Abstract: A source IPv6 mobile router is configured for establishing an IPv4 tunnel with destination IPv6 mobile router using a synthetic tag address, specifying a forwarding protocol, and IPv4 source and destination addresses. If an optional transport header is used (e.g, UDP port), the source port and destination port also are added to the synthetic tag address. The IPv6 packet includes a reverse routing header that enables the destination IPv6 mobile router to recover routing information for reaching the source IPv6 mobile router via the IPv4 network. Hence, all IPv4 routing information that may be needed by the destination IPv6 mobile router in sending an IPv6 reply packet back to the source IPv6 mobile router is maintained in the routing header specified in the IPv6 reply packet.

Abstract: A network-enabled user interface device, for example a VoIP telephony device, includes a display, a user input interface, an interface controller, and an application controller. The interface controller is configured for generating display elements for the display, controlling the user input interface based on received commands, and outputting responses to the application controller. The application controller is configured for outputting the commands to the interface controller based on application operations executed locally (i.e., within the user interface device), or remotely (e.g., by a server in communication with the user interface device). Remote application operations may include communications between the application controller and the remote server. The interface controller also may generate the display elements based on messages received from the network.

Abstract: A unified communications system is configured for receiving, storing, and/or sending SMS messages based on management of the SMS messages according to a prescribed open standards-based protocol. The unified communications system includes an SMS module configured for receiving SMS messages according to SMPP protocol. The SMS module obtains source and destination information from the received SMS message, and accesses a subscriber profile directory for subscriber profile information based on at least one of the source and destination information. The SMS module generates a common format message that includes the SMS message, and selectively supplies the common format message to selected destinations based on the accessed subscriber profile information. The common format message may be stored in a prescribed message store, for example an IMAP directory, or output as an e-mail message to a prescribed destination.

Abstract: A user computer, for example a personal computer (PC), includes a recorder configured for generating an audio file having an identifier enabling a unified messaging system to identify the audio file as a voice message, and an e-mail client configured for sending the audio file to a prescribed destination corresponding to a unified messaging subscriber voice mail inbox, enabling the unified messaging subscriber to use the unified messaging system for retrieval of the audio file as a stored voice message.

Abstract: A signaling network node is configured for identifying a received signaling message as belonging to one of a plurality of prescribed originating node groups, based on prescribed message class selection criteria. Each originating node group has at least one originating node and is assigned at least one congestion threshold relative to a message rate (e.g., messages per unit time or number of bytes per unit time). The signaling network node determines the message rate for each corresponding originating node group; if the message rate for an originating node group exceeds the corresponding at least one congestion threshold, the signaling network node outputs a congestion signaling message to the originating node having sent the signaling message, for reduction of the corresponding message rate.

Abstract: A unified web-based voice messaging system uses an application server, configured for executing a voice application defined by XML documents, that accesses subscriber attributes from a standardized information database server (such as LDAP), and messages from a standardized messaging server (such as IMAP), regardless of message format. The application server, upon receiving a request from a browser serving a user, accesses the standardized database server to obtain attribute information for responding to the voice application operation request. The application server generates an HTML document having media content and control tags for personalized execution of the voice application operation based on the attribute information obtained from the standardized database server. The application server also is configured for storing messages for a called party in the standardized messaging server by storing within the message format information that specifies the corresponding message format.

Abstract: A notification system includes a dial map configured for converting the destination telephone number within a notification message from an original format to another format recognizable by a messaging server, based on execution of a mapping rule selected based on a match between the mapping rule and the destination telephone number having the original format.

Abstract: A processor based system includes a parser, configured for identifying whether an input word received from a user is a new command word relative to a character-based command parse tree, and a tree management process configured for managing the character-based command parse tree. The character-based command parse tree includes multiple element levels for respective character positions for each known command word, and at least one character element at each level. Each character element specifies at least one corresponding character component and a corresponding at least one index value.

Abstract: 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.

Abstract: A method is provided in an application server configured for responding to hypertext transport protocol (HTTP) requests. The method includes storing, in response to a first HTTP request, an XML document that specifies for a user, a call number of a second party. The stored XML document is retrieved based on a second HTTP request by the user. A first HTML document is generated based on the retrieved XML document. The first HTML document has instructions including the call number for accessing the second party. A second HTML document is generated based on a prescribed input received from the second party. The second HTML document has instructions for connecting the second party with the user. Hence, a user may speak with a called party without ever having to remain on hold.

Abstract: 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.

Abstract: A unified web-based voice messaging system provides voice application control between a web browser and an application server via an hypertext transport protocol (HTTP) connection on an Internet Protocol (IP) network. The application server executes the voice-enabled web application by runtime execution of a first set of extensible markup language (XML) documents that define the voice-enabled web application to be executed. The application server generates an HTML form specifying selected application parameters from an XML document executable by the voice application. The HTML form is supplied to a browser, enabling a user of the browser to input or modify application parameters for the corresponding XML document into the form. The application server inserts the received input application parameters into the XML document, and stores the document.

Abstract: A unified web-based voice messaging system provides voice application control between a web browser and an application server via an hypertext transport protocol (HTTP) connection on an Internet Protocol (IP) network. 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. The application server executes the voice-enabled web application by runtime execution of extensible markup language (XML) documents that define the voice-enabled web application to be executed. The application server includes a runtime environment that establishes an efficient, high-speed connection to a web server. The application server, in response to receiving a user request from a user, accesses a selected XML page that defines at least a part of the voice application to be executed for the user.

Abstract: Provided is an arrangement that enables deployment of international prompts in a voice messaging system based on a server configured for dynamically generating a selected concatenated prompt, from stored prompt fragments, in a selected language.

Abstract: A recording device at a calling party premises is configured for recording a message for a messaging subscriber by the calling party, and sending the recorded message, via a data network, to a destination message store serving the messaging subscriber. The recording device sends the recorded message based on messaging subscriber attributes retrieved via the data network. Hence, a calling party can utilize a recording device at the calling party premises for recording a message for a messaging subscriber, and sending the message via the data network to a destination message store, eliminating the necessity of establishing any voice-grade connection or a media connection between the calling party and any server (e.g., messaging server, media gateway server, etc.).

Abstract: An IP-based subscriber endpoint, for example a VoIP telephony device or an IP telephony user agent, is configured for retrieving recorded audio caller information in response to receiving an incoming call that includes a calling party identifier. The IP-based subscriber endpoint, in response to receiving the calling party identifier, generates an IP-based query via an IP network to an identifiable server for identification of a source of the recorded audio caller information corresponding to the calling party identifier. The IP-based subscriber endpoint outputs a request to the source of the recorded audio caller information, and plays the recorded audio caller information in response to reception via the IP-based network. Hence, the IP-based subscriber endpoint is able to retrieve existing recorded audio caller information from servers configured for storing recorded names, eliminating the necessity for text to speech resources.

Abstract: 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.

Abstract: A router (e.g., a home agent for an IPv6 mobile router) is configured for determining a destination router (e.g., the IPv6 mobile router) for a received packet based on accessing a routing table having multiple routing entries, each routing entry including a routing key and a routing field that specifies one of a prescribed address specifying the destination router and a computation tag. The computation tag specifies a prescribed function to be executed to calculate a determined address for the destination router (e.g., the home address for the IPv6 mobile router). The router identifies, for each received packet, the matching routing entry based on the corresponding routing key, and in response to detecting the computation tag in the routing field, selectively executes the corresponding function to calculate the determined address for the destination router.

Abstract: An application server, configured for executing an application defined by XML documents for delivering voice portal services, accesses subscriber attributes from a standardized information database server (such as LDAP). The application server, upon receiving a request from a proxy browser serving a user, accesses the standardized database server to obtain attribute information for responding to the application request. The application server sends a request to a content server for media content based on the HTTP request and the attribute information. The application server generates an HTML page, for execution of the prescribed voice portal service application having XML tags configured for controlling delivery of the media content in an audible format, based on the HTTP request.

Abstract: A unified web-based voice messaging system provides voice application control between a web browser and an application server via an hypertext transport protocol (HTTP) connection on an Internet Protocol (IP) network. The application server, configured for executing a voice application defined by XML documents, selects an XML document for execution of a corresponding voice application operation based on a determined presence of a user-specific XML document that specifies the corresponding voice application operation. The application server, upon receiving a voice application operation request from a browser serving a user, determines whether a personalized, user specific XML document exists for the user and for the corresponding voice application operation.