基于TFO的移动话音安全传输应用研究

如何在移动通信系统中建立语音业务的安全数据流透明传输通道,是移动通信系统安全语音方案设计的关键技术。移动通信标准中的TFO技术可以避免核心网侧设备的语音编解码功能,实现语音数据流的端到端透明传输通道。研究了GSM、CDMA等移动通信系统中的TFO协商流程、协商原则,重点分析和研究如何基于TFO实现移动通信安全语音业务的技术,为实现高音质的移动通信安全语音业务打下了良好基础。     

A Low-rate Data Transfer Technique for Compressed Voice Channels

This study considers the problem of transmitting generic (non-speech) data through compressed voice channels such as those used in wireless communications networks. These networks employ voice codecs that are designed to efficiently encode and reproduce the relatively slow-changing signals of human speech, which leads to communication channels that are nonlinear and have long-term memory. A data modem is presented that utilizes finite alphabets of waveforms that are numerically optimized to be as separable as possible after passing through the voice codec. The optimization of the finite alphabet is performed using a pattern search algorithm. When used with the GSM Enhanced Full-Rate Voice Codec, this system demonstrated improved performance, in terms of error rates, compared to previously reported results. Simulation results for four other voice codecs are also presented.     

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。     

Blind and nonblind turbo estimation for fast fading GSM channels

Wireless communication systems such as global system for mobile (GSM) communications are playing a growing role for data transmission. In order to ensure reliable transmission, the channel impulse response has to be estimated accurately. This is a difficult task particularly for fast fading channels caused by high-speed mobile units. We deal with the application of nonblind and blind channel estimation approaches to identify the full rate data Traffic CHannel (TCHF9.6) of GSM. We present a new iterative channel estimation scheme leading to a significant performance improvement especially for high-speed propagation with Doppler frequencies up to 500 Hz. Furthermore, it is shown that blind channel estimation schemes could be as efficient as nonblind methods when regarding bit error rates (BERs) after channel decoding in terms of the E¯_b/N₀ ratio. Moreover, a solution for the scalar ambiguity inherent in all blind estimation approaches is suggested     

Performance of the π/4-DQPSK, GMSK, and QAM modulation schemesin mobile radio with multipath fading and nonlinearities

As mobile communications have become so indispensable, every possible effort should be spent to achieve the optimum operating conditions. In addition to the normal impairments associated with wireless communications, in general, the mobile channel suffers from particular limitations that confine the performance of a mobile radio system. Among those impairments are the bandwidth limitation, interference, and multipath fading. With the strong motivation toward portable radio and personal communication systems, power limitation has manifest itself in the picture, and, consequently, nonlinear operation of the amplifiers involved (hence, the channel) will have to be dealt with. Constant envelope modulation schemes have been used in digital mobile radio systems recently installed. The Gaussian minimum shift keying (GMSK) is employed in the Global System for Mobile (GSM) communications installed in Europe, while in the North American IS-54 system, the modulation scheme used is the π/4-DQPSK. As the quest for higher data rates has kept on increasing, multilevel modulation schemes have been proposed with their performance over nonlinear channels having been overlooked. The paper provides a comparative study, based on simulation, and tests the performance of various modulation schemes operating over a wide variety of mobile radio channel conditions. The effective throughput of all systems is also considered     

Theoretical analyses of data communications integrated intocordless voice channels

Dynamic integration of data into voice channels of second-generation cordless systems provides effective channel utilization. This paper proposes and theoretically examines an inhibit and random multiple access (IRMA) protocol for data terminals in an integrated voice and data system by assuming that real-time voice traffic has priority over data. Analytical expressions are derived to quantify the effect of data inhibition on data performance, i.e., throughput and delay for both infinite and finite population models for data terminals. In order to find the possible data throughputs with our channel access scheme for data communication using voice channels while not affecting the voice quality, we investigate the data performance under two extreme situations: 1) no voice load and 2) full voice load. The numerical results indicate that IRMA data performance is comparable to that of slotted ALOHA (S-ALOHA) at lighter loads of less than 0.2 while efficiently sharing the network resources between real-time voice and nonreal-time data traffic. For the data loads above 0.2, IRMA pays a price, but an affordable one, in terms of throughput performance for its ability to accommodate data while assuring quality of voice even when all channels are occupied by voice traffic. An optimum number of terminals and the range of data-transmission probabilities have been deduced as 16 and 0.05-0.15, respectively, in order to achieve maximum throughput with minimum delay while maintaining stable data transmission and voice quality in an integrated system     

应用坎贝尔方法配置GPRS／EDGE的静态信道及数据流量对话音信道配置的影响

为了给移动网络中的每个小区配置合适的GPRS／EDGE静态信道数，采用了坎贝尔模型。本文具体阐述了坎贝尔方法的流程。由于将GSM的话音信道TCH的部分或全部设置为GPRS／EDGE的动态信道，必须研究数据流量对话音信道配置的影响，以对话音信道进行适当的扩容。     

Outage Probability in GSM-GPRS Cellular Systems With and Without Frequency Hopping

General packet radio service(GPRS) is designed for transmitting packet data andis supposed to take its radio resource from the pool ofunused channels of GSM voice services. Obviously, theintroduction of GPRS has an impact on the voice services.In this paper, we present a method to calculate theoutage probability of the GSM-GPRS network for bothnon-frequency hopping and frequency hoppingsystems. This method takes into account Rayleighfading, power control (with error), discontinuoustransmission, and frequency hopping (if applied). Theoutage probability of voice services affected by theintroduction of GPRS is discussed. The number ofunused voice channels allocated to GPRS depends on thedifference between the outage level of the existingGSM network and the maximum acceptable level. Thefrequency hopping system can accommodate more GPRStraffic than the non-frequency hopping system. Thepower control error has more impact on systemperformance when more channels are allocated to GPRS.Beyond our expectations, for the non-frequency hoppingsystem, the channels provided for GPRS are not muchdifferent between high channel occupancy and lowchannel occupancy of voice services. In contrast, for thefrequency hopping system, the system can provide morechannels for GPRS at low channel occupancy. The cellservice area decreases by about 10% 20% for eachadditional channel allocated to GPRS.     

Performance comparisons of dynamic resource allocation with/without channel de-allocationin GSM/GPRS networks

In this letter, we analyzed and compared the performance of dynamic resource allocation with/without channel de-allocation in GSM/GPRS networks. It is quite known that dynamic resource allocation allows communication systems to utilize their resources more efficiently than the traditional fixed allocation schemes. In GPRS, multiple channels may be allocated to a user to increase the transmission rate. In the case when there are no free channels in the system, some of these channels may be de-allocated to serve higher priority calls. The results show that with channel de-allocation mechanism, the voice blocking probability can be greatly reduced, especially at high GPRS traffic load. Besides, the scheme with channel de-allocation mechanism can achieve higher channel utilization.     

A suboptimal approach to channel equalization based on the nearestneighbor rule

Applications of clustering and neural network techniques to channel equalization have revealed the classification nature of this problem. This paper illustrates an implementation of a global system for mobile communications (GSM) receiver in which channel equalization and demodulation are realized by means of the nearest neighbor (NN) classifier algorithm. The most important advantage in using such techniques is the significant reduction in terms of the computational complexity compared with the maximum likelihood sequence estimation (MLSE) equalizer. The proposed approach involves symbol-by-symbol interpretation and the knowledge of the channel is embedded in the mapping process of the received symbols over the symbols of the training sequence. This means that no explicit channel estimation need be carried out, either with correlative blocks or using neural networks thus speeding up the entire process. The performance of the proposed receiver, evaluated through a channel simulator for mobile radio communications, is compared with the results obtained by means of a 16-state Viterbi algorithm and other suboptimal receivers. It is shown that the presented algorithm increases the bit error rate (BER) compared with the MLSE demodulator, but the performance degradation, despite the simplicity of the receiver, is kept within the limits imposed by the GSM specifications     

Design and Implementation to Authentication over a GSM System Using Certificate-Less Public Key Cryptography (CL-PKC)

Recent years, the mobile technology has experienced a great increment in the number of its users. The GSM’s architecture provides different security features like authentication, data/signaling confidentiality and secrecy of user yet the channel is susceptible to replay and interleaved. It always remains relevant as it is important in all types of application. Global system for mobile (GSM) communications has become the most popular standard for digital cellular communication. The GSM security system depends on encryption, authentication algorithms and information from SIM card. In this research paper, we proposed the design and implementation of a new authentication scheme by using certificate-less public key cryptography (CL-PKC) over the GSM system was attempted to miss some system detail. This research paper, we also proposed the GSM system and its security and public key cryptography with a focus in the CL-PKC; the CL-PKC is a simple, useful and robust security scheme designed and implemented over GSM. Our approach is more efficient than other competing topologies. We solved the GSM problem in A3 algorithm such as eavesdropping and this problem solved by CL-PKC because of its robustness against this type of attack by providing mutual authentication make the system more secure.     

基于短信的通信系统信息保障协议研究

通信与信息已密不可分。短消息业务(Short Messaging Service,SMS)是GSM网络提供的主要电信业务之一,他是通过GSM网络的七号信令(557)承载较短数据包来实现个人简易数据通信的一种方式。作为一种既不占用无线话音信道资源,又能够提供终端用户点对点之间便捷、价格低廉的信息交流方式,短消息业务越来越受到广大移动用户的欢迎。短信系统作为一种数据传输手段,其通信信道是不可靠的,所以必须有相应的信息保障协议对其数据传输进行保护,确保其可靠性。     

Performance of M-QAM modulations for indoor radio communications with combined diversity and equalization techniques

Digital indoor radio communications systems have become an attractive proposal as they could provide the user high quality voice and low rate data services with total mobility. However, indoor communications are affected by multipath propagation that restricts their application imposing an upper limit on the system data rate. To overcome this drawback diversity and channel equalization could be considered. In this paper the performances of 4, 16 and 64-QAM modulations in a radio communications indoor environment are presented. The results show that a system without protection has very limited performance. Diversity techniques are efficient when the radio channel can be assumed to be non-selective. For systems operating in selective fading channels joint equalization and diversity techniques are necessary to fight the degrading effect introduced by the channel. Moreover, if correlation between the two diversity channels is taken into account, the system performances remain almost unchanged, assuming that the correlation coefficient between the two diversity branches is lower than 0.6 to 0.7 approximately.     

动态多通道卫星话音／数据综合的S－ALOHA性能分析

本文分析了一种新的多通道话音／数据综合ＶＳＡＴ系统的Ｓ－ＡＬＯＨＡ性能，并给出了数值计算结果；从中可看到该协议既能提供较高的吞吐量，又能有效地减小话音时延，并能协调话音和数据之间对性能要求的差异，文中考虑了Ｓ－ＡＬＯＨＡ中分组达到时间抖动的影响．文中还讨论了话音在重负载的条件下可占用数据通道的动态分配方案，并给出了动态分配的算法。     

L’égalisation numérique en communications

This tutorial contribution explains how digital equalization permits very high transmission rates, even with severe channels, by adaptive (real time) correction of the distortion. Equalizers are in general digital filters. Transversal equalizers are only suitable for mild channels. In data transmission, where symbols have discrete levels, severe channels can be equalized by adding a recursive path that is filled in with detected symbols. Engineers have realized optimal equalizers for more than twenty years thanks to the adaptive tracking of the channel time variations, at a very low computational cost. However adaptation requires the periodic transmission of a training sequence deprived of information content. This supervised learning technique is acceptable only in an end-to-end communication system. The most recent equalization methods are usable in multiuser systems such as networks, broadcasted communications, etc., because they employ self-learning or unsupervised equalization. Then adaptation is controlled by the very information data flow. It only takes advantage of an a priori statistical knowledge on the emitted data, e.g. their whiteness, a property that is ensured thanks to jamming. Let us conjecture that all equalizers will be self-learning in a near future.     

Block adaptive techniques for channel identification and datademodulation over band-limited channels

A new approach to the problem of data detection for communications over band-limited channels with unknown parameters is introduced. We propose a new way to implement the Viterbi algorithm (VA) for maximum-likelihood data sequence estimation (MLSE) in a known channel environment and utilize it to derive block adaptive techniques for joint channel and data estimation, when the channel-impulse response (CIR) is unknown. We show, via simulations, that we can achieve a probability of error very close to that of the known channel environment and nearly reach a mean-square error in the channel estimate as predicted by analytical bounds, operating on static channels, which exhibit deep nulls in their magnitude response and nonlinear phase. The proposed schemes accomplish channel acquisition after processing a few hundred symbols while operating without a training sequence, whereas linear blind equalizers, such as Sato's (1975) algorithm, fail to converge at all. The application of block processing to adaptive MLSE is also investigated for time-varying frequency-selective Rayleigh-fading channels, which are used for modeling mobile communication systems. In such environments it is shown that the proposed scheme exhibits improved performance compared to the conventional adaptive MLSE receiver using tentative delayed decisions     

安全控制管理中心实现GSM安全通信

安全控制管理中心通过蜂窝移动通信网络,可构建蜂窝移动通信安全服务系统,为专用手机提供安全信息自动产生和分配、被动式和主动式安全信息销毁、安全数据控制以及失控专用手机追踪等安全通信管理服务,确保专用手机的使用、操作安全。论文提出专用手机利用在GSM提供的数据信道上建立的虚拟专网(VPN)来实现数字安全话音和安全数据通信,适合在国内外现有的GSM蜂窝移动通信系统上使用。     

Non-Uniform Time Slot Based Modulation for Visible Light Communication System

Recently, visible light communication (VLC) has got increasing attention in the field of indoor wireless communications. VLC has the obligation of lighting besides data communications. The IEEE 802.15.7 task group proposed a VLC modulation format named color shift keying in which data symbols are defined based on the diffused light wavelength. This paper presents a new modulation scheme for indoor wireless visible light communication system named non-uniform time slot modulation (NUTS). The NUTS modulation scheme defines the feeder signals of red, green and blue LEDs as rectangular pulses with appropriate width and their positions. In this method, the time axes of all the three light sources (channels) are slotted to some non-uniform durations that all symbols are spanned over these slots as orthogonal dimensions. The proposed scheme has some advantages with respect to other approaches such as low bit error rate and high bit rate. Furthermore, it provides appropriate synchronization capability. Besides, in the design of NUTS modulation scheme it has been tried to deliberate all practical constraints, as far as possible.     

An adaptive modulation scheme for simultaneous voice and datatransmission over fading channels

We propose a new adaptive modulation technique for simultaneous voice and data transmission over fading channels and study its performance. The proposed scheme takes advantage of the time-varying nature of fading to dynamically allocate the transmitted power between the inphase (I) and quadrature (Q) channels. It uses fixed-rate binary phase shift keying (BPSK) modulation on the Q channel for voice, and variable-rate M-ary amplitude modulation (M-AM) on the I channel for data. For favorable channel conditions, most of the power is allocated to high rate data transmission on the I channel. The remaining power is used to support the variable-power voice transmission on the Q channel. As the channel degrades, the modulation gradually reduces its data throughput and reallocates most of its available power to ensure a continuous and satisfactory voice transmission. The scheme is intended to provide a high average spectral efficiency for data communications while meeting the stringent delay requirements imposed by voice. We present closed-form expressions as well as numerical and simulation results for the outage probability, average allocated power, achievable spectral efficiency, and average bit error rate (BER) for both voice and data transmission over Nakagami-m fading channels. We also discuss the features and advantages of the proposed scheme. For example, in Rayleigh fading with an average signal-to-noise ratio (SNR) of 20 dB, our scheme is able to transmit about 2 bits/s/Hz of data at an average BER of 10 ^-5 while sending about 1 bit/s/Hz of voice at an average BER of 10^-2