详细介绍了JTAPI的组成及使用方法，并在此基础上论证了用JTAPI实现军用指挥网中话音传输优点，最后论述了在军网中用JTAPI开发语音通信的方法。
This thesis gives a detailed introduction to the JTAPI composition and operation method and demonstrates the advantages of JTAPI on realizing voice transmission in commanding autoimmunization network.
呼之欲出的第三代(3G)移动通信不仅能提供现有的各种移动电话业务，还能提供高速率的宽带视频业务，支持高质量的话音、分组数据业务以及实时的视频传输。
The incoming 3g could not only support the existing mobile operations, but also broad band video operations high quality voice packet data transmission and real time video communication.
双音多频是一种在话音信道用音调来表示数字的方法，它可以用来在模拟话音信道传输信令，因此在通信中有广泛的应用。
Provide is a voice channel and tone to represent Numbers method, which can be used to transmit analog voice, so in communication has been widely used.
一种数字私线服务，传输话音、数据、视频以及其他的数字信号。
A digital private line service that transmits voice, data, video and other digital signals.
一种传输电话信令信息方式，所使用的路径与话音路径不同。
A way of carrying telephone signaling information along a path different from the path used to carry voice.
模拟结果表明，该协议能满足话音的实时要求，性能良好，且数据传输性能没有明显下降。
Simulation result shows that the protocol is suitable for real-time voice communications and the performance for data transmission has no significant degradation.
一种数据信道，其传输的信息可取该信道规定界限内的任何值。话音信道就是模拟信道。
A data channel on which the information transmitted can take any value between the limits defined by the channel. Voice - grade channels are analog channels.
该协议在保证话音终端的优先权的情况下，允许数据终端在报文的传输期间在连续多个帧中预约多个信息时隙。
Under the condition of giving high priority to voice terminals, data terminals are allowed to reserve multiple information slots in consecutive time frames.
本文分析了公用通信网中非话音信号传输的特点，提出并实现了万门话路非话信号检测系统的研制方案。
The features of non-voice signals on public communication nets are analysed and an approach to distinguish such signals is introduced in this paper.
双音多频(DTMF)是一种在话音信道用音调来表示数字的方法，它可以用来在模拟话音信道传输信令，因此在通信中有广泛的应用。
Provide (DTMF) is a voice channel and tone to represent Numbers method, which can be used to transmit analog voice, so in communication has been widely used.
该文介绍了一种基于802.3以太网协议传输话音和数据的局域通信系统的设计。
A design of local communication system which is used to transfer speech and data based on802.3Ethernet protocol is introduced in this paper.
因此，该系统具有广泛的应用前景，可以应用于诸如安全通信、话音多路传输、卫星通讯、多媒体应用、蜂窝电话等多种语音处理的场合。
Thus the system has widely potential application at voice processing aspects, such as secure communications, voice multiplexing, satellite communications, multimedia applications, cellular telephony.
话音信号是一个模拟信号，通过采样、量化和编码，我们可以实现话音信号的数字传输。
The speech signal is a analog one, and we can perform its digital transmission by sampling, quantizing and coding.
介绍了一种基于802.3以太网协议传输话音和数据的局域通信系统的设计。
A design of local communication system which is used to transfer speech and data based on 802.3 Ethernet protocol is introduced in this paper.
在分析现有利用AAL1传输话音的各项技术后，提出了利用AAL5实现话音AT M交换的方法。
After analyzing the technologies of using AAL1 to transmit the voice, this paper provides a method of using AAL5 to transfer voice over ATM.
该系统不但实现了话音的高速传输，并为开发MAC层协议和网络层协议的研发人员提供了与实际环境相符的测试平台。
The system realizes a high speed transmission for voice and provides a testing platform matching the actual environment for researchers who study on MAC and network protocol.
统一的消息传输把电子邮件、传真和话音邮件等消息统统汇集到可由pc机访问的邮箱中。
Unified messaging funnels email, fax and voice-mail messages into a mailbox that can be accessed by a PC.
即IP电话，是将话音编码、压缩转换成数据包，在IP网络中进行传输的技术。
VoIP( Voice over IP), i. e. IP phone, tries to digitalize voice, compress and transmit it through IP networks.
ATM技术的主要目的是使数据业务、视频业务、话音业务等能在统一的全数字网络中实现传输和交换，同时提供灵活简捷有效的动态资源分配。
The main intention of ATM is to integrate data, video, audio services as an unique service background, meanwhile to offer flexible and dynamic resource allocation methods.
IP电话系统不仅能传输话音，而且能提供多种智能服务，完全可以取代传统的个人分支交换(PBX)系统。
IP telephony system can not only provide voice application, but also provide many kinds of intelligent services, it can replace the traditional Private Branch eXchange (PBX) system completely.
根据电离层影响无线电波传输的有关理论，本文设计并实现了一种基于DSP的话音带宽短波信道模拟器。
In this paper, a DSP-based HF channel simulator within audio bandwidth is designed and implemented based on the concerned theory on the effects caused by the ionosphere in radio wave.
处理话音数据，通过压缩话音和去除抖动来制造供传输用的话音取样。
VOIP solutions use a digital signal processor (DSP) to process the voice data, preparing the voice sample for transmission by compressing voice and removing jitter.
为了实时传输话音信息，传真业务只能采用非实时方式，利用存储转发技术来实现。
The FAX service in MUX is non real time one, which is implemented by means of store&forward. The implementation of FIU should be deeply considered the compatibility issue with other fax machines.
数字化防空旅的通信网组网效能评估采用排队论和ADC效能模型，从话音通信和数据传输两方面进行评估。
Using Markov process theory of random process theory and adopting ADC efficiency model proposed by WSEIAC of USA, the paper establishes efficiency evaluation model of the system.
数字化防空旅的通信网组网效能评估采用排队论和ADC效能模型，从话音通信和数据传输两方面进行评估。
Using Markov process theory of random process theory and adopting ADC efficiency model proposed by WSEIAC of USA, the paper establishes efficiency evaluation model of the system.
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