Abstract:
A methoc to access and use an integrated web site from a mobile environment is disclosed. The method includes communicating voice activated commands, using voice recognition technology, to an integrated web site via a wireless communication network and a global informational network. The integrated web site comprises a network-based media station integrated with a global search engine. The voice activated commands are used to access the integrated web site, initiate the transmission of streaming digital media content from the media station to a wireless mobile network terminal, and perform key word searches of t ie global informational network using the global search engine.
Abstract:
A voice application creation and deployment system includes a voice application server for creating and serving voice applications to clients over a communication network; at least one voice portal node having access to the communication network, the portal node for facilitating client interaction with the voice applications; and an inference engine executable from the application server. In a preferred embodiment the inference engine is called during one or more predetermined points of an ongoing voice interaction to decide whether an inference of client need can be made based on analysis of existing data related to the interaction during a pre-determined point in an active call flow of the served voice application, and if an inference is warranted, determines which inference dialog will be executed and inserted into the call flow.
Abstract:
A system (10) and method (100) for providing a message-based communications infrastructure for automated call center operation is described. A call (54) from a user (12) into a call center (11) is accepted. The accepted call (54) includes a stream of transcribed verbal speech utterances (57). Each transcribed verbal speech utterance (57) is recorded as a user message (22). The accepted call (54) is assigned to a session (20), which is then assigned to an agent (81). The call (54) is progressively processed in the assigned session (20) by presenting each user message (22) to the assigned agent (81), executing commands responsive to the assigned agent (81), and sending an agent message (23) to the user (12). The agent message (23) includes a stream of synthesized audible speech utterances (63).
Abstract:
Some embodiments of the present invention combine two or more separate networks into a unified tool for multimedia information delivery. In particular, the unified tool provides at least an aspect (for example, a visual aspect (400)) of the multimedia information over one of the separate networks and provides at least another aspect (for example an audio aspect (250)) of the multimedia information over another one of the separate networks. In still other embodiments of the present invention, the unified tool provides synchronized multimedia information delivery over the two or more separate networks. Some embodiments of the present invention enhance the speed of delivery of web content to users. In particular, an embodiment of the present invention is a method for speeding access to web content by a user which includes the steps of: a) at or prior to display of web content, accessing a list of identifiers of further web content; b) requesting web content pertaining to at least one of the identifiers before it is actually needed and; c) optionally, tagging the requested web content with header information which indicates that the requested content will be considered "fresh" for some period of time and not to re-request the web content during its "fresh" time period.
Abstract in simplified Chinese:经由通信系统的通信网络从源用户的远程源用户设备接收说源语言的源用户与说目标语言的目标用户之间的调用的调用音频,调用音频包含源用户用源语言的说话。自动说话辨识进程系在调用音频上运行。使用说话辨识进程的结果以产生源用户说话的目标语言的翻译。源用户说话的翻译合成说话音频版本系与源用户的调用音频及/或与目标用户说话的源语言的翻译音频混合。混合音频信号经由通信网络发送到目标用户的远程目标用户设备,以用于在调用期间输出给至少一目标用户。
Abstract in simplified Chinese:本发明揭示一种使电话用户能够参与一基于实时传讯之会议的系统(10)及方法(50),其可包括以下步骤:借由电话会议系统(24)接收(52)来自电话(26或28)的语音输入;将该语音输入转录(54)为一第一文本消息并将该第一文本消息发送(58)至复数个设备(18、20、26或28),该等设备耦接至一属于基于实时传讯之会议的实时传讯网络。该方法可进一步包括以下步骤:接收(60)来自该基于实时传讯之会议上的该等复数个设备中之任一设备的一第二文本消息;将该第二文本消息转换(62)为一语音输出;并将该语音输出经由电话会议系统而发送(68)至电话。
Abstract in simplified Chinese:本发明系揭露一种在包含一语音辨认模块、一时间管理模块以及一语音产生模块之一系统内,用于提供一服务给一用户之一方法其包含经由语音辨识模块接收一谈话;使用属于一数据模型之词典,将谈话转换为至少一结构;使用结构辨认谈话内之概念;若所提供之谈话包含足够的信息,依据概念选取一服务;依据选取的服务产生一文本消息;且使用语音产生模块将文本消息转换为一语音频息。
Abstract in simplified Chinese:本发明系关于通信系统以及在阻塞的通信信道中维持音频通信之方法,阻塞的通信信道目前承受发送方与接收方之间音频通信中的语音发送,通信信道具有至少一传讯信道以及至少一具有服务品质之酬载信道。酬载信道之服务品质在音频通信期间遭到监测。若酬载信道之服务品质低于阈值,则将各自发送方之语音转换成文本;并且透过保留之通信信道予以发送至各自接收方。可于接收方将文本转换回语音。