Abstract:
Protocol control devices each have a database for storing application information concerning each of a number of IP (Internet Protocol) terminals controlled by the protocol control device. A dedicated maintenance terminal has a database for temporarily storing application information possessed by each protocol control device. An IP-capable PBX receives the application information of a designated IP terminal from the relevant protocol control device in response to a back-up command from the dedicated maintenance terminal. The dedicated maintenance terminal receives this application information and saves it temporarily for back-up purposes in a database.
Abstract:
A Voice Over Internet Protocol gateway support of supplementary services in a communication system. In one embodiment, a gateway includes a main state machine that is adapted to process a plurality of different type supplemental services with a single process.
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.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.
Abstract:
Various embodiments of the invention provide novel apparatus, methods and systems for providing relatively high-speed bandwidth to enable, inter alia, video transmission services over media previously unable to support such services. In accordance with certain embodiments, a device located at the telecommunication service provider can logically couple two or more physical media to provide a single, consolidated source of bandwidth, which can be used to transmit data, which can represent a video signal. In accordance with other embodiments, a device located at the subscriber's location can be used to receive the data from each of the two physical media and recreate the video signal from the data, such that the video signal can be transmitted to a display device, such as a television, monitor, etc.
Abstract:
When a line for communication of audio signals between first and second portable-telephone devices through a first portable-telephone base station, a second portable-telephone base station and a portable-telephone line network has been connected, IP addresses are exchanged. The IP addresses are required for communicating image data by first and second terminal adapters associated with the first and second portable-telephone devices respectively through an OCN. Image data is then transmitted to the IP addresses received in the communication through the OCN. To put it concretely, an image of the user of the first portable-telephone device is transmitted by the first terminal adapter through the OCN to the IP address of the second terminal adapter to be displayed by a second television receiver associated with the second terminal adapter. By the same token, an image of the user of the second portable-telephone device is transmitted by the second terminal adapter through the OCN to the IP address of the first terminal adapter to be displayed by a first television receiver associated with the first terminal adapter. Thus, by merely taking a portable-telephone device to a place like a store selling home electrical appliances including a television.
Abstract:
An improved services gateway environment is provided within a gateway framework. The improvement comprising a SIP service architecture that enables SIP entities to register with itself and translates such registrations into gateway aware registrations.
Abstract:
A digital residential entertainment system is disclosed recording video data of an event. The apparatuses include a processor communicating with memory. The memory stores video data of the event captured by a camera, and the video data includes a series of picture frames of the event. A loop buffer also stores video data of the event captured by the camera. A rule-based engine stored in the memory uses a set of rules to store the contents of the loop buffer in the memory. The apparatus utilizes the loop buffer to provide video data prior to occurrence of the event.
Abstract:
In order to provide a single common cost-efficient architecture for real time communication services for audio, video, and data over internet protocol, a voice over internet protocol (VoIP) system and architecture is provided by placing border elements (BEs) at the interface boundaries between the access network the user devices use and the VoIP infrastructure. The BEs use SIP protocol as the access call control protocol over any access networking technologies, for example, IP, Ethernet, ATM, and FR, and provides all services transparently to the end users that use SIP-enabled devices. To enable a scalable system, the SIP BEs are decomposed into separate communicating entities that make the SIP BE scalable and provide new capabilities not previously available by a self-contained SIP BE. Further, multiple levels of decomposition of a SIP BE can be provided by the present invention supporting a flexible and scalable SIP BE design that further improves system efficiencies and cost advantages as compared to use of single integrated border or edge elements. Further, a scalable SIP BE, made up of a plurality of physical entities for optimization of a large scale design, acts as a single integrated functional entity to logically execute a set of functions at the border of a VoIP infrastructure.
Abstract:
A virtual private network includes an internet protocol (IP) network and a public switched telephone network (PSTN). An enterprise gateway is operably connected to the IP network. The enterprise gateway is operably connected to a switch of the PSTN through a direct access line (DAL). The set-up signaling for virtual private network calls and the calls themselves are transported across the internet protocol network and the public switched telephone network through the direct access line.
Abstract:
A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.