7891results about "Multiplex system selection arrangements" patented technology

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.

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 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.

Private Multimedia Network (PMN) complements, and is an improved alternative to digital videoconferencing and multimedia delivery systems. PMN's desktop and meeting room delivery system is designed to support the exponential growth of enterprise team-based initiatives. PMN provides “one-stop-shopping” for the full multimedia rubric. It delivers user-friendly control and cost/effective TV and broadcast quality videoconferencing and other multimedia services to organizations with “critical mass” campuses and building complexes. Though digital systems dominate the videoconferencing marketplace, PMN's hybrid digital/analog architecture has no digital peer in breadth or quality of service within or between campuses. The novel architecture leverages advances in analog video short-haul technology, digital long-haul technology, and telephony audio and control technology to deliver four-level multimedia services: 1) premise; 2) campus; 3) multi-site; and 4) ubiquitous (any site with ITU compatible multimedia equipment (e.g., videoconferencing) and communication links). On balance, the price/performance afforded by PMN's centralized Telco-based control and audio delivery combined with its decentralized broadcast quality video distribution raise videoconferencing and other multimedia services to a new level of ubiquity. Just as telephones and PC LANs, PMN delivers expensive Boardroom and mobile cart videoconferencing capabilities to every desktop via existing multimedia wall plates. The key phases for this invention are: Lip-synchronization across differing network communication links and protocols; Ubiquitous multimedia service; Cost/effective room and desktop deployment; Telco control and audio; Broadcast quality video; Isochronous Quality; Centralized control and distributed operation; and Interoperable architecture.

The invention provides a method and apparatus for re-ordering data traffic units, such as IP data packets, that may have been miss-ordered during a transmission over a multi-pathway link between a source node and a destination node in a network. The re-ordering apparatus includes a storage medium for intercepting the IP data packets and holding the IP data packets to allow IP data packets delayed on slower pathways to catch-up. The IP data packets in the storage medium are re-ordered based on their sequence number in an attempt to restore the original order of the IP data packets. A maximal time delay variable determines how long a certain IP data packet can be held in the storage medium. The TP data packet is released prior to the maximal time delay variable or as the maximal time delay variable is exceeded.

A method and apparatus for adding a member to an active call in a group communication network provides for receiving a member list from a user and sending a request to a server to add the member list to the active group call. The method and apparatus further provides for announcing each member in the member list that they are being added to the group call, receiving acknowledgement from a member who wishes to participate in the group call, and forwarding media to the member. The method and apparatus also provides for a significant reduction in the actual total dormancy wakeup time and latency by exchanging group call signaling even when mobiles are dormant and no traffic channel is active.

An apparatus, system, and method to transcribe a voice chat session initiated from a text chat session. The system includes a chat server, a voice server, and a transcription engine. The chat server is configured to facilitate a text chat session between multiple instant messaging clients. The voice server is coupled to the chat server and configured to facilitate a transition from the text chat session to a voice chat session between the multiple instant messaging clients. The transcription engine is coupled to the voice server and configured to generate a voice transcription of the voice chat session. The voice transcription may be aggregated into a text chat history.

A method and system of performing IP multicast includes a client at one of many downstream networks that sends a request signal to an upstream network via a return channel (e.g., the Internet), and the upstream network sends the request to a media server. The media server, and for live data, a media encoder, processes a media stream to generate a real-time IP multicast communication that is output to a unidirectional satellite that transmits the IP multicast communication to the client without delay via the downstream network, and can convert the IP multicast to unicast. The bidirectional return channel allows the source to calculate billing information based on client usage statistics, and transmits confirmation acknowledgement based on a confirmation request from the upstream network. Because the routing configuration is transparent to the rest of the network, the invention applies to multi-hop networks on both sides of the satellite link.

AIN based methods and apparatus for transitioning telephone numbers and customers from the PSTN to a VOIP network are described. AIN line number portability features are used to allow a few gateway switches that interconnect the PSTN and VOIP networks to service customers whose telephone numbers were originally serviced by several remote PSTN switches. AIN LNP triggers are used to forward PSTN calls, directed to the PSTN switch previously used to service a telephone number, to the gateway switch assigned to route such calls to the IP network. AIN triggers set at the gateway switch insure that the subscriber with the ported telephone number continues to receive AIN services provided before the telephone number was ported to the IP network. Calls from ported telephone numbers to telephones in the PSTN are billed from PSTN switches through the use of AIN functionality and triggers set at the gateway switch.

A method for implementing self-chosen caller identification and instantaneous transmission of caller identification information before communications have been established is described. Aspects of the invention improve upon modem methods and bridge the gap between current technology that allows users to self-identify in communicating with other users and technology that transmits this information before a telephone call is answered. Furthermore, aspects of the invention allow a user to select his or her own identifying information (e.g., picture, ring tone, video) and send it to contacts in his or her contact database. A background transmission process transmits the identifying information upon selection, creating a fast and seamless method of self-identification for caller ID technology. The identifying information may be stored in an association database using factors such as events and identification type. An option may also exist that allows users to override any caller identification information transmitted to them.

Methods and systems are provided for sharing images and chat text over a network. A determination is made as to whether a first user of a first terminal has changed focus from a first image to a second image during a chat session conducted with a second user of a second terminal. At least partly in response to determining that the first user of the first terminal has changed focus from the first image to the second image, a second image identifier is automatically sent to the second terminal. Chat text is transmitted from the first terminal to the second terminal. The second terminal accesses the second image using the second image identifier, and displays the chat text in association with the second image.

An apparatus for optimizing communications between session border controllers in a network environment is provided that includes a session border controller (SBC) operable to communicate with a first endpoint and a second endpoint. The SBC is also operable to communicate with a data path border element (DBE) and a signaling path border element (SBE). The DBE is operable to provide a media relay for a communication session involving the endpoints, the media relay being invoked by either of the endpoints using a traversal using relay network address translation (TURN) protocol, which allows the SBC to operate in either of two modes depending on protocol support present in the endpoints.

A system and method for performing messaging between telephones in a data network telephone system. The message communications may be managed by a messaging application connected to the data network. The messaging application detects when a telephone is connected to another telephone and sends a messaging input screen to the first telephone. The messaging application may determine the availability of other users for messaging and may control messages communicated between the first telephone and the other users. The messages may be text or voice messages.

A method, system, and network elements for authentication and authorization of a mobile terminal (MT) roaming to or in a foreign network different from its home network is provided, the home network having an authentication and authorization home server (AAAH), and the foreign network having a plurality of domains each of which comprises at least one local server (AAAL1, AAAL2) for authentication, authorization and accounting, each of which local servers being connected to at least one network access server (NAS) for handling access for mobile terminals roaming to or in the foreign network, wherein an authentication and authorization of the mobile terminal is performed whenever the mobile terminal performs a roaming, wherein the authentication and authorization is performed according to a procedure pursuant to one of a plurality of hierarchy levels, whereby a combination of network elements involved in the roaming determines the hierarchy level to be used.

A teleconferencing system includes a network. The system includes a plurality of nodes in communication with each other to form a conference. Each node having a video display with a display layout, at least one of the nodes individually controlling at least partially the display layout of each node in the conference with a specific format able to be unique for each node. A method for providing a teleconference including the steps of forming a conference with a plurality of nodes in communication with each other through a network. Each node having a video display with a display layout. There is the step of controlling individually with at least one of the nodes at least partially the display layout of each node in the conference with a specific format able to be unique for each node. A teleconferencing node. A teleconferencing system includes a network. The system includes a plurality of nodes in communication with each other through the network to form a conference. Each node communicating only a change to the other nodes when the change occurs. A method for conducting a telecommunications conference between at least three parties includes the steps of establishing a conference between the party. There is the step of making a change to the conference. There is the step of communicating only the change to the parties.

Upper layer applications discover, establish and manage the QoS connectivity through a SIP (session initiation protocol) that serves as a unified interface. In one implementation, this is IP Multimedia Subsystem-centric, further supporting applications that utilize SIP for session control. This capability extends to the data card universe allowing UMTS data card vendors to establish concurrent QoS-based sessions using multiple primary PDP (packet data protocol) contexts based on a set of SIP triggers, further allowing applications running on a computing system to transparently utilize the established pipes based on the individual QoS requirements.

Provided is a system and method for a multicast routing algorithm to work in infrastructure based mesh networks. It chooses access points, fixed infrastructure gateway nodes connected to each other and/or the global internet via a wired/wireless backbone, as a group of local multicast group leaders to form a multicast group leader cloud. Each local multicast group leader is elected on-demand according to the local multicast group member's request. Each local multicast group leader forms a local multicast tree rooted at this leader connecting all multicast group members associated with the AP. The processes of electing and maintaining local multicast trees rooted at APs enable efficient coordination with underlying unicast routing to exploit the advantages of fixed infrastructure nodes. Therefore, routing overhead and multicast tree convergence time are reduced. The method can support large networks with fast topology change due to fast convergence and reduced routing overhead.