One example of the type of application / IP phone integration that LiteScape provides is conferencing support using WebEx/Microsoft LiveMeeting.
以下是一个此类应用与IP电话集成的例子，LiteScape提供了基于WebEx/MicrosoftLiveMeeting的会议支持。
The technology is also becoming a platform that other companies such as Infosys and WebEx are using to extend their products and services to the IP phone level.
该技术也成为许多其它公司竞相使用的平台，例如Infosys和WebEx利用该平台将自身的产品和服务扩展到IP电话级别。
Figure 4: The Process of Launching a WebEx Session from an ip-phone.
图4：通过IP电话启动WebEx会话的流程。
The IP phone is the most representative product.
网络电话就是其中最具有代表性的产品。
Then, the thesis shows a scheme of secrecy IP Phone.
接着，论文提出了一种IP保密电话机的设计方案。
Gatekeeper constitutes a very important part in IP phone system.
网守是IP电话系统中重要的组成部分。
The security of IP phone is the foundation of its large scale application.
IP语音业务的安全性是其得到大规模应用的重要前提之一。
How to make the IP phone more stable and easy to use like traditional phone?
如何将IP电话做成和传统电话一样稳定且简单易操作呢？
Application of IP phone net is introduced to save national and international long-distance fees.
介绍了如何运用IP电话网络来节省国际，国内长途通信费用。
VoIP( Voice over IP), i. e. IP phone, tries to digitalize voice, compress and transmit it through IP networks.
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。
This article describes the concepts of IP phone, key technology, realization principles, software and hardware supports.
本文作者阐述了IP电话的概念、关键技术、实现方式及所需的软、硬件设施。
JY-type single IP phone charging device is used to process phone users call timing and charging of the measuring instruments.
JY型单路IP电话计费器是用于对电话用户的通话过程进行计时计费的计量器具。
When the INFO button is pushed, the analog or IP phone automatically dials a preprogrammed number connecting any user with an operator.
当按下INFO按钮，模拟电话或IP电话将自动拨打操作员预置过的电话。
Traditional IP phone must depend on PSTN or PC when working, so this thesis proposed a Design of Voice-Terminal for network-conference Based on DSP.
目前的IP电话都必须依赖传统的公众电话网（PSTN）或PC机来实现，针对这种现状，本论文提出了一种基于DSP技术的网络会议电话的语音终端系统实现方案。
SIP is proposed by IETF as IP phone call order protocol which is totally a text modality that to be used to establish, modify and terminate multimedia calls and conversation.
会话初始化协议(SIP)是IETF提出的IP电话呼叫信令协议，是一个基于文本的协议，用来创建、修改和终止多媒体呼叫与会话。
Cisco TelePresence and IP Phone BU (TIPBU) is always focused on customer and partner success by designing and developing the world class voice and video collaboration endpoint.
思科TIPBU研发团队一直致力于通过设计和开发业界最先进的语音与视频终端来提高思科客户和合作伙伴的满意度。
This paper introduces the design of IP Phone terminal using SIP (Session Initiation Protocol) upon the OMAP5912 platform of a vehicle electronic multi-media video and audio system.
以车载汽车电子多媒体视听系统为研发背景，在OMAP5912平台上采用SIP(会话发起协议)设计了IP电话终端。
We can easily transplant the system into other domains such as faxes encrypting, IP Phone and private data network by changing the interfacing circuit and adding or deleting software modules.
通过适当地修改接口电路和增删软件模块，可以很方便地把所提出的系统方案移植到其它类似的应用领域中，如传真加密、IP电话加密、专用网络的数据加密等。
We can easily transplant the system into other domains such as faxes encrypting, IP Phone and private data network by changing the interfacing circuit and adding or deleting software modules.
通过适当地修改接口电路和增删软件模块，可以很方便地把所提出的系统方案移植到其它类似的应用领域中，如传真加密、IP电话加密、专用网络的数据加密等。
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