移动通信端到端语音传输安全问题与对策分析

随着移动通信的广泛应用,语音通信安全问题日益突出.移动通信网的现有安全机制仅对无线信道进行了加密,未能提供地面核心网络的安全措施.文中针对移动语音通信安全现状,探讨了实现端到端语音保密通信的方法,分析了加密语音在移动网内传输面l临的主要问题,研究了加密语音基于移动数据业务信道传输和基于话音业务信道传输的两种实现方案,分析了方案的可行性,指出了下一步的研究方向.     

综合信息传输平台技术探讨

科学考察船的特殊性要求对通信业务的综合集成。阐述了科学考察船各类通信业务的需求和传输特点,提出了通信业务在综合信息传输平台的集成及业务交换原理和实现方法,主要描述了程控业务和视频业务嵌入的技术,重点解决了程控业务中的信令传输、语音交换和视频数据流封装等关键技术。综合业务传输平台的运用同时解决了多种业务综合传输的稳定性和可靠性,为科学考察船的通信保障发挥了重要作用。     

GSM数字移动通信中的语音信道编码

GSM数字移动通信系统(以下简称GSM)采用的信道编码技术,使其业务信道和信令信道数据在无线环境下的传输特性达到了最优选择。以GSM业务信道中的全速率语音信道编码技术为例,以列表和图形示意的方式对GSM的信道编码过程做了较系统的介绍。     

LTE/EPC部署中的语音业务实现

语音业务是移动通信中最重要、最基本的通信业务。对3GPP提出的使用CS回退（CSFallBack）技术实现LTE/EPC部署中的语音业务方案进行了研究,分析了其网络架构和CS语音的实现过程,并对其他两种LTE运营商可能采用的语音实现方法进行了分析。     

基于STM32的无线语音传输系统设计与实现

随着移动通信技术的不断发展和完善,人们相互之间的沟通交流方式有了翻天覆地的变化,从过去的以固定电话作为远距离沟通的主要形式,逐渐转化为以移动设备无线通信作为日常生活的主要沟通方式,代表着无线语音传输技术的不断进步。为了进一步探究基于STM32芯片的无线语音传输系统的构建与实现,文章从STM32芯片的特点和无线语音传输系统的含义入手,详细分析了基于STM32芯片的无线语音传输系统的整体设计思路和大体结构,对该无线语音传输系统的硬件和软件结构进行了分类阐述,展现了基于STM32芯片构建的无线语音传输系统的实现。     

TrFO技术探析

通过说明TFO和TrFO技术的实现原理,讨论了TFO技术要二次编解码的问题,以及针对此问题引入的TrFO技术具有在呼叫双方采用相同语音编解码类型的情况下,可实现压缩语音透传的优势。比较了TFO和TrFO技术的区别和共同点。最后讨论了TrFO与BICC协议的关系,以及BICC协议完成Codec协商的过程。     

蜂窝移动通信的手机（五）

第二部分 手机的基带基本技术 这里准备介绍蜂窝移动通信手机的基带部分主要设计技术,包括语言编解码设计、信道编解码设计、蜂窝接入设计。１ 手机的语音编码设计１．１ 语音编码技术概述 语音编码是实现语音在数字通信系统中经济有效地传输的基本技术环节,蜂窝移动通信中的语音编码应力求： 语音编码速率尽可能低,以增加系统用户容量。 延迟时间合理。让通话人感觉自然可接受。 计算的复杂度与器件发展水平相适应。此中最关键的是计算量,同时,对ＲＡＭ和ＲＯＭ的需求也是考虑因素。 话音质量适当。话音质量以主观评定统计平均打分来…     

基于DSP的音频信号压缩通信系统的设计与实现

针对数据采集系统中语音传输受传输距离、成本等方面限制的问题,设计了一种低成本的语音信号压缩实时通信系统。采用模块化的方法进行硬件设计,通过将精简的TCP/IP协议移植到以太网控制芯片中来实现局域网通信,在DSP上优化实现了G.723.1语音编码算法,编码后的语音数据通过socket传送到PC端,PC端接收解码并实时播放。实验结果表明在语音质量略有退化的情况下,算法复杂度降低了,系统实现了语音信号远距离的实时传输。     

全IP组网下3G语音杂音优化思路研究

目前移动通信网络的全IP组网,给解决语音业务带来新的挑战;除了传统的无线设备故障、无线网络质量差、干扰以及终端等问题外还需掌握网络组网架构、传输误码以及各环节带宽负荷等网络状况。基于真实案例,逐步研究分析排查各个方面的问题,提出IP组网下该问题的端到端解决思路。     

面向宽带业务的网络全IP化解决方案

1 引言 传统的网络组织架构和电信传输网络面临从满足窄带的移动语音业务到满足移动分组业务和宽带业务转变的挑战,从技术角度看,这种挑战体现在移动网络的IP化和移动通信网络的宽带化.更具体一些,就是传统移动通信网络向3G网络的演进,而与此同时传统电信传输网络也需要实现IP化,以适应移动宽带业务的要求.     

NATIONAL SECURITY/EMERGENCY PREPAREDNESS AND DISASTER RECOVERY COMMUNICATIONS VIA ACTS

During the period from early 1993 to 1994, the U.S. National Communication System (NCS), a government agency, sponsored the development and execution of several fixed and mobile experiments using the advanced communications technology satellite (ACTS). The purpose of these experiments was to evaluate the feasibility of integrating an ACTS-like communications system into national security/emergency preparedness (NS/EP) operational scenarios. These experiments focused on clear and secure voice communications in both the fixed and mobile environment. For the mobile experiments, the focus was on the performance of both clear and secure voice in the land-mobile satellite communications channel. For the fixed experiments, the focus was on the use of a T1 terminal for restoration of communication services in the event of a disaster and for remote user communications. The results of these experiments are described in this paper.     

An integrated voice and data multiple-access scheme for a land-mobile satellite system

A Land-Mobile Satellite System (LMSS) is a satellite-based communications network which provides voice and data communications to mobile users in a vast geographical area. By placing a "relay tower" at a height of 22300 mi, an LMSS can provide ubiquitous radio communication to vehicles roaming in remote or thinly populated area. LMSS is capable of supporting a variety of services, such as two-way alphanumeric service, paging service, full-duplex voice service, and half-duplex dispatch service. A Network Management Center (NMC) will handle the channel requests, channel assignments, and in general the network control functions. A pool of channels is managed at the NMC to be shared by all mobile users. An integrated demand-assigned multiple-access protocol has been developed for the experimental LMSS. The pool of channels is divided into reservation channels and information channels. The information channels can be assigned by the NMC to be either voice channels or data channels. Each mobile user must send a request through one of the reservation channels to the NMC via the ALOHA random-access scheme. Once the request is received and processed, the NMC will examine the current traffic condition and assign an information channel to the user. NMC will periodically update the partitions between the reservation channels, voice channels, and data channels to optimize system performance. Data channel requests are queued at the NMC while voice channel requests are blocked calls cleared. Various operational scenarios have been investigated. Tradeoffs between the data and voice users for a given delay requirement and a given voice call blocking probability have been studied. In addition, performance impacts of such technological advancements as satellite on-board switching and variable bandwidth assignment are discussed.     

OFDM技术在GSM语音信道中的应用

远距离数据传输以及数据传输的保密性向来是人们比较关注的问题,使用语音信道传输数据信息是一种可选的方案。利用GSM信道,结合OFDM技术,将电脑、手机、A/D、D/A组成一个系统,可以实现一种远距离数据传输平台。针对OFDM技术特点,实现了GSM语音信道OFDM调制解调系统,经过多次实验测试,平均误码率曲线表明设计达到了最初的设计要求,在话音保密数据通信及其他数据传输上有较好的应用前景。     

Protocol design for mobile radio group communications oversatellite

A protocol design for a mobile radio service supporting half-duplex push-to-talk voice communications over mobile satellite systems is presented. The service enables a closed group of mobile radio users to communicate among themselves, so that a transmission from any user is received by all. A key issue is the efficient utilization of satellite channels. Demand-assignment multiple access (DAMA) is employed to allocate channels only to those user groups in active sessions of conversations. Owing to the unique characteristics of the half-duplex point-to-multipoint group communications, a new DAMA protocol is proposed for this service. Within a channel assigned to a user group, access contentions exacerbated by very long round-trip delay can occur. A signaling protocol is devised to arbitrate access contentions so as to improve the utilization of the assigned channel. The proposed access control protocol offers 65% or 18% improvement in throughput capacity compared to manual carrier-sensed access without and with collision detection, respectively     

基于AMBE-2000的数字语音传输系统

介绍了AMBE-2000的工作原理及其应用,并结合实例给出了数字语音传输系统中AMBE-2000与TLV320AIC10的A/D、D/A接口电路。AMBE-2000是DVSI(Digital Voice System·Inc)公司的语音压缩解压缩芯片,它采用AMBE语音压缩技术,具有语音音质好和编码波特率低、功耗低等优点;TLV320AIC10是TI公司的低功耗∑-Δ型16位A/D、D/A音频接口芯片。应用AMBE-2000的数字语音传输系统可以在低速率下提供优良的语音质量,并能实现全双工、实时语音压缩解压缩功能,因此在卫星通信、移动通信等领域有着广泛的应用。     

Combined rate and power adaptation in DS/CDMA communications overNakagami fading channels

We consider combined rate and power adaptations in direct-sequence code-division multiple-access communications, where the transmission power and the data rate are adapted relative to channel variations. We discuss the power gain that the combined adaptations provide over power adaptation. Then, we consider an integrated voice and data transmission system that offers a constant bit rate voice service, using power adaptation and a variable bit rate data service with rate adaptation. We present an expression for the required average transmission power of each traffic type having different quality-of-service specifications and discuss the capacity gain over power adaptation for voice and data     

Dynamic participation in a secure conference scheme for mobilecommunications

We propose a scheme to implement secure digital mobile communications. The scheme can both enable multiple users to hold a secure teleconference and also resolve the problem of allowing a participant to join dynamically or to quit a teleconference already in progress. Essentially, teleconference is a synchronous collaboration session in which participants at remote locations cooperate through wireless communications. Two requirements for the system are: privacy and authentication. Privacy signifies that an eavesdropper cannot intercept conversations of a conference. Authentication ensures that the service is not obtained fraudulently in order to avoid usage charge usage. We present a conference key distribution scheme for digital mobile communications, according to which users can share a common secret key to hold a secure teleconference over a public channel. The participants need not alter their secret information when a participant joins late or quits the conference early     

A secure end‐to‐end SMS‐based mobile banking protocol

Short message service (SMS) provides a wide channel of communication for banking in mobile commerce and mobile payment. The transmission of SMS is not secure in the network using global system for mobile communications or general packet radio service. Security threats in SMS restricted the use of SMS in mobile banking within certain limits. This paper proposed a model to address the security of SMS using elliptic curve cryptography. The proposed model provides end‐to‐end SMS communication between the customer and the bank through the mobile application. The main objective of the proposed model is to design and develop a security framework for SMS banking. Further, the protocol is verified for its correctness and security properties because most of the protocols are not having the facility to be verified by using the formal methods. Our proposed framework is experimentally validated by formal methods using model checking tool called automated validation of internet security protocols and Scyther tools. Security analysis shows that the proposed mechanism works better compared to existing SMS payment protocols for real‐world applications.     

Voice QoS in third-generation mobile systems

We analyze voice quality and system performance in third-generation mobile communication systems. We argue that in these networks that are expected to integrate voice, data, and multimedia services, the transport network can no longer be considered as a lossless transparent traffic channel. Rather, proper dimensioning is needed even for plain voice services. Furthermore, the efficient utilization of the network resources shared by voice and data traffic requires sophisticated traffic management in the statistical multiplexing environment. The contribution of the paper is twofold: first, voice quality in the cellular transport network is analyzed and network dimensioning criteria are derived. Second, building on advances reported in the literature and taking into consideration current standardization activities, a joint performance model is established for the air interface (AI) and the transmission network (TN) allowing for the analysis of end-to-end service performance within a uniform framework. To demonstrate the applicability of this model, we point to some performance problems in the multiservice environment and suggest and evaluate traffic management actions to overcome these