Cisco Technology, Inc.

David William Geen

Satish Joshi

Lewis Dean Dodrill

Ryan Alan Danner

Steven J Martin

Narasimha K Nayak

Geetha Ravishankar

Susan Harrow Barban

Zhiwei Zhang

Swaminathan Ravishankar

William Bateman Willaford, Iv

Tribhuvan Kambham

Melissa Lee Denbar

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).

IP unified agent using an XML voice enabled web based application server

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.

Calling service using voice enabled web based application server

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.

Scalable voice over IP system configured for dynamically switching codecs during a call

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.

Voice over IP voice mail system configured for placing an outgoing call and returning subscriber to mailbox after call completion

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.

Apparatus and method for providing server state and attribute management for multiple-threaded voice enabled web applications

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.

Proxy browser providing voice enabled web application audio control for telephony devices

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.

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King George, VA, United States of America

David William Geen