Dipankar Raychaudhuri

Overview of the ORBIT radio grid testbed for evaluation of next-generation wireless network protocols

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.

Performance Analysis of Random Access Packet-Switched Code Division Multiple Access Systems

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.

Data link control protocols for wireless ATM access channels

Describes data link control procedures for wireless ATM access channels based on a dynamic TDMA/TDD frame-work. The system provides integrated ATM services including available bit-rate (ABR) data and constant/variable bit-rate (CBR/VBR) voice or video through the addition of wireless-specific medium access control (MAC) and data link control (DLC) protocol layers between the physical and ATM network layers. The purpose of the DLC protocol layer is to insulate the ATM network layer from wireless channel impairments by selective retransmission of erroneous or lost cells before they are released to the ATM layer. The DLC methods described exploit the on-demand ABR burst transmission capability of the dynamic TDMA channel to retransmit unacknowledged cells in available slots not allocated to service data. Specific error recovery procedures are outlined for both (asynchronous) ABR and (isochronous) CBR services. For ABR, the DLC operation follows a group ACK/NACK based selective reject (SREJ) procedure on a burst-burst basis, without time limits for completion. For CBR, the retransmission procedure is constrained to complete within a specified time interval, so that isochronous delivery of cells to the ATM layer can be maintained. The proposed protocols have been validated using a software emulator which incorporates a choice of radio channel models, dynamic TDMA/TDD MAC, and ABR or CBR DLC. Numerical results show that the data link control protocols under consideration can significantly improve wireless ATM service quality over impaired radio channels for both packer data ABR and stream type CBR virtual circuits.

Packet CDMA versus dynamic TDMA for multiple access in an integrated voice/data PCN

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. >

Emerging Wireless Technologies and the Future Mobile Internet

This book provides a preview of emerging wireless technologies and their architectural impact on the future mobile Internet. The reader will find an overview of architectural considerations for the mobile Internet, along with more detailed technical discussion of new protocol concepts currently being considered at the research stage. The first chapter starts with a discussion of anticipated mobile/wireless usage scenarios, leading to an identification of new protocol features for the future Internet. This is followed by several chapters that provide in-depth coverage of next-generation wireless standards, ad hoc and mesh network protocols, opportunistic delivery and delay tolerant networks, sensor network architectures and protocols, cognitive radio networks, vehicular networks, security and privacy, and experimental systems for future Internet research. Each of these contributed chapters includes a discussion of new networking requirements for the wireless scenario under consideration, architectural concepts, and specific protocol designs, many still at research stage.

An UPC-based traffic control framework for ATM networks

Future broadband networks have to support new bandwidth intensive applications that will place stringent requirements on the network in terms of the QOS (quality of service) they receive. In spite of the the promise of abundant bandwidth, ATM would not be able to support such services unless adequate control is exercised within the network. In this paper, we propose a traffic control framework for ATM, that would provide guarantees on QOS for these new applications. Central to this framework is a traffic classification scheme that maps applications to specific traffic classes based on their QOS requirements and the statistical characteristics of their traffic, and a set of controls layered in time based on the time constant of events they control. The proposed traffic control framework can work across LANs, MANs, WANs, private and public networks in a seamless manner. The controls are designed to scale with distance, traffic volume and traffic type. A notable feature of this framework is that it depends on a minimum set of control functions to handle a wide range of traffic types. Further, it uses parameters for control that have already been standardized by ITU and the ATM Forum.

Statistical multiplexing of VBR MPEG compressed video on ATM networks

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.<<ETX>>

Bandwidth renegotiation for VBR video over ATM networks

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 sources 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.

Statistically based buffer control policies for constant rate transmission of compressed digital video

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. >

Error concealment algorithms for robust decoding of MPEG compressed video

Abstract This paper provides some research results on the topic of error resilience for robust decoding of MPEG (Moving Picture Experts Group) compressed video. It introduces and characterizes the performance of a general class of error concealment algorithms. Such receiver-based error concealment techniques are essential for many practical video transmission scenarios such as terrestrial HDTV broadcasting, packet network based teleconferencing/multimedia, and digital SDTV/HDTV delivery via ATM (asynchronous transfer mode). Error concealment is intended to ameliorate the impact of channel impairments (i.e., bit-errors in noisy channels, or cell-loss in ATM networks) by utilizing available picture redundancy to provide a subjectively acceptable rendition of affected picture regions. The concealment process must be supported by an appropriate transport format which helps to identify image pixel regions which correspond to lost or damaged data. Once the image regions (i.e., macroblocks, slices, etc.) to be concealed are identified, a combination of spatial and temporal replacement techniques may be applied to fill in lost picture elements. A specific class of spatio-temporal error concealment algorithms for MPEG video is described and alternative realizations are compared via detailed end-to-end simulations for both one- or two-tier transmission media. Several algorithm enhancements based on directional interpolation, ‘I-picture motion vectors’, and use of MPEG-2 ‘scalability’ features are also presented. In each case, achievable performance improvements are estimated via simulation. Overall, these results demonstrate that the proposed class of error concealment algorithms provides significant robustness for MPEG video delivery in the presence of channel impairments, permitting useful operation at ATM cell-loss rates in the region of 10 −4 to 10 −3 and 10 −2 to 10 −1 for one- and two-tier transmission scenarios, respectively.