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1. 发明授权

US06735288B1 Voice over IP voice mail system configured for placing an outgoing call and returning subscriber to mailbox after call completion 有权
标题翻译：语音IP语音邮件系统被配置为在呼叫完成之后发出呼出呼叫并将用户返回给邮箱
公开(公告)号：US06735288B1
公开(公告)日：2004-05-11
申请号：US09479235
申请日：2000-01-07
申请人： David William Geen , Narasimha K. Nayak
发明人： David William Geen , Narasimha K. Nayak
IPC分类号： H04M164
CPC分类号： H04L29/06027 , H04L65/1009 , H04L65/1026 , H04L65/103 , H04L65/1036 , H04L65/104 , H04L65/608 , H04L65/80 , H04M3/533 , H04M3/53333 , H04M3/58
摘要： An IP telephony gateway and a voice mail resource enable a voice mail subscriber to place an outgoing call to a destination party from a voice mail session via a first Real Time Protocol (RTP) data stream according to the voice over IP (H.323) protocol, and resume the voice mail session upon completion of the outgoing call with the destination party. The voice mail resource initiates a second RTP data stream to a destination party, and uses the Empty Capability Set feature in the H.323 standard to cause the IP telephony gateway to close the first and second RTP data streams to the voice mail resource. The voice mail resource then issues Non-Empty Capability Set messages to the IP telephony gateway for the first and second RTP data streams, causing the IP telephony gateway to internally bridge the first and second RTP data streams, and later to resume the voice mail session.
摘要翻译： IP电话网关和语音邮件资源使得语音邮件订户能够根据IP语音（H.323）通过第一实时协议（RTP）数据流从语音邮件会话向目的地方发出呼叫，协议，并且在与目的地方的呼出完成时恢复语音邮件会话。语音邮件资源向目标方发起第二RTP数据流，并使用H.323标准中的空能力集功能使IP电话网关将第一和第二RTP数据流关闭到语音邮件资源。语音邮件资源然后向第一和第二RTP数据流的IP电话网关发出非空能力设置消息，导致IP电话网关内部桥接第一和第二RTP数据流，并且稍后恢复语音邮件会话。

2. 发明授权

US06977911B1 Scalable voice over IP system configured for dynamically switching codecs during a call 有权
标题翻译：配置为在呼叫期间动态切换编解码器的可扩展IP语音系统
公开(公告)号：US06977911B1
公开(公告)日：2005-12-20
申请号：US09629057
申请日：2000-07-31
申请人： David William Geen , Narasimha K. Nayak , Tribhuvan Kambham
发明人： David William Geen , Narasimha K. Nayak , Tribhuvan Kambham
IPC分类号： H04L12/28 , H04L29/06
CPC分类号： H04L65/1043 , H04L29/06027 , H04L65/1083
摘要： An IP telephony gateway and a media server are configured for changing media streams during an existing call according to the voice over IP (H.323) protocol, where a first H.245 media channel configured for transmitting a first media stream at a corresponding first compression such as G.723 or G.729A is closed and a second H.245 media channel configured for transmitting a second media stream at a corresponding second compression such as G.711 is opened during the same H.225 call.
摘要翻译： IP电话网关和媒体服务器被配置用于根据IP语音（H.323）协议在现有呼叫期间改变媒体流，其中第一H.245媒体信道被配置用于在对应的第一诸如G.723或G.729A的压缩被关闭，并且被配置用于在相应的第二压缩（例如G.711）处发送第二媒体流的第二H.245媒体信道在相同的H.225呼叫期间被打开。

3. 发明授权

US06954518B1 Voice over IP voice mail system configured for placing an outgoing call and returning subscriber to mailbox after call completion 有权
标题翻译：语音IP语音邮件系统被配置为在呼叫完成之后发出呼出呼叫并将用户返回给邮箱
公开(公告)号：US06954518B1
公开(公告)日：2005-10-11
申请号：US10819119
申请日：2004-04-07
申请人： David William Geen , Narasimha K. Nayak
发明人： David William Geen , Narasimha K. Nayak
IPC分类号： H04J3/16 , H04J3/22 , H04L12/16 , H04L12/66 , H04L29/06 , H04M1/64 , H04M3/533 , H04M3/58 , H04M7/00 , H04Q11/00
CPC分类号： H04L29/06027 , H04L65/1009 , H04L65/1026 , H04L65/103 , H04L65/1036 , H04L65/104 , H04L65/608 , H04L65/80 , H04M3/533 , H04M3/53333 , H04M3/58
摘要： An IP telephony gateway and a voice mail resource enable a voice mail subscriber to place an outgoing call to a destination party from a voice mail session according to the voice over IP (H.323) protocol, and resume the voice mail session upon completion of the outgoing call with the destination party. The IP telephony gateway establishes a voice mail session for the voice mail subscriber with the voice mail resource across a first Real Time Protocol (RTP) data stream. The voice mail resource initiates a second RTP data stream to a destination party in response to reception of a prescribed command from the voice mail subscriber. Although an RTP bridge connecting the first and second RTP data streams can be maintained by the voice mail resource, the voice mail resource may also use the Empty Capability Set feature in the H.323 standard to cause the IP telephony gateway to close the first and second RTP data streams to the voice mail resource. The voice mail resource then issues Non-Empty Capability Set messages to the IP telephony gateway for the first and second RTP data streams, causing the IP telephony gateway to internally bridge the first and second RTP data streams. The voice mail resource monitors connections between the voice mail subscriber and the destination party, and upon detecting a disconnect by the destination party causes the IP telephony gateway to resume the voice mail session, by repeating the sequence of sending Empty Capability Set and Non-Empty Capability Set messages to the IP telephony gateway to break down the bridge and re-establish the connection between the voice mail subscriber and the voice mail resource.
摘要翻译： IP电话网关和语音邮件资源使得语音邮件订户能够根据IP语音（H.323）协议从语音邮件会话向目的地方发出呼叫，并且在完成后继续语音邮件会话与目的地方的呼出。 IP电话网关通过第一实时协议（RTP）数据流与语音邮件资源建立语音邮件用户的语音邮件会话。响应于从语音邮件订户接收到规定的命令，语音邮件资源向目的地发起第二RTP数据流。虽然可以通过语音邮件资源来维护连接第一和第二RTP数据流的RTP桥，语音邮件资源也可以使用H.323标准中的空能力集特征来使得IP电话网关关闭第一和第第二RTP数据流到语音邮件资源。语音邮件资源然后向第一和第二RTP数据流的IP电话网关发出非空能力设置消息，导致IP电话网关内部桥接第一和第二RTP数据流。语音邮件资源监视语音邮件订户和目的方之间的连接，并且在检测到目的方之间的断开连接时，会使IP电话网关通过重复发送空能力集和非空的顺序来恢复语音邮件会话能力将消息设置到IP电话网关以分解网桥，并重新建立语音邮件订阅者和语音邮件资源之间的连接。

4. 发明申请

US20090021639A1 Audio and Video Communication 审中-公开
标题翻译：音视频通信
公开(公告)号：US20090021639A1
公开(公告)日：2009-01-22
申请号：US12224216
申请日：2007-02-21
申请人： David William Geen , Robert Lockwood , Jingyi Hu
发明人： David William Geen , Robert Lockwood , Jingyi Hu
IPC分类号： H04N5/04 , G06F15/16
CPC分类号： H04N21/4341 , H04L29/06027 , H04L65/1069 , H04L65/4007 , H04L65/80 , H04L69/40 , H04N7/147 , H04N21/234318 , H04N21/2368 , H04N21/4307 , H04N21/8547
摘要： In order to correct the skew experienced by the end user, a ‘reverse skew’ is applied by a video IVR, resulting in synchronized data at the edge. This is achieved by ‘sliding’ the time-bases of audio relative to video prior to delivery. Therefore, the data as received by the end user is synchronized. Media interfaces towards the video IVR are full duplex; the server corrects the skew in the respective halves of the duplex, particularly dependent on the type of service being deployed on the video IVR. For messaging applications, correcting the skew of the received data is important prior to the actual storage of the data. By applying the same technique as used for play-out, the skew can be corrected. The video IVR slides the time-base of audio relative to video before saving the multimedia data to the storage device. As a result, data saved is synchronized.
摘要翻译：为了纠正终端用户经历的偏斜，视频IVR应用“反向偏斜”，导致边缘处的同步数据。这是通过在交付之前“滑动”音频相对于视频的时基来实现的。因此，最终用户接收的数据是同步的。到视频IVR的媒体接口是全双工的; 服务器校正双工相应两半的偏斜，特别是取决于部署在视频IVR上的服务类型。对于消息传递应用，在实际存储数据之前，校正接收到的数据的偏斜是重要的。通过应用与播放相同的技术，可以纠正偏斜。在将多媒体数据保存到存储设备之前，视频IVR相对于视频滑动音频的时基。结果，保存的数据是同步的。

5. 发明授权

US06883015B1 Apparatus and method for providing server state and attribute management for multiple-threaded voice enabled web applications 有权
标题翻译：用于为多线程语音启用的Web应用程序提供服务器状态和属性管理的装置和方法
公开(公告)号：US06883015B1
公开(公告)日：2005-04-19
申请号：US09538899
申请日：2000-03-30
申请人： David William Geen , Geetha Ravishankar , Satish Joshi , Melissa L. Denbar , William Bateman Willaford, IV , Zhiwei Zhang
发明人： David William Geen , Geetha Ravishankar , Satish Joshi , Melissa L. Denbar , William Bateman Willaford, IV , Zhiwei Zhang
IPC分类号： G06F15/16 , G10L15/28
CPC分类号： G10L15/30
摘要： An application server generates and maintains a server-side data record, also referred to as a “brownie”, that includes application state information and user attribute information for multiple users within a single session controlled by a web-based browser. The brownie includes a session identifier that uniquely identifies the session, and a subsession identifier that uniquely identifies each corresponding user of the application session. As each new user is added to the session, for example by initiating a call to the new user, the application server stores the subsession identifier and corresponding application state information for the new user in the same brownie. In response to receiving a second web page request from the browser that includes the session identifier, the application server initiates a new web application instance, and recovers the brownie from the memory based on the session identifier included in the second page request.
摘要翻译：应用服务器生成并维护也称为“布朗”的服务器端数据记录，其包括由基于web的浏览器控制的单个会话内的多个用户的应用状态信息和用户属性信息。布朗尼包括唯一地标识会话的会话标识符，以及唯一标识应用会话的每个对应用户的子会话标识符。当每个新用户被添加到会话中时，例如通过发起对新用户的呼叫，应用服务器将新用户的子会话标识符和相应的应用状态信息存储在相同的布朗尼中。响应于从包括会话标识符的浏览器接收到第二网页请求，应用服务器启动新的web应用实例，并且基于包括在第二寻呼请求中的会话标识符从存储器恢复布朗。

6. 发明授权

US06738803B1 Proxy browser providing voice enabled web application audio control for telephony devices 有权
标题翻译：代理浏览器为电话设备提供语音启用的Web应用程序音频控制
公开(公告)号：US06738803B1
公开(公告)日：2004-05-18
申请号：US09459927
申请日：1999-12-14
申请人： Lewis Dean Dodrill , David William Geen , Satish Joshi , Ryan Alan Danner , Steven J. Martin
发明人： Lewis Dean Dodrill , David William Geen , Satish Joshi , Ryan Alan Danner , Steven J. Martin
IPC分类号： G06F1516
CPC分类号： H04L67/2819 , G10L15/22 , G10L15/26 , H04L29/06 , H04L67/28 , H04L67/34 , H04L69/329 , H04M3/4938 , H04M3/53 , H04M3/533 , H04M7/12 , H04M2203/253 , H04M2203/4509
摘要： A unified web-based voice messaging system provides voice application control between a proxy browser having a web browser, and an application server via an hypertext transport protocol (HTTP) connection on an Internet Protocol (IP) network. The proxy browser serves as an HTTP interface for a user device that lacks HTML and HTTP processing capabilites, such as an analog telephone, a cellular telephone, a voice over IP telephone, and the like. The web browser receives an HTML page from the application server having an XML element that defines data for an audio operation to be performed by an executable audio resource within the proxy browser. The audio resource, also referred to as a media resource, selectively executes the HTML tags and the audio operation based on the determined capabilities of the user device. If the user device does not have audio capabilities, the media resource ignores the audio operation, and merely presents the HTML information, assuming the user device has a display. If the media resource determines that the user device has at least a speaker and possibly a microphone, the media resource executes the audio operation based on enhanced audio control specified by the XML element. Similarly, if the media resource determines that the user device does not have a display, the HTML tag information is discarded by the media resource. Hence, a proxy browser can be used by user devices to access enhanced voice control for voice enabled web applications.
摘要翻译：统一的基于网络的语音消息系统在具有网络浏览器的代理浏览器和经由互联网协议（IP）网络上的超文本传输协议（HTTP））连接的应用服务器之间提供语音应用控制。代理浏览器用作缺乏HTML和HTTP处理能力的用户设备的HTTP接口，例如模拟电话，蜂窝电话，IP电话等。 Web浏览器从具有XML元素的应用服务器接收HTML页面，该XML元素定义要由代理浏览器内的可执行音频资源执行的音频操作的数据。也称为媒体资源的音频资源基于所确定的用户设备的能力来选择性地执行HTML标签和音频操作。如果用户设备没有音频功能，则媒体资源忽略音频操作，并且假设用户设备具有显示器，仅呈现HTML信息。如果媒体资源确定用户设备至少具有扬声器和可能的麦克风，则媒体资源基于由XML元素指定的增强音频控制来执行音频操作。类似地，如果媒体资源确定用户设备没有显示，则HTML标签信息被媒体资源丢弃。因此，用户设备可以使用代理浏览器来访问用于支持语音的Web应用的增强的语音控制。

7. 发明授权

US07836188B1 IP unified agent using an XML voice enabled web based application server 有权
标题翻译： IP统一代理使用基于XML语音的基于Web的应用服务器
公开(公告)号：US07836188B1
公开(公告)日：2010-11-16
申请号：US09605848
申请日：2000-06-29
申请人： Lewis Dean Dodrill , David William Geen , Geetha Ravishankar , Satish Joshi , Ryan Alan Danner , Susan Harrow Barban , Steven J. Martin , Swaminathan Ravishankar
发明人： Lewis Dean Dodrill , David William Geen , Geetha Ravishankar , Satish Joshi , Ryan Alan Danner , Susan Harrow Barban , Steven J. Martin , Swaminathan Ravishankar
IPC分类号： G06F15/16
CPC分类号： H04L67/02 , H04L65/1046 , H04L65/1063
摘要： A method for inter-process communication between non-persistent application instances includes establishing a first non-persistent application instance serving a first party and establishing a second non-persistent application instance serving a second party. In the first application instance, an HTML page is generated having instructions for a persistent browser instance having received the HTML page to initiate a new application session for the second party. Thus, inter-application communication is possible where one non-persistent application instance is able to notify and/or interrupt another non-persistent application instance by way of an associated real time component (web browser or proxy browser).
摘要翻译：用于非持久应用实例之间的进程间通信的方法包括：建立服务于第一方的第一非持久应用实例，以及建立服务第二方的第二非持久应用实例。在第一应用实例中，生成HTML页面，其具有已经接收到HTML页面的持久浏览器实例的指令，以启动用于第二方的新应用程序会话。因此，应用间通信是可能的，其中一个非持久应用实例能够通过相关联的实时组件（web浏览器或代理浏览器）通知和/或中断另一个非持久性应用程序实例。

8. 发明授权

US07502993B1 Calling service using voice enabled web based application server 有权
标题翻译：使用支持语音功能的基于Web的应用服务器呼叫服务
公开(公告)号：US07502993B1
公开(公告)日：2009-03-10
申请号：US09604654
申请日：2000-06-27
申请人： Lewis Dean Dodrill , David William Geen , Satish Joshi , Ryan Alan Danner , Steven J. Martin
发明人： Lewis Dean Dodrill , David William Geen , Satish Joshi , Ryan Alan Danner , Steven J. Martin
IPC分类号： G06N3/00
CPC分类号： H04L67/02 , H04L65/1046 , H04L65/1063
摘要： A method is provided in an application server for executing a calling application. The method includes receiving an HTTP request for execution of a calling application operation for a caller. A selected extensible markup language (XML) document is accessed in response to reception of the HTML request. Based on the XML document, a first HTML page including prompts is generated for the caller. A directory is accessed based on an input from the caller to obtain called party information. A second HTML page is generated having instructions for contacting the called party. Hence, calling services may be deployed on a platform that is customizable, scalable, and built upon open standards such as Internet protocol. By directly contacting an application server upon picking-up a telephone device, an intelligent system is provided for making telephone calls over an IP network.
摘要翻译：在应用服务器中提供一种用于执行呼叫应用的方法。该方法包括接收用于执行呼叫者的呼叫应用操作的HTTP请求。响应于HTML请求的接收来访问所选择的可扩展标记语言（XML）文档。基于XML文档，为呼叫者生成包含提示的第一个HTML页面。基于来自呼叫者的输入来访问目录以获得被叫方信息。生成具有与被叫方联系的指令的第二HTML页面。因此，呼叫服务可以部署在可定制，可扩展和建立在诸如因特网协议之类的开放标准的平台上。通过在拾取电话设备时直接联系应用服务器，提供智能系统用于通过IP网络进行电话呼叫。

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