Abstract: A broker application server for facilitating communication between one or more senders and one or more receivers over a digital packet network. Each receiver receives data in a preferred format. Information identifying the preferred data format of each receiver is stored in a database in the broker application server. The broker application server receives digital packets addressed to a receiver. It extracts data and address information from the packet and optionally decrypts the data. It examines the data to identify the data format. If the data is in the preferred format of the addressed receiver, the broker application server optionally encrypts the data, formats it into a packet, and transmits it to the appropriate receiver.

Abstract: A system and method is disclosed for automatically answering and playing a desired telephone call from a desired calling party. In one embodiment, the called telephone is a cellular phone with a speaker or headset for use in a car. The called party is located and local traffic conditions are report to the called party over the speaker or headset of the called party's telephone which automatically answers the incoming call as a desired phone call. In other embodiments, information requested comprises concierge services, reminder services or intermittent and/or periodic announcements of desired information.

Abstract: In a method for reducing latency in packet telephony caused by anti-jitter buffering, audio data elements are received and placed in a telephony input buffer used for anti-jitter buffering. Rather than wait until the buffer is full, the audio data elements are clocked, or played, out of the buffer at a rate slower than the normal play rate. In this way, latency due to the initial buffer fill period is reduced or eliminated. Audio data elements continue to be played out at a slower than normal rate until the buffer fill level reaches a threshold. At that time, the play rate for sending data elements out of the telephony input buffer is adjusted to the normal play rate. In an alternative embodiment of the present invention, the fill level of the telephony input buffer is controlled within a desired range by speeding up or slowing down the rate at which audio data elements are played out of the telephony input buffer.

Abstract: An operator receives a request for directory information about a listed party over a telephone connection. It is then determined if enhanced directory information for the listed party is available in an enhanced information database. The enhanced directory information associated with the listed party is retrieved, from the enhanced information database, as a script that incorporates the enhanced directory information. The enhanced directory information can then be provided using the script. The enhanced directory information can be stored, for example, on a Web server that is updated by the listed party and the information can be transferred using a Phone Markup Language script.

Abstract: A digital data packet transmission process and system provides more efficient and higher quality of service in applications such as Internet telephony. In one aspect of this approach, transmission control protocol (“TCP”) is used to send data from a first user or client over standard telephone lines to a local Internet service provider (“ISP”). At the ISP, the data packets are converted from TCP to user datagram protocol (“UDP”). The UDP packets are then transmitted, typically over a higher bandwidth link to another local ISP serving the recipient. The UDP packets are translated back to TCP packets and routed to the receiver. Because many existing systems currently employ UDP packets, the present approach is largely backwards compatible should a recipient be hooked up to an ISP that does not employ a TCP/UDP converter. A bidirectional TCP/UDP converter is preferable for two way communication such as Internet telephony.

Abstract: A system and method for establishing a communication path for a call between a video telephone/teleconference call and a packet network telephone terminal through a packet network. The call is routed through a multimedia gateway which performs a conversion process between a video telephone/teleconference domain and a packet network telephone domain.

Abstract: A server platform in the PSTN performs the service of calling the subscriber if the subscriber's customary commuting route is congested. The service makes an initial call when the fastest commute route is congested. It also makes additional calls each time it finds a better alternate route. The server can accommodate any subscriber commuting pattern. If a street on the subscriber's customary route is found to be congested, then the server sequentially tests additional streets having geographic coordinates nearby, dynamically searching for an alternate route that has an acceptable level of congestion. The server has a voice response unit (VRU) in the platform, to announce to the subscriber that the subscriber's customary commuting route is congested. Following an initial call to the subscriber, the server continues to search for better alternate routes having the least congestion and it calls the subscriber again if it finds a better alternate route.

Abstract: The invention provides a network paging system that includes a method and apparatus for sending messages to a subscriber. When one party attempts to contact another party, via a page message for example, the system may determine if the paged party is logged onto the network system. If the paged party is logged onto the network, the page message may be sent to the paged party's personal computer. If the paged party is not logged onto the network, the page message may be sent to the paged party's pager. Advantageously, the invention eliminates the need or expense of sending a message both via the Internet and a pager.

Abstract: A digital data packet transmission process and system provides more efficient and higher quality of service in applications such as Internet telephony. In one aspect of this approach, transmission control protocol (“TCP”) is used to send data from a first user or client over standard telephone lines to a local Internet service provider (“ISP”). At the ISP, the data packets are converted from TCP to user datagram protocol (“UDP”). The UDP packets are then transmitted, typically over a higher bandwidth link to another local ISP serving the recipient. The UDP packets are translated back to TCP packets and routed to the receiver. Because many existing systems currently employ UDP packets, the present approach is largely backwards compatible should a recipient be hooked up to an ISP that does not employ a TCP/UDP converter. A bidirectional TCP/UDP converter is preferable for two way communication such as Internet telephony.

Abstract: An apparatus and method is provided for placing a phone call when dialing out of the caller's local area. A database resident in a telephone connection mechanism contains the access codes required to place the out-of-local area call. A user input device is utilized by the calling party to retrieve the out-of-area access codes for the called party's telephone from the database. The user input device is also utilized by the calling party to input the called party's local telephone number into the telephone connection mechanism. The telephone connection mechanism automatically combines the local calling number input by the user with the out-of-area access codes retrieved from the database. The telephone connection mechanism then transmits a telephone connection request by utilizing the combined local telephone number and the out-of-area access codes.

Abstract: A billing method and server for gathering and processing billing information for Internet telephony connections. Originating and terminating clients and Internet Service Providers ("ISPs") are identified by the server from information contained in digital packets. The digital packets also identify the beginning and ending time of the connection, as well as enhancements chosen for the particular connection, and choices of information to be displayed during the connection. The billing server transmits selected information to each client during the connection, such information including packet routing, packet density, elapsed time of connection, and available and selected enhancements. The server constructs a billing detail for each connection and constructs a billing record for each ISP and client over a specific time period. The server periodically forwards each billing record to its associated ISP.

Abstract: A digital data packet transmission process and system provides more efficient and higher quality of service in applications such as Internet telephony. In one aspect of this approach, transmission control protocol ("TCP") is used to send data from a first user or client over standard telephone lines to a local Internet service provider ("ISP"). At the ISP, the data packets are converted from TCP to user datagram protocol ("UDP"). The UDP packets are then transmitted, typically over a higher bandwidth link to another local ISP serving the recipient. The UDP packets are translated back to TCP packets and routed to the receiver. Because many existing systems currently employ UDP packets, the present approach is largely backwards compatible should a recipient be hooked up to an ISP that does not employ a TCP/UDP converter. A bidirectional TCP/UDP converter is preferable for two way communication such as Internet telephony.

Abstract: An enduser at a POTS analog voice-only endpoint (136) and endusers at H.320 standard multimedia terminals (101, 102, 103, 104), which each communicate over separate voice, video and data streams, engage in a videoconference with each other in a pseudo multimedia manner through a central platform (135) that provides call conversion capabilities. A document to be shared by a user at the POTS endpoint with users at the multimedia endpoints is transmitted as a data signal from a facsimile machine (137) or PC terminal (138) associated with the POTS user to a server (146) in the platform. The received data signal is then inputted to a multimedia bridge (124) and transmitted on the data stream to each multimedia endpoint for display on a window on each multimedia terminal. Similarly, a document to be shared by a multimedia endpoint is transmitted on a data stream to the multimedia bridge, where it is bridged on the data stream transmitted to the other multimedia endpoints and to the server.

Abstract: A method and a system for accumulating charges for telecommunications services used during a calling card call, whether the calling card is periodically billed or prepaid. According to the invention, call information detail information, such as a user identification, a beginning time of the call, and setup parameters of the call, is received for a calling-card call. Services rate information is received for telecommunications services used during the calling-card call. A depletion rate of an available funds account associated with the calling-card call is then determined in real time, and the available funds account is depleted at the determined depletion rate during the calling-card call, also in real time.

Abstract: Disclosed is a networked multimedia information system which may be utilized to record, store and distribute multimedia presentations together with any supplemental materials that may be referenced during the presentation. The recorded presentation, together with the associated supplemental materials, may be simultaneously presented on a display containing two separate viewing windows. The effects of network congestion are minimized by prefetching audio and video data for storage in audio and video buffers. An adaptive control algorithm compensates for network congestion by dynamically varying the rate at which video frames are retrieved over the network, in response to network traffic conditions. The audio playback speed is reduced if the audio data does not arrive fast enough over the network to maintain the desired size of the audio buffer after the amount of video data transmitted across the network has been reduced to a minimum value.

Abstract: A method and system is disclosed for handling multimedia messages when the intended recipient is unavailable. The telecommunication based method and system allows a party to leave a greeting message which will be furnished to a caller when the party is unavailable. The method and system further allows a caller to leave a multimedia message for the unavailable intended recipient. The method and system further alerts the party that a multimedia message has been recorded and allows the party to retrieve the multimedia message.

Abstract: Internet access is provided to a multimedia terminal (101, 102, 103, 104) supporting the H.320 protocol through a gateway server (125) that is connected to the Internet. The data stream of the H.320 bearer channel (ISDN phone line or switched 56 kbps facilities) is used for the exchange of data between the accessing terminal and the gateway server, which outputs and receives such data onto and from the Internet, respectively. The gateway server is implemented to function to directly exchange on the H.320 data stream information requests and responsive data in the TCP/IP format used for Internet transport. Alternatively, the gateway server is implemented to function as a proxy to which requests are passed from the terminal over the H.320 data stream, and where function calls are made and locally converted to TCP/IP format for transport onto the Internet.

Abstract: Disclosed is a networked multimedia information system which may be utilized to record, store and distribute multimedia presentations together with any supplemental materials that may be referenced during the presentation. The recorded presentation, together with the associated supplemental materials, may be simultaneously presented on a display containing two separate viewing windows. The effects of network congestion are minimized by prefetching audio and video data for storage in audio and video buffers. An adaptive control algorithm compensates for network congestion by dynamically varying the rate at which video frames are retrieved over the network, in response to network traffic conditions. The audio playback speed is reduced if the audio data does not arrive fast enough over the network to maintain the desired size of the audio buffer after the amount of video data transmitted across the network has been reduced to a minimum value.