This paper proposes a multipacket-per-slot reservation-based random access protocol with multiuser detection (MD) and automatic repeat request (ARQ), called MPRMD, and analyzes its performance by computer simulations. In MPRMD, before data packet (DP) transmission, a user terminal (UT) transmits a small access request packet (AP) that is composed of an orthogonal preamble sequence and a UT identifier (UT-ID) in a randomly selected minislot during a short dedicated period. Even when several APs collide, a base station (BS) distinguishes them by matched filtering against the preamble part and then extracts the UT-IDs after separating each AP by MD. If the APs are not successfully detected, a small number of minislots are additionally arranged to retransmit them. Thus, by using MD under AP crowded conditions, BS can maximally detect the access requests in a short period, which results in reducing the overhead. Furthermore, in the assignment of a slot, BS intentionally assigns one slot to multiple UTs in order to enhance the efficiency and separates UT's DPs by MD. Since MPRMD can detect a multitude of access requests by utilizing MD in the short period and efficiently assign the slot to separable DPs by MD, it can enhance the system throughput. Computer simulations are conducted to demonstrate the effectiveness of MPRMD. It is shown that the maximum throughputs of MPRMD with the average SNR of 30dB reach 1.4 and 1.7 packets/slot when a data packet is 10 times and 50 times as long as a control packet, respectively.

Energy-efficient transmission scheme is very essential for Wireless Personal Area Networks (WPNs) for maximizing the lifetime of energy-constrained wireless devices and assuring the required QoS in the actual physical transmission at each allocated TDMA time slot. We therefore propose the minimum energy (ME) criterion based adaptive transmission scheme which determines the optimum combination of transmit power, physical data rate and fragment size required to simultaneously minimize the energy consumption and satisfy the required QoS in each assigned time duration. The improved performances offered by the proposed algorithm are demonstrated via computer simulation in terms of throughput and energy consumption.

To analyze the saturation performance of IEEE 802.11 DCF, several discrete-time Markov chain models for a station and sets of channel equations have been introduced. We take into account a frame retry limit, freezing of backoff counter, and the dependence of backoff procedure on the previous channel status all together. Our method is simple even though it is accurate under the assumption of no consecutive transmissions over the shared channel.

We aim to establish a highly efficient transmitting power control (TPC) scheme suitable for the reverse link of high-speed CDMA packet communication systems. Reservation-based access is assumed to be used for packet transmission in the reverse link. First, we describe a hybrid TPC that we created to cope with average interference changes. The target receiving power in the hybrid TPC is set according to the interference averaged over a comparatively long period of time. We show, using experiments on our high-speed packet communication experimental system, that hybrid TPC can effectively reduce transmission power consumption and PER compared with basic receiving power based TPC. Furthermore, we need to change the transmitting power according to the instantaneous interference to cope with instantaneous interference changes slot by slot. However, in a high-speed packet communication system, the interference level can change dramatically in a very short period of time. The TPC of cdma2000 or W-CDMA cannot efficiently cope with rapidly and greatly changing interference levels. Therefore, we created another two novel TPCs. Interference is divided in these TPCs into intra-cell and inter-cell interference. The supposed inter-cell interference level is changed according to the change in the probability distribution of the inter-cell interference, and the necessary transmitting power for a packet is calculated based on intra-cell allocation information and the supposed inter-cell interference level. Computer simulations show that, with the proposed TPCs, throughput can be increased by more than 200% compared with the type of TPC used in cdma2000 or W-CDMA, and the transmitting power consumption in a mobile host (MH) can also be vastly reduced.

We propose a p-persistent protocol for slow-frequency-hopped slotted random access networks, assuming that all the users know the number of backlog users in a slot. The proposed protocol delays new packet transmission until the number of users with a packet to be retransmitted decreases to the threshold or less. Performance of the proposed protocol is evaluated with a two-dimensional Markov chain for systems with finite population in terms of throughput efficiency and the average transmission delay. Numerical results show that the proposed protocol can achieve better performance by suppressing new packet transmission, compared to the conventional frequency-hopped slotted ALOHA. The optimum value of the threshold is also numerically derived.

Session length and group size are two most significant factors in achieving efficient scheduling for unicast and multicast traffic in single-hop wavelength division multiplexing (WDM) local area networks (LANs). This paper presents a hybrid protocol to schedule both unicast and multicast traffic in broadcast WDM networks. The protocol makes an important assumption that unicast traffic is the major portion of the overall traffic and is usually scheduled with a pre-allocation-based protocol. On the other hand, multicast traffic is a smaller portion of the overall traffic with multicast sessions and multicast groups, and is scheduled with a reservation-based protocol. The concept of multicast threshold, a function of random variables including the multicast session length and the multicast group size, is also proposed to partition the multicast traffic into two types. If the transmission threshold of a multicast request is larger than the multicast threshold, the request is handled with a reservation-based protocol. Otherwise, the multicast request is handled similar to unicast traffic; that is, each packet in the multicast session is replicated and sent to the unicast queues of destinations. The results show that the hybrid protocol can achieve better channel utilization efficiency and packet delay for unicast traffic under the multicast scenarios with moderate session length and group size. However, separate scheduling or broadcasting will be more suitable for a multicast scenario with very large session length and group size, which is not common on most realistic networks.

Channel-state-dependent (CSD) radio-resource scheduling algorithms for wireless message transport using a framed ALOHA-reservation access protocol are presented. In future wireless systems that provide Mbps-class high-speed wireless links using high frequencies, burst packet errors, which last a certain number of packets in time, would cause serious performance degradation. CSD resource scheduling algorithms utilize channel-state information for increasing overall throughput. These algorithms were comparatively evaluated in terms of average allocation plus transfer delay, average throughput, variance in throughput, and utilization of resources. Computer simulation results showed that the CSD mechanism has a good effect, especially on equal sharing (ES)-based algorithms, and also CSD-ES provides low allocation plus transfer delay, high average throughput, low variance in throughput, and efficient utilization of radio resources.

This paper proposes a new MAC protocol and physical channel control schemes for TDMA-TDD multi-slot packet channel. The goal of this study is to support both circuit-switched and packet-switched communications on the same resources and to enable high-speed packet transmission using a multi-slot packet channel. In the proposed channel control schemes, three points are taken into account; 1) effective sharing of time slots and frequencies with minimum impact on circuit communications, 2) compatibility with the existing access protocol and equipment, and 3) dynamic allocation of uplink and downlink slots. As for the MAC protocol, we adopt BRS (Block Reservation Scheme) and adaptive access control scheme to the proposed MAC protocol. In addition, to overcome the inherent disadvantage of TDD channels, packet scheduling and access randomizing control are newly proposed in this paper. The results of throughput and delay evaluations confirm that downlink capacity can be drastically enhanced by the dynamic allocation of uplink and downlink slots while corruption under heavy traffic loads is prevented by applying the adaptive traffic load control scheme.

This paper proposes the reserved idle signal multiple access with collision resolution (R-ISMA/CR) protocol, which can maintain stability without being influenced by the retransmission probability. The R-ISMA/CR makes use of the multi-idle signals at the base station and the manipulating counter at the terminals. The performance of this protocol is evaluated by using the maximum throughput analysis and simulations.

We propose two multipriority reservation protocols for wavelength division multiplexing (WDM) networks. The network architecture is a single-hop with control channel-based passive star topology. Each station is equipped with two pairs of laser and filter. One pair of laser and filter is always tuned to wavelength λ0 for control and the other pair of laser and filter can be tuned to any of data wavelengths, λ1, λ2, ..., λN. According to the access methods of the control channel, one protocol is called slotted ALOHA-based protocol and the other protocol is called TDM-based protocol. The two protocols have the following properties. First, each of them has its own priority control scheme which easily accommodates multipriority traffics. Second, they can be employed in the network with limited channels, i.e. the number of stations in the system is not restricted by the number of data channels. Third, they are conflict-free protocols. By using a reservation scheme and a distributed arbitration algorithm, channel collision and destination conflict can be avoided. For the performance point of view, the TDM-based protocol gives an optimal solution for the priority control. However it is less scalable than the slotted ALOHA-based protocol. The slotted ALOHA-based protocol also performs good priority control even though it is not an optimal solution. We analyze their performances using a discrete time Markov model and verify the results by simulation.

In this paper, an efficient voice/data integrated access algorithm for future personal communication networks (PCNs) is proposed and analyzed based on an equilibrium point analysis (EPA) method. A practical wireless communication channel may be impaired by noise and multipath distortion, and thus corrupted real-time packets have to recompete immediately in order to be transmitted within the stringent delay constraint. Also, real-time traffic users have to transmit their packets irrespective of the amount of non real-time data messages so that heavy non real-time traffic does not degrade the quality of real-time traffic. In the proposed algorithm, request subslots are distributed in the beginning of every slot to reduce access delay of real-time traffic. Moreover, slots are assigned to real-time traffic first and the remaining idle slots are assigned later to non real-time traffic by using the scheme of contention separation. We analyze the throughput and delay characteristics of this system based on an EPA mothod, and validate their performances by simulations. This scheme can support different quality of services (QoSs) imposed by different services efficiently and show good quality of real-time traffic, especially voice packets, no matter how heavy non real-time traffic is.

In ths letter, we suggest a collision resolution algorithm when the packet length is random, and analyze its throughput and delay performance. Here, three different packet length distributions and two feedback schemes (ternary and binary success/failure feedback) are considered.

This paper investigates the problems which inhibits the use of today's WDM networks. These are propagation delay, packet processing overhead, bit & frame synchronization, and tuning latency. So far, these problems, especially propagation delay, have been ignored in most performance analysis papers. They have always hindered network designers, but they are magnified by the order of magnitude increase in speed of optical communications systems as compared to previous media. This paper examines the impact of the propagation delay on the performance of WDM protocols with variations in the number of channels, packet length and system size, specifically in two reservation based protocols with control channels and two pre-allocation protocols without the control channels. Also the impact of three delay factors (packet processing overhead, bit & frame synchronization and tuning latency) are studied with different propagation delay parameters. In reservation protocols, each node has one agile transmitter and two receivers; one of them is fixed and the other one is agile. The fixed receiver continuously monitors the control channel, receives all control packets, and updates their own status tables in order to track the availability of the other nodes as a target and data channels to avoid the destination collisions and the data channel collisions, respectively. In pre-allocation protocols, each node has a tunable transmitter, a fixed or slow tunable receiver, and its own home channel to receive the packets. The performance of this protocol is evaluated through the discrete-event simulation in terms of the average packet delay and network throughput.

This paper deals with the method of integration of voice and data in wireless communication systems. By applying the DSV (Data Steal into Voice) technique to D-TDMA (Dynamic Time Division Multiple Access) systems, this paper presents an MAC (Media Access Control) method of integration of voice and data for the systems such as cellular radios and cordless phones. After a brief review of the D-TDMA scheme and the DSV technique, the protocol called D-TDMA/DSV is described. Then, a static analysis to derive the channel capacity and a dynamic analysis to evaluate the throughput and delay performance are presented. An extension of TFA (Transient Fluid Approximation) analysis is employed in the dynamic analysis. With the same system parameters, the capacity of D-TDMA/DSV is compared with that of the traditional D-TDMA. Under the limitation of the blocking probability required for cellular radios, some numerical examples of dynamic analysis are given to show the throughput and delay performance of the system.

We study behavior of multi-zone MCA (Multi-Channel Access) mobile communication systems with a finite number of channels in each zone. Three queueing schemes for channel requests named holding scheme, ready-nonready scheme, and optimum scheme are investigated. The delay performance of channel requests is studied through computer simulations.

In this paper, we propose DSVMA (Data Steal into Voice Multiple Access) scheme for integration of voice and data in wireless information networks. By using speech activity detectors and effective downstream control signals, DSVMA enables data terminals to transmit multi-packet messages when voice terminals are in silent periods. The S-G (throughput versus offered load) performance of the DSVMA system and the blocking probabilities of both the second generation systems and the DSVMA systems are evaluated by the static analysis. A dynamic analysis of a system with finite number of terminals is also presented using an approximate Markov analysis method. Some numerical examples are given in the paper. As a result, it is shown that DSVMA can improve the channel utility efficiency of a circuit-switched TDMA (Time Division Multiple Access) wireless communication system and is directly applicable for second generation wireless information systems.

In our preceding paper, I-ISMA (Idle Signal Multiple Access for Integrated services), a combination of ISMA and time reservation technique, was proposed to transmit an integrated voice and data traffic in third generation wireless communication networks. There, the channel capacity of I-ISMA was evaluated by the static analysis. To fully estimate performance of contention-based channel access protocols, however, we also need dynamic analysis to evaluate stability, delay, etc. Particularly, in systems concerning real-time voice transmission, delay is one of the most important performance measures. A six-mode model to describe an I-ISMA system is set up. With some assumptions for simplification, the dynamic behavior of the system is approximated by a Markov process so that the EPA (Equilibrium Point Analysis), a fluid approximation method, can be applied to the analysis. Then, numerical and simulation results are obtained for some examples. By means of the same analysis method and under the same conditions, the performance of PRMA is evaluated and compared briefly with that of I-ISMA.

Corresponding to the development of B-ISDN, integrated services for data, voice, etc. are imperatively required for the so called third generation wireless communication networks. In this paper, I-ISMA (Idle Signal Multiple Access for Integrated services) is proposed to transmit integrated voice and data traffic from dispersed terminals to a base station. In the system, data packets and the first packets of talkspurts of conversational speeches are transmitted using ISMA protocol over a shared channel while subsequent packets of talkspurts are sent with time reservation technique. The channel capacity of I-ISMA is evaluated and compared with that of PRMA. The region in which I-ISMA has larger capacity than PRMA is figured out. Generally speaking, I-ISMA has larger capacity than PRMA when the duration for transmitting and detecting an idle signal is not too long and the channel is not too congested by the reserved voice transmissions. When we concern real time voice transmission, delay is one of the most important performance measures. Only is a qualitative discussion on delay performance given here. The quantitative evaluation is obtained by the dynamic analysis in our succeeding paper.