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公开(公告)号：US08543704B2
公开(公告)日：2013-09-24
申请号：US11910301
申请日：2006-04-06
摘要： A voice server can be located, temporarily allocated, and sent audio. The results are returned to a voice client, and the voice server is deallocated for use by the next person talking into their client browser. Voice channels and IVR ports are initially set up by a switch and the IVR using conventional audio protocols. The voice channels are not initially connected to the client. The switch handles the allocation and deallocation of IVR voice channels without having to communicate further with the IVR. A user indicates to the client device that he wishes to initiate a voice interaction during an X+V session. This translates to a request on the CTRL channel to synchronise XHTML and VXML forms as a trigger for the VXML browser to execute a conversational turn. A multiplexer intercepts this control command and establishes a virtual voice circuit between the client device and an existing open but unattached voice port. The virtual circuit is established without having to set up an RTP channel. The CTRL signal is then forwarded to an interaction manager so that the conversation can take place. At the end of the conversation the virtual circuit is disconnected.
摘要翻译： 语音服务器可以被定位，临时分配和发送音频。 结果返回到语音客户端，并且语音服务器被释放以供下一个人在其客户端浏览器中使用。 语音通道和IVR端口最初由交换机和IVR使用常规音频协议设置。 语音通道最初没有连接到客户端。 交换机处理IVR语音信道的分配和释放，而不必进一步与IVR通信。 用户向客户机指示他希望在X + V会话期间发起语音交互。 这转换为CTRL通道上的请求，以将XHTML和VXML表单同步为VXML浏览器执行会话转弯的触发器。 多路复用器拦截此控制命令，并在客户端设备和现有的已打开但未附加的语音端口之间建立虚拟语音电路。 建立虚拟电路而不必设置RTP通道。 然后将CTRL信号转发给交互管理器，以便进行通话。 在会话结束时，虚拟电路断开连接。
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公开(公告)号：US20090144428A1
公开(公告)日：2009-06-04
申请号：US11910301
申请日：2006-04-06
IPC分类号： G06F15/16
摘要： This invention is based on being able to locate a voice server, temporarily allocate it, send it the audio of you saying “When is flight 683 due to arrive?”, getting the results of what you said back in the browser, and deallocating the voice server for use by the next person talking into their browser. Voice channels and IVR ports are initially set up by a switch and the IVR using conventional audio protocols. The Voice channels are not initially connected to the client. The switch handles the allocation and deallocation of IVR voice channels without having to communication further with the IVR. A user indicates (usually by pressing a PTT button) to the client device that he wishes to initiate a voice interaction during an X+V session. This translates to a request on the CTRL channel to synchronise the XHTML and VXML forms which the embodiment uses as a trigger for the VXML browser to execute a conversational turn. The multiplexer intercepts this control command and connects the virtual voice circuit between the device and an existing open but unattached voice port. The virtual circuit is connected without having to set up an RTP channel. The CTRL signal is then forwarded to the interaction manager so that the conversation can take place. At the end of the conversation the virtual circuit is disconnected.
摘要翻译： 本发明基于能够定位语音服务器，临时分配，发送您的“你什么时候到达飞行683”的音频，在浏览器中获取你所说的结果，并取消分配 语音服务器供下一个人在浏览器中使用。 语音通道和IVR端口最初由交换机和IVR使用常规音频协议设置。 语音通道最初没有连接到客户端。 交换机处理IVR语音信道的分配和释放，而不必与IVR进一步通信。 在X + V会话期间，用户指示（通常通过按PTT按钮）到客户端设备，他希望发起语音交互。 这转换为CTRL通道上的请求，以同步XHTML和VXML形式，该实施例用作VXML浏览器的触发器来执行会话转弯。 多路复用器拦截该控制命令，并将虚拟语音电路连接到设备和现有的开放但未附加的语音端口之间。 虚拟电路连接而不必设置RTP通道。 然后将CTRL信号转发到交互管理器，以便进行通话。 在会话结束时，虚拟电路断开连接。
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公开(公告)号：US07877498B2
公开(公告)日：2011-01-25
申请号：US11421094
申请日：2006-05-31
IPC分类号： G06F15/16
CPC分类号： H04L65/605 , H04L65/608 , H04L65/80
摘要： The invention streams data by identifying an existing streaming data channel and disabling the channel so that data can not be streamed; breaking the channel to form at least one pair of channel connection points; connecting at least one streaming data plug-in between the connection points; and enabling the channel so that streaming data can flow through the channel via the plug-in, wherein the at least one plug-in can process the streaming data as it flows trough the channel.
摘要翻译： 本发明通过识别现有流数据信道并禁用信道来流数据，使得数据不能被流传输; 断开通道以形成至少一对通道连接点; 在连接点之间连接至少一个流数据插件; 以及启用所述通道，使得流数据可以经由所述插件流过所述通道，其中所述至少一个插件可以在所述流数据流经所述通道时处理所述流数据。
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4.发明授权System and method for optimizing a device driver by incorporating debugging and tracing 失效通过集成调试和跟踪来优化设备驱动程序的系统和方法公开(公告)号：US06526567B1
公开(公告)日：2003-02-25
申请号：US09375063
申请日：1999-08-16
IPC分类号： G06F945
CPC分类号： G06F11/366 , G06F8/54 , G06F11/3624 , G06F11/3636
摘要： The invention relates to a method for executing, in a computer system 5, a device driver 10 which is used by a number of applications running on that system. A first 12 and second version 11 of the device driver are provided, with the first version being optimised and the second version containing debug code for providing trace information. Upon discovering a problem with the device driver, it is possible to selectively switch to the debug version of the code without any of the applications already using the device driver being aware of the switch. Having gathered the necessary trace data, it is then possible to transparently switch back to the optimised version. Conventionally, device drivers form part of the operating system within a computer and so are difficult to remove and re-load without taking the whole system down and re-booting. The invention however affords continuous service to any application already using the device driver and does not require a time-wasting reboot which may in any event remove the error condition.
摘要翻译： 本发明涉及一种用于在计算机系统5中执行由在该系统上运行的多个应用程序使用的设备驱动程序10的方法。 提供设备驱动程序的前12和第二版本11，其中第一版本被优化，并且第二版本包含用于提供跟踪信息的调试代码。 当发现设备驱动程序的问题时，可以选择性地切换到代码的调试版本，而没有任何已经使用设备驱动程序知道该开关的应用程序。 收集必要的跟踪数据后，可以透明地切换回优化版本。 通常，设备驱动程序构成计算机内的操作系统的一部分，因此难以在不将整个系统关闭并重新启动的情况下去除和重新加载。 然而，本发明对已经使用设备驱动程序的任何应用程序提供持续服务，并且不需要浪费时间的重新启动，其可以在任何情况下消除错误状况。
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5.发明授权公开(公告)号：US06748055B1
公开(公告)日：2004-06-08
申请号：US09221018
申请日：1998-12-23
IPC分类号： H04M164
CPC分类号： H04M3/487
摘要： A voice processing system for connection to a telephone line for providing telephony support to voice processing applications having a voice processing application and a media object, the media object having an output element for outputting a set of output components in accordance with one or more presentation criteria such as locale and style; a receiving element for receiving the media object from the voice processing application, the media object representing desired output over a telephone line; and the outputting element outputting the output components over the telephone line.
摘要翻译： 一种用于连接到电话线的语音处理系统，用于向具有语音处理应用和媒体对象的语音处理应用提供电话支持，所述媒体对象具有用于根据一个或多个呈现准则输出一组输出组件的输出元件 如地域和风格; 用于从所述语音处理应用接收所述媒体对象的接收元件，所述媒体对象通过电话线表示期望的输出; 并且输出元件通过电话线输出输出分量。
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公开(公告)号：US08365189B2
公开(公告)日：2013-01-29
申请号：US12039100
申请日：2008-02-28
IPC分类号： G06F3/00
CPC分类号： H04L67/2823 , G06Q10/107 , H04L65/1016 , H04L65/40 , H04L67/28 , H04L67/327
摘要： This invention relates to a method, system and computer program product for managing a service message in a service oriented architecture system including a service provider, a service consumer and a set of control services, the method, system and computer program product comprising the following steps: receiving a service message; selecting a group of rules from a set of rule groups depending on the type of service message; selecting a control service from a set of control services and instructing the selected control service according to one or more of the rules from the selected rules group applied to the service message.
摘要翻译： 本发明涉及一种用于在面向服务的架构系统中管理服务消息的方法，系统和计算机程序产品，该系统包括服务提供商，服务使用者和一组控制服务，所述方法，系统和计算机程序产品包括以下步骤 ：接收服务消息; 根据服务消息的类型从一组规则组中选择一组规则; 从一组控制服务中选择一个控制服务，并根据应用于服务消息的所选择的规则组中的一个或多个规则来指示所选择的控制服务。
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公开(公告)号：US20090144060A1
公开(公告)日：2009-06-04
申请号：US12326030
申请日：2008-12-01
CPC分类号： G10L13/00
摘要： Disclosed is a system and method for generating a web podcast interview that allows a single user to create his own multi-voices interview from his computer. The method allows the user to enter a set of questions from a text file using a text editor. (Answers may also be entered from a text file although this is not the more preferred embodiment.) For each question, the user may select one particular interviewer voice among a plurality of predefined interviewer voices, and by using a text-to-speech module in a text-to-speech server, each question is converted into an audio question having the selected interviewer voice. Then, the user preferably records answers to each audio question using a telephone. And a questions/answers sequence in a podcast compliant format is generated.
摘要翻译： 公开了一种用于生成网络播客面试的系统和方法，其允许单个用户从他的计算机创建他自己的多个语音面试。 该方法允许用户使用文本编辑器从文本文件输入一组问题。 （尽管这不是更优选的实施例，但也可以从文本文件输入答案。）对于每个问题，用户可以在多个预定义的访问者语音中选择一个特定的访问者语音，并且通过使用文本到语音模块 在文本到语音服务器中，每个问题被转换成具有所选访问者声音的音频问题。 然后，用户最好使用电话记录每个音频问题的答案。 并且生成符合播客格式的问题/答案序列。
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公开(公告)号：US6141724A
公开(公告)日：2000-10-31
申请号：US115921
申请日：1998-07-15
申请人： Nicholas David Butler , Jacqueline Bowden , Steven John Hyatt , David Seager Renshaw , Yuk-Lun Wong
发明人： Nicholas David Butler , Jacqueline Bowden , Steven John Hyatt , David Seager Renshaw , Yuk-Lun Wong
摘要： A system for remotely developing an telephony application for a call handling server comprises: program code components for each process used by the call handling server and an associated reduced code components for each process; an application designer for combining the reduced code components into an application design; networking capability for sending the application designer and reduced code components to a client and receiving a completed application design; and an application builder for assembling, according to the application design, the program code components into an self contained application.
摘要翻译： 用于远程开发用于呼叫处理服务器的电话应用的系统包括：由呼叫处理服务器使用的每个进程的程序代码组件和用于每个进程的相关联的减少的代码组件; 用于将缩减的代码组件组合成应用程序设计的应用程序设计器; 将应用程序设计器和减少的代码组件发送到客户端并接收完整的应用程序设计的联网功能; 以及应用程序构建器，用于根据应用设计将程序代码组件组装成自包含的应用程序。
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9.发明授权Generating a web podcast interview by selecting interview voices through text-to-speech synthesis 失效通过文本到语音合成选择面试声音来生成网络播客面试公开(公告)号：US08255221B2
公开(公告)日：2012-08-28
申请号：US12326030
申请日：2008-12-01
IPC分类号： G10L13/00
CPC分类号： G10L13/00
摘要： Disclosed is a system and method for generating a web podcast interview that allows a single user to create his own multi-voices interview from his computer. The method allows the user to enter a set of questions from a text file using a text editor. (Answers may also be entered from a text file although this is not the more preferred embodiment.) For each question, the user may select one particular interviewer voice among a plurality of predefined interviewer voices, and by using a text-to-speech module in a text-to-speech server, each question is converted into an audio question having the selected interviewer voice. Then, the user preferably records answers to each audio question using a telephone. And a questions/answers sequence in a podcast compliant format is generated.
摘要翻译： 公开了一种用于生成网络播客面试的系统和方法，其允许单个用户从他的计算机创建他自己的多个语音面试。 该方法允许用户使用文本编辑器从文本文件输入一组问题。 （尽管这不是更优选的实施例，但也可以从文本文件输入答案。）对于每个问题，用户可以在多个预定义的访问者语音中选择一个特定的访问者语音，并且通过使用文本到语音模块 在文本到语音服务器中，每个问题被转换成具有所选访问者声音的音频问题。 然后，用户最好使用电话记录每个音频问题的答案。 并且生成符合播客格式的问题/答案序列。
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公开(公告)号：US07660399B2
公开(公告)日：2010-02-09
申请号：US10818837
申请日：2004-04-06
IPC分类号： H04M11/00
CPC分类号： H04M3/487
摘要： A voice processing complex has a plurality of host machines, each host machine supporting one or more voice applications, at least one host in the complex including telephony hardware for providing access to a plurality of telephone lines. Such a host provides telephony functions to the complex. This is achieved by maintaining a mapping of voice applications to telephone lines, and creating a call object in response to an incoming call on one of the lines. It is then determined which voice application to associate with the incoming call on the basis of this mapping, and an identifier to the call object is then passed to the determined voice application. Thereafter the call object is responsive to requests from this voice application for providing telephony functions for the call. This includes receiving a media object from the voice processing application which represents desired output over a telephone line. This media object is then processed into a plurality of output components in accordance with one or more presentation criteria, and these components are then output over the telephone line.
摘要翻译： 语音处理复合体具有多个主机，每个主机支持一个或多个语音应用，所述复合体中的至少一个主机包括用于提供对多条电话线路的接入的电话硬件。 这样的主机向复合体提供电话功能。 这通过将语音应用程序映射到电话线并且响应于其中一条线路上的来话呼叫创建呼叫对象来实现。 然后，基于该映射确定哪个语音应用与呼入呼叫相关联，并且将呼叫对象的标识符传递到所确定的语音应用。 此后，呼叫对象响应于来自该语音应用的用于为呼叫提供电话功能的请求。 这包括从语音处理应用接收媒体对象，该媒体对象通过电话线表示期望的输出。 然后根据一个或多个呈现标准将该媒体对象处理成多个输出组件，然后通过电话线输出这些组件。
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