Abstract: A method and apparatus for wideband voice and optional data conferencing over a telecommunications network channel between at least two wideband communications devices. An exemplary method comprises establishing an audio link, verifying wideband capability between the at least two wideband communications devices, training modems of the at least two wideband communications device to line conditions, and adjusting the telecommunications connection line conditions between the communications devices. Once a wideband connection has been established, audio and data may be simultaneously exchanged.

Abstract: Disclosed herein is a system and method for storing and transmitting a videoconference system name (identifier) in the languages of double byte character sets. Although described in terms of a videoconferencing system, the techniques described herein have applicability to various other systems that rely upon inter-language compatibility, including telephonic communication such as cell phones or IP phones, as well as various computer network applications in which computers having different default languages are in communication. In a preferred embodiment, the system identifier is stored in a text or ASCII format rather than native (binary) format. Additionally, a system employing a preferred embodiment of the teachings described herein allows a particular system to have multiple site names in multiple different languages. This allows such a system to interface and exchange identifier data with other systems having different language settings during a caller ID exchange.

Abstract: A conference link between devices in teleconference system at one site is disclosed. The linked devices include video conference units, speakerphones or stand-alone loudspeakers. Audio data can be transmitted through the conference link between devices. Audio data processing can be performed in either a video conference unit or a speakerphone. The microphones and loudspeakers in the video conference unit may be eliminated. The microphones and loudspeakers in the speakerphone are used. Other data, for example directories of dialing information, may also be transmitted through the conference link and the data processing may be allocated among processors in devices connected by conference links. The conference link may be wired or wireless, analog or digital. The devices may be linked by conference link in parallel or series. A dialing program can adapt the dialing stream with the locations of the near end site and the dialed far end site.

Abstract: A multipoint communication system uses Internet protocol trunking to facilitate communication between media control units (for sending and receiving multipoint communication signals between end-point devices), a media gateway (for translating between non-Internet protocol multipoint communication signals and Internet protocol communication signals), and a controller (for establishing and controlling a multipoint communication session between the end-point devices). In addition, a multimedia gateway (for use in a multipoint communication system) is described that incorporates an interactive voice response unit through which users of non-Internet protocol devices (connected to the multimedia gateway) interact to establish a communication session with a multipoint communication system.

Abstract: In audio processing for an audio or video conference, a terminal receives audio packets having transform coefficients for reconstructing an audio signal that has undergone transform coding. When receiving the packets, the terminal determines whether there are any missing packets and interpolates transform coefficients from the preceding and following good frames. To interpolate the missing coefficients, the terminal weights first coefficients from the preceding good frame with a first weighting, weights second coefficients from the following good frame with a second weighting, and sums these weighted coefficients together for insertion into the missing packets. The weightings can be based on the audio frequency and/or the number of missing packets involved. From this interpolation, the terminal produces an output audio signal by inverse transforming the coefficients.

Abstract: During a conference, a multipoint control unit (MCU) designates priority and non-priority endpoints. The MCU forms priority audio from the priority endpoint and sends that audio to the other endpoints at a normal level. However, the MCU forms non-priority audio from the non-priority endpoint and handles that audio based on whether the input audio from the priority endpoint is from speaking or not. If the priority endpoint's audio indicates a participant at that endpoint is speaking, then the MCU sends the non-priority audio to the other endpoints at a reduced level. Designation of which endpoint has priority can be based on which endpoint has a current duration of audio indicative of speech that is longer than other endpoints. Alternatively, the designation can be based on which endpoint is currently presenting content during the conference or based on a mix of speech audio and content presentation.

Abstract: A video conference system provides “in the room” telepresence to near end participants. The video conference system includes a frameless or bezelless display device placed in front of a front wall for displaying edge-to-edge 3D images of far end participants. Color of the near end front wall is configured to be the same as the color of the far end rear wall. As a result, images displayed on the display device can merge or blend into the near end front wall, giving the near end participants the perception that the far end participants are actually in the near end conference room. Brightness of the faces/bodies of the near end participants can also be adjusted to match the brightness of the face/bodies of the images of the far end participants as they appear on the display device—therefore enhancing the in the room perception of the far end participants.

Abstract: Disclosed herein are methods, systems, and devices for improved audio, video, and data conferencing. The present invention provides a conferencing system comprising a plurality of endpoints communicating data including audio data and control data according to a communication protocol. A local conference endpoint may control or be controlled by a remote conference endpoint. Data comprising control signals may be exchanged between the local endpoint and remote endpoint via various communication protocols. In other embodiments, the present invention provides for improved bridge architecture for controlling functions of conference endpoints including controlling functions of the bridge.

Abstract: Disclosed herein is a middleware server for interfacing a Multipoint Conferencing Unit (MCU) with a communication system and/or a management system within the context of an organization. The middleware server uses modular architecture, making the middleware server highly adaptable to different types of communication systems and management systems. The middleware server is useful for interfacing a MCU with communication devices including Plain Old Telephone Service (POTS) telephones, Internet Protocol telephones (IP-phones), and multimedia endpoints and with management applications such as exchange servers and databases. The middleware server is useful for managing multimedia sessions according to organizational policies.

Abstract: A videoconferencing system includes a first videoconferencing unit coupled to a network and associated with a first instant messaging identity. The first videoconferencing unit obtains a second instant messaging identity and automatically sends a request instant message requesting videoconferencing connection information to the second instant messaging identity. A second videoconferencing unit is coupled to the network and is associated with the second instant messaging identity. The second videoconferencing unit receives the request instant message and automatically returns a response instant message including videoconferencing connection information to the first instant messaging identity. The first videoconferencing unit receives the response instant message and automatically obtains the videoconferencing connection information from the response instant message.

Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. Digital samples of an audio signal are transformed from the time domain to the frequency domain. The resulting transform coefficients are coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. At least two transforms are applied on an audio signal, each with different transform periods for better resolutions at both low and high frequencies. The transform coefficients are selected and combined such that the data rate remains similar as a single transform. The transform coefficients may be coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: A new speakerphone wherein a loudspeaker is oriented to face the listeners of an audio conference is disclosed. New speakerphones which have a bass loudspeaker and at least one tweeter loudspeaker wherein the tweeter loudspeaker is oriented towards listeners of an audio conference are also disclosed.

Abstract: Devices and techniques for overcoming lost packets and avoiding congestion when conducting media conferences over packet switched networks are described herein. To avoid the problem of lost packets, redundant information is inserted into the media stream that permits the receiver to reconstruct any lost packets from the redundant information. Congestion avoidance techniques include adjusting the bitrate of the media stream to find the highest bitrate that can be supported without packet loss due to congestion. When increasing the bitrate to a higher rate, the additional bits can come from the redundant information used for lost packet recovery so that any lost packets caused by network congestion will not adversely affect the bitstream.

Abstract: A system and method is disclosed for adapting a continuous presence videoconferencing layout according to interactions between conferees. Using regions of interest found in video images, the arrangement of images of conferees may be dynamically arranged as displayed by endpoints. Arrangements may be responsive to various metrics, including the position of conferees in a room and dominant conferees in the videoconference. Video images may be manipulated as part of the arrangement, including cropping and mirroring the video image. As interactions between conferees change, the layout may be automatically rearranged responsive to the changed interactions.

Abstract: A system and method particularly useful in handling the video of a “large” conference is disclosed. The system and method initially defines certain conferees as panel members, which can provide input video signals to the multipoint control unit and can receive output video signals or a composite layout for the conference from the multipoint control unit. The system and method also initially defines audience members, which only passively receive the output video signals or layout from the multipoint control unit and do not have their video signals incorporated into the layout of the videoconference. In one aspect, an audience terminal can request to participate in the conference. By making an appropriate request, the audience terminal can, if system resources are adequate, participate fully as a panel member and may supply video signals to the conference for at least some portion of time.

Abstract: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

Abstract: Systems, methods, and devices for improving speaker performance with an acoustic damped port are disclosed. In accordance with various embodiments of the present invention, a damping material is placed around a vented frame of a speaker driver, and substantially covers or fills the vents of the frame. In some embodiments, the damping material results in improved impedance matching with acoustic delay, without required dimensional tuning (as with conventional ports). In some embodiments, it also reduces the air velocity gradients minimizing higher order frequency distortion components. In some embodiments, the damping material also acts as an absorber of energy by coupling to the sound wave. In some embodiments, low-frequency performance is improved in a small-scale design.

Abstract: A method, microphone module and a system for full-duplex audio system implementing full duplex audio, audio echo cancellation and audio codec. The audio signals from the microphones and to the loudspeakers are encoded to reduce the transmission bandwidth requirement during wired or wireless transmission and decoded when needed for further processing. Each microphone has transceivers to receive signals fed to loudspeakers within the microphone module. An audio echo canceller is installed within each microphone module to generate echo free audio signal for further signal processing. Problems due to the combination of lossy codec and echo cancellation are avoided.