Digital Mobile Radio (DMR)

도시 인구 집중과 이동 인구의 증가로 이동 통신 시스템은 90년대 이후 급격한 발전을 하고 있다. 그러나 현재 널리 사용되고 있는 2세대 이동 통신 시스템은 음성에만 치중하여 개발되었고 시장이 거의 포화상태에 이르렀다. 따라서 앞으로 다가올 UMTS와 같은 제 3세대 이동 통신 시스템에서는 무선 인터넷과 같은 본격적인 데이타 서비스를 위해 패킷 데이타 통신이 매우 중요한 부분으로 떠오르고 있다. 음성 서비스와는 달리 패킷 데이타 특성에 갖는 패킷 데이타 서비스를 제공하기 위해서 UMTS 시스템에서는 전송하기 전에 반드시 데이타를 버퍼에 저장해야 한다. 하지만 지금까지 패킷 데이타 서비스를 위한 버퍼가 어떻게 얼만큼 필요하고 트래픽에 따라 그 상태가 어떻게 변화하는지에 관한 연구는 거의 진행되어 있지 않다. 본 논문에서는 UMTS 시스템에서 BS (Base Station)에 놓이는 버퍼 상태가 어떻게 변화하는지 알아 보기 위해 버퍼 상태 분석을 위한 UMTS 시스템 모델을 구성하고 실험을 통해 그 결과를 관찰 분석한다. 그리고 버퍼 상태를 확률을 통해 수식적으로도 분석한다. The mobile telecommunication system has been growing exponentially after 1990s due to the high population in a city and the growth of mobile user. In this time, the current mobile system mainly concentrates on the voice communication. However, in the next generation, mobile users want to get very diverse services via mobile terminal such as the Internet access, web access, multimedia communication, and etc. For this reason, the next generation system, such as the UMTS system, has to support the packet data service and it will play the major role in the system. In different from the voice service, the packet data service should store packet data in a buffer at base station before it forwards. Consequently, the performance closely related to the buffer management and its operation. However, until now, there are rare study on the buffer behavior and management. In this paper, we will observe the buffer behavior in the UMTS systems by using simulation and analyze the results. For this research, we generate packet data traffics and model the UMTS system for a simulation. Then we analyze the buffer state by a simulation and calculate the buffer overflow probability by mathematical methods.

cdma2000 is a US proposal for third generation cellular systems submitted to the International Telecommunications Union (ITU). This proposal is based on a smooth migration from IS-95 legacy systems and provides extensions to the existing IS-95 based data services that satisfy the IMT-2000 requirements for third generation systems. We describe a methodology for evaluating the performance of this proposal. Specifically, simulation methods are described for computing the spectral efficiency (capacity) of cdma2000 for voice, circuit and packet data services in a 5 MHz carrier. Voice data is serviced at 9.6 kbps peak rate and assumes 50% activity. Circuit data is serviced at higher data rates and assumes 100% activity. Two types of packet data services are evaluated-fixed and variable rate. In a fixed rate service all mobiles in the service area can support only one data rate. For variable rate services, the mobiles can support a range of rates from a specified maximum value down to a specified minimum.

Narrowing the 'Digital Divide': The Role of Complementarities between Fixed and Mobile Data in South Africa

In this paper we consider the problem of resource management in GSM/GPRS cellular networks offering not only mobile telephony services, but also data services for the wireless access to the Internet. In particular we investigate channel allocation policies that can provide a good tradeoff between the QoS guaranteed to voice and data services end users, considering three different alternatives, and developing analytical techniques for the assessment of their relative merits. The first channel allocation policy is called voice priority, since it gives priority to voice in the access to radio channels; we show that this policy cannot provide acceptable performance to data services, and we discuss the reasons for this shortcoming. The second channel allocation policy is called R-reservation; it statically reserves a fixed number of channels to data services, thus drastically improving their performance, but subtracting resources from voice users, even when these are not needed for data, thus inducing an unnecessary performance degradation for voice services. The third channel allocation policy is called dynamic reservation; as the name implies, it dynamically allocates channels to data when necessary, using the information about the queue length of GPRS data units within the base station. A threshold on the queue length is used in order to decide when channels must be allocated to data. Numerical results, show that the dynamic reservation channel allocation policy can provide very effective performance tradeoffs for data and voice services, with the additional advantage of being easily managed through the setting of the threshold value.

The growing demand for mobile and wireless access to Internet services has generated interest in how to use broadband infrastructures to provide such services. In this paper we provide, as a first step, an introductory presentation of alternative architectures for providing wireless Internet access, and in particular, the mobile IP service, when the fixed backbone network, either in the Internet or the public switched telephone network (PSTN) contains ATM entities. All the architectures use an interworking function (IWF) which encapsulates the functions of protocol adaptation, inter-BSC handoff encryption, authentication and packet routing and addressing. The first architecture we consider provides a wireless Internet access service only, without integration with voice services or provision for non-Internet data services (e.g. messaging or fax), and uses a centralized IWF. The second architecture is similar but uses a distributed IWF. We then consider two architectures where voice and data services are integrated. In architecture 3, voice and data services share the radio system only, with switching and other functions being provided separately for each service. architecture 4 not only allows voice and data to share the PSTN switching resources, but also considers the interworking of ATM switches in the PSTN backbones with narrowband PSTN switches as well as Internet ATM transport.

The emerging demand for low-cost data services over GSM networks has driven the ETSI to promote the definition of a General Packet Radio Service (GPRS), in which data traffic is carried according to the packet switched mode. We compare the performance of two possible GPRS implementations, respectively based on the static and dynamic allocation of traffic channels. In the dynamic allocation case, packet data services use traffic channels temporarily unused by circuit switched services, without affecting their performance. Moreover, the combined use of the static and dynamic schemes is evaluated and shown to provide enhanced performance for GPRS.

In this work output power distributions of mobile radio base stations have been analyzed for 2G and 3G telecommunication systems. The approach is based on measurements in selected networks using performance surveillance tools part of the network Operational Support System (OSS). For the 3G network considered, direct measurements of output power levels were possible, while for the 2G networks, output power levels were estimated from measurements of traffic volumes. Both voice and data services were included in the investigation. Measurements were conducted for large geographical areas, to ensure good overall statistics, as well as for smaller areas to investigate the impact of different environments. For high traffic hours, the 90th percentile of the averaged output power was found to be below 65% and 45% of the available output power for the 2G and 3G systems, respectively.

The telecommunications industry (TI) is challenged by a significant increase in the complexity of information transfer due to a recent proliferation of data mining technologies, techniques and applications. As the result, TI is facing a fundamental paradigm shift, with the convergence of voice and data services as well as ever expanding technologies to its users. This technological movement towards a convergence of telephony and computer technologies, web-based networks, and wired and wireless services is creating areas of tremendous opportunities. These areas of opportunity are for continuous quality improvements and applications of the voice and data convergence mining techniques and their implementations. The TI’s implementation of the data mining algorithms reduces information overload, increases data integrity and accuracy, and effectively manages its global networks.

Further wireless networks are expected to support different multimedia, voice and data services. During a different traffic call setup, a base station (BS) must decide whether to admit a new call or a handover call. This depends on the availability of bandwidth to support the call. The amount of bandwidth allocated to a new call is based on various statistical properties of the call and on the number of calls already in the session. When the aggregate call arrival rates exceed the BS's capacity, a BS can loss a call which is specified by its QoS requirements. This paper proposed the design of bandwidth allocation policies (overflow and rearrangement strategies) for multi-service mobile cellular environment and analytical formulations have been derived to compute QoS parameters such as blocking probabilities of each type of calls and carried traffic. Simulation models are also developed with Monte Carlo simulation method. The performance measures in this paper include blocking probabilities of each type of calls under varying offered traffics.

The crucial issues for service providers in wireless cellular networks include providing guaranteed quality of service (QoS), minimizing dropping rate (DR) for handoff calls, reducing the blocking rate (BR) for new calls and most importantly ensuring efficient utilization of network resources to maximize profits. This paper proposes an adaptive call admission control (CAC) algorithm for a wideband code division multiple access (WCDMA) network. The proposed algorithm utilizes three major concepts: cell breathing by considering the distance of a new and handoff call from a base station, bandwidth degradation of multimedia calls to provide priority for voice and video calls and transfer of ongoing calls to other cells to provide the highest priority to voice calls. This work is useful in the case when a voice or video call cannot be accepted into a cell due to unavailability of bandwidth and also when a new or handoff voice call cannot be accepted because the cell has already reached its maximum number of calls. Simulation results are presented for relations between call blocking, dropping probability and bandwidth utilization for different types of traffic (i.e. multimedia, video and voice calls).

Development of Unstructured Architecture for Voice and Data Services in Mobile Communication

The proposed evolution of TDMA, as governed by the next revision of IS-136, proposes enhanced voice and data services, which are enabled by using 8-level modulation. The proposed voice service uses the US1 vocoder. The proposed packet data service uses GPRS at higher layers with customized physical and MAC layers. In this paper, the performance of coherent and differential and 8-PSK are compared in light of the slot format and channel coding considered in ANSI-136. The comparison shows that the coherent modulation option is the preferable one. Also, throughput results for the proposed data service are presented.

In traditional CDMA wireless networks, real-time services, such as circuit-switched voice, are transported over the air via synchronous channels because of their stringent delay requirements. In the third generation networks (3G), IxEV-DV [A.Soong et al., 2003] and HSDPA [E. Dahlman et al., 1998], an additional time shared channel was introduced on the forward link to support data services for which delay requirements are less stringent. The IxEV-DO [P. Bender et al., 2000] standard provides a single time-shared forward link channel and no synchronous channels since it was designed primarily for data services. It has recently been suggested that certain voice services (e.g.. Voice over IP (VoIP) and push-to-talk (PTT)), can be efficiently transported over such time-shared channels since they have less stringent delay requirements compared to circuit-switched voice. In this paper we investigate the capacity of VoIP users over these time-shared channels and investigate the sensitivity to various base station (BS) and mobile station (MS) design parameters, algorithms and features. Note that detailed simulations of each standard is not provided but rather a comparative approach is used whereby we investigate specific features of each standard. Our focus is on the forward channel since code division multiplexing is performed in the reverse link and hence comparable user capacities are achieved.

TETRA was designed from the outset with security as one of its principal features. Standard security features include strong mutual authentication, dynamic cipher key encryption and secure terminal disabling. Some users require an additional degree of security and a methodology has been developed by the TETRA MoU Security and Fraud Prevention Group (SFPG) to implement end to end encryption for voice and short data services. This paper examines the reasons for choosing end to end encryption as a counter to perceived threats, shows the detail of the end to end encryption recommendation and examines the limitations of end to end encryption and the benefits of using it in conjunction with the standard TETRA security mechanisms. It also shows methods of key management in large systems for both end to end encryption and air interface encryption. (12 pages)

This paper deals with the study of the uplink mechanisms of medium access within an UTRA system which offers the possibility of holding on to voice and data services. The voice service is beamed off through a dedicated channel whereas a common channel is used for a very widespread service nowadays like the Web traffic. The first investigations on homogeneous systems reveal that the access channel is longer occupied in the data system than in the voice system. The simulations related to the integrated voice and data system show that, in spite of the fact that the voice experiences a greater delay than in the homogeneous system, the QoS parameters for voice and data are guaranteed with no need to use all the available system resources. Following this, the integrated system allows for certain losses in the common RACH (random access channel).