Nortel Networks Corporation

Avaya Inc.

Genband US LLC

Rockstar Bidco, LP

Gregory Thomas Osterhout

Moni Manor

R Alberto Villarica

William Janning

Mark Sosebee

Trip Ingle

Dany Sylvain

Patrick N Sollee

Christopher L Jessen

Raheel Yuhanna

Craik R Pyke

David R Creech

Parameshwaran Krishnan

Jon Louis Bentley

Stephen R Whynot

Philip K Edholm

Lin Lin

Peter Tarle

Kim Brian Holmes

Jim A McAlear

James Allan McAlear

Julie R Simons

Samuel A Weaver, Iii

Alex Beck

A method and system for allocating bandwidth among communication sessions are disclosed. In one embodiment, a method includes performing a first allocation to a first communication session and performing a second allocation to a second communication session. The method includes performing a reallocation of bandwidth from the first communication session to the second communication session based on a predetermined criteria.

Method and system for communication session bandwidth control based on actual bandwidth usage

A video conference is set up by a video conference system between three or more video terminals. Each video terminal in the video conference sends its video stream to the video conference system. The video conference system in turn sends out the video streams to video terminals to set up a video conference in the traditional manner. A video controller receives an indication of a change in how at least one of the other video terminals is displaying a video stream from one of the video terminals. Based on the indication of change, a change to the bit rate of the video stream is made. This results in the video conference system be able to send reduced bandwidth based on who is viewing the video conference.

Video bandwidth management system and method

Once an active video conference is set up and a user is viewing the active video conference at a video terminal, the video terminal looks for different events that indicate a change in focus of the user to or from the active video conference. For example, the user brings up another application and starts using the application or the user has minimized a window that is displaying the active video conference. The video terminal sends a change of focus message based on the event to a video conference bridge or another video terminal that is streaming the active video conference to the user. The video conference bridge/video terminal processes the message and changes video portion of the stream of the active video conference based on the message. The result is that there is improved use of bandwidth between the video terminal and the video conference bridge/video terminal.

Changing bandwidth usage based on user events

The present invention provides a redundancy mechanism for packet-based communication environments without requiring localized redundancy of session control entities. These session control entities are generally referred to as service nodes, which may act as proxies on behalf of the communication terminals being served. Instead of having a dedicated backup service node, other active service nodes, which primarily serve different groups of communication terminals, are configured to provide backup support when another service node fails. In operation, a communication terminal will periodically attempt to communicate with its primary service node. When the communication terminal detects that the attempts to communicate with the service node are unsuccessful, the communication terminal will attempt to register with a selected alternate service node, the alternate service node being a currently active service node primarily supporting another group of communication terminals. The alternate service node can then accept and support the communication terminal.

Geographic redundancy in communication networks

A method and apparatus for handling subscriber services from different sources without requiring interaction with different devices. The present invention also provides a method and apparatus for receiving multimedia services using session initiation protocol (SIP) messages received from a source, such as a server on a network. In one embodiment, a set-top device is provided in which a SIP control is provided to register a subscriber with a server using SIP messages. A connection to a network, such as the Internet, is provided to communicate with the server. The set-top device is placed in line with the audio and video source to control audio and video streams presented to the subscriber. The set-top device is also configured to receive voice calls from a public switched telephone network (PSTN). Further, the set-top device includes a remote control mechanism, such as an infrared (IR) transmitter and receiver to receive input from the subscriber and to control other devices using IR signals. Also, subscriber services customized for a particular subscriber may be directed to the subscriber at a location when the subscriber registers with the server through a set-top device at that location using SIP. Services, such as, for example, call screening, channel guides, video on demand, and Web browsing are provided through the set-top device and messaging with a server using SIP messages.

Session initiation protocol enabled set-top device

A method and apparatus of communicating over a data network includes providing a user interface in a control system for call control and to display information relating to a call session. The control system communicates one or more control messages (e.g., Session Initiation Protocol or SIP messages) over the data network to establish a call session with a remote device in response to receipt of a request through the user interface. One or more commands are transmitted to a voice device associated with the control system to establish the call session between the voice device and the remote device over the data network. A Real-Time Protocol (RTP) link may be established between the voice device and the remote device.

Controlling voice communications over a data network

In one embodiment, a service node will recognize an attempt to initiate a call from a first terminal to a second terminal, and automatically provide information to media clients associated with the first and second terminals such that a media session can be readily established between the media clients in association with the call. The service node may be configured to interact with telephony switches that support the first or second terminals, directly or indirectly via a signaling adaptor. In a second embodiment, the service node will recognize an attempt to initiate a call and will route the call to a gateway, which is controllable by the service node. Once the call is sent to the gateway, the service node may provide instructions to the gateway for routing or otherwise processing the call.

Convergence of circuit-switched voice and packet-based media services

A system for an adaptive network phone includes a SIP-enabled handset or other telephone device to be connected to a host computer, for instance via a USB connection. The host computer may initiate a call event when the user picks up the SIP device and wishes to make a call, in one regard by testing line conditions for a data connection to the Internet or other network to determine if a packet-based call will meet minimum noise, delay or other criteria of the user. If the conditions are satisfactory, the call may proceed as a packet network transmission, otherwise the call may revert to the public switched telephone network for fallback completion. The SIP-enabled handset may also interface to multimedia functions in the host computer, for instance to permit paging, answering machine, cordless operation or other features.

System and method for network phone having adaptive transmission modes

A method of redirecting a call from a data processing system to another address. In a preferred embodiment, a notice of an incoming call received from a server at a data processing system. This notice may include caller identification information as well. The user of the data processing system is prompted for an address to which the user wishes the call to be redirected. The user then identifies and sends to the server a new address to which the incoming call is to be redirected. The server then redirects the call to the new address.

Portable call management system

A communications system includes call devices that are capable of communicating over a packet-based data network. The packet-based data network may be according to an Internet Protocol (IP), for example. The call devices are capable of performing call communications, such as according to a Session Initiation Protocol (SIP) or an H.323 Recommendation. Each of the call devices includes a user interface that is customizable to one or multiple configurations depending on which user is using the call device. User identification information is communicated from the call device to a proxy or server that has access to a database. The identification information is used to locate configuration information associated with the user. The corresponding user configuration information is then retrieved from the database and communicated back to the call device to configure the call device user interface.

Configuring user interfaces of call devices

A communications system includes a packet-based data network coupled to various network elements, including a gateway that provides ports to various peripheral devices. One type of peripheral device includes a Universal Serial Bus (USB) device. A network element coupled to the data network may establish Session Initiation Protocol (SIP) sessions with the gateway. Once a SIP session is established, communications may occur between the network element and the peripheral device. SIP messaging is exchanged between the network element and the gateway. USB commands and data are exchanged between the gateway and the USB device. The gateway converts between the SIP messaging and the USB commands and data.

Growing community of inventors

Coppell, TX, United States of America

Gregory Thomas Osterhout