彩铃——电信增值业务新亮点

“彩铃”是电信业务较新推出的一项增值服务。“彩铃”是其商业化名称，其实质为“个性化回铃音业务”(Coloring Ring Back Tone，简称CRBT)，即由被叫用户定制，在主叫用户发起呼叫之后且被叫用户尚未接听之前，为主叫用户提供一段悦耳的音乐或一句问候语来替代原来普通的“嘟嘟”叫铃音的业务。     

扩音呼叫电话机的设计

介绍了基于AT89C2051单片机的扩音呼叫电话机的硬件与软件设计.该电话机能根据预定振铃数目后自动对主叫话音进行扩音,这一特点在石油、化工、煤炭等强噪音的生产场所能使主叫用户快速、简便的与被叫取得联系.     

来电显示纵横谈

来电显示业务,它的专业名称为主叫识别信息传送及显示业务(calling identity delivexy),简称CID,是向普通用户提供的一种电信新业务,可在被叫用户终端设备(话机或显示器)上显示主叫电话号码、呼叫日期和时间等信息,并进行存储以供查阅。现已在一些发达国家和地区广泛推行,据不完全统计,全世界现有1200万用户使用该业务。     

基于IMS的视频电话通行证业务的设计与实现

视频电话通行证(Video phone Passport,VPP)业务是指在主叫用户拨打被叫用户电话的时候,首先录制一段视频来表明自己的身份,被叫通过观看视频,确认用户身份后决定是否接听电话的业务.基于IMS(IP Multimedia Subsystem,IP多媒体子系统)网络的体系结构以及视频电话通行证业务的业务特征,本文提出了一种利用VPP业务平台来实现该业务的解决方案,其中包括组网方案、系统实现方案以及基于SIP(Session Initiation Protocol,会话初始协议)消息的信令流程.最后,对业务进行了特点分析和总结.     

移动终端智能应答系统的实现

本系统针对当前移动手机用户在不方便接听、不愿意接听电话例如在开会、开车等各种各样的情形下．通过委婉的方式表达个人处境,达到礼貌告知对方的目的而开发的智能应答系统。业务的基本模式就是捕获到用户忙的事件,如果可能还需要区分是用户通话忙还是拒接忙,然后通过USSD向被叫推送一个提示如何回复用户的操作菜单,最后通过被叫用户的选择,给主叫用户一个答复,例如一个预定义的短信、IVR放音、转接语音信箱等多种方式。     

单片机来电显示器的设计与实现

介绍一种利用单片机AT89C52接收并显示电话主叫号码的来电显示器。该显示器在被叫挂机 状态下接收以频移键控(FSK)方式传送的主叫识别信息,可显示主叫电话号码、呼叫日期、 时间等信息,并可存储多达120条的主叫号码信息。用户使用上、下翻转键及删除键可方便 地查阅或删除来电信息。     

单片机来电显示器的设计与实现

介绍一种利用单片机AT89C52接收并显示电话主叫号码的来电显示器.该显示器在被叫挂机状态下接收以频移键控(FSK)方式传送的主叫识别信息,可显示主叫电话号码、呼叫日期、时间等信息,并可存储多达120条的主叫号码信息.用户使用上、下翻转键及删除键可方便地查阅或删除来电信息.     

倡导智能业务

“我是Cathy,我现在无法接听您的电话,如有急事,请留下您的电话、留言;稍后,我再与您联系。”如果您是一位商务繁忙的人士,正在与客户洽谈一笔重要的业务,但是又担心因此错过另一个也许更重要的客户,除了在办公室电话上留言,您还可以申请自动秘书业务、在手机上开通此服务功能,这一业务主要用于无线用户呼叫另一无线用户,如果被叫忙或无应答,则无线系统自动将该主叫接入独立的IP系统。目前该业务在泰国Telecom Asia电信运营公司已经投入应用,由国研北邮通信技术有限公司作为系统集成商。     

MD110低版本交换机来电显示的设计与实现

利用单片机控制技术,设计一个“用户模块”,在电话初次振铃时,给交换机送一个假取机用户信令,使主叫号码通过音频通道送到被叫话机上,实现来电显示功能。     

用单片机改造长话计费系统

<正> 一、问题的提出DD16-ⅡA 型长途电话对端设备的计费系统采用DBJ-Z80型单板机控制。其主要功能是实时采集长途电话通话过程中的主叫用户电话号码、被叫所在城市     

专用语音信箱数据采集与处理

介绍的专用语音信箱由２ 条中继线、７ 台分机、１ 个语音处理单元以及呼叫处理程序和语音信箱管理程序组成。内外线电话能够互相呼叫或拨号访问信箱,并拥有多种程控业务新功能。语音信箱具有查询、留言和播放公众信息等功能,在被叫忙音或无应答时自动进入信箱。呼叫处理程序使用状态迁移法解决呼叫信号采集、处理的多重性问题;使用时间调度技术解决多用户的实时处理问题;使用ＶｉｓｕａｌＢａｓｉｃ的ＭＳＣＯＭＭ 控件实现了串行口交互通信;将多媒体ＭＩＣ控件用于语音信息的记录、储存和重放过程。该语音信箱有明显的实用价值     

基于Kano模型的家庭成员通信需求的研究与设计

随着人民日益增长的数字化美好生活需要,家庭内部成员间的通信交流方式已从“单一”的语音通话向“多元化”的信息通信方式转变。目前家庭网主要提供成员间语音通话服务,已较难满足成员间视频化、社区化等多样化的通信需求。本文使用Kano模型针对家庭成员间的通信需求进行调研与分析,并将需求优先级排序转化为设计目标的重要程度,在满足用户需求的前提下,提升办理体验,同时为类似通信类产品研究与设计提供参考。     

Enhancing VoIP service for ubiquitous communication in a campus WLAN with partial coverage

The IEEE 802.11 WLAN technology has become the de facto standard for wireless Internet access. The spotty coverage of WLAN access points, however, confines the applicability of many real-time services such as VoIP within the boundary of the WLAN service area. In this paper, we investigate the problem of enhancing VoIP service for ubiquitous communication in a WLAN with spotty service area. We consider a university campus that has an established infrastructure for supporting SIP-based VoIP service through either wired or wireless data networks. The campus WLAN service does not have 100% full coverage, and hence users cannot make untethered VoIP calls anywhere on campus. The goal of this paper is to overcome the limitations of such “dead spots” for motivating the use of campus IP telephony service. To proceed, we start with two approaches called one-hop extension and dual-mode communication. The first approach uses multi-hop relay to extend the WLAN coverage, while the second approach leverages the availability of dual-mode handsets for ubiquitous voice communication. We implement the two approaches, and evaluate their performance in the campus testbed environment. We find that while the two approaches can effectively allow voice communication in WLAN dead spots, they have one common problem as the potential lack of support for voice call continuity that can cause degradation of the speech quality to an active call. We adopt a cross-layer solution based on signal processing algorithms to address the problem, thus achieving seamless voice call continuity while enabling ubiquitous voice communication on campus. Testbed evaluation shows promising results for future research along the proposed direction.     

Transference and retrieval of voice message over low signal strength in satellite communication

This paper presents a method of transferring voice using short messaging service in satellite communication system. The method is especially applicable in a situation where signal strength is low and voice call is not possible. In a tunnel, basement or environment with bad climate conditions, signal strength usually gets weak which make voice call difficult but SMS works in such situation. An application has been developed using J2ME language in order to test the proposed method. For experimentation, Thuraya SG-2520 satellite phone has been used.     

基于SIP的矿用应急通信业务系统设计

设计了一种基于SIP的矿用应急通信业务系统,详细介绍了系统架构设计及紧急呼叫、通信终端区域定位、多路对讲和语音广播等业务流程的设计。该系统遵循标准SIP协议及3GPP提出的标准SIP协议扩展框架,实现了具有较高集成度的矿用调度通信与应急通信业务系统,为矿山井下通信联络、安全生产提供了有效的联络手段与通信保障,提高了井下通信系统的综合利用率。     

基于SIP的远程呼叫中心的设计方案

呼叫中心是指通过互动式语音应答和人工坐席通信为客户提供协助和咨询的交互式增值服务的系统。传统的呼叫中心更多的依赖于硬件实现,且不易扩展。此文提出了一种基于SIP协议的呼叫中心,与传统的基于CTI技术的呼叫中心相比,有更好的扩展性和灵活性;同时此系统可以通过Internet网络传输远程呼叫大大节省了通话费用。     

Talking Call Waiting: An Application of Text-to-Speech

Call Waiting is a service provided by most telephone companies that alerts a subscriber to an incoming call while he/she is engaged in a prior call. The Talking Call Waiting service at Ameritech enhances Call Waiting by converting Caller ID information into a spoken utterance using text-to-speech technology. A subscriber to Talking Call Waiting hears the name associated with the line that originates a call to them while he/she is on the phone.Customer acceptance of hearing a text-to-speech synthesized spoken name that interrupts an ongoing conversation was a concern during the concept and design phases of this application. We designed a set of experiments that enabled us to predict customer acceptance of the product based on three factors: 1) the tolerance of the subscriber for interruption in his/her current conversation; 2) the intelligibility of the text-to-speech synthesis (i.e., how successful subscribers are at understanding the name of the call waiting caller); and 3) the perceived quality of the text-to-speech synthesis. We also designed a process that formats the name data for optimal text-to-speech synthesis and a mechanism to respond to possible customer dissatisfaction with the synthesis of particular names. Our research with the service prototype and field system indicate that intelligibility of names was good, quality of the sound was acceptable, and disruption of the call in progress was tolerable. The value of the information about the calling party, conveyed via the spoken name, we believe outweighed the fact that the sound of the speech lacked the quality some customers expected.     

支持多媒体和移动业务的软交换系统设计

软交换是NGN中的一项重要技术,目前,支持语音通信的软交换技术已经比较成熟,但在支持多媒体和移动业务方面仍然需要进一步研究。文中分析了支持多媒体和移动业务的软交换系统的软件功能结构,简要说明了软交换系统的呼叫控制、承载控制、切换管理、业务适配的设计思路。     

保护信道和排队相结合的信道分配策略

提出了一种适用于语音、数据呼叫的蜂窝移动通信系统的信道分配策略。该策略为数据呼叫提供保护信道,降低数据呼叫的阻塞率。同时,采取语音呼叫排队策略抑制数据保护信道引起的语音呼叫阻塞率的恶化。为了进一步提高系统的性能,在策略中引入了不耐烦顾客,并建立了带有不耐烦顾客的排队模型。仿真结果表明该策略能够有效地降低语音呼叫和数据呼叫阻塞率,改善系统性能。     

VoIP development in China

Several factors have spurred the explosive growth of VoIP phone use in China, including customer incentives such as improved voice quality and lower cost per call, and provider incentives such as higher profits and upgrade paths to next-generation technologies. The voice over Internet protocol, also called IP telephony, offers a new type of service that uses the Internet protocol, intranets, and extranets to deliver voice information. In contrast to traditional telephone services, which operate through a circuit-switched network, VoIP uses a packet-switched network. This distinction results in differences in implementation, quality of service (QoS), and operating costs. Since the service was introduced to the public in China in April 1999, VoIP toll telephone traffic has increased with astonishing speed. By the end of 2002, VoIP toll telephone traffic had surpassed traditional toll telephone traffic in China in both domestic longdistance and international call areas, including phone calls to and from Hong Kong, Macao, and Taiwan.