19464results about "Multiplex system selection arrangements" patented technology

A conference call facility is described in which one (2a) of a group of communication devices (2a,2b,2c,2d) connected to a low power radio frequency network (9) is able to set up a call to a party external of the network (9) and then selectively add further devices (2a,2b,2c) to the call under the control of the user of the one device (2a). The users of the other devices (2b,2c) are able to enable or disable the selection of their device in a conference call. One or more of the communication devices may be a mobile radio telephone equipped with the necessary network interface (1).

An approach is provided for supporting multi-media conferencing. Textual information (e.g., text associated with a presentation) is received for display during a conference session to multiple participants. For a particular participant, configuration information corresponding to the participant specifies language assistance for the textual information. Language assistance involves augmenting the textual information according to the configuration information for comprehension of the textual information by the particular participant. In one embodiment, the augmented textual information includes language translation of the textual information. Additionally, the augmented textual information can encompass definitions of terms, including acronyms. The augmented textual information is forwarded for display to the participant during the conference session.

Systems, methods, devices, processes, procedures, algorithms, networks, and network elements are described for time-scheduled and/or time-reserved dat networks. Invention provides capabilities for synchronizing data networks and/or data network links; for establishing time-schedules, time-reservations, time-schedule reservations, and/or reservation time-slots for packets, cells, frames, and/or datagrams; and for transferring, transmitting, switching, routing, and/or receiving time-sensitive, high-reliability, urgent, and/or other time-scheduled, time-reserved, time-allocated, and/or time-scheduled-reservation packets, cells, frames, and/or datagrams, such as real-time and high-priority messages over these networks. The invention(s) enables packet-, cell-, datagram- and/or frame-based networks to thereby efficiently, reliably, and in guaranteed real-time, to switch and/or route data such as voice, video, streaming, and other real-time, high-priority, high-reliability, and/or expedited data with guaranteed delivery and guaranteed quality of service. Networks may be fixed, point-to-point, mobile, ad-hoc, optical, electrical, and/or wireless.

A system and method for interactive management of multiple wireless devices (such as PDA's or desktop or laptop computers) making graphical or video presentations while connected wirelessly to a display device whose wireless module converts the transmitted data from the wireless devices to the required format for display.

A method and apparatus are provided that allows for the representation of a larger number of classes of network traffic and logical queues than is physically available on a per port basis within a network device. A number of logical queues, whose number can match the number of classes of network traffic a network device handles, may be supported across an aggregated set of ports even though the network device has fewer physical queues per port than there are classes of network traffic. The method and apparatus improve the management of network traffic sensitive to time delay and jitter, and further facilitates the operation of these applications in a simultaneous or near simultaneous manner.

A method and system to provide push-to-talk from one user to another in a wireless packet data telecommunications network is described. The system may include: a wireless communication network including push-to-talk (PTT) functionality, with a Session Initiation Protocol (SIP) Proxy Server; a SIP Registrar and Location Server operable to store contact addresses of active mobile devices; a Realtime Transport Protocol (RTP) Media Gateway (PTT Server) operable to function as a call endpoint for each of a plurality of mobile devices wherein the plurality of mobile devices are segmented into membership groups, the PTT Server further operable to multicast a communication from one member of the group to the other members of the group; and an Internet Protocol (IP) network interconnecting the SIP Proxy server, the SIP Registrar and Location Server, and the PTT Server.

A hierarchical network management system (NMS) in which a plurality of NMS managers, each responsible for different portions or aggregations of a communications network, are logically arranged in a tree structure. The NMS managers are further organized into various sub-groups. The NMS managers within each sub-group monitor the status of one another in order to detect when one of them is no longer operational. If this happens, the remaining operational NMS managers of the sub-group collectively elect one of them to assume the responsibility of the non-operational NMS manager. The NMS is thus "self-healing" in the sense that one NMS manager can dynamically, without operator intervention, assume the responsibilities for another NMS manager.

Reduction of administrative overhead in maintaining network information, rapid convergence on an optimal routing path through the data network, and utilization of only required network resources are realized by a novel method for establishing a call path between network users. The method is based upon deployment of a network information server that stores network topology information and that is addressable by each end user. In this method, the network information server receives a request to establish a call path. The request identifies at least the calling party. In response to the request, the network information server determines a network traversal between the calling party and a root network wherein the network traversal includes call path information about the sub-networks between the calling party and the root network. The request for establishing a call path can also identify the called party. Based on the calling and called party identification, the network information server also determines a second network traversal between the called party and the root network. The second network traversal is sent to either the calling party or the called party or to both the calling and called parties. The server can determine an intersection of the traversals and send the intersection information to the parties. The intersection information is known as a merge point and represents an optimal call path between the parties.

Embodiments of the present invention are directed generally to recording communication of a call utilizing a multi-channel recording technique. According to one exemplary embodiment, inbound communication from each party to a call (e.g., from each communication device that is party to a call) to a recording system is assigned to a separate channel, and communication on each channel is independently recorded. Further, during the call, a control channel is generated that correlates the multiple communication channels. The independently recorded communication channels and control channel may be used to analyze a recorded call from any desired perspective. For instance, communication from a given party may be analyzed in isolation. Further, the control channel enables the recorded multiple communication channels to be correlated such that the communication received (e.g., heard) by any selected party may be accurately re-created for analysis thereof.

A system for scheduling a conference between callers includes a database that stores scheduling information indicating at least a start time, a duration, and a maximum number of callers for one or more scheduled conferences, the scheduling information reflecting available conferencing resources. A server complex coupled to the database communicates, to a requesting Internet Protocol (IP) user, at least one page including one or more scheduling input fields. The server complex receives scheduling input from the requesting IP user for a requested conference according to the scheduling input fields. The server complex accesses the database to determine, according to the scheduling input, whether sufficient resources are available for the requested conference. If so, the server complex allocates at least some available resources to the requested conference and generates confirmations of the requested conference for communication to the callers.

A system and method for providing services to packet flows in a communication network. The services include enhanced charging, stateful firewalls, traffic performance optimization (TPO) and advanced services such as content differentiated charging, per-subscriber stateful firewalls, and VPNs, for example. A gateway can be used to route each packet flow through the services on a per-subscriber basis by inspecting accounting messages and checking subscriber profiles prior to setting up a new packet flow session. By directing the packet flow through the services requested latency can be reduced as well as the need to have equipment for each session regardless of whether the service is provided to the packet flow. The services can be provided in-line as well.

An automatic messaging system for a telecommunications network is disclosed. A call processing entity such as a switch or central control point recognizes a call-processing event and responsively generates and sends an informational message to a specified destination. The informational message may be a text description of the event, such as an indication that "calling party just placed a call to called party" or may take other forms, such as a video message, an audio message or a fax message.

A method to systematically analyze a next generation telecommunications network to result in creating a provisioning plan for provisioning the network to provide services for one or more subscribers. In one specific embodiment, the method involves creating and storing information that represents a logical decomposition of the next generation network into a plurality of discrete functional areas. The information representing the functional areas is analyzed to identify one or more provisioning requirements for each of the functional areas. One or more provisioning procedures are determined, and one or more required provisioning tools are identified for each of the functional areas, based on the provisioning requirements. A sequence of execution of the procedures and tools is created and stored as a provisioning plan.

An approach provides network-based call processing. A packetized voice call is received from a first station. The packetized voice call is queued at a pre-designated queue maintained within a network of a service provider. Further, the packetized voice call is selectively forwarded to a second station.

A system and method are disclosed for packet voice conferencing. The system and method divide a conferencing presentation sound field into sectors, and allocate one or more sectors to each conferencing endpoint. At some point between capture and playout, the voice data from each endpoint is mapped into its designated sector or sectors. Thereafter, when the voice data from a plurality of participants from multiple endpoints is combined, a listener can identify a unique apparent location within the presentation sound field for each participant. The system allows a conference participant to increase their comprehension when multiple participants speak simultaneously, as well as alleviate confusion as to who is speaking at any given time.

A system and method for continuous voice calls when a user switches between packet data and circuit switched access networks. In one example embodiment, the present innovations include an interworking system that supports voice call continuity for a user that moves between IP-CAN and CS networks (e.g., PSTN or GSM). In one example embodiment, the present innovations comprise a voice call continuity application server (VCC-AS) that serves as an anchor point for a voice call (i.e., it is the node from which a handover is initiated) and controls and handles voice calls to and from the user equipment (UE) regardless of the access network.

A device is provided including a processor and a display configured to list more than one presence enabled contact. A user agent is configured to indicate an availability status of the device to a presence server. The user agent is at least partially executed by the processor. Moreover, at least one input is configured to enable a device user to request a communications session with at least one of the plurality of presence enabled contacts. The request is initiated by the user agent when the requested presence enabled contact is available.

Packet-based central conference bridges, packet-based network interfaces and packet-based terminals are used for voice communications over a packet-based network. Modifications to these apparatuses can reduce the latency and the signal processing requirements while increasing the signal quality within a voice conference as well as point-to-point communications. For instance, by selecting the talkers prior to the decompression of the voice signals, decreases in the latency and increases in signal quality within the voice conference can result due to a possible removal of the decompression and subsequent compression operations in a conference bridge unnecessary in some circumstances. Further, the removal of the jitter buffers within the conference bridges and the moving of the mixing operation to the individual terminals and/or network interfaces are modifications that can cause lower latency and transcoding within the voice conference.

Methods for providing video on demand service from an Application Service Provider (“ASP”) to a user over a Regional/Access Network (“RAN”) are provided. A request for a video on demand application flow may be received from the user. In response to the request, information may be obtained from the RAN regarding the capabilities of the user's Customer Premises Equipment (“CPE”). A profile may then be forwarded from the ASP to the RAN that specifies at least one policy regarding the transmission of data associated with the video on demand application flow across the RAN. The data associated with the video on demand application flow may then be forwarded across the RAN in accordance with the profile. The RAN may also be provided a default profile that specifies default policies that apply with respect to video on demand application flows that are transmitted across the RAN prior to obtaining information from the RAN regarding the capabilities of the CPE of any specific user. The user specific profiles may then be used to alter one or more policies specified in the default profile.