PRMA/TDD多址协议下的数据┐话音综合

本文提出了一个在ＰＲＭＡ／ＴＤＤ多址协议下的数据－话音综合方案，该方案以动态边界分离话音和数据业务。动态边界的位置由话音子系统的业务负荷确定，使系统在保证话音业务质量的前提下，让数据业务获得最大带宽。通过马尔可夫链建模，推导出了数据业务最大化公式，并进行了计算机系统仿真。分析结果表明，与ＰＲＭＡ数－话综合方案相比，该方案可使系统性能明显提高。动态边界也使话音业务子系统独立于数据业务子系统，因此，数据业务的突发性和潜在不稳定性不会影响话音业务。     

PRMA／TDD多址协议下的数据—话音综合

本文提出了一个在ＰＲＭＡ／ＴＤＤ多址协议下的数据－话音综合方案，该方案以动态边界分离话音和数据业务。动态边界的位置由话音子系统的业务负荷确定，使系统在保证话音业务质量的前提下，让数据业务获得最大带宽，通过马尔可失链建模，推导出了数据业务最大化公式，并进行了计算机系统仿真。分析结果表明，与ＰＲＭＡ数－话综合方案相比，该方案可使系统性能明显提高。动态边界也使话音业务子系统独立于数据业务子系统，因此，数     

固定通信网络向IMS网络演进的方案探讨

在通信网络向全IP化驱动下,话音业务网适用全IP化运营和业务发展而向IMS(IP多媒体子系统)网络演进.通过对IMS技术特征分析,提出IMS网络承接固网话音通信业务和满足视频通信业务需求的部署以及固网网络演进方案.     

数据网络中的回声消除技术

互联网在全球的普及导致了创新和应用的爆炸性发展,因此电信运营商注定要在数据通信占主导的现代网络基础设施中支持话音业务.尽管数据通信现在已经超过了话音业务,服务提供商的主要收入仍然来自传统的话音业务.为了保护这个收入来源,电信运营商必须保持当前话音业务的质量,并且创建支持各种新型数据业务的网络.话音业务在无线、有线、电路交换以及分组交换网络中,均至关重要,而话音质量作为一个保持因素,是一项基本要求.     

12580精细化营销的系统实现和思考

12580系统精细化营销涉及多种数据类型.通过分析和设计,实现了12580系统精细化营销支撑模块.同时,也分析了支撑模块架构对营销案的迅速支撑和高可扩展性等优点,为今后12580乃至其它业务系统的后续系统开发总结相应的设计原则和经验.     

LTE话音业务解决方案

话音业务作为传统电信网络最重要的一种业务在LTE/EPC下如何实现是运营商和制造商非常关注的问题。本文讨论了LTE话音业务的意义和需求,在此基础上进一步探讨LTE话音业务的实现和部署方案。     

TD-LTE CSFB话音解决方案的理论研究

话音业务是移动通信网为用户提供的基础业务之一,也是运营商最重要的业务收入来源之一。在2G/3G网络,话音业务由核心网电路域(CS)提供支持。在LTE时代如何顺利实现对话音业务的承接,成为一个重要的研究课题。     

基于IMS的广电话音网体系架构分析

为适应三网融合的推进,创新业务形态、丰富服务内容是探索发展的一项重点,广电话音业务是基于有线电视网络提供的国内IP业务,本文介绍了IP多媒体子系统技术的技术体系和标准进展,以及业务方案和业务策略,并对其发展前景进行了分析.     

中国移动综合信息服务门户12580整合创新

分析了中国移动综合信息服务门户12580目前的市场环境和业务现状,以及与"号码百事通"等竞争对手的实力对比,从角色定位、业务开展与融合及营销手段等多方面对12580的建设、提升竞争力的策略等进行了探讨.认为中国移动正在力争建设一流的通信企业和从"移动通信专家型"向"移动信息专家型"转变,亟需在信息服务领域提升竞争力.在"一个中国移动"理念的指导下,实现12580信息与数据的共享;需要加大业务整合的力度,丰富业务的内容,开展分层、有针对性的市场营销;需要充分利用中国移动客户群的优势,提高用户的参与度,增强客户的黏性和忠诚度等,使12580成为真正的综合信息服务门户和业务开展与推广的新平台.     

呼叫中心 干警的好帮手

为丰富商业模式的多元化，提升12580影响力和业务知晓度，陕西移动目前上线“12580生活播报”业务。     

A New Performance Measurement for Voice Transmission in Burst and Packet Switching

Voice transmission in burst switching is characterized by the process of talkspurt clipping, while in packet switching, it is characterized by the process of packet delay. In most analyses, the talkspurt clipping has been measured by the clipping probability averaged over all bits, and the packet delay has been measured by the delay performance averaged over all packets. The resulting measures overlook the duration of clipping in a talkspurt and the significant difference of delay in packets arriving at different times. Because of the nature of voice, different effects of these may result in substantially different degrees of voice distortion. This paper studies the worst case performance of both processes. The voice traffic is modeled as a process alternating between overload and underload periods. Statistically, more clipping and delay will be incurred while in the overload period. By worst case we mean that, in burst switching, we measure the worst case of talkspurt clipping duration in an overload period, while in packet switching, we measure the worst case of packet delay in an overload period. Furthermore, a simple closed form equation is derived which gives a very good approximation of the worst case mean packet delay performance. This equation can be more generally applied when the packet service time is to be geometrically distributed or when voice and data are to be integrated. The voice performances in burst switching and packet switching are also compared.     

Voice/Data Integration Using Circuit Switched Networks

This paper assesses the impact of integrating voice and data over circuit switched networks. Three main types of circuit switching are considered: 1) traditional circuit switching, 2)fast circuit switchingemploying advanced switching speeds, and 3) enhanced circuit switchingemploying time assigned speech interpolation (TASI) and adaptive data multiplexing (ADM) techniques. The circuit switching networks are evaluated in terms of two main network performance parameters: transmission efficiency and delay. In addition, an evaluation is made of such things as protocol and error control, precedence and preemption, routing and flow control, synchronization, voice continuity, probability of error or loss, and classmarking flexibility. One of the main conclusions of this paper is that circuit switching technologies have several deficiencies associated with providing integrated voice/data service and that the future lies in the effective use of packet and hybrid (circuit/packet) switching technologies.     

无比特丢弃的AAL2分组话音系统最优分组长度的研究

该文根据分组话音业务的特点,结合分组话音业务服务质量要求,特别是分组丢弃概率和端到端分组传送时延的要求,研究AAL2分组话音系统中AAL2分组最优长度的确定方法,得出结论:对于无比特丢弃的AAL2分组话音系统,当话音采用32kb/s的编码时,AAL2分组的最优长度大约为31个字节;当话音采用16kb/s的编码时,AAL2分组的最优长度大约为27个字节。此时AAL2分组的分组头开销小,话音分组的丢弃概率和端到端分组传送时延低,所得的分组话音质量高。     

VoIP网络信息安全研究

VoIP是基于分组交换的话音传输技术,与基于电路交换的传统电话架构有很大的不同。导致了VoIP网络特有的安全问题。另外,VoIP网络的实时性要求高,由于IP网络服务自身的特性以及各种安全策略的应用,给VoIP网络的性能和服务质量提出了挑战。对VoIP网络存在的安全威胁及各种安全措施进行分析,并对VoIP系统的部署和VoIP管理政策的制定提出建议。     

基于SIP协议的VoIP网关的设计和实现

在传统方式下,话音业务和数据业务是由两个互相独立的网络———公共电话交换网(PSTN)和数据通信网分别提供的。公共电话交换网基于电路交换的结构,这种技术保证语音有优良的品质,但同时也给PSTN带来线路资源分配方案的效率低和通话传真成本高的局限性。数据通信网则是基于包交换,由于采用了简单健壮的IP协议,使得其在这几年发展迅速。分析了目前在IP通信领域的两大主流协议H.323和SIP的特点和差异,给出了对支持SIP协议的VOIP网关的软硬件设计方案。     

AAL2分组话音复接器带宽分配算法的研究

本文根据分组话音业务的特点,结合分组话音业务服务质量的要求,特别是分组丢弃概率和平均分组排队时延的要求,研究AAL2分组话音复接器带宽分配算法.得出结论:对于无比特丢弃的AAL2分组话音复接器,按平均速率分配带宽基本上可以满足分组话音业务服务质量的要求;如果适当降低ATM VC的带宽利用率ρ(例如:令ρ=0.9),则可以进一步提高话音质量,获得令人满意的话音;对于带比特丢弃的AAL2分组话音复接器,按平均速率分配带宽,可以很好地满足分组话音业务服务质量要求,获得较高质量的话音.计算机仿真证实了上述结论是正确的.     

LTE语音模式选择及建设方案

由于LTE只能提供分组业务,而对语音、短信业务的需求还存在长期的需求。本文针对在LTE网络下,语音/短信业务实现的几种方式进行深入的研究和分析,尤其在CSFB和IMS语音共存下,对EPC核心网的改造要求,其中重点阐述了CSFB实现流程、VOLTE-IMS语音实现和eSRVCC切换技术等。     

The cost of quality in Internet-style networks

In broad terms, the quality of service (QoS) of a wide-area network is a measure: of how well it does its job-how quickly and reliably it transfers various kinds of data, including digitized voice and video traffic, from source to destination. With the advent of packet switching and the proliferation of many kinds of communications traffic (time-sensitive financial transactions, still images, large data files, voice, video, and so on), there are more than one set of criteria to satisfy. The the data rate needed for satisfactory voice communication may take an intolerable time to transfer high-resolution images. Conversely, the degree of network latency acceptable in transferring some files may not be adequate for real-time voice. So QoS has become a hot topic, and the contracts that specify it, called service level agreements (SLAs), are becoming more and more common, at least between service providers and their largest customers. As incumbent providers of telecommunications service are increasingly being challenged by competitive carriers, QoS has become a convenient marketing tool for both. The ability truly to deliver quality of service will separate the winner from the losers in the packet-switched future. The paper defines QoS, its priorities, and improvement     

A Distributed Experimental Communications System

The packet experimental communications system (packet XCS) is a new experimental voice and data switch. It uses a local-area network (LAN) for digital voice transmission, with local intelligence for switching. The packet XCS also has highly distributed control. The individual sites cooperate to provide user services as well as internal data management. We have learned that several local networks, including CSMA/CD networks, can be made to work well for voice transmission and that highly distributed control is practical in such a system. A system has been constructed which is used as a testbed for distributed voice and data communications experiments. This system is purely for experimentation and does not indicate a direction for future Bell System product offerings.