Abstract: A mechanism for forward error correction (FEC) coding, suitable for use where multiple payload streams are simultaneously transmitted from end-to-end. Instead of deriving parity information based on payload information carried within a given stream, the invention involves FEC encoding across multiple parallel streams and thereby deriving parallel parity information. The parallel parity information may then be transmitted to the receiving end in parallel with the underlying payload information. Beneficially, the invention can substantially reduce the time it takes for the transmitting end to derive parity information or for the receiving end to receive the information necessary to recover from data loss. The invention is especially suitable for use in IP telephony and particularly for implementation in an IP telephony gateway.

Abstract: A system and method for communicating screen display images on a computer to another computer using a telephony network. A screen share button on a data network telephone initiates a screen shot request to a first computer associated with the data network telephone. The first computer uses a screen shot application to retrieve the image on the computer display and send it to the data network telephone. The data network telephone receives the screen shot image and sends the image to another data network telephone which is a party to a telephone conversation with the first data network telephone. The first data network telephone preferably uses the SIP invite with a session description protocol that permits the transfer of image data on SIP messages. The second data network telephone implements the image data session description protocol and communicates the screen shot image to the second computer.

Abstract: A dial plan for a data network telephony system is provided. A variable-length-code scheme is used to assign communication numeral strings to user identifiers and/or device identifiers, such as SIP URIs (Universal Resource Identifier). The dial plan enables a user to enter a user identifier (or device identifier) into a conventional keypad of a data network telephone, such as an Internet telephony phone. Additionally, embodiments of the disclosed dial plan include techniques for dialing from one data network telephone to another data network telephone utilizing the same service provider, dialing from one data network telephone utilizing a first service provider to a second data network telephone utilizing a second service provider, and dialing from a data network telephone to a PSTN phone and vice versa.

Abstract: Gateway routers for real-time networks have the ability to collect delay, loss, and jitter statistics on a per-connection basis. It is possible to use this information not only to monitor the quality of individual voice calls and other real-time connections, but also to evaluate the overall performance of the underlying network. This paper describes a method for monitoring and managing the performance of a real-time data network that supports voice, video and other real-time services. In the described embodiments, the RTCP mechanisms of RTP for sender and receiver reporting be used to relay performance information to one or more network monitoring sites for analysis and interpretation. In addition, gateway routers are organized and managed within a hierarchy that allows the monitoring function to localize it view of network conditions within the hierarchy; and the monitoring of network performance can occur on various time scales.

Abstract: A system for automatically populating an IP telephony address mapper with mapping information for an IP telephony gateway or other device. A self-registering gateway is configured to automatically provide an address mapper with mapping indicia defining the IP address of the gateway and the telephone numbers served by the gateway. In one embodiment, for instance, the gateway places a PSTN modem-to-modem call to the address mapper and conveys its IP address as a modulated data stream, while the address mapper may determine the telephone number of the gateway through established automatic calling number delivery procedures. Based on the telephone number of the gateway, the address mapper may then refer to a telephone number mapping database to identify other telephone numbers that could be served by the gateway. The address mapper may then update its address mapping data files accordingly.

Abstract: A system and method for providing location information and other information about a calling telephone to the caller during a telephone connection in a data network telephony system. Data network telephones may be provisioned and otherwise configured for operation with an extensive database and other user account information. The user's account may include a location identifier that identifies the physical location of the telephone. The location identifier may provide address information, latitude and longitude configuration, directions, or other information. In one embodiment of the present invention, the location information is communicated in a data communications channel between the caller and the callee. In another embodiment of the present invention, the location information is communicated to the caller during call setup. The embodiments of the present invention are useful in providing emergency dispatch services, such as 911 in a data network telephony system.

Abstract: A system and method for providing call handling services on a data network telephony system. A portable information device, such as a PDA, associated with a user contains the user's address and/or appointment book information. The address and/or appointment book information includes priority information comprising one or more address/appointment entries having associated priority levels. The priority information is transmitted from the portable information device across a link to a data network telephone or other voice communication device, where it is stored in a call handling database. When an incoming call request is received at the data network telephone, the call is handled according to the priority information contained in the call handling database.

Abstract: A high-fidelity voice/audio communication system including a high-fidelity SLIC (HSLIC) device that combines traditional BORSCHT functionality with high fidelity sampling and compression techniques. The HSLIC preferably resides on a single plug-in line card contained within a multi-cards chassis. The line card includes an analog interface that connects to a two-wire subscriber line, a high fidelity codec for sampling the analog signal at a high resolution and converting high rate digital signals to an analog signal, a voice processing client running on a microprocessor and associated digital memory. The high fidelity codec preferably has a sample rate of at least twenty thousand samples per second, and no less that 250 quantization levels.

Abstract: A method is shown for compression of packet header information of packets transmitted on a point to point link. First and second endpoints of the point to point link negotiate a profile for packet header information for packets transmitted from the first endpoint to the second endpoint on the point to point link. The profile includes a predetermined default value for a predetermined header field of the packet header information. A packet sent from the first endpoint to the second endpoint over the point to point link includes a profile identifier for the profile and excludes the predetermined header field. The second endpoint uses the profile identifier to access the profile. The second endpoint then uses the predetermined default value for the predetermined header field from the profile to decode the packet.

Abstract: A system for adjusting or establishing transmission fees based on delay and/or jitter. A network element may evaluate transmission delay and/or jitter over a given transmission system, which may be or is in effect for a particular real-time media transmission. A determination may then be made whether the delay and/or jitter exceeds a predetermined threshold, which may suggest that the transmission quality would be lower than desired. If so, then the fees that a billing entity would charge for the transmission may be decreased in view to account for the lesser quality of service. Conversely, where the delay and/or jitter is lower than a predetermined threshold, then the billing entity may increase its fees accordingly. Other fee schedules based on delay may be established as well.

Abstract: A system and method for providing user mobility services on a data network telephony system. User attributes may be transmitted from a portable information device, such as a personal digital assistant, to a voice communication device, such as an Ethernet-based telephone. The voice communication device receives the user attributes from the portable information device and may transmit a registration request to a registration server. The registration request may include the user attributes, and is used by the registration server to register the user to the voice communication device in a registration data base. When a call is placed to the user, the registration server may reference the registration data base to direct the call to the voice communication device.

Abstract: A method and apparatus for measurement-based conformance testing of service level agreements in networks. The method includes first collecting quality of service information from network traffic over a plurality of network nodes. Then, the collected quality of service information is compared to a plurality of specified quality of service levels. A plurality of possible virtual quality of service pathways through a plurality of network nodes is provided, based on the compared quality of service information. One embodiment of the method includes the additional step of creating a virtual connection using the compared quality of service information. In another embodiment of the method, the step of collecting quality of service information from network traffic over a plurality of network nodes includes first transmitting test traffic from a source to a destination over a plurality of network nodes.

Abstract: A system for dynamically jitter buffering a sequence of packets based on substantially synchronized time signals maintained at the transmitting and receiving ends of a communication system. The time signals may be synchronized, for instance, by equipping both the transmitting and receiving ends with global positioning system receivers. The transmitting end may mark each outgoing packet with a sender-time based on the time signal at the transmitting end. Dynamic jitter buffering may then be provided by scheduling delayed play-out of each packet. For instance, the jitter buffer at the receiving end may be configured to delay play-out of each packet until the time signal at the receiving end indicates a time that is substantially a predetermined end-to-end delay period after the sender-time for the packet.

Abstract: A toll saving method and apparatus for a remote access system is provided. A first communication device receives a connection request from a second communication device. The first communication device then decodes caller identification information from the connection request to determine a user identity, without completing the connection request. The first communication device determines whether an event associated with the user identity has occurred. If the event associated with the user identity has not occurred, the connection request is not completed. In a preferred embodiment, the toll saving method and apparatus allows an email user to call an ISP to determine whether unaccessed email is present in the user's account. If no email is present in the user's account, the call is never answered and no toll charges are incurred.

Abstract: A method for improving perception of electronic content from a computer network such as the Internet or an intranet. Network latencies and the type of electronic content such as text, graphical images, animation, voice, video and other electronic content interact to influence user perception of the quality of information provided. As network latency increases and becomes more variable, users typically become less satisfied. The method dynamically adjusts the amount of electronic content presented to user based on a determined network latency. The amount of electronic content is also adjusted progressively and underlying transport protocol such as Transmission Control Protocol (“TCP”) and User Datagram Protocol (“UDP”) are adaptively adjusted based on the type of electronic content requested (e.g., TCP for text, UDP for graphical images, etc.).

Abstract: A complete collision avoidance contention resolution protocol is provided for networks with stations communicating over a shared access communication medium. The protocol is used where a physical layer connected to the communication medium uses a carrier modulation scheme with multiple frequencies. Stations on the network that desire access to the communication medium use a carrier modulation scheme with multiple frequencies. Stations that desire access simultaneously transmit a single frequency selected at random from a set of physical layer frequencies during an open-contention-interval. At the same time, the stations listen to a combined frequency signal and analyzes it for frequency content. If one station whose own frequency matches the highest frequency signal in the combined signal, the station gains access to a communication medium. If more than one station have the same highest frequency signal in the combined signal, no station gains access to the medium.