth allocation policy to support each ABR connection's MCR requirement, as well as its peak cell rate (PCR) constraint. Furthermore, we develop two explicit-rate (ER) based ABR algorithms consistent with the ATM Forum ABR traffic management framework to achieve this rate allocation policy. The first ABR implementation is a simple heuristic algorithm which does not require per-VC accounting. It requires minimal implementation complexity and offers satisfactory performance in a LAN environment. The second ABR implementation employs per-VC accounting and is proven to converge to our rate allocation policy for any network topology and any set of link distances.

We investigate packet discarding schemes for TCP over ATM with UBR service. In doing so, we tested the effective throughput of two existing schemes, Partial Packet Discard (PPD) and Early Packet Discard (EPD), as compared to the Random Cell Discard (RCD) scheme which discards any incoming cells after buffer overflow. We observed that PPD alleviates the effect of packet fragmentation so that it gets effective throughput enhancement over RCD, and EPD provides further enhancement over PPD. After closer investigation, we found that there is a sustained congestion problem other than packet fragmentation that causes the effective throughput to be degraded. We noted that sustained congestion resulted in the synchronization of TCP window expansion and shrinkage. To provide a solution for this problem, we propose the Early Selective Packet Discard (ESPD) policy, a strategy which makes sessions take turns in accessing network capacity by discarding packets from selected sessions rather than randomly. Our results shows that ESPD achieves throughput and fairness enhancement over EPD with only a modest increase in implementation complexity.

The authors analyze the effect of fixed delay in conjunction with queueing and resequencing delay on the optimal distribution of traffic on multiple disjoint paths. They study a system of two hosts or end nodes, connected by a high speed network communicating on two virtual channels which follow disjoint physical paths. The paths have a different number of hops and/or physical length which leads to a different amount of constant delay for each of them. The variable delay on each path is modelled by a queue with exponential service. Furthermore the destination node delivers packets an the order they arrived at the source node, which entails additional resequencing delay. They find the optimal split of traffic, so as to minimize the total average system time (including the resequencing delay). The results show that the optimal splitting probability may be heavily dependant on the difference in the fixed delays on the two paths. Numerical examples are presented to illustrate the effect of fixed delay on the fraction of traffic routed to different paths. Performance can be further improved when they do a deterministic split of the traffic.>

Study the problem of optimal buffer space priority control in an ATM network node. The buffer of a transmission link is shared among the cells of several traffic classes waiting for transmission through the link. When the number of cells to be stored in the buffer exceeds the available buffer space, certain cells have to be dropped. Different traffic classes have different sensitivities to cell losses. By appropriately selecting the classes of cells which are dropped or blocked in case of overflow, one can have the more sensitive classes suffering smaller cell losses. Depending on the control that on the system, three classes of policies are distinguished. In each one, policies that schedule the buffer allocation in some optimal manner are identified.>

The problem of optimal buffer space priority control in an asynchronous transfer mode (ATM) network node is studied. The buffer of a transmission link is shared among the cells of several traffic classes waiting for transmission through the link. When the number of cells to be stored in the buffer exceed the available buffer space, certain cells have to be dropped. Different traffic classes have different sensitivities to cell losses. By appropriate selection of the classes of cells that are dropped in case of overflow, the more sensitive classes can be made to suffer smaller cell losses. Arriving cells might be blocked from entering the system or they may be dropped after they are already in the buffer. Depending on the control that is on the system, three classes of policies are distinguished. In each one, policies that schedule the buffer allocation in some optimal manner are identified.>

The authors describe a methodology for designing interconnected LAN/MAN networks with the objective of minimizing the average network delay. They consider IEEE 802 standard LANs interconnected by transparent bridges. These bridges are required to form a spanning tree topology. The authors propose a simulated annealing-based algorithm for designing minimum delay spanning tree topologies. In order to measure the quality of the solutions, a lower bound for the average network delay is found. The algorithm is extended to design the overall LAN/MAN topology consisting of a MAN or high-speed data service interconnecting several clusters of bridged LANs. Comparison with the lower bound and several other measures show that the solutions are not very far from the global minimum.>

Collision resolution algorithms (CRAs) for the ternary feedback multiple access channel with time constraints are considered. The authors wish to maximize the number of packets successfully transmitted within a fixed deadline K after their arrival for transmission. Packet arrivals are assumed to be Poisson. A nonnested CRA is described and its performance is compared with a nested CRA for values of K>

The authors describe a methodology for designing interconnected local area network/metropolitan area network (LAN-MAN) networks with the objective of minimizing the average network delay. They consider IEEE 802.3-5 LANs interconnected by transparent bridges. These bridges are required to form a spanning tree topology. The optimization algorithm for finding a minimum delay spanning tree topology is based on simulated annealing. In order to measure the quality of the solutions, a lower bound for the average network delay is found. The comparison of results with this lower bound and several other goodness measures shows that the solutions are not very far from the global minimum. The authors extend the present algorithm for finding minimum delay LAN-MAN topologies consisting of fiber distributed data interface (FDDI) MANs or switched multi-megabit data service (SMDS) interconnecting several clusters of bridged LANs.>

A method of flow control that requires very few buffers to be allocated at each node to virtual circuits (or sessions) that have to transverse many links is described. Transmissions are scheduled using the golden ratio policy of A. Itai and Z. Rosberg (1984). It is shown that the buffer requirements of a session grow at most logarithmically with the number of slots allotted to it. As an immediate consequence, intra-network delays are bounded.>

The problem of scheduling impatient customers in a non-preemptive G/GI/1 queue is considered. Every customer has a random deadline to the beginning of its service. Given the distribution of the customer deadlines (rather than their exact values), a scheduling policy decides the customer service order and also which customer(s) to reject. The objective is to find an optimal policy which maximizes the number of customers served before their deadlines. It is shown that LIFO (last-in first-out) is an optimal service order when the deadlines are i.i.d. (independently identically distributed) random variables with a concave cumulative distribution function. There is an optimal policy in the LIFO-TO (time-out) class, as defined by the authors. For the M/GI/1 queue, it is proved that unforced idle times are not allowed under this optimal policy. It is also shown that the optimal LIFO-TO policy assigns a fixed critical time (i.e., its maximum waiting time) to every customer. When the customer waiting times are unknown, the optimal policy for an M/M/1 queue becomes the LIFO-PO (push-out) policy, with a fixed buffer size used as a rejection threshold.>

The static and centralized routing of voice and data traffic in burst switched networks is addressed. It is assumed that the routing allows random bifurcation in voice and data paths and preemptive priorities for voice requirements. A study is made of routing of voice only, by using a multicommodity flow model with linearized link losses and average network loss as a minimization objective. Solving the resulting linear program, it is observed that optimal routing strategies prefer to freeze a requirement at an early stage of its path rather than those requirements that are close to their destinations. A study is made of the voice-data interaction at the link level using an available fluid-flow model, and the combined link performance is translated as a maximum flow constraint on a link. This constraint may have undesirable effects on the voice, such as introducing routes with flow absorbing loops, and unfair freezing of some requirements. All conflicting multiple objectives and constraints are included in a linear programming formulation, and it is shown how parameters can be tuned to produce desirable voice and data paths.>

A model of a packet radio network in which transmitters with range R are distributed according to a two-dimensional Poisson point process with density D is examined. To ensure network connectivity, it is shown that pi R/sup 2/D, the expected number of nearest neighbors of a transmitter, must grow logarithmically with the area of the network. For an infinite area there exists an infinite connected component with nonzero probability if pi R/sup 2/D>N/sub 0/, for some critical value N/sub 0/. It is shown that 2.195>

Many problems can be modeled as single-server queues with impatient customers. An example is that of the transmission of voice packets over a packet-switched network. If the voice packets do not reach their destination within a certain time interval of their transmission, they are useless to the receiver and considered lost. It is therefore desirable to schedule the customers such that the fraction of customers served within their respective deadlines is maximized. For this measure of performance, it is shown that the shortest time to extinction (STE) policy is optimal for a class of continuous and discrete time nonpreemptive M/G/1 queues that do not allow unforced idle times. When unforced idle times are allowed, the best policies belong to the class of shortest time to extinction with inserted idle time (STEI) policies. An STEI policy requires that the customer closest to his or her deadline be scheduled whenever it schedules a customer. It also has the choice of inserting idle times while the queue is nonempty. It is also shown that the STE policy is optimal for the discrete time G/D/1 queue where all customers receive one unit of service. The paper concludes with a comparison of the expected customer loss using an STE policy with that of the first-come, first-served (FCFS) scheduling policy for one specific queue.

It is shown that, for a ternary feedback random access channel with a Poisson arrival process,0.5is an upper bound to the throughput for all "degenerate intersection" algorithms (DIA's) and first-come first-served algorithms (FCFSA's). As a by-product, the nested FCFSA with the largest throughput is found for the random access channel with a Bernoulli arrival process with parameterp. Forp \geq 0.018, this algorithm has the highest throughput over all DIA's and FCFSA's. Lastly, it is shown that. for some values ofp, a non-DIA, non-FCFSA has a higher throughput than the optimum DIA or FCFSA.