公开(公告)号：US20080273673A1

公开(公告)日：2008-11-06

申请号：US11742660

申请日：2007-05-01

Applicant: Magnus OTTOSSON ,  Mats ORMIN ,  Hanshenric CARENBORN ,  Russel HUGHES ,  Robin PETERSSON

Inventor： Magnus OTTOSSON ,  Mats ORMIN ,  Hanshenric CARENBORN ,  Russel HUGHES ,  Robin PETERSSON

IPC: H04M1/64

CPC classification number: H04M3/436 ,  G10L15/26 ,  H04M1/6505 ,  H04M1/665 ,  H04M3/42034 ,  H04M3/42059 ,  H04M3/42093 ,  H04M2201/41 ,  H04M2203/2005 ,  H04M2242/22

Abstract: A method of handling an incoming call from a first communication device addressed to a second communication device, the first and second communication devices being operative in a communication network, wherein the second communication device or the communication network comprises an answering unit. The answering unit includes a privacy data unit comprising a privacy code message for bypassing the answering unit. The method comprises receiving, by means of the answering unit, the incoming call intended for and addressed to the second communication device. Furthermore, the method comprises determining whether also a code message is received by the answering unit. If so, the method comprises comparing the received code message with the privacy code message stored in the privacy data unit to determine whether the incoming call should bypass the answering unit and, thereby, be directly forwarded to the second communication device being addressed by the first communication device in the incoming call. Thereby, a user of the second communication device may be permitted to intercept the incoming call from the calling first communication device.

Abstract translation: 一种处理来自寻址到第二通信设备的第一通信设备的呼入的方法，所述第一和第二通信设备在通信网络中可操作，其中所述第二通信设备或所述通信网络包括应答单元。 应答单元包括隐私数据单元，其包括用于绕过应答单元的隐私代码消息。 该方法包括通过应答单元接收旨在和寻址到第二通信设备的呼入。 此外，该方法包括确定应答单元是否接收到代码消息。 如果是，则该方法包括将接收到的代码消息与存储在隐私数据单元中的隐私代码消息进行比较，以确定来话呼叫是否应该绕过应答单元，从而直接转发到由第一通信设备寻址的第二通信设备 通话设备在来电。 从而，可以允许第二通信设备的用户拦截来自呼叫第一通信设备的呼入。

公开(公告)号：US20080192901A1

公开(公告)日：2008-08-14

申请号：US11780120

申请日：2007-07-19

Applicant: Marc Mumm ,  Raja Kuppuswamy

Inventor： Marc Mumm ,  Raja Kuppuswamy

IPC: H04M11/00

CPC classification number: H04L63/0861 ,  G10L17/22 ,  H04M3/38 ,  H04M3/385 ,  H04M2201/41 ,  H04W12/06 ,  H04W12/08

Abstract: Digital process for authentication of a user of a telecommunications or data network for access to protected data or a service reserved for a defined circle of users or for the use of data currently entered by the user, wherein a voice sample currently enunciated during an access attempt by the user on a telecommunications or VoIP terminal device is routed to a voice analysis unit and, herein, a current voice profile is computed and this is compared in a voice profile comparison unit against a previously stored initial voice profile and, in response to a positive comparison result, the user is authenticated and a first control signal enabling access, but in response to a negative comparison result a second control signal disabling access or triggering a substitute authentication procedure is generated.

Abstract translation: 用于认证用于访问受保护数据的电信或数据网络的数字处理或为定义的用户圈保留的服务或用于使用当前由用户输入的数据的数字处理，其中当前在接入尝试期间被发送的语音样本 由电信或VoIP终端设备上的用户被路由到语音分析单元，并且在此，计算当前语音配置文件，并且在语音配置文件比较单元中与先前存储的初始语音配置文件进行比较，并且响应于 肯定的比较结果，用户被认证，并且第一控制信号能够访问，但是响应于否定的比较结果，产生禁用访问或触发替代认证过程的第二控制信号。

公开(公告)号：US20070263846A1

公开(公告)日：2007-11-15

申请号：US11396789

申请日：2006-04-03

Applicant: Roger Fratti

Inventor： Roger Fratti

IPC: H04M1/00 ,  H04M9/00

CPC classification number: H04M3/56 ,  H04M3/568 ,  H04M2201/41

Abstract: The audio signals associated with different co-located groups of talkers in a teleconference are detected (e.g., by comparing the voiceprint for the current talker group with stored voiceprints corresponding to all of the co-located teleconference participants) and processed using different and appropriate automatic gain control (AGC) levels, where each group has a corresponding stored AGC level. Depending on the embodiment, each group may have one or more participants.

Abstract translation: 检测到与电话会议中的不同同位置的通话者组相关联的音频信号（例如，通过将当前讲话者组的声波图与对应于所有同位的电话会议参与者的已存储的声纹相比较）并使用不同且适当的自动 增益控制（AGC）电平，其中每个组具有相应的存储的AGC电平。 根据实施例，每个组可以具有一个或多个参与者。

公开(公告)号：US20070223677A1

公开(公告)日：2007-09-27

申请号：US11727135

申请日：2007-03-23

Applicant: Yoshihiro Ono

Inventor： Yoshihiro Ono

IPC: H04M3/42

CPC classification number: H04M3/569 ,  H04M3/4872 ,  H04M2201/41 ,  H04M2203/352 ,  H04M2203/6045

Abstract: The present invention provides a multi-party communication system in which the speaking party can be identified aurally and the speech contents can be accurately transmitted to the party at the receiving terminal. A multi-party communication server and a plurality of terminal devices with a communication function make up the multi-party communication system. Each terminal device with the communication function includes a speech right management unit, a speaking party name output unit and a buffer unit. The speaking party name output unit outputs the voice data of the speaking party identification information such as the name of the speaking party. The buffer unit accumulates the speech voice of the user as voice data. The speech right management unit controls the buffer unit to produce an output after the speaking party output unit. The speech right management unit issues a request to cancel the right to speak after completion of the output of the speech voice data.

Abstract translation: 本发明提供一种多方通信系统，其中可以听觉地识别发言方，并且可以将语音内容准确地发送到接收终端处的一方。 具有通信功能的多方通信服务器和多个终端装置构成多方通信系统。 具有通信功能的每个终端设备包括语音权限管理单元，会话人姓名输出单元和缓冲单元。 讲话人姓名输出单元输出会话者身份信息的语音数据，例如讲话人的姓名。 缓冲单元将用户的语音语音累积为语音数据。 语音权限管理单元控制缓冲单元以在会话输出单元之后产生输出。 言语权限管理单元在完成语音语音数据的输出之后发出取消发言权的请求。

公开(公告)号：US07246061B2

公开(公告)日：2007-07-17

申请号：US10181153

申请日：2000-12-15

Applicant: Fred Runge ,  Christel Mueller ,  Marian Trinkel ,  Thomas Ziem

Inventor： Fred Runge ,  Christel Mueller ,  Marian Trinkel ,  Thomas Ziem

IPC: G10L21/00 ,  H04M1/65

CPC classification number: G10L17/04 ,  G10L17/24 ,  H04M3/42059 ,  H04M2201/41

Abstract: A method for the voice-operated identification of the user of a telecommunications line in a telecommunications network is provided in the course of a dialog with a voice-operated dialog system. Utterances spoken by a caller from a group of callers limited to one telecommunications line are used during a human-to-human and/or human-to-machine dialog to apply a reference pattern for the caller. For each reference pattern, a user identifier is stored which is activated once the caller is identified, and, together with the CLI and/or ANI identifier of the telecommunications line, are made available to a server having a voice-controlled dialog system. On the basis of the CLI, including the user identifier, data previously stored for this user are ascertained by the system and made available for the dialog interface with the customer.

Abstract translation: 在与语音操作的对话系统对话的过程中提供用于在电信网络中的电信线路的用户的语音操作识别的方法。 来自限于一条电信线路的呼叫者的呼叫者所说的话语在人与人和/或人对机对话中被使用以对呼叫者应用参考模式。 对于每个参考模式，存储一个用户标识符，一旦识别出呼叫者，该用户识别符被激活，并且与电话线的CLI和/或ANI标识符一起使得具有语音控制对话系统的服务器可用。 在CLI的基础上，包括用户标识符，系统确定了先前为此用户存储的数据，并为客户提供对话界面。

公开(公告)号：US20070036289A1

公开(公告)日：2007-02-15

申请号：US11424627

申请日：2006-06-16

Applicant: Guo Fu ,  Feng Shi ,  Zhi Wang ,  Yu Zhou

Inventor： Guo Fu ,  Feng Shi ,  Zhi Wang ,  Yu Zhou

IPC: H04M1/64

CPC classification number: H04L63/0861 ,  G07C9/00158 ,  G10L17/00 ,  H04M3/385 ,  H04M2201/41 ,  H04M2203/553 ,  H04M2203/554

Abstract: A voice authentication system using a removable voice ID card comprises: at server side, a voiceprint database for storing the voiceprints of all authorized users; a voiceprint updating means for updating the voiceprints in said voiceprint database; and a voiceprint digest generator for generating a voiceprint digest according to a request from a client; at client side, a voice ID card for storing the voiceprint of an authorized user; a validation means for validating the voiceprint in the voice ID card on the basis of the voiceprint digest from the server; an audio device for performing voice interaction with a user; and a voice authentication means for determining whether the voiceprint from said voice ID card is of the same speaker as the voice from said audio device. The present invention can significantly avoid the abuse of a voice ID card when it is lost or stolen by using the voiceprint digest stored at server side to verify the voiceprint in the voice ID card.

Abstract translation: 一种使用可移动语音识别卡的语音认证系统包括：在服务器侧，用于存储所有授权用户的声纹的声纹数据库; 声纹更新装置，用于更新所述声纹数据库中的声纹; 以及声纹摘要生成器，用于根据客户端的请求生成声纹摘要; 在客户侧，存储用于存储授权用户的声纹的语音ID卡; 验证装置，用于基于来自服务器的声纹摘要来验证语音ID卡中的声纹; 用于与用户进行语音交互的音频设备; 以及语音认证装置，用于确定来自所述语音ID卡的声纹是否与来自所述音频设备的语音具有相同的扬声器。 本发明可以通过使用存储在服务器端的声纹摘要丢失或被盗，来大大地避免语音ID卡的滥用来验证语音ID卡中的声纹。

公开(公告)号：US20060140356A1

公开(公告)日：2006-06-29

申请号：US11326307

申请日：2006-01-05

Applicant: Robert Bossemeyer ,  Raymond Bennett ,  Donald Liebrecht ,  Barry Sullivan

Inventor： Robert Bossemeyer ,  Raymond Bennett ,  Donald Liebrecht ,  Barry Sullivan

IPC: H04M1/64 ,  H04M15/06

CPC classification number: H04L12/2818 ,  H04M1/57 ,  H04M1/578 ,  H04M1/642 ,  H04M1/663 ,  H04M3/42059 ,  H04M3/4288 ,  H04M3/533 ,  H04M15/06 ,  H04M2201/39 ,  H04M2201/41 ,  H04W84/14

Abstract: A home gateway system has a transceiver (70) capable of establishing a wireless local loop connection (72). A voice processing system (74) is coupled to the transceiver (70). The voice processing system (74) is capable of storing a message from an incoming call. A caller identification processing system (76) is coupled to the transceiver (70). The caller identification processing system (76) determines a telephone number of the incoming call and routes the incoming call to the voice processing system (74), if the telephone number belongs to a screened group of telephone numbers.

Abstract translation: 家庭网关系统具有能够建立无线本地环路连接（72）的收发机（70）。 语音处理系统（74）耦合到收发器（70）。 语音处理系统（74）能够存储来自呼叫的消息。 呼叫者识别处理系统（76）耦合到收发信机（70）。 呼叫者识别处理系统（76）确定来电的电话号码，并将呼入路由到语音处理系统（74），如果电话号码属于被屏蔽的一组电话号码。

公开(公告)号：US07065198B2

公开(公告)日：2006-06-20

申请号：US10279674

申请日：2002-10-23

Applicant: Michael Wayne Brown ,  Joseph Herbert McIntyre ,  Victor S. Moore ,  Michael A. Paolini ,  Scott Lee Winters

Inventor： Michael Wayne Brown ,  Joseph Herbert McIntyre ,  Victor S. Moore ,  Michael A. Paolini ,  Scott Lee Winters

IPC: H04M3/42

CPC classification number: H04M3/42221 ,  H04M3/42042 ,  H04M3/42059 ,  H04M3/56 ,  H04M2201/40 ,  H04M2201/41 ,  H04M2242/22

Abstract: A system and method for recording a telephone conference and replaying a portion of the recording during the conference. Users participate by connecting through different types of networks using a device having a communication line connection. The recording can be in audio format, text format, or both. Thus, users can recall and replay textual information in addition to the recorded audio. Other information-such as time and user data-may also be recorded along with the audio and text. Users in the conference are identified to enable the association with them each user's contribution to the conference. The user or the user's device can assist by providing identification information. User identification may also be accomplished by associating each user's contribution with the particular line the user is calling from. Caller ID information may also be used to identify the user. Voice analysis may also performed to accomplish user identification.

Abstract translation: 一种用于在会议期间记录电话会议并重播记录的一部分的系统和方法。 用户通过使用具有通信线路连接的设备通过不同类型的网络进行连接。 录音可以是音频格式，文本格式或两者兼容。 因此，除了记录的音频之外，用户还可以调用和重播文本信息。 其他信息（例如时间和用户数据）也可以与音频和文本一起被记录。 会议中的用户被确定为能够与每个用户对会议的贡献相关联。 用户或用户的设备可以通过提供识别信息来辅助。 用户识别也可以通过将每个用户的贡献与用户所呼叫的特定线相关联来实现。 来电显示信息也可用于识别用户。 也可执行语音分析以完成用户识别。

公开(公告)号：US20050216258A1

公开(公告)日：2005-09-29

申请号：US10510955

申请日：2004-02-06

Applicant: Kazunori Kobayashi ,  Kenichi Furuya

Inventor： Kazunori Kobayashi ,  Kenichi Furuya

IPC: G10L11/02 ,  G10L21/02 ,  H04R3/00 ,  G10L19/14

CPC classification number: H04M3/56 ,  G10L21/0364 ,  G10L25/78 ,  G10L2021/02166 ,  H04M2201/41 ,  H04R3/005

Abstract: Upon detecting an utterance period by a state decision part 14, a sound source position detecting part 15 detects the positions of sound sources 91 to 9K are detected by a sound source position detecting part 15, then covariance matrix of acquired signals are calculated by a covariance matrix calculating part 18 in correspondence to the respective sound sources, and stored in a covariance matrix storage part 18 in correspondence to the respective sound sources. The acquired sound level for each sound source is estimated by an acquired sound level estimating part 19 from the stored covariance matrix, and filter coefficients are determined by a filter coefficient calculating part 21 from the estimated acquired sound levels and the covariance matrices, and the filter coefficients are set in filters 121 to 12M. Acquired signals from the respective microphones are filtered by the filters, then the filtered outputs are added together by an adder 13, and the added output is provided as a send signal; by this, it is possible to generate send signals of desired levels irrespective of the positions of sound sources.

Abstract translation: 在通过状态判定部14检测出发声周期的情况下，声源位置检测部15检测声源9〜9的位置，由声源检测 位置检测部15，则通过与各声源对应的协方差矩阵运算部18计算所获取信号的协方差矩阵，并存储在与各声源对应的协方差矩阵存储部18中。 通过获取的声级估计部分19从存储的协方差矩阵中估计每个声源的获取的声级，并且滤波器系数由估计的获取的声级和协方差矩阵由滤波器系数计算部分21确定，滤波器 系数被设置在滤波器12 1到12 M中。 来自相应麦克风的获取信号由滤波器滤波，然后滤波后的输出由加法器13相加在一起，并将相加的输出作为发送信号提供; 由此，无论声源的位置如何，都可以产生所需水平的发送信号。

公开(公告)号：US20050207554A1

公开(公告)日：2005-09-22

申请号：US11129253

申请日：2005-05-13

Applicant: William Ortel

Inventor： William Ortel

IPC: H04M1/64 ,  H04M3/42 ,  H04M3/56

CPC classification number: H04M3/56 ,  H04M3/567 ,  H04M2201/38 ,  H04M2201/41

Abstract: There is provided a system for facilitating a conference call. The system includes a module to generate a real-time voiceprint from a voice input of a participant in the conference call, and a module to provide information indicative of the participant based on the real-time voiceprint.