公开(公告)号：US20010054085A1

公开(公告)日：2001-12-20

申请号：US09777406

申请日：2001-02-06

Inventor： Alexander Kurganov

IPC: G06F015/16

CPC classification number: H04M3/4938 ,  G06F3/167 ,  G06F17/30769 ,  G06F17/30861 ,  G10L13/08 ,  G10L15/02 ,  G10L15/06 ,  G10L15/08 ,  G10L15/22 ,  G10L15/26 ,  G10L17/24 ,  G10L25/54 ,  G10L2015/223 ,  H04L29/0809 ,  H04L67/02 ,  H04M2201/39 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2207/40

Abstract: The present invention relates to a system for retrieving information from a network such as the Internet. A user creates a user-defined record in a database that identifies an information source, such as a web site, containing information of interest to the user. This record identifies the location of the information source and also contains a recognition grammar based upon a speech command assigned by the user. Upon receiving the speech command from the user that is described within the recognition grammar, a network interface system accesses the information source and retrieves the information requested by the user.

Abstract translation: 本发明涉及一种用于从诸如因特网的网络检索信息的系统。 用户在数据库中创建用户定义的记录，该记录标识包含用户感兴趣的信息的信息源（例如网站）。 该记录识别信息源的位置，并且还包含基于用户分配的语音命令的识别语法。 在接收到在识别语法中描述的用户的语音命令之后，网络接口系统访问信息源并检索用户请求的信息。

公开(公告)号：US6167119A

公开(公告)日：2000-12-26

申请号：US006033

申请日：1998-01-12

Applicant: Dale L. Bartholomew ,  Robert D. Farris ,  Alexander I. McAllister ,  Michael J. Strauss

Inventor： Dale L. Bartholomew ,  Robert D. Farris ,  Alexander I. McAllister ,  Michael J. Strauss

IPC: H04M3/22 ,  H04M3/38 ,  H04M3/42 ,  H04M3/533 ,  H04M7/12 ,  H04Q3/00

CPC classification number: H04M3/38 ,  H04M3/2281 ,  H04M3/382 ,  H04M3/385 ,  H04M3/42 ,  H04M3/42017 ,  H04M3/42059 ,  H04M3/42068 ,  H04M3/42102 ,  H04M3/4211 ,  H04M3/42229 ,  H04M3/533 ,  H04M3/53383 ,  H04M7/06 ,  H04M7/12 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2207/12 ,  H04Q3/0029 ,  H04Q3/0045

Abstract: An intelligent telephone network provides personalized communication services based on subscriber prescribed double speech signal processing of utterances of both calling and answering parties on a subscriber line having multiple subscribers with a single directory number. Specifically, when one of the multiple subscribers has personalized voice mail service and a busy/no answer call is received, the network uses speech processing of an utterance of the calling party to identify a customer service profile of the called party in the terminating switch. This contains instructions inviting storage of a voice message left by the caller. Upon the subscriber going off-hook, the customer profile of the subscriber line is installed in the switch. The subscriber transmits an utterance and the personalized customer profile of the subscriber is identified by a virtual office equipment number and the profile installed in the switch. The subscriber may then retrieve the stored message.

Abstract translation: 智能电话网络基于在具有单个目录号码的多个订户的用户线上的呼叫和应答方的话语的用户规定的双语言信号处理来提供个性化通信服务。 具体地，当多个用户中的一个具有个性化语音邮件服务并且接收到忙/不应答呼叫时，网络使用主叫方的话语的语音处理来识别终止交换机中被叫方的客户服务简档。 这包含邀请​​存储呼叫者留下的语音消息的指令。 当用户摘机时，用户线路的客户资料安装在交换机中。 用户发送话语，并且用户的个性化客户简档由虚拟办公室设备号码和安装在交换机中的简档来标识。 订户然后可以检索存储的消息。

公开(公告)号：KR1020120046552A

公开(公告)日：2012-05-10

申请号：KR1020100108265

申请日：2010-11-02

Applicant: 엘지전자 주식회사

Inventor： 김태준

IPC: H04M1/725

CPC classification number: H04M1/72544 ,  H04M1/7255 ,  H04M1/72555 ,  H04M1/72572 ,  H04M2201/405 ,  H04M2201/50 ,  H04M2250/66 ,  H04M2250/68 ,  H04M2250/74

Abstract: PURPOSE: A portable terminal and operation method thereof are provided to display content which is able to interact with people who creates voice signals by using a preview image of a camera unit. CONSTITUTION: A second control unit deducts link information or registered content from a database in case a voice signal is stored(S730, S740). The second control uni transmits a voice response message to a portable terminal(S750). The second control unit registers the content within a content registration message or the link information in the database when the content registration message is transmitted to the portable terminal(S760, S770). The second control unit transmits content confirmation message to the portable terminal(S780).

Abstract translation: 目的：提供便携式终端及其操作方法，以便通过使用相机单元的预览图像来显示能够与创建语音信号的人交互的内容。 构成：在存储语音信号的情况下，第二控制单元从数据库中扣除链接信息或注册的内容（S730，S740）。 第二控制单元向便携式终端发送语音响应消息（S750）。 当将内容注册消息发送到便携式终端时，第二控制单元将内容登记在内容注册消息内或数据库中的链接信息（S760，S770）。 第二控制单元向便携式终端发送内容确认消息（S780）。

公开(公告)号：KR1020110096329A

公开(公告)日：2011-08-30

申请号：KR1020100015714

申请日：2010-02-22

Applicant: 이문섭

Inventor： 이문섭

IPC: H04M1/2745 ,  H04M3/44

CPC classification number: H04M1/274533 ,  H04M3/44 ,  H04M2201/40 ,  H04M2201/405 ,  H04M2250/74

Abstract: PURPOSE: A personal telephone number unit data base estimating method, method for serving automatic call connection using the same, system thereof using voice recognition are provided to automatically connecting a call with a desired person by using a voice. CONSTITUTION: A voice and receiving telephone number about a name of a caller is inputted through a call network from a service user which a personal telephone number unit estimation is requested. The caller telephone number of a service user is obtained. A speaker authentication voice characteristics and speaker-dependent voice recognition name characteristics is extracted at the same time from name of a caller by using a voice recognition(306). By using the obtained caller number as a key, the speaker-dependent voice recognition characteristics per the extracted speaker authentication voice characteristics and the inputted calling number are stored(308).

Abstract translation: 目的：提供一种个人电话号码单元数据库估计方法，用于使用其自动呼叫连接的方法，使用其语音识别的系统，以通过使用语音自动地将呼叫连接到期望的人。 构成：通过来自请求个人电话号码单元估计的服务用户的呼叫网络输入关于呼叫者姓名的语音和接收电话号码。 获得服务用户的主叫电话号码。 通过使用语音识别从呼叫者的姓名同时提取扬声器认证语音特性和与扬声器相关的语音识别名称特征（306）。 通过使用获得的呼叫者号码作为密钥，存储所提取的扬声器认证语音特性和输入的主叫号码的与扬声器相关的语音识别特性（308）。

公开(公告)号：KR1020080079662A

公开(公告)日：2008-09-01

申请号：KR1020087015918

申请日：2006-12-29

Applicant: 마이크로소프트 코포레이션

Inventor： 호웰,데이비드,에이. ,  선다라라만,스리드할 ,  퐁,데이비드,티. ,  세이드,프랭크

IPC: G06Q50/32 ,  H04M3/53

CPC classification number: H04M3/5307 ,  H04M2201/405 ,  H04M2201/60 ,  H04M2203/253 ,  H04M2203/4509 ,  G06Q50/32

Abstract: Improved systems and methods are provided for transcribing audio files of voice mails sent over a unified messaging system. Customized grammars specific to a voice mail recipient are created and utilized to transcribe a received voice mail by comparing the audio file to commonly utilized words, names, acronyms, and phrases used by the recipient. Key elements are identified from the resulting text transcription to aid the recipient in processing received voice mails based on the significant content contained in the voice mail.

Abstract translation: 提供了改进的系统和方法，用于转录通过统一消息系统发送的语音邮件的音频文件。 通过将音频文件与收件人使用的常用词，名称，首字母缩略词和短语进行比较，创建和利用特定于语音邮件收件人的定制语法来转录接收到的语音邮件。 从所得到的文本转录中识别关键要素，以帮助接收者基于语音邮件中包含的重要内容来处理接收到的语音邮件。

公开(公告)号：US20240363125A1

公开(公告)日：2024-10-31

申请号：US18646493

申请日：2024-04-25

Applicant: Pindrop Security, Inc.

Inventor： Elie KHOURY ,  Ganesh SIVARAMAN ,  Tianxiang CHEN ,  Nikolay GAUBITCH ,  David LOONEY ,  Amit GUPTA ,  Vijay BALASUBRAMANIYAN ,  Nicholas KLEIN ,  Anthony STANKUS

IPC: G10L17/26 ,  G10L17/02 ,  G10L17/04 ,  G10L25/60 ,  H04M3/22 ,  H04M3/51

CPC classification number: G10L17/26 ,  G10L17/02 ,  G10L17/04 ,  G10L25/60 ,  H04M3/2281 ,  H04M3/5183 ,  H04M2201/405

Abstract: Disclosed are systems and methods including software processes executed by a server that detect audio-based synthetic speech (“deepfakes”) in a call conversation. Embodiments include systems and methods for detecting fraudulent presentation attacks using multiple functional engines that implement various fraud-detection techniques, to produce calibrated scores and/or fused scores. A computer may, for example, evaluate the audio quality of speech signals within audio signals, where speech signals contain the speech portions having speaker utterances.

公开(公告)号：US20240064230A1

公开(公告)日：2024-02-22

申请号：US17890779

申请日：2022-08-18

Applicant: Mitel Networks Corporation

Inventor： Venkata Ajaresh Gudipati

IPC: H04M3/22 ,  H04M3/42 ,  G10L15/18 ,  G10L15/30

CPC classification number: H04M3/2281 ,  H04M3/42221 ,  G10L15/18 ,  G10L15/30 ,  H04M2201/42 ,  H04M2201/405 ,  H04M2242/04

Abstract: A communication system and method for identifying a threatening and/or illegal call includes (1) a telephony network, (2) a first user device in communication with the telephony network, wherein the first user device is assigned to a first user who is an organization member, (3) a second user device in communication with the telephony network, wherein the second user device is assigned to a second user who is not an organization member, (4) a communications server in communication with at least the first user device, (5) a natural language processor (NLP) in communication with the communications server; and (6) a threatening language database in communication with the communications server and/or the NLP. The NLP is configured to monitor language used by the second user during a communication with first user to identify threatening language by comparing the language used by the second user to the threatening language in the threatening language database.

公开(公告)号：US20230412734A1

公开(公告)日：2023-12-21

申请号：US17843731

申请日：2022-06-17

Applicant: Microsoft Technology Licensing, LLC

Inventor： Mrinal Kumar SHARMA

IPC: H04M3/56 ,  H04L65/80 ,  G06V40/16 ,  H04L65/403 ,  G10L25/57 ,  G10L15/02 ,  G10L15/22 ,  G10L15/06

CPC classification number: H04M3/563 ,  H04L65/80 ,  G06V40/168 ,  H04L65/403 ,  G10L25/57 ,  G10L15/02 ,  G10L15/22 ,  G10L15/063 ,  H04M2201/405 ,  G10L2015/025

Abstract: Methods, systems, and computer storage media for providing a disrupted-speech assistance service associated with a disrupted-speech management engine of a meeting management system. The disrupted-speech assistance service is an accessibility service that supports accessibility operations of a disrupted-speech management engine to provide disrupted-speech assistance features in a meeting management system. In operation, meeting data comprising audio data is accessed. The audio data is analyzed to determine that the audio data comprises disrupted-speech at a threshold level of disrupted-speech. Based on the audio data comprising disrupted-speech at the threshold level of disrupted-speech, one or more disrupted-speech assistance operations for a meeting can be executed. The one or more disrupted-speech assistance operations comprises identifying a disrupted-speech word; determining an alternative word for the disrupted-speech word. A disrupted-speech assistance interface is generated based on the one or more disrupted-speech assistance operations. The disrupted-speech assistance interface comprises the alternative word for the disrupted-speech word.

公开(公告)号：US20230208976A1

公开(公告)日：2023-06-29

申请号：US18175253

申请日：2023-02-27

Applicant: NICE LTD.

Inventor： Roman FRENKEL ,  Tal RASKIN ,  Adi BEN ZEEV ,  Stav MISHORY ,  Dan TEPLITSKI ,  Hadas KATZ

IPC: H04M3/523 ,  H04M3/51 ,  G06F40/295 ,  G10L15/26 ,  G10L15/02

CPC classification number: H04M3/5235 ,  H04M3/5166 ,  H04M3/5183 ,  G06F40/295 ,  G10L15/26 ,  G10L15/02 ,  H04M3/5116 ,  H04M2201/405

Abstract: Systems for and methods of determining the priority of a call interaction include receiving a call interaction from a call center; validating, by a validation and transcription engine, that the call interaction is authentic; converting, by the validation and transcription engine, the call interaction into text; calculating, by the data calculation engine, a priority of the call interaction from the text and organization, location, and time information in the text by determining an important of words in the text and correlating the words to a priority class using a pre-trained algorithm that is trained on emergency-type and emergency services-type language; determining that the call interaction should be transmitted to the call center for initial handling by a call center agent; and transmitting the call interaction, the calculated priority, and the extracted information to the call center.

公开(公告)号：US20190141190A1

公开(公告)日：2019-05-09

申请号：US15802695

申请日：2017-11-03

Applicant: SONY CORPORATION

Inventor： RASHMI GUPTA ,  AVNEET SINGH

IPC: H04M3/493 ,  G10L17/14

CPC classification number: H04M3/493 ,  A61B5/024 ,  A61B5/1112 ,  A61B5/6802 ,  G10L15/265 ,  G10L17/14 ,  H04M3/4936 ,  H04M2201/405 ,  H04M2203/6054

Abstract: An electronic call assistant based on a callee-status and a caller-status, comprises one or more sensors, a memory, and a circuitry. The memory in the first electronic device may be configured to store a plurality of applications. The circuitry in the first electronic device is configured to determine an identity of a caller of a voice call received by the first electronic device from a second electronic device. A current callee-status of a callee associated with the first electronic device, is detected. The circuitry is configured to extract a text query from a speech signal in the received voice call. The circuitry is further configured to communicate a custom audio response to the second electronic device based on the extracted text query, the determined identity of the caller, current callee-status of the callee, and the level of access associated with the determined identity.