LTE系统上行链路质量提升的方法研究

随着VoLTE业务的商用以及4G用户大规模增加，4G网络不仅承担数据业务，还承担着语音业务，这对上行链路质量提出了更高的要求。在网络结构和无线环境日趋复杂的情况下，LTE链路质量在很大程度上制约着LTE网络性能的发挥，LTE网络受干扰、覆盖或容量受限程度直接影响了通话质量、上传下载速率、业务时延等用户的实际感知，还会引起无法接通、掉话、切换失败等一系列问题事件。本文利用功率余量值PH表征LTE上行链路质量，通过挖掘并调整上行性能参数来快速的提升上行链路质量。结合上海核心城区低功率余量TOP小区的实际情况，探究上海低PHR小区占比高的主要原因，提出一套区别于传统优化方式的参数优化方案以实现低PHR小区占比降低，达到上行链路质量提升的目的。通过试点验证可知，目前核心城区上行低PHR问题小区占比已由3.94%降至1.48%。     

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     

Mobility and Handover Management for Heterogeneous Networks in LTE-Advanced

3GPP Long Term Evolution-Advanced (LTE-A) aims at enhancement of LTE performance in many respects including the system capacity and network coverage. This enhancement can be accomplished by heterogeneous networks (HetNets) where additional micro-nodes that require lower transmission power are efficiently deployed. More careful management of mobility and handover (HO) might be required in HetNets compared to homogeneous networks where all nodes require the same transmission power. In this article, we provide a technical overview of mobility and HO management for HetNets in LTE-A. Moreover, we investigate the A3-event which requires a certain criterion to be met for HO. The criterion involves the reference symbol received power/quality of user equipment (UE), hysteresis margin, and a number of offset parameters based on proper HO timing, i.e., time-to-trigger (TTT). Optimum setting of these parameters are not trivial task, and has to be determined depending on UE speed, propagation environment, system load, deployed HetNets configuration, etc. Therefore, adaptive TTT values with given hysteresis margin for the lowest ping pong rate within 2 % of radio link failure rate depending on UE speed and deployed HetNets configuration are investigated in this article.     

Call-Level and Packet-Level Quality of Service and User Utility in Rate-Adaptive Cellular CDMA Networks: A Queuing Analysis

A queuing analytical model is presented to evaluate call-level and packet-level quality of service (QoS) metrics in the uplink of a voice/data cellular code division multiple access (CDMA) network. In this model, a threshold-based call admission control (CAC) is used to limit the number of admitted calls in a cell and also to prioritize handoff calls over new calls. The transmission rates for data calls can be adjusted to accommodate more voice and/or data calls while satisfying the minimum signal-to-interference ratio (SIR)/ transmission rate requirement. Also, automatic repeat request (ARQ)-based error control is used for improved reliability of data packets. Call-level performance measures for both voice and data calls and packet-level performance measures specifically for data calls can be obtained from the analytical model. The interdependencies among call-level and packet-level QoS metrics are investigated under different CAC, rate adaptation, and error control parameter settings. To this end, the level of users' satisfaction (or user utility) is formulated as a function of the QoS metrics and an optimization formulation is presented to obtain the local-optimal system parameters     

Call admission control for voice/data integration, in.broadband, wireless networks

This paper addresses bandwidth allocation for an integrated voice/data broadband mobile wireless network. Specifically, we propose a new admission control scheme called EFGC, which is an extension of the well-known fractional guard channel scheme proposed for cellular networks supporting voice traffic. The main idea is to use two acceptance ratios, one for voice calls and the other for data calls in order to maintain the proportional service quality for voice and data traffic while guaranteeing a target handoff failure probability for voice calls. We describe two variations of the proposed scheme: EFGC-REST, a conservative approach which aims at preserving the proportional service quality by sacrificing the bandwidth utilization, and EFGC-UTIL, a greedy approach which achieves higher bandwidth utilization at the expense of increasing the handoff failure probability for voice calls. Extensive simulation results show that our schemes satisfy the hard constraints on handoff failure probability and service differentiation while maintaining a high bandwidth utilization.     

SBS TDMA-DA system with VAC and DAC

This paper describes many of the design considerations involved in developlng the demand assignment (DA) subsystem of the TDMA satellite communication system for Satellite Business Systems. Each earth station has a satellite communications controller (SCC) which requests capacity to meet current voice and data traffic demands. A central reference station frequently reallocates capacity based on the requests of all the earth stations in a network. To minimize the amount of satellite capacity required, the SCC has a circuit-switching capability for voice calls as well as for digital data calls. Furthermore, the SCC employs voice activity, compression (VAC) and data activity cornpression (DAC), In which the amount of capacity requested is based on measured average speech activity as well as the number of off-hook voice ports and the number of off-hook data ports. Data calls are queued on a first-come, first-served basis when capacity is not immediateiy available. The reference station distributes excess capacity according to a nonlinear table lookup procedure so that the voice call blocking probability is equalized across the network. The DA system makes much more efficient use of satellite transmission capacity than a design with fixed-capacity trunk routes.     

Novel satellite digital video TDMA system for business videocommunications

The authors propose a satellite video signal transmission system that uses a time division multiple access (TDMA) scheme for digital video signal transmission and a code division multiple access (CDMA) scheme for voice signal transmission from a video signal reception-only earth station (backward voice signal transmission). The adoption of a TDMA scheme makes it possible to transmit video signals from various places and to communicate in multipoint-to-(multi)point mode. For backward voice channel transmission from video signal reception-only earth stations, a superposed CDMA voice signal transmission over TDMA video signals by using the same transponder is proposed. The employment of high coding gain forward error correction and development of a cochannel interference cancellation technique have made it possible for the proposed system to transmit a practical number of voice channels. The performance of the proposed system has been experimentally evaluated and the results show the proposed scheme can transmit about 20 CDMA voice channels simultaneously     

Performance evaluation on IEEE 802.11e considering emergency calls in congested situation

To enhance the quality of service (QoS) support in IEE 802.11, IEEE 802.11e has been studied, which introduces the so-called hybrid coordination function (HCF). HCF includes two medium access mechanisms contention-based channel access (EDCA), and contention-free channel access (HCCA). Although IEEE 802.11e has provided differentiated channel access mechanism, when call demand rises for important festivals such as New Year's Day or large scale natural disasters such as earthquakes, the delay of voice will increase and the QoS of voice nodes will drop down rapidly. Through our simulation study, in order to guarantee the QoS of emergency voice calls in congested situation, a higher priority for these calls will be required.     

A framework for network quality monitoring in the VoIP environment

Monitoring speech quality in Voice over IP (VoIP) networks is important to ensure a minimal acceptable level of speech quality for IP calls running through a managed network. Information such as packet loss, codec type, jitter, end‐to‐end delay and overall speech quality enables the network manager to verify and accurately tune parameters in order to adjust network problems. The present article proposes the deployment of a monitoring architecture that collects, stores and displays speech quality information about concluded voice calls. This architecture is based on our proposed MIB (Management Information Base) VOIPQOS, deployed for speech quality monitoring purposes. Currently, the architecture is totally implemented, but under adjustment and validation tests. In the future, the VOIPQOS MIB can be expanded to automatically analyze collected data and control VoIP clients and network parameters for tuning the overall speech quality of ongoing calls. Copyright © 2006 John Wiley & Sons, Ltd.     

语音数字信号压缩技术在TES系统中的应用

介绍ADPCM标准、RLPC编码原理,编、解码器方框图及工作过程。民航卫星通信网TES系统为节省卫星转发器频率资源对传输的语音信号进行压缩处理,其信道单元基带信号处理器对语音信号进行CCITT推荐的G.721-ADPCM编码和修斯公司专利技术开发的RLPC编码处理,将64 kb/s语音数字信号压缩至32 kb/s,16 kb/s,9.6 kb/s传输,实现语音质量满足一般通信要求的低速率语音信号传输。     

千兆宽带全光室内远端覆盖研究

为解决室内网络传输速度慢、信号覆盖范围小等问题,文中对全光远端覆盖技术（FTTR）进行了研究。首先,介绍了千兆宽带技术的发展现状,然后分析了影响信号覆盖的主要因素,最后提出了优化室内WIFI覆盖方案、FTTR技术应用等措施。根据测试结果可知,优化后的室内网络传输速度明显加快,且信号覆盖均匀,可为网络通信优化改造提供参考。     

Performance Optimizations for Deploying VoIP Services in Mesh Networks

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients     

Transmission quality of north American cellular, personalcommunications, and public switched telephone networks

Future long distance, and especially international calls, will involve an increasing number of multilink circuits of cellular, personal communications, mobile satellite, and public switched telephone network (PSTN) type of connections incorporating a variety of speech coding devices. In particular, the rapid growth of cellular communications has highlighted the need to characterize the quality of switched networks when cellular terminals are attached at their termination nodes. At the same time, the nonlinear nature of low-rate parametric speech coding has rendered questionable analytical methods for estimating end-to-end voice quality of interconnected telecommunications networks. Instead, quantification of transmission performance appears to require direct subjective evaluation of the pertinent conditions of interest. In this paper the quality of interconnected North American digital cellular and future microcellular terminals with 16 kbit/s and 32 kbit/s DCME/PCME-based switched and private telephone networks is quantified. From these assessments it can be concluded that cellular networks employing the TIA IS-54 8 kbits/s VSELP algorithm may meet the end-to-end transmission planning criteria when interconnected with the switched network     

Planning UMTS base station location: optimization models with power control and algorithms

Classical coverage models, adopted for second-generation cellular systems, are not suited for planning Universal Mobile Telecommunication System (UMTS) base station (BS) location because they are only based on signal predictions and do not consider the traffic distribution, the signal quality requirements, and the power control (PC) mechanism. We propose discrete optimization models and algorithms aimed at supporting the decisions in the process of planning where to locate new BSs. These models consider the signal-to-interference ratio as quality measure and capture at different levels of detail the signal quality requirements and the specific PC mechanism of the wideband CDMA air interface. Given that these UMTS BS location models are nonpolynomial (NP)-hard, we propose two randomized greedy procedures and a tabu search algorithm for the uplink (mobile to BS) direction which is the most stringent one from the traffic point of view in the presence of balanced connections such as voice calls. The different models, which take into account installation costs, signal quality and traffic coverage, and the corresponding algorithms, are compared on families of small to large-size instances generated by using classical propagation models.     

TD-SCDMA网络质差问题分析方法研究

由于TD-SCDMA无线网络的质差问题日益呈现严重趋势,本文根据TD-SCDMA网络的MR数据以及网络配置数据,探讨了TD-SCDMA话音质差问题的分析方法,剖析了干扰与覆盖对质差的影响,对于网络负荷、重叠覆盖度、频率复用度等各类因素对于质差的相关影响进行了详细分析。     

3G中的编解码协商技术

由于在接入网和核心网传送的码流格式不一致，造成话音信号进行两次编码和解码，导致语音质量降低。因此必须采用编解码协商技术来统一全程的编解码。编解码协商技术主要有无码型变换器操作（TrFO）、无二次编解码操作（TFO）和审计网络质量优选编解码技术。TrFO是呼叫建立过程中优选的一种机制，它尝试去建立用户设备（UE）到UE的无需使用码形变换器（TC）的连接，如果成功，能够最有效地使用带觅；TFO作为TrFO的备用技术，是一种带内的编解码协商协议，因为用户面码流不再需要通过语音编解码器的压缩、解压缩处理，可以改善话音质量；审计网络质量优选编解码技术依据呼叫的接入数来灵活地选择采用G．711或G．729来编解码，作到既不过分加重网络的负担，同时又可以接入新的呼叫。     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

On the impact of mobility and soft vertical handoff on voice admission control in loosely coupled 3G/WLAN networks

Loose coupling between 3G and WLAN ensures flexibility and openness. However, providing an ubiquitousmobile voice service in a loosely coupled 3G/WLAN network requires both packet-level and call-level quality of service (QoS) guarantees using soft vertical handoff (SVHO) and call admission control (CAC). In this paper, we evaluate the impact of both SVHO and WLAN mobility on call blocking and dropping probabilities rederived for the integrated network. For this purpose, we propose a new multi-region mobility model that accurately estimate these probabilities under a resource-efficient dynamicthreshold SVHO compared to a standard static-threshold SVHO. Results show us that the resource-efficient SVHO blocks and drops much less voice calls than the static one when very low mean and high variability of multi-mode mobile station velocities are noticed. Therefore, resource-efficient SVHO implementations are highly recommended in these mobility environments.     

Integrated system interoperability testing with applications to VoIP

This work has been motivated by the need to test interoperability of systems carrying voice calls over the IP network. The voice over IP (VoIP) systems must be integrated and interoperate with the existing public switched telephone network (PSTN) before they are widely adopted. Standards have been developed to address this problem, but unfortunately different standards bodies and commercial consortiums have defined different standards. Furthermore, the prevailing VoIP standard such as H.323 is incomplete, complex, and presents the implementers with "vendors latitudes". As a result, there is no guarantee that the integrated VoIP systems would interoperate properly even if the implementations are all H.323-compliant. Thus interoperability testing has become indispensable. We want to test all the system interoperations by exercising all the required patterns of "interoperating behaviors". On the other hand, test execution in real environment is expensive, and we want to minimize the number of tests while maintaining the coverage. We present a general method for automatic generation of test cases, which cover all the required system interoperations and contain a minimal number of tests. We study data structures and efficient test generation algorithms, which take time proportional to the total test case size. Finally, we report experimental results on VoIP systems.     

层次分析法在ZigBee网络语音通信中的应用研究

在自组织网络语音通信中,针对音频传输中存在的延时、丢包等主要问题,在ZigBee网络路由的基础上建立层次分析法评价模型,设计了一种音频传输路由算法AHP-RP。通过分析路径链路质量、音频负载值、路径存活时间和路径长度等因素对音频质量的影响,构建以网络的4个因素为因子的比较矩阵,选择最优传输路径。仿真及实际通信平台验证表明,该算法能有效地适应网络状态,明显改善了语音通话质量。