Shoretel, Inc.

Michael A. Dolan

Edwin Basart

Pascal R Crausaz

David B Rucinski

Dale C Tonogai

Michael S W Tovino

Thomas E Miller

Greg T Sueoka

Nam Do

Dieter Werner Rencken

Jeffrey W Ridley

Dave Dix

David Cornelius

Michael Dolan

Michael Joseph Delpier

A method for joining a teleconference includes receiving a schedule for the teleconference at a conferencing system. The schedule may include a start time for the teleconference and one or more conference rooms for attendees to use during the teleconference. A first message is sent from the conferencing system to the at least one communication device. The first message indicates to the at least one communication device to provide a user notification associated with the teleconference. A second message is received at the conferencing system. The second message is sent in response to an action by an attendee of the teleconference. The second message indicates to the conferencing system to initiate the teleconference with the at least one communication device or to join the at least one communication device to the teleconference.

Joining a teleconference

The cloud computing VoIP system includes a software as a service (SaaS) site and one or more additional sites. The SaaS site includes a gateway and a call manager and is coupled for communication across an IP network and the PSTN. The call manager includes a media monitoring module, a bypass module, a PSTN control module and a GUI module. The SaaS site handles and controls voice and video calls over the IP network and the PSTN for endpoints at the one or more additional sites. In another embodiment, the cloud computing VoIP system is also coupled to a cellular network and the endpoints include an associated cellular phone. The cloud computing VoIP system also includes a method for falling back to a secondary network, namely the cellular network, when there is insufficient quality on a primary network.

Cloud VoIP system with bypass for IP media

A system, method and graphical user interface for determining a speaker's position and a generating a display showing the position of the speaker. In one embodiment, the system comprises a first speakerphone system and a second speakerphone system communicatively coupled to send and receive data. The speakerphone system comprises a display, an input device, a microphone array, a speaker, and a position processing module. The position processing module is coupled to receive acoustic signals from the microphone array. The position processing module uses these acoustic signals to determine a position of the speaker. The position information is then sent to other speakerphone system for presentation on the display.

Speaker position identification and user interface for its representation

A system for exchanging information for improved call quality includes an exchange engine for switching between networks or communication channels and a monitoring module for determining quality of communication over a network or communication channels and initiating switching in response to poor call quality. The monitoring module may be responsive to input from the user or to call quality as determined for example by statistical measurement. The exchange engine identifies the network or communication channel in use for the current call; determines the call quality for a second network or communication channel not being used for the current call; determines whether call quality of the second network is better than the first network; and switches the current call between a first user and the second user from the first network to the second network if the call quality of the second network is better than the first network.

Phone URL exchange for improved call quality

A system and method for providing PBX functionality to a mobile phone includes a private branch exchange having a mobile phone proxy. A call control path is established between the mobile phone and a private branch exchange. The call control path is a direct trunk line, a mobile voice network, an IP data network, a mobile data network or a 4G network. The private branch exchange receives a request to perform the PBX function. The private branch exchange uses the call control path to perform the PBX function.

Mobile phone integration with a private branch exchange in a distributed telephony system

Routing an incoming call for an IP phone system is disclosed. A location data for a mobile device is received. A way point based at least in part on the location data is determined. A state for a user of the IP phone system is determined based at least in part on the way point. An incoming call is routed to the IP phone system based at least in part on the state.

IP phone routing and call initiation based on location information

A system and for exchanging information to enable embodiment IP communication comprises a first endpoint corresponding to a first user, a second endpoint corresponding to a second user, and an information exchange (PURLX) unit. The PURLX unit comprises: a processor, a table and a phone URL exchange engine. The processor is configured for communication over the communications channel with a first endpoint and a second endpoint. The table stores information about the first user and the second user including a first association between the first user and one or more endpoint device addresses and a second association between the second user and one or more endpoint device addresses. The table is coupled for communication with the processor. The exchange engine establishes IP communication between the first user and the second user using information from the table. The exchange engine is coupled to retrieve information from the table and store information in the table and coupled to the processor for communication with the first endpoint and the second endpoint.

Phone URL exchange for unified communications

A button box provides additional keys that can be used in conjunction with a telephone. In one embodiment, a button box is configured by associating it with a telephone and by programming its keys. When a key on the button box is activated, a key press event is sent from the button box to the associated phone. The phone treats the key press event as if a key on the phone itself had been activated and forwards the event to the voice switch appliance associated with the phone. (Alternatively, the key press event is sent from the button box directly to a voice switch appliance.) The button box's user interface is controlled by a voice switch appliance.

Phone-independent key expansion module

Telephone

A system and method for falling back to a secondary network when there is insufficient quality on a primary network is disclosed. In one embodiment, a call manager has a media monitoring module, a bypass module, a PSTN control module and a GUI module. Once a call has been established between endpoints using a primary IP network, the media monitoring module monitors for degradation in call quality. If degradation in call quality is detected, the GUI module presents an interface to the user that allows selection of the secondary network or the primary IP network for continuation of the call. If the user selected the secondary network, the bypass module is activated to transfer the data communication to a secondary network such as a PSTN while the call control continues to be over the primary network.

User activated bypass for IP media

A system and method for using a PSTN as a fallback for the IP media when the IP network cannot pass the media acceptably and while continuing to use the IP network for call control and other features. In one embodiment, a call controller has a media monitoring module, a bypass module and a PSTN control module. Once a call has been established between endpoints using a primary IP network, the media monitoring module monitors for degradation in call quality. If degradation in call quality is detected, the bypass module transfers the data communication to a secondary network such as a PSTN while the call control continues to be over the primary network. The present invention also includes a variety of methods including a method for falling back to a secondary network and a method for dynamically switching between a primary and a secondary network to ensure call quality.

PSTN bypass for IP media

A telephone comprising one or more speaker apertures, a microphone indicator disposed on a user-facing surface of the telephone, and one or more microphone apertures disposed proximate to the microphone indicator on a surface of the telephone other than the user-facing surface, where at least one of the microphone apertures is operable to allow audio to pass from the user to a microphone disposed in the telephone, and where the microphone indicator is arranged to direct the user to speak in a direction toward at least one of the microphone apertures.

Telephone microphone

A computer system (a method) for exchanging information such as a Universal Resource Locator (URL) or a Universal Resource Identifier (URI) over a communication channel in general and over a telephone communications channel in particular in a unified communications system is disclosed. In one embodiment, the system comprises a phone URL exchange unit configured to receive a phone URL exchange request from a first endpoint, to determine the URL to be sent and URL encoding method based on the request, to encode the URL into audio data or as a part of the communications control signals using the URL encoding method and to send the encoded data to a second endpoint over the telephone communications channel. The phone URL exchange unit is further configured to receive audio data or the communications control signals embedded with a URL and to decode the received data to reveal the URL.

Phone URL exchange

A system and method for determining a speaker's position and a generating a display showing the position of the speaker. In one embodiment, the system comprises a first speakerphone system and a second speakerphone system communicatively coupled to send and receive data. The speakerphone system comprises a display, an input device, a microphone array, a speaker, and a position processing module. The position processing module is coupled to receive acoustic signals from the microphone array. The position processing module uses these acoustic signals to determine a position of the speaker. The position information is then sent to other speakerphone system for presentation on the display. In one embodiment, the position processing module comprises an auto-detection module, a position analysis module, a tracking module and an identity matching module for the detection of sound, the determination of position and transmission of position information over the network. The position processing module comprises a position display module and a position translation module for receiving position information from the network and generating a user interface for display. The present invention also includes a method for determining speaker position and presenting position information.

Growing community of inventors

Los Altos, CA, United States of America

Edwin Basart