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1. 发明授权

US08918431B2 Adaptive ontology 有权
标题翻译：自适应本体论
公开(公告)号：US08918431B2
公开(公告)日：2014-12-23
申请号：US13229591
申请日：2011-09-09
申请人： William Scott Mark , Thierry Donneau-Golencer , Madhu Yarlagadda
发明人： William Scott Mark , Thierry Donneau-Golencer , Madhu Yarlagadda
IPC分类号： G06F17/30
CPC分类号： G06F17/30734
摘要： A computing system has a user interface allowing a user to view and input data related to concepts in a knowledge area associated with the user, an observation sub-system to centralize data and to identify a plurality of knowledge concepts, a conceptualization sub-system to generate a plurality of nodes within an ontological mapping, wherein each of the nodes corresponds to a certain one of the plurality of identified knowledge concepts identified by the observation sub-system, a relationship identification sub-system to create relationships between at least some of the plurality of identified knowledge concepts, and attribute affinity weights to the relationships, a change refinement sub-system to modify at least one of the plurality of nodes, affinity weights and relationships based upon information associated with the user, and a non-transitory knowledge store to store the information associated with the user pertaining to a sub-plurality of the plurality of identified knowledge concepts.
摘要翻译：计算系统具有允许用户查看和输入与与用户相关联的知识区域中的概念的数据的用户界面，用于集中数据并识别多个知识概念的观察子系统，以及识别多个知识概念，概念化子系统在本体映射中生成多个节点，其中每个节点对应于由观察子系统识别的多个识别的知识概念中的某一个，关系识别子系统以在至少一些多个识别的知识概念和对关系的属性关联权重，改变细化子系统以修改多个节点中的至少一个节点，基于与用户相关联的信息的关联权重和关系，以及非暂时性知识存储存储与所述用户有关的与所述多个识别知识con中的多个子集有关的信息小心点

2. 发明授权

US07821953B2 Dynamically selecting CODECS for managing an audio message 有权
标题翻译：动态选择用于管理音频消息的CODECS
公开(公告)号：US07821953B2
公开(公告)日：2010-10-26
申请号：US11129666
申请日：2005-05-13
申请人： Madhu Yarlagadda , Ramkumar Ramani
发明人： Madhu Yarlagadda , Ramkumar Ramani
IPC分类号： H04L12/26 , H04L1/00
CPC分类号： H04L29/06027 , H04L51/04 , H04L65/80 , H04M7/006
摘要： A system, method, and apparatus are directed towards a system, method, and apparatus for managing a communication session by dynamically selecting a CODEC. A client device requests a communication session with a receiver device. If available, historical information may be used to establish an initial CODEC and its associated sampling frequency for the communication session. Signals, such as a white noise signal, and/or a ring tone may be sent between the client device and the receiver to determine a metric for the communication session. The metric may be used to adjust the CODEC and/or its sampling frequency for the communication session. In one embodiment, if it is determined that the sampling frequency is less than a minimum determined value, a message may be sent to the client device advising that the current communication session be terminated.
摘要翻译：系统，方法和装置针对通过动态选择CODEC来管理通信会话的系统，方法和装置。客户机设备请求与接收机设备进行通信会话。如果可用，可以使用历史信息来建立通信会话的初始CODEC及其相关联的采样频率。可以在客户端设备和接收机之间发送诸如白噪声信号和/或铃声的信号，以确定通信会话的度量。该度量可用于调整通信会话的CODEC和/或其采样频率。在一个实施例中，如果确定采样频率小于最小确定值，则可以向客户端设备发送消息以通知当前通信会话终止。

3. 发明授权

US07313134B2 Proxy server for relaying VOIP messages 有权
标题翻译：用于中继VOIP消息的代理服务器
公开(公告)号：US07313134B2
公开(公告)日：2007-12-25
申请号：US11128634
申请日：2005-05-12
申请人： Madhu Yarlagadda
发明人： Madhu Yarlagadda
IPC分类号： H04L12/66
CPC分类号： H04L29/06027 , H04L65/1026 , H04L65/1036
摘要： A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) message over a network, where the VOIP message may employ the Real-time Transport Protocol (RTP) and possibly Session Initiation Protocol (SIP) over the User Datagram Protocol (UDP). The invention enables a VOIP client device, such as an IP phone, and the like, to communicate a message to a local proxy residing on a local computing device. The communications between the VOIP device and the local proxy may employ SIP/RTP over UDP. Upon receipt of the communications, the local proxy converts the transport protocol to another transport protocol, such as Transmission Control Protocol (TCP). The local proxy may also perform a port translation on the message. The converted communications may then be sent to a remote server, where it may be employed in its present SIP/RTP over TCP format, or be converted back to using UDP.
摘要翻译：系统，方法和装置旨在通过网络管理IP语音（VOIP）消息，其中VOIP消息可以使用用户数据报中的实时传输协议（RTP）和可能的会话发起协议（SIP）协议（UDP）。本发明使诸如IP电话之类的VOIP客户端设备能够将消息传送到驻留在本地计算设备上的本地代理。 VOIP设备和本地代理之间的通信可以通过UDP采用SIP / RTP。在接收到通信时，本地代理将传输协议转换为另一传输协议，例如传输控制协议（TCP）。本地代理还可以对消息进行端口转换。转换的通信然后可以被发送到远程服务器，其中它可以以其当前的SIP / RTP over TCP格式采用，或者被转换回使用UDP。

4. 发明授权

US07095733B1 Voice integrated VOIP system 有权
标题翻译：语音集成VOIP系统
公开(公告)号：US07095733B1
公开(公告)日：2006-08-22
申请号：US09658771
申请日：2000-09-11
申请人： Madhu Yarlagadda , Patrick Loo , David H. Nakayama
发明人： Madhu Yarlagadda , Patrick Loo , David H. Nakayama
IPC分类号： H04L12/66
CPC分类号： H04M3/493 , H04L29/06 , H04L29/06027 , H04L65/103 , H04L65/104 , H04L65/1043 , H04L65/80 , H04M3/2272 , H04M3/533 , H04M7/12 , H04M7/1225 , H04M2201/40 , H04M2201/60
摘要： An integrated VoIP unified message processing system includes a voice platform that processes data in native VoIP format. There is no use of hardware telephone interface cards (TICs) or software transcoding to transform data to PCM or other formats. Cost reductions are achieved by the elimination of expensive dedicated hardware and scalability is achieved by obviating the need for software transcoding.
摘要翻译：集成的VoIP统一消息处理系统包括处理本机VoIP格式的数据的语音平台。没有使用硬件电话接口卡（TIC）或软件代码转换来将数据转换为PCM或其他格式。通过消除昂贵的专用硬件来实现成本降低，并且通过消除对软件代码转换的需要来实现可扩展性。

5. 发明申请

US20060023695A1 Predictive communication routing scheme using instant messaging preferences and/or settings for routing packet traffic for voice, video and other multimedia communications 有权
标题翻译：使用即时消息收发偏好和/或用于路由用于语音，视频和其他多媒体通信的分组业务的设置的预测通信路由方案
公开(公告)号：US20060023695A1
公开(公告)日：2006-02-02
申请号：US11056485
申请日：2005-02-10
申请人： Madhu Yarlagadda , Warren Wan
发明人： Madhu Yarlagadda , Warren Wan
IPC分类号： H04L12/66
CPC分类号： H04M3/42263 , H04L29/06027 , H04L51/04 , H04L51/066 , H04L51/14 , H04L65/1069 , H04L2012/6443 , H04M3/2218 , H04M3/42374 , H04M3/4931 , H04M7/0024 , H04M7/0045 , H04M7/1205 , H04M15/41 , H04M15/43 , H04M2203/4536 , H04M2203/651 , H04M2215/0164 , H04M2215/2026 , H04M2215/32 , H04Q3/0045 , H04Q3/76 , H04Q2213/13034 , H04Q2213/1307 , H04Q2213/13097 , H04Q2213/13103 , H04Q2213/13141 , H04Q2213/1315 , H04Q2213/13173 , H04Q2213/1318 , H04Q2213/13282 , H04Q2213/13335 , H04Q2213/13336 , H04Q2213/13389 , H04W4/12 , H04W4/24 , H04W8/18 , H04W8/26 , H04W40/02 , H04W40/18 , H04W80/00
摘要： Techniques for using instant messaging preferences and settings are used for predictive communications, including routing of data traffic, are provided. A voice call can be routed over a packet-based network if a voice network is not available, or for other reasons. In addition to voice calls, embodiments of the invention can route video calls, multimedia communications and other communications. In some implementations, an instant message presence can be queried for in determining how to route a voice-over-IP or PSTN telephone call. Thus, a call might be routed to a computer or computing device unrelated to the location of the PSTN telephone line being called, if an associated instant messenger presence is detected on that computer or computing device.
摘要翻译：提供了使用即时通讯偏好和设置的技术用于预测通信，包括数据流量的路由。如果语音网络不可用或由于其他原因，语音呼叫可以通过基于分组的网络路由。除了语音呼叫之外，本发明的实施例可以路由视频呼叫，多媒体通信和其他通信。在一些实现中，可以查询即时消息存在以确定如何路由IP或PSTN电话呼叫。因此，如果在该计算机或计算设备上检测到相关联的即时消息存在，则呼叫可以被路由到与所呼叫的PSTN电话线路的位置无关的计算机或计算设备。

6. 发明授权

US07634072B2 Integrated instant messaging, routing and telephone services billing system 有权
标题翻译：集成即时通讯，路由和电话业务计费系统
公开(公告)号：US07634072B2
公开(公告)日：2009-12-15
申请号：US11056212
申请日：2005-02-10
申请人： Madhu Yarlagadda , Warren Wan
发明人： Madhu Yarlagadda , Warren Wan
IPC分类号： H04M1/56 , H04M15/06 , H04M1/64
CPC分类号： H04M3/42263 , H04L29/06027 , H04L51/04 , H04L51/066 , H04L51/14 , H04L65/1069 , H04L2012/6443 , H04M3/2218 , H04M3/42374 , H04M3/4931 , H04M7/0024 , H04M7/0045 , H04M7/1205 , H04M15/41 , H04M15/43 , H04M2203/4536 , H04M2203/651 , H04M2215/0164 , H04M2215/2026 , H04M2215/32 , H04Q3/0045 , H04Q3/76 , H04Q2213/13034 , H04Q2213/1307 , H04Q2213/13097 , H04Q2213/13103 , H04Q2213/13141 , H04Q2213/1315 , H04Q2213/13173 , H04Q2213/1318 , H04Q2213/13282 , H04Q2213/13335 , H04Q2213/13336 , H04Q2213/13389 , H04W4/12 , H04W4/24 , H04W8/18 , H04W8/26 , H04W40/02 , H04W40/18 , H04W80/00
摘要： Techniques for using billing services to manage a personal address book are provided. The techniques comprise receiving a call detail record (CDR) for a plain old telephone system (POTS) or a voice-over-IP (VOIP) call by a user. The CDR is created by a telecommunication carrier that routed the call. The CDR may be stored on a telecommunications server associated with the telecommunications carrier. Address information for the call is determined from the CDR. This address information is added to the personal address book for the user. This address information may be available for the user when using an IM client.
摘要翻译：提供了使用计费服务来管理个人通讯录的技术。这些技术包括接收用户的普通老式电话系统（POTS）或IP语音（VOIP）呼叫的呼叫详细记录（CDR）。 CDR由通过呼叫路由的电信运营商创建。 CDR可以存储在与电信运营商相关联的电信服务器上。呼叫的地址信息由CDR确定。该地址信息被添加到用户的个人通讯录中。使用IM客户端时，该地址信息可能对用户可用。

7. 发明申请

US20060268828A1 Selecting a network based on metrics for real time communication 有权
标题翻译：基于实时通信指标选择网络
公开(公告)号：US20060268828A1
公开(公告)日：2006-11-30
申请号：US11128519
申请日：2005-05-12
申请人： Madhu Yarlagadda
发明人： Madhu Yarlagadda
IPC分类号： H04L12/66
CPC分类号： H04M3/42289 , H04L43/0829 , H04L45/123 , H04M7/006
摘要： A system, method, and apparatus are directed towards routing a Voice over IP (VOIP) messages over a network. The VOIP messages are sent by a source client device to a destination client device through a portal service that has access to multiple routing services such as peering partners, carriers, etc. A VOIP system of the portal service aggregates call quality data after each VOIP call over each routing service. The call quality data is analyzed to determine a perception factor for each routing service at various times of day, days of the week, day of the year, geographic areas, and the like. When a VOIP call is requested through the portal service, the VOIP system determines a current cost, a current quality of service, and the perception factor for each routing service. A weighting is applied to each criterion and a routing service is selected for routing the VOIP call.
摘要翻译：系统，方法和装置被引导通过网络路由IP语音（VOIP）消息。 VOIP消息由源客户端设备通过门户服务发送到目的客户端设备，门户服务可以访问多个路由服务，如对等伙伴，运营商等。门户服务的VOIP系统在每个VOIP呼叫之后聚合呼叫质量数据通过每个路由服务。对通话质量数据进行分析，以确定每天路线服务的感知因素，一天中的几天，一天中的某一天，地理区域等。当通过门户服务请求VOIP呼叫时，VOIP系统确定当前成本，当前服务质量以及每个路由服务的感知因子。对每个标准应用加权，并且选择路由服务来路由VOIP呼叫。

8. 发明申请

US20060256776A1 Statistical approach to automatic gain control for managing audio messages over a network 有权
标题翻译：用于通过网络管理音频消息的自动增益控制的统计方法
公开(公告)号：US20060256776A1
公开(公告)日：2006-11-16
申请号：US11131459
申请日：2005-05-16
申请人： Eugene Gladyshev , Ramkumar Ramani , Madhu Yarlagadda , Erik Reed
发明人： Eugene Gladyshev , Ramkumar Ramani , Madhu Yarlagadda , Erik Reed
IPC分类号： H04L12/66
CPC分类号： H04L29/06027 , H04L65/605
摘要： A system, method, and apparatus are directed towards managing an audio message, such as a Voice over Internet Protocol (VOIP) message over a network. The invention employs a statistical mechanism to automatically optimize a gain control for setting a volume of an audio message being sent by a client device. An initial gain value is automatically adjusted based, in part, on a statistical sampling of energy levels in the audio message. Environmental factors, such as a sound card within the client device, background noise, and the like, may also be considered through a setting of a servo coefficient that may be used to map between volume levels and decibel levels. The servo coefficient may also be adjusted based, at least in part, on decibel (dB) feedback information from a destination device for which the audio message is intended.
摘要翻译：系统，方法和装置旨在通过网络管理音频消息，例如因特网协议语音（VOIP）语音消息。本发明采用统计机制来自动优化用于设置由客户端设备发送的音频消息的音量的增益控制。部分地基于音频消息中能量级别的统计采样来自动调整初始增益值。环境因素，例如客户端设备内的声卡，背景噪声等，也可以通过设置可用于在音量级别和分贝电平之间映射的伺服系数来考虑。伺服系数也可以至少部分地基于来自音频消息所针对的目的地设备的分贝（dB）反馈信息进行调整。

9. 发明申请

US20060256772A1 Selecting a network for routing real-time audio 审中-公开
标题翻译：选择用于路由实时音频的网络
公开(公告)号：US20060256772A1
公开(公告)日：2006-11-16
申请号：US11128646
申请日：2005-05-12
申请人： Madhu Yarlagadda
发明人： Madhu Yarlagadda
IPC分类号： H04L12/66
CPC分类号： H04L12/5692 , H04L12/6418 , H04L69/14 , H04M7/0075
摘要： A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) message over a network. A computing device may be configured to select a network connection for which to send the message to a destination based on a variety of factors. Duplicate message packets may be communicated to the destination device through multiple network connections. The multiple network connections may include a peer-to-peer network connection, a peer network connection, an ad-hoc network connection, or the like. Metrics may be collected about the multiple network connections. A determination may be made based, in part, on the metrics whether one network connection is optimal over another network connection. If so, that network connection may be selected to continue to provide the message packets, and the communication of the duplicate packets is ceased.
摘要翻译：系统，方法和装置旨在通过网络管理IP语音（VOIP）消息。计算设备可以被配置为基于各种因素来选择要向其发送消息的网络连接。可以通过多个网络连接将重复的消息分组传送到目的地设备。多个网络连接可以包括对等网络连接，对等网络连接，自组织网络连接等。可能收集有关多个网络连接的度量。可以部分地基于度量来确定一个网络连接是否比另一个网络连接最佳。如果是这样，则可以选择该网络连接来继续提供消息分组，并且停止重复分组的通信。

10. 发明申请

US20060256721A1 Dynamically selecting codecs for managing an audio message 有权
标题翻译：动态选择用于管理音频消息的编解码器
公开(公告)号：US20060256721A1
公开(公告)日：2006-11-16
申请号：US11128897
申请日：2005-05-13
申请人： Madhu Yarlagadda , Jamie Wiegand
发明人： Madhu Yarlagadda , Jamie Wiegand
IPC分类号： H04J1/16 , H04L12/66
CPC分类号： H04L29/06027 , H04L65/1006 , H04L65/103 , H04L65/104 , H04L65/1043 , H04L65/608 , H04L65/80
摘要： A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) messages over a network, employing the Real-time Transport Protocol (RTP) and Session Initiation Protocol (SIP) over the Transmission Control Protocol (TCP). The VOIP messages are sent by a source device to a destination device through a relay server. The relay server may throttle the VOIP messages employing buffer management. When the buffer is substantially full, the relay server will drop packets from the source device. Indication of the lost packets may be provided to the source device through a Real-time Transport Control Protocol (RTCP) report. The source device may then employ the RTCP report to modify a type of codec employed, and thereby adjust a rate of flow of VOIP packets sent towards the destination device. Additionally, the relay server may provide port translation services for RTP/RTCP packets between the source and destination devices.
摘要翻译：系统，方法和装置旨在通过在传输控制协议（TCP）上采用实时传输协议（RTP）和会话发起协议（SIP）的网络来管理IP语音（VOIP）消息。 VOIP消息由源设备通过中继服务器发送到目的设备。中继服务器可以通过缓冲管理来调节VOIP消息。当缓冲区基本为满时，中继服务器将丢弃源设备的数据包。丢失分组的指示可以通过实时传输控制协议（RTCP）报告提供给源设备。然后，源设备可以使用RTCP报告来修改所采用的编解码器的类型，从而调整向目的地设备发送的VOIP分组的流量。此外，中继服务器可以在源设备和目的设备之间为RTP / RTCP分组提供端口转换服务。

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