This paper describes the design, implementation and testing of wideband code division multiple access (CDMA) base station demodulator for the international mobile telecommunication-2000 (IMT-2000) system test plant based on cdma2000 radio transmission technology (RTT). The performance of the implemented base station demodulator is measured and compared with the theoretical performance bound. The system test plant equipped with this demodulator provides wireless services, such as high quality speech (9.6 kbps), real-time video (384 kbps) and internet protocol (IP) based data services (144 kbps) in a mobile radio environment.

In this paper, we propose a Call Admission Control (CAC) scheme for the Direct Sequence-Code Division Multiple Access (DS-CDMA) systems supporting voice and data services and analyze the Erlang capacity under the proposed CAC scheme. Service groups are classified by Quality of Service (QoS) requirements such as the required Bit Error Rate (BER) and information bit rate, and Grade of Service (GoS) requirement such as required call blocking probability. Different traffics require different system resources based on their QoS requirements. In the proposed CAC scheme, some system resources are reserved exclusively for handoff calls to have high priority over new calls. Additionally, the queueing is allowed for both new and handoff data traffics that are not sensitive to delay. As a performance measure of the suggested CAC scheme, Erlang capacity is introduced. For the performance analysis, a four-dimensional Markov chain model is developed. As a numerical example, Erlang capacity of an IS-95B type system is depicted, and optimum values of system parameters such as the number of reservation channels and queue lengths are found. Finally, it is observed that Erlang capacity is improved more than 2 times by properly selecting the system parameters with the proposed CAC scheme. Also, the effect of handoff parameters on the Erlang capacity is observed.

The Direct-Sequence Code Division Multiple Access (DS-CDMA) is one of the most likely candidates for the 3rd generation mobile communication systems. On the other hand, a network supporting integrated data transfer and providing predictable qualities of service (QoS) to such application is an attractive issue. This paper proposes several schemes based on the majority rule concerned with resource reservation to investigate the performance of a CDMA cellular system. The admission control is based on the residue capacity and required signal-to-noise ratio (SNR) for multimedia traffic including voice, data and video. The system performance is explored with variable spreading gain and multi-code CDMA systems under the proposed schemes. Moreover, several system parameters including the handoff probability (lifetime in a cell) and the traffic load are investigated to study the impact to system performance. The results show the trade-off between the blocking probability of new calls and dropping probability of handoff calls under diversified situations. However, it is inefficient under resource reservation due to unused capacity. In this paper, we also propose a resource-borrowing scheme to improve system performance. Simulation results show that our proposed algorithm really improves the system performance in a multimedia CDMA cellular system with resource reservation.

Orthogonal Frequency Division Multiplexing-Direct Sequence/Code Division Multiple Access (OFDM-DS/CDMA) systems provide frequency diversity gain avoiding inter symbol interference (ISI) in a frequency selective fading channel. However, path diversity gain can not be obtained by using conventional OFDM-DS/CDMA schemes. This paper proposes a new multiple antenna transmission system with combined path diversity and frequency diversity. Signal of each antenna is delayed by several chips to create artificial path diversity as well as frequency diversity of multi-carrier transmission in which can then be combined by using a RAKE receiver. Therefore multiple antenna transmission scheme creates a path diversity effect on uncorrelated signals in multi-carriers from each antenna. The received uncorrelated signals can be processed by Maximum Ratio Combining (MRC) diversity without ISI at a RAKE receiver even when we use FFT modulation. As a result, we can obtain combined path diversity and frequency diversity gain effectively by the RAKE system with the combination of multiple antennas.

We have proposed the packet SS-CDMA scheme for downlink of SS-CDMA flexible wireless cellular network. Transmission packet is framed with synchronization block with 11 chip Barker code and information block with orthogonal spreading code. The chip synchronization is carried out using short code surface acoustic wave (SAW) matched filter. The code de-spreading is carried out using in-line de-spreader. Multi-channel downlink of 63 channels can be designed using orthogonal m-sequence. Simulation results show more than 15 channels without degradation from theoretical value can be used under multi-path environment. The packet SS-CDMA modem has been implemented using a 2.4 GHz front-end SAW matched filter. Degradation of E_b/N₀ of less than 0.5 dB is experimentally achieved with four-channel multiplex.

Mobile IP can provide macro mobility for a mobile node (MN) instead of the original IP communication. Since the orignal Mobile IP does not consider micro mobility, it can not avoid handoff latency. The latency causes packets loss or large variations of the delivery times. To prevent it, several methods are proposed to improve the orignal Mobile IP handoff method. In our previous work, we proposed the fast registration method which enabled an MN to have two communication routes from the home agent (HA), and realized lower latency in the handoff procedures by the doublecasting on the two communication routes. But, the simulation results also show that the proposed method still has large overhead. This paper presents a seamless handoff method, which improves the fast registration method. In the method, the MN can know the first and second suitable base stations for the communications through information from the MAC media. The control helps for seamless handoff in case that the MN must frequently change its current base station (BS). The computer simulations for Constant Bit Rate (CBR) over real-time transport protocol (RTP) flow shows that the proposed method provides no handoff latency with low overhead even if the MN must occur frequent handoffs in a second.

In this paper, we consider a mobile, multimedia, multihop (M³) ad hoc network. Key characteristics of this system are the mobility of users, energy constraints, and the need to operate without a fixed (wired or wireless) infrastructure. In this environment, with the advent of multimedia communications, the use of the cluster architecture has been revisited to support the resource reservation and Quality-of-Service routing. We proposed an access-based clustering protocol (ABCP) that allows the network to self-organize into a cluster architecture. Three advantages are claimed by ABCP. First, by the access-based criterion, it minimizes the overhead on cluster formation so that the protocol has short execution time and good scalability. Second, ABCP unifies the algorithms for cluster initialization and maintenance, i.e., the same set of clustering functions are used by a node regardless of whether it just becomes active or is in leaving its current cluster. Third, simulation results demonstrate that the cluster structure behaves stably amid topology changes compared with techniques previously proposed. Together with the access-based criterion, a multiple access scheme is also proposed for the broadcast of control messages.

The third generation cellular system will provide high bandwidth data service and multimedia applications are proposed on the high-speed data link. On the other hand, cellular handsets are getting equipped with a short-range radio communication device that is mainly targeted to connect cellular handsets to computers. In this paper, we propose a percolating data delivery mechanism for a cellular-ad hoc integrated network utilizing multicast/broadcast communication, which is endorsed by data synchronization, and single-hop ad hoc networks for information shower/exchange services. The mechanism lessens data traffic in both the cellular and the ad hoc network for data delivery and tolerates unexpected disconnection in the ad hoc network. The mechanism also compensates data delivery in areas out of the cellular service. Our implementation of the data delivery mechanism utilizes Bluetooth for an ad hoc network configuration, and delivers data in the cellular network to the Bluetooth ad hoc network via the data synchronization mechanism. We evaluate communication traffic and delivery time on the prototype system and discuss about the efficiency of the mechanism.

Differentiated services (Diffserv) model is one of the possible solutions for providing quality of service (QoS) on the Internet. Most existing approaches assume that the packet loss is an indication of network congestion and thus reduce the sending rates of sources. For wireless networks, the assumption is not proper since packet losses may be caused by other reasons, such as fading and interference of the signal. Therefore, these approaches do not work well in wireless networks. In this paper, we propose an approach which is able to provide service differentiation in wireless environments. In our approach, the rate share of a connection is determined by the associated weight. By keeping a proper amount of extra data in the network, the proposed approach can achieve weighted proportional fairness, which can provide selective QoS without any particular support from the network. We use the ns simulator to evaluate our approach. Simulation results show the validity of the proposed approach.

To satisfy huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes two different procedures of multi-channel multiple access schemes with the slotted ALOHA operation for both data and voice traffic and presents an exact analysis to numerically evaluate the performance of the systems. In scheme I, there is no limitation on access between data transmissions and voice transmissions, i.e., all channels can be accessed by all transmissions. In scheme II, a channel reservation policy is applied, where a number of channels are used exclusively for voice packets while the remaining channels are used for both data packets and voice packets. We call the system using scheme I "Non-reservation system" and call the system using scheme II "Reservation system. " Performance characteristics we obtained include loss probability for voice traffic, average packet delay for data traffic and channel utilization for both traffic. The performance of the schemes and the effects of the design parameters are numerically evaluated and compared to a wide-bandwidth conventional single-channel slotted ALOHA system with single data traffic. The analysis presented in this paper will be not only useful for the performance evaluation and the optimum design of multi-channel multi-traffic systems in wireless environments, but also applicable to evaluate other performance measures in priority networks, cellular mobile radio networks or multi-hop wireless networks.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.

Understanding traffic characteristics in mobile communications is invaluable for planning, designing, and operating cellular networks, and various mobility models have therefore come to be developed to predict traffic characteristics. In this paper, cell-dwell-time distribution and transition probability in a virtual cellular system are first estimated from the results of measuring taxi motion using the Global Positioning System (GPS) for large-city and small-city ranges of motion. Then, on the basis of simulations using these estimations, traffic characteristics like handoff rate and channel blocking probability in a cellular system are evaluated. It was found that a difference between large and small cities could be observed in speed distribution and direction-of-travel probability, but only a slight difference in cell-dwell-time distribution.

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.

Currently, the issues in Quality of Service, fairness and pricing strategies should have expedited the emergence of service differentiation in wireless access networks. In this paper, we propose a novel scheduling algorithm, called the Wireless Differentiated Fair Queueing (WDFQ) algorithm, to accommodate such need by providing delay/jitter controls, and fair residual bandwidth sharing for real-time and non-real-time traffic streams simultaneously. We show that the WDFQ scheme can achieve excellent performance, including timely delivery of real-time traffic, virtually error-free transmission of non-real-time traffic, and fair usage of channel bandwidth among remote stations. In addition, the location-dependent channel error property, as appeared in most wireless networks, are considered in the model and the temporary short error bursts are compensated by credits of bandwidth. The simulation results suggest that the length of retransmission period should be adapted to the error length to achieve good performance and maintain low implementation complexity.

We propose a dynamic assignment method of radio channels using their qualities and data packet sizes. In this method, a high quality channel is assigned for long packets and a low quality channel is assigned for short packets. Assuming a packet transfer method based on PHS (Personal Handy System) procedure, the transfer speed and delay characteristics are investigated by computer simulation and compared with conventional method (i.e. random assignment). Both the transfer speed characteristics and delay performance are improved under light traffic load.

Infostations serve a demand for multimedia services with high quality and high data rate. The challenge for Infostation system design is how to download as many information bits when a mobile user drives through an Infostation cell. A variable rate TDMA/TDD radio architecture is chosen for study and experimental implementation because of its flexibility in a multi-user environment. A symbol rate and rate compatible punctured convolutional code controlled adaptive transmission scheme is proposed for the Infostation radio design. Data throughput is analyzed for a mobile user passing through an Infostation cell. The analysis indicates that three data rates at 0.25, 1.0 and 2.0 Mbps achieve greater than 85% of the theoretical throughput while simplifying the implementation. The punctured convolutional code is studied to maximize the Infostation throughput. A recommendation is made to use rate 1/2 coding scheme. An efficient rate adaptation algorithm is proposed and it can track the Rayleigh fading channel well and the capacity loss is negligible in various conditions.

To improve the reliability and efficiency of multicast transmissions in wireless systems, a novel retransmission procedure is desired. In this paper, the representative acknowledgment scheme for reliable wireless multicast communications is proposed that offers quite a low packet loss rate. The proposed protocol carries out retransmissions in the datalink layer within the wireless region, and retransmissions do not affect the traffic in the wired region. The representative acknowledgment scheme employs both positive acknowledgment (ACK) and negative acknowledgment (NACK) to achieve reliable multicast transmissions and reduces the number of responses to be returned by forming groups of stations in the cell. One of the members in a group, called a representative station, returns a response for a received data frame while the others return a NACK if necessary. With this scheme, reliable multicast transmissions are enabled in wireless communications without spending much time as in conventional reliable multicast protocols. The performance of the proposed protocol is evaluated by numerical analyses and by computer simulation. The results show that 30% or more decrease in transmission time is achieved in a typical wireless environment.

We describe a control scheme for wireless-link layers and the TCP/IP layers in which wireless link states, such as signal strength and transmission rate, are transparent to the upper network layers and govern the behavior of the upper layers. Monitoring and notifying functions for wireless link states are incorporated into mobile hosts, and prefetching functions for the mobility agent lists are added to the present MobileIP functionalities of the network layers. In addition, we give the mobile hosts functions for controlling the TCP advertised windows in the transport layers and give the base-stations buffering functions to deal with the variable signal strength of the wireless links. A simulation (using ns-2) of this control scheme shows that mobile agent can be switched at the network layers soon after base-station hand-over and that errors in packet routing, packet loss and communication throughput reduction can be avoided. Moreover, communications can be interrupted without having to shrink the size of the congestion windows of the TCP senders, which improves overall throughput.

Due to the lack of support in mobility and quality of service, today's IP-based networks have some inherent limitations for delivering multimedia services in a mobile environment. In the past few years, these issues have been addressed in the research community and the resulting techniques are being standardised. However, these developments have been done in isolation and become incompatible with each other. In addition to these technical issues, the future infrastructure for charging and accounting mobile multimedia services is expected to be increasingly complicated. In this paper we present a home-proxy based framework which can facilitate the integration of mobility support and QoS management. Furthermore, it enables centralised accounting, which simplifies the cost recovery processes of roaming services. To prove the viability of our design, we have built and tested this framework on a DiffServ wide-area backbone using an MP3 streaming application.

The rapid growth of mobile systems and the exponential spread of the Internet have requested technologies for efficient and seamless delivery of IP data to mobile users. However, the Mobile Internet Protocol and the IETF Mobile IPv6 proposal are not scalable and capable of handling real time applications. The Mobile Internet Protocol employs mobility agents to support Internet-wide mobility, and mobile node employs the concept of care-of address to communicate with its correspondent node when it changes its point of attachment to the Internet. This paper proposes a new addressing scheme for mobile node based on IPv6. The concept of Mobile Internet is introduced, which is a logical subnet of IPv6 Internet and supports IP layer mobility. Mobile Internet is geographically overlaid on the Internet. It has a fixed subnet prefix, and each mobile node in it is only identified by its home IP address, regardless of its current location. Some new kinds of mobility agents (LRPC, LRPS/FLR) are defined. The proposed scheme is considered as a long-term solution for the Internet with mobile computers, several defects in the current Mobile IP protocol are solved.

The IETF Mobile IPv6 enables any IPv6 node to both cache the Care-of Address associated with a mobile node's home address, and to directly send packets addressed to a mobile node at the Care-of Address using the IPv6 routing header. Support for optimizing the route is built in as a fundamental part of the protocol. Several hierarchical schemes have been presented recently on top of the Mobile IPv6. These schemes separate micro-mobility from macro-mobility and exploit a mobile node's locality. They can reduce the number of signaling messages sent to a home network and improve hand-off performance. However, existing hierarchical schemes do not achieve route optimization. When external correspondent nodes send packets to a mobile node, these packets are intercepted by an intermediate mobility agent encapsulated and routed to the mobile node. In this paper, we propose a new hierarchical scheme that enables any correspondent node to cache two Care-of Addresses; the mobile node's temporary address and the intermediate mobility agent's address. Also, we introduce two lifetimes managing the two Care-of Addresses. Until the lifetime associated with the mobile node's temporary address expires, a correspondent node can send packets directly to the mobile node. If the lifetime expires but the lifetime associated with the intermediate mobility agent's address has not expired, the correspondent node sends packets to the intermediate mobility agent. This proposal can reduce delay in packet delivery and optimize routing. Furthermore, based on the mobility of a mobile node, we introduce more reduced frequency of binding update and longer period of the lifetimes than the existing hierarchical schemes. Therefore, our proposal can reduce the binding update bandwidth as well as the packet delivery bandwidth lower than those of the IETF IPv6 and the existing hierarchical schemes.

This paper proposes a new satellite communication system that enables high-speed communication in a mobile environment. The system configuration combines a terrestrial mobile network and an existing satellite system, and includes a tracking antenna that was newly developed to receive 30 Mbit/s signals from commercial communication satellites. A prototype system comprising the mobile network, the satellite system and a vehicle in which the tracking antenna is installed was constructed for purposes of evaluation and demonstration. A LAN system was incorporated in the experimental vehicle by using the tracking antenna, a satellite router and a Personal Digital Cellular phone. The validity of the proposed system was verified by the tracking antenna driving tests, system UDP tests and FTP throughput tests in a mobile environment.

We propose a mechanism to facilitate the development of component-based mobile applications with adaptive behaviors. The design principles and communication patterns of legacy software systems will greatly change in a forthcoming environment, where a variety of computing devices become embedded in home and office environments, users move around with/without portable computing devices, and all the devices are interconnected through wired/wireless networks. In the proposed mechanism, Improvised Assembly Mechanism (IAM), we realize functionality to compose an application in an ad hoc manner and to achieve the adaptation of applications by adding, replacing, supplementing, and relocating components at system runtime according to various environmental changes such as the locational changes of computing devices and users. The mechanism is implemented as a built-in functionality of the Soul component, which is one of the fundamental elements in the Possession model.

In this paper, we propose a new filesystem named Ad-hoc Filesystem which targets ad-hoc wireless networks. Ad-hoc Filesystem is a serverless filesystem which automatically generates temporary shared space among multiple mobile machines when they gather in a communicable range. The generated space is freely available for participating machines and can be used for any temporary work there. The design of Ad-hoc Filesystem is strongly motivated by the recent wireless network technologies and advances of ad-hoc network techniques. However, targeting such an unstable network forces us a completely different design from existing filesystems. We have designed an initial prototype of Ad-hoc Filesystem and have evaluated the simulation results. Our design is strongly based on an assumption such that people who wish to work together would form a stable group in a range, and replicating data on two different machines would be enough to provide a proper level of availability. Hence, Ad-hoc Filesystem distributes files among multiple machines and duplicates them, in order to keep the files available even if some of participating machines arbitrarily leave the communication range. The directory entries are kept as a soft-state mechanism maintained by exchanging broadcast packets. The simulation results show that using broadcast to maintain system state hardly affects the overall system performance, and optimizations such that delaying replies with broadcast and sensing other packets promiscuously during the delay reduce the overhead significantly. Also the result indicates that our strategy that keeps two replicas for each data is promising to provide availability. Although further investigation based on real implementation remains important work, we believe that the prototype has a contribution to realize the possibility of ad-hoc filesystem by synthesizing number of previous works in areas such as network filesystems, server replication and ad-hoc wireless networks.

High-speed digital data transmission services with mobile equipment are becoming available. Though the visual signal is one of the expected media to be used with such transmission capabilities, the bandwidth of visual signal is much broader than the provided transmission bandwidth in general. Therefore efficient video encoding algorithms have to be introduced. The ITU-T Recommendation H.263 and ISO/IEC MPEG-4 are very powerful encoding algorithms for a wide range of video sequences. But a large amount of bits are generated in highly active scenes to encode them using such conventional methods. This results in frame skipping and degradation of decoded picture quality. In order to keep these degradations as low as possible, we proposed a Dynamic Resolution Conversion (DRC) method of the prediction error. In the method, a reduced resolution encoding is carried out when the input scene is highly active. Simulation results show that the proposed scheme can improve both coding frame rate and picture quality in a highly active scene. We also present in this paper that some analysis for the performance of the DRC method under the error prone environment that is inevitable with mobile communications.

Error concealment is an essential part of reliable video communication systems because transmission errors are inevitable even when the coded bitstream is highly protected. The problem of temporal EC can be factored into two parts regarding candidate motion vectors (MVs) employed and the matching criterion to evaluate the fitness of each candidate MV. In order to obtain more faithful EC results, this paper proposes a novel iterative EC algorithm, in which an efficient way to provide candidate MVs and a new fitness measure are presented. The proposed approach for candidate MVs systematically utilizes all the available neighboring MVs by exploiting a well-known spatiotemporal correlation of block MVs. Also, in order to remove the dependency of a damaged block's quality of concealment on the already concealed adjacent blocks, we develope a new matching criterion. The objective of the proposed fitness measure is to minimize the total boundary matching errors induced by the whole corrupted blocks. Simulations performed using an H.263 codec demonstrate a significant improvement on the subjective and objective concealed video qualities, especially when the corrupted area is wider than a single row of coding blocks.

In this paper, a slot-averaged SIR model is derived to analyze the performances of 1/N-slotted DS-CDMA packet radio networks. DS-CDMA packet radio networks can be analyzed with channel threshold model, but it is not appropriate for a 1/N-slotted access scheme since the interference level varies slot by slot during a packet transmission time. A packet in a DS-CDMA system will be channel-encoded and interleaved so that an excessive interference of some slots does not result in a corruption of a packet directly. In a 1/N-slotted DS-CDMA packet system, interleaving and channel coding can cope with the excessive interference in some slots. Proposed slot-averaged SIR model reflects this effect and is simple enough to be used in the performance analyses of various packet access control schemes. This paper verifies the slot-averaged SIR model and evaluates the throughputs of 1/N-slotted access schemes using it. This paper also compares the results with those of the conventional channel threshold model. Packet access schemes such as slotted ALOHA, slotted CLSP and slotted CLSP/CC are analyzed. The results show that many different aspects from the results of the previous works can be evaluated and this work will offer more accurate view on the behavior of 1/N-slotted DS-CDMA packet radio networks.

We evaluate the performance of indoor infrared wireless systems using on-off keying code division multiple access (OOK-CDMA) with decision-feedback equalizer (DFE) on diffuse channels. To estimate the impulse response, we use the training sequence that alternates '1' and '0. ' We show that the OOK-CDMA with DFE and the training sequence can achieve better performance than the OOK-CDMA without DFE. We also show that the OOK-CDMA with DFE and the training sequence can achieve almost the same performance as the OOK-CDMA with DFE and the known impulse response.

In this paper, we propose an efficient channel estimation technique for orthogonal frequency-division multiplexing (OFDM) systems with transmitter diversity. The proposed technique estimates uniquely all channel frequency responses needed in space-time coded OFDM systems using "comb-type" training symbols. The computational complexity of the proposed technique is reduced dramatically, compared with the previous minimum mean-squared error (MMSE) technique, due to the processing made all in the frequency-domain. Also, several other techniques for mitigating random noise effect and tracking channel variation are discussed to further improve the performance of the proposed approach. The performances of the proposed approach are demonstrated by computer simulation for mobile wireless channels.

In this paper, a medium access control protocol is proposed for the integrated voice and data services in the local CDMA communication systems. Based on WB-TD/CDMA, uplink channels for the proposed protocol are composed of time slots with multiple spreading codes for each slot. During a talkspurt, a voice terminal transmits its entire packets over a reserved code. On the other hand, a data terminal transmits its packet after sensing the spreading code status. The base station broadcasts the status of spreading codes for each slot. In this protocol, voice packets never contend with data packets. The numerical results show that this protocol increases the system capacity for voice service by applying the reservation scheme. Despite the low access priority of data terminal, the data traffic performance can be increased in proportion to the number of spreading codes.

This paper presents the design, implementation and test results of a complete wideband CDMA modem with a modulator and a demodulator for use in next-generation cell sites. The modem is based on Japanese proposal for the next generation W-CDMA air interface. The modulator has a flexible architecture, which can transmit the data of different physical channels such as common control physical channels, and dedicated physical channels by setting the channel selection registers of the modulator. The modulator performs the transmission power control digitally according to the physical channel setting. The receiver consists of a searcher, four fingers and a combiner. The searcher supports path selected search operation considering antenna diversity which is employed to reduce the effect of excessively deep fades, and early dump operation to speed up the searching process. Since the simulation results to determine the number of finger showed that the system with more than four fingers had no essential performance improvement, demodulation performed with four-finger rake receiver. The proposed cell site modem was implemented with FPGAs and verified by the board-level experiment. Experiments have verified that the modem fully complied with the specifications.

In this paper, we first investigate the problems of the existing branch point algorithms for available bit rate (ABR) multicast connections in ATM networks, and then propose various solutions for resolving those problems. By combining these solutions, we also propose a new efficient and scalable branch point algorithm. In the proposed algorithm, each branch point stores the feedback information on a per-branch basis for each virtual connection and only passes BRM cells returning from the farthest destination. Simulation results show that the proposed algorithm can provide a good fairness, a higher efficiency and an excellent scalability, compared with the existing algorithms.

This paper proposes an adaptive zone selection (AZS) scheme with adaptive modulation for wireless packet transmission systems to achieve high throughput and low delay performances even under non-uniform traffic conditions. In the proposed system, based on the measurement of the propagation path characteristics to each access point (AP) as well as broadcasted blocking probability information from each AP, a terminal autonomously selects an AP and modulation parameters that give the minimum transmission failure probability determined by both the call blocking rate due to lack of radio resource and packet error rate due to severe channel conditions. Computer simulation confirms that the proposed scheme greatly improves throughput and delay performances especially under non-uniform traffic conditions.

In this paper, we propose a new location tracking method for pointer forwarding with distributed HLRs schemes to decrease the location tracking cost. Similar to the distributed HLR schemes, the location information is replicated on all HLRs in our scheme. However, unlike the traditional distributed HLR schemes, we propagate the most recent location changes from an HLR to reset the forwarding pointers in other HLRs, and thus reduce the cost of location tracking for call deliveries. The signaling cost of location propagation may be too costly to be practical. Therefore, we allow some inconsistency to exist among the location data of HLRs and defer the propagation of location changes to reduce the propagation cost. Three delay propagation policies are also proposed and analyzed using simulation. The performance result shows that the delay propagation policies reduce the propagation cost and the propagation resetting process degrades the location tracking cost. Our method preserves a low total cost for call deliveries, registrations and propagations in distributed HLRs.

In this paper, we proposed two models, the full multiple MCS (Multicast Server) model and the hybrid multiple MCS model to support multiple MCS over a single large cluster in ATM (Asynchronous Transfer Mode) networks. Also, we presented two methods for MCS assignment which are known as 2PSPMT (2 Phase Shortest Path based on Multicast tree) and hybrid-2PSPMT, and evaluated its performance by simulation. When an ATM host requests joining a specific multicast group, the MARS (Multicast Address Resolution Server) designates a proper MCS among the multiple MCSs for the group member to minimize the average path delay between the sender and the group members. Each method for MCS assignment construct a 2-phase partial multicast tree based on the shortest path algorithm. We reduced the average path delay in the multicast tree using these methods with various cluster topologies and MCS distribution scenarios in addition to distributing the load among multiple MCSs.

The problem of finding a minimum-cast multicast tree (Steiner tree) is known as NP complete. Heuristic based algorithms for this problem to achieve good performance are usually time-consuming. In this paper, we propose a new strategy called tree-caching for efficient multicast connection setup in connection-oriented networks. In this scheme, the tree topologies that have been computed are cached in a database of the source nodes. This can reduce the connection establishment time for subsequent connection requests which have some common multicast members, by an efficient reuse of cached trees without having to re-run a multicast routing algorithm for the whole group. This method can provide an efficient way to eliminate, when ever possible, the expensive tree computation algorithm that has to be performed in setting up a multicast connection. We first formulate the problem of tree-caching and then propose a tree-caching algorithm to reduce the complexity of the tree computations when a new connection is to be established. Through simulations, we find that the proposed tree-caching strategy performs very well and can significantly reduce the computation complexity for setting up multicast connections.

We examine delay performance of packets from constant bit rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the Σ D_i/G/1 queue with vacations. Our analysis allows heterogeneous service time and heterogeneous interarrival time. Thus, we can get the impact of packet length of a stream on the delay time of other streams. We then give various numerical results for enterprise multimedia networks, which include voice, video and data communication services. From our quantitative evaluation, we conclude that packet length of video traffic has large influence on the delay time of voice traffic while voice traffic gives a little impact on the delay time of video traffic.

Broadcast data delivery is attractive for large-size data distribution where a large user community is connected to a server through a network. It is important to consider a broadcast scheduling method which minimizes the average response time. The scheme should also guarantee the expected waiting time at the time of request. In this paper, we propose a method which divides all titles into several groups and assigns FIFO to each group. The proposed method can guarantee the waiting time for each user at his request, and is superior to FIFO (in high load) and a fixed allocation method (in low load).

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.

A low sidelobe two-beam waveguide slot array is designed and measured. The antenna structure should be symmetrical for realizing two symmetrical beams which imposes restriction in slot design for the sidelobe and the gain. The slot coupling distribution is optimized numerically for side-lobe suppression under the condition of the structural symmetry. The first side-lobe level is minimized for the specific antenna efficiency in the continuous source model. This synthesis is reinforced by the full wave slot analysis using the method of moments. The design is confirmed by experiments using a one-dimensional array at 12 GHz and the good agreements between the predictions and the measurements are observed.

In this paper, we describe statistical properties of timing jitter of symbol timing recovery circuit for carrierless amplitude/phase modulation (CAP)-based very high-rate digital subscriber line (VDSL) system. Analytical expressions of the timing jitter for envelope-based timing recovery system, such as squarer-based timing recovery (S-TR) and absolute-value-based timing recovery (A-TR) schemes, are derived in the presence of additive white Gaussian noise (AWGN) or far-end crosstalk (FEXT). In particular, the analytical and simulation results of the timing jitter performance are presented and compared for a 51.84 Mb/s 16-CAP VDSL system. The A-TR system implemented digitally meets the DAVIC's VDSL system requirement, which specifies the maximum peak-to-peak jitter value of 1.5 nsec and the acquisition time of 20 msec.

In this letter, we propose an efficient channel allocation scheme to provide multicast traffic in multitier cellular systems. Our proposed scheme allocates microcell/macrocell channels based on the 'microcell-group size' and probability. Also, we analyze the performance of the scheme in view of the call blocking probability in case of considering overflow traffic or not. Numerical results show that our proposed scheme exhibits a better performance than conventional schemes.

The common packet channel (CPCH) adopted by 3GPP is used to carry small and medium sized packets, such as SMS (short message service) and web data. We analyze the system performance of the non-real time channel monitoring scheme when the periodic broadcast method is deployed. If user equipment (UE) can continuously monitor the CPCH status, the performance of the channel monitoring scheme is satisfactory. However, in practice, a CPCH status broadcast period exists and, therefore, an UE can request a CPCH channel based on incorrect CPCH status information. This incorrect CPCH status information results in performance degradation in the channel monitoring scheme. We analyze this performance degradation and show the relationship between system performance and the status broadcast period.

The VLSI implication of the guard interval in orthogonal frequency-division multiplexing (OFDM) systems is described. A new OFDM transmission scheme using cyclic suffix is proposed to reduce the hardware complexity required for implementing the guard interval in the transmitter, and is shown to have the same performance as the conventional approach using cyclic prefix, even with a significantly lower hardware complexity (smaller buffer size and no processing delay).