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    • 1. 发明授权
    • Dynamic determination of initial filter criteria
    • 动态确定初始过滤标准
    • US09148453B1
    • 2015-09-29
    • US13047118
    • 2011-03-14
    • Bejoy PankajakshanCharles Brent HirschmanJeremy R. Breau
    • Bejoy PankajakshanCharles Brent HirschmanJeremy R. Breau
    • H04L29/06H04L29/08H04W24/00
    • H04L65/105H04L65/1006H04L65/1046H04L65/1069H04L65/1073H04L65/1076H04L67/16
    • An IP multimedia subsystem comprises a CSCF and an HSS. The CSCF is configured to receive a SIP registration message transferred by a user communication device and responsively transfer a registration request to the HSS. The HSS is configured to process the registration request to obtain IFC for the user communication device, wherein the IFC comprises processing instructions that direct the CSCF to perform an operation to dynamically obtain and load data into the IFC. The CSCF is configured to receive a SIP invite message transferred by the user communication device, process text from the SIP invite message against a set of keywords to determine a dynamic parameter based on a keyword match, process one of the processing instructions in the IFC to obtain and load the dynamic parameter into the IFC, process the IFC with the dynamic parameter to determine a communication session parameter, and transfer a SIP message with the communication session parameter responsive to the SIP invite message.
    • IP多媒体子系统包括CSCF和HSS。 CSCF被配置为接收由用户通信设备传送的SIP注册消息,并且响应地将注册请求传送到HSS。 HSS被配置为处理注册请求以获得用于用户通信设备的IFC,其中IFC包括指示CSCF执行操作以动态获取并将数据加载到IFC中的处理指令。 CSCF被配置为接收由用户通信设备传送的SIP邀请消息,针对一组关键字处理来自SIP邀请消息的文本,以基于关键字匹配确定动态参数,将IFC中的处理指令之一处理为 获得并将动态参数加载到IFC中,使用动态参数处理IFC以确定通信会话参数,并响应于SIP邀请消息传送具有通信会话参数的SIP消息。
    • 2. 发明授权
    • Transferring an active wireless voice call to a packet network
    • 将有源无线语音呼叫转移到分组网络
    • US08599776B1
    • 2013-12-03
    • US12862472
    • 2010-08-24
    • Jeremy R. BreauBejoy PankajakshanCharles Brent Hirschman
    • Jeremy R. BreauBejoy PankajakshanCharles Brent Hirschman
    • H04B7/216
    • H04W36/0022
    • Systems and methods are provided herein for transferring an active voice call to a packet communication network. In a particular embodiment, a method provides establishing a voice call having an initial signaling channel and an initial voice channel between a first wireless device and a second wireless device over a first and second wireless access system. The method further includes establishing a new signaling channel for the voice call between the first wireless device and a separate packet network. Also, the method provides extending the new signaling channel for the voice call from the separate packet network to the second wireless access system over the first wireless access system. Furthermore, the method includes using the new signaling channel to establish a new voice channel for the voice call over the separate packet network, wherein the first wireless access system remains in the new signaling channel.
    • 本文提供的系统和方法用于将活动语音呼叫传送到分组通信网络。 在特定实施例中,一种方法提供在第一和第二无线接入系统上建立具有初始信令信道和第一无线设备与第二无线设备之间的初始语音信道的语音呼叫。 该方法还包括为第一无线设备和单独的分组网络之间的语音呼叫建立新的信令信道。 而且,该方法提供了通过第一无线接入系统将语音呼叫的新信令信道从单独的分组网络延伸到第二无线接入系统。 此外,该方法包括使用新的信令信道在单独的分组网络上为语音呼叫建立新的语音信道,其中第一无线接入系统保持在新的信令信道中。
    • 3. 发明授权
    • Timer based logic component for initial filter criteria in a wireless communication system
    • 基于定时器的逻辑组件,用于无线通信系统中的初始过滤器标准
    • US08537843B1
    • 2013-09-17
    • US12858099
    • 2010-08-17
    • Jeremy R. BreauBejoy PankajakshanCharles Brent Hirschman
    • Jeremy R. BreauBejoy PankajakshanCharles Brent Hirschman
    • H04L12/28H04L12/56
    • H04L65/1016H04L65/1069H04L67/2819H04L69/28
    • Systems and methods are disclosed herein for providing initial filter criteria with a time limit for providing an application service. In a particular embodiment, a method includes receiving a service request for an application into an IP Multimedia Subsystem (IMS). The method further provides, in the IMS, processing the service request to generate initial filter criteria that includes a first logic component that has a first time limit to control provision of the application to the user and transferring the initial filter criteria from the IMS and receiving the initial filter criteria into an application server. Additionally, the method includes, in the application server, before the first time limit expires, providing the application to the user based on the initial filter criteria with the first logic component, and after the first time limit expires, providing the application to the user based on the initial filter criteria without the first logic component.
    • 本文公开的系统和方法用于提供具有用于提供应用服务的时间限制的初始过滤器标准。 在特定实施例中,一种方法包括将应用的服务请求接收到IP多媒体子系统(IMS)中。 该方法还在IMS中提供处理服务请求以产生初始过滤器标准,该初始过滤器标准包括具有第一时间限制的第一逻辑组件以控制向用户提供应用程序并从IMS传送初始过滤标准并接收 初始过滤条件到应用服务器。 此外,该方法在应用服务器中包括在第一时间限制到期之前,基于具有第一逻辑组件的初始过滤标准向用户提供应用,并且在第一时间限制到期之后,向用户提供应用 基于没有第一逻辑组件的初始过滤标准。
    • 5. 发明授权
    • Translation of congestion notification indicators in a base station system
    • 基站系统中拥塞通知指示符的翻译
    • US08681626B1
    • 2014-03-25
    • US12707460
    • 2010-02-17
    • Lyle Walter PaczkowskiPallavur SankaranaraynanBejoy Pankajakshan
    • Lyle Walter PaczkowskiPallavur SankaranaraynanBejoy Pankajakshan
    • G01R31/08
    • H04W28/0247H04W28/0205H04W76/12
    • A base station system comprises a processing system, an IP communication interface, and a wireless communication interface. The processing system is configured to direct the IP communication interface to establish communication connectivity over a communication link to a service node on a communication network, register with the service node over the communication link and the communication network, and establish a tunnel with the service node, direct the wireless communication interface to exchange user communications with a plurality of wireless communication devices in a wireless protocol, direct the IP communication interface to exchange the user communications with the service node over the tunnel and receive an IP layer three congestion notification indicator over the tunnel, process the IP layer three congestion notification indicator to generate a layer two congestion notification indicator in the wireless protocol, and direct the wireless communication interface to transfer the layer two congestion notification indicator in the wireless protocol to at least one wireless communication device.
    • 基站系统包括处理系统,IP通信接口和无线通信接口。 所述处理系统被配置为指示所述IP通信接口通过通信链路建立到通信网络上的服务节点的通信连接,通过所述通信链路和所述通信网络向所述服务节点注册,并与所述服务节点建立隧道 指导无线通信接口与无线协议中的多个无线通信设备交换用户通信,指示IP通信接口通过隧道交换与服务节点的用户通信,并在该接口上接收IP层三个拥塞通知指示符 隧道,处理IP层三拥塞通知指示符,在无线协议中生成二层拥塞通知指示符,并指示无线通信接口将无线协议中的第二层拥塞通知指示符传送给至少一个无线通信设备。
    • 6. 发明授权
    • Reconnecting dropped calls using an internet protocol multimedia subsystem
    • 使用互联网协议多媒体子系统重新连接掉话
    • US08644218B1
    • 2014-02-04
    • US13070796
    • 2011-03-24
    • Bejoy PankajakshanWarren B. CopePallavur Sankaranaraynan
    • Bejoy PankajakshanWarren B. CopePallavur Sankaranaraynan
    • H04W4/00H04L12/66
    • H04W76/19H04L65/1006H04L65/1016H04L65/1073
    • A call session control function (CSCF) controls a communication session between a first communication device and a second communication device. The first communication device detects a loss of signal and transmits a modified session initiation protocol (SIP) bye message. The modified SIP bye message includes a code that indicates a loss of the communication session and an identifier for the second communication device. The CSCF receives the modified SIP bye message and in response to the code, subscribes with a home subscriber server (HSS) for a registration notification for the second communication device. The second communication device subsequently transfers a SIP registration request and the CSCF responsively receives the registration notification for the second communication device from the HSS that indicates a network address for the second communication device. In response to the registration notification, the CSCF transfers SIP invite messages to the first and second communication devices to re-establish the communication session.
    • 呼叫会话控制功能(CSCF)控制第一通信设备和第二通信设备之间的通信会话。 第一通信设备检测信号丢失并发送修改的会话发起协议(SIP)再生消息。 经修改的SIP再生消息包括指示通信会话丢失的代码和用于第二通信设备的标识符。 CSCF接收经修改的SIP再次消息,并且响应于该代码,订阅用于第二通信设备的注册通知的归属订户服务器(HSS)。 第二通信设备随后传送SIP注册请求,并且CSCF响应地从指示第二通信设备的网络地址的HSS接收第二通信设备的注册通知。 响应于注册通知,CSCF将SIP邀请消息传送到第一和第二通信设备以重新建立通信会话。
    • 7. 发明授权
    • Tiered subscriber service with advanced header compression methods in a VOIP system
    • 在VOIP系统中使用高级头压缩方法的分层用户服务
    • US08971185B1
    • 2015-03-03
    • US13549574
    • 2012-07-16
    • Bejoy Pankajakshan
    • Bejoy Pankajakshan
    • H04J1/16H04L12/66H04L12/28G06F15/173
    • H04L47/38H04L47/14H04L47/19H04L47/20H04L47/22H04L47/2416H04L47/2441H04L47/39H04L65/1069H04L69/04H04W28/06H04W72/048
    • One or more computer readable media, methods and systems that allow phone calls involving a mobile telephone to be to be conducted using VOIP technology with header compression of the voice packets and dynamically adjustable tiered subscriber service. A dynamic, tiered-subscriber service allows the bandwidth allotted to users subscribing to different levels of service to be adjusted between sessions to accommodate changing network conditions. Header compression involves compressing the voice packet header into a reduced number of bites. Embodiments of the present invention enable simultaneous header compression and dynamic tiered-subscriber service through a new call flow set up that includes a new policy. The new policy is given preference over the other installed policies and includes new attribute value pairs that enables both header compression and dynamic bandwidth adjustments.
    • 一个或多个计算机可读介质,方法和系统,其允许涉及移动电话的电话通过语音分组的头部压缩和动态可调的分层用户服务的VOIP技术来进行。 动态分层订户服务允许分配给用户订阅不同级别的服务的带宽在会话之间进行调整,以适应不断变化的网络条件。 报头压缩包括将语音数据包报头压缩成减少的位数。 本发明的实施例通过包括新策略的新呼叫流设置能够同时进行报头压缩和动态分层用户服务。 新策略优先于其他已安装的策略,并包括新的属性值对,可实现头压缩和动态带宽调整。
    • 10. 发明授权
    • Tiered subscriber service with advanced header compression methods in a VOIP system
    • 在VOIP系统中使用高级头压缩方法的分层用户服务
    • US08264958B1
    • 2012-09-11
    • US12126090
    • 2008-05-23
    • Bejoy Pankajakshan
    • Bejoy Pankajakshan
    • H04J1/16H04W4/00H04L12/66H04L12/56
    • H04L47/38H04L47/14H04L47/19H04L47/20H04L47/22H04L47/2416H04L47/2441H04L47/39H04L65/1069H04L69/04H04W28/06H04W72/048
    • One or more computer readable media, methods and systems that allow phone calls involving a mobile telephone to be to be conducted using VOIP technology with header compression of the voice packets and dynamically adjustable tiered subscriber service. A dynamic, tiered-subscriber service allows the bandwidth allotted to users subscribing to different levels of service to be adjusted between sessions to accommodate changing network conditions. Header compression involves compressing the voice packet header into a reduced number of bites. Embodiments of the present invention enable simultaneous header compression and dynamic tiered-subscriber service through a new call flow set up that includes a new policy. The new policy is given preference over the other installed policies and includes new attribute value pairs that enables both header compression and dynamic bandwidth adjustments.
    • 一个或多个计算机可读介质,方法和系统,其允许涉及移动电话的电话通过语音分组的头部压缩和动态可调的分层用户服务的VOIP技术来进行。 动态分层订户服务允许分配给用户订阅不同级别的服务的带宽在会话之间进行调整,以适应不断变化的网络条件。 报头压缩包括将语音数据包报头压缩成减少的位数。 本发明的实施例通过包括新策略的新呼叫流设置能够同时进行报头压缩和动态分层用户服务。 新策略优先于其他已安装的策略,并包括新的属性值对,可实现头压缩和动态带宽调整。