Abstract: A personal information device (PID) is coupled to an IP Telephony phone in order to provide end-to-end connectivity to another PID through a network. The architecture disclosed includes a pair of internet-enabled phones that are able to establish a call session using a Session Initiation Protocol (SIP) and a Session Description Protocol (SDP). Each phone is also provided with an interface configured to communicate with a PID. Each PID is registered to a corresponding internet-enabled phone using each PID user's SIP URL. The user of a first PID connected to a first phone requests a call to a SIP URL corresponding to the user of the second PID that is connected to a second phone. The SIP URL for the user of the second PID is resolved to the network address of the second phone and connection is established between the first and second phones. The connection includes a media stream for transferring data between each of the PIDs.

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 for increasing call capacity of an internet telephony system in an emergency mode, such as where there is a surge of call requests in a particular region. A management system in the internet telephony system may detect the presence of an actual or potential state of congestion of call traffic and/or signaling traffic. In order to admit a higher rate of call traffic to the telephony system, action is taken to reduce the workload of the signaling system, in order to allow the telephony system to focus more on serving call requests. For example, the signaling system may reduce the rate of signaling request messages processed with respect to requested and ongoing calls and/or reduce the amount of processing involved in handling such messages.

Abstract: A method and apparatus for increasing call capacity in a packet switched telephony system in an emergency mode. The telephony system includes a management system, which detects an emergency, such as a surge in incoming media streams, a surge in traffic elsewhere in the system, a potential surge in traffic, and/or a convention emergency situation such as a natural disaster that may cause a surge in traffic. The management system alerts an internet telephony machine, such as a gateway, to the congestion problem. The internet telephony machine responds by switching to a lower bandwidth coder, thereby outputting each incoming media stream at a lower output rate, and thereby effectively increasing the capacity of the receive and handle calls.

Abstract: A method and system for assigning priority or classes of service of messages delivered in a packet-based network. The method and system allows for implementation of differentiated classes of service according to the requirements of the network application or user.

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 system and method for using a portable information device, such as a personal digital assistant, to establish a conference call on a telephony network. In one embodiment, a user of the portable information device may select communication partners from an address book application located in the portable information device. Communication information related to the communication partners is transmitted across a link from the portable information device to a voice communication device, such as a data network telephone. The data network telephone may then set up the conference call.

Abstract: A method and apparatus for communicating a real time media input over a network. The apparatus encodes the input into data packets having a number of frames ordered according to a first variable. A receiving device unpacks and buffers the unpacked data packets for playout according to a second variable. The receiving device generates utility parameters for evaluating a dynamic characteristic of the network that transports the data packets. The receiving device selects a preferred utility parameter and adjusts the first and second variable according to the selected utility parameter. The method includes encoding an analog input into data packets that are transported to a receiving device. The method also includes unpacking the data packets, buffering the unpacked data packets according to a second variable, and generating at least two utility parameters that represent a dynamic characteristic of a network.

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 method and apparatus for communicating a real time media input over a network. The apparatus encodes the input into data packets having a number of frames ordered according to a first variable. A receiving device unpacks and buffers the unpacked data packets for playout according to a second variable. The receiving device generates utility parameters for evaluating a dynamic characteristic of the network that transports the data packets. The receiving device selects a preferred utility parameter and adjusts the first and second variable according to the selected utility parameter. The method includes encoding an analog input into data packets that are transported to a receiving device. The method also includes unpacking the data packets, buffering the unpacked data packets according to a second variable, and generating at least two utility parameters that represent a dynamic characteristic of a network.

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 system and method for receiving a transported stream of data packets includes a buffer management device for receiving the data packets, unpacking the data packets, and forwarding a stream of data frames. The system and method further includes a first jitter buffer for receiving the data frames from the buffer management device and buffering the data frames, and a second jitter buffer for receiving the data frames from the buffer management device and buffering the data frames. In addition, the system and method includes a computationally-desirable jitter buffer selected from the first jitter buffer or the second jitter buffer by comparing a first jitter buffer quality and a second jitter buffer quality. The system and method also includes a decoder for receiving buffered data frames from the computationally-desirable jitter buffer.

Abstract: A system and method for an Options Based Address Reuse (“OBAR”) is provided. The system includes a first network having a host device with a host address and a host port, and a second network external to the first network and having a server device. The also comprises a combination network address for identifying the host device to the server device. The combination network address includes the host address, the host port, and an option. Similarly, the method involves sending a packet from the host device to the server device through a router, and adding the option to the packet.

Abstract: A method and apparatus for communication system buffer size and error correction coding selection. A method includes the steps of receiving a stream of data packets by a real time receiver that includes a buffer management device, a first plurality of jitter buffers, and a second plurality jitter buffers. The first and second plurality of jitter buffers are evaluated and a first and a second optimal jitter buffer is chosen. The first and the second optimal jitter buffer has an associated conditional optimal performance characteristic. The conditional characteristics are compared and a preferred buffer of the receiver is selected. The apparatus includes a receiving device including a first set of jitter buffers and a second set of jitter buffers with error coding. The first set includes a plurality of buffers and a second plurality of buffers maintained in the second set of buffers.

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.