AT&T Labs Research — Leading Invention, Driving Innovation

The present invention provides a system and method for providing handwritten messages with electronic communication devices. For example the electronic communication device may be a mobile communication device and may include a touch screen, screen buffer memory, processor, and transceiver. In preferred embodiments, the mobile communication device may be a personal digital assistant (PDA), a two-way pager, or a mobile telephone. The mobile communication device user may create a handwritten message on the touch screen using, for example, a stylus. In one exemplary embodiment, the handwritten message displayed on the touch screen may be stored to the buffer memory as an image, directed by the processor to the transceiver, and transmitted via a wireless connection to, for example, a wireless base station and the Internet. The Internet may route the handwritten message image to a desired Internet address. The handwritten message may be displayed in real time on the recipient's display, e.g., an instant message (IM) session, or stored in their email mailbox for later retrieval and display. The recipient may type in a text message in response to the hand written message, and this message will be transmitted to mobile communication device and displayed on the mobile communication device. In another embodiment, the receiving party may also have handwritten messaging capabilities (included with, for example, a PC, PDA or a mobile phone) and may respond with a handwritten message.

A personal communications device may be placed in, or removed from, a docking station. In response, a predetermined command may be selected from a plurality of predetermined commands and transmitted from the docking station to at least one of a plurality of network devices. Each predetermined command may include the address of the network device and may be configured to cause the network device to respond. In this way, a user's communication infrastructure may be reconfigured in a simple and intuitive manner.

The present invention provides a system and method for providing handwritten messages with electronic communication devices. For example the electronic communication device may be a mobile communication device and may include a touch screen, screen buffer memory, processor, and transceiver. In preferred embodiments, the mobile communication device may be a personal digital assistant (PDA), a two-way pager, or a mobile telephone. The mobile communication device user may create a handwritten message on the touch screen using, for example, a stylus. In one exemplary embodiment, the handwritten message displayed on the touch screen may be stored to the buffer memory as an image, directed by the processor to the transceiver, and transmitted via a wireless connection to, for example, a wireless base station and the Internet. The Internet may route the handwritten message image to a desired Internet address. The handwritten message may be displayed in real time on the recipient's display, e.g., an instant message (IM) session, or stored in their email mailbox for later retrieval and display. The recipient may type in a text message in response to the hand written message, and this message will be transmitted to mobile communication device and displayed on the mobile communication device. In another embodiment, the receiving party may also have handwritten messaging capabilities (included with, for example, a PC, PDA or a mobile phone) and may respond with a handwritten message.

A personal communications device may be placed in, or removed from, a docking station. In response, a predetermined command may be selected from a plurality of predetermined commands and transmitted from the docking station to at least one of a plurality of network devices. Each predetermined command may include the address of the network device and may be configured to cause the network device to respond. In this way, a user's communication infrastructure may be reconfigured in a simple and intuitive manner.

The present invention enables a party which is placed on hold in a first communication session to enter a second communication session and continue to monitor the status of the first communication. For example, in a broadband communication system telephone call or multimedia call, two parties may be in an active communication session with one another and a first of the two parties to the call may receive another incoming call. The first party then places the second party on hold. At this time the second party may wish to place the communication session with the first party on hold to initiate another communication session with a third party. However, the second party would also like to know when the first party takes the communication session off hold so that they can resume their communication. As a result, the second party places the communication on monitor hold. While the communication is on monitor hold, the second party is able to monitor the on hold communication session to recognize when the first party returns. For example, the monitored call on hold may be maintained at a reduced volume or a video display of the video in a multimedia communication session may be displayed in a reduced window size on a display. Further, the present invention provides prompts to various parties of a communication session when one or more parties to the communication session are placed on hold or taken off hold. For example, the system may generate a prompt to one or more parties when a communication session is placed on hold or monitor hold, or when a communication session on hold or monitor hold is taken off hold.

Methods and systems for routing information in an integrated global communications network comprising a signalling system interconnecting a source router, a plurality of intermediate routers, a destination router and a routing processor for routing the multimedia information. The source router receives the information including the routing requirements associated with the information and transmits a routing query signal to the routing processor. The routing processor evaluates the routing requirements, determines which routers and communication paths are capable and available to route the information, evaluates the statistical availability of such routers and selects an optimal routing path comprising select ones of the available routers and communication paths.

In a method for reducing latency in packet telephony caused by buffering at the conversion stage between analog audio signals and digital audio data, analog audio is sampled at a rate far greater than necessary for telephony. The increased sampling rate allows the audio data to pass much more rapidly through the data conversion buffer. After passing through the buffer, the data is downsampled to a rate normally used for telephony. To handle audio data for speaker output, the data is upsampled to a rate far in excess of the rate necessary for processing telephony-grade voice signals. The increased sampling rate allows the audio data to pass much more rapidly through the data conversion buffer. After passing through the buffer, the data is converted into an analog audio signal for sending to the speaker. In this way, latency due to the buffering that accompanies the process of converting audio signals to digital data, or vice versa, is minimized.

A method and system are disclosed for updating announcements on a packet network site such as a website. An exemplary method includes receiving an announcement through an audio interface and storing the announcement on a website server. In accordance with the exemplary method, merchants without Internet access can now update announcements on their websites quickly and economically by using an audio interface such as a conventional telephone. The updated announcements may be either audio or data. If the announcement received in the exemplary method is an audio announcement, it is converted from a first audio format into a second audio format prior to the step of storing the announcement. Whether the announcements are audio or data, the merchants' customers can then retrieve them by accessing the merchants' websites via the Internet.

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. The present invention also verifies out-of-area access codes that are directly input by the calling party to ensure their accuracy and corrects any incorrect access codes input by the caller.

A method and apparatus that provide enhanced directory assistance information. 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.

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.

A method and system for accessing a remote system via a computer connected to a packet data network, comprises a Packet Data Network Gateway (PDNG) is connected to the packet data network. The PDNG receives a signal representing a DTMF tone transmitted from the computer, generates the DTMF tone at corresponding to the received signal, and transmits the generated DTMF tone to the remote system. The PDNG receives a response from the remote system and creates at least one voice packet for transmission to the computer over the packet data network. The remote system may be a voice messaging system or an interactive voice response (IVR) system.

The present invention provides an apparatus and method that allows the communication mode for communicating between two user devices to be selected based on the abilities of the user devices and/or the preferences of the users. Using the present invention, when an operator of a user device enters the destination address for another user device with which communication is sought, a Client Routing Assessment Profile Services (CRAPS) processor receives the signals from the source user device. The signals include information of the identity of the source user device and the destination user device. The CRAPS processor retrieves profile information from a database corresponding to the identity of the source user device and/or the destination user device. Based on the profile information retrieved, the CRAPS processor sends boot signals to the user devices involved in the communication instructing them to change to the appropriate communication mode. Once the user devices are operating in the same mode, the CRAPS processor enables the communication between the user devices until an end condition occurs.

A method and a system for routing data calls in which an incoming data call having a destination number is received. The data call is routed to a first destination over a circuit-switched telecommunications network based on the destination number for the data call. Transaction information is then sent to the first destination. Alternatively, the incoming data call is received at an input port. The incoming data call is associated with an automatic number identification and the input port is associated with an input port identification. The automatic number identification is stored in a database in association with the input port identification. A transfer message for the data call is received from the first destination. The transfer message contains information relating to an address for a second destination. The information relating to the address for the second destination is stored in the database in association with the automatic number identification and input port identification for the data call. The data call is disconnected from the first destination, and the data call is routed to the second destination over the circuit-switched telecommunications network.

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.

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. In yet another alternative embodiment, the amount of latency jitter in the packet network is measured and the size of the telephony input buffer is adjusted based upon the relative amount of jitter, such that the relative size of the buffer is reduced when the packet network is quiet, and the size of the buffer is increased when the network is relatively jittery.

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.

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.

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.

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.

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.

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. The present invention also verifies out-of-area access codes that are directly input by the calling party to ensure their accuracy and corrects any incorrect access codes input by the caller.

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.

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.

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.

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.

A method and apparatus for recording and indexing audio information exchanged during an audio conference call, or video, audio and data information exchanged during a multimedia conference. For a multimedia conference, the method and apparatus utilize the voice activated switching functionality of a multipoint control unit (MCU) to provide a video signal, which is input to the MCU from a workstation from which an audio signal is detected, to each of the other workstations participating in the conference. A workstation and/or participant-identifying signal generated by the multipoint control unit is stored, together or in correspondence with the audio signal and video information, for subsequent ready retrieval of the stored multimedia information. For an audio conference, a computer is connected to an audio bridge for recording the audio information along with an identification signal for correlating each conference participant with that participant's statements.