Abstract: A system for progressively synchronizing stored copies of indexed media transmitted between nodes on a network. The system includes a transmitter at the sending node configured to progressively transmit available indexed media to a receiving node with a packet size and packetization interval sufficient to enable the near real-time rendering of the indexed media, wherein the near real-time rendering of the indexed media provides a recipient with an experience of reviewing the transmitted media live. The system also includes a receiver at the receiving node that progressively receives the transmitted indexed media and continually notes any indexed media that is not already locally stored at the receiving node. The receiver also continually generates and transmits to the sending node requests as needed for the noted indexed media. In response, the transmitter at the sending node transmits the noted indexed media to the receiving node.

Abstract: A method of automatically establishing an input language for a handheld electronic device includes receiving a request to initiate the composition of a new message, receiving a selection of a particular contact that is the intended recipient of the message, and determining whether the particular contact has a stored preferred input language. If a preferred input language is stored, the method includes determining whether the preferred input language is different than the current input language (which may be a pre-established default input language). If the preferred input language is different, the method further includes switching the current input language to the preferred input language. Also provided is an improved handheld electronic device having one or more routines adapted to implement the method. Also provided is a method of indicating a preferred language in association with a call received by or placed to the device.

Abstract: A system and method for performing echo suppression on a server in browser-based online audio conferences without downloading or installing software on a participant's computing device is disclosed. Streams of audio communication data from the participants in an audio conference are received at the server. An echo suppression application determines the first party that speaks by analyzing the streams to locate speech data, and assigns that party as the “owner” of the audio channel. The speech data is sent to the other participants in the conference. The application then determines whether newly received audio from the owner of the channel is new speech; if so, then the party remains the owner of the channel, and the new speech data is also sent to the other parties in the conference. The channel is surrendered if no new speech is received from the owner in a defined period, and the next party that speaks becomes the new owner of the channel. The other audio data from the participants is replaced by silence.

Abstract: Architecture for accessing and purchasing multimedia services in a mobile terminal cellular environment. The architecture brings together the on-the-fly discovery of music and connectivity for the purchase or preview of multimedia services (e.g., music tracks). Provided is a music client on a cell phone or other mobile communications device that facilitates the discovery, storage, and play of multimedia content or portions thereof such as music. A user can listen to music transmitted to the mobile terminal device (e.g., cell phone) via broadcast, satellite radio, streaming internet radio, etc., review the music, and buy tracks or whole albums over-the-air via the mobile terminal.

Abstract: VoIP packets as many as the number transmittable within a time period equal to a window T1 minus a window T2 are transmitted to a radio IP network 10A in a case where the bandwidth of the radio IP network 10A is determined to be of a size not capable of transferring VoIP packets within a predetermined delay time period. In addition, a remaining number of VoIP packets excluding the number of VoIP packets transmittable within the time period equal to the window T1 minus the window T2 are transmitted to a radio IP network 10B.

Abstract: A voice communication apparatus includes an analysis unit to analyze an outgoing conversation signal transmitted from a user of the voice communication apparatus to another party in communication or an incoming conversation signal from the another party in communication received by the voice communication apparatus, a storage unit to store a voice message to be played back, a mixing unit to perform mixing of the voice message with the incoming conversation signal, and a control unit to control a start timing of the mixing so that the mixing of the voice message with the incoming conversation signal is started between utterances of the user on the basis of an analysis result of the analysis unit.

Abstract: A method and system is provided for peer-to-peer communication among a plurality of sites in a land mobile radio (LMR) system. The sites are interconnected by a data network, such as an IP network. When a site, referred to as the initiating site, intends to communicate with other sites, the initiating site transmits a call request message. The call request message is multicast over the data network to the intended recipients. The initiating site receives an acknowledgement message from the other sites confirming their willingness to communicate with the initiating site. The initiating site then sends a voice message, via multicasting, to the other sites interested in the call.

Abstract: A wireless mesh network system includes a plurality of wireless devices adapted to forward a message packet. Each of the wireless devices includes a transmitting unit, a receiving unit, a communications controller, and a RAM. Further, the communications controller controls such that an original source address contained in a message packet forwarded by a wireless device, which is an address of the wireless device that has firstly transmitted the message packet, and a packet number that identifies the message packet are stored in the RAM, and controls such that a message packet received is forwarded if information contained in the message packet received is not consistent with information of the original source address and the packet number stored in the RAM.

Abstract: Aspects of the present disclosure involve systems, methods, computer program products, and the like, for collaboration conferencing with multiple participants over a communications network, and more specifically for a conferencing routing service for managing and routing collaboration participants. In one embodiment, the conferencing routing server is configured to respond to a failure in a conference bridge to allow for repair to the network and/or account for split conferences that may occur due to the bridge failure. In one embodiment, the CCRS may account for moving a conference bridge into an offline state by removing the conference bridge from consideration for hosting a collaboration conference.

Abstract: An approach for providing custom call waiting with media messaging is disclosed. A request is received from a first voice station to establish a voice communication session with a called party station. A detection is made that the called party station is currently in communication with a second voice station. A content designated from a call-waiting condition for presentation to a caller is retrieved. The content to either the first voice station or the second voice station is retrieved in response to a user input from the called party device.

Abstract: A system and method for automatically adjusting floor controls based on conversational characteristics is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold comprising a minimum number of timeslices for at least one of the current configuration and one of the possible configurations is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: During a conference, a multipoint control unit (MCU) designates priority and non-priority endpoints. The MCU forms priority audio from the priority endpoint and sends that audio to the other endpoints at a normal level. However, the MCU forms non-priority audio from the non-priority endpoint and handles that audio based on whether the input audio from the priority endpoint is from speaking or not. If the priority endpoint's audio indicates a participant at that endpoint is speaking, then the MCU sends the non-priority audio to the other endpoints at a reduced level. Designation of which endpoint has priority can be based on which endpoint has a current duration of audio indicative of speech that is longer than other endpoints. Alternatively, the designation can be based on which endpoint is currently presenting content during the conference or based on a mix of speech audio and content presentation.

Abstract: In one embodiment, a method includes receiving data that indicates a maximum number of end nodes with which to set up real-time data sessions that is less than a conference number of nodes whose data is mixed in a single real-time data conference. Real-time data sessions are established with a connected set of a connected number of nodes not greater than the maximum number. A mix of the real-time data received from the connected set is presented to a user of the first node. Local real-time data is received from a user. The received real-time data is distributed by mixing and sending. This includes, for each individual node in the connected set, mixing received local real-time data with real-time data received from other nodes in the connected set different from the individual node to form individual mixed real-time data, and sending the individual mixed real-time data to the individual node.

Abstract: Processing images in conference between a plurality of video conferencing terminal. The method includes providing properties of terminals, defining a number of groups based on the properties of the terminals, then determining an ordered list of the a number of groups, and determining a pane layout for displaying images from the terminals. The pane layout is determined based on the provided properties of the terminal the pane layout is displayed on and the properties of the other terminals in the conference. Then, mapping the ordered list of the number of groups to the respective determined pane layout, and providing at least one composite image stream to each of the terminals according to the selected pane layout and mapping of the a number of groups. Different types of telepresence endpoints, (e.g. different manufacturers, numbers of screens/cameras, etc.) can thus interoperate well in the same video conference.

Abstract: A video teleconferencing system and method for providing reservationless conferencing is disclosed. The method involves creating a conference profile identifying selected connection parameters; allocating a dial-in number; notifying participants of the dial-in number; detecting a call from a requesting participant; determining if sufficient resources for the participant are available; and connecting the requesting participant into the reservationless conference. Additionally, the video teleconferencing method allows the conference time to be automatically extended. This is done using timers and reallocating available resources when the original end of conference time arrives. The method also enables the conference to automatically terminate after all conference participants leave the conference.

Abstract: In one embodiment, a method comprises receiving through a telephony user interface from a user at the telephony user interface an indication of a first direction of a sound source along a first axis with respect to the user and a second direction of the sound source along a second axis with respect to the user. The first axis is substantially perpendicular to the second axis. The method further involves communicating the indication of the first direction and the second direction to enable audio communication of sound originating at the sound source to the user as if coming from the first direction and the second direction with respect to the user.

Abstract: Methods and apparatus are provided for a timeslot teleconference. A meeting invite is created for a timeslot teleconference on a computer. Timeslots are created for the timeslot teleconference in the meeting invite on the computer. Participants are designated for the timeslots in the meeting invite on the computer, and the timeslots are different time periods during the timeslot teleconference, which can have variable start and end times, as determined by a host. Reach numbers are added for each of the participants in the meeting invite on the computer, and the reach numbers are contact numbers to call the participants for their respective timeslots. The meeting invite is transmitted to participants on the computer. To participate in the timeslot teleconference, the timeslot teleconference is configured for the reach numbers to be called for the participants to correspond to their respective timeslots and/or the participants to call in for their respective timeslots.

Abstract: A system for effecting conference calling in a telecommunication network includes: (a) a plurality of subscriber units coupled with the telecommunication network; and (b) a conference control unit coupled with the telecommunication network. The conference control unit responds to a request message from a requesting subscriber unit to effect coupling of a conference-set of subscriber units in a conference calling configuration. The conference control unit controls inclusion in the conference-set by respective subscriber units. The system may provide that the conference control unit provides and controls conferencing services for the conference-set of subscriber units.

Abstract: A method of dynamically switching voice channels between sampling rates in a telecommunication system including a plurality of conferenced channels that have a telephony side, a network side, and a conferencing unit therebetween. During conferencing the presence of a switching stimuli is detected, and at least a first affected voice channel affected by the switching stimuli is identified from the plurality of conferenced voice channels. Responsive to the switching stimuli, based on a telephony side sampling rate of the first affected voice channel, a network side sampling rate of the first affected voice channel, and respective sampling rates of other conferenced voice channels, it is determined whether or not to switch a sampling rate for at least the first affected voice channel using a switching criteria. Provided the determining decides to switch, the sampling rate is switched for at least the first affected voice channel.

Abstract: An IP communication apparatus employed in a telephone voice/moving picture recording system is comprised of: an IP packet transmitting/receiving I/F for transmitting/receiving an IP packet; an IP address acquiring unit of acquiring an IP address corresponding to a transmission source of the IP packet; a signal judging unit for performing a signal judging operation by employing data contained in an IP packet; a recording unit for recording the data in relation to the IP address based upon a judgment result made by the signal judging unit; and a recording control unit for controlling the recording unit.

Abstract: Techniques are provided herein to add multicast media streams to teleconferencing sessions without the unwanted side effects that occur when a speaker's own media is echoed back to the speaker. Multicast and unicast media streams are generated by a network device, e.g., a conference bridge, and distributed to any number of endpoints, e.g., conference endpoints. A media selector is provided that selects the appropriate multicast or unicast media stream for forwarding to an associated network endpoint.

Abstract: A method and apparatus for interworking between push-to-talk over cellular (PoC) systems and instant messaging (IM) systems is provided. An interworking entity that behaves like a PoC client to a PoC server and as an IM client to the IM server is provided. The interworking entity may be implemented as part of the PoC server or the IM server. The interworking entity listens to talk bursts from the PoC server and when it receives a talk burst the interworking entity sends the message to the IM client via the IM server. In an embodiment, the interworking entity listens to talk bursts or audio clips from the IM server and when it receives them, stores them in a temporary storage memory and converts them to a streaming format. The interworking entity then requests the floor from the PoC server, and, once the floor is granted, streams the stored audio clip or talk burst to the PoC server. The PoC server then distributes the clips to the PoC clients.

Abstract: A method and apparatus for conferencing dynamically determines composite image layout based on properties of received conferencing signals that include video signals and control signals. Video signals received from connected sites are processed to provide a composite video signal according to the defined composite image layout and the composite video signal is transmitted to the connected sites. Upon a change in properties of one or more received video signals, the composite image layout may be redefined. The received video signals are processed according to the redefined composite image layout to provide a second composite video signal that is transmitted to the connected sites.

Abstract: Systems and methods are described for implementing an application-level routing protocol for multiparty audio-video conferencing. In one implementation, application-level per-stream routing techniques separately control audio data and video data between conference members hosted on a network. Different audio application-level multicast (ALM) trees are generated by each member, are dynamically updated according to shortest-path-first selection of data delivery paths, and are used to send audio data to the other members of the videoconference. Likewise, different video ALM trees are generated by each member, are dynamically updated according to broadest-path-first selection of data delivery paths, and are used to send video data to the other members of the videoconference. Separate audio and video ALM trees for each member can utilize IP multicast in segments of the network in which IP multicast is enabled.

Abstract: A network entity, method and computer program product are provided for effectuating a conference session. The method may include receiving a plurality of signals representative of voice communication of the participants. In this regard, the signals may be received from a plurality of terminals of a respective plurality of participants at one of the locations, each of at least some of the terminals otherwise being configured for voice communication independent of at least some of the other terminals. The method of this aspect also includes classifying speech activity of the conference session according to a speech pause, or one or more actively-speaking participants, during the conference session. The signals of the respective participants may then be mixed into a at least one mixed signal for output to one or more other participants at one or more other locations, the signals being mixed based upon classification of the speech activity.

Abstract: In one embodiment, a network device attaches a header including an identifier for an original call to an outgoing refer message and may also modify a table entry for a consultative call to include the identifier. The network device then monitors incoming invite messages for a header containing the identifier and observes that an incoming invite message containing the header correlates to the original call. When a header is not included, the network device extracts information from the incoming invite messages and compares the extracted information to the modified table entry to correlate one of the incoming invite messages to the original call.

Abstract: Methods for inter-mixer communication enable a “self-formed” cascaded mixer node tree topology. One method enables inter-mixer node signaling and includes first selecting a plurality of media flow payloads at a first cascaded mixer node. A set of conference payloads are generated and each include the media flow payloads from the plurality of media flow payloads that are part of the same conference call, an identifier for the conference call, and a node identifier for each media flow in the conference call. For each media flow in the conference call, each conference payload further identifies whether the first cascaded mixer node is a root node or a leaf node. The set of conference payloads are bundled into a single packet and sent to a second cascaded mixer node. Another method enables the “self-forming” of the mixer node tree topology. Yet another method enables active talker identification within the “self-formed” tree topology.

Abstract: A management system may test network paths prior to scheduled calls. Testing network paths may include one endpoint placing a test call to an endpoint on the other end of the path. Various characteristics of the network path may be tested. If the network path fails and/or in some way does not meet the predefined characteristics for the call, an administrator may be notified and/or alternate network path may be detected and tested. The call may be rescheduled if a satisfactory network path is not found in time.

Abstract: A system and method for automatically adjusting floor controls for a conversation is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: The improved vehicle crew communication system provides an increase in the number of wireless headsets that can be used with one base station, an increase range from the vehicle for operation of said headsets, full duplex operation communication over digitally encrypted DECT protocol links from one base station to a plurality of headsets, pairing of each headset to a particular base station, use of wireless technology inside and outside of the vehicle, automatic channel and link selection to transparently avoid interfered-with channels and links, removal of the need to manually select a channel on the headset, and the combination of the noise-cancelling microphone with noise threshold, noise attenuation, and line echo cancellation tuned to intercom parameters.

Abstract: In a computer system having a memory, a processor, and a network interface, a method for listening on multiple conferencing interfaces having the steps of loading a set of transport components into the memory; initializing each transport components of the set of transport components to listen on a particular conferencing interface using the network interface, each transport component of the set of transport components listening to a different conferencing interface; receiving an incoming call signal on the network interface having an incoming conferencing interface; processing the incoming call signal to detect the incoming conferencing interface; and launching an application based on the incoming conferencing interface.

Abstract: A method and system for peer-to-peer communication among a plurality of sites in a land mobile radio (LMR) system. The sites are interconnected by a data network, such as an IP network. Each site transmits a periodic broadcast message, which includes the identity of the transmitting site and a list of target groups that the transmitting site is interested in participating. The message is received by the other sites, thus each site becoming aware of the presence of all other sites in the system and Talk Groups they are interested in. When a site, referred to as the initiating site, intends to communicate with other sites, the initiating site transmits a call request message. The call request message is multicast over the data network to the intended recipients. The initiating site receives an acknowledgement message from the other sites confirming their willingness to communicate with the initiating site.

Abstract: A group communication method capable of performing group communication without additional functions provided on a network and with preventing excessive network traffic. Each communication device first transmits an enrollment message. A providing device groups the communication devices, creates a group key specifying the communication devices that are identified as a same subgroup, and transmits the group key to the communication devices. The communication devices receive and store the group key. To transmit distribution data to a subgroup, the providing device transmits the data together with data corresponding to the group key to a selected communication device of the subgroup. When the communication devices recognize the distribution data, the communication devices determine whether to receive the distribution data, based on the distribution data and the stored group key.

Abstract: A multiparty calling method, which includes initiating a multiparty call between at least a first, second and third terminal, and setting a flag in the first terminal to inform a processor of the first terminal to skip a process of closing an existing communication path between the first terminal and the second terminal, when the first terminal is disconnected with the third terminal. Further, the set flag is released when the multiparty call between the at least first, second and third terminals has been established for a predetermined time period. In addition, the first terminal preferably sets the flag when the multiparty call join command is transmitted to the network.

Abstract: The quality of signals during SDMA is raised. In an uplink, a signal processing unit receives signals respectively from a plurality of terminal apparatuses which have been multiple-accessed by division of time. It derives receiving channel characteristics corresponding to the plurality of terminal apparatuses, respectively, for each time slot. In a downlink, the signal processing unit derives transmitting channel characteristics from the receiving channel characteristics derived and, based on the transmitting channel characteristics derived, it transmits signals respectively to the plurality of terminal apparatuses to which SDMA has been performed.

Abstract: Methods and apparatus are provided for a timeslot teleconference. A meeting invite is created for a timeslot teleconference on a computer. Timeslots are created for the timeslot teleconference in the meeting invite on the computer. Participants are designated for the timeslots in the meeting invite on the computer, and the timeslots are different time periods during the timeslot teleconference, which can have variable start and end times, as determined by a host. Reach numbers are added for each of the participants in the meeting invite on the computer, and the reach numbers are contact numbers to call the participants for their respective timeslots. The meeting invite is transmitted to participants on the computer. To participate in the timeslot teleconference, the timeslot teleconference is configured for the reach numbers to be called for the participants to correspond to their respective timeslots and/or the participants to call in for their respective timeslots.

Abstract: A method of providing conference call connections includes initiating a conference call between multiple conferees using a conferencing controller, placing each of the multiple conferees in a half-duplex listen only mode using the conferencing controller, and receiving a signal at the conferencing controller from at least one of the multiple conferees to actively participate in the conference call. Active participation can be in the form of asking questions, giving comments, or providing a keypress to vote or provide other information. The conference call can also be broadcast to increase participants. Also, after the conference call is initiated, additional potential conferees can be added.

Abstract: An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.

Abstract: One embodiment of the invention is a computer controlled method for use with a communication system. The method includes a step of receiving a plurality of communications, where each one of the plurality of communications is from one of a plurality of communication sources; includes a step of mixing (that is responsive to a plurality of floor controls) the plurality of communications for a plurality of outputs associated with plurality of communication sources; and includes a step of analyzing, for a plurality of users associated with the plurality of communication sources, one or more conversational characteristics of two or more of the plurality of users. The method also includes a step of automatically adjusting the plurality of floor controls responsive to the step of analyzing. Other embodiments include systems and devices that use the method as well as program products that cause a computer to execute the method.

Abstract: A method for managing conference resources includes receiving from an endpoint of a user a call to join a multipoint conference between a plurality of participants. Each participant communicates on the multipoint conference through a respective endpoint. The method includes determining that an insufficient amount of resources exist to allow the user to communicate on the multipoint conference and determining participation instructions to enable the user to communicate on the multipoint conference. The method also includes communicating the participation instructions. Location information associated with the user and location information associated with at least one of the plurality of participants may be used to determine the participation instructions to enable the user to communicate on the multipoint conference.

Abstract: A method and arrangement for establishing a conference call from a mobile station in a call centre environment. A user may choose conference call participants from his mobile phone, and a client application in the mobile phone will communicate relevant parameters to the call centre server, which will then call the conference initiator and other participants via the gateway server. The calls are handed over to the gateway server, which will digitize each call to an audio and/or video stream, typically an RTP stream. The streams are relayed to a computer where the streams are combined to form a successful conference call. The conference call is then recorded to a data file on the computer, and may be sent to participants or stored to a database with relevant metadata. The network infrastructure of the telecommunications network operator need not be consulted, nor is there any need for additional hardware.

Abstract: A voice conference apparatus includes: a communication unit that transmits and receives a first voice signal to and from another apparatus via a network; an output unit that reproduces the first voice signal received by the communication unit and outputs the reproduced first voice signal as a first voice; an input unit that collects the first voice and a second voice generated by a voice source, as a second voice signal; a separating unit that separates the second voice signal into a third voice signal originated from the first voice and a fourth voice signal originated from the second voice; and a selecting unit that selects one of the third voice signal and the fourth voice signal. The communication unit transmits the one of the third voice signal and the fourth voice signal selected by the selecting unit to the other apparatus.

Abstract: In a communications system where a plurality of endpoints share a virtual address, all of the endpoints are involved in a communication with one of them, but only one has its talk path enabled at any one time. Each endpoint has an actuator that, when actuated, enables that endpoint's talk path and disables the other endpoints' talk paths. Preferably, one of the endpoints is either an announcement facility or a teletypewriter, while another one of the endpoints is a telephone.

Abstract: A method of controlling teleconference signals includes receiving, at a teleconference bridge, endpoint-generated audio signals from each of a plurality of participating endpoints arranged at a plurality of locations. At least two of the participating endpoints are acoustically collocated at one of the locations. A bridge-generated audio signal is generated for each of the participating endpoints based on a set of signals. The set of signals includes all of the endpoint-generated signals received at the teleconference bridge, exclusive of the endpoint-generated signals transmitted to the bridge from each of the acoustically collocated endpoints.

Abstract: A method and system for coordinating a Voice Over Internet Protocol (VoIP) teleconference. A teleconference operator drags the name of invited attendees into a hot window, which connects the invited attendees to a channel for the teleconference. Each invited attendee's name has a status icon associated with it, indicating if the invited attendee is active in the teleconference. Each invited attendee has access to a Graphical User Interface (GUI) that permits the attendee to toggle back and forth between different teleconferences.

Abstract: In various embodiments, a management system may query connection devices for their capabilities and store the information in a data structure. In some embodiments, the data structure may be in a form of a Deducible Capability Numbering (DCN) number. In some embodiments, the DCN number may describe the capabilities of a connection device using a series of representative numbers. After connecting the connection devices, the management system may continue monitoring capabilities of the connection devices and line failures. The management system may reconnect connection devices based on their capabilities if a line failure occurs.

Abstract: A telephone system that allows multiplexing and demultiplexing plural group calls through simple operations is provided. The telephone system includes a conference trunk creating unit that creates a conference trunk as required and has a multiplexing/demultiplexing special number table prestoring a multiplexing special number and demultiplexing special number corresponding to plural group calls. After plural group calls start, if the multiplexing special number is input, a control unit causes the conference trunk creating unit to create a large-conference trunk, switches and connects the voices of plural group calls corresponding to the multiplexing special number to the large-conference trunk and starts the multiplexed group call.

Abstract: A method for transferring data based on an event function is provided. The method includes receiving an event function from an event initiator. The event function is distributed to one or more destinations associated with event performers. A response message is received as a function of destination and time. It is determined whether the time is less than or equal to a limit time, and it is further determined whether an event parameter is greater than or equal to an event constant if the time is less than or equal to the limit time. Data is transferred from a data origination system accessible by the event initiator to a data destination system accessible by the event performer if the event parameter is greater than or equal to the event constant and if the time is less than or equal to the limit time.

Abstract: A method, computer program product, system and apparatus are presented for reducing wasted bandwidth due to supercasting multicast cells through a router switch fabric. In one embodiment of the present invention, signatures of a switch fabric destination address are generated and compared. A signature is an information-rich representation of the fabric destination address that is generated using the fabric destination address. Therefore, supercasting can be minimized by combining fabric destination addresses with like signatures. Aspects of the present invention include generating the signatures using random permutation maps of the set of switch fabric ports or determining intersections of a fabric destination address with a selection of subsets of the switch fabric ports.

Abstract: Systems and methods by which voice/data communications may occur in multiple modes/protocols are disclosed. In particular, systems and methods are provided for multiple native mode/protocol voice and data transmissions and receptions with a computing system having a multi-bus structure, including, for example, a TDM bus and a packet bus, and multi-protocol framing engines. Such systems preferably include subsystem functions such as PBX, voice mail and other telephony functions, LAN hub and data router. In preferred embodiments, a TDM bus and a packet bus are intelligently bridged and managed, thereby enabling such multiple mode/protocol voice and data transmissions to be intelligently managed and controlled with a single, integrated system. A computer or other processor includes a local area network controller, which provides routing and hub(s) for one or more packet networks. The computer also is coupled to a buffer/framer, which serves to frame/deframe data to/from the computer from TDM bus.