Wireless Internet access for mobile subscribers based on theGPRS/UMTS network

With the advent of IP technologies and the tremendous growth in data traffic, the wireless industry is evolving its core networks toward IP technology. Enabling wireless Internet access is one of the upcoming challenges for mobile radio network operators. The General Packet Radio Service is the packet-switched extension of GSM and was developed to facilitate access to IP-based services better than existing circuit-switched services provided by GSM. We illustrate how a visited mobile subscriber on a GPRS/UMTS network can access his/her home network via the gateway GPRS support node (GGSN). We also propose some implementation ideas on wireless Internet access for a remote mobile subscriber based on a GPRS/UMTS network     

Toward an all-IP-based UMTS system architecture

Looking into the future, two main drivers for the mobile telecommunications market can be identified: third-generation mobile systems (e.g., UMTS) and the Internet (e.g., the introduction of IP technologies like voice/multimedia over IP in mobile networks). UMTS is seen as the enabler of wireless multimedia applications and portability of a personalized service set across network/terminal boundaries, as defined within the virtual home environment (VHE) system concept. In light of these evolutions, this article investigates the impact of the evolution toward an all-IP UMTS network architecture on the UMTS service architecture, which is based on the VHE concept. The article discusses two possible scenarios for supporting VoIP services in the UMTS service architecture and analyzes their applicability in an all-IP-based UMTS network. The first is based on the traditional centralized IN service architecture. The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces     

Handoff with DSP Support: Enabling Seamless Voice Communications across Heterogeneous Telephony Systems on Dual-Mode Mobile Devices

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.     

Unified Mobility Manager: enabling efficient SIP/UMTS mobile network control

Internet telephony is viewed as an emerging technology not only for wireline networks, but also for third-generation wireless networks. Although IP end to end is considered the ultimate approach to future wireless voice services, there is still a long way to go before IP voice packets can be effectively transported over the air. Therefore, Internet telephony and today's circuit-switched wireless network will coexist for years to come, and it is essential to effectively perform interworking between these networks. This article proposes the Unified Mobility Manager (UMM) that achieves efficient interworking between traditional wireless networks and Internet telephony networks. The main characteristic of the UMM is that it combines UMTS HLR and SIP proxy functionality in one logical entity, which helps eliminate the performance degradation due to interworking between SIP and UMTS. This article identifies seven potential network architectures with and without the UMM and with varying degrees of IP penetration in the wireless core networks, and performs comparative analysis in terms of their call setup signaling latency. Our performance results show that for SIP originated calls, the architecture with the UMM can achieve better performance than existing UMTS networks without the UMM. Our results further show that when the backbone network is fully IP-enabled, dramatic performance gains can be accomplished with the UMM for PSTN originated calls as well as for SIP originated calls. The article also demonstrates that the UMM allows graceful migration from today's circuit-switched wireless networks to hybrid SIP/circuit-switched wireless networks, and toward the IMS architecture for all-IP UMTS networks in the future.     

乡村环境无线接入系统的设计

在农村电网自动化调度系统中,业务存在话音、电路数据和分组数据,是个典型的无线接入系统。其对于数据传输速率要求不是很高,但对数据的传输可靠度提出较高的要求。结合工程实际需求,提出采用扩频加时分机制构建了一个有中心的,一点对多点的无线接入的应用方案。基于对乡村环境传播特性的简述,提出了信道分配方案,详细介绍了分组数据可靠的传输的方法。     

Cost optimization methods in the design of next generation networks

A key development of telecommunication systems during the past two decades has been the evolution from the circuit-switched network toward the packet-switched network paradigm. Many operators are now migrating their PSTNs from circuit switched networks. This new approach is often called the next-generation network NGN enables network operators to run all services (i.e., voice, data and video) on one network. In this article the migration of Iceland Telecom's circuit-switched PSTN towards NGN is described. A cost model of the telecommunications system has been developed to enable cost and benefits analysis of transforming the network to NGN. Methods of optimization and their application to determine the optimal number and position of nodes in the future network are described. The optimization produces a network structure with the lowest possible total cost of ownership, and the model can also indicate how deviations from the optimum affect cost. The feasibility of NGN is assessed by comparing the cost of NGN migration to that of maintaining the current circuit-switched network.     

Cordless personal communications

The role of digital cordless technology in the transition to personal communication systems (PCSs) is discussed. The evolution of the pan-European digital cordless standards-CT2 and DECT-and industry progress in equipment availability and service implementation are reviewed. The application of cordless access to the domestic market in the United Kingdom and European telepoint, wireless PABXs, and radio drops are described. CT2 product development and availability and cordless data applications are outlined. The migration of microcellular cordless techniques for wider area systems including ETSI's universal mobile telecommunications system (UMTS) and the future public land mobile telecommunications systems (FPLMTS), now under development by CCIR with assistance from CCITT, are also discussed     

Intelligent MBWIMA/UMTS protocol using cascade fuzzy logic control for UTRA TDD mode

An intelligent medium-access-control (MAC) protocol based on cascade fuzzy-logic-control (CFLC), consisting of a fuzzy V/sub max/ (maximum number of voice/video slots) control and a fuzzy data-rate control for universal mobile telecommunications system (UMTS) terrestrial radio access time-division duplex (UTRA TDD) mode is presented. Voice, data, and video are integrated for transmission using CFLC-based movable-boundary wireless multiple access (MBWIMA) in the UMTS protocol. The performance of the CFLC-based MBWIMA/UMTS protocol with data-rate control is optimized by using the CFLC to generate the proper maximum number of voice/video slots and a suitable data rate. Simulation results demonstrate that the data-packet delay for the CFLC-based MBWIMA/UMTS protocol with data-rate control is less than that for the CFLC-based MBWIMA/UMTS protocol without data-rate control. The CFLC-based MBWIMA/UMTS protocol with data-rate control can greatly improve both the voice-video dropping probability and the data-packet delay of the MBWIMA/UMTS and general packet-radio system/UMTS protocols at the expense of a small decrease in the data-packet throughput. Voice-activity detector, single cell, and multicell environments are also considered.     

基于GPRS的IP电话技术研究

文章研究了一种的新的无线IP电话技术GPRS-VoIP,是一种基于GPRS接入的IP电话技术,可以实现和传统的基于电路交换的语音通话进行无缝切换.文中分析了该技术下的通话和传统GSM语音通话的无缝切换.文中还详细的分析了时廷、丢包、通话不连续等因素对基于GPRS接入的IP通话的影响和其对于带宽的需求,并提出了相应的解决方法.     

An all-IP approach for UMTS third-generation mobile networks

This article describes the UMTS all-IP approach for third-generation mobile systems, with emphasis on the core network architecture. Following the introduction of the core network nodes, we elaborate on application-level registration, circuit-switched call origination, packet-switched call origination, and packet-switched call termination.     

Slot allocation for an integrated voice/data TDMA mobile radiosystem with a finite population of buffered users

Slot allocation for voice and data in an integrated TDMA mobile radio system is investigated. In the proposed system, voice traffic is circuit-switched and data traffic is packet-switched using slotted ALOHA for channel access; the data traffic model is practically assumed to have a finite number of users with finite buffer capacity. The authors apply an equilibrium point analysis (EPA) technique to analyze the data performance and present a heuristic performance criterion to obtain an optimal slot allocation for voice and data in the integrated TDMA mobile radio system     

Wireless mobile communications at the start of the 21st century

At the start of the 21st century, the wireless mobile markets are witnessing unprecedented growth fueled by an information explosion and a technology revolution. In the radio frequency arena, the trend is to move from narrowband to wideband with a family of standards tailored to a variety of application needs. Many enabling technologies including wideband code-division multiple access, software-defined radio, intelligent antennas, and digital processing devices are greatly improving the spectral efficiency of third-generation systems. In the mobile network area, the trend is to move from traditional circuit-switched systems to packet-switched programmable networks that integrate both voice and packet services, and eventually evolve toward an all-IP network. Furthermore, accompanied by wireless mobile location technology, wireless mobile Internet is expected to revolutionize the services that can be provided to consumers in the right place and at the right time. Wireless mobile communications may not only complement the well established wireline network; it may also become a serious competitor in years to come. We review the history of the wireless mobile communications, examine the current progress in standards and technologies, and discuss possible trends for wireless mobile solutions     

Broadband satellite networks-the global IT bridge

Next-generation broadband satellite networks are being developed to carry bursty Internet and multimedia traffic in addition to the traditional circuit-switched traffic (mainly voice) on a global basis. These satellites provide direct network access for personal applications as well as interconnectivity to the terrestrial remote network segments. The main requirement in success of these networks is that they should be able to transmit high data rate traffic with prescribed quality of service (QoS). Thus, the broadband satellite network has no choice other than the rise of ATM technology and to be optimized for Internet-based traffic. ATM is the promising technology for supporting high-speed data transfer potentially suitable for all varieties of private and public telecommunications networks. IP, on the other hand is the fast-growing Internet layer protocol that is applicable over any data link layer Internet-based applications are the emerging source of traffic in the future wireless networks and broadband satellite networks should consider Internet as the primary service. In this paper, we discuss the traditional ATM and wireless ATM networks and explain the characteristics of the wireless IP networks. The paper then uses those concepts in defining the criteria for the broadband satellite networks such as the QoS and traffic management. Application of the broadband satellite networks is also proposed     

Channel Allocation for GPRS with Buffering Mechanisms

General Packet Radio Service (GPRS) provides mobile users end-to-end packet-switched services by sharing the radio channels with voice and circuit-switched services. In such a system, radio resource allocation for circuit-switched and packet-switched services is an important issue, which may affect the QoS for both services significantly. In this paper, we propose two algorithms: Dynamic Resource Allocation with Voice and Packet queues (DRAVP) and Dynamic Resource Allocation with Packet and Voice queues (DRAPV) for channel allocation of the voice calls and packets. We propose analytic and simulation models to investigate the performance of DRAVP and DRAPV in terms of voice call incompletion probability, packet dropping probability, average voice call waiting time, and average packet waiting time. Our study indicates that the buffering mechanism for GPRS packets significantly increase the acceptance rate of GPRS packets at the cost of slightly degrading the performance of voice calls.     

Operation, administration, and maintenance in MPLS networks

The boundaries between packet and circuit networks have long disappeared, with many traditional circuit-switched applications such as voice and video now being carried over packet-switched IP/MPLS or Ethernet networks. However, this transition has happened so fast that many OAM functions supported by circuit-switched networks such as SONET/SDH are still unmatched in packet networks. In order to match the quality sustained by circuit-switched networks, OAM functions must also be developed for such packet networks. A number of recent efforts have started to address OAM functions for IP/MPLS and Ethernet packet technologies. Service providers and carriers alike are the driving force behind the work, as there is general recognition that to generate sustained revenues, services must be efficiently managed. In this article we discuss issues in providing OAM features and capabilities for MPLS-based packet networks.     

A ubiquitous mobile communication architecture for next-generation heterogeneous wireless systems

Rapid progress in research and development of wireless networking and communication technologies have created different types of wireless systems (e.g., Bluetooth, IEEE 802.11, UMTS, and satellite networks). These systems are envisioned to coordinate with each other to provide ubiquitous high-data-rate services to mobile users. In this article, the architecture for ubiquitous mobile communications (AMC) is introduced that integrates these heterogeneous wireless systems. AMC eliminates the need for direct service level agreements among service providers by using a third party, a network interoperating agent. Instead of deploying a totally new infrastructure, AMC extends the existing infrastructure to integrate heterogeneous wireless systems. It uses IP as the interconnection protocol. By using IP as the gluing protocol, transparency to the heterogeneities of the individual systems is achieved in AMC. Third-party-based authentication and billing algorithms are designed for AMC. New mobility management protocols are also developed to support seamless roaming between different wireless systems.     

An IPv4-IPv6 translation mechanism for SIP overlay network in UMTS all-IP environment

Both IPv6 and session initiation protocol (SIP) are default protocols for Universal Mobile Telecommunications System (UMTS) all-Internet protocol (IP) network. In the existing mobile telecommunications environments, an IPv6-based UMTS all-IP network needs to interwork with other Internet protocol version 4 (IPv4)-based SIP networks. Therefore, mobile SIP applications are typically offered through an overlay structure over the IPv4-Internet protocol version 6 (IPv6) interworking environments. Based on 3GPP 23.228, we propose an IPv4-IPv6 translation mechanism (i.e., SIPv6 translator) that integrates different IP infrastructures (i.e., IPv4 and IPv6) to provide an overlay network for transparent SIP application deployment. In this paper, we present the architecture and the call flows of the SIPv6 translator. An analytic model is proposed to investigate the fault tolerance issue of our approach. Our study provides guidelines to select appropriate number of processors for fault tolerance.     

Internetworking connectionless and connection oriented networks

The use of connection-oriented (CO) networks for the transport of IP traffic is seen to have value to both users and service providers. Given the expectation that most endpoint-generated traffic will be in the form of connectionless (CL) IP datagrams, we address the problem of how to internetwork a CL (IP) network with a CO network. CO networks can be packet-switched or circuit-switched. Examples of packet-switched CO networks include ATM and MPLS networks, in which resource reservations are made at the ATM or shim layer, and IP-switch-based networks, in which resource reservations are made at the IP layer. Examples of circuit-switched networks include SONET/SDH and WDM networks that consist of programmable optical crossconnects. We consider the internetworking problem for two modes of operation of CO networks: provisioned, in which connections are set up a priori, and switched, in which connections are set up on demand. The main focus of this article is on the more complex problem: the internetworking of CL IP networks with CO networks operated in a switched mode. Our solution consists of (i) interworking user plane protocols with protocol conversion in some cases instead of always using protocol encapsulation, (ii) interworking routing protocols by either simply having gateways know routing information of both networks or having all nodes know routing information of both networks, and (iii) interworking signaling protocols by using application- or transport-layer end-to-end handshakes to trigger connection setups through the CO network. We demonstrate throughput improvements with our integrated routing interworking scheme over the MPOA IP-ATM internetworking solution for two example networks     

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     

Voice quality evaluation in wireless packet communication systems: a tutorial and performance results for RHC

As wireless systems evolve toward supporting a wide array of services, including traditional voice service, using packet-switched transport, it becomes increasingly important to assess the impact of packet-switched transport protocols on voice quality, in this article we present a tutorial on voice quality evaluation for wireless packet-switched systems. We introduce an evaluation methodology that combines elementary objective voice quality metrics with a frame synchronization mechanism. The methodology allows networking researchers to conduct effective and accurate quality evaluation of packet voice. To illustrate the use of the described evaluation methodology and interpretation of the results, we conduct a case study of the impact of robust header compression (ROHC) on the voice quality achieved with real-time transmission of GSM encoded voice over a wireless link.