518 results about "Conference call" 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).

A mechanism for providing a connection from an IP-based network to a circuit-switched network, such as a GSM network is disclosed. A temporary routing number for the circuit-switched network, such as an E.164 number, is delivered to a user terminal, and a circuit-switched call leg is established from the user terminal to the IP-based network using the routing number. Thereby, IMS-services are provided for end users which are located in the radio access network not having sufficient QoS required for voice over IP. In the example of a conference call service, a request for a conference call may forwarded via a data channel or data path to an application server which provides that conference call service. The application server then selects a conference routing number and returns the routing number to the conference host terminal via the data channel. Using the received conference routing number, the conference host terminal can then set up a circuit-switched connection as a call leg of the conference call.

A method and system for distributing voice prompts and messaging closer to the calling or called party via the Voice Access Port (VAP) of a wireless centrex system comprises the steps of transmitting a message request to the VAP from a network server platform, retrieving a selected message and forwarding the message, for example, to the calling party, for example, when the called party is not available or during conference call set-up, call hold or call screening. In accordance with a further embodiment, the voice prompt or message may be personalized with the calling / called party's name, address or other personalized information as appropriate.

In some embodiments, spatially realistic audio may be provided for a conference call. Voices from participants on the left side of a display, in a conference call, may be directed through audio on the left side of the display at the other conferencing system in the conference call (similarly for voices from the center and right side of the display). In some embodiments, two speakers may be used in the system to create synthesized stereo sound at a location specified by directional information received as side information along with the existing audio channel. The location may be determined by using beamforming with integrated microphones on a camera or speakerphone. In some embodiments, the audio signal and directional information may be sent in the form of a left audio channel and a right audio channel.

A system for enabling setup and control of conference calls using short message service (SMS) messages (SMs). The SMS messages may be used to set up, or control a scheduled or real-time conference call. The SMS messages can also be used to exercise real-time control over a conference call. Invitations can be sent in real time to add participants to a call in progress, or to transfer the call from any participant's telephone to another telephone connected to the PSTN.

An apparatus and methods for Real Time Protocol (RTP) based bridging to create calls to and From any communication devices connected to the Public Switched Telephone Network (PSTN) or any public or private Internet Protocol network. The methods are represented in controls residing either in an RTP Bridge apparatus or as middleware on a service creation platform, softswitch, or SIP proxy services incorporating SIP bridging and SIP call relay technologies, with an interface to a TDM (Time Division Multiplexing) switch operated by a Local Exchange Carrier (LEC) or Competitive LEC (CLEC), or hosted by an Internet Telephony Service Provider (ITSP) with connectivity to the PSTN through media gateways. The method permits a party to request an on demand conference call by either dialing into the apparatus from the PSTN or from any type of RTP communication device such as an IP phone; or using a form of signaling from RTP translation device. The moderator initiates the request, enters the participant(s) to be included in the call, and the launch sequence is initiated and the apparatus makes contact with the phone / end point and uses RTP bridging to interlace the packet streams and deliver one stream back to all the devices. The advantage is an on demand conference call to one or many participants, on similar or different communications platforms.

Recording and indexing audio and video conference calls allows topic-based notification, navigation and playback of one or many recordings. An external conferencing system with five internal modules, recording, indexing, management, notification, and playback allows a plurality of meeting participants to record a conference call, and a plurality of subscribers to access, search, and be notified of the occurrence of an event of interest. The system automatically creates contextual or topic based time intervals which are used to immediately notify a subscriber of an event occurrence. The subscriber can use the time stamps of the topic intervals to navigate and playback the recording. The system also indexes project management files and correlates recording intervals to project tasks, and subsequently correlates multiple recording intervals.

Two advanced group-based voice services, known as Push-to-Conference (P2C) and Push-to-Message (P2M) services are provided by a real-time exchange (RTX) that interfaces to the wireless network. The RTX provides a full-duplex Push-to-Conference (P2C) session between an initiator and two or more other participants, wherein the P2C session comprises a full-duplex conference call, and both the real time exchange and handsets participating in the P2C session communicate with each other using call setup and in-band signaling within the wireless network. The realtime exchange may be coupled to and work with a Push-to-Message (P2M) server to deliver multimedia messages in a non-real time manner from an originator to one or more recipients, without establishing voice paths between the originator and recipients, wherein the P2M server, and an optional Voice Mail Server, provide a message storage facility for the multimedia messages. Additionally, a technique based on handset speech buffering permits zero delay call set-ups to be achieved.

Communications methods for allowing groups of individuals, such as family members, to contact one another when no one is available to answer calls to a primary, e.g., family, telephone number are described. Calls which go unanswered are connected to a telephony device capable of storing messages and initiating conference calls. If the calling party is determined to be a family member the telephony device accesses a table of contact information corresponding to the unanswered telephone number and offers the caller the opportunity to hear messages from other family members and / or initiate a conference call to one or more family members who have supplied a contact telephone number. Family members can access and update the table of messages and contact number information by telephone or the Internet. Table information can be accessed and updated remotely thereby providing a method of communicating with family members even when no one is at home.

A portable adapter that provides non-repudiable telecommunications services to bar-code reading hand-held computers and palm-top or tablet-type mobile computers is disclosed. The adapter provides supplemental power supply and processing capacity that supports API communications functions, such as interactive voice recognition, conference calling, data encryption, VoIP packetization and other signal-format conversions that are not implemented on mobile computers. In particular, the device automatically logs IP packet identifiers and DOV dialing and status signals, without the user having access to edit this information, thereby providing a "non-repudiation" record of all communications. The adapter also supports intensive use of the host computer's serial port by supplementing the power available from the host computer's battery, or replacing that battery with a connector. For plant inspection and inventory auditing, ground-based cellular communications are implemented for supporting on-site work, including conference calling to discuss apparent pilferage or imminent safety hazards, and removable WORM recording media for documenting these discussions. For repair shop use, a standard phone jack or 10-base-T connector allows the device to upload engine test-data, with reports or estimates dictated by the mechanic to the repair shop's LAN server. For hospital use, the device includes removable WORM media for logging patient test results and examination reports. The hospital device also uses a dedicated local-area RF or IR transmitter, with location-specific encryption, to protect the privacy of lab reports received by the device, and to limit use of these devices to hospital's own premises.

According to one embodiment of the invention, a method for conducting a conference call between two or more participants is provided. The method includes receiving an indication of a request for text from a participant. The method also includes converting, in response to the indication, any speech of the other participants of the conference call into text. The method also includes sending the text to a device associated with the participant who requested test. The device is operable to display the text.

A handheld communication device and method is provided that facilitates improved device usability. The handheld communication device and method uses a touch screen interface, where the touch screen comprises a proximity sensor adapted to detect object motion in a sensing region, a display screen overlapping the sensing region, and a processor. The touch screen is adapted to provide user interface functionality on the communication device by facilitating the display of user interface elements and the selection and activation of corresponding functions. The handheld communication device and method are configured to display representations of calls on the display screen, and are further configured to initiate conference calls responsive to sensed object motion beginning at a first call representation and continuing toward a second call representation. Thus, a user can initiate a conference call with a relatively simple and easy to perform gesture on the touch screen.

A system and method of providing a subscriber service to service users in a telecommunications network. In networks utilizing Session Initiation Protocol (SIP) control signaling for call setup and control, the SIP REGISTER message is modified to indicate service capability information and optionally a traffic load indication for service providers. The REGISTER message is sent to a modified Presence and Instant Messaging (PIM) server that stores presence information and the service capability information for registered service providers. The PIM server then notifies subscribing service users of the identity of the service provider that is registered on the network. The PIM server may utilize the traffic load information to balance the traffic load between service providers by providing users with the identity of the service provider that is the most lightly loaded.

A system and method are disclosed for enhanced IP conferencing. In one embodiment, the enhanced IP conferencing allows for joining a conference call through a calendaring application. A web page or GUI is created that keeps track of all conference call participants, and monitors who is speaking along with speaking data, tracks the speakers and maintains a condensed transcript of the conference call.

A method and system for fast setup of group Voice over Internet Protocol (VoIP) communications. The method and system provide a lower VoIP signaling overhead and a faster setup-time of group calls such as N-way audio and / or audio-video conference calls. Multi-party VoIP connections are quickly set-up over one or more virtual communications channels established between plural network devices using a session layer application, one or more group transport layer ports and a broadcast destination network address.

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.

A method and a system for supporting rendezvous based instant group conferencing among mobile users allows a conference originator to initiate and setup instant group conferencing to a set of participants even when the participants may not be immediately available to commit to the conference call at the time of its initiation. All negotiations for obtaining commitments from the participants are performed by the system, thus relieving the originator of the time consuming task of negotiating with the participants. The system automatically decides as to when to actually dial out to the participants to setup the conference call after obtaining commitment from all (or a quorum of) required participants.

A real-time interface between the public switched telephone network (PSTN) and an Internet Protocol (IP) network provides voice to data and data to voice conversion between the PSTN and the IP network in a seamless process. The interface, a central communication network, performs Class 5 switching between the PSTN and the IP network, besides providing enhanced services. Receiving a call, the central communication network simultaneously routes the call to a plurality of pre-programmed numbers on the PSTN and on the IP network. The central communication network provides call screening, takes voice messages and converts them to e-mail messages, takes e-mail or facsimile messages and converts them to voice messages. Communication between a PSTN phone on a local PSTN, a computer hooked up to the IP network, a phone hooked up to the IP network by a gateway, a private branch exchange (PBX) on a local PSTN, a wireless communication system with pagers and / or cell phones hooked up to a local PSTN, and facsimile machines on a local PSTN, for example, is provided by the central communication network. Through the central communication network, a computer hooked up to the IP network can exchange voice messages and facsimile messages with a PSTN connected device and conduct conference calling with a plurality of PSTN devices.

A method and apparatus is provided for mixed mode conference call setup. The method includes the steps of inviting users to a mixed mode conference call by providing invitation information from an inviting user to a server and thereby providing invitations from the server to each of the users. The invitation information indicates the users, the modes acceptable for the conference call and the mode accepted by the inviting user. The invitations from the server include information indicating the modes acceptable for the conference call. The users provide a response to the server which includes information indicating a selected mode accepted by each user. The server sets up a conference call where each of the users and the inviting user communicate in the selected modes accepted by each.

A system and method for recording and / or otherwise monitoring IP multimedia sessions. The present invention features a recording and / or monitoring device, referred to hereinafter as “a recording device” for the purposes of clarity only and without any intention of being limiting. The recording device is a participant in the IP multimedia session, although preferably the recording device only receives data for recording and / or otherwise monitoring the session. Therefore, the IP multimedia session is preferably a multi-user session, such as a “conference call” for example, even if data is being provided for recording from only one of the participants in the session.

Provided is a system which takes advantage of the integrated capabilities of different communication devices and employs those capabilities so that participation in any particular conference call is more tightly controlled than possible with current systems. A “vInvitation” programming object is defined that establishes parameters for a particular conference call. A conference call coordinator sends a vInvitation object to each prospective participant of a particular conference call. Each vInvitation is attached to a vCalendar programming object, which includes information related to the scheduling of the conference call. When activated by a receiving device, the vCalendar object automatically enters information about the particular conference call into a prospective participant's electronic calendar. At the appropriate time, the vCalendar object is activated, which in turn activates the vInvitation object. The vInvitation object establishes the corresponding participant's connection to the conference call, maintaining the secrecy of pass code and access number.

A system, method and access gateway for minimizing the number of transcodings of a speech signal during a Voice-over-IP (VoIP) conference call in a packet-switched network in which Tandem Free Operation (TFO) is utilized. The system includes a first gateway connecting the first mobile subscriber to the network, a second gateway connecting the second subscriber, and a third gateway connecting the third subscriber. The second gateway sends a message to the first gateway indicating a speech coding mode being utilized between the second gateway and the second subscriber. The third gateway sends a message to the first gateway indicating a speech coding mode being utilized between the third gateway and the third subscriber. When a three-way conference call is initiated, the first gateway encodes the call path to the second subscriber with the speech coding mode being utilized between the second gateway and the second subscriber. The first gateway also encodes the speech signal for the call leg to the third subscriber with the speech coding mode being utilized between the third gateway and the third subscriber.

A method and apparatus for allocating call resources during a conference call are disclosed. A conference call between a plurality of clients is conducted on a first call resource. The clients communicate media streams to the first call resource for mixing. A media gateway identifies a second call resource to conduct the conference call and mix the media streams, and transfers the conference call from the first call resource to the second call resource without suspending communication of a plurality of mixed media streams received by the clients.

Methods are disclosed for using an Internet server with access to a data base of PSAP (Public Safety Answering Point) jurisdiction boundaries and telephone numbers to track the location of a wireless communication device and to determine a PSAP telephone number to call to request emergency services at the location of said wireless communication device using telephone company provided location information. Methods are disclosed for establishing conference call connections between a person, a PSAP and a remote wireless communication device. Also disclosed are methods for establishing an Explicit Call Transfer call connection that redirects a call from a person to a wireless communication device to the telephone of a PSAP having jurisdiction over the location of said remote wireless communication device. Alzheimer tracking devices, offender monitoring devices, medical alert notification devices, satellite tracking devices and child safety telephone embodiments of the current invention are also disclosed.

In various embodiments, multiple participants (at endpoints) may connect through a video conferencing system. A screen may display other participants in the conference call. For example, multiple participants may be displayed in respective portions of the screen. In some embodiments, each of the participants may use video conferencing systems that support different resolutions. In some embodiments, a transcoder may resolution transcode signals to allow the conference call participants to be displayed on each participant's respective screen at up to the highest possible resolution supported by the respective screen. For example, low resolution video conferencing systems may be displayed on a high definition screen of a high resolution system at the low resolution of the low resolution video conferencing system while another image on the same high definition screen from a different, higher resolution video conferencing system, may be displayed at the higher resolution.

A method for setting up a multicast session between clients. The method includes the steps of assigning a common multicast IP address so that the clients may participate in the videoconference session, and transmitting the common multicast IP address to at least one of the clients.

A conference phone system includes personal headsets and a base unit. The personal headsets individually capture audios of local participants on a conference call (“local audios”) and transmit the local audios in separate and identifiable channels to the base unit. The base unit receives the local audios and transmits the local audios in separate and identifiable audio streams over a network to a network client. For a remote participant on the conference call, the network client reproduces the local audios and indicates one or more participants who are presently speaking. The network client can also virtualize the local audios so that the remote participant can distinguish the participants by their relative positions, whether virtual or actual. The network client uses the audio source identification information of various participants to enable conference features to mute, enhance, or hold private sidebar conversations.