Abstract: According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Abstract: A customer premises router identifies contacts based on web sites visited and searches conducted by a user device. The customer premises router also detects phone calls initiated by a user device (e.g., a Voice over IP telephony device), and identifies an association between a phone call to an identified called destination (e.g., a destination telephone number) and an identified contact, based on a prescribed association attribute. The identified associations are stored in nonvolatile memory. Any subsequent web request by a user device and related to the contact causes the customer premises router to supply to the user device the identified association specifying the contact and the identified called destination, enabling a user of the user device to determine the identified called destination for the contact, even if the visited web sites do not provide the identified called destination.

Abstract: A method for executing a video conference is provided that includes receiving one or more audio streams associated with a video conference from one or more end points and determining an active speaker associated with one of the end points. Audio information associated with the active speaker may be received at one or more media switches. One or more video streams may be suppressed except for a selected video stream associated with the active speaker, the selected video stream propagating to one or more of the media switches during the video conference. The selected video stream may be replicated such that it may be communicated to one or more of the end points associated with a selected one of the media switches.

Abstract: In one embodiment, a method can include: (i) establishing an internet protocol (IP) connection; (ii) forming a buffered version of a plurality of voice frame slices from received audio packets; and (iii) when an erasure is detected, performing a packet loss concealment (PLC) to provide a synthesized speech signal for the erasure, where the PLC can include: (a) identifying first and second pitches from the buffered version of the plurality of voice frame slices; and (b) forming the synthesized speech signal by using the first and second pitches, and more if needed, followed by an overlay-add (OLA).

Abstract: According to an embodiment of the present invention, an apparatus for performing video conferencing is provided that includes an I-frame injector element operable to intercept I-frame requests from one or more end points and to attempt to service the I-frame requests such that at least a portion of the requests are prevented from propagating back to an originating sender. In more specific embodiments, when a receiver endpoint sends a fast video update (FVU) request upstream, it is intercepted by the I-frame injector element and rather than passing the FVU request to the sender the I-frame injector element replaces a next P-frame from the sender with an I-frame, whereby the I-frame is constructed so that when decoded, it matches the P-frame that it replaced. In still more detailed embodiments, the I-frame injector element operates in one of three modes that are associated with bandwidth parameters.

Abstract: In one embodiment, a method can include: (i) establishing an internet protocol (IP) connection; (ii) forming a buffered version of a plurality of voice frame slices from received audio packets; and (iii) when an erasure is detected, performing a packet loss concealment (PLC) to provide a synthesized speech signal for the erasure, where the PLC can include: (a) identifying first and second pitches from the buffered version of the plurality of voice frame slices; and (b) forming the synthesized speech signal by using the first and second pitches, and more if needed, followed by an overlay-add (OLA).

Abstract: According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Abstract: In one embodiment, a method that includes receiving a plurality of packet streams input from different endpoints, packets of each stream including encrypted and unencrypted portions, the unencrypted portion containing audio power level information. The audio power level information contained in the packets of each of the packet streams is then compared to select N packet streams with loudest audio. The N packet streams are then decrypted to obtain audio content, and the audio content of the N packet streams mixed to produce one or more output packet streams. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.

Abstract: According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Abstract: A method for a client to receive streaming media over a network includes receiving data having the streaming media encoded therein and storing the received data in a de-jitter buffer thereby increasing a fullness of the de-jitter buffer. The method further includes, when the fullness reaches a fill level, initiating play out of the stored data from the de-jitter buffer, and changing the fill level while playing out the stored data.

Abstract: A conferencing system provides automated recording of conference media using a recording module. The recording module connects with conferences and records these conferences automatically. In certain circumstances, such as when all key invitees have joined a conference, the recording module stops recording and deletes recorded media for the conference.

Abstract: A system includes multiple packet-based communication devices that support audio output and other suitable functionality and an acoustic space management device that manages these devices to provide telephony services within acoustic spaces. These devices can coordinate their actions to provide audio services, such as conferencing, paging, broadcasting, and other appropriate audio services.

Abstract: A method for executing a video conference is provided that includes receiving one or more audio streams associated with a video conference from one or more end points and determining an active speaker associated with one of the end points. Audio information associated with the active speaker may be received at one or more media switches. One or more video streams may be suppressed except for a selected video stream associated with the active speaker, the selected video stream propagating to one or more of the media switches during the video conference. The selected video stream may be replicated such that it may be communicated to one or more of the end points associated with a selected one of the media switches.

Abstract: Devices, softwares and methods generate, in real time, indexing metadata for select portions of a telephone conversation or conference. The indexing metadata is generated responsive to inputs received while the conversation is being recorded live. The inputs are either by a user pressing a soft key on a telephone device, or by a voice conference bridge determining who is the dominant speaker in a multi-party conference.

Abstract: A method and system for call answer while connected to voice mail includes generating real-time packets for transmission of a message toward a voice mail system. Generation of the real-time packets is interrupted upon a call answer by a party generating the message. Generation of the real-time packets for transmission of the message toward the voice mail system is resumed after a delay associated with the call answer. An indication is provided to the voice mail system that the real-time packets generated subsequent to the delay follow the real-time packets generated prior to the delay in the message by a time amount less than the delay.

Abstract: A method for a client to receive streaming media over a network includes receiving data having the streaming media encoded therein and storing the received data in a de-jitter buffer thereby increasing a fullness of the de-jitter buffer. The method further includes, when the fullness reaches a fill level, initiating play out of the stored data from the de-jitter buffer, and changing the fill level while playing out the stored data.

Abstract: A communication system includes a destination that receives voice samples and a voice parameter generated by a source. The destination uses the voice samples and voice parameter to reconstruct voice information in response to a packet loss. The destination may reconstruct voice information from multiple sources.

Abstract: In one embodiment, a method that includes receiving a plurality of packet streams input from different endpoints, packets of each stream including encrypted and unencrypted portions, the unencrypted portion containing audio power level information. The audio power level information contained in the packets of each of the packet streams is then compared to select N packet streams with loudest audio. The N packet streams are then decrypted to obtain audio content, and the audio content of the N packet streams mixed to produce one or more output packet streams. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.

Abstract: Devices and methods are provided for minimizing the startup delay of streaming media transmitted through networks. A server maintains a portion of the media stream stored in an initial burst transmit buffer. At startup, the stored portion is transmitted at a rate higher than the fixed frame rate, exploiting the full available bandwidth. The initial burst transmission fills up the de-jitter receive buffer at the receiving end faster, thereby shortening the startup delay. Then transmission is switched to the regular rate, from the regular buffer. A variable bit rate transcoder is optionally used for the data of the initial transmission. The transcoder diminishes the size of these frames, so they can be transmitted faster. This shortens the start up delay even more. A receiver has a buffer with a fill level started at a value lower than a final value. This triggers the beginning of play out faster, further shortening the delay time.

Abstract: A customer premises router identifies contacts based on web sites visited and searches conducted by a user device. The customer premises router also detects phone calls initiated by a user device (e.g., a Voice over IP telephony device), and identifies an association between a phone call to an identified called destination (e.g., a destination telephone number) and an identified contact, based on a prescribed association attribute. The identified associations are stored in nonvolatile memory. Any subsequent web request by a user device and related to the contact causes the customer premises router to supply to the user device the identified association specifying the contact and the identified called destination, enabling a user of the user device to determine the identified called destination for the contact, even if the visited web sites do not provide the identified called destination.