CDMA网络中多媒体业务的接入控制

从码分多址(CDMA)系统的特点出发，采用多维马尔可夫过程来分析多媒体业务共存时基于干扰水平的接入控制策略，得到在小区容量可变情况下的系统性能。文中考虑到切换用户的优先级以及不同类型用户的不同QoS要求，还在系统中设置了多级接入门限，从而实现在扩大系统容量的同时，又力求保证系统已有的链路质量，实现接入控制的目的。     

Call admission control in wireless multimedia networks

Call admission control (CAC) is a mechanism used in networks to administer quality of service (QoS). Whereas the CAC problem in time-division multiple access (TDMA)-based cellular networks is simply related to the number of physical channels available in the network, it is strongly related to the physical layer performance in code-division multiple access (CDMA) networks since the multi-access interference in them is a function of the number of users and is a limiting factor in ensuring QoS. In this article, the CAC issues in multimedia DS-CDMA systems are reviewed by illustrating the basic principles underlying various schemes that have been proposed progressively from the simplest to the complex. The article also introduces SIR as a measure of QoS and describes the relatively simple schemes to administer CAC. The expression for SIR resulting from linear minimum mean-squared error processing is also presented. This article illustrates how CAC for multiple class service can be casted into an optimality framework and then discuss the recent work addressing self-similar multiple access interference.     

Effective bandwidth of multimedia traffic in packet wireless CDMA networks with LMMSE receivers: a cross-layer perspective

We propose a novel concept of cross-layer effective bandwidth that characterizes the unified resource usage taking into account both physical layer linear minimum mean square error (LMMSE) receivers and statistical characteristics of the packet traffic in code division multiple access (CDMA) networks. Based on the concept of cross-layer effective bandwidth, we develop an optimal connection admission control (CAC) scheme for variable bit rate packet traffic with QoS constraints at both physical and network layers. By introducing a small signal-to-interference ratio (SIR) outage probability using the concept of cross-layer effective bandwidth, the capacity of CDMA networks in the proposed CAC scheme can be increased significantly compared to some existing schemes. The effectiveness of the proposed approaches is demonstrated by numerical examples.     

QoS-oriented packet scheduling for wireless multimedia CDMA communications

In the third-generation (and beyond) wireless communication systems, there will be a mixture of different traffic classes, each having its own transmission rate characteristics and quality-of-service (QoS) requirements. In this paper, a QoS-oriented medium access control (MAC) protocol with fair packet loss sharing (FPLS) scheduling is proposed for wireless code-division multiple access (CDMA) communications. The QoS parameters under consideration are the transmission bit error rate (BER), packet loss, and delay requirements. The MAC protocol exploits both time-division and code-division statistical multiplexing. The BER requirements are guaranteed by properly arranging simultaneous packet transmissions and controlling there transmit power levels, whereas the packet loss and delay requirements are guaranteed by proper packet scheduling. The basic idea of FPLS is to schedule the transmission of multimedia packets in such a way that all the users have a fair share of packet loss according to their QoS requirements, which maximizes the number of the served users under the QoS constraints. Simulation results demonstrate effectiveness of the FPLS scheduler, in comparison with other previously proposed scheduling algorithms.     

Jointly optimal power and admission control for delay sensitive traffic in CDMA networks with LMMSE receivers

Quality-of-service (QoS) guarantees for multiclass code division multiple access networks are provided by means of cross-layer optimization across the physical and network layers. At the physical layer, the QoS requirements are specified in terms of a target signal-to-interference ratio (SIR) requirement, and optimal target powers are dynamically adjusted according to the current number of users in the system. At the network layer, the QoS requirements are the blocking probabilities and the call connection delays. The network layer guarantees that both physical layer and network layer QoS are met by employing admission control. An optimal admission control policy is proposed based on a semi-Markov decision process formulation. The tradeoff between blocking and delay is discussed for various buffer configurations. The advantage of advanced signal processing receivers is established using a comparative capacity analysis and simulation with the classical scenario in which the system uses matched filter receivers.     

Scheduling in multimedia CDMA wireless networks

Wireless systems in the future will have to provide multimedia services where different users have different physical-layer quality of service (QoS) requirements (e.g., bit energy per interference power density, E/sub b//N/sub 0/, or bit error rate and power constraints) and network-layer QoS requirements (e.g., delay bound, delay-jitter, throughput, and loss). We investigate the use of power control, processing gain and/or multiple codes, and scheduling in CDMA systems to accommodate these diverse service requirements. We first show that the instantaneous capacity region, given in terms of the set of user bit rates that can be supported simultaneously subject to peak power and E/sub b//N/sub 0/ constraints, is nonconvex. This suggests that by time-sharing the channel, one may be able to get better system throughput. We define the capacity region as the convex hull of the instantaneous capacity region and we show that it may be obtained by time sharing between operating points, where each user either uses full power or is silent (bang-bang control). We then consider the problem of scheduling so as to meet prespecified delay bounds or minimum service curve requirements for traffic streams, which are specified in terms of a traffic profile such as a sigma-rho constraint (enforced by a leaky bucket) and a guarantee that the system is stable.     

A call admission control strategy for multiservice wireless cellular packet networks

A simple connection control system for multiservice cellular wireless networks is presented. Mobile stations are classified depending on the traffic they generate (e.g., voice, data). Within each class, two subclasses are also identified: stations which have originated inside the cell and stations which come from adjacent cells. The connection control mechanism is carried out by considering a number of priorities among the various classes and their subclasses. It works on two levels: static and dynamic. The static level looks at packet-level quality of service (QoS), such as cell loss and delay, while the dynamic level takes care of connection dynamics and allows the load of the system to be driven with respect to the various subclasses. Results that illustrate the performance of this control mechanism are presented.     

Predictive QoS-based admission control for multiclass traffic incellular wireless networks

We develop the notion of quality of service (QoS) for multimedia traffic in terms of maximum call dropping probabilities independent of system load and a predefined call blocking probability profile for the different traffic classes for wireless networks of arbitrary shape and dimension. We describe two distributed predictive admission control algorithms, independent multiclass one-step prediction (IMOSP-CS and IMOSP-RES), which provide each traffic class with a given call dropping probability and a desired call blocking probability profile. Both algorithms may be seen as extensions of the multimedia one-step prediction (MMOSPRED) algorithm previously reported, which uses prediction of the overload probability in the home and neighbor cells in deciding whether to admit new users into a multiclass cellular system. The two algorithms differ in their approach to handoff call admission. The first algorithm completely shares the bandwidth among the entering handoff users while the second implements a partition-based reservation scheme. In this paper, we additionally impose a call blocking criterion that ensures a system-imposed call priority independent of the traffic in the system and which adapts to changes in the offered load. In comparing these algorithms to each other, we focus on system throughput and class independence. Both algorithms provide appropriate throughput under both homogeneous and heterogeneous traffic loading conditions while maintaining steady call dropping probabilities for each traffic class     

End-to-end performance models for variable bit rate voice overtandem links in packet networks

Analytical models are presented for computing the end-to-end voice call performance in a packet network that drops the less significant bits in voice packets during periods of congestion. These models provide information about the end-to-end quality likely to be experienced in future packet-switched integrated services networks. An existing single-node bit-dropping model is modified to include the situation resulting when the overall arrival process at an internal node consists of a mix of packets of different sizes due to bit dropping at previous nodes. A detailed model to capture bit-dropping effects in a tandem connection of nodes is presented. The model includes the effect of load fluctuations at each node, and also takes into account the dependencies in bit dropping experienced by a voice packet at successive nodes in a tandem connection. The model also incorporates the internodal dependence when reductions in packet service times occur at intermediate nodes due to bit dropping at previous nodes. Two approximation procedures are discussed that serve as upper and lower bounds. In particular, the upper bound is shown to be very tight for a practical range of loads, and hence serves as a good approximation with significant computational simplicity     

Ratioed power and rate control for CDMA wireless networks

We propose in this paper the ratioed power and rate control (RPRC) algorithm to satisfy the requirements of both the data rate and the signal quality. During the execution of the RPRC algorithm, only the representative user in each cell adjusts its power level according to certain distributed power control algorithm, and the ratio of the power level for the non‐representative user to that for the representative user is kept constant. When the RPRC algorithm is finished, the power levels for all users remain unchanged and the transmission rate is determined by the available rate. We show that the RPRC algorithm can be used for allocating the bandwidth of each cell and achieving the required rate of each user. Furthermore, simulation results reveal that the RPRC algorithm results in larger throughput and finds a feasible power set faster than the other algorithm. Copyright © 2008 John Wiley & Sons, Ltd.     

Realistic cell-oriented adaptive admission control for QoS support in wireless multimedia networks

An important quality-of-service (QoS) issue in wireless multimedia networks is how to control handoff drops. We propose admission-control algorithms that adaptively control the admission threshold in each cell in order to keep the handoff-dropping probability below a predefined level. The admission threshold is dynamically adjusted based on handoff-dropping events. We first present a simple admission-control scheme that brings out an important performance evaluation criterion - intercell fairness - and serves as a reference point. We then investigate the intercell unfairness problem and develop two enhanced schemes to overcome this problem. The performance of these protocols is benchmarked and compared with other competitive schemes. The results indicate that our schemes perform very well while, in addition, achieving significantly reduced complexity and signaling load.     

Performance evaluation of wireless multimedia CDMA networks usingadaptive transmission control

When multimedia information including speech, image, data etc. are transmitted, the difference in information rates, required quality, and traffic performance should be taken into account. A wireless spread spectrum system can achieve a flexible balance of these differences because of inherent asynchronous code-division multiple-access (CDMA) capability. In order to accomplish wireless multimedia communication, three parameters are investigated: the transmission power, packet length, and time slot. We propose and investigate two methods to accomplish wireless multimedia communication by using these parameters: (1) a method based on control of both transmission power and transmission packet length and (2) a method based on control of both transmission power and adaptive time slot. We derive the optimal transmission packet length and transmission power in (1) and derive optimal adaptive time slot and transmission power in (2) by using traffic of measurement and nonlinear programming. By the comparison, the two proposed methods are compared to the minislotted alternating priority (MSAP). Numerical results demonstrate that the proposed methods, when the priority of each medium is graded, show better performance than other systems in terms of average throughput and delay time of all media     

Power control for wireless multimedia CDMA systems

In a wireless multimedia communication system where different types of traffic such as speech, data, and video have different data rates, good power control for individual traffic can ensure quality of service (QoS) and achieve a better throughput at the same time. The authors propose a closed-form power control function based on a parameter called the traffic exponent that simplifies the power control processing and improves the overall CDMA system performance     

Routing for multimedia traffic in wireless frequency-hopcommunication networks

A new method is described for routing multimedia traffic in a frequency-hop (FH) store-and-forward packet radio network. The method is illustrated for traffic of two types, each type having its own throughput, delay, and error-rate requirements. A typical application is the routing of voice and data packets in a distributed multiple-hop network. In such an application, voice packets cannot tolerate much delay, but they are allowed to contain a small number of frame erasures while data packets must be delivered error-free even if a moderate delay is required to do so. The fully distributed routing protocol presented in the paper takes into account the type of service required for each type of traffic, and it adapts to the interference as seen by the FH radio receivers in the network. Our approach to multimedia routing is based on least-resistance routing with different link and path resistance metrics for different message types. Each of the resistance metrics for a link reflects the ability of the link to provide the service required by the one of the message types. This includes, but is not limited to, a measure of the likelihood of successful reception by the FH radio receiver for that link. The route selection for a particular type of packet depends on the resistances of the links along the routes from that packet's source to its destination. In general, different routes may be selected for different types of packets. The primary conclusion of this paper is that the quality of service increases for each of the two types of multimedia traffic if the routing protocol accounts for the type of message that is being relayed     

Nonlinear power assignment in multimedia CDMA wireless networks

In this letter, we propose a new technique for power assignment in outdoor code-division multiple access cell supporting bursty multimedia traffic. The proposed scheme takes into consideration both average signal-to-interference ratios and outage probability required by multimedia traffic     

Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

ARC: an integrated admission and rate control framework for competitive wireless CDMA data networks using noncooperative games

The competition among wireless data service providers brings in an option for the unsatisfied customers to switch their providers, which is called churning. The implementation of Wireless Local Number Portability (WLNP) is expected to further increase the churn rate (the probability of users switching the provider). However, the existing resource management algorithms for wireless networks fail to fully capture the far-reaching impact of this unforeseen competitiveness. From this perspective, we first formulate noncooperative games between the service providers and the users. A user's decision to leave or join a provider is based on a finite set of strategies. A service provider can also construct its game strategy set so as to maximize their utility (revenue) considering the churn rate. Based on the game theoretic framework, we propose an integrated admission and rate control (ARC) framework for CDMA-based wireless data networks. The admission control is at the session (macro) level while the rate control is at the link layer packet (micro) level. Two admission control modes are considered - one-by-one mode and batch processing mode, in which multiple users are admitted at a time. We show that: 1) for the one-by-one mode, the Nash equilibrium using pure strategy can be established for both under-loaded and fully-loaded systems and 2) for batch processing mode, there is either an equilibrium in pure strategy or a dominant strategy exists for the service provider. Therefore, the providers have clearly defined admission criteria as outcome of the game. Users are categorized into multiple classes and offered differentiated services based on the price they pay and the service degradation they can tolerate. We show that the proposed ARC framework significantly increases the provider's revenue and also successfully offers differentiated QoS to the users.     

Effective bandwidth approach to connection admission control formultimedia traffic in ATM networks

The authors propose an effective bandwidth approach to connection admission control in ATM networks. The aggregate arrival traffic is accurately modelled by a two-state Markov modulated Poisson process (MMPP) via the matching of four important statistics. If the buffer is large, admission control can be achieved by computing the effective bandwidth of the two-state MMPP. Simulation tests show that approach is simple and results in higher utilisation compared with conventional methods     

Modeling and call admission control algorithm of variable bit ratevideo in ATM networks

The authors propose a new method for the modeling and call admission control (CAC) of variable bit rate video source, which come to the front of ATM networks as hot issues nowadays. First, the modeling of video source is accomplished using the three-state Markov chains including the effects of scene change at which the bit rate of video source is abruptly increased. Also, using two AR models, they improve the defects which an AR model has in modeling a video source. In addition, they represent the analytical model of a video source so that a network manager can acquire the information which is very important in managing the entire networks. CAC is accomplished using the previously defined analytical model. A routing manager calculates the cell loss probability of a chosen VP where a new call is connected so that the routing manager decides whether this new call is accepted or not. This calculation is accomplished through the GB/D/1-S queuing system. Using BIA (bandwidth increasing algorithm), they check whether the calls rejected by the routing manager could be accepted if possible. Finally, the applicable procedures to suitable allocate bandwidth to each VP on a link are presented in detail