Abstract: A system and a method for quantitatively measuring voice transmission quality within a voice-over-data-network such as a telephony-enabled LAN utilize speech recognition to measure the quality of voice transmission. A first aspect of the invention involves determining the suitability of the LAN for voice communications. A voice sample is selected by a first terminal and is transmitted to a second terminal on the LAN a number of times. The first terminal introduces an incrementally larger quantity of noise into each transmission of the voice packet. The second terminal performs speech recognition for each successively received voice sample and determines the accuracy for each speech recognition session. The amount of noise which drops the speech recognition accuracy below a threshold level provides a measure of the suitability of the LAN for voice communications. During normal operation of the LAN, speech recognition accuracy tests are performed between various endpoints to monitor voice transmission quality.

Abstract: A system and a method for quantitatively measuring voice transmission quality within a voice-over-data-network such as a telephony-enabled LAN utilize speech recognition to measure the quality of voice transmission. A first aspect of the invention involves determining the suitability of the LAN for voice communications. A voice sample is selected by a first terminal and is transmitted to a second terminal on the LAN a number of times. The first terminal introduces an incrementally larger quantity of noise into each transmission of the voice packet. The second terminal performs speech recognition for each successively received voice sample and determines the accuracy for each speech recognition session. The amount of noise which drops the speech recognition accuracy below a threshold level provides a measure of the suitability of the LAN for voice communications. During normal operation of the LAN, speech recognition accuracy tests are performed between various endpoints to monitor voice transmission quality.

Abstract: A system and method for supporting communications among multiple interconnected networks include assigning multiple dynamic telephony addresses to each wireless communication device that registers in more than one network. The networks assign the addresses independently of each other. When an incoming call is directed to a particular wireless device via a first network, if the wireless device is beyond the transmission range of the first network, a locate-wireless-communication-device message may be single-cast, multicast or broadcast to remote networks, with instructions to return dynamic telephony addresses assigned to the device. While the telephony addresses are different in each network, each wireless device is associated with a device identifier that is universally applied in the internetwork. Upon receiving a telephony address from a remote network, the address is stored in local memory at the first network, thereby allowing access for subsequent incoming calls.

Abstract: A system and method for supporting an interactive decision tree with placeholder capability include an interactive decision tree-based expert unit with an input enabled to receive user data that includes a user identification and with an output enabled to transmit system data. A memory is connected to the expert unit for storing the system data. Optionally, the user memory stores the identification and an associated decision tree node identifier representing a node in the decision tree visited by a user prior to termination of a communications link between the expert unit and the user. A device is provided for accessing the decision tree node identifier from the memory utilizing the associated user identification, so that a connecting device can automatically connect the user to a node within the decision tree based solely upon the user identification. Alternatively, the decision tree node identifier is transmitted to the user.

Abstract: A method of transmitting DTMF signals over a compressed computer network (24) comprises sending a packet containing a compressed DTMF signal followed by a confirmation packet. The confirmation packet preferably comprises an uncompressed DTMF signal which is transmitted on the same logical channel as the packets that represent voice signals. To detect the DTMF signal transmitted, a telephony Internet server (28) decompresses packets as they are received. If a packet appears to contain a compressed DTMF signal, the telephony Internet server looks for the presence of the confirmation packet before processing the previously received packet as a DTMF signal. In an alternative embodiment, the confirmation packet comprises a code that represents a DTMF signal. The packets that represent the voice signal and the code representing the DTMF signal are sent on separate virtual channels of the same logical channel.

Abstract: A user interface is organized using different personae. When in a master persona mode, a user is provided access to a full list of contacts and to a full set of data files. When in a first alternate persona mode, user access is provided to only a first subset of the full list of contacts and to only a first subset of the full set of data files.

Abstract: A user interface is organized using different personae. When in a master persona mode, a user is provided access to a full list of contacts and to a full set of data files. When in a first alternate persona mode, user access is provided to only a first subset of the full list of contacts and to only a first subset of the full set of data files.

Abstract: A method of transmitting DTMF signals over a compressed computer network comprises sending a packet containing a compressed DTMF signal followed by a confirmation packet. The confirmation packet preferably comprises an uncompressed DTMF signal. To detect the DTMF signal transmitted, a telephony Internet server decompresses packets as they are received. If a packet appears to contain a compressed DTMF signal, the telephony Internet server looks for the presence of the confirmation packet before processing the previously received packet as a DTMF signal. In another embodiment, an output of a speech compressor and a DTMF signal detector are transmitted on separate virtual channels of a real time protocol data stream.

Abstract: A method of transmitting DTMF signals over a compressed computer network (24) comprises sending a packet containing a compressed DTMF signal followed by a confirmation packet. The confirmation packet preferably comprises an uncompressed DTMF signal which is transmitted on the same logical channel as the packets that represent voice signals. To detect the DTMF signal transmitted, a telephony Internet server (28) decompresses packets as they are received. If a packet appears to contain a compressed DTMF signal, the telephony Internet server looks for the presence of the confirmation packet before processing the previously received packet as a DTMF signal. In an alternative embodiment, the confirmation packet comprises a code that represents a DTMF signal. The packets that represent the voice signal and the code representing the DTMF signal are sent on separate virtual channels of the same logical channel.

Abstract: An apparatus and method for selectively compressing information to be transmitted over a communications network includes an analog line monitor connected to an analog telephone and an analog data communications device. The analog line monitor is configured for detection of carrier tone signals associated with data transmissions. An analog-to-digital/digital-to-analog (AD/DA) converter is connected to the analog line monitor to convert the information contained in transmissions from the analog telephone and the analog data communications device from an analog format to a digital format. Optionally, the AD/DA circuitry is configured for demodulating data prior to transmission over the network. Compression circuitry is responsive to detection of a carrier tone signal by the analog line monitor, such that detection of a carrier tone signal suppresses compression of data transmitted by the data communications device.

Abstract: A conferencing system (100) improves audio quality by identifying and selectively allowing a dominant audio channel to bypass a lossy compression transformation included in a server 102. The system (100) places conference calls over a network (106) and includes the conference server (102), having an input interface (200), a decompressor (202), a summation circuit (212), a correlation unit (206) a compressor (214), and an output interface (220). The input interface (200) receives a plurality of audio channels, each being compressed using a lossy compression algorithm. The decompressor (202) decompresses each of the incoming channels to produce a plurality of decompressed audio channels, which are then summed by the summation circuit (212) to produce a summation stream. The compressor (214) compresses the summation stream to produce a first audio output. The correlation unit (206) identifies a dominant channel from the incoming compressed audio channels.

Abstract: A method and system for delivering multimedia data to a user. The multimedia data is stored on a computer network which can consist of a private network or intranet or a public network like the Internet. The multimedia data to be delivered is typically stored in a combination of hypertext markup language (HTML) and audio file formats (WAV, MPEG, AU, etc.) and typically delivered using hypertext transfer protocol (HTTP). The multimedia data contains a visual portion of data such as text and graphics and an audio portion of data such as music. The visual portion of the data is delivered to the user's computer via a computer network. The audio portion of the data is delivered to the user's telephone via a telecommunications network.

Abstract: A video conferencing system (10) for providing enhanced security to a conference room. A local video conferencing system (10) establishes a conference call with a remote video conferencing system. A security system within the local video conferencing system (10) monitors activity in the local video conferencing system's (10) conference room and disconnects the conference call if no activity is detected after a predetermined period of time. If activity is detected after a period of nonactivity, users of the local video conferencing system (10) are alerted that a call is in progress. The users are then given the option of continuing or disconnecting the call. In addition, the security system detects the presence of a black screen or no video signal from the remote video conferencing system and if detected, alerts the users that a call is in progress.

Abstract: A calendar system within a telephone system includes an interface for obtaining information from a calling party, calendar data which includes a calendar for a called party, and an agent. The agent, in response to a request to schedule a telephone call with the called party, searches the calendar of the called party to determine an available slot for the telephone call. After confirming that the available slot for the telephone call is acceptable to the calling party, the agent schedules the telephone call in the available slot of the calendar for the called party.

Abstract: A mechanical keypad is attached to the outside of a protective cover. A switch allows a personal intelligent communicator to be operated either with the user interface of a mobile telephone handset or with a graphical user interface. When the cover is closed, the mechanical keypad serves for dialing telephone numbers. When the cover is open, the mechanical keypad is disabled, and the graphical user interface is revealed. The microphone and speaker of the personal intelligent communicator together operate as a speaker phone when the cover is open, and as a mobile telephone handset when the cover is closed. By use of an override switch, the personal intelligent communicator can also operate as a speaker phone when the cover is open.