Abstract: A multi-media terminal adapter couples to a network access module over a communication link. The network access module communicates IP frames over a frame switched network. The multi-media terminal adapter comprises a wide area network interface coupled to the communication link for exchanging IP frames with the access module and a local area network interface for receiving outbound data client IP frames from each of a plurality of data clients. Each outbound data client IP frame comprises a socket information that includes a local area network IP address and a data client port number. A VoIP module generates outbound VoIP frames. Each outbound VoIP frame comprises socket information that includes a VoIP port number selected from a first group of port numbers exclusively reserved for use by the VoIP module. A router module is coupled between the wide area network interface and each of the VoIP module and the local area network interface.

Abstract: A video mail server comprises a video call signaling module coupled to an internet protocol network for establishing a first internet protocol channel with a caller remote internet video device to support a recording session over the internet protocol network and establishing a second internet protocol channel with a user remote internet video device to support a playback session over the internet protocol network. A media interface comprises a recording module for obtaining a recording sequence of compressed images representing motion video images from the caller remote internet video device and storing a video mail file representing the recording sequence of compressed images. Each compressed image frame is one of: i) an independent frame from which an image frame can be recovered utilizing only the independent frame; and ii) a dependent frame from which the image frame can only be recovered utilizing both the dependent frame and an independent frame preceding the dependent frame in the sequence.

Abstract: An IP telephony gateway operates in conjunction with a base station of a hybrid micro cellular wireless telephone network. The base station supports roaming of a wireless telephone into a cell controlled by the base station. The IP telephony gateway comprises: i) a circuit switched network interface for exchanging system messaging, call signaling, and audio media with a circuit switched system controller; ii) a base station network interface for exchanging system messaging, call signaling, and audio media with a base station; and iii) a VoIP client for exchanging system messaging, call signaling and audio media with a remote VoIP device over a packet switched network. A management module interfaces messages exchanged with the base station and each of the circuit switched system controller and the soft switch to support roaming of the wireless telephone into the cell controlled by the base station.

Abstract: A method of diverting a hunting function of a PSTN central office switch is provided. The method diverts a hunting function in the event a channel within a hunting group is available but a call can not be completed. The method comprises detecting a status of a plurality of user channels to determine whether all user channels associated with the hunting group are in use and providing a hunting diversion message to the PSTN central office switch in response to determining that all user channels associated with the hunting group are in use. The method also comprises detecting at least one user channel associated with the hunting group is available for an inbound call and providing a hunting enable message to the PSTN central office switch in response to determining that at least one user channel associated with the hunting group is available for an inbound call.

Abstract: A network access module interconnects a stand alone multi-media terminal adapter with a network controller of a frame switched network. The network access module comprises a frame switched network interface coupled to the frame switched network for communicating with the network controller. The network access module further comprises a communication link interface for communicating with the stand alone-multi media terminal adapter. A service flow module is coupled to the frame switched network interface and coupled to the communication link interface. The service flow module receives a plurality of frames of IP traffic from the multi-media terminal adapter and sorts the frames such that each frame is delivered to the frame switched network interface at a time that corresponds to a time division logical channel which corresponds to the frame. A QoS module is coupled to the service flow module and coupled to communication link interface.

Abstract: A stand-alone multi-media terminal adapter controls a dynamic quality of service management system of a broad band network access module. The multi-media terminal adapter provides the dynamic quality of service management system with instructions to reserve, commit, and release time division logical channels on a broad band network as well as discrimination identification to be used by the network access module for identifying IP traffic that corresponds to a time division logical channel. The multi-media terminal adapter receives acknowledgement of a time division logical channel that comprises identification of a frame frequency and a frame size. The multi-media terminal adapter encapsulates compressed digital audio data representing a VoIP session into IP frames with a frame size, frame frequency, and discrimination identification that corresponds to the time division logical channel.

Abstract: A tunneling address translation engine enables a Internet protocol packet audio conversation to be maintained between two device. At least one of the devices may be behind a firewall or behind a network address translator, or proxy server. The tunneling address translation engine routes frames of audio data to an intended destination utilizing unreliable data protocols which do not include source address information. An audio conversation may be maintained through a sequence of tunneling address translation engines.

Abstract: A real time streaming media session client operates with a communication service provider system. The client generates a session signaling message to the system identifying an endpoint and receives a response message. The response message includes an endpoint address for the real time streaming media session. The client commences a real time streaming media session with destination address. The client periodically sends session status responses to the system in response to session status requests sent by the system. The client also terminates the session and reports the session termination time along with a session identifier to the system in response to a call detail record backup request by the system.

Abstract: A communication service provider accounting server is used to track communication session usage by a client of the service provider. The accounting server receives an indication identifying a client that has commenced a communication session and an indication identifying that the client has terminated the communication session. The accounting server further provides periodic status request messages to the client and receives a sequence of status response messages in response thereto from the client. The accounting server determines a session termination time as the earlier of receipt of the indication that the client has terminated the communication session and making a determination that the sequence of status messages have ceased.

Abstract: The present invention relates to displaying caller ID information, and more specifically, to displaying caller ID information in an environment with broadband telephony service. The invention provides a method and apparatus for receiving caller ID information from a service provider and providing the caller ID information for display as an overlay over a television programming image provided by the service provider and displayed on a television.

Abstract: The improved AES processing method provides an efficient alternative to both Mips intensive multiplication and to conventional table lookup, used to multiply terms over a Galois field (GF). The improved method takes advantage of the fact that in the GF, any non zero element X can be represented by a power of a primitive element P. The improved method thereby results in a 2 by 256 table. The log base P of the terms being multiplied are looked up and summed, and the anti-log of the sum is looked up in the same table.

Abstract: A microphone for a full duplex speaker phone comprises a loudspeaker, a uni-directional microphone, a omni-directional microphone, and a background noise suppression circuit. The uni-directional microphone is a directional microphone such that the intensity of an acoustic source detected at the uni-directional microphone is dependent on the angular direction of the acoustic source relative to a reference direction. The omni-directional microphone is a non-directional microphone such that the intensity of the acoustic source detected at the omni-directional microphone is substantially independent of the angular direction of the acoustic source relative to the reference direction. The background noise suppression circuit determines the direction of the acoustic source by calculating a quotient between a signal strength of the acoustic source as detected by the two microphones and provides for suppressing an acoustic source that is not within a threshold angle of the reference angle.

Abstract: A directory server provides a media session channel for communication of real time streaming media data from a remote client to a client served by an address translation firewall. The directory server includes a client registration module for receiving a registration datagram originated by the client, source network address and a source port number from the registration datagram, and providing a session signaling message from the remote client to the client utilizing the extracted source network address and source port number.

Abstract: A device provides for sending datagrams representing real time streaming media frames to a client independent of whether the client is served by a network address proxy. The device receives a datagram originated by the client that includes an indicated network address and an indicated port number for receipt of the datagrams representing real time streaming media frames, extracts a source network address and a source port number from the datagram originated by the client, and compares the indicated network address to the source network address. The device addresses the datagrams representing real time streaming media frames to the source network address and source port number if the indicated network address and the source network address are not the same.

Abstract: A method for communicating real time media data between a first client and a second client across a packet switched data network is provided. The method includes receiving an indication of a first client network address for use as a destination network address for sending media datagrams to the first client. A media datagram originated by the first client is also received. A comparison between the first client network address and a source network address extracted from the media datagram originated by the first client is made. A media datagram is sent to the first client using the source network address if the source network address and the first client network receiving address are not the same and the media datagram is sent to the first client using the first client network address if the source network address and the first client network address are the same.

Abstract: A system and method enable wireless communication from one data processing system to a telephone network. The system and method establish a connection from a local transceiver to a base transceiver using echo cancellation techniques to remove various tones and noise. The system and method enable full duplex communication between the local transceiver and the base transceiver by using forward error correction algorithms to encode and decode messages therebetween, and by using an interleaving algorithm to protect against burst type errors.

Abstract: A method and apparatus for improving burst error correction. The method and apparatus involve transmitting block codes with interleaving, and predicting a burst error in the received block codes. Burst error prediction includes decoding received block codes and determining if the decoding is successful. The method includes receiving a code having an error correction capability, an erasure detection capability and a burst error length associated with it. A decoding step is performed to find an error in the code. The error has a length associated with it. The decoding is successful if the error length found is less than twice the error correction capability. If an initial decode attempt is not successful, then a burst error length is associated with the code and another decode is performed. The burst error length has an initial value that is less than the error correction capability.

Abstract: A computationally efficient method and apparatus for motion estimation by producing accurate motion vectors with minimal computational effort. A preferred embodiment of the present invention first identifies an approximate match between a reference frame and a current frame of video data. Once an approximate match is found, the method performs at least two searches at a finer pixel level, until a motion estimate is reached.

Abstract: A data signal compression technique for real-time voice and data processing where the digitized signal is first compressed to obtain a first compressed signal, and the first compressed signal is then compressed again to obtain a second compressed signal. Within a digital signal processor, digital signals first undergo time scale compression after which the compressed signals undergo audio compression to achieve multiple-compressed signals. Upon reception in a second digital signal processor, the multiple-compressed signals are correspondingly decompressed to achieve a high-quality estimation of the original digital signals.

Abstract: A method and system for packet-switched voice communication between two remote points over a randomly connected network such as the Internet includes use of a first index to serialize the outgoing packets, and a return index to confirm receipt. In the event the return index indicates an unacceptable number of packets have not been received at their destination within a limited time, the system causes at least some of the lost packets to be retransmitted, allowing a more complete recreation of the sender's voice of the random connection.