Abstract:
According to this invention, a digital key telephone system connected to an analog public network NW having a function of transmitting a ringing signal including identification information of a calling line through a subscriber line (CO line), accommodating a plurality of extension lines each connected to a digital key telephone (DKT) 2 or a standard telephone (STT) 4 as an extension terminal, and having a function of switching and connecting the subscriber line to the plurality of extension lines or the extension lines to each other includes a called party storage means storing, in advance, information representing the correlation between the calling line and the extension terminals 2 and 4 as a called terminal. When a ringing signal arrives from the analog communication network NW, calling line identification information (caller ID) contained in the ringing signal is detected by a calling line identification information interface unit (RCIU) 12. A control unit (RCTU) 16 determines the called extension terminal on the basis of the detected caller ID and the information stored in the storage means, so the extension terminal receives the call from the digital key telephone interface unit (RDKU) 13 or a standard telephone interface unit (RSTU) 15.
Abstract:
Methods and apparatus are described for facilitating multiple data communications with a first platform (100, 102, and 104) using a single communication conduit (108) having first and second channels. First data are transmitted between the first platform and a second platform (114) via the first and second channels (202). It is determined whether an incoming call comprises second data of a particular type (206). Where the incoming call comprises the second data of the particular type (208), transmission of the first data via the second channel is inhibited (226) thereby enabling reception of the second data by the first platform via the second channel while the first data continue to be transmitted via the first channel. The second data are then received at the first platform via the second channel (230). Transmission of the first data via the first and second channels is resumed in response to termination of transmission of the second data (266).
Abstract:
In a simultaneous voice and data communication system, a stream of signal points is partitioned into a plurality of symbol blocks, each symbol block including a data segment and a control segment. The data segment carries information from a user, i.e., user data, while the control segment provides control information. A voice signal is then added to at least a portion, or all, of the signal points of each symbol block to provide for simultaneous voice and data transmission to an opposite endpoint. The control information may represent information from a secondary data source, and/or may include information about the characteristics of the succeeding block, e.g., the user data rate, and information pertaining to characteristics of the communications channel.
Abstract:
A single medium (audio) voice grade call initiated by an ISDN video phone and directed to another ISDN video phone is converted into a multimedia (audio and video) ISDN call by a) maintaining the connection for the voice grade call while the ISDN audio and video connection is being established over a bearer channel of the ISDN subscriber loop different from the bearer channel being used by the voice grade connection, b) switching the audio signals from the voice grade call to the ISDN audio and video connection, once the ISDN audio and video connection is established, and c) tearing down the initial voice grade call.
Abstract:
A packet communications network and apparatus for communicating information in voice and data packets transmits and receives voice and data in accordance with standardized frames of a standardized communications format such as a standard DS-1 type trunk. An apparatus is coupled to multiplex standardized channels into a single channel wherein voice or data signals are packetized into independently addressable packets synchronized on, for example, the DS-1 frame. The network includes voice and packet data concentration apparatus operable in a multiple node trunk environment to concentrate signals into independently addressable synchronously switchable packets, thereby to provide an efficient (high density capacity) interface between trunk terminations. Up to four times as much information can be transferred between nodes with the ability to switch between nodes as compared to conventional TDM and PCM communication without compression without the ability to switch between nodes.
Abstract:
A transmission system comprises apparatus for the transmission of voice signals only, data signals only, or a combination of both voice and data signals in a multiplexed stream of eight bit time slots over a single, bidirectional digital channel for a point-to-point connection. In the combined mode, the encoded voice signals, using low bit rate voice encoders, are assigned to four bit positions of the eight bit time slot; the data signals are assigned to the remaining four bits. Up to two bit positions normally used for data may be used for signature bits, thereby identifying whether the multiplexed stream comprises voice signals only, data signals only, or a combination of both voice and data signals. Also, a minimum of one "1" bit per eight bit time slot is thereby guaranteed.
Abstract:
A service node in a telecommunication network is arranged to receive a Multi-Media video call setup for a terminal to be called. The called terminal is called by the service node according to either a video or a voice call setup, depending on a current allowed answer mode of the called terminal. Responding to the call setup will, depending on either answering or rejecting the call setup by the called terminal, and depending whether the call setup is for a voice or a video call, be handled by the service node according to a predetermined method. When a call setup is answered by the called terminal, the service node further supports a method of toggling between a video call and a voice call at the called terminal.
Abstract:
A method and computer program product which allows both phone-based and IP-based clients to participate in a single audio conference. The method enables at least two multi-point control units (MCUs) (i.e., conferencing servers) to connect via a standard data linkage (i.e., full-duplex dial-up or IP link). The method and computer program product enables the phone-based MCU to handle the phone clients and the IP-based MCU to handle the IP-based clients, while connecting the two to allow each participating client to hear all other participating clients.
Abstract:
A method and computer program product which allows both phone-based and IP-based clients to participate in a single audio conference. The method enables at least two multi-point control units (MCUs) (i.e., conferencing servers) to connect via a standard data linkage (i.e., full-duplex dial-up or IP link). The method and computer program product enables the phone-based MCU to handle the phone clients and the IP-based MCU to handle the IP-based clients, while connecting the two to allow each participating client to hear all other participating clients.