公开(公告)号：US20050232250A1

公开(公告)日：2005-10-20

申请号：US11155498

申请日：2005-06-20

申请人： James Craig ,  Kellly Phillips ,  Narasimha Nayak

发明人： James Craig ,  Kellly Phillips ,  Narasimha 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 user interface resource enable a subscriber to place an outgoing call according to the voice over IP (H.323) protocol to a destination party from a user interface session of an intelligent dial tone service such as voice activated dialing, and resume the user interface session upon completion of the outgoing call with the destination party. The IP telephony gateway establishes a user interface session for the subscriber with the user interface resource across a first Real Time Protocol (RTP) data stream. The user interface resource initiates a second RTP data stream to a destination party in response to reception of a prescribed command from the subscriber. Although an RTP bridge connecting the first and second RTP data streams can be maintained by the user interface resource, the user interface 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 user interface resource. The user interface 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 user interface resource monitors connections between the subscriber and the destination party, and upon detecting a disconnect by the destination party causes the IP telephony gateway to resume the user interface 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 subscriber and the user interface resource.

公开(公告)号：US06697964B1

公开(公告)日：2004-02-24

申请号：US09534121

申请日：2000-03-23

申请人： Lewis Dean Dodrill ,  Narasimha Nayak ,  Ryan Alan Danner ,  Steven J. Martin

发明人： Lewis Dean Dodrill ,  Narasimha Nayak ,  Ryan Alan Danner ,  Steven J. Martin

IPC分类号： G06F1100

CPC分类号： G06F11/3414 ,  G06F11/3457 ,  G06F2201/875

摘要： A testing system for an application server configured for providing a unified web-based voice messaging system using voice application control between a web browser via an hypertext transport protocol (HTTP) connection. The testing system is configured for outputting HTTP-based application-specific requests to the application server based on execution of prescribed scripts that specify generation of the application-specific requests according to the state-aware application. Each script specifies parameters for a prescribed sequence of requests for a prescribed application operation for the application server, enabling the load generator to provide a sequence of requests, simulating the sequence of requests that would normally be sent by a user of the application server during a user application session. The testing system also includes a HTTP-based user interface that enables a user to select the scripts, as well as the number of instances for simultaneous execution of the selected number of iterations.

摘要翻译： 一种用于应用服务器的测试系统，被配置为通过超文本传输​​协议（HTTP）连接在web浏览器之间提供使用语音应用控制的统一的基于web的语音消息系统。 测试系统被配置为基于指定根据状态感知应用生成特定于应用的请求的指定脚本的执行向应用服务器输出基于HTTP的应用特定请求。 每个脚本指定用于针对应用服务器的规定应用操作的规定的请求序列的参数，使得负载生成器能够提供一系列请求，模拟通常在应用服务器的用户期间由应用服务器的用户发送的请求序列 用户应用程序会话。 测试系统还包括基于HTTP的用户界面，使用户能够选择脚本，以及用于同时执行所选迭代次数的实例数量。