In this paper, a network dimensioning approach suitable to the Internet is discussed. Differently from the traditional telephone networks, it is difficult to guarantee QoS for end-users even in a statistically sense due to an essential nature of an end-to-end communication architecture in the Internet. We should therefore adopt another approach, based on the traffic measurement. In the approach, the traffic measurement is performed for monitoring the end-to-end QoS. Then, the network adaptively controls the link capacities to meet the user's QoS demands. For this purpose, the underlying network should support such a capability that the link capacities can be flexibly reused. With the WDM network as an underlying network, an example scenario for network provisioning is finally illustrated.

This paper describes IP traffic, especially the control of VoIP traffic, on the carrier-scale, and proposes algorithms for it. It examines a case that has already been introduced in the United States and discusses the trend of standardization for this control. Control techniques that will be introduced into the IP network in the future are considered from the viewpoints of both "quality" that users receive and the "control" that carriers perform.

The provisioning of QoS in the Internet is gaining an increasing attention, thus the importance of methods capable of estimating the bandwidth requirement of traffic flows is constantly growing. This information can be used for a wide range of purposes. Admission control, QoS routing and load sharing all need the same basic information in order to be able to make decisions. This paper describes a number of methods that can be used to arrive at precise estimates of the bandwidth requirement focusing on those that are based on the theory of large deviations. A methodology is presented that allows the reformulation of earlier solutions based on the estimation of some form of an overflow probability so that their output becomes a bandwidth-type quantity, the format preferred by Internet control applications. The methodology provides two tracks for the conversion: an indirect method that encapsulates the overflow probability-type approach as an embedded calculation and a direct method that immediately results in the estimate of the bandwidth requirement. The paper introduces a novel method for the direct computation of the bandwidth requirement of Internet traffic flows using the many sources asymptotic regime of the large deviation theory. The direct bandwidth estimator method reduces the computational complexity of the calculations, since it results directly in the bandwidth requirement, allowing the omission of the frequent and costly computation of the buffer overflow probability. The savings arising from the reduction in computational complexity are demonstrated in a numerical example.

For several years, more and more people are joining the Internet and various kind of packets (so called transaction-, block-, and stream-types) have been transmitted in the same network, so that poor network conditions cause loss of the stream-type data packets, such as voices, which request smaller transmission delay time than others. We consider a switching node (router) in a network as an N-series M/G/1-type queueing model and have mainly evaluated the fluctuation of packet delay time and end-to-end delay time, using the two moments matching method with initial value, then define the delay jitter D of a network which consists of jointed N switching nodes. It is clarified that this network is not suitable for voice packets transmission media without measures.

Guarantees of quality-of-service (QoS) in the real-time transmission of video on the Internet is a challenging task for the success of many video applications. The Internet Engineering Task Force (IETF) has proposed the Guaranteed Service (GS) in the Integrated Service model with firm delay and bandwidth guarantees. For the GS, it is necessary to provide traffic sources with the capability of calculating the traffic characteristics to be declared to the network on the basis of a limited set of parameters statistically characterizing the traffic and the required level of QoS. In this paper, we develop an algorithm for the evaluation of the traffic parameters which characterize the video stream when a QoS requirement is given. To this end an analytical traffic model for the VBR MPEG video is introduced. Simulation results show that the proposed method can evaluate the traffic parameters precisely and efficiently.

This paper studies effect of packet scheduling algorithms at routers on media synchronization quality in live audio and video transmission by experiment. In the experiment, we deal with four packet scheduling algorithms: First-In First-Out, Priority Queueing, Class-Based Queueing and Weighted Fair Queueing. We assess the synchronization quality of both intra-stream and inter-stream with and without media synchronization control. The paper clarifies the features of each algorithm from a media synchronization point of view. A comparison of the experimental results shows that Weighted Fair Queueing is the most efficient packet scheduling algorithm for continuous media among the four.

We have focused on the RIO queueing mechanism in statistical bandwidth allocation service, which uses AF-PHB. We have studied the parameterization of RIO to achieve both high throughput and low delay. We were able to parameterize RIO for that purpose in terms of both minth and maxp used in dropping OUT packets. Furthermore, we have also examined the parameterization regarding EWMA (Exponential Weighted Moving Average), i.e., weight factor w_qout, and have shown that dropping OUT packets should depend upon the queue length without much delay unlike in RED. From our simulation results, we could see that our parameterization provided high throughput performance and also limited the queue length in a narrow range more effectively.

In the Internet, because of huge scale and distributed administration, it is of practical importance to infer network-internal characteristics that cannot be measured directly. In this paper, based on a general framework we proposed previously, we present a feasible method of inferring packet loss rates of individual links from end-to-end measurement of unicast probe packets. Compared with methods using multicast probes, unicast-based inference methods are more flexible and widely applicable, whereas they have a problem with imperfect correlation in concurrent events on paths. Our method can infer link loss rates under this problem, and is applicable to various path-topologies including trees, inverse trees and their combinations. We also show simulation results which indicate potential of our unicast-based method.

In a shared buffer packet switch, a good buffer management scheme is needed to reduce the overall packet loss probability and improve the fairness between different users. In this paper, a novel buffer control scheme called partial sharing and partial partitioning (PSPP) is proposed. The PSPP is an adaptive scheme that can be dynamically adjusted to the changing traffic conditions while simple to implement. The key idea of the PSPP is that part of the buffer space, proportional to the number of inactive output ports, is reserved for sharing between inactive output ports. This portion of buffer is called PS buffer. The residual buffer space, called PP buffer, is partitioned and distributed to active output ports equally. From the analysis results, we only need to reserve a small amount of PS buffer space to get good performance for the entire system. Computer simulation shows the PSPP control is very robust and very close to the performance of pushout (PO) buffer management scheme which is a scheme considered as optimal in terms of fairness and total loss ratio while too complicated for implementation.

A feedback-based congestion control mechanism is essential to realize an efficient data transfer service in packed-switched networks. TCP (Transmission Control Protocol) is a feedback-based congestion control mechanism, and has been widely used in the current Internet. An improved version of TCP called TCP Vegas has been proposed and studied in the literature. It can achieve better performance than TCP Reno. In previous studies, performance analysis of a window-based flow control mechanism based on TCP Vegas only for a simple network topology has been performed. In this paper, we extend the analysis to a generic network topology where each connection is allowed to have a different propagation delay and to traverse multiple bottleneck links. We first derive equilibrium values of window sizes of TCP connections and the number of packets waiting in a router's buffer. We also derive throughput of each TCP connection in steady state, and investigate the effect of control parameters of TCP Vegas on fairness among TCP connections. We then present several numerical examples, showing how control parameters of TCP Vegas should be configured for achieving both stability and better transient performance.

Recently, as multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) will co-exist. In addition, classified service based on Diff-Serv (Differentiated Service), MPLS (Multi-Protocol Label Switching),etc., have come into wide use. Today's Internet environment requires routers to perform control mechanisms in order to guarantee various QoSs. In this paper, we propose a buffer management scheme for the Internet router that uses class-based priority control. This paper focuses on per-flow queueing, and evaluates the performance of the proposed buffer management scheme. Realization of differentiated services and dissolution of buffer occupation by specific flow is expected by the proposed control.

Several gateway-based congestion control mechanisms have been proposed to support an end-to-end congestion control mechanism of TCP (Transmission Control Protocol). One of promising gateway-based congestion control mechanisms is a RED (Random Early Detection) gateway. Although effectiveness of the RED gateway is fully dependent on a choice of control parameters, it has not been fully investigated how to configure its control parameters. In this paper, we analyze the steady state behavior of the RED gateway by explicitly modeling the congestion control mechanism of TCP. We first derive the equilibrium values of the TCP window size and the buffer occupancy of the RED gateway. Also derived are the stability condition and the transient performance index of the network using a control theoretic approach. Numerical examples as well as simulation results are presented to clearly show relations between control parameters and the steady state behavior.

The differentiated services (Diffserv) architecture is a potential solution for providing quality of service (QoS) on the Internet. Most existing studies focus on providing service differentiation among few service classes. In this paper, we propose an approach which can achieve per-flow weighted fair rate allocation in a differentiated services network. Following the design philosophy of the Diffserv model, in the proposed approach core routers do not need to keep per-flow information. An edge router adjusts the transmission rate of a flow based on the feedback carried on control packets, which are inserted by the ingress edge router and returned by the egress edge router. Core routers periodically estimate the fair share rate of each virtual flow and mark the results in control packets. We use both simulations and analysis to evaluate the performance of the proposed approach. The analytical results show that our approach allows a system to converge to weighted fair rate allocations in limited time. Through the simulation results, we can further validate the analytical results, and demonstrate that better throughput can be achieved.

Most applications can adapt their coding techniques and sending rates according to the network congestion and the resource needed can be provided at the beginning of the transmission. So traditional Differentiated Services (DiffServ) model is too rigid to them. In this paper, we are seeking a balance between the relative DiffServ and the absolute DiffServ and propose a new Diffserv model, a relative Differentiated Service model with admission control, which suits the adaptive application. By providing the proportional differentiated services in core routers and loss-rate based CAC control in edge routers, we can make both the network and the users adaptive: the network is adaptive to the traffic load and the users is adaptive to the network congestion. This model is promising to the elastic but unpredictable traffic, such as IP telephony or other multimedia applications.

At present, the global Internet consists of many ASes. Each AS pays a pre-determined connection fee to another AS for connecting its network with that AS's network. The connection fee type charging may be rational in case of transferring the best-effort type traffic. However, usage charging is necessary to transferring the resource guaranteed type traffic such as the Intserv traffic and the Diffserv traffic. In this case, each AS pays a per-flow fee to another AS every time it routes a flow into another AS. The per-flow fee paid by each AS becomes a part of the cost for that AS. Thus, each AS needs to select a route with the lowest price to improve its own profit. In this paper, we call such an inter-AS routing scheme a price-based inter-AS routing scheme. When each AS has a request to route an inter-AS flow, it can select an inter-AS route with the lowest price to improve its own profit by this routing scheme. Cost-dependent pricing scheme is suitable for the price-based inter-AS routing scheme because it can reduce frequency of price information exchange between ASes. However, in the cost-dependent pricing scheme, profit in each AS depends on the distribution of path costs in that AS. Generally, ASes with narrow ranges of path costs cannot obtain sufficient profits compared to ASes with wide ranges of path costs. Thus, we propose a routing policy for ASes with narrow ranges of path costs to improve their profits efficiently and evaluate its effect using a simple routing model.

The explosive growth of World Wide Web usage is causing a number of performance problems, including slow response times, network congestion, and denial of service. Web site that has a huge number of accesses and requires high quality of services, such as a site offering hosting services, or content delivery services, usually uses a cache server to reduce the load on the original server offering the original content. To increase the throughput of the caching process and to improve service availability, multiple cache servers are often positioned in front of the original server. This requires a switch to direct incoming requests to one of the multiple cache servers. In this paper, we propose a routing algorithm for such a switch in front of clustered multiple cache servers and evaluate its performance by simulation. The results show that our routing algorithm is effective when content has request locality and a short period of validity, for example, news, map data, road traffic data, or weather information. We also identify points to consider when the proposed algorithm is applied to a real system.

The requirement to realize large-capacity, high-speed and guaranteed Quality of Service (QoS) communications in IP networks is a recent development. A technique to satisfy these requirements, Multi-Protocol Label Switching (MPLS) is the focus of this paper. In the future, it is expected that congestion and faults on a Label Switched Path (LSP) will seriously affect service contents because various applications are densely served in a large area. In MPLS, however, methods to solve these problems are not clear. Therefore, this study proposes a concrete traffic engineering method to avoid heavy congestion, and at the same time, endeavors to realize a fault-tolerant network by autonomous restoration, or self-healing.

LSR (Label Switching Router)s in MPLS (Multiprotocol Label Switching) networks map arriving IP flows into some labels on Layer 2 switching fabric and establish LSP (Label Switching Path)s. By using LSPs, LSRs not only transmit IP datagrams fast by cut-through mechanism, but also solve traffic engineering issue to optimize the delay of some IP datagram flows. So far, we have analyzed the performance of LSR focusing only on the maximum number of LSPs which can be set on Layer 2. In this paper, we will also consider the bandwidth allocated to each LSP and analyze the IP datagram transmission delay and the cut-through rate of LSR. We suppose the label mapping method as the data-driven scheme in the analytical model, so that the physical bandwidth of LSR is shared by both the default LSP for hop-by-hop transmission and the cut-through LSPs. Thus, we will investigate the impact of the bandwidth allocation among these LSPs on the performance.

Grouping multiple voice frames into a single IP packet ("frame grouping") is a commonly mentioned approach to saving bandwidth in IP telephony. But little is known as to when, how, and how much frame grouping should be done in Internet environment. This paper explores the feasibility and the methods of frame grouping based on Internet delay measurement. Specifically, we propose an adaptive frame grouping method that minimizes the delay violation while reducing the bandwidth usage by as much as a factor of two under real Internet delay fluctuations. The performance of the method is evaluated as it is used against a single voice stream and then against multiple voice streams.

In the paper, we propose a congestion control scheme for reliable multicast communication which enables TCP fairness and prevents a drop-to-zero problem. The proposed congestion control scheme is rate-based one based on NAKs from receivers and cooperatively works with a flow control scheme. The congestion control scheme consists of two components of a rate-based controller and a selection mechanism of a representative. The rate-based controller runs between the sender and the representative and achieves TCP fairness and fast response to losses at the representative. The selection mechanism of the representative allows the sender to select the representative in a scalable manner, in which the sender makes use of NAKs from receivers to select it. In the paper, we also propose the switchover mechanism of the flow and congestion control schemes which enables the sender to use either of them adaptively based on network situations. When the network is congested, the congestion control scheme works to share network resources fairly with competing TCP flows. Otherwise, the flow control scheme works to adapt the transmission rate to the slowest receiver. We verify the performance of our proposed schemes by using computer simulation.

In this paper, we propose a channel assignment scheme for integrated voice and data traffic in reservation multiple access protocol. In the proposed scheme, a voice packet never contends with a data packet and takes over the slot which is previously assigned to a data packet. Thus, a larger number of voice terminals can be accommodated without degradation of quality and throughput even in the situation that data were integrated. We evaluate the voice packet dropping probability, throughput and packet delay through computer simulation. The results show that the proposed scheme has better performance than the conventional PRMA and DQRUMA systems.

Teletraffic characteristics of a mobile packet communication network, which supports mobile Internet, were quantitatively evaluated by using a terminal migration model in which the cell dwell time possesses self-similarity. I used a migration model in which the migration speed of the terminal is determined by the density of the dwell terminals in a cell (determined from measured vehicular mobility characteristics). The transmission rates per terminal in a cell were estimated as teletraffic on the mobile packet communication networks using this migration model. I found that when there is self-similarity in the terminal cell dwell time, communicating terminals may be concentrated in the cell and restricted for an indefinite period of time to using only a narrow bandwidth.

We will discuss a novel non-contact removal technique of optical fiber coating in continuous and uninterrupted manner with hot air stream. We observed little degradation of the tensile strength of the optical fiber after removing the protective polymer coating and the mean breaking tensile strength of the stripped optical fiber using non-contact removal method was 5.1 GPa.

This paper introduces a new application of adaptive control theory to power control in a code division multiple access (CDMA) cellular system operating over mobile fading radio channels. Conventional feedback power control algorithms allow the base station to send a power command to either raise or lower each user's transmission power according to a bang-bang-like control policy. In this paper, we present an adaptive minimum-variance power control methodology which can be shown to improve power control performance consistently against a random nature of the near-far effect, shadowing and fast varying fading. Two adaptive implementations are considered: direct and indirect control. In the indirect adaptive control, a minimum-variance controller is combined with a constrained estimation algorithm to ensure the stability of a link gain model. In the direct adaptive control, the controller parameters are obtained directly from a standard estimation algorithm. Our simulations have shown that the proposed adaptive minimum-variance power control schemes provide much smaller error variance than the conventional fixed-step bang-bang control scheme and consequently the reverse channel capacity of the CDMA system can be significantly increased.

OFDM modulation has attracted attention for fourth-generation mobile communication systems and high-speed wireless LANs. However, it has a very serious problem of large peak power. PTS (partial transmit sequences) has been proposed as one solution to this problem. In PTS, the OFDM subcarriers are divided into several clusters, and the phase of each cluster is rotated by a complex weight to minimize the PAPR (peak-to-average power ratio). However, the weight of the phase rotation must be sent to the mobile terminal by using a side information channel. In this paper, we propose two weight estimation methods at the receiver to avoid weight transmission in side information channels. The first method uses pilot signals, while the second is a blind estimation method that changes the weight pattern. We evaluate the performance of these methods by computer simulation.

The maximum likelihood sequence estimator (MLSE) is usually implemented by the Viterbi algorithm (VA). The computational complexity of the VA grows exponentially with the length of the channel response. With some performance reduction, a decision-feedback equalizer (DFE) can be used to shorten the channel response. This greatly reduces the computational requirement for the VA. However, for many real-world applications, the complexity of the DFE/MLSE approach may be still too high. In this paper, we propose a constrained DFE that offers much lower VA computational complexity. The basic idea is to pose some constraints on the DFE such that the postcursors of the shortened channel response have only discrete values. As a result, the multiplication operations can be replaced by shift operations making the VA almost multiplication free. This will greatly facilitate the real world applications of the MLSE algorithm. Simulation results show that while the proposed algorithm basically offers the same performance as the original MLSE performance, the VA is much more efficient than the conventional approach.

In this paper, we present several unitary transform-domain filtering techniques based on Karhaunen-Loeve Transform (KLT) for narrow-band interference rejection in CDMA communication systems. The reason for selecting the KLT is that it is an optimum unitary transform in the sense of packing the energy of the narrow-band interference. As a result after applying this transform, a small portion of the transformed signal would be interfered by the narrow-band interference, and thus must be set to zero. Due to unavailability of the optimum transform (KLT), several sub-optimum transforms are presented and their performances are compared with the well-known conventional transform methods such as Discrete Fourier Transform (DFT) and Discrete Cosine Transform (DCT) in the presence of both Auto Regressive (AR) and sinusoidal narrow-band interference. Our simulation results show that the proposed transform methods significantly outperform the conventional methods.

Fast packet switches for variable-size packets have become an everyday necessity with the rapid growth in the volume of Internet traffic. Such switches can be designed in two different ways, either by segmenting packets into smaller fixed-size cells and designing packet switches for such cells, or by designing generic packet switches for variable-size packets, where packet segmentation and reassembly can be omitted. The second option is investigated in this paper. The synchronous operation mode with time-limited bulk service is selected. The switching fabric is assumed to be internally non-blocking and provided with input queues. A previous maximum switch throughput analysis has been done under the assumption that the length of the time slot is fixed set to its minimum allowed value (T_min). In this work, a so-called time-slot stretch factor (SF) is introduced. The actual time-slot length is determined by multiplying T_min with the SF, where SF. Next, a so-called Internet traffic-source model is proposed based on findings from real IP traffic measurements. The performance implications of the proposed time-slot length modification are analyzed by discrete-event computer simulation. The maximum switch throughput is increased by increasing the SF value, e.g. for uniform packet size distribution and SF=10, the maximum switch throughput is increased from 75% to 97%. The influence of the traffic-source characteristics on the maximum switch throughput is decreased when SF value is increased. In order to prevent any possible throughput degradations, it is advisable to use integer SF values. Packet delay analysis has revealed that by increasing the SF value, the mean packet delay is also increased. Nevertheless, it is shown that the number of switch input and output ports is the most important factor to be considered when packet delay is at stake. Service class differentiation inside investigated packet switch is possible and is not affected by the increasing SF value. Such a packet switch is suitable for implementation in wide area networks, due to high transmission speeds and the small number of switch ports.

The future third generation mobile communications system, named IMT-2000, is expected to provide mobile users with voice, high-rate data and their combined multi-media services with the same QoS as in the fixed networks. As a radio access standard for the IMT-2000, W-CDMA and cdma2000 have been selected in Europe and North America, respectively. In this paper, we present an analytic model of the cdma2000 data mobile servicing a connected data service. In order to do this, we first model the traffic generated at mobile by a discrete-time Batch Markovian Arrival Process (D-BMAP). Next, we model the Radio Link Protocol (RLP) Queue in the cdma2000 MAC protocol by a D-BMAP/D/1 queueing system with batch service and setup times. Finally, we analyze this queueing system and get the performance measures such as the mean delay and the loss probability. Analytic results are compared with simulation ones for accuracy.

In this paper we addresses the problem of finding feasible solutions for the Group Multicast Routing Problem (GMRP). This problem is a generalization of the multicast routing problem whereby every member of the group is allowed to multicast messages to other members from the same group. The routing problem involves the construction of a set of low cost multicast trees with bandwidth requirements for all the group members in the network. We first prove that the problem of finding feasible solutions to GMRP is NP-complete. Following that we propose a new heuristic algorithm for constructing feasible solutions for GMRP. Simulation results show that our proposed algorithm is able to achieve good performance in terms of its ability of finding feasible solutions whenever one exist.

The Quality of Service (QoS) reservations in Differentiated Service (DiffServ) networks can be classified into two sets: Book-ahead (BA) requests and Instantaneous Requests (IRs). When an admitted BA request becomes active, some ongoing IRs is dropped when the bandwidth is insufficient for supporting both IRs and BA requests. The admission control should predict the lifetime, i.e. look-ahead time, of the IRs to prevent the admitted IRs from being dropped. The control should then check whether the available bandwidth during the look-ahead time is sufficient for the incoming IRs. We propose an application-aware look-ahead admission control for IRs, which determines the look-ahead time for specific types of IR applications. An admitted BA request might block subsequent ones that could bring more effective revenue. Thus, we propose the deferrable model of the admission control for BA requests. Simulation results indicate that the application-aware look-ahead admission control successfully reduces the dropping probability and wasted revenue of IRs by up to 10 times and 30%, respectively. Besides, the deferrable model indeed results in more BA effective revenue.

We present a simple framework for multicasting video in an active network, in which we overcome heterogeneity in the quality requests by filtering the video stream at some properly located active nodes. The framework includes the requirements for the underlying active network and outlines the video multicast application. We then introduce a heuristic algorithm for electing the filtering nodes to conform a multicast distribution tree, in which we use an objective function to, for example, minimize the required bandwidth. We evaluate the performance of our algorithm comparing it with simulcast and layered encoded transmission through simulation experiments, showing some advantages of using active filtering.

In this paper, we propose an algorithm to calculate the higher moments of the busy period length of a discrete-time M/G/1 type queue with finite buffer. The queueing model has a level-dependent transition probability matrix. Our algorithm is given as a set of recursive formulas which are derived from the relationship among the generating function matrices of the fundamental period. As an example of our algorithm, we provide an approximate analysis of a HOL (Head Of Line) priority control queue.

We proposed an improved Hierarchical Packet Fair Queueing (H-PFQ) mechanism, using ACK Spacing, for efficient bandwidth management of TCP traffic over Internet. According to the pre-determined bandwidth sharing and the class hierarchy of all TCP sessions, we design an algorithm to calculate the required time intervals between consecutive ACK packets of each TCP session to avoid packet drops due to buffer overflow. We demonstrated via computer simulations that the proposed improvement techniques may result in much better performance than merely original H-PFQ mechanism used in the forward direction in the sense that not only effective throughput of the bottleneck link is improved but also the fairness among TCP sessions can be maintained.

In the previous work, Lampson et al. proposed an IP lookup algorithm which performs binary search on prefixes (BSP). The algorithm is attractive, even for IPv6, because of its bounded worst-case memory requirement. To achieve fast forwarding, it may need to slow down the insertion speed. Although this can be justified, the routing-table reconstruction in BSP is too time-consuming to handle the frequent route updates. In this work, we propose a fast forwarding-table construction algorithm which can accomplish more than 4,000 route updates per second. Moreover, it is simple enough to fulfill the need of fast packet forwarding. With the enhanced multiway search tree, we further reduced the depth of the tree and eliminated the pointer storage; this reduces the forwarding table size and shortens the lookup time.

In this paper, we propose a novel electronic bearer check system (EBC). This system allows the consumer to pay any amount of money below an upper-boundary on the Internet within an expiration period. During each transaction, the consumer does not need to contact the bank's server. Furthermore, this electronic bearer check can be transferred to any third party. The off-line characteristic of our system is very convenient for the consumer. Moreover, the double spending and double depositing problem will not occur in this system. More importantly, the framework of this system provides anonymity to protect customer privacy.

The backward direction Soft Output Viterbi Algorithm (a backward SOVA) is compared with the conventional SOVA (a forward SOVA) in turbo code decoding. We find noticeable performance improvement for the backward SOVA when it is not terminated, which turns out to be due to a smaller reliability value, indicating that the termination conditions of the turbo encoder strongly affect the performance of the backward SOVA decoder. We also propose a hardware efficient serially mixed SOVA decoder composed of a forward SOVA decoder and a backward SOVA decoder. Simulation results show that the proposed serially mixed SOVA decoder has a 0.2 dB coding gain at 2.0 dB E_b/N_o over the forward SOVA for a typical turbo code example.

We propose a call admission control (CAC) scheme for the reverse link of direct sequence-code division multiple access (DS-CDMA) systems with multi-class traffic, in which the admissibility of the set of requested channels is decided by checking the outage probability of the total composite power at a cell-site receiver. The reverse link capacities under various traffic conditions are evaluated. From numerical results, we see that the proposed scheme can utilize a given radio resource more effectively as compared with the existing scheme using constraints on the individual power levels.

The frequency- and time-domain expressions are derived for the signal-to-noise power ratio (SNR) of an ideal Rake combiner output in a direct sequence spread spectrum (DS-SS) mobile communication system. The derived SNR expressions make it possible to estimate the SNR statistics after Rake combining for an arbitrary spreading chip rate in the frequency-selective multipath channel.

The reverse link signal power required for multimedia traffic in multipath faded single-code (SC-) and multi-code CDMA (MC-CDMA) systems is investigated. The effect of orthogonality loss among multiple spreading code channels is characterized by introducing the orthogonality factor. The required signal power in both CDMA systems is analyzed with varying system parameters of spreading bandwidth, the orthogonality factor, and the number of spreading codes. Analytical results show that MC-CDMA users transmitting only a single traffic type require significantly more power than SC-CDMA users with only a single traffic type. On the other hand, MC-CDMA users transmitting multimedia traffic require power levels approximately identical to SC-CDMA users with multimedia traffic.

Under cutoff and threshold priority policies, we mathematically formulate a prioritized channel allocation problem which is combinatorial in nature. We then reduce that problem using the concept of pattern, and apply a simulated annealing approach to the reduced problem. Computational experiments show that our method works very well and the cutoff priority policy outperforms the non-prioritized complete sharing policy and the threshold priority policy.

A high isolation hybrid circuit composed of a pair of transformers, a voltage control resistance (VCR) circuit and an automatic impedance control device is designed for a two-wire full duplex cable modem to adapt variable line impedance. A binary frequency shift keying (BFSK) cable modem using the new hybrid circuit with an isolation of 52 dB to 58 dB in the line impedance variation range of 400 to 950 ohm is demonstrated. The isolation of the new hybrid circuit is increased by more than 30 dB over the traditional hybrid circuit for a two-wire full duplex modem in the preset line impedance range.

We propose a new and fast full search (FS) motion estimation algorithm for video coding. The computational reduction comes from sequential rejection of impossible candidates with derived formula and subblock norms. Our algorithm reduces more the computations than the recent fast full search (FS) motion estimation algorithms.