Abstract: A data processing system comprises an audio processing means receiving data within said data processing system for converting the data into an analog signal with a first and a second analog channel components; an audio output means receiving the analog signal and providing a first and second output signal wherein the first output signal is provided for a loudspeaker and the output second signal is provided for a headset. The control of the audio processing means by a control signal providing an audio signal on the first or on the second channel or on both.

Abstract: A method and system for supporting a real-time communication on an integrated packet-based network involve accumulating information about the communication, referred to as signature data, while the communication is being setup and/or is in progress and storing the signature data in a database that is openly accessible to any applications that are connected to the network. Once signature data regarding an active real-time communication is accumulated, portions of the signature data may be utilized to further process the active real-time communication. Providing an openly accessible communication profile that is accumulated while the communication is in progress allows applications to provide a higher level of service by tailoring actions to the profile of the specific communication.

Abstract: Reconfiguration operations in a networking environment are actively monitored to automatically detect faulty network traffic routing conditions, such as routing loop. The autodetection includes identifying a condition in which one of a number of routers is switched to a reconfiguration mode. In response, a network monitoring state is initiated in which operational parameters of each router are recurringly analyzed to detect an unexpected deviation that evidences the faulty network traffic condition.

Abstract: A telecommunication system includes a source private branch exchange (12) that transmits telephone calls from a source to a destination private branch exchange (26) over a public switched telephone network (18). As an alternative to transmitting the calls over the public switched telephone network, the private branch exchange is coupled to a telephony Internet server 30 that can transmit telephone calls over a global wide area computer network such as the Internet (32). The private branch exchange (12) queries the destination private branch exchange (26) to determine if it similarly coupled to a telephony Internet server. If so, the bandwidth used by the telephony Internet server will allow the transmission of the telephone call, the call is routed over the Internet (32) to the intended recipient.

Abstract: An H.323 client terminal (102) according to an embodiment of the invention employs primary and secondary H.323 control units or state machines (110a, 110b). The primary control unit (110a) sends signaling messages to a primary gatekeeper (108a) and triggers the secondary control unit (110b) to send a message with an appropriate correlation identifier directly to a second client terminal. The primary control unit (110a) establishes a call per standard H.323 protocols. The secondary control unit (110b) also establishes a direct signaling connection to the second client terminal as back up. However, no media channels are established for this backup signaling. The secondary control unit (110b) checks the status of the call with the primary control unit (110a). If the call signaling on the primary control unit (110a) is proceeding normally, no further action is taken.

Abstract: An SIP-H.323 gateway (210) is provided, which functions as a multi-alias H.323 endpoint (204) with one alias representing one SIP agent (206). The SIP-H.323 gateway (210) processes the call signaling conversions and stores and passes media port information to the signaling parties, so that the existence of SIP agents (206) is transparent to the H.323 endpoints (204) and the features are transparently available to the H.323 endpoints.

Abstract: A partial emulation technique in which only the answers to poll frames and uplink information frames, or the slave part of the protocol, are emulated. All messages from the processor are passed down-link, and emulation is dependent upon the contents of an attendant list (452). In operation, the attendant list (452) is built by detecting responses to poll frames from the main controller (401). Once entries are made in the attendant list, partial emulation of responses to down link poll frames and up link information frames for each entry is allowed. No emulation of down link information frames occurs.

Abstract: A fiber-optic cable includes a conductive coating formed around an optical fiber. The conductive coating may be in the shape of a band or ring located at a terminal end of the fiber, or may extend along the entire length of the fiber or a portion thereof. For protection purposes, the fiber is covered with a buffer made from a plastic or polymer. The fiber-optic cable may be attached to a plug or receptacle of a connector. Preferably, the cable is attached to the plug. Under these circumstances, the receptacle would be equipped with a verification circuit for confirming that the plug has been properly installed into the receptacle housing. An electronic device may be used to alert a user as to whether proper installation has occurred. In addition to providing an indication that an optical connection has been made, the conductive coating may be adapted to carry electrical signals. In this way, the fiber-optic cable would operate as an electrical transmission line and an optical transmission line.

Abstract: A multipoint control unit (104) is provided which allows for dynamic codec selection. According to one embodiment, the MCU (104) causes endpoints (102, 106) to renegotiate their codec selections if a most-commonly available codec is not being used, upon entry of new parties to a teleconference. Alternatively, the codec renegotiation may be performed each time a user speaks, to optimize for maximum transmission quality or for minimizing transcoding.

Abstract: Techniques for validating an audio configuration of a computer are provided. Validation tests are performed on the audio configuration and results from the tests are displayed to a user as a report. The report can include an indication of the audio configuration settings that will adversely affect the audio quality for such uses as computer telephony. Additionally, the report can indicate a recommendation on changing the audio configuration to improve audio quality and can assist the user in making the audio configuration change.

Abstract: Methods, apparatus and computer system for efficiently utilizing a conference bridge capacity in order to support a multipoint conference call are described. Each conference bridge within multiple communication groups determines which local communication unit is a dominant local communication unit, quantizes that dominant local communication unit's signal, and transmits the quantized signal of the dominant local communication unit via signaling channels to the other conference bridges. Each conference bridge then compares the various quantized signals of the various dominant local communication units to determine an overall dominant communication unit of the multipoint conference. Then the signal of the overall dominant communication unit is then transmitted to all communication units participating in the conference call in various ways for different embodiments of the invention.

Abstract: A method and device for decoding a sequence of signals in which information is derived, in part, from a duration of (a) one of the signals and/or (b) an inter-signal pause between two of the signals. The method and device may also be used to decode a signal to form decoded information, derived, in part, from a duration of the signal.

Abstract: Apparatus and systems for recording Call ID data and transmitting the Caller ID data to remote telecommunications devices on demand, to fully or partially automate the dialing process for remotely calling back a message sender.

Abstract: A system and method for predicting the wait time of a caller to a call center is disclosed. The call center associates a set of agents to which the caller may be queued. This set of agents selected may depend on the skills that each agent possesses, the type of service request made by the caller, caller priority, time of day, day of week and other conditions. An initial wait time estimate may then be given to the caller who is just queued. As a caller's conditions may dynamically change, a caller's position in the queue may also change as well as the pool of available agents. Periodic wait time estimate updates may also be given to the queued caller. A caller's wait time may be estimated based upon mean inter-arrival times for recently past calls into the call center. An average inter-arrival time may be calculated for the last several calls. Alternatively, a caller's wait time may be estimated based upon calls that are recently queued and dequeued.

Abstract: To improve the quality of audio signals transmitted over a computer network, a source computer (12) and a destination computer (30) that participate in the call exchange information concerning the gain and the frequency characteristics response of the audio components of each participating computer. The parameters of the audio components of the source and/or destination computer are then adjusted to produce an audio signal having one or more desired characteristics.

Abstract: A method and an apparatus for protecting call privacy across telecommunication networks allow a calling party of a first telecommunication network to control the privacy of a telephone call to a called party of a second telecommunication network. This is achieved by selecting among a group of privacy options that represent varying degrees of privacy, assigning each of the selected privacy options to the call, and enforcing the selected privacy options within the second telecommunication network, thereby eliminating the need for external enforcement. The privacy options selected for the call limit the ability of a called party to manipulate the call. For example, the privacy options may prevent the called party from forwarding the call, transferring the call, conferencing the call, and/or listening to the call via a hands-free speaker phone.

Abstract: A communications system and interface (109, 210) for interfacing a proprietary digital telephone (108, 114) to a local area network (LAN) (104). The interface may be embodied in a computer expansion card or a stand-alone unit. The interface converts signals from the proprietary digital telephone's communication protocol to the LAN's communication protocol. In a specific embodiment, the LAN communication protocol is the H.323 protocol.

Abstract: An H.323 client terminal (102) according to an embodiment of the invention employs primary and secondary H.323 control units or state machines (110a, 110b). The primary control unit (110a) sends signaling messages to a primary gatekeeper (108a) and triggers the secondary control unit (110b) to send a message with an appropriate correlation identifier to a secondary gatekeeper (108b). The primary control unit (110a) establishes a call per standard H.323 protocols. The secondary control unit (110b) also establishes a call via a secondary gatekeeper (108b). However, no media channels are established for this back up call. As the secondary control unit (110b) receives signaling from the secondary gatekeeper (108b), it checks the status of the call with the primary control unit (110a). If the call signaling on the primary control unit (110a) is proceeding normally, no further action is taken. If the call signaling with the primary gatekeeper (108a) fails, the secondary control unit (110b) takes over communication.

Abstract: Systems and methods for interfacing a service component written in any one of a variety of programming languages to a native operating system application program interface (API) are described. For example, in one embodiment, a generic interface between the Win32 API (application program interface) and Windows NT service components written in C, C++, and JAVA is provided. In one aspect, an interface module is configured to load a service component written in a non-native programming language. In another aspect, an interface module is configured to retrieve service component information from a configuration database.

Abstract: A TAPI method according to an implementation of the present invention includes installing a TAPI service provider and associating a first PID with the TAPI service provider; re-installing the TAPI service provider and associating a second PID with the TAPI service provider; and automatically associating TAPI devices that had been associated with the first PID with the second PID.