Virtual guard channel for handoff calls in integrated voice/data wireless networks

In this paper, we propose a virtual guard channel (VGC) scheme for handoff calls in integrated voice/data wireless networks. By utilizing the multi-channel capability of data service, the proposed scheme can provide better performance in quality of service (QoS) provisioning and utilize the limited channel resources more efficiently compared with the conventional guard channel (GC) scheme.     

PMCAP/CDMA协议在语音与数据混合业务中的应用研究

该文将CDMA移动通信系统上行信道中的一种MAC协议PMCAP／CDMA协议应用到语音与数据混合业务的情况，对协议建立数学模型，并对性能进行理论计算和仿真。为了保证对语音请求的优先分配，将PN码集分为语音可用码集，数据可用码集及语音与数据的预防码集。提出了新的动态码集分配方案。仿真表现，该方案较固定码集方案更好地提高了系统的综合性能，而负指数码字分配方案提供了语音与数据性能的很好折衷。     

Performance modeling and evaluation of data/voice services in wireless networks

Application-level performance is a key to the adoption and success of the CDMA 2000. To predict this performance in advance, a detailed end-to-end simulation model of a CDMA network is built to include application traffic characteristics, network architecture, network element details using the proposed simulation methodology. We assess the user-perceived application performance when a RAN and a CN adopt different transport architectures such as ATM and IP. To evaluate the user-perceived quality of voice service, we compare the end-to-end packet delay for different vocoder schemes such as G.711, G.726 (PCM), G.726 (ADPCM), and vocoder bypass scheme. By the simulation results, the vocoder bypass scenario shows 30% performance improvement over the others. We also compare the quality of voice service with and without DPS scheduling scheme. We know that DPS scheme keep the voice delay bound even if the service traffic is high. For data packet performance, HTTP v.1.1 shows better performance than that of HTTP v.1.0 due to the pipelining and TCP persistent connection. We may conclude that IP transport technology is better solution for higher FER environment since the packet overhead of IP is smaller than that of ATM for web browsing data traffic, while it shows opposite effect to the small size voice packet in RAN architecture. We show that the 3G-1X EV-DO system gives much better packet delay performance than 3G-1X RTT. The main conclusion is that end-to-end application-level performance is affected by various elements and layers of the network and thus it must be considered in all phases of the development process. Jae-Hyun Kim He received the B.S., M.S., and Ph.D. degrees, all in computer science and engineering, from Hanyang University, Ansan, Korea, in 1991, 1993, and 1996 respectively. In 1996, he was with the Communication Research Laboratory, Tokyo, Japan, as a Visiting Scholar. From April 1997 to October 1998, he was a post-doctoral fellow at the department of electrical engineering, University of California, Los Angeles. From November 1998 to February 2003, he worked as a member of technical staff in Performance Modeling and QoS management department, Bell laboratories, Lucent Technologies, Holmdel, NJ. He has been with the department of electrical engineering, Ajou University, Suwon, Korea, as an assistant professor since 2003. His research interests include QoS issues and cross layer optimization for high-speed wireless communication. Dr. Kim was the recipient of the LGIC Thesis Prize and Samsung Human-Tech Thesis Prize in 1993 and 1997, respectively. He is a member of the Korean Institute of Communication Sciences (KICS), Korea Institute of Telematics and Electronis (KITE), Korea Information Science Society (KISS), and IEEE. Hyun-Jin Lee received the B.S. degree in electrical engineering from Ajou University, Suwon, Korea, in 2004, and is working toward the M.S. degree and Ph. D. degree in electrical engineering at Ajou University. He has been awarded Samsung Human-Tech Thesis Prize in 2004. His research interests QoS, especially network optimization and wireless packet scheduling. He is a member of the KICS. Sung-Min Oh received the B.S. and M. S. degrees in electrical engineering form Ajou University, Suwon, Korea, in 2004, and is working toward the Ph. D. degree in electrical engineering at Ajou University. His research interests QoS performance analysis and 4G network. He is a member of the KICS. Sung-Hyun Cho received his B.S., M.S., and Ph.D. in computer science and engineering from Hanyang University, Korea, in 1995, 1997, and 2001, respectively. From 2001 to 2005, he has been with Samsung Advanced Institute of Technology, where he has been engaged in the design and standardization of MAC and upper layers of B3G, IEEE 802.16e, and WiBro systems. He is currently a MAC part leader in the telecommunication R&D center of Samsung Electronics. His research interests include 4G air interface design, radio resource management, cross layer design, and handoff in wireless systems.     

Performance analysis of a multiple access protocol for voice and data support in multiuser broadband wireless LANs

Wireless Local Area Networks (WLANs) is one of the most promising access technologies for the upcoming fourth-generation wireless communication systems. In the last few years, several research efforts have been devoted to investigate possible multiple access schemes capable of supporting real-time traffic as well as best-effort data transmissions. In particular, the use of suitable transmission schemes allows not only to achieve higher data-rates, but also to perform resource allocation in order to guarantee specific service requirements. In this paper we propose a medium access control (MAC) scheme for a WLAN supporting real-time (voice) and best-effort (data) services, based on Orthogonal Frequency Division Multiplexing (OFDM) technique. A suitable analytical approach is proposed in order to derive the performance of the proposed MAC scheme. In particular, it is shown in the paper that a high overall network capacity in terms of simultaneously active users is achieved by effectively exploiting the multiuser capabilities offered by OFDM, together with a proper service differentiation. Work partially supported by MIUR within the WOMEN project. Romano Fantacci, (M’87,SM’91, F’05) born in Pistoia, Italy, graduated from the Engineering School of the Universit di Firenze, Florence, Italy, with a degree in electronics in 1982. He received his Ph.D. degree in telecommunications in 1987. After joining the Dipartimento di Elettronica e Telecomunicazioni as an assistant professor, he was appointed associate professor in 1991 and full professor in 1999. His current research interests are digital communications, computer communications, queuing theory, satellite communication systems, wireless broadband communication networks, ad-hoc and sensor networks. He has been involved in several European Space Agency (ESA) and INTELSAT advanced research projects. He is the author of numerous articles published in prestigious communication science journals. He guest edited special issues in IEEE Journals and magazines and served as symposium chair of several IEEE conferences, including VTC, ICC and Globecom. Professor Fantacci received the IEE IERE Benefactor premium in 1990 and IEEE COMSOC Award Distinguished Contributions to Satellite Communications in 2002. He is currently serving as Editor for Telecommunication Systems, IEEE Trans. Commun. and IEEE Transactions on Wireless Communications. Gianluca Vannuccini born in Florence, Italy, graduated in Electronics Engineering in 1999. He received his Ph.D. degree in Telematics and Information Society in 2003 from the Electronics and Telecommunications Department of the University of Florence. During 2002 he was for six months in IBM Zurich Research Laboratory, Zurich, Switzerland, working in a research project on IEEE 802.11e performance evaluation. During the Ph.D. course, he has published on several IEEE conferences and served as reviewer for several journals on the telecommunications and telematics research area. He has been IEEE student member since 1999 and IEEE member since 2004. He is now with the IT department of the Florence local government organization, where he works as a program manager on data quality and integration and e-government IT projects. Gabriele Vestri was born in Florence (Italy) in November 1976. He received his degree in telecommunications from the Universit di Firenze, Florence, Italy, in March 2002. He has been research scientist for CSO Ophthalmic (Florence) since September 2002. His current research interests include ophthalmic instruments, contact lens design, image processing, the effects of optical aberrations and of retinal architecture of the eye on visual performance.     

An integrated voice/data system for mobile indoor radio networks

A centralized, integrated voice/data radio network for fading multipath indoor radio channels is proposed and analyzed. The packets of voice and data are integrated through a movable boundary method. The uplink channel access uses a framed-polling protocol whereas the downlink uses a time-division multiple-access (TDMA) scheme. This system dynamically switches between two transmission rates and uses multiple antennas to maximize the throughput in the fading multipath indoor environment. Throughput and delay characteristics of the system are analyzed using four different techniques. The results are compared with those of Monte Carlo computer simulations. A simple relationship between the number of voice terminals and the throughput of the data traffic are derived for an upper bound of 10-ms delay for the data packets     

Performance analysis of burst switching for integrated voice/dataservices

The voice cutoff fraction and mean data waiting time of an integrated burst-switched system are analyzed. Several approximate approaches are proposed for the analysis. Comparison between simulation and approximate results shows that the accuracy of the approximations depends on the relationship between μ_d and μ_v where 1/μ_d and 1/μ_v are the mean lengths of data and voice packets (bursts), respectively. In general, the approximations give good accuracy for μ_d≲μ_v. For μ_d>μ_v, the approximations for the mean data waiting time give less accurate but adequate results. From the analysis, it is found that a burst-switched system can achieve a higher bandwidth efficiency when data traffic has a shorter burst length (e.g. interactive data). Furthermore, it is also observed that the relative burst lengths between voice and data can significantly affect the characteristic of the voice clipping process. The analysis shows that, under normal operating condition where voice traffic is predominant and the constraint that the voice cutoff fraction is less than 0.5%, the data queue is always stable     

Call admission control for voice/data integrated cellular networks: performance analysis and comparative study

In this paper, we propose a new call admission control scheme called dual threshold bandwidth reservation, or DTBR scheme. The main novelty is that it builds upon a complete sharing approach, in which the channels in each cell are shared among the different traffic types and multiple thresholds are used to meet the specific quality-of-service (QoS) requirements. We present a detailed comparative study based on mathematical and simulation models, and quantitatively demonstrate that the DTBR is capable of providing the QoS guarantee for each type of traffic, while at the same time leading to much better channel efficiency. We further show that the DTBR scheme with elastic data service can offer both service guarantee and service differentiation for voice and data services, and enhance the bandwidth utilization.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Performance evaluation of a modified PRMA protocol for joint voiceand data packet wireless networks

In this paper, a modified version of the packet reservation multiple-access (PRMA) protocol is proposed to provide spatially dispersed voice and data user terminals wireless access to a base station over a common short-range radio channel. An analytical approach is presented in order to derive system performance in terms of mean data message delay and voice packet dropping probability. A suitable permission probability design is also proposed to enhance system performance. Performance comparisons with an extension of the PRMA protocol to voice data systems previously reported in literature are shown to highlight the better behavior of this approach     

Interference and SIR in integrated voice/data wireless DS-CDMAnetworks-a simulation study

The objective of this study is to characterize the multiple-access interference in a DS-CDMA integrated voice/data wireless network. The system model is designed to accommodate bursty packet-based data services in addition to stream-based services on a shared spectrum basis. A common packet data channel (CPDC) is employed to transport bursty data services, Consequently, this study incorporates features associated with packet-based services, such as higher transmitter power, short inter-arrival times, and short service time durations, as well as features associated with stream-based services, in order to provide a complete interference characterization. The results of a detailed and precise simulation study are presented in which the influence of traffic burstiness, voice activity, spatial distribution of mobiles, and transmitter powers on the fluctuation of the interference and signal-to-interference ratio (SIR) is assessed. The results obtained also quantify the impact of interference caused by CPDC services on stream-based services. The interference and SIR statistics are evaluated in terms of cumulative distribution functions. Since a DS-CDMA network is interference limited, studies of this type are essential for system design, capacity evaluation, and bandwidth management     

Economic analysis of integrated voice and data networks: A case study

This paper describes the results of a study to evaluate alternative switching strategies for future integrated voice and data networks. Three fundamental problems are addressed: 1) the economics of integrating voice and data applicatiom in a common communications system; 2) the comparison of alternative switching technologies for integrated voice and data networks; 3) the cost-effectiveness of alternative voice digitization rates and strategies. Three broad switching technologies and variations thereof are investigated and compared. The switching technologies are: circuit switching, packet switching, and hybrid (circuit-packet) switching. Each switching technology can accommodate either voice or data applications separately or combined voice and data requirements in an integrated fashion. Results of studies regarding communications systems option are provided and the sensitivity of the results are tested with respect to traffic variations, cost trends of switching and transmission, and network performance variables. The significant variables which affect the results are identified and quantified. The intent of this study is to identify and quantify network technologies which demonstrate long-term low operating costs, This is a necessary effort to provide the basis for determining the most cost-effective evolutionary path for future communication systems. It is recognized that transition problems and associated costs may be other important factors determining the ultimate evolutionary path. However, determining these costs was not an objective of this study. Neverthless, this study provides a framework and a target technology for detailed evolution, planning, and cost analysis.     

Fair data collection in wireless sensor networks: analysis and protocol

In general, wireless sensor networks (WSNs) consist of many sensors which transmit data to a central node, called the sink, possibly over multiple hops. This many-to-one data routing paradigm leads to nonuniform traffic distribution for the different sensors (e.g., nodes closer to the sink transfer more traffic than those farther away). In this paper, we perform an analysis of the fairness issue by presenting a tree-based WSN and derive the throughput, delay, and energy distribution for each sensor under the fairness constraint. Based on the analysis, we design our fair data collection protocol in which each node decides its media access and packet forwarding strategies in a distributed manner. Finally, we demonstrate the effectiveness of our solution through simulations. The results for the proposed protocol show the accuracy of the analysis and show that the protocol ensures the fair delivery of packets and reduces end-to-end delay. Based on the analysis, we also quantitatively determine the energy required for each of the nodes and show that a nonuniform energy distribution can maximize the network lifetime for the WSN scenario under study.     

Performance analysis of an enhanced DQRUMA/MC-CDMA protocol for voice traffic

The original distributed-queueing request update multiple-access (DQRUMA)/multicode code-division multiple-access (MC-CDMA) protocol was developed as a channel access protocol for wireless packet CDMA networks. This protocol has recently attracted considerable attention. We modify the original protocol, which was designed for data traffic only, to additionally accommodate voice traffic and call it the A-Protocol. We propose a new packet CDMA protocol that enhances the A-Protocol by improving the utilization of receivers in a base station and call it the E-Protocol. In the E-Protocol, an access request is attempted with a randomly chosen code at a request minislot. We analytically evaluate the performance of both protocols and compare analytical results with computer simulation. Analytical results agree well with simulation results.     

Call admission control for voice/data integration, in.broadband, wireless networks

This paper addresses bandwidth allocation for an integrated voice/data broadband mobile wireless network. Specifically, we propose a new admission control scheme called EFGC, which is an extension of the well-known fractional guard channel scheme proposed for cellular networks supporting voice traffic. The main idea is to use two acceptance ratios, one for voice calls and the other for data calls in order to maintain the proportional service quality for voice and data traffic while guaranteeing a target handoff failure probability for voice calls. We describe two variations of the proposed scheme: EFGC-REST, a conservative approach which aims at preserving the proportional service quality by sacrificing the bandwidth utilization, and EFGC-UTIL, a greedy approach which achieves higher bandwidth utilization at the expense of increasing the handoff failure probability for voice calls. Extensive simulation results show that our schemes satisfy the hard constraints on handoff failure probability and service differentiation while maintaining a high bandwidth utilization.     

CRDA: a collision resolution and dynamic allocation MAC protocol tointegrate data and voice in wireless networks

One key requirement for radio access in advanced, third generation (3G) mobile communication systems is the ability to accommodate a variety of services via a flexible and efficient medium access control (MAC) protocol. The paper presents and evaluates a new multiple access protocol termed CRDA (collision resolution and dynamic allocation), which has the potential to meet the above requirement. CRDA is basically a slotted packet-reservation multiple access technique with dedicated reservation slots, which allows the main shortcoming of previous PRMA schemes, i.e., contention, to be overcome through the integration in the MAC protocol of a code-division multiple-access (CDMA) transmission mode used to access the reservation slots. This prevents collisions during the reservation phase and enhances channel throughput, notably in the case of mixed voice/data traffic. Our simulations of the CRDA MAC integrate voice channels with data sources, generating what we call advanced data traffic, which has a very similar shape to the actual traffic generated by World Wide Web (WWW) applications. The standard assumption of Poissonian data traffic is also considered. Our conclusion is that the CRDA MAC protocol satisfactorily accommodates both types of traffic     

Performance of cellular packet CDMA in an integrated voice/data network

This paper presents a simulation-based study of cellular packet CDMA systems operating in an integrated voice/data traffic scenario. Spread-spectrum CDMA provides a suitable framework for resource-shared packet transport capable of combining isochronous (voice, ISDN) and bursty data services. In this work, a general network model for cellular packet CDMA with mixed voice/data traffic is described and used to evaluate the capacity/performance impact of several key system parameters. First, the effect of spreading factor (N) and forward error correction (FEC) rate are studied, confirming earlier work indicating a weak dependence onN and a well-defined optimum code rate in the range of 0.5–0.7 (with BCH coding). Next, the effect of propagation loss coefficient () on network capacity is investigated over a range of possible assumptions for, including both constant and distance-dependent models. The results show that system capacity depends strongly on, varying by as much as a factor of 2 over the range of parameters considered. For a given distance-dependent assumption, performance results are also obtained for different cell sizes in order to understand the overall spatial reuse efficiency achievable in different cellular and microcellular scenarios. This is followed by an investigation of traffic source model effects: first the capacity improvement from voice activity detection VAD) is presented, showing the expected 21 gains. Results for varying proportions of voice and data traffic intensities indicate that the operating efficiency does not change significantly as the proportion of bursty data relative to voice is varied.     

An address-light, integrated MAC and routing protocol for wireless sensor networks

We propose an address-light, integrated MAC and routing protocol (abbreviated AIMRP) for wireless sensor networks (WSNs). Due to the broad spectrum of WSN applications, there is a need for protocol solutions optimized for specific application classes. AIMRP is proposed for WSNs deployed for detecting rare events which require prompt detection and response. AIMRP organizes the network into concentric tiers around the sink(s), and routes event reports by forwarding them from one tier to another, in the direction of (one of) the sink(s). AIMRP is address-light in that it does not employ unique per-node addressing, and integrated since the MAC control packets are also responsible for finding the next-hop node to relay the data, via an anycast query. For reducing the energy expenditure due to idle-listening, AIMRP provides a power-saving algorithm which requires absolutely no synchronization or information exchange. We evaluate AIMRP through analysis and simulations, and compare it with another MAC protocol proposed for WSNs, S-MAC. AIMRP outperforms S-MAC for event-detection applications, in terms of total average power consumption, while satisfying identical sensor-to-sink latency constraints.     

An efficient protocol for providing integrated voice/data services to mobiles over power-limited satellite channels

Various mobile satellite communication systems are being developed for providing integrated voice/data services over a shared satellite transponder which is power-limited in nature. A common strategy is to use slotted ALOHA request channels to request channel assignments for voice/data calls from a network management station. To maximize efficiency in a system with a power-limited satellite transponder, it is proposed that the bursty nature of voice sources be exploited by the NMS to ‘over-assign’ channels. This may cause problems of inefficiency, potential instability as well as a degradation in the quality of service offered to the customers. Augmenting this with the introduction of simple state-dependent control procedures provides systems which exhibit more desirable operational features. This approach is proposed and examined in this paper.     

A reservation protocol for packet voice and data integration inunidirectional bus networks

A movable boundary protocol is proposed for integrating packet voice and data in unidirectional bus networks. The head station on the bus learns the number of ready-to-transmit voice stations by reading a `request' bit in the header of the received packets and allocates the exact number of voice slots needed in each frame. The protocol guarantees that the maximum delay to transmit a voice packet will be less than the round-trip propagation delay at the head station plus twice the time needed to form the packet. The average data packet delay is evaluated via approximate analysis and simulation, for the case in which the voice-reserved slots in a frame are contiguous and for the case in which they are evenly distributed