WCDMA-HSDPA技术研究及改进方案

本文将证明HSDPA缺乏技术理论基础,利用现有的CDMA基本原理无法实现HSDPA应该达到的目标:在小区内利用一个载波给多个用户同时提供高速数据和语音业务的应用,显然此时要求提供的下行业务容量应比单独提供语音业务时更大.同时给出一种利用码分多址/时分多址改进HSDPA的方案,它将利用码分多址区分语音和高速数据业务信道,在语音和高速数据业务信道中分别选用码分和时分多址方式.这种方法可改进WCDMA标准上下行容量基本对称的技术缺陷,使其符合移动因特网的需求.     

Cross-layer resource allocation for integrated Voice/Data traffic in wireless cellular networks

A major task in next-generation wireless cellular networks is provisioning of quality of service (QoS) over the bandwidth limited and error-prone wireless link. In this paper, we propose a cross-layer design scheme to provide QoS for voice and data traffic in wireless cellular networks with differentiated services (DiffServ) backbone. The scheme combines the transport layer protocols and link layer resource allocation to both guarantee the QoS requirements in the transport layer and achieve efficient resource utilization in the link layer. Optimal resource allocation problems for voice and data flows are formulated to guarantee pre-specified QoS with minimal required resources. For integrated voice/data traffic in a cell, a hybrid time-division/code-division medium access control (MAC) scheme is presented to achieve efficient multiplexing. Theoretical analysis and simulation results demonstrate the effectiveness of the proposed cross-layer approach.     

Performance analysis of a packet-switched synchronous voice/datatransmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system     

Optimal resource allocation and adaptive call admission control for voice/data integrated cellular networks

Resource allocation and call admission control (CAC) are key management functions in future cellular networks, in order to provide multimedia applications to mobiles users with quality of service (QoS) guarantees and efficient resource utilization. In this paper, we propose and analyze a priority based resource sharing scheme for voice/data integrated cellular networks. The unique features of the proposed scheme are that 1) the maximum resource utilization can be achieved, since all the leftover capacity after serving the high priority voice traffic can be utilized by the data traffic; 2) a Markovian model for the proposed scheme is established, which takes account of the complex interaction of voice and data traffic sharing the total resources; 3) optimal CAC parameters for both voice and data calls are determined, from the perspective of minimizing resource requirement and maximizing new call admission rate, respectively; 4) load adaption and bandwidth allocation adjustment policies are proposed for adaptive CAC to cope with traffic load variations in a wireless mobile environment. Numerical results demonstrate that the proposed CAC scheme is able to simultaneously provide satisfactory QoS to both voice and data users and maintain a relatively high resource utilization in a dynamic traffic load environment. The recent measurement-based modeling shows that the Internet data file size follows a lognormal distribution, instead of the exponential distribution used in our analysis. We use computer simulations to demonstrate that the impact of the lognormal distribution can be compensated for by conservatively applying the Markovian analysis results.     

Self-adaptive transmission scheme of integrated services over an IEEE 802.11 WLAN

A novel self-adaptive transmission scheme to support integrated data and voice transmission over an IEEE 802.11 WLAN is proposed. The simulation results show that the scheme can improve the data traffic performance and decrease efficiently the voice traffic delay jitter, and then increase the WLAN capacity. It is easy to realise as no change in the MAC protocol is required.     

Performance evaluation of a CDMA protocol for voice and dataintegration in personal communication networks

Future wireless personal communication networks (PCN's) will require voice and data service integration on the radio link. The multiaccess capability of the code-division multiple-access (CDMA) technique has been widely investigated in the recent literature. The aim of this paper is to propose a CDMA-based protocol for joint voice and data transmissions in PCN's. The performance of such a protocol has been derived by means of an analytical approach both in terms of voice packet dropping probability and mean data packet delay. Voice traffic has been modeled as having alternated talkspurts and silences, with generation of voice packets at constant rate during talkspurts and no packet generation during silence gaps. A general arrival process is assumed for the data traffic. However, numerical results are derived in the case of a Poisson process. Simulation results are given to validate our analytical predictions. The main result derived here is that the proposed CDMA-based protocol efficiently handles both voice and data traffic. In particular, it is shown that the performance of the voice subsystem is independent of the data traffic     

A Medium Access Control Protocol for Voice/Data Integrated Wireless CDMA Systems

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay‐sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.     

An adaptive reservation scheme for VBR voice traffic in wireless ATM networks

A reservation scheme, named dynamic hybrid partitioning, is proposed for the Medium Access Control (MAC) protocol of wireless ATM (WATM) networks operating in Time Division Duplex (TDD) mode. The goal is to improve the performance of the real-time Variable Bit Rate (VBR) voice traffic in networks with mixed voice/data traffic. In most proposed MAC protocols for WATM networks, the reservation phase treats all traffic equally, whether delay-sensitive or not. Hence, delay-sensitive VBR traffic sources have to compete for reservation each time they wake up from idle mode. This causes large and variable channel access delays, and increases the delay and delay variation (jitter) experienced by ATM cells of VBR traffic. In the proposed scheme, the reservation phase of the MAC protocol is dynamically divided into a contention-free partition for delay-sensitive idle VBR traffic, and a contention partition for other traffic. Adaptive algorithms dynamically adjust the partition sizes to minimize the channel bandwidth overhead. Simulation results show that the delay performance of delay-sensitive VBR traffic is improved while minimizing the overhead.     

Analysis and engineering of a voice/data packet multiplexer

This paper investigates performance and engineering issues concerning a multiplexer scheme that has been implemented in AT&T's Integrated Access Terminal (IAT) to transport packetized voice and data traffic on shared facilities. The multiplexer serves voice and data traffic according to a dynamic bandwidth allocation scheme in order to simultaneously meet their performance requirements. A bit-dropping procedure is employed for voice packets to provide a graceful degradation of voice quality under overload conditions. An analytical model is developed for the multiplexer service scheme that estimates performance parameters given the voice and data offered loads. The model is used to demonstrate the capacity advantages of dynamic bandwidth allocation, and to generate load-service curves that illustrate the tradeoffs of carrying different combinations of voice and data traffic on the multiplexer. Sensitivity of voice and data performance to the multiplexer time-slice parameters is also investigated. The model is readily embedded in a design approach that determines the bandwidth required to carry the voice and data traffic demands while satisfying all desired performance objectives     

Performance analysis for voice/data integration on a finite-buffermobile system

Personal communication service (PCS) networks offer mobile users diverse telecommunication applications, such as voice, data, and image, with different bandwidth and quality-of-service (QoS) requirements. This paper proposes an analytical model to investigate the performance of an integrated voice/data mobile network with finite data buffer in terms of voice-call blocking probability, data loss probability, and mean data delay. The model is based on the movable-boundary scheme that dynamically adjusts the number of channels for voice and data traffic. With the movable-boundary scheme, the bandwidth can be utilized efficiently while satisfying the QoS requirements for voice and data traffic. Using our model, the impact of hot-spot traffic in the heterogeneous PCS networks, in which the parameters (e.g., number of channels, voice, and data arrival rates) of cells can be varied, can be effectively analyzed. In addition, an iterative algorithm based on our model is proposed to determine the handoff traffic, which computes the system performance in polynomial-bounded time. The analytical model is validated by simulation     

Simulation analysis of a communication link with statisticallymultiplexed bursty noise sources

Variable bit rate (VBR) coding techniques have received great research interest as very promising tools for transmitting bursty multimedia traffic with low bandwidth requirements over a communication link. Statistically multiplexing the multimedia bursty traffic is a very efficient method of maximizing the utilization of the link capacity. The application of computer simulation techniques in analyzing a rate-based access control scheme for multimedia traffic such as voice traffic is discussed. The control scheme regulates the packetized bursty traffic at the user network interface of the link. Using a suitable congestion measure, namely, the multiplexer buffer length, the scheme dynamically controls the arrival rate by switching the coder to a different compression ratio (i.e., changing the coding rate). VBR coding methods can be adaptively adjusted to transmit at a lower rate with very little degradation in the voice quality. Reported results prove that the scheme greatly improves the link performance, in terms of reducing the probability of call blocking and enhancing the statistical multiplexing gain     

A new protocol for the integration of voice and data over PRMA

In packet reservation multiple access (PRMA) the receiver in the mobile terminal is required to listen continuously to monitor the acknowledgment messages broadcasted at the end of every time slot. A new scheme for the integration of voice and data based on PRMA is proposed. The voice and the data subsystems are logically separated. The total available bandwidth is divided into three regions-voice information, voice contention, and data regions. The available bandwidth is dynamically partitioned between the above three regions subject to the fulfillment of the quality of service (QoS) requirements of the voice users. The voice subsystem has been modeled as a Markov chain and an exact analytical method used to compute the voice packet dropping probability is described. A nonlinear programming problem is formulated to optimize the bandwidth allocated for the data users. Solutions to this nonlinear programming problem that are very close to optimum have been obtained heuristically. Numerical results indicate that a significant amount of data traffic can be supported without sacrificing the voice capacity of the system     

Wireless access to the World Wide Web in an integrated CDMA system

We propose a wireless access mechanism for web traffic packets in an integrated wireless code-division multiple-access system that has both voice and Web traffic. The proposed scheme is a medium-access control layer/link layer (MAC/LL) scheduling algorithm that consists of a two level control: admission control and packet scheduling. The admission control restricts the number of users in the system such that quality-of-service requirements [target signal-to-interference ratio (SIR) and delay] for both voice and Web traffic can be met. The packet scheduling balances the system interference on a slot-by-slot basis such that the target SIRs can be achieved for all users (voice and Web browsing sessions) with a higher scheduling priority for voice. Designing admission control for Web users based on the average offered rate per session is difficult due to the high variations in the offered load generated by heavy tailed distributions for Web traffic. To overcome this problem, we propose an admission control algorithm that adaptively estimates the aggregate average load based on load measurements using a sliding observation window.     

Call admission control for integrated on/off voice and best-effort data services in mobile cellular communications

This paper proposes a call admission control (CAC) policy for a cellular system supporting voice and data services, and providing a higher priority to handoff calls than to new calls. A procedure for searching the optimal admission region is given. The traffic flow is characterized by a three-dimensional (3-D) birth-death model, which captures the complex interaction between the on/off voice and best-effort data traffic sharing the total resources without partition. To reduce complexity, the 3-D model is simplified to an exact (approximate) 2-D model for voice (data). The mathematical expressions are then derived for the performance measures and for the minimal amount of resources required for quality-of-service (QoS) provisioning. Numerical results demonstrate that: 1) the proposed CAC policy performs well in terms of QoS satisfaction and resource utilization; 2) the approximate 2-D model for data traffic can achieve a high accuracy in the traffic flow characterization; and 3) the admission regions obtained by the proposed search method agree very well with those obtained by numerically solving the mathematical equations. Furthermore, computer simulation results demonstrate that the impact of lognormal distributed data file size is not significant, and may be compensated by conservatively applying the Markovian analysis results.     

Analysis of a polling system for telephony traffic with application to wireless LANs

Recently, polling has been included as a resource sharing mechanism in the medium access control (MAC) protocol of several communication systems, such as the IEEE 802.11 wireless local area network, primarily to support real-time traffic. Furthermore, to allow these communication systems to support multimedia traffic, the polling scheme often coexists with other MAC schemes such as random access. Motivated by these systems, we develop a model for a polling system with vacations, where the vacations represent the time periods in which the resource sharing mechanism used is a non-polling mode. The real-time traffic served by the polling mode in our study is telephony. We use an on-off Markov modulated fluid (MMF) model to characterize telephony sources. Our analytical study and a counterpart validating simulation study show the following. Since voice codec rates are much smaller than link transmission rates, the queueing delay that arises from waiting for a poll dominates the total delay experienced by a voice packet. To keep delays low, the number of telephone calls that can be admitted must be chosen carefully according to delay tolerance, loss tolerance, codec rates, protocol overheads and the amount of bandwidth allocated to the polling mode. The effect of statistical multiplexing gain obtained by exploiting the on-off characteristics of telephony traffic is more noticeable when the impact of polling overhead is small.     

Radio resource management for cellular CDMA systems supporting heterogeneous services

A novel radio resource management (RRM) scheme for the support of packet-switched transmission in cellular CDMA systems is proposed by jointly considering the physical, link, and network layer characteristics. The proposed resource management scheme is comprised of a combination of power distribution, rate allocation, service scheduling, and connection admission control. Power distribution allows individual connections to achieve their required signal-to-interference-plus-noise ratio, while rate allocation guarantees the required delay/jitter for real-time traffic and the minimum transmission rate requirement for non-real-time traffic. Efficient rate allocation is achieved by making use of the randomness and burstiness; of the packet generation process. At the link layer, a packet scheduling scheme is developed based on information derived from power distribution and rate allocation to achieve quality of service (QoS) guarantee. Packet scheduling efficiently utilizes the system resources in every time slot and improves the packet throughput for non-real-time traffic. At the network layer, a connection admission control (CAC) scheme based on the lower layer resource allocation information is proposed. The CAC scheme makes use of user mobility information to reduce handoff connection dropping probability (HCDP). Theoretical analysis of the grade of service performance, in terms of new connection blocking probability, HCDP, and resource utilization, is given. Numerical results show that the proposed RRM scheme can achieve both effective QoS guarantee and efficient resource utilization.     

Managing bandwidth in ATM networks with bursty traffic

Three approaches to the bandwidth management problem that have been proposed and studied by various groups are reviewed to illustrate three distinctly different approaches and identify their strengths and weaknesses. Based on these approaches, a bandwidth management and congestion control scheme for asynchronous transfer mode (ATM) networks that supports both point-to-point and one-to-many multicast virtual circuits is proposed. It is shown that the method can handle fully heterogeneous traffic and can be effectively implemented. The algorithm for making virtual circuit acceptance decisions is straightforward and fast, and the hardware mechanisms needed to implement buffer allocation and traffic monitoring at the user-network interface have acceptable complexities. It is also shown, through numerical examples, that the approach can achieve reasonable link efficiencies even in the presence of very bursty traffic. No advance reservation required, simplifying the interface between the network and the user and avoiding an initial network round trip delay before data can be transmitted     

Analysis of Bandwidth Gain Over Various Timer$_$CU of AAL2 for Voice Traffic Multiplexing

An asynchronous transfer mode adaptation layer type 2 (AAL2) transmission scheme commonly is used to deliver the voice and tha data traffic between Node-B and the radio network controller on the universal mobile terrestrial device network. To predict the AAL2 multiplexing performance, we analyzed the bandwidth gain and the cell-packing density using discrete Markov chain model for the voice service and validated these results with simulations. We also performed a detailed simulation for the voice and the data services in a concentrator. Based on the analysis, we proposed an engineering guideline for selecting the optimal Timer$_$CU in a Node-B. We found that there is no major benefit in using the AAL2 multiplexing in a concentrator. The benefit of the AAL2 multiplexing in$ I_ ub$for the data service was much less than that for the voice service. They also depended heavily on the traffic load.     

A Distributed Channel Access Scheme with Guaranteed Priority and Enhanced Fairness

Although the IEEE 802.11e enhanced distributed channel access (EDCA) can differentiate high priority traffic such as real-time voice from low priority traffic such as delay- tolerant data, it can only provide statistical priority, and is characterized by inherent short-term unfairness. In this paper, we propose a new distributed channel access scheme through minor modifications to EDCA. Guaranteed priority is provided to real time voice traffic over data traffic, while a certain service time and short-term fairness enhancement are provided to data traffic. We also present analytical models to calculate the percentage of time to serve voice traffic and the achieved data throughput. Both analysis and simulation demonstrate the effectiveness of our proposed scheme.     

Packet telephony support for the IEEE 802.11 wireless LAN

Packet telephony is one of the most promising applications in the Internet. In this paper, we propose a modified MAC protocol supporting voice traffic over the IEEE 802.11 WLAN. The proposed scheme adapts the power-saved mode of the IEEE 802.11 specifications in such a way that it approaches the TDM access mode carrying voice traffic, and is compatible with the IEEE 802.11 standard. Simulation results show that the proposed scheme does not degrade the performance of the IEEE 802.11 WLAN using the DCF and also provides good voice quality