This demo presents Edge-DC, a decentralized edge platform that enables resource sharing across autonomous Kubernetes clusters without a global coordinator. Each cluster hosts a lightweight Domain Controller (DC) that summarizes local workload and network conditions, exchanges brief updates with peers, and ranks service endpoints for client requests. The demonstration illustrates how minimal inter-cluster coordination has potential to improve responsiveness and stability under variable load while preserving local autonomy. By emphasizing explainable, local-first decision making, Edge-DC provides a practical starting point for cooperative edge computing.

Dynamic Spectrum Access (DSA) is a key mechanism for meeting the ever-increasing demand for emerging wireless services. DSA involves managing and assigning available spectrum resources in a way that minimizes interference and allows RF coexistence between heterogeneous devices and systems. Such co-existence mechanisms, if they are to succeed when heterogeneous RF devices managed by different entities need to operate in a given area and frequency band (licensed and/or unlicensed), require a common mechanism for expressing the boundaries of spectrum use of each device so that spectrum use deconfliction methods can be built and verified. Spectrum Consumption Models (SCMs) – defined in the IEEE 1900.5.2 standard – offer a mechanism for RF devices to: (i) declare the characteristics of their intended spectrum use and their interference protection needs; and (ii) determine compatibility (non-interference) with existing devices. In this paper, we propose a novel SCM-based Spectrum Deconfliction (SD) algorithm that dynamically configures RF operational parameters (e.g., center frequency and transmission power) of a target transmitter-receiver pair aiming to minimize interference with existing devices/systems. We also propose sequential and distributed DSA methods that use the SD algorithm for assigning spectrum in large-scale networks. To evaluate the performance of our methods in terms of computation time, spectrum assignment efficiency, and overhead, we use two custom-made simulation platforms. Finally, to experimentally demonstrate the feasibility of our methods, we build a proof-of-concept implementation in the NSF PAWR COSMOS wireless testbed. The results reveal the advantages of using SCMs and their capabilities to conduct spectrum assignments in dynamic and congested communication environments.

Private 5G networks will soon be ubiquitous across the future-generation smart wireless access infrastructures hosting a wide range of performance-critical applications. A high-performing User Plane Function (UPF) in the data plane is critical to achieving such stringent performance goals, as it governs fast packet processing and supports several key control-plane operations. Based on a private 5G prototype imple-mentation and analysis, it is imperative to perform dynamic resource management and orchestration at the UPF. This paper leverages Mobile Edge Cloud-Intelligent Agent (MEC-IA), a logically centralized entity that proactively distributes resources at UPF for various service types, significantly reducing the tail latency experienced by the user requests while maximizing resource utilization. Extending the MEC-IA functionality to MEC layers further incurs data plane latency reduction. Based on our extensive simulations, under skewed uRLLC traffic arrival, the MEC-IA assisted bestfit UPF-MEC scheme reduces the worst-case latency of UE requests by up to 77.8% w.r.t. baseline. Additionally, the system can increase uRLLC connectivity gain by 2.40× while obtaining 40% CapEx savings.

This paper presents mmCPTP, a cross-layer end-toend protocol for fast delivery of data over mmWave channels associated with emerging 5G services. Recent measurement studies of mmWave channels in urban micro cellular deployments show considerable fluctuation in received signal strength along with intermittent outages resulting from user mobility. This results in significant impairment of end-to-end data transfer throughput when regular TCP is used to transport data over such mmWave channels. To address this issue, we propose mmCPTP, a novel cross-layer end-to-end data transfer protocol that sets up a transport plug-in at or near the base station and uses feedback from the lower layer (RLC/MAC) to opportunistically pull data at the mobile client without the slow start and probing delays associated with TCP. The system model and end-to-end protocol architecture are described and compared with TCP and IndirectTCP (I-TCP) in terms of achievable data rate. The proposed mmCPTP protocol is evaluated using NS3 simulation for 5G NR (New Radio) considering a high-speed mobile user scenario. The system is further validated using a proof-of-concept prototype which emulates the high-speed mmWave/NR access link with traffic shaping over Gbps ethernet. Results show significant performance gains for mmCPTP over TCP and I-TCP (2.5x to 17.2x, depending on the version).

Next generation wireless services and applications, including Augmented Reality, Internet-of-Things, and Smart-Cities, will increasingly rely on Dynamic Spectrum Access (DSA) methods that can manage spectrum resources rapidly and efficiently. Advances in regulatory policies, standardization, networking, and wireless technology are enabling DSA methods on a more granular basis in terms of time, frequency, and geographical location which are key for the operation of 5G and beyond-5G networks. In this context, this paper proposes a novel DSA algorithm that leverages IEEE 1900.5.2 Spectrum Consumption Models (SCMs) which offer a mechanism for RF devices to: (i) "announce" or "declare" their intention to use the spectrum and their needs in terms of interference protection; and (ii) determine compatibility (i.e., non-interference) with existing devices. In this paper, we develop an SCM-based DSA algorithm for spectrum deconfliction in large-scale wireless network environments and evaluate this algorithm in terms of computation time, efficiency of spectrum allocation, and number of device reconfigurations due to interference using a custom simulation platform. The results demonstrate the benefits of using SCMs and their capabilities to perform fine grained spectrum assignments in dynamic and dense communication environments.

As mobile devices continuously generate streams of images and videos, a new class of mobile deep vision applications are rapidly emerging, which usually involve running deep neural networks on these multimedia data in real-time. To support such applications, having mobile devices offload the computation, especially the neural network inference, to edge clouds has proved effective. Existing solutions often assume there exists a dedicated and powerful server, to which the entire inference can be offloaded. In reality, however, we may not be able to find such a server but need to make do with less powerful ones. To address these more practical situations, we propose to partition the video frame and offload the partial inference tasks to multiple servers for parallel processing. This paper presents the design of Elf, a framework to accelerate the mobile deep vision applications with any server provisioning through the parallel offloading. Elf employs a recurrent region proposal prediction algorithm, a region proposal centric frame partitioning, and a resource-aware multi-offloading scheme. We implement and evaluate Elf upon Linux and Android platforms using four commercial mobile devices and three deep vision applications with ten state-of-the-art models. The comprehensive experiments show that Elf can speed up the applications by 4.85× with saving bandwidth usage by 52.6%, while with <1% application accuracy sacrifice.

This paper describes a novel distributed mobility management (DMM) scheme for the "named-object" information centric network (ICN) architecture in which the routers forward data based on unique identifiers which are dynamically mapped to the current network addresses of a device. The work proposes and evaluates two specific handover schemes namely, hard handoff with rebinding and soft handoff with multihoming intended to provide seamless data transfer with improved throughput during handovers. The evaluation of the proposed handover schemes using system simulation along with proof-of-concept implementation in ORBIT testbed is described. The proposed handoff and scheduling throughput gains are 12.5% and 44% respectively over multiple interfaces when compared to traditional IP network with equal share split scheme. The handover performance with respect to RTT and throughput demonstrate the benefits of clean slate network architecture for beyond 5G networks.

This paper presents a novel control plane protocol designed to enable cooperative resource sharing in heterogeneous edge cloud scenarios. While edge clouds offer the advantage of potentially lower latency for time critical applications, computing load generated by mobile users at the network edge can be very bursty as compared with aggregated traffic served by a data center. This motivates the design of a shared control plane which enables dynamic resource sharing between edge clouds in a region. The proposed control plane is designed to exchange key compute and network parameters (such as CPU GIPS, % utilization, and network bandwidth) needed for cooperation between heterogeneous edge clouds across network domains. The protocol thus enables sharing mechanisms such as dynamic resource assignment, compute offloading, load balancing, multi-node orchestration, and service migration. A specific distributed control plane (DISCO) based on overlay neighbor distribution with hop-count limit is described and evaluated in terms of control overhead and performance using an experimental proto-type running on the ORBIT radio grid testbed. The prototype system implements a heterogeneous network with 18 autonomous systems each with a compute cluster that participates in the control plane protocol and executes specified resource sharing algorithms. Experimental results are given comparing the performance of the baseline with no cooperation to that of cooperative algorithms for compute offloading, cluster computing and service chaining. An application level evaluation of latency vs. offered load is also carried out for an example time-critical application (image analysis for traffic lane detection). The results show significant performance gains (as much as 45% for the cluster computing example) vs. the no cooperation baseline in each case at the cost of relatively modest complexity and overhead.

This paper focuses on COSMOS - Cloud enhanced Open Software defined MObile wireless testbed for city-Scale deployment. The COSMOS testbed is being deployed in West Harlem (New York City) as part of the NSF Platforms for Advanced Wireless Research (PAWR) program. It will enable researchers to explore the technology "sweet spot" of ultra-high bandwidth and ultra-low latency in the most demanding real-world environment. We describe the testbed's architecture, the design and deployment challenges, and the experience gained during the design and pilot deployment. Specifically, we describe COSMOS' computing and network architectures, the critical building blocks, and its programmability at different layers. The building blocks include software-defined radios, 28 GHz millimeter-wave phased array modules, optical transport network, core and edge cloud, and control and management software. We describe COSMOS' deployment phases in a dense urban environment, the research areas that could be studied in the testbed, and specific example experiments. Finally, we discuss our experience with using COSMOS as an educational tool.

The Cloud-Enhanced Open Software Defined Mobile Wireless Testbed for City-Scale Deployment (COSMOS) platform is a programmable city-scale shared multi-user advanced wireless testbed that is being deployed in West Harlem of New York City [1]. To keep pace with the significantly increased wireless link bandwidth and to effectively integrate the emerging C-RANs, COSMOS is designed to incorporate a fast programmable core network for providing connections across different computing layers. A key feature of COSMOS is its dark fiber based optical x-haul network that enables both highly flexible, user defined network topologies and experimentation directly in the optical physical layer. The optical architecture of COSMOS was presented in [2]. In this abstract, we present the tools and services designed to configure and monitor the performance of optical paths and topologies of the COSMOS testbed. In particular, we present the SDN framework that allows testbed users to implement experiments with application-driven control of optical and data networking functionalities.

Running computer vision algorithms on images or videos collected by mobile devices represent a new class of latency-sensitive applications that expect to benefit from edge cloud computing. These applications often demand real-time responses (e.g., <;100 ms), which can not be satisfied by traditional cloud computing. However, the edge cloud architecture is inherently distributed and heterogeneous, requiring new approaches to resource allocation and orchestration. This paper presents the design and evaluation of a latency-aware edge computing platform, aiming to minimize the end-to-end latency for edge applications. The proposed platform is built on Apache Storm, and consists of multiple edge servers with heterogeneous computation (including both GPUs and CPUs) and networking resources. Central to our platform is an orchestration framework that breaks down an edge application into Storm tasks as defined by a directed acyclic graph (DAG) and then maps these tasks onto heterogeneous edge servers for efficient execution. An experimental proof-of-concept testbed is used to demonstrate that the proposed platform can indeed achieve low end-to-end latency: considering a real-time 3D scene reconstruction application, it is shown that the testbed can support up to 30 concurrent streams with an average perframe latency of 32ms, and can achieve 40% latency reduction relative to the baseline Storm scheduling approach.

As the deployment of software-defined networks increases, so does the manageability of local and wide area networks. Designing intelligent solutions that respond to traffic changes automatically will soon become a mandatory requirement in production networks. In this paper, we focus on designing an intelligent control plane for the MobilityFirst Future Internet architecture. This architecture proposes novel mechanisms to replace the Internet Protocol to better support content delivery and mobility, such as hop-by-hop transfer, storage-aware routing and separation of identifiers and network addresses. In earlier work, we have argued that these mechanisms can be bypassed for certain data flows. Indeed, when there is no mobility involved, it is more convenient to implement cut-through switching at lower layers to bypass the routing mechanisms. In this paper, we propose an inter-domain framework capable of cut-through switching in MobilityFirst. The proposed framework is capable of adding and removing flows from tunnels automatically. It is also capable of creating inter-domain tunnels based on flow behavior and inter-domain latency. Our implementation experiments show that the control plane delay can be reduced by 75% when using inter-domain tunnels. Furthermore, the results also show how our framework needs fewer messages than current protocols such as label distribution protocol to setup intra-domain and inter-domain tunnels.

This work describes a clean-slate inter-domain routing protocol designed to meet the needs of the future mobile Internet. In particular, we describe the edge-aware inter-domain routing (EIR) protocol which provides new abstractions of aggregated-nodes (aNodes) and virtual-links (vLinks) for expressing network topologies and edge network properties necessary to address next-generation mobility related routing scenarios which are inadequately supported by the border gateway protocol (BGP) in use today. Specific use-cases addressed by EIR include emerging mobility service scenarios such as multi-homing across WiFi and cellular, multipath routing over several access networks, and anycast access from mobile devices to replicated cloud services. Simulation results for protocol overhead are presented for a global-scale Caida topology, leading to an identification of parameters necessary to obtain a good balance between overhead and routing table convergence time. A Click-based proof-of-concept implementation of EIR on the ORBIT testbed is described and used to validate performance and functionality for selected mobility use-cases, including mobile data services with open WiFi access points and mobile platforms such as buses operating in an urban area.

This work introduces a new model for cloud gaming systems aimed at optimizing the timeliness of video frames based on an age of information (AoI) metric. Mobile clients submit actions through an access network to a game server. The game server generates video frames at a constant frame rate. At the mobile device, the display of these frames represent game status updates. We develop a Markov model to characterize the frame delivery process in low-latency edge cloud gaming systems. Based on this model, we derive a simple formula for the average status age of a tightly synchronized low-latency mobile gaming system in which the inter-frame period is a significant contributor to the system latency. We validate the model by ns-3 simulation of a low-latency edge cloud gaming system. Our evaluation scenarios included single-player games as well as multi-player games in which the game processing was conducted by a combination of a centralized game server and edge cloud renderers.

In the recent years it has become increasingly evident that the current end-to-end host-centric communication paradigm will not be capable of meeting the ongoing demand for massive data rates and ultra-low latency. With the advent of fifth generation of cellular architecture (5G) to support these requirements on the wireless edge of the network, the need for core network solutions to play a complementary role is conspicuous. In this paper we present and tackle some of the challenges of deploying a Future Internet Architecture (FIA), called MobilityFirst (MF), specifically for 5G use case scenarios. We report our findings of the deployment based on a setup on a small- scale testbed (ORBIT) and a nation-wide distributed testbed (GENI), and illustrate some results for the use case of device multihoming, in comparison with current TCP/IP based solution, i.e. Multipath TCP.

The introduction of Dynamic Adaptive Streaming over HTTP (DASH) helped reduce the consumption of resources in video delivery, but its client-based rate adaptation is unable to optimally use the available end-to-end network bandwidth. We consider the problem of optimizing the delivery of video content to mobile clients while meeting the constraints imposed by the available network resources. Observing the bandwidth available in the network's two main components, core network, transferring the video from the servers to edge nodes close to the client, and the edge network, which is in charge of transferring the content to the user via wireless links, we aim to find an optimal solution by exploiting the predictability of future user requests of sequential video segments, as well as the knowledge of available infrastructural resources at the core and edge wireless networks in a given future time window. Instead of regarding the bottleneck of the end-to-end connection as our throughput, we distribute the traffic load over time and use intermediate nodes between the server and the client for buffering video content to achieve higher throughput, and ultimately significantly improve the Quality of Experience for the end user in comparison with current solutions.

Edge cloud computing moves cloud services to the edge of the network, thereby allowing clients to access services with a significantly reduced network delay. This service migration is intended to enable a range of latency sensitive mobile applications. In this paper, we propose to manage user QoS by actively migrating services to different edge clouds in response to degraded server or network performance. Previous studies have proposed a distance-based Markov Decision Process (MDP) for optimizing migration decisions. These models provide the feasibility of applying MDP to edge cloud service migration decisions. However, these models fail to consider dynamic network and server states in migration decisions. In this work, we address these limitations by designing a comprehensive edge cloud migration decision system, which we call SEGUE. SEGUE achieves optimal migration decisions by providing a long-term optimal QoS to mobile users in the presence of link quality and server load variation. The basis of SEGUE is in its QoS-aware service migration and its state based MDP model which effectively incorporates the two dominant factors in making migration decisions: 1) network state, and 2) server state. An evaluation of SEGUE performance is given through an augmented reality application. Our results demonstrate that SEGUE reduces the response time of this application by 27.21% and 53.70% compared to the lowest load migration model and the least hop migration model, respectively.

This paper presents the design and evaluation of a mobile data service which exploits parallel multi-path transmission over multiple cellular networks to achieve significant performance gains. Multi-network access is motivated by the fact that multi-radio mobile devices are fast becoming a reality, making it possible for end-users to increase their service speed and availability via network diversity. In contrast to previously proposed techniques at the application or transport layers, we propose a novel network assisted architecture for multi-homed (NAMH) cellular access. In the proposed system, network elements such as routers and base stations provide the necessary multihoming functionality including identification of a bifurcation router and dynamic splitting of the data stream corresponding to currently achievable bit-rates on the cellular base stations. A detailed evaluation is provided for the multi-network service with LTE base stations, using both ns3 simulation and trace-driven emulations. The results show that significant gains are achieved with multi-network access with bit-rates of ~1.9x using two LTE networks in parallel. Comparison with the well-known MPTCP method is given, showing gains of about 40% with NAMH for the trace-driven two-network scenario.

This paper presents a novel approach to achieving scalable push multicast services using the distributed global name resolution service associated with emerging name-based network architectures. The proposed named-object multicast (NOMA) scheme employs unique names to identify multicast groups while using the global name resolution service (GNRS) to store the tree structure and maintain current mappings to mobile end-user addresses. The NOMA scheme achieves improved scalability and performance over conventional multicast protocols such as PIM-SM and MDSP by taking advantage of the GNRS to simplify tree management and limit control overhead. Performance evaluation results including comparisons with IP multicast are given using a combination of analysis and NS-3 simulation. The results show good scalability properties along with low control overhead for medium to large multicast groups. In addition, NOMA seamlessly handles mobility for end-hosts subscribed to a group, avoiding data losses upon mobility events. Results further demonstrate how separating names from addresses enables NOMA to dynamically forward traffic to mobile users. In conclusion, we describe a proof-of-concept prototype developed for further experimental validation of the proposed NOMA multicast routing scheme.

Future Internet projects such as MobilityFirst and Named Data Networking have proposed novel mechanisms to replace the Internet Protocol to better support content delivery and mobility. However, the problem of efficient data transfer across the network core has not been adequately investigated. We tackle the challenge of inter-domain cut-through switching using software-defined networking (SDN). First, we propose and solve an optimization problem that minimizes the total transfer time using inter-domain tunnels. Second, we propose an SDN-based routing framework for the MobilityFirst architecture capable of dynamically creating such tunnels. The main novelty of this framework is to name tunnels as network objects to simplify how tunnels are created and maintained. To validate our framework, we implement on the GENI (Global Environment for Network Innovations) testbed a prototype for the MobilityFirst architecture. Our experiments with the optimization problem show that the inter-domain latency between controllers plays a key role on how tunnels are setup. Furthermore, our implementation experiments show that the control plane delay can be reduced by 75% when using inter-domain tunnels. Finally, we show how our framework needs fewer messages than current protocols such as label distribution protocol (LDP) to setup intra-domain and inter-domain tunnels.

MobilityFirst (MF), as a realization of Information Centric Network architecture, places intelligent functionality, such as storage and reliability, inside the network to assist with data delivery. The MF architecture requires effective congestion and flow control to efficiently support data delivery at scale. Traditional end-to-end, window-based congestion control like that used by TCP is unsuitable as it is unable to take advantage of such in-network functionality. We design network-layer assisted congestion control schemes tailored to MF. One approach that works well for hop-by-hop reliable networks is using per-flow queueing and backpressure to alleviate congestion. However, it could become impractical in the presence of a large number of flows, which leads to substantial memory consumption and computational complexity. Building on a more scalable per-interface queueing model, we design congestion control mechanism that embodies traffic source rate control and explicit congestion notification from routers. Sample results show that the proposed scheme is able to achieve similar link utilization and better fairness compared with a per-flow queueing scheme.

In this paper, QoS characteristics such as TCP throughput is investigated for densely deployed mobile wireless LANs (WLANs). Factors affecting throughput characteristics are discussed and evaluated by using real machines such as smartphones and portable APs. In IEEE 802.11 WLANs, a rate adaptation mechanism controls the transmission rate and one of the dominant factors for QoS. In order to understand the behavior of the rate adaptation control, 1 to 18 sets mobile WLANs are examined under different parameters. Since a behavior of the rate adaptation control is a vender specific one and it strongly depends on interference, signal strength and etc., the real terminals such as smartphones were used in the experiments. Performance anomaly drastically reduces the throughput not only in the WLAN which has a terminal with low transmission rate but also in the neighboring WLANs that share the same channel. In order to avoid unnecessary transmission rate degradation by the rate adaptation control, Context Aware multi Rate Control (CARC) is proposed and evaluated. In CARC, Turning the rate adaptation control on/off is controlled according to a context, for example, signal strength. The evaluation results show that the CARC can be cost-effectively implemented and improves the throughput performance of whole WLANs by 3.5 times than that without the application of CARC.

With exponential growth of IoT devices [1], the 5G network will experience a variety of traffic patterns not prevalent in earlier applications. These, often will transmit short sporadic messages, which are not well suited to the connection-oriented modes associated with legacy 3GPP network resulting in high service latency and excessive control overhead. It is acknowledged that current 4G network could be overwhelmed by the surge in both traffic and control plane signaling load. For 5G, it is necessary to redesign the mobile network to provide a low delay, low control overhead IoT mode that will work efficiently for emerging application scenarios. The MAC layer has to be designed in such a way that IoT messages experience low access latency across both the radio access network and core network. The goal is to operate in the same band as current LTE, thus not requiring any separate bandwidth allocation and is backward compatible with the current 4G system. In this paper, we propose a cross-layer MAC and Physical layer solution for low power, low bitrate devices that require low access delay and long range for communication.

Network topology plays a critical role while designing and evaluating network protocols. Most existing topology generators are insufficient to reflect the real world network demands to a topology or to capture the Internet topology evolution such as the "flattening" Internet. They focused on the graph properties of a topology, thus, lacking of ability to model engineering features of the network. Some state-of-art topology generators that consider engineering factors fail to capture trends in both intra-AS and inter-AS connections, which are equally important for evaluating future network protocols. We have developed a topology generator GeoTopo, which is to our best knowledge the first scalable topology generator modeling engineering factors for both intra-AS and inter-AS topology generation. The engineering factors that GeoTopo considers include demographic and geographic features as well as business interests of ASes. We use GeoTopo to create and study three classes of topologies: the topology characterized mainly by graph-properties (Status Quo topology), the topology driven by peering at Internet Exchange Points (IXP topology) and the topology characterized by country backbones (CB topology). The SQ topology follows the degree-based model and serves as a baseline for capturing topology features. The IXP and CB topologies model two major directions of the Internet "flattening". The three classes of topologies enable us to analyze the impact of engineering factors on topology generation such as AS peering policies, IXP deployment and AS geo-settings. GeoTopo's ability to generate projected future Internet topologies make it a valuable tool for the design and evaluation of Future Internet Architectures that is currently under consideration in the research community. We use the evaluation of Global Name Resolution Service (GNRS), a key component shared by name-based network architectures, as an example application to demonstrate GeoTopo's capability to capture the mobility of network entities, the locality of the traffic, and the impact of the evolving network.

This paper investigates Quality of Service (QoS) of the personal mobile wireless LANs (m-WLANs). The situations in the m-WLANs differs from the situations in the normal use of WLANs; the access point (AP) and the associated terminals (TEs) are in proximity. In the m-WLANs, the capture effect (CE) significantly affects on the throughput performance. To measure the impact on the QoS by the CE, the experimental study considering the interference from other power sources (APs and TEs) are required. However, since it is difficult to understand the detailed relationships between the QoS factors, the analytical calculations were also performed. With the experimental and analytical results, we demonstrated that the auto rate fallback algorithm of WLAN causes degradation of the QoS performance. We propose two transmission rates controlling schemes to improve the QoS performance.

Traffic from mobile wireless networks has been growing at a fast pace in recent years and is expected to surpass wired traffic very soon. Service providers face significant challenges at such scales including providing seamless mobility, efficient data delivery, security, and provisioning capacity at the wireless edge. In the Mobility First project, we have been exploring clean slate enhancements to the network protocols that can inherently provide support for at-scale mobility and trustworthiness in the Internet. An extensible data plane using pluggable compute-layer services is a key component of this architecture. We believe these extensions can be used to implement in-network services to enhance mobile end-user experience by either off-loading work and/or traffic from mobile devices, or by enabling en-route service-adaptation through context-awareness (e.g., Knowing contemporary access bandwidth). In this work we present details of the architectural support for in-network services within Mobility First, and propose protocol and service-API extensions to flexibly address these pluggable services from end-points. As a demonstrative example, we implement an in network service that does rate adaptation when delivering video streams to mobile devices that experience variable connection quality. We present details of our deployment and evaluation of the non-IP protocols along with compute-layer extensions on the GENI test bed, where we used a set of programmable nodes across 7 distributed sites to configure a Mobility First network with hosts, routers, and in-network compute services.

With the increasing interest in connected vehicles, it is useful to evaluate the capability of delivering large content over a WiFi infrastructure to vehicles. The throughput achieved over WiFi channels can be highly variable and also rapidly degrades as the distance from the access point increases. While this behavior is well understood at the data link layer, the interactions across the various protocol layers (data link and up through the transport layer) and the effect of mobility may reduce the amount of content transferred to the vehicle, as it travels along the roadway. This paper examines the throughput achieved at the TCP layer over a carefully designed outdoor WiFi environment and the interactions across the layers that impact the performance achieved, as a function of the receiver mobility. The experimental studies conducted reveal that impairments over the WiFi link (frame loss, ARQ and increased delay) and the residual loss seen by TCP causes a cascade of duplicate ACKs to be generated. This triggers large congestion window reductions at the sender, leading to a drastic degradation of throughput to the vehicular client. To ensure outdoor WiFi infrastructures have the potential to sustain reasonable downlink throughput for drive-by vehicles, we speculate that there is a need to adapt how WiFi and TCP (as well as mobility protocols) function for such vehicular applications.

The prevalence of mobile devices has attracted research on mobile content delivery techniques. The MobilityFirst(MF) project, discussed in this paper, proposes a clean-slate Internet architecture that enables mobile content-oriented operations at the network level. We describe the design details of the architecture in realizing this and provides a preliminary evaluation on scalability and performance.

This paper presents a technique for enabling multihoming in the emerging heterogeneous (“hetnet”) mobile wireless access scenarios, where mobile devices have dual wireless interfaces (such as Wi-Fi and LTE) and can use either or both to achieve significant improvements in performance and service quality. A novel network-assisted technique for multihoming is introduced, enabled by the globally unique identifier (GUID) based routing in the proposed MobilityFirst Future Internet architecture, now under development. In particular, the approach shifts the burden of policy expression and data-striping from end-nodes to in-network nodes, and utilizes named object routing with GUIDs to establish multiple paths to destination mobile devices. The proposed multihoming technique uses hop-by-hop backpressure for data striping at the bifurcation router and includes a robust mechanism to reduce reordering of packets at the receive buffer. We quantify the performance gains using detailed NS3 based simulations and present results from a thorough parametric study to determine the effects of datarate, delay and hop-count difference between multiple available paths. We also show that when multiple interfaces are available, simultaneous use of both the interfaces is beneficial only under certain conditions depending on the ratio of the data-rate of the interfaces and the size of the flow.

To develop unified IoT platforms where objects can be made accessible to applications across organizations and domains, popular solutions are based on client-server overlays on today's Internet. These solutions, however, inherit the inefficiencies of the current Internet - especially in terms of mobility, scalability, and communication reliability. To address this problem, we propose to build the unified IoT platform leveraging the salient feats of Information-Centric Network (ICN) architectures, which we call ICN-IoT. Specifically, we explore two ICN architectures - MobilityFirst and NDN - to support IoT, and refer to them as MF-IoT and NDN-IoT, respectively. Through detailed simulations, we find that though these two architectures fare comparably, MF-IoT incurs lower control overheads.

Recent technology advances are poised to enable low-cost, low-power communications in the 7 GHz of unlicensed spectrum at 60 GHz millimeter wave (mmW) frequencies. In 60 GHz networks, transmitters and receivers employ directional antennas and point their main beams toward each other to overcome high propagation losses and achieve high data rates. However, CSMA based directional MAC (DMAC) protocols suffer from the "deafness" problem which causes unfairness and low channel utilization. This paper examines the deafness problem from a new perspective and shows that unfairness and low channel utilization are caused by the exponential backoff mechanism. We propose an enhanced DMAC (EDMAC) protocol that does not use an exponential backoff mechanism, instead employing a low control overhead protocol that enables receivers to adaptively tune senders' contention window sizes. NS-2 simulation results are given to demonstrate that EDMAC compares favorably to DMAC, achieving similar capacity and lower delay jitter in single hop networks, and significantly higher capacity in multi-hop ad hoc network scenarios.

This paper presents a study of mobile wireless LAN (WLAN) hotspots which are used to provide cellular-WiFi tethering service to personal devices. A dense deployment scenario for fixed and mobile WLAN is described and potential performance problems due to interference are identified. An analytical model for coexisting fixed and mobile WLAN hotspots with heterogeneous traffic is presented. The model is used to evaluate the performance of a mobile WLAN as it transits through a set of densely deployed fixed access points (APs), and performance problems due to lack of frequency coordination are identified. An adaptive channel assignment (ACA) scheme for improving mobile AP performance is proposed and evaluated. It is shown that significant performance gains can be achieved with ACA with maximum absolute and percentage throughput gains up to 1.24 Mbps and 42.8% respectively. We also show that setting the scanning interval in ACA requires consideration of the speed at which the mobile WLAN is moving in order to compensate for the throughput losses during channel scanning.

Vehicular networking, both vehicle-to-vehicle (V2V) and vehicle-to-infrastructure (V2I), is an increasingly important usage scenario for future mobile Internet services. Radio technologies such as 3G/4G and WAVE/802.11p now enable vehicles to communicate with each other and connect to the Internet, but there is still the lack of a unifying network protocol architecture for delivery of services across both V2V and V2I modes. The MobilityFirst future Internet architecture, discussed in this paper, is a clean-slate protocol design in which the requirements of untethered nodes and dynamically formed networks are considered from the ground-up, making it particularly suitable for vehicular applications. Here we describe the vehicular networking specific features and protocol design details of the architecture and present evaluation results on performance and scalability.

The prominence of wireless, mobile devices on the Internet today has motivated numerous protocols and architectures, such as the MobilityFirst Future Internet Architecture project. In this work, we present a robust, local-scale, storage-aware routing approach, called GSTAR, for use in Mobility-First networks. GSTAR unifies techniques from MANET and DTN routing protocols. This unification with in-network storage enables it to overcome mobility-related challenges such as link quality variation, node disconnection, and network partitioning. Through NS3-based simulation, we show that GSTAR outperforms traditional link-state protocols for both wireless and hybrid wired-wireless network environments.

This paper presents the design and evaluation of a novel distributed shared hosting approach, DMap, for managing dynamic identifier to locator mappings in the global Internet. DMap is the foundation for a fast global name resolution service necessary to enable emerging Internet services such as seamless mobility support, content delivery and cloud computing. Our approach distributes identifier to locator mappings among Autonomous Systems (ASs) by directly applying K>1 consistent hash functions on the identifier to produce network addresses of the AS gateway routers at which the mapping will be stored. This direct mapping technique leverages the reach ability information of the underlying routing mechanism that is already available at the network layer, and achieves low lookup latencies through a single overlay hop without additional maintenance overheads. The proposed DMap technique is described in detail and specific design problems such as address space fragmentation, reducing latency through replication, taking advantage of spatial locality, as well as coping with inconsistent entries are addressed. Evaluation results are presented from a large-scale discrete event simulation of the Internet with ~26,000 ASs using real-world traffic traces from the DIMES repository. The results show that the proposed method evenly balances storage load across the global network while achieving lookup latencies with a mean value of ~50 ms and 95th percentile value of ~100 ms, considered adequate for support of dynamic mobility across the global Internet.

In the emerging paradigm of pervasive computing, applications change their behaviors in response to their environmental context, which is provided by the smart objects in the Internet of Things (IoT). Due to the inherent heterogeneity of physical world objects, realizing the IoT requires service layers to fill the gap between the low level interfaces of networked objects and the applications which use them. In this paper, we show that the MobilityFirst Future Internet Architecture is an ideal platform for realizing pervasive computing in an IoT framework. In particular, MobilityFirst's identity based routing, overloaded identities, content caching and in-network compute plane are excellent building blocks for IoT applications. We then present a detailed example of a location based service built using MobilityFirst.

The field of wireless and mobile communication has a remarkable history that spans over a century of technology innovations from Marconi's first transatlantic transmission in 1899 to the worldwide adoption of cellular mobile services by over four billion people today. Wireless has become one of the most pervasive core technology enablers for a diverse variety of computing and communications applications ranging from third-generation/fourth-generation (3G/4G) cellular devices, broadband access, indoor WiFi networks, vehicle-to-vehicle (V2V) systems to embedded sensor and radio-frequency identification (RFID) applications. This has led to an accelerating pace of research and development in the wireless area with the promise of significant new breakthroughs over the next decade and beyond. This paper provides a perspective of some of the research frontiers of wireless and mobile communications, identifying early stage key technologies of strategic importance and the new applications that they will enable. Specific new radio technologies discussed include dynamic spectrum access (DSA), white space, cognitive software-defined radio (SDR), antenna beam steering and multiple-input-multiple-output (MIMO), 60-GHz transmission, and cooperative communications. Taken together, these approaches have the potential for dramatically increasing radio link speeds from current megabit per second rates to gigabit per second, while also improving radio system capacity and spectrum efficiency significantly. The paper also introduces a number of emerging wireless/mobile networking concepts including multihoming, ad hoc and multihop mesh, delay-tolerant routing, and mobile content caching, providing a discussion of the protocol capabilities needed to support each of these usage scenarios. In conclusion, the paper briefly discusses the impact of these wireless technologies and networking techniques on the design of emerging audiovisual and multimedia applications as they migrate to mobile Internet platforms.

With the rapid advance in the technology area of data storage, storage capacities have increased substantially while the price has been dropping fast. Motivated by this trend, it has been proposed in the Cache-and-Forward architecture that storage is incorporated into each intermediate CNF router. Content can be cached at CNF routers when they flow through the network, and therefore, routers can serve the subsequent requests later on, without forwarding the requests to the host server, we refer to this caching paradigm as In-Network Caching. In this paper, the content caching is enhanced by Content Broadcast(CB), by which a CNF router broadcasts the information of cached content to its neighboring nodes. In order to solve the problem that with limited storage, how an intermediate CNF router optimally decides which passing content should be cached, we develop a mathematical model for CB to minimize the average content retrieval latency, and propose the Independent Allocation algorithm. We compare the average content retrieval latencies of the proposed caching scheme with two other commonly used cache replacement policies. We study the impact of cache size and locality parameter. The proposed scheme is shown to provide significant performance improvement under various settings by as large as 65%.

With integrated in-network caching diagram, each router advertises the cached content to its immediate neighbors. Although the baseline content broadcast strategy can significantly improve the performance, it leads to low overall cache utilization while each router makes independent caching decisions. In this paper, we enhance the efficiency of content broadcast by providing implicit coordination among neighboring routers. Through detailed simulations, we show that the proposed technique has dramatic performance improvement over the baseline content broadcast scheme. More importantly, this performance gain can be achieved with a minimal communication overhead.

Virtual network (VN) mapping is a useful tool for mapping VNs to physical mesh networks. This study extends the idea of mapping VNs from the wired world to the wireless domain by showing its potential applications. Since the generic VN mapping problem is NP-Hard, this study shows how the wireless VN mapping problem can be simplified and be used instead as a mechanism for provisioning wireless points of presences (POPs) as additions to conventional cellular voice and data services. Two heuristic algorithms GSA and GDR are proposed for producing a 2-phase solution to the mapping problem, which corresponds to conventional network deployment process. The results obtained from the VN mapping algorithms proposed here can be used for comparison of overall performance achieved by deploying a particular type of physical network. Further, using this setup, the network operator can determine the costs and benefits associated with setting wired or wireless links on the physical network. Performance is determined based on perceived revenue, and substrate utilization.

This paper presents an adaptive interference control method to mitigate undesirable interference from femtocells to macrocell users in hierarchical cellular networks. Such mechanisms usually require over-the-air signalling for estimation of interference resulting significant bandwidth overhead. The proposed `Adaptive Interference Scaling' (AIS) method uses geolocation information for femtocell power control for interference avoidance. In this approach, each femtocell calculates interference contributed to nearby macrocell users and adjusts the power to meet specific target signal-to-interference-plus-noise (SINR) level. Results from simulations show that AIS is able to increase the number of macrocell users achieving target data rates by up to 158% relative to baseline without adaptive control, while resulting in only 12.2% femtocell users receiving rates below the target. AIS achieves improved performance by using location information to calculate and limit the interference power contributed by femtocells to macrocell users, while allowing the network operator to set any desired target rates.

The increasing density and data rate of unlicensed band wireless devices in small office and home (SOHO) environments has led to significant inter- and intra-radio interference problems. Multiple competing standards such as the IEEE 802.11b/g, Bluetooth and ZigBee, all of which operate in the 2.4 GHz ISM band, can interfere with each other when used in typical indoor environments, potentially causing significant performance degradation. This paper presents detailed experimental results (using the ORBIT radio grid testbed) to quantify the effects of such interference in representative SOHO scenarios. In particular, different topologies, traffic loads and number of interfering devices are emulated to show the impact of multi-radio interference and to characterize each kind of interference. Further, a cross-layer, multi-radio interference diagnosis framework (called “spectrum MRI”) is described with the aim of isolating and classifying multi-radio interference problems using heuristic and model-based methods. A specific example of identifying interference problems which may affect an 802.11g video link is given to illustrate the proposed measurement and diagnosis framework.

Routing protocols for mesh and ad-hoc networks are designed under the assumption that the underlying channel is unreliable and nodes may be mobile. Therefore, route failure detection and recovery are important components of the design. However, under high traffic scenarios, an aggressive route failure detection technique may create a chain of routing messages which interfere with the data communication triggering another route failure detection. This paper investigates the chain effect in two popular routing protocols, OLSR and AODV, with emphasis on cross layer feedback based route error detection mechanism. NS2 based simulations are performed in static and mobile scenarios. Random as well as group mobility scenarios were evaluated. Simulation results reveal that under high traffic load, a chain of route recoveries and MAC collisions are triggered due to localized MAC layer contention rather than mobility in AODV due to direct MAC feedback.

This paper presents an adaptive MAC (AMAC) protocol for supporting MAC layer adaptation in cognitive radio networks. MAC protocol adaptation is motivated by the flexibility of emerging software-defined radios which makes it feasible to dynamically adjust radio protocols and parameters in order to maintain communications quality. Dynamic changes to the MAC layer may be useful in tactical or vehicular networking scenarios, where radio node density, traffic volumes and service requirements can vary widely over time. A specific control framework for the proposed AMAC algorithm is described based on the ”CogNet” protocol stack which uses a Global Control Plane (GCP) to distribute control information between nearby radios. An AMAC prototype which switches between CSMA and TDMA is evaluated for various traffic scenarios using the NS-2 simulator. In addition, a proof-of-concept AMAC protocol is implemented using GNUradio/USRP platforms on the ORBIT radio grid testbed. Detailed simulation and experimental results are given for both UDP and TCP traffic with different usage scenarios and application models. The results show that AMAC can provide improved performance relative to a conventional static system and can be implemented with reasonable control protocol overhead and latency.

This paper presents a novel storage aware routing (STAR) protocol designed to provide a general networking solution over a broad range of mobile and wireless scenarios. STAR enables routing policies which adapt seamlessly from a well-connected wired network to a disconnected wireless network using a 2-Dl routing metric composed of a short and a long term route cost and storage availability on downstream routers. Temporary in-network storage is preferred over forwarding along a path that is slower than average and opportunistic transmission is encouraged when a faster route becomes available. Results from ns2 based simulations show that STAR achieves 20% higher throughput compared to OLSR under varying link conditions and delivers 20% more files in DTN scenarios. Experimental evaluation of STAR on the ORBIT testbed demonstrates significant performance improvements with 25% higher peak throughput compared to OLSR in a wireless mesh network.

Providing air-time guarantees across a group of clients forms a fundamental building block in sharing an access point (AP) across different virtual network providers. Though this problem has a relatively simple solution for downlink group scheduling through traffic engineering at the AP, solving this problem for uplink (UL) traffic presents a challenge for fair sharing of wireless hotspots. Among other issues, the mechanism for uplink traffic control has to scale across a large user base, and provide flexible operation irrespective of the client channel conditions and network loads. In this study, we propose the SplitAP architecture that address the problem of sharing uplink airtime across groups of users by extending the idea of network virtualization. Our architecture allows us to deploy different algorithms for enforcing UL airtime fairness across client groups. In this study, we will highlight the design features of the SplitAP architecture, and present results from evaluation on a prototype deployed with: (1) LPFC and (2) LPFC+, two algorithms for controlling UL group fairness. Performance comparisons on the ORBIT testbed show that the proposed algorithms are capable of providing group air-time fairness across wireless clients irrespective of the network volume, and traffic type. The algorithms show up to 40% improvement with a modified Jain fairness index.

The 802.16e standard for broadband wireless access mandates the presence of QoS classes, but does not specify guidelines for the scheduler implementation or mechanisms to ensure air time fairness. Our study demonstrates the feasibility of controlling downlink airtime fairness for slices while running above a proprietary WiMAX basestation (BS) scheduler. We design and implement a virtualized infrastructure that allows users to obtain at least an allocated percentage of BS resources in the presence of saturation and link degradation. Using Kernel virtual machines for creating slices and Click modular router for implementing the virtual network traffic shaping engine we show that it is possible to adaptively control slice usage for downlink traffic on a WiMAX Basestation. The fairness index and coupling coefficient show an improvement of up to 42%, and 73% with preliminary indoor walking mobility experiments. Outdoor vehicular measurements show an improvement of up to 27%, and 70% with the fairness index and coupling coefficient respectively

The Cache-and-Forward (CNF) protocol architecture was proposed to support efficient mobile content delivery services in the future Internet. In contrast to the TCP/IP protocol stack which is based upon the assumption of reliable end-to-end path through the network, the CNF architecture considers varying access link speed/quality and periods of disconnection as inherent properties of the network. Routers in a CNF network are built with large memory space for in-network caching and temporary storage to support transient disconnections due to mobility or link quality variation. Content delivery through the network follows a hop-by-hop transport method in which files move as single entities from one router to the next rather than as end-to-end packet streams. A novel storage aware routing protocol (STAR) is proposed to efficiently support mobile and wireless end-users through the use of a two-dimensional metric that takes into account both short-term and long-term path quality in making forwarding and storage decisions. A reliable link layer provides per hop file transfer reliability. This paper provides an outline of the three basic protocol components of CNF i.e., transport, routing and link layers and describes a proof-of-concept implementation of the protocol stack on the ORBIT testbed. Performance evaluation results in multi-hop wireless scenarios with lossy link conditions show 66% improvement in wireless network throughput compared to TCP and 60% lower packet loss rate when compared to UDP.

Frame aggregation is a wireless link optimization mechanism that aims to reduce transmission overheads by sending multiple frames as the payload of a single MAC frame. Static assignment of frame aggregation parameters can result in delay penalties due to variations in traffic type or load levels. Another possible side effect is an increase in packet error rate in noisy environments due to large aggregated frame size. Adaptive aggregation methods have previously been proposed to deal with the above problems independently, but there is still a need for a unified adaptation algorithm that addresses both aspects in an integrated manner. In this study, a backlogged queue (BQ) aggregation approach is proposed that considers both these aspects, and also ensures inter-operability with other WLAN devices that are not capable of frame aggregation. Performance evaluation of the proposed algorithm on the ORBIT testbed shows throughput improvements of up to 56% in the presence of channel noise and 25% in scenarios with high contention over using a simple txop. An experimental case study shows improvement in FTP file transfer times of up to 11% while preserving performance for real time traffic.

Wireless network emulator testbeds have become increasingly important for realistic, at-scale experimental evaluation of new network architectures and protocols. Typically, wireless network performance measurements are made at multiple layers of the wireless protocol stack, i.e. link layer, MAC layer and network layer. This study highlights the impact of layer 2 frame aggregation that is enabled by default in the software drivers for commodity wireless 802.11 devices while it is still not a part of the core 802.11 standard. Using experimental measurements, it is shown that this feature has an impact across a diverse set of wireless experiments and should be considered while comparing results. Measurements on the ORBIT testbed show that throughput measurements can vary up to a startling 25% for certain packet sizes and the variance in receiver side interframe delays can almost double if MAC aggregation and preset transmission opportunities are not taken into consideration. Further results for VoIP traffic show a deterioration in jitter of up to 8 times when coupled with MAC layer aggregation in 802.11.

In order to meet the increasing demands of content dissemination in Internet, we propose a novel architecture for the future Internet called cache-and-forward (CNF), which transports content as "packages" in a hop-by-hop manner towards the destination, instead of transporting a stream of fragmented packets along an established TCP/IP connection. In this paper, we discuss how the CNF network architecture can be designed for efficient content retrieval. We first introduce several specific services provided in CNF network which are centered around content handling and mobile access. We then give an overview of the CNF protocol stack, which is built on top of IP, and consists of a data plane and a control plane. We provide detailed descriptions of each protocol within both planes. Then we present two caching algorithms, where one involves each CNF router making independent decisions on content caching while the other coordinates node caching within an autonomous system (AS) through hashing. Finally, we gave the initial simulation results to show the performance benefits of hop-by-hop transport and content caching.

This paper presents an adaptive MAC (AMAC) protocol for supporting MAC layer adaptation in cognitive radio networks. MAC protocol adaptation is motivated by the flexibility of emerging software-defined radios which make it feasible to dynamically adjust radio protocols and parameters. Dynamic changes to the MAC layer may be useful in wireless networking scenarios such as tactical or vehicular communications where the radio node density and service requirements can vary widely over time. A specific control framework for the proposed AMAC is described based on the "CogNet" protocol stack which uses a "global control plane (GCP)" to distribute control information between nearby radios. A proof-of-concept AMAC prototype which switches between CSMA and TDMA is implemented using GNU radio platforms on the ORBIT radio grid testbed. Experimental results are given for both UDP and TCP with dynamic traffic variations. The results show that adaptive MAC can be implemented with reasonable control protocol overhead and latency, and that the adaptive network achieves improved performance relative to a conventional static system.

In multi-hop infrastructure wireless mesh networks (WMNs), the association mechanism, by which a client station (STA) affiliates with a mesh access point (MAP), and the routing algorithm, through which MAPs form a multi-hop backhaul for relaying STAs' traffic, determine a two-tier logical topology. Apparently the STA-MAP association mechanism and the backhaul routing impact the available bandwidth that can be allocated to each STA. In this paper, we formulate a joint optimization problem of STA-MAP association, backhaul routing and bandwidth allocation. Our rigorous framework maximizes the network throughput while guaranteeing network-wide fairness among STAs, taking into account the bandwidth constraints of both access and backhaul links, as well as the wireless interference. We then develop approximation algorithms for efficiently solving the joint optimization problem. A method to decouple topology construction and bandwidth allocation is proposed to simplify the optimization problem under integral association and single- path routing, which is NP hard in the original formulation. We also use the clique approximation to alleviate the complexity for constructing the wireless interference constraints. Furthermore a scheduling algorithm is proposed, which coordinates channel access to provide bandwidth guarantee and can recover certain performance loss due to the clique approximation. Our evaluation demonstrates that constructing a good logical topology can improve throughput while enhancing fairness, and our algorithms can achieve performance close to the optimal solution to the joint association, routing and bandwidth allocation problem.

In a wireless mesh network, a client station needs to associate with a mesh access point for network access. Conventional association mechanisms assume a high-speed backhaul and only the access link being the bottleneck. This assumption holds for most WLANs, but in wireless mesh networks traffic could be bottlenecked either by the access link or by the bandwidth-limited wireless backhaul. In this paper, we propose an association mechanism for wireless mesh networks to improve stations' end- to-end communication performance with the Internet. A station makes its association decision by jointly considering the quality of the access link between the station and the candidate mesh access point as well as the cost of the multi-hop path from the mesh access point to the gateway. In addition, we design two access link metrics, Contention Aware Expected Transmission Time (CAETT) and Load Aware Expected Transmission Time (LAETT). The main strength of CAETT is incorporating the impact of 802.11 MAC layer contention on the bandwidth sharing among the multi-rate stations. LAETT further captures the traffic load. We evaluate the performance of our system through simulations and demonstrate that the proposed joint association mechanism with the CAETT/LAETT metric can significantly improve the end-to-end performance for wireless mesh networks by up to 60%.

We report the design, implementation and evaluation of Merged Hybrid ARQ with staggered FEC (MHARQ) system for video multicast over wireless LANs. MHARQ combines the advantages of receiver-driven staggered FEC and hybrid ARQ schemes to compensate the large dynamic range of WLAN channels and to achieve high reliability, scalability and wireless bandwidth efficiency for video multicast. The FEC packets generated by a cross-packet FEC code are divided into multiple streams according to the pre-configured overhead and are transmitted in different multiple IP multicast groups. Certain FEC streams are delayed from the original video stream. The receivers dynamically join/leave the FEC multicast groups based on the channel conditions. For efficient utilization of WLAN bandwidth, FEC data for a multicast group would not be transmitted by the APs in wireless networks if no receiver joins this group. The time shift between the video stream and the FEC streams introduces temporal diversity and compensates for the client join delay and handoff interruption. In addition, when delayed FEC packets are not enough to recover the lost packets, the receivers can send a hybrid ARQ request to the video server. We design a channel estimation algorithm for a receiver to dynamically determine the delayed FEC multicast groups to join and/or send ARQ NACK to request for retransmission. Using the ORBIT radio grid testbed, we have investigated the performance of the proposed MHARQ system with various numbers of users per AP and different number of APs per video server. It is demonstrated via real system implementation on ORBIT that MHARQ improves wireless bandwidth efficiency and scalability for reliable video multicast, compared with existing reliable multicast schemes.

This paper presents an integrated routing and MAC scheduling protocol (IRMA) for multihop wireless mesh networks. The IRMA approach is motivated by the fact that the overall performance achieved by conventional layered approaches (802.11 MAC combined with independent ad hoc routing protocols) is significantly lower than the underlying network capacity. We propose to integrate the routing and MAC into a single protocol layer and use joint optimization techniques to establish end-to-end path and TDMA schedules for flows across the network. This approach achieves non-conflicting allocation of channel resources based on global or local traffic flow specifications and the network graph. The proposed method not only establishes interference-free MAC link schedules, but also helps to find optimal routes which can route around congested areas of the network. Two specific IRMA algorithms are proposed and evaluated in this paper. The first method solves min-hop routing, then optimizes link scheduling based on routing results and real-time flow demands. The second approach attempts to optimize routing and scheduling decisions simultaneously, using available MAC bandwidth information to route around congested areas. Both centralized and distributed algorithms based on these methods are proposed and evaluated with detailed simulations. Results show significant 2–3x improvements in network throughput when compared with baseline 802.11-based mesh networks using independent routing protocols.

This paper investigates a three-tier hierarchical hybrid wireless network designed to provide significant improvements in system capacity and performance relative to conventional "flat" ad hoc and two-tier "hybrid" networking approaches. The analytical results illustrate that in a three-tier hierarchical network with nAaccess points (AP), nFdual-radio forwarding nodes (FN), and nMmobile nodes (MN), linear scaling of low-tier capacity can be achieved when the low-tier transmission range satisfies rL= O(1/radic(nF)); linear scaling of high-tier capacity can be approached in the scaling regime nA= Omega(radic(nF)) and nA= O(nF/log nF), when the high-tier transmission range satisfies rH= O(1/radic(nA)). The scaling properties of the three-tier hierarchical ad hoc network are studied using ns-2 based system simulation models. In particular, we investigate the impact of relative node densities and traffic pattern on the scalability of the network. The simulation results are consistent with the analysis, and demonstrate that system throughput can be scaled by using the right proportions of FN's and AP's, making it possible to design high-capacity, low-cost mesh networks with a moderate number of radio FN's and only a few wired AP's.

This paper presents an architecture for adaptive cognitive radio networks based on the concept of a "global control plane". The proposed control architecture uses a predetermined common coordination channel for spectrum etiquette, network establishment and adaptation to changing interference environments. The focus of this work is on design and evaluation of three key components of the control protocol - bootstrapping, discovery and naming/addressing. The bootstrapping protocol uses beacons to inform neighboring nodes about a node's PHY/MAC capabilities and current status. The network discovery protocol helps nodes to obtain a global view of reachability and end-to-end paths in the network by exchanging and propagating local link states. Further, nodes obtain their IP addresses and perform name to network address translations using a distributed naming/addressing scheme. An ns2 simulation model of the cognitive radio network with global control has been developed and used to evaluate performance in terms of network setup time, control overhead and achievable data throughput.

We report the implementation experience and experimental evaluation of a staggered adaptive forward error correction (FEC) system for video multicast over wireless LANs. In the system, the parity packets generated by a cross-packet FEC code are transmitted at a time delay from the original video packets, i.e. staggercasting video stream and FEC stream in different multicast groups. The delay provides temporal diversity to improve the robustness of video multicast, especially to enable the clients to correct burst packet loss using FEC and to achieve seamless handoff. A wireless client dynamically joins the FEC multicast groups based upon its channel conditions and handoff events. We have implemented the system including the streaming server and client proxy. A novel software architecture is designed to integrate the FEC functionality in the clients without requirement for changing the existing video player software. We conduct extensive experiments to investigate the impact of FEC overhead and the delay between the video stream and FEC stream to the video quality under different interference levels and mobile handoff durations. The efficacy of staggered adaptive FEC system on improving video multicast quality is demonstrated in real system implementation.

Conventional topology-based routing protocols such as AODV, DSR and ZRP are not suitable for highly mobile ad hoc network, where frequent route re-establishment may cause high end-to-end delay and packet loss. This paper presents a new ad hoc routing protocol called the multi-hop MAC forwarding protocol (MMFP). The MMFP avoids explicit path setup in order to reduce the control overhead associated with it, but instead uses the reachability information towards the destination at each hop. Next-hop nodes are determined on-the-fly by contention based on a priority value. The basic operations of the MMFP are conceptually similar to that of MAC bridges and position-based ad-hoc routing protocols. The MMFP is designed to be integrated with the IEEE 802.11 MAC protocol in order to achieve higher efficiency and accuracy in its time-critical operations. Simulation results show that the MMFP outperforms the AODV in a practical ad hoc network scenario in terms of both the end-to-end delay and packet delivery ratio.

Compared with unicast, video multicast over 802.11 WLANs should handle varying channel conditions of multiple users and user topology changes as well as scalability to achieve good quality for all users in the serving area. This paper analyzes error control strategies available in different layers of the network stack, including modulation and channel coding in physical layer, cross-packet Forward Error Correction (FEC), packet size optimization and scalable video coding in application layer. By combining and adapting these schemes jointly, an adaptive cross layer optimization algorithm is proposed for scalable video multicast over 802.11 WLANs. Based on a variety of criteria, improvement in overall video quality for all the targeted users can be achieved.

Video streaming over wireless networks is a challenging problem because of limited bandwidth, high bit error rates and fluctuations in channel quality. This paper investigates the problem of video multicast over an infrastructure-mode 802.11b wireless local area network. In particular, a multirate simulcasting model is considered to deal with variations in channel conditions, and both distributed and centralized algorithms are developed for receiver rate adaptation. OPNET simulations are used to validate the algorithms which aim to maximize the user satisfaction index. Simulation results show that the centralized scheme is better than the distributed scheme as it maximizes the aggregate user satisfaction index under the available bandwidth and channel conditions. It is also observed that FEC increases the user satisfaction index when the channel errors are low; however with higher channel errors EEC cannot increase the user satisfaction index. Based on these results, both open and closed loop algorithms have been developed to maximize the received video quality

This paper investigates the use of reactive cognitive radio algorithms to enable co-existence between IEEE 802.11b and 802.16a networks in the same unlicensed band. In particular, we develop a system model in which the two wireless systems share radio resources in frequency, space and time, and reactive coordination methods are used to reduce the mutual interference and improve link throughput. Reactive cognitive radio schemes utilize the available degrees of freedom in frequency, power and time, and react to observations in these dimensions to avoid interference. Dynamic frequency selection (DFS) enables radios to choose the band with the least interference. power control (PC) allows communications at the least possible transmit power. Time agility (TA) enables radios to adapt to each other's traffic patterns and avoid increasing interference in poor channel conditions. Simulation results are given for the following scenarios: (i) single 802.16a cell with single 802.11b hotspot; (ii) multiple 802.16a cells with multiple 802.11b hotspots. The results demonstrate that reactive cognitive radio schemes can provide significant improvements in 802.11b and/or 802.16a throughputs in the typical operating scenarios considered.

This paper investigates the TCP simultaneous-send problem (SSP) which arises in infrastructure mode WLANs. In particular it has been observed that for file transfer traffic, 802.11 wireless nodes have a sustained supply of packets to send and hence experience a relatively high rate of MAC contention. For TCP, this results in competition among data and ACK packets for channel access which causes the SSP that deteriorates flow throughput. This SSP can be alleviated by skipping TCP ACKs. Detailed simulation results are presented to demonstrate the usefulness of ACK skipping in various network scenarios such as with MAC retries and multiple TCP flows. The largest improvement is seen for the case of a single TCP flow, and moderate gains are also achieved in cases with multiple streams. For the single TCP stream case with 1 ACK skip and no MAC retries, TCP throughput improves 30% for short-lived and 98% for long-lived TCP transmissions. The paper concludes with potential cross-layer solutions that potentially provide further improvements, including the use of the point coordination function (PCF) to reduce contention between multiple TCP streams and returning ACK packets.

This paper presents an overview of the ORBIT (open access research testbed for next-generation wireless networks) radio grid testbed, that is currently being developed for scalable and reproducible evaluation of next-generation wireless network protocols. The ORBIT testbed consists of an indoor radio grid emulator for controlled experimentation and an outdoor field trial network for end-user evaluations in real-world settings. The radio grid system architecture is described in further detail, including an identification of key hardware and software components. Software design considerations are discussed for the open-access radio node, and for the system-level controller that handles management and control. The process of specifying and running experiments on the ORBIT testbed is explained using simple examples. Experimental scripts and sample results are also provided.

The paper describes the design and evaluation of ad-hoc extensions to the IEEE 802.15.3 medium access control (MAC) layer for wireless personal area networks (WPANs). The proposed protocol allows communication between ad-hoc devices without the intervention of any central entity and, at the same time, ensures bounded delays for isochronous traffic. Ad-hoc communication is made possible without the hidden terminal problem. Features from both IEEE 802.15.3 and IEEE 802.11 standards are used - in particular the TDMA structure from 802.15.3 and RTS-CTS based contention from 802.11. The protocol includes certain other ingredients, like a decentralized synchronization procedure using randomized beaconing, periodically interspersed contention periods, bit maps to convey reservation information and a mechanism to estimate and react to channel errors. The MAC has been simulated in ns-2 and simulation results are reported.

Ad-hoc wireless networks with multi-rate radios (such as 802.11a, b, g) require a new class of MAC/PHY aware metrics that take into account factors such as physical-layer link speed and MAC-layer channel congestion. Conventional "layer 3" ad-hoc routing algorithms typically make routing decisions based on the minimum hop-count (MH). Use of the MH metric leads to the selection of paths with few hops, but one or more of these hops may turn out to be low-speed radio links due to adaptive rate selection at the physical layer We investigate a new cross-layer routing metric that takes into account both physical layer link speed and estimated channel congestion, thus aiming to minimize end-to-end delay that includes both transmission and access times. The proposed "PARMA" routing metric thus helps to spread the traffic across the "good links and nodes" in the network, increasing network capacity and reducing packet loss and delay. The paper presents the design and implementation of the proposed PARMA metric for proactive ad-hoc routing protocols, such as DSDV. DSDV modifications for incorporating the MAC/PHY aware metric into an ns-2 simulation model are given. Simulation results for typical multi-rate 802.11 ad-hoc network scenarios show that the proposed cross-layer PHY/MAC aware metric achieves significantly higher network throughput and decreases network congestion by selecting paths with high bit-rate links, while also avoiding areas of MAC congestion.

This paper describes a beacon assisted discovery mechanism for self-organizing hierarchical ad-hoc networks. The discovery protocol, which operates between the medium access control (MAC) and network layers, is responsible for topology formation in the ad-hoc network taking into account performance objectives such as throughput, delay, energy consumption and robustness. The proposed discovery protocol operates by listening to augmented MAC-layer beacons from neighboring radio nodes and then selects a subset of these for routing associations based on specified criteria. A distributed heuristic algorithm for topology formation is considered and compared with upper-bound centralized algorithms with optimization objectives such as maximum throughput, minimum delay or minimum energy. Simulation results (based on ns-2 models) are given for the performance of the proposed discovery methods, demonstrating significant improvements in routing overhead when compared to an ad hoc network without discovery. A proof-of-concept prototype implementation for an IEEE 802.11b-based three-tier hierarchical ad hoc network is briefly described in conclusion.

The paper presents a novel medium access control (MAC) protocol for QoS support in multi-hop ad-hoc wireless networks. The proposed D-LSMA (distributed link scheduling multiple access) protocol uses an extension of the 802.11 CSMA/CA procedure as the basis for a distributed link scheduling algorithm which results in dynamic TDMA-like bandwidth allocation among neighboring wireless nodes without the need for global synchronization. In addition to supporting QoS, the proposed scheduling technique also solves the "exposed node" problem in ad-hoc 802.11, thus resulting in improved throughput in many scenarios. Simulation results from an ns-2 model are presented for a 15-node random ad-hoc network. The results demonstrate significant performance improvements relative to ad-hoc 802.11, with capacity increases typically /spl sim/20% for the example considered. Also, the D-LSMA network is shown to offer far better real-time packet delay and fairness properties than 802.11, particularly under overload and heavy contention conditions.

This paper describes the design and implementation of a novel 802.11-based self-organizing hierarchical ad-hoc wireless network (SOHAN), and presents some initial experimental results obtained from a proof-of-concept prototype. The proposed network has a three-tier hierarchy consisting of low-power mobile nodes (MNs) at the lowest layer, forwarding nodes (FNs) with higher power and multi-hop routing capability at the middle layer, and wired access points (APs) without power constraints at the highest layer. Specifics of new protocols used for bootstrapping, node discovery, and multi-hop routing are presented, and the overall operation of the complete hierarchical ad-hoc network is explained. A prototype implementation of the SOHAN network is outlined in terms of major hardware and software components, and initial experimental results are given.

A novel self-organizing hierarchical architecture is proposed for improving the performance and scalability properties of ad hoc wireless networks. This paper presents the results of a simulation study of the performance and throughput capacity of a specific three-tier hierarchical ad hoc network with 802.11 radios, forwarding nodes and access points. The performance of the proposed hierarchical network is evaluated for two well-known classes of ad hoc routing protocols: dynamic source routing (DSR) and ad hoc on-demand distance vector (AODV), and compared with that of a conventional "flat" ad hoc network. The results for an example sensor network scenario have shown significant capacity increases with the hierarchical architecture for both DSR and AODV cases. Modifications to ad hoc routing metrics for energy efficiency are also considered. The scalability properties of the three-tier hierarchy are studied further in terms of the achievable system capacity as a function of the relative densities of sensor nodes, forwarding nodes and access points. It is shown that the capacity of the three-tier hierarchical network scales well when the number of forwarding nodes and access points are increased in the right proportions.

This paper presents a spectrum etiquette protocol for efficient coordination of radio communication devices in unlicensed (e.g. 2.4 GHz ISM and 5 GHz U-NII) frequency bands. The proposed etiquette method enables spectrum coordination between multiple wireless devices using different radio technologies such as IEEE 802.11.x, 802.15.x, Bluetooth, Hiperlan, etc. The basic idea is to standardize a simple common protocol for announcement of radio and service parameters, called the "common spectrum coordination channel (CSCC)". The CSCC mechanism is based on the low bit-rate mode of the 802.11b physical layer, along with a periodic broadcast protocol at the MAC layer. The CSCC protocol is "policy neutral" in the sense that it provides a general mechanism, which can accommodate a wide range of specific spectrum sharing rules. One possible CSCC protocol implementation is described in terms of the packet formats used and related channel access rules. Proof-of-concept experimental results from a CSCC prototype are presented for an example scenario in which nearby 802.11b and Bluetooth devices contend for 2.4 GHz ISM band access. Results showing file transfer delay with and without CSCC etiquette are given for comparison purposes.

This paper studies the effects of user mobility and handoff path rerouting on the traffic distributions in a mobile network environment. In mobile ATM networks, extra traffic load may be added to network links due to user mobility and handoff path rerouting. This requires higher network link capacity and possible topology reengineering in order to support the same quality of service (QoS) for mobile services. To capture the dynamic variations in mobile ATM networks, we propose to use a flow model. The model represents the mobile-generated traffic as a set of stochastic flows over a set of origin-destination (OD) pairs. The user mobility is defined by transfer probabilities of the flows and the handoff path rerouting algorithm is modeled by a transformation between the routing functions for traffic flows. The analysis shows that user mobility may cause temporal variations as well as smoothing effects on the network traffic. Using the flow network model, typical handoff path rerouting algorithms are evaluated through both analytical and experimental approaches. The evaluation methodology can be used for either redesigning the network topology for a given path rerouting algorithm or selecting a path rerouting algorithm for a given network topology under a specific mobile service scenario.

A dynamic framework for quality-of-service (QoS) control of video in distributed multimedia applications is presented The framework allows flexible and efficient video delivery with application-level QoS support. Key components of the framework are: client QoS renegotiation, server source rate control and dynamic bandwidth allocation. The soft-QoS requirements of the application are communicated among the framework components using distributed software objects, called service contracts. A proof-of-concept prototype of a video browser with user-level control of soft-QoS is implemented within the proposed framework. Experiences with the prototype and its performance are discussed. Results suggest that the framework is particularly suitable for the mobile multimedia scenario allowing mobile applications to maintain soft-QoS while roaming without compromising network utilization.

This paper studies mobile-generated traffic distributions in mobile ATM networks and evaluates the performance of path rerouting algorithms for handoff control. In mobile ATM networks, user mobility and handoff path rerouting may produce extra traffic load over network links, requiring larger network capacity to support the same QoS. We propose a flow model for mobile ATM networks. The model represents the mobile-generated traffic as a set of stochastic flows over a set of OD (origin-destination) pairs. The user mobility is defined by transfer probabilities of the flows and the handoff path rerouting algorithm is modeled by a transformation between the routing functions for traffic flows. The analysis shows that user mobility may cause temporal variations as well as smoothing effects on the network traffic. Using the flow network model, typical handoff path rerouting algorithms are evaluated through both analytical and experimental approaches. The evaluation methodology can be used for either redesigning the network topology for a given path rerouting algorithm or selecting a path rerouting algorithm for a given network topology under a specific mobile service scenario.

The concept of "wireless ATM" (WATM), first proposed in 1992, is now being actively considered as a potential framework for new-generation wireless communication networks capable of supporting integrated, quality-of service (QoS) based multimedia services. In this review paper, we outline the technological rationale for wireless ATM, present a system-level architecture, and discuss key design issues for both mobile ATM switching infrastructure and radio access subsystems. The WATM radio access layer issues covered in this paper include: spectrum allocation; spectrum etiquette; modem technology; and medium access/data link control (MAC/DLC) protocols. Mobile ATM aspects such as ATM signaling extensions for handoff control, location management, and mobile QoS control are discussed. A summary of current wireless/mobile ATM technology development and standardization status is given, including an outline of our WATMnet prototype. The paper concludes with a discussion of future directions for wireless ATM technology such as Internet protocol (IP) integration and mobile multimedia terminals/applications.

This paper studies handoff control mechanisms based on a 'mobile ATM' concept. Mobile ATM refers to an ATM infrastructure which supports ubiquitous mobile services through extending ATM signaling functions within the ATM backbone. In order to meet the performance criteria required by a diverse set of mobile services, we propose a unified handoff control protocol which is independent of wireless access technologies (ATM/nonATM) and path rerouting algorithms. Such a protocol can provide (1) a common software architecture for system implementation, (2) a common measurement base for performance evaluation and (3) a common signaling syntax for standardization. A prototype system is implemented based on this protocol, which provides an IP-over-ATM service with WaveLAN access.

This paper presents a selection of current topics related to emerging wireless and mobile ATM network technologies. A summary of the wireless ATM system architecture and related radio access layer and "M" UNI/NNI protocols is given. As technical approaches for these radio access and mobile network protocols are now entering the convergence phase, we identify several systems-oriented issues that need to be addressed next for efficient and flexible use of the mobile/wireless ATM networking platform. Specific topics discussed include: (1) quality-of-service (QoS) control in wireless ATM systems; (2) interworking with legacy wireless services such as GSM and IS-95/136; and (3) IP support alternatives such as IP-over-mobile ATM, IP switching (Ipsofacto), and IP routing over WATM link layer.

This paper presents a medium access level VBR bandwidth allocation scheme for wireless ATM (WATM) networks. After introducing a wireless ATM system model and the associated dynamic TDMA/TDD medium access control (MAC) scheme, an algorithm for VBR slot allocation is proposed. The algorithm combines information from user parameter control (UPC) at the call admission control (CAC) level with rapid MAC level estimates of the requirements for each VBR virtual circuit (VC). The method described is applicable to both conventional VBR and VBR+ in which UPC parameters are dynamically renegotiated as source bit rate requirements vary. Simulation experiments using statistically multiplexed MPEG-1 video are carried out for a typical 25 Mbps dynamic TDMA/TDD wireless ATM access link scenario. Experimental results show that, for the VBR video under consideration, the proposed scheme can achieve throughputs in the range of 60-70% while maintaining reasonable QoS.

A prototype microcellular wireless asynchronous transfer mode network (WATMnet) capable of providing integrated multimedia communication services to mobile terminals is described in this paper. The experimental system's hardware consists of laptop computers (NEC Versa-M) with WATMnet interface cards, multiple VME/i960 processor-based WATMnet base stations, and a mobility-enhanced local-area ATM switch. The prototype wireless network interface cards operate at peak bit-rates up to 8 Mb/s, using low-power 2.4 GHz industrial, scientific, and medical (ISM)-band modems. Wireless network protocols at the portable terminal and base station interfaces support available bit rate (ABR), variable bit rate (VBR), and constant bit rate (CBR) transport services compatible with ATM using a dynamic time-division multiple-access/time-division duplex (TDMA/TDD) MAC protocol for channel sharing and data link control (DLC) protocol for error recovery. A custom wireless control protocol is also implemented between the portable and base units for support of radio link related functions such as user registration and handoff. All network entities including the portable, base and switch use a mobility-enhanced version of ATM ("Q.2931+") signaling for switched virtual circuit (SVC) connection control functions, including handoff. In the first stage of the prototype, the application-level API is TCP/UP over ATM ABR service class using AAL5. Early experiments with the WATMnet prototype have been conducted to validate major protocol and software aspects, including DLC, wireless control, and mobility signaling for handoff, Selected network-based multimedia/video applications requiring moderate bit-rates (/spl sim/0.5-1 Mb/s) in the ABR mode have been successfully demonstrated on the laptop PC.

A scheme for delivery or variable bit-rate (VBR) video over asynchronous transfer mode (ATM) networks where bandwidth can be renegotiated during the duration of a call between the video source and the network is considered. Renegotiation can be initiated by either the video source or the network. The video bandwidth requirement is characterized by a usage parameter control (UPC) consisting, in general, of peak rate, burst length, and sustained rate. A baseline design is outlined where rate-control adjusts the source's rate while a new UPC is requested from the network. When granted, the new UPC allows the source to maintain its target quantization and delay requirements. Rate control epochs may be extended when the network blocks UPC requests or sets a lower UPC value to temporally deal with congestion. Simulation results are presented for VBR MPEG video. The results show that with a moderate renegotiation rate the scheme tracks the bandwidth requirements of the source. As a result, the video quality and bandwidth efficiency can be maintained.

This paper presents an ATM-based transport architecture for next-generation multiservices personal communication networks (PCN). Such "multimedia capable" integrated services wireless networks are motivated by an anticipated demand for wireless extensions to future broadband networks. An ATM compatible wireless network concept capable of supporting a mix of broadband ISDN services including constant bit-rate (CBR), variable bit-rate (VBR), and packet data transport is explored from an architectural viewpoint. The proposed system uses a hierarchical ATM switching network for interconnection of PCN microcells, each of which is serviced by high-speed, shared-access radio links based on ATM-compatible cell, relay principles. Design issues related to the physical (modulation), media access control (MAC), and data-link layers of the ATM-based radio link are discussed, and preliminary technical approaches are identified in each case. An example multiservice dynamic reservation (MDR) TDMA media access protocol is then considered in further detail, and simulation results are presented for an example voice/data scenario with a proportion of time-critical (i.e., multimedia) packet data. Time-of-expiry (TOE) based queue service disciplines are also investigated as a mechanism for improving the quality-of-service (QoS) in this scenario.>

The letter considers modeling methodologies of variable bit-rate (VBR) video sources for performance evaluation of integrated networks. The authors consider an example in which compressed H.261 standard VBR video is transmitted over a local area network carrying both video and data. A group-of-block (GOB) level source model has been constructed using TES (transform-expand-sample)-a new general methodology for generating stationary autocorrelated time series, which can simultaneously capture a prescribed marginal distribution and autocorrelation function. The TES model is used to drive CSMA/CD network simulations for a performance evaluation study. Numerical comparisons with a conventional autoregressive (AR) modeling approach are given.>

A variable-bit-rate (VBR) MPEG video compression encoder is introduced, and the performance of a statistically multiplexed asynchronous transfer mode (ATM) network supporting a number of such VBR video sources is evaluated. Bit-rate characteristics obtained from a detailed simulation are provided for a VBR MPEG encoder for CCIR601 video (operating in the 5-10 Mb/s regime) appropriate for medium-quality multimedia or broadcasting applications. The results presented include bit-rate traces and signal-to-noise-ratio data for typical test sequences, along with summary statistics such as the marginal distribution of frame rate. Data from a study of statistical multiplexing on an ATM network are also given. Simulation results for an ATM statistical multiplexer with N>>1 VBR MPEG sources are presented in terms of key performance measures such as cell loss rate and delay versus throughput. The results confirm that ATM channel efficiencies of approximately 80-90% can be obtained at reasonable cell loss rate and delay levels.>

The packet-oriented transport approach used in the advanced digital television (ADTV) system for terrestrial HDTV broadcast is described. ADTV achieves robust HDTV delivery on terrestrial simulcast channels via MPEG video compression, prioritization of MPEG data, and 'cell-relay' type packet transport in conjunction with a two-tier physical transmission scheme. General design issues relevant to the development of the proposed transport protocol are discussed. ADTV's prioritization algorithm for partitioning MPEG-encoded video into high-priority (HP) and standard-priority (SP) bit streams is outlined. The data transport format supporting these prioritized compressed video bit streams is described. The three principal sublayers of the ADTV transport protocol are discussed in terms of specific functions, impact of system performance, and hardware implementation factors. A proof-of-concept simulation model that incorporates transport encoding and decoding functionality is outlined, and performance evaluation results are given for illustrative transmission scenarios.>

A comparative evaluation of dynamic time-division multiple access (TDMA) and spread-spectrum packet code-division multiple access (CDMA) approaches to multiple access in an integrated voice/data personal communications network (PCN) environment are presented. After briefly outlining a cellular packet-switching architecture for voice/data PCN systems, dynamic TDMA and packet CDMA protocols appropriate for such traffic scenarios are described. Simulation-based network models which have been developed for performance evaluation of these competing access techniques are then outlined. These models are exercised with example integrated voice/data traffic models to obtain comparative system performance measures such as channel utilization, voice blocking probability, and data delay. Operating points based on typical performance constraints such as voice blocking probability 0.01 (for TDMA), voice packet loss rate 10/sup -3/ (for CDMA), and data delay 250 ms are obtained, and results are presented.>

An analysis of unslotted random-access direct-sequence spread-spectrum multiple-access (DS/SSMA) channels with block forward error correction (FEC) coding is presented. Extending a methodology that was introduced in an earlier paper on unslotted packet code-division multiple access (CDMA) without coding, a procedure for calculating the error probability of an L-bit packet in the variable message length, FEC-coded, DS/SSMA environment is described. This procedure is then used in conjunction with appropriate flow equilibrium traffic models to compute channel throughput. Using BCH block coding as an example, the analytical model is exercised to obtain throughput versus channel traffic curves over a range of code rates, leading to an assessment of maximum achievable throughput and the associated optimum FEC code rate. The results show that the use of block FEC coding provides a significant improvement in the bandwidth-normalized channel throughput (utilization), approaching values competitive with those for comparable narrowband ALOHA channels.>

The authors consider modeling methodologies of variable-bit-rate video sources for performance evaluation of integrated networks. They consider an example in which compressed video is transmitted over a local area network carrying both video and data. A new modeling methodology called transform-expand-sample (TES) which is a method for generating autocorrelated variates, is used. TES is a nonparametric method which can accurately capture the histogram and approximate autocorrelation function of any data set. The TES model was used to drive a network simulation for performance measures.>

Spread spectrum CDMA based cellular systems are currently under consideration for a range of personal communication network (PCN) applications. While CDMA has been shown to offer significant capacity advantage over narrowband TDMA alternatives, these gains are very sensitive to the propagation loss coefficient gamma . Most analytical studies for CDMA system capacity are based on an assumption of propagation loss coefficient gamma =4, consistent with large cell sizes associated with present day cellular radio. However, in microcellular (i.e., cell sizes with radius approximately 100's of meters) environments, the path loss coefficient can vary in the range of 2 to 4. In addition, attenuation measurements using low antenna heights for microcells have indicated that the path loss coefficient may not be a constant, increasing with distance between the base station and the subscriber terminal. A distance-dependent model for the propagation loss coefficient gamma is considered and used to compute the capacity of a cellular packet CDMA system operating in an integrated voice/data PCN environment. System-level performance measures such as channel utilization, voice packet loss probability and data delay are derived from a detailed channel event based simulation model, and compared for alternative scenarios with constant gamma and distance dependent gamma .>

An investigation of statistically based approaches to the design of buffer control algorithms for interfacing a compressed digital video source to a constant rate channel is presented. As a first step to a quantitative methodology for the design and evaluation of adaptive buffer control algorithms, a detailed statistical characterization of the various encoding modes of an example broadcast quality intra/interframe differential-pulse-code-modulation (DPCM) algorithm is obtained from extensive simulation. The statistics presented include simple measures such as the overall encoded rate distributions for each encoding mode and more detailed intermode and intramode statistics. Two specific adaptive mode control algorithms are proposed. Using simulation over a large sea of representative images, the performance of the proposed statistically based algorithms is compared to that of a conventional buffer-level-based control heuristic which does not require source characterization.>

A simulation study is presented of the dynamic behavior of ALOHA-type protocols used in interactive-data very small aperture terminal (VSAT) networks. A simulation model for quantitative evaluation of random access channel stability in terms of the transient response to a traffic overload pulse is described, and the usefulness of a single performance measure called backlog fall time is established. The variation of backlog fall time with selectable protocol parameters (such as average retransmission delay for nonadaptive systems or retransmission backoff policy parameters for adaptive systems) is investigated, for example, for ALOHA and selective reject ALOHA channels. A methodology for joint optimization of steady-state and dynamic performance based on obtaining contours of average delay versus backlog fall time over the variation range of selectable protocol parameters is outlined and demonstrated. It is shown that, generally speaking, a suitable operating point is easily identified from these contours because they exhibit a characteristic knee region in which both delay and fall time are close to the minimum values that can be independently obtained.>

An analytical investigation is presented of generalized retransmission backoff policies for slotted-ALOHA random-access channels. Backoff techniques, of which the well-known exponential backoff is a special case, are based on adaptation of average retransmission delay as a function of the number of collisions experienced by each message accessing the contention channel. An analytical model applicable to slotted-ALOHA channels using general backoff functions has been developed and used to assess the performance advantages offered by either exponential backoff or alternative policies motivated by heuristic considerations. Numerical results for a sample satellite channel scenario are presented, demonstrating that the use of appropriate backoff policies can result in significant improvements in stable throughput-delay characteristics relative to those of nonadaptive systems.>

An investigation is made of the stability and dynamic behavior of ALOHA-type protocols used in very small aperture terminal (VSAT) networks that support bursty, interactive data sources. A simulation model has been developed for dynamic performance analysis of unslotted ALOHA and selective reject (SREJ) ALOHA channels and used to obtain quantitative measures of stability in terms of transient response to pulsed traffic overload. For a typical 56-kb/s interactive station VSAT channel, the dependence is examined of pulse response (summarized in terms of a quantity called the backlog fall time) on selectable protocol parameters such as average transmission delays for nonadaptive systems or backoff policy parameters for adaptive systems. These results are augmented with average delay simulations, to obtain characteristic delay versus fall-time contours that permit the designer to select protocol parameters yielding the optimum combination of nominal steady-state and dynamic performance. For the ALOHA and SREJ-ALOHA channels under consideration, a surprisingly well-defined optimum region (in the form of a knee in the delay versus fall-time curves) is generally found to exist, underscoring the value of the joint steady-state/dynamic response design methodology proposed.>

The problem of designing shared access packet-transport-based transmission systems for compressed video signals is studied. The feasibility of using conventional link-level and transport-level protocol services to transmit compressed video is examined by focusing on two practically important scenarios for compressed video transmission: (1) multipoint-to-multipoint video transmission using a 200 Mb/s implicit token passing (ITP) fiber-optic local area network (LAN); and (2) point-to-multipoint broadcast video distribution using a 90 Mb/s packet time-division multiplexing (packet-TDM) direct-broadcast satellite channel. To evaluate the performance of such shared-access broadband packet video systems accurate simulation models were developed that were driven by realistic 'broadcast quality' compressed video sources for the ITP-LAN and packet-TDM systems. The models were used to determine design tradeoffs between channel throughput, video quality (measured by clipping probability), and the transport-level and media-access-level protocol features and parameters implemented in the packet video network interface unit.>

Simulation models are described for the performance evaluation of satellite multiple-access protocols used in VSAT (very small aperture terminal) data networks. Since the design of VSAT-based wide-area networks for interactive applications is critically dependent on the delay-throughput characteristics of the channel access protocol used, detailed simulation models have been developed to augment analytical performance-evaluation methods. Simulation models for several candidate VSAT access protocols, including random-access techniques such as Aloha, selective reject Aloha and slotted Aloha and demand-assignment multiple-access based techniques, are discussed in terms of traffic-source models, access algorithms, and performance-evaluation criteria. Numerical results for a realistic interactive data VSAT network environment are presented.>

An analytical performance evaluation model is presented for selective reject (SREJ) ALOHA, a recently proposed high-capacity protocol for unslotted channels with high propagation delay. SREJALOHA approaches the theoretical asynchronous multiaccess throughput limit of 0.368 for fixed or variable length message traffic, using subpacketization of messages in conjunction with a selective reject retransmission policy. A finite user flow equilibrium-based model for SREJ-ALOHA with variable length messages is derived and used to characterize stability, throughput, and delay. Particular attention is given to the development of a stability analysis methodology for unslotted ALOHA-type protocols with variable length transmissions. Numerical results based on a satellite channel example are presented to demonstrate the significant performance advantages offered by SREJ-ALOHA, even after accounting for realistic subpacket overheads.

A finite user model for stability analysis of announced retransmission random access (ARRA) systems is presented. It is shown that as in other ALOHA-type random access protocols, bistable behavior and saturation can occur unless a suitably long retransmission delay is used. The higher expected delay associated with an unconditionally stable nonadaptive system can be avoided by adaptively varying retransmission parameters with system backlog. Optimal retransmission control (ORC) policies of this type are derived and evaluated.

A new contention-based broadcast multiaccess protocol called announced retransmission random access (ARRA) is introduced and analyzed in this paper. The proposed class of protocols increases the capacity of a slotted random access channel by adding a small amount of potentially useful control information to every message transmission. In particular, the ARRA technique requires users to announce the intended location of their potential retransmission slot (to be used in the event of a collision) over a low-rate announcement subchannel, so that conflicts between new and retransmitted messages can be prevented. The proposed protocols provide significant capacity improvements over conventional contention protocols, without sacrificing the simplicity and operational convenience associated with random access. It is shown that with a modest amount of processing and storage at the terminals, capacities in the range of 0.5-0.6 can be achieved, as compared with 0.37 for slotted ALOHA and 0.4-0.5 for the more efficient tree algorithm-based collision resolution algorithms (CRA's). Specifically, the basic ARRA protocol, in which announced retransmissions take place unconditionally, has a capacity of about 0.53. An improved algorithm (called extended ARRA), which involves aborting predictably unsuccessful retransmissions, has a capacity of 0.6. In addition, because of delay considerations, the proposed ARRA protocols are well suited for operation on satellite channels.

The throughput of slotted ALOHA systems with multipacket message transmissions is evaluated. The conventional strategy of retransmitting the entire message when collisions occur results in a reduction in throughput when compared with the single packet case. However, it is proved analytically that the use of ARQ-type retransmission protocols can provide significant performance improvement. In particular, it is shown that a simple go-back-N(GBN) protocol can offer a modest throughput increase, while a selective reject (SREJ) strategy provides exactly the same throughput as a single packet slotted ALOHA system, irrespective of message length statistics. These results motivate a new scheme (referred to as unslotted selective reject (SREJ) ALOHA) for increasing the throughput of unsynchronized random access channels. It is demonstrated that unslotted SREJ ALOHA with optimized parameters can achieve typical maximum throughputs (after accounting for all overheads) at least 25-40 percent higher than conventional pure ALOHA.

Simple techniques for unauthorized descrambling of a random line inversion scrambled television signal are proposed. An analysis of the complexity of the proposed descrambling process is presented and it is shown that high-quality estimates of moderately long encoding sequences can be obtained with reasonable memory and delay. An improvement to the basic scheme which substantially reduces memory requirements with only a slight increase in delay is also evaluated. Finally, it is shown that varying code lengths among a finite number of alternatives does not significantly increase descrambler complexity.

Analytical techniques for performance evaluation of synchronous random access packet switching in code division multiple access (CDMA) systems are presented. Steady-state throughput characteristics using several packet generation models are obtained. A number of example random access CDMA systems are compared in terms of their throughput versus offered traffic and utilization-delay characteristics. Numerical results indicate that appropriate use of multiaccess coding can provide utilization-delay characteristics superior to that of ALOHA. System stability is evaluated using a general finite user model, and the dynamic behavior of some example random access CDMA schemes is investigated.

The performance of orthogonal and quasi-orthogonal tree codes is considered for incoherent and coherent systems. In addition, several alternative receiver structures using hard quantizing and "greatest-of" detection are treated. Upper bounds on error probability are derived and are used as a basis for system comparison. An application of quasi-orthogonal tree codes to a multiple access environment is presented as an example and discussed.