Bus connection set up and tear down
    1.
    发明授权
    Bus connection set up and tear down 失效
    总线连接设置和拆除

    公开(公告)号:US06272146B1

    公开(公告)日:2001-08-07

    申请号:US08965365

    申请日:1997-11-06

    IPC分类号: H04L1240

    摘要: A bus connection controller in a voice processing is for managing the connection of a timeslot on a time-division multiplex (TDM) bus to a port on an adapter. The voice processing system includes basic time-division multiplex (TDM) connection management to enable the coordination of connections between resources such as channels on line cards (SPacks or VPacks), and channels on digital signal processor (DSPs) cards that provide, amongst others things, voice recognition, text-to-speech, fax capabilities and so on. Problems are encountered when a telephone call in a voice processing system ends suddenly because one of the callers hangs up. If the telephony channel has connections with other channels or resources via a TDM bus, callers may hear spurious data. To address this problem each call is associated with its corresponding connection on the TDM bus and each connection is associated with its connection details including the adapters and ports involved in connecting the calls. When one of the calls ends all the relevant ports involved with the connection are immediately disconnected.

    摘要翻译: 语音处理中的总线连接控制器用于管理时分复用(TDM)总线上的时隙与适配器上的端口的连接。 语音处理系统包括基本的时分多路复用(TDM)连接管理,以实现诸如线路卡(SPack或VPack)之间的信道之类的资源之间的连接的协调,以及数字信号处理器(DSP) 事物,语音识别,文字转语音,传真功能等。 当语音处理系统中的电话呼叫由于其中一个呼叫者挂断而突然结束时遇到问题。 如果电话信道通过TDM总线与其他信道或资源连接,则呼叫者可能会收到虚假数据。 为了解决这个问题,每个呼叫都与其在TDM总线上的对应连接相关联,并且每个连接与其连接细节相关联,包括连接呼叫所涉及的适配器和端口。 当其中一个呼叫结束时,连接涉及的所有相关端口立即断开连接。

    Variation in playback speed of a stored audio data signal encoded using a history based encoding technique
    2.
    发明授权
    Variation in playback speed of a stored audio data signal encoded using a history based encoding technique 失效
    使用基于历史的编码技术编码的存储的音频数据信号的播放速度的变化

    公开(公告)号:US06223153B1

    公开(公告)日:2001-04-24

    申请号:US08594054

    申请日:1996-01-30

    IPC分类号: G10L100

    摘要: The invention relates to a voice processing system capable of varying the speed of output of digitized audio data stored therein. The digitized audio data is stored using blocks of LPC coefficients. Each block is sufficient to allow twenty milliseconds of speech to be generated therefrom. Periodically, or selectably, the utilization of particular blocks is repeated resulting in a decrease in the speed of output of the speech synthesized therefrom. Alternatively, selectably blocks of LPC coefficients are omitted from use thereby producing a corresponding increase in speech output.

    摘要翻译: 本发明涉及能够改变存储在其中的数字化音频数据的输出速度的语音处理系统。 数字化的音频数据使用LPC系数的块来存储。 每个块足以允许从其生成二十毫秒的语音。 周期地或可选择地重复使用特定块,导致从其合成的语音的输出速度降低。 或者,可选择地省略LPC系数的块,从而产生相应的语音输出增加。

    Client-server system
    3.
    发明授权
    Client-server system 失效
    客户端 - 服务器系统

    公开(公告)号:US6052367A

    公开(公告)日:2000-04-18

    申请号:US777723

    申请日:1996-12-20

    摘要: Using the Internet World Wide Web (WWW) network 320, a WWW Client 310 can communicate with a WWW Server 330 to request the reconfiguration of or to generate software which controls a voice processing application. A voice response system client communicates with a voice response system server to alter the configuration of the voice response system or control the execution of software on said voice response system which enables a voice application program to be generated. The output of the voice response system ordinarily destined for display on a visual display unit of a local terminal is directed to the voice response system server. The voice response system server forwards the data to a voice response system client. The voice response system client generates data in a first format useable by said WWW client terminal from data in a second format received from said voice response system and generates data in said second format useable by said voice response system from data in said first format received from WWW client terminal, that is, the voice response system client dynamically generates HTML data from the data generated by the voice response system for transmission to and subsequent display at the WWW client terminal and visa versa.

    摘要翻译: 使用互联网万维网(WWW)网络320,WWW客户端310可以与WWW服务器330进行通信,以请求重新配置或产生控制语音处理应用的软件。 语音响应系统客户端与语音响应系统服务器进行通信以改变语音响应系统的配置或控制所述语音应答系统上的软件的执行,从而能够生成语音应用程序。 通常用于在本地终端的可视显示单元上显示的语音应答系统的输出被引导到语音应答系统服务器。 语音应答系统服务器将数据转发到语音应答系统客户端。 语音应答系统客户端以所述语音响应系统接收的第二格式的数据生成所述WWW客户端终端能够使用的第一格式的数据,并以所述第一格式从所述第一格式的数据生成所述第二格式的数据, WWW客户终端,即语音响应系统客户机从语音响应系统产生的数据动态生成HTML数据,以便传送到WWW客户终端并随后在WWW客户终端显示,反之亦然。

    Voice processing system
    4.
    发明授权
    Voice processing system 失效
    语音处理系统

    公开(公告)号:US06335964B1

    公开(公告)日:2002-01-01

    申请号:US09069378

    申请日:1998-04-29

    IPC分类号: H04M1100

    CPC分类号: H04M3/493

    摘要: A voice processing system is connected to a switch via multiple telephone lines, and provides a set of line objects, each line object being associated with one of the physical telephone lines. The line object allows a demarcation to be made between the underlying voice processing system software, and external business applications. Thus a line object supports a set of methods such as Get DTMF Tone, Play Audio, Answer Call, and End Call, to allow the external business applications to perform desired operations on a telephone line. These methods are invoked via a set of corresponding IVR action objects, which in turn are integrated into the business application. The business application itself, and its IVR actions, regard the line objects effectively as servers to provide IVR functionality. The business application may therefore run partially or completely on a separate physical machine from the IVR system itself.

    摘要翻译: 语音处理系统经由多条电话线连接到交换机,并提供一组线路对象,每条线路对象与物理电话线路之一相关联。 线对象允许在底层语音处理系统软件和外部业务应用之间进行划分。 因此,线对象支持一组方法,例如获取DTMF音调,播放音频,应答呼叫和结束呼叫,以允许外部业务应用程序在电话线路上执行所需的操作。 这些方法通过一组相应的IVR动作对象进行调用,后者又被集成到业务应用程序中。 业务应用程序本身及其IVR操作将线对象有效地视为服务器来提供IVR功能。 因此,业务应用程序可以在独立的物理机器上部分或完全地从IVR系统本身运行。

    Method of and system for supporting voice over internet protocol (VoIP) media codecs on a voice processing system
    5.
    发明授权
    Method of and system for supporting voice over internet protocol (VoIP) media codecs on a voice processing system 有权
    在语音处理系统上支持语音互联网协议(VoIP)媒体编解码器的方法和系统

    公开(公告)号:US07529234B1

    公开(公告)日:2009-05-05

    申请号:US12131204

    申请日:2008-06-02

    IPC分类号: H04J3/16

    摘要: A method of supporting Voice over Internet Protocol (VOIP) media codecs on a voice processing system stores voice response segments in each of a plurality network native formats. Each of the network native formats corresponds to a media codec. The method receives a call from a caller over an IP network. The method determines a negotiated codec for the call. If the call requires a required voice response segment, the method retrieves the required voice segment in the network native format corresponding to the negotiated codec. The method then sends the required voice segment in the network native format corresponding to the negotiated codec to the caller over IP network. If the call includes a voice message from the caller, the method stores the voice message from the caller in the network native format corresponding to the negotiated codec. If the call requires retrieval of a voice mail message for the caller, the method retrieves the voice mail message. If the voice mail message is in the network native format corresponding to the negotiated codec, the method sends the voice mail message in the network native format corresponding to the negotiated codec to the caller over the IP network. If the voice mail message is not in the network native format corresponding to the negotiated codec, the method converts the voice mail message into the network native format corresponding to the negotiated codec, and sends the converted voice mail message in the network native format corresponding to the negotiated codec to the caller over the IP network.

    摘要翻译: 支持语音处理系统上的语音互联网协议(VOIP)媒体编解码器的方法以多种网络本机格式存储语音响应段。 每个网络本机格式对应于媒体编解码器。 该方法通过IP网络接收来自呼叫者的呼叫。 该方法确定用于该呼叫的协商编解码器。 如果呼叫需要所需的语音响应段,则该方法以对应于协商的编解码器的网络本机格式检索所需的语音段。 然后,该方法通过IP网络将与协商的编解码器对应的网络本地格式的所需语音段发送给呼叫者。 如果呼叫包括来自呼叫者的语音消息,则该方法以对应于协商的编解码器的网络本机格式存储来自呼叫者的语音消息。 如果呼叫需要检索呼叫者的语音邮件消息,则该方法检索语音邮件消息。 如果语音邮件消息是与协商的编解码器对应的网络本地格式,则该方法通过IP网络将与协商的编解码器对应的网络本地格式的语音邮件消息发送给呼叫者。 如果语音邮件消息不是与协商的编解码器对应的网络本机格式,则该方法将语音邮件消息转换为与协商的编解码器相对应的网络本机格式,并将转换的语音邮件消息以对应于 通过IP网络向呼叫者通过协商的编解码器。

    Voice mail on the internet
    6.
    发明授权
    Voice mail on the internet 失效
    互联网上的语音信箱

    公开(公告)号:US06282269B1

    公开(公告)日:2001-08-28

    申请号:US08763156

    申请日:1996-12-10

    IPC分类号: H04M164

    摘要: A first Internet telephone system 620 attempts to call with a second Internet telephone system 630 via the Internet 600. However, the second Internet telephone system 630 is not logged onto the Internet at the time of the call. In response to the failed attempt to call, the first Internet telephone system prompts the user to send voice mail to the user of the second Internet telephone system. This results in a phone call over the Internet between a voice mail system 610 and the first Internet telephone system, allowing a greeting to be heard, and a message to be stored. This message may be subsequently retrieved, either using an Internet telephone system over the Internet, or using a standard phone over the conventional telephone network.

    摘要翻译: 第一互联网电话系统620尝试经由互联网600与第二互联网电话系统630呼叫。然而,在通话时,第二互联网电话系统630未登录到因特网。 响应于呼叫失败的尝试,第一互联网电话系统提示用户向第二互联网电话系统的用户发送语音邮件。 这导致通过因特网在语音邮件系统610和第一互联网电话系统之间的电话呼叫,允许听到问候语和要被存储的消息。 可以随后通过因特网使用因特网电话系统或者通过常规电话网络使用标准电话来检索该消息。

    Internet telephony signal conversion
    7.
    发明授权
    Internet telephony signal conversion 失效
    互联网电话信号转换

    公开(公告)号:US06195358B1

    公开(公告)日:2001-02-27

    申请号:US08914111

    申请日:1997-08-19

    IPC分类号: H04L1266

    摘要: The disclosure concerns a method of processing Internet telephony messages at gateway computer such as an IBM RISC/6000 system, comprising the steps of: receiving an Internet telephony message in a first compression scheme from a computer running an Internet telephone software application which uses the first compression scheme; converting the Internet telephony message into a compression scheme optionally via an intermediate format; sending the Internet message telephony to another computer running another Internet telephone software application using the second format.

    摘要翻译: 本公开涉及在诸如IBM RISC / 6000系统的网关计算机处处理因特网电话消息的方法,包括以下步骤:从运行使用第一个的互联网电话软件应用的计算机接收来自第一压缩方案的因特网电话消息 压缩方案; 将所述因特网电话消息转换为可选地经由中间格式的压缩方案; 将所述因特网消息电话发送到使用所述第二格式运行另一互联网电话软件应用的另一计算机。

    Method and apparatus for multimodal voice and web services
    8.
    发明授权
    Method and apparatus for multimodal voice and web services 有权
    多模式语音和Web服务的方法和装置

    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。

    Voice recognition of telephone conversations
    9.
    发明授权
    Voice recognition of telephone conversations 失效
    语音识别电话对话

    公开(公告)号:US06278772B1

    公开(公告)日:2001-08-21

    申请号:US09107686

    申请日:1998-06-30

    IPC分类号: H04M164

    CPC分类号: H04M3/42221 H04M2201/60

    摘要: Voice data requires large storage resources even when compressed and takes a long time to retrieve. Further the required information cannot normally be directly located and it is difficult to analyze the voice data for statistical information. There is described a method for performing a voice recognition function on a voice telephone conversation to convert the conversation into text data using a voice processing system. The method comprises receiving voice data representing the telephone conversation comprising a first series of speech data from an agent interspersed with a second series of speech data from a client and storing the first and second series of speech data as a single body of voice data for later retrieval. Then a voice recognition function is performed on the voice data to convert it into text data representing the telephone conversation and the text data is stored for later retrieval. Such a solution allows entire days/weeks or even months of conversation to be stored and accessed. Since the memory space required is considerably smaller for text storage it is possible to keep many days of conversation in directly accessible memory that may be searched by a computer. Furthermore it is possible to search for keywords typed from the keyboard and it is not necessary to manually scan the entire conversation for the desired topic.

    摘要翻译: 语音数据即使在压缩并需要很长时间的时候也需要大量的存储资源。 此外,所需的信息通常不能直接定位,并且难以分析用于统计信息的语音数据。 描述了一种用于在语音电话会话上执行语音识别功能的方法,以使用语音处理系统将会话转换成文本数据。 该方法包括接收表示电话对话的语音数据,该语音数据包括来自客户端的散布有第二系列语音数据的代理的第一系列语音数据,并将第一和第二系列语音数据存储为单个语音数据主体以供以后使用 恢复。 然后对语音数据执行语音识别功能,将其转换为表示电话会话的文本数据,并存储文本数据以备以后检索。 这样的解决方案允许存储和访问整个几天或几个月甚至几个月的会话。 由于文本存储所需的存储空间相对较小,因此可以在可由计算机搜索的可直接访问的存储器中保持多天的会话。 此外,可以搜索从键盘输入的关键字,而不需要手动扫描整个会话以获得所需主题。

    Communication method and system
    10.
    发明授权
    Communication method and system 失效
    通信方式和系统

    公开(公告)号:US5970126A

    公开(公告)日:1999-10-19

    申请号:US868085

    申请日:1997-06-03

    摘要: The present invention relates to establishing a communication channel between two communication systems having computer telephony integration (CTI). Many CTI systems are configured into incoming and outgoing lines according to the anticipated demands. Consequently, if sufficient outgoing or incoming capacity is unavailable at a given time to support a communication channel of required characteristics, the users of such a system must conventionally wait until sufficient capacity becomes available. However, the present invention determines who the intended addressee is and can instruct the CTI system associated with that addressee to instigate the establishment of a communication channel to the user who originally desired the connection. The instructions can be sent to the intended addressee using for example, another communication network, such as the Internet or other data communications network. This arrangement also allows the call to be placed in the reverse direction, where this is favourable for tariff reasons. The invention also includes the automatic selection of a telephone network carrier based on tariff data requested over a data network such as the Internet.

    摘要翻译: 本发明涉及在具有计算机电话集成(CTI)的两个通信系统之间建立通信信道。 根据预期的需求,许多CTI系统被配置为进入和离开的线路。 因此,如果在给定时间内足够的输出或进入容量不可用以支持所需特性的通信信道,则这种系统的用户必须经常地等待直到足够的容量变得可用。 然而,本发明确定预期的接收者是谁,并且可以指示与该收件人相关联的CTI系统向最初希望该连接的用户启动建立通信信道。 可以使用例如另一个通信网络(例如因特网或其他数据通信网络)将指令发送到预期的收件人。 这种安排也可以将呼叫放在相反的方向,这样做有利于关税原因。 本发明还包括基于通过诸如因特网的数据网络请求的资费数据自动选择电话网络运营商。