Design and performance analysis for data transmission in GSM/GPRSsystem with voice activity detection

To efficiently utilize the bandwidth of cellular mobile systems and offer service of high quality to both voice and data users, we propose a protocol to integrate packet-switched data traffic into current time-division multiple-access (TDMA)-type circuit-switched digital voice systems. We analyze the performance of the proposed system, which transmits data packets in the silent periods of a conversation with voice activity detection and adapts itself to the GSM/GPRS system, which uses the idle channels to provide data services. We show that the proposed protocol can increase the bandwidth utilization efficiency and improve the throughput/delay performance of the data transmission while minimizing the impact on the current GSM/GPRS service     

The European (r)evolution of wireless digital networks

Some of the European developments of two different personal communication services, digital cellular telephony and mobile data networks, are compared with each other and with developments in the United States. The related agendas for research and development of wireless technologies in the past decade are also reviewed and compared. The present status of European cellular telephony, the implementation of mobile data networks in Europe, cellular engineering issues, and economic and regulatory issues are discussed. GSM, the pan-European digital cellular telephony standard, and some essential technology differences between the circuit-switched and packet-switched radio systems and their relative merits in the increasingly competitive European environment are also reviewed     

Internet telephony: going like crazy

Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive     

Transport of mobile application part signaling over Internetprotocol

The convergence of voice, data, and video networks is creating a new environment for telecommunications. In response to the changes, telecommunications equipment manufacturers and service providers are competing fiercely to bring an optimum solution to customers. The evolution of GSM to GPRS and to UMTS is a cellular wireless industry endeavour to meet this demand. This evolution will see the core wireless network infrastructure change from circuit-switched to packet-switched where voice and data are transported using IP as the common protocol. However, this poses a number of challenges, one of which is how to run the key mobile application part signaling protocols over IP. MAP defines the application protocols between switches and databases (e.g., MSC, VLR, SGSN, HLR) for supporting mobility management, security management, radio resource management, and mobile equipment management. UMTS supports both circuit-switched and packet-switched services     

An overview of EGPRS: the packet data component of EDGE

The explosive growth of the Internet and the subsequent demand for wireless data communications have led to data services being a major component in the standardisation process of many cellular mobile radio systems across the world. In GSM, the standardisation of the first phase of a new data service, initiated by the European Telecommunications Standards Institute (ETSI) and collectively known as Enhanced Data rates for GSM Evolution (EDGE), was finalised in 1999. The Enhanced Circuit Switched Data (ECSD) and Enhanced General Packet Radio Service (EGPRS) components of EDGE will enhance the existing GSM circuit-switched and packet-switched data services, respectively. This paper provides an overview of EGPRS and its performance     

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     

SS7 over IP: signaling interworking vulnerabilities

Public telephony - the preferred choice for two-way voice communication over a long time - has enjoyed remarkable popularity for providing acceptable voice quality with negligible connection delays, perhaps due to its circuit-switched heritage. Recently, IP telephony, a packet-based telephone service that runs as an application over the IP protocol, has been gaining popularity. To provide seamless interconnectivity between these two competing services, the Internet Engineering Task Force (IETF) has designed a signaling interface commonly referred to as SIGTRAN. This seamless intersignaling provided by SIGTRAN facilitates any subscriber in one network to reach any other subscriber in the other network, passing through any heterogeneous maze of networks consisting of either of these. Unfortunately, the same intersignaling potentially can be exploited from either side to disrupt the services provided on the other side. We show how this can be done and propose a solution based on access control, signal screening, and detecting anomalous signaling. We argue that to be effective, the latter two should consider syntactic correctness, semantic validity of the signal content, and the appropriateness of a particular signal in the context of earlier exchanged messages     

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     

Architecture and testbed implementation of vertical handovers based on SIP session border controllers

This paper describes a Session Initiation Protocol (SIP) based solution for mobility management that provides seamless mobile multimedia services in a heterogeneous scenario where different radio access technologies are used (802.11/ WiFi, Bluetooth, 2.5G/3G networks). The solution relies on the so called “Session Border Controllers” which are now widely used in many commercial SIP telephony solutions, mainly to deal with NAT traversal. Session Border Controller functionality has been extended to support seamless mobility for multimedia applications. A prototype of the proposed solution focused on VoIP services has been implemented in a test bed which is able to perform seamless handovers (and NAT traversal) using the 802.11, Bluetooth and 3G (UMTS) access networks. Measurements results are reported which analyze the performance of the solution in a real world environment, using commercial WiFi and 3G services.     

Isochronous communications in an integrated circuit/packetmetropolitan area network

A simple protocol is presented for establishing and terminating circuit-switched communications within a metropolitan area network (MAN) that supports both circuit-switched communication (for voice and video) and packet-switched communication (for data and file transfers). A mathematical model of the MAN is used to study its performance with respect to the circuit-switched communication and its effects on the pocket-switched communication. The models are numerically solved by the mean-value-analysis algorithm     

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.     

Applications of voice processing to telecommunications

The ways in which people communicate are changing rapidly. The options are many and diverse, ranging from voice calls over wireless networks, to video calls over the conventional wired network, ISDN video, FAX, e-mail, voice mail, beeper services, data services, audio teleconferencing, video teleconferencing, and so-called scribble phone service (transmission of arbitrary handwritten input). This revolution in communications is being fueled by several sources, including the availability of low-cost, low-power, computation in both DSP and RISC chips, larger and cheaper memory chips, improved algorithms for communications (e.g., modems, signaling) and signal processing, and finally the creation of world-wide standards for transmission, signal compression, and communication protocols. The broad goal of the communications revolution is to provide seamless and high-quality communications between people (or groups of people), anywhere, anytime, and at a reasonable price. Although there are many technologies that form the bases for the communications environment of the twenty-first century, one of the key technologies for making the vision a reality is voice processing. In this paper we attempt to show, by example, how voice processing has been applied to specific problems in telecommunications, and how it will grow to become an even more essential component of the communications systems of the twenty-first century     

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.     

A Low Latency Handoff Algorithm for Voice over IP Traffics in the Next Generation Packet-Based Cellular Networks: Cellular Mobile IPv6

In the mobile communication environments, Mobile IP is defined to provide users roaming everywhere and transmit information freely. It integrates communication and network systems into Internet. The Mobile IPv6 concepts are similar to Mobile IP, and some new functions of IPv6 bring new features and schemes for mobility support. Two major problems in mobile environments are packet loss and handoff. To solve those problems, a mobile management scheme – the cellular mobile IPv6 (CMIv6) is proposed. Our approach isbased on the Internet Protocol version 6 and is compatible with the Mobile IPv6 standard. Besides, it also combines with the cellular technologies which is an inevitable architecture for the future Personal Communication Service system (PCS). In this paper, {Cellular Mobile IPv6 (CMIv6)}, a new solutionmigrated from Mobile IPv6, is proposed for mobile nodes moving among small wireless cells at high speed. This is important for future mobile communication trends. CMIv6 can solve the problems of communication break off within smaller cellular coverage during high-speed movement when packet-switched data or the real-time voice messages are transmitted. Voice over IP (VoIP) packets were chosen to verify this system. The G.723.1 Codec scheme was selected because it has better jitter resistance than GSM and G729 in a packet-based cellular network. Simulation results using OPNET show smooth and non-breaking handoffs during high-speed movement.     

Error-resilient packet-switched mobile video telephony with channel-adaptive rateless coding and early reference picture selection

Providing high-quality video for packet-switched wireless video telephony on handheld devices is a challenging task due to packet loss, varying bandwidth, and end-to-end delay constraints. While many error resilience techniques have been proposed for video transmission over wireless channels, only a few were specifically designed for mobile video telephony. We propose a low-complexity channel-adaptive error resilience technique for packet-switched mobile video telephony, which combines rateless coding, feedback, and reference picture selection. In contrast to previous approaches, our technique uses cumulative feedback at every transmission opportunity and predicts when decoding is likely to fail so that reference picture selection can be triggered at an early stage. Experimental results for H.264 video sequences show that the proposed technique can achieve improvements of 1.64 dB in peak signal-to-noise ratio over benchmark techniques in simulated Long-Term Evolution networks.     

Context-aware cooperative testbed for energy analysis in beyond 4G networks

The complexity of heterogeneous wireless networks in synergy with battery powered mobile devices is driving new stringent requirements in terms of power efficiency to ensure that battery life, environmental and thermal criteria can be met. Modern mobile devices are equipped with multiple interfaces, which allow them to exploit the benefits offered by heterogeneous networking environments, but on the other hand, drain battery swiftly. In this paper, architecture for a context-based node and a testbed platform for the analysis of energy consumption of heterogeneous cooperative communications are presented. The demonstrative testbed comprises a WiFi Access Point, which provides WiFi coverage in the infrastructure mode, as well as nodes capable of communicating through short-range ultra-wideband WiMedia. The testbed includes a context aware module that provides and stores information related to different nodes in the system. The paper shows how context information can be used to save the energy of mobile devices and extend their battery lifetime using short-range communications. The testbed is used as a proof-of-concept for the practical implementation of the cooperative communications concept. The obtained results show that significant amount of energy can be saved using context information along cooperation among multiple interfaces, in comparison to direct communications.     

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.     

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     

A Data Modem for GSM Voice Channel

This paper introduces a novel approach to data communication over the global system for mobile communications (GSM) voice channel. It is based on the concept of ldquosymbolsrdquo-a set of predefined signals with finite bandwidths. Data are encoded into the symbols, and the symbols are voice coded as they were speech, modulated into the GSM signal, sent over the air, GSM demodulated, voice decoded, and converted back to data. The symbols are synthesized by a genetic algorithm with the aim of maintaining separability after passing them through the voice codec. This method enables data transfer over communication networks that do not have dedicated data channels and could also be used in conjunction with other data services to balance the system load between data and voice channels, allowing optimization of system resources. We present the full algorithmic structure of the system, which performs data communications over the GSM voice channel, and we also give the results of the performance tests.     

由TMS320VC5049与C8051F020组成的数据采集传输系统

设计了一种基于低功耗的TMS320VC5409 DSP(数字信号处理器)和C8051F020单片机组成的数据采集传输系统,它能够采集、处理和通过无线移动网传送和接收数据.该系统以CPU单元为核心,主要由DSP、CPU、GSM(全球移动通信系统)3个单元组成,DSP单元用于数据处理,GSM单元用于数据传输.该系统具有体积小、便于携带、功耗低、可使用电池供电的特点,因而主要应用在要求移动数据采集传输的小型或便携仪器上.