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We describe and evaluate a new suite of protocols for the delivery of information from each node to every other node in a multi-hop ad hoc wireless network. The protocol suite is integrated with random linear network coding to mitigate the effects of packet losses due to noise and time-varying fading on the radio links. The suite includes a practical three-mode protocol for packet-by-packet selection of the channel code and modulation based on control information from statistics derived by the nodes themselves. Our performance metric is energy efficiency, and our emphasis is on robust performance in practical networks for which optimization is not feasible. [ABSTRACT FROM AUTHOR]

In this paper, we review the key figures of merit to assess the performance of advanced random access (RA) schemes exploiting physical layer coding, repetitions, and collision resolution techniques. We then investigate RA modeling aspects and their impact on the figures of merit for the exemplary advanced RA schemes: contention resolution diversity slotted ALOHA (CRDSA), irregular repetition slotted ALOHA (IRSA), coded slotted ALOHA (CSA) and enhanced spread-spectrum ALOHA (E-SSA). We show that the typical simplifications of the reception model when used to optimize RA schemes lead to inaccurate findings, both in terms of parameter optimization and figures of merit, such as the packet loss ratio (PLR) and throughput. We also derive a generic RA energy efficiency model able to compare the schemes in terms of the energy required to transmit a packet. The combination of achievable RA throughput at the target PLR and energy efficiency, for the same average user power investment per frame and occupied bandwidth, shows that E-SSA, which is an unslotted scheme, provides the best overall performance, while, in terms of the slotted schemes, CRDSA outperforms the more elaborated IRSA and CSA. This surprising results are due to the fact that the IRSA and CSA optimization has so far been performed using RA channel models that are not accurately reflecting the physical layer receiver behavior. We conclude by providing insights on how to include more accurate reception models in the IRSA and CSA design and optimization. [ABSTRACT FROM PUBLISHER]

There is an urgent need for substantially enhancing random access schemes to be able to support native massive machine-to-machine communications in the next generation wireless systems. Lately, contention resolution diversity slotted ALOHA (CRDSA) and its variants have contributed to a drastic improvement in throughput compared with the traditional slotted ALOHA protocols. In this paper, we further enhance the CRDSA flexibility and performance along different physical layer dimensions. Based on simulation results, we show that there is still room for improvement in terms of throughput and, using heuristic methods, we find an optimized distribution of the packets power that is entering the demodulator for an additive white Gaussian noise channel. Applying this power distribution, which is easy to implement at the terminal in open loop and in a decentralized manner, is shown to achieve significantly enhanced performance. By introducing multi-frequency CRDSA, a drastic reduction in the terminal’s peak power requirement is attained. Furthermore, we investigate spread spectrum CRDSA, which improves the robustness of the scheme against irresolvable packet collisions allowing for enhanced energy efficiency. To support the work presented herein, we have developed a computationally efficient yet accurate methodology for estimating CRDSA performance, a novel approximate centrally coordinated CRDSA achievable sum rate analysis accounting for modulation and forward error correction constraints, as well as a more accurate loop probability analysis. [ABSTRACT FROM PUBLISHER]

Fountain coding is a valuable tool for multicast transmissions in half-duplex packet radio networks if it is supported by an adaptive transmission protocol that compensates for fluctuations in the propagation losses on the radio links. Our goal is to devise low-complexity packet-by-packet adaptive modulation and channel coding protocols that provide high throughput for a broad range of multicast channels and operate with limited feedback and no knowledge of the fade levels or even the probability distribution for the fading on the radio links. The only control information for the protocols is derived from simple receiver statistics that are obtained by the demodulators and channel decoders in the receiving radios. Performance results are presented for practical fountain coding with low-complexity adaptive modulation and channel coding. Results on ideal fountain coding with adaptive transmission based on hypothetical perfect channel-state information are employed as performance benchmarks. [ABSTRACT FROM AUTHOR]

Analytical methods are provided for use in performance evaluations, tradeoff studies, and preliminary designs of adaptive modulation and coding protocols that obtain their control information from practical sources. Our approach employs capacity limits and Markov chain models of fading to give analytical results on the throughput achieved by adaptive transmission in packet radio systems. The analytical methods replace simulations of the modem, iterative decoder, and time-varying fading channel. The amount of processor time required for protocol performance evaluations is decreased by several orders of magnitude. [ABSTRACT FROM PUBLISHER]

For time-varying fading, adaptation of the physical-layer channel code greatly enhances the throughput of a multicast transmission system, even if the system employs an application-layer fountain code for packet erasure correction. Performance evaluations for adaptive channel-coding protocols typically require extensive simulations of the fading channel, the iterative decoding process, and the generation of the control information that is used for the adaptation of the channel code. We present a method for the evaluation of such protocols that avoids a substantial amount of the computation required by traditional simulations, and we demonstrate that some important performance measures can be evaluated entirely by analysis. By the use of capacity-achieving channel codes in our investigation, we also decouple the performance of the adaptive coding protocol from the performance of a particular type of channel code and decoding method, which facilitates comparisons among different protocols. The application of our methods is demonstrated for two adaptive channel-coding protocols that are candidates for fountain-coded multicast transmission in packet radio systems. [ABSTRACT FROM PUBLISHER]

In this paper, we address various aspects of packet data transmission in terrestrial trunked radio (TETRA) networks giving particular emphasis on the performance of applications transmitting small messages between a number of radio terminals and a fixed server. The utilization of such applications is constantly increasing in public safety networks and so does the need to dimension and configure TETRA networks to meet their reliability, delay and loss requirements. We present simulation results for a variety of practical scenarios such as loading the packet data channel (PDCH) with different mobile station (MS) populations engaged in transmitting packet data (PD) at various rates. We quantify key performance indicators of the PDCH such as packet delay and packet loss rate. Furthermore, we provide an in-depth analysis of the impact of non-PD MS population—i.e., the loading of main control channel (MCCH)—on the average packet delay. Our results provide an insight of how network loading parameters interact with the protocol performance and may help in dimensioning and planning TETRA networks, and in particular, in determining the number of PD terminals that can be supported in a cell for a given traffic profile and QoS requirements. [ABSTRACT FROM AUTHOR]

This paper evaluates the packet data performance of Internet Protocol (IP) over TErrestrial Trunked RAdio (TETRA), focusing on the effects of packet size and message transmission rate. We performed a range of simulations to evaluate the average packet delay and packet failure probability corresponding to IP datagram transmission on the Packet Data CHannel (PDCH) in a TETRA cell. We employed IP packets of different sizes and we considered a number of different scenarios with respect to the number of users in a cell and datagram transmission rate. The simulation results offer TETRA operators the means for facilitating network planning and optimization of datagram-based services. Apart from the simulation, we also conducted an extensive measurement survey on the performance of IP over the TETRA PDCH. In particular, we carried out measurements in a client–server communication scenario with different datagram sizes and typical user mobility and radio propagation conditions. Measurement results include throughput, delay jitter, percentage of lost datagrams and out-of-order datagrams. The results obtained through our measurement survey can extent and enhance the simulation results because they are more realistic and consider user mobility as well as channel impairments. [ABSTRACT FROM AUTHOR]

This paper studies the impact of voice traffic on the performance of packet data (PD) transmission in terrestrial trunked radio (TETRA) networks. Particular emphasis is given on the performance of applications characterized by occasionally transmitting small messages between a number of radio terminals and a fixed server while these terminals are also conducting group voice calls. Such applications are constantly increasing in TETRA networks, and so does the need to dimension and configure these networks to meet their reliability, delay, and loss requirements. However, the impact of group voice calls on PD transmission has not been sufficiently studied, and this constitutes the main motivation of our performance analysis. We present simulation results for a variety of practical scenarios, and we discuss how key performance indicators such as PD delay and loss rate are affected by the group call activity of the radio terminals. Our results can help in dimensioning and planning TETRA networks, particularly in determining the number of PD terminals that can be supported in a cell for a given traffic profile and for quality-of- service requirements. [ABSTRACT FROM AUTHOR]

Modern communication systems are required to provide services based on high data rates burst-mode packet-data transmission, capable of operating at very low SNR conditions. Turbo codes enable the operation at low SNR, close to the Shannon limit. However, carrier frequency and phase synchronization, needed for optimal coherent performance of the receiver, still remains a problem in low SNR and short bursts conditions. This paper proposes a new carrier synchronization method, the Maximum-Likelihood Iterative-Soft-Decision-Directed (ML-ISDD), which uses the turbo-decoder soft decisions to improve the carrier synchronization performance at low SNR values. The ML-ISDD method operates iteratively and jointly with the turbo decoder, enhancing both the turbo-decoder and the synchronization performance. The ML-ISDD method has been shown by simulation to significantly increase the allowed initial frequency and phase uncertainty region, thus allowing the use of very short training sequences for initial carrier synchronization. [ABSTRACT FROM AUTHOR]

In this paper we design novel generalized likelihood ratio test (GLRT)-type packet-data detectors for general multi-access/multiuser digital communication systems and we develop analytical performance evaluation tools for finite data packet sizes. For the known channel case, we derive a coherent GLRT packet-data detector while for the unknown channel case we derive both a coherent pilot assisted GLRT packet-data detector and a differential phase-shift-keying (DPSK) GLRT packet-data detector. Efficient suboptimum implementations of the above schemes that exhibit complexity linear in the packet size are also considered. Simulation studies evaluate the performance of the proposed schemes in the context of packet-data code-division multiple access (CDMA) communications. [ABSTRACT FROM AUTHOR]

Cognitive radios that are employed in a network with dynamic frequency assignments must operate efficiently in the presence of uncertainties and variations in the propagation characteristics of the network's communication links. A low-complexity adaptive transmission protocol is described and evaluated for use in cognitive radio networks whose links have unknown and possibly time-varying propagation losses as a result of such phenomena as slow fading or variations in shadowing. The cognitive radios are required to derive only simple statistics in the receivers in order to provide the information that is needed by our protocol; no estimates or measurements of received power or channel gain are used. The protocol's primary mechanism for responding to changes in propagation loss is to adjust the modulation and coding. Because of disruptions that can be caused by higher levels of interference to other radios in the network, the transmitter power is increased only if the most powerful combination of coding and modulation is inadequate. We employ finite-state Markov models for slowly varying channels, and we demonstrate that for such channels our protocol performs nearly as well as an ideal protocol that is told the exact value of the propagation loss for each packet transmission. Thus, the additional complexity that is required to enable cognitive radios to obtain precise channel-gain estimates is not justified and would lead to only negligible improvement in throughput. The throughput of our adaptive transmission protocol is compared with an upper bound that is derived from information theory for a hypothetical ideal protocol that is given perfect channel-state information, and some preliminary results on learning the adaptation decision intervals are included. [ABSTRACT FROM AUTHOR]

This paper investigates the benefit of adaptive modulation based on channel state information (CSI) in direct-sequence/ code-division multiple-access (DS/CDMA) multihop packet radio networks. By exploiting varying channel conditions, adaptive modulation can be used in ad hoc networks to provide upper layers with higher capacity links over which to relay traffic. Using the a-stable interference model, the distribution of the signal-to-interference ratio (SIR) is obtained for a slotted system of randomly, uniformly distributed nodes using multilevel coherent modulation schemes. Performance is evaluated in terms of the information efficiency, which is a new progress-related measure for multihop networks. Three types of adaptivity are analyzed, differing in the level of CSI available: 1) full knowledge of the SIR at the receiver; 2) knowledge of only the signal attenuation due to fading; and 3) knowledge of only the slow fading component of the signal attenuation. The effect of imperfect channel information is also investigated. Sample results are given for interference-limited networks experiencing fourth-power path loss with distance, Ricean fading, and lognormal shadowing. [ABSTRACT FROM AUTHOR]

This paper analyzes the benefit of adaptive routing based on knowledge of the channel state information in multihop, ad hoc wireless networks that use direct-sequence code-division multiple access. Cross-layer, channel-adaptive routing exploits the inherent spatial diversity of multihop wireless networks to select links with favorable channel conditions. The information efficiency, an extension of a previously used measure called expected progress, is used to evaluate performance. Results show that, combined with adaptive modulation, adaptive routing can improve performance in ad hoc networks by a factor of four to five in channels with Rayleigh fading and lognormal shadowing. The lack of position information in the routing decision would reduce performance by 25%. New approaches to channel-adaptive routing that enable rapid adaptivity to channel conditions are discussed. [ABSTRACT FROM AUTHOR]

In this part of the paper, we investigate the throughput bound of CDMA unslotted ALOHA system with a hybrid type II automatic-repeat-request (ARQ) implemented using rate compatible punctured turbo codes. The system under consideration can be regarded as an enhanced version of the classical CDMA packet radio network. When applying a hybrid type-II ARQ, the probability of packet success and packet length is generally different from attempt to attempt. Since the conventional analytical model, customarily employed for the ALOHA system with pure or hybrid type-I ARQ, cannot be applied for this case, we introduce an expanded analytical model, which can be regarded as a network of queues, and apply Jackson and Burke's theorems to simplify the analysis. When using code combining, soft-decision decoding is usually considered, in which case, it is very difficult to find the actual probability of packet success, especially when the number of terminals changes during a packet time duration. Hence, instead of finding the actual probability of packet success, we evaluate a lower and an upper bound. Together with the expanded model and the packet success probability bound, we obtain the corresponding throughput bounds by iteratively searching the steady-state arrival rate, λ, and the average departure rate, μ, for given composite offered traffic, G=λ/μ. The results show that significant improvement in throughput can be achieved by exploiting high coding gain of turbo codes and channel utilization efficiency of incremental redundancy retransmission protocol. [ABSTRACT FROM PUBLISHER]

The Internet has been evolving from its origins in the early 1970s, based on work sponsored by the U.S. Defense Advanced Research Projects Agency. While the basic design was known in 1973 and first published in 1974 and the system essentially deployed in the academic and military communities on January 1, 1983, much has happened in the intervening 20 years. The first commercial Internet services emerged in 1989 after the interconnection of the Internet to commercial e-mail services. By 1993, commercial versions of the World Wide Web had appeared, and by 2003, voice over IP service was growing rapidly, after its first commercial introduction around 1995 (See Vocaltec: http://www. vocaltec.com/html/about/company.shtml). The Internet of the future will be shaped by the tectonic forces of regulation, commercialization, technological change, and a wide range of policy concerns expressed at local, national, regional and international levels. In this paper, the effect of these forces is considered and an attempt made to project their effects into the future. [ABSTRACT FROM AUTHOR]

A common problem in a packet radio network (PRN) environment is to construct a multicasting network from a single source to a set of remote destinations which minimizes the number of transmissions. This problem is known to be NP-complete, thus computing an optimal solution may be infeasible for sizable networks. This paper provides two alternative solutions to this problem. The first is a heuristic algorithm which iteratively builds a spanning tree from the destinations to the source. A second solution, included for comparative purposes, is based on the Hopfield neural network whose dynamics are governed by a motion equation and a set of constraints. Both solutions are tested on a variety of instances against an optimal algorithm. Results show the approaches form good solutions (the number of transmissions is within about 3% of the optimum) and run in a fraction of the time required to form the optimal solution. [ABSTRACT FROM AUTHOR]

Network protocols are often analyzed using simulations. We demonstrate how to extend such simulations to check propositions expressing safety properties of network event traces in an extended form of linear temporal logic. Our technique uses the NS simulator together with a component of the MaC system to provide a uniform framework. We demonstrate its effectiveness by analyzing simulations of the Ad Hoc On-Demand Distance Vector (AODV) routing protocol for packet radio networks. Our analysis finds violations of significant properties and we discuss the faults that cause them. Novel aspects of our approach include modest integration costs with other simulation objectives such as performance evaluation, greatly increased flexibility in specifying properties to be checked and techniques for analyzing complex traces of alarms raised by the monitoring software. [ABSTRACT FROM AUTHOR]

The first part of this paper presents a novel scheme for handover provisioning in wireless asynchronous transfer mode (W-ATM) networks based on in-band signaling. First, the network architecture and principles of in-band signaling are described, discussing advantages and interaction with other procedures and signaling techniques. Then, loss-free protocols for the handover procedures are presented and compared with existing proposals. The second part of the paper is devoted to performance analysis of the handover procedures. A general methodology for evaluating handover delays and required buffer capacity is introduced and exemplified for one of the protocols introduced before. Numerical results give insight into the handover procedure characteristics and are compared with estimates provided by detailed discrete event simulations for validation purposes. Finally, additional simulation results are presented for parallel, concurrent handovers, evaluating the requirements posed to the network by different handover procedures [ABSTRACT FROM PUBLISHER]

This paper proposes a new access protocol, time division multiple access with parallel transmission (TDMA/PT), for packet radio networks. With this protocol, transmission power is controlled at terminals and more than one throughput performance can be obtained, even when a single channel is used for an entire area. With TDMA/PT, an ID number is assigned to each terminal and transmission with TDMA takes place in the order of ID numbers. Each terminal examines the control field and address field of a packet transmitted from another terminal that has higher priority at the slot. When it is clear that a collision will not occur, the terminal transmits a packet simultaneously with another terminal having higher priority. Moreover, each terminal can be informed of the number of slots necessary for the message transmitted from another terminal; therefore, we can expect fair performance even if the load on each terminal is not equal and/or propagation delay is big. Because performance improvement depends on ID assignment, effective ID assignment method is also studied. © 1997 Scripta Technica, Inc. Electron Comm Jpn Pt 1, 80 (5): 1–10, 1996 [ABSTRACT FROM AUTHOR]

New distributed protocols for channel access and adaptive spreading are described and evaluated for use in direct-sequence spread-spectrum packet radio networks that have no central controllers, access points, or base stations. The protocols rely on demodulator statistics to provide the information needed for efficient sharing of the frequency band and compensation for changes in the communications environment in a network of half-duplex radios for which simultaneous two-way communications is not possible. [ABSTRACT FROM AUTHOR]