Abstract: A conferencing system provides automated conference moderation at appropriate times to suppress disorder in an on-going conference. According to particular embodiments, a conference bridge provides multi-mode speaker selection to provide automated conference moderation for a disorderly conference.

Abstract: According to one embodiment of the invention, a method for managing communication impairments between Internet Protocol devices is provided. The method includes determining a transmission of a signal comprising a comfort noise, where the signal is transmitted from a first endpoint to a second endpoint. The method also includes sending a notice signal from the first endpoint to the second endpoint indicating that the signal is transmitted. The method also includes suppressing the signal at the second endpoint in response to the notice signal. According to another embodiment of the invention, a method for managing communication impairments between an Internet Protocol (“IP”) phone and an IP device is provided. The method provides sending a status signal to the device indicating that the phone is operating as a speakerphone. The method also includes suppressing the transmission of any comfort noise to the phone in response to the status signal.

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: 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 synchronizing media in a call includes receiving a first input stream of packets of a first media type at a first call resource, generating a first output stream in response to the media streams, receiving synchronization information from a second call resource, and communicating the output stream from the first call resource to a endpoint in synchronization with the second call resource. A video conference bridge includes a first interface operable to receive a video input stream, a processor operable to generate video output, and a second interface operable to receive synchronization information from an audio conference bridge.

Abstract: A server is provided with a transmit buffer, a transcoder, and a redundancy encoder. The server is for transmitting streaming media to a client through a communication network. A control unit senses when the network is temporarily congested, and activates the transcoder and the redundancy encoder. The frames require fewer bits, and thus arrive in the client's de-jitter buffer without interruption, notwithstanding the network congestion. This prevents network jitter.

Abstract: Methods for concealing audible distortions resulting from changes in jitter buffer size include receiving an audio stream, storing the audio stream in a jitter buffer, and determining a pitch period associated with the audio stream. To expand the jitter buffer, a method includes inserting additional audio data that has a duration corresponding to an integer multiple of the pitch period into the audio stream. To contract the jitter buffer, a method includes removing a portion of the audio stream having a duration corresponding to an integer multiple of the pitch period.

Abstract: An audio mixer on a first device receives one or more incoming audio streams. Each of the one or more incoming audio streams has an associated timestamp. The audio mixer generates a mixed audio stream from the one or more incoming audio streams. The audio mixer determines differences in the time base of each of the one or more incoming audio streams and the time base for the mixed audio stream. The audio mixer generates mapping parameters associated with the determined differences and transforms the timestamp of each of the one or more incoming audio streams to a corresponding output timestamp associated with the mixed audio stream according to the mapping parameters. the mapping parameters are provided to a video mixer for similar processing and transformation such that the mixed audio stream is in synchronization with a mixed video stream.

Abstract: A communication system includes a server, a call manager, and a plurality of network endpoints. The network endpoints are assigned to network regions to allow selection of a proper compression/decompression algorithm (codec) for a call between the network endpoints. In one embodiment, the server receives a request for a network address from a network endpoint, and communicates the network address and a network location parameter identifying a network region to the endpoint. In another embodiment, the call manager receives a call request from a network endpoint and determines a network region for the endpoint based on the endpoint's network address. In yet another embodiment, an endpoint determines its own network region based on its network address.

Abstract: According to one embodiment of the invention, a method for managing call requests in a limited bandwidth environment includes receiving a call request from a user, determining if a voice channel on a communication link is available for the call request, and providing a feedback signal for the user indicating when a voice channel will be available.

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: Devices, software and methods are provided for generating aggregate comfort noise for teleconferencing over IP networks. A transcoding component includes a decoder for decoding streams of packets. A summing component has a summer with summing inputs to receive the decoded streams of packets. The summing component has at least one silence flag input, and an additional signaling path is used by the transcoding component to signal to the silence flag input if any of the decoded streams of packets includes a silence identification packet. In another embodiment, the summing component may or may not include the silence flag input, but the device includes an aggregate comfort noise generation component. The aggregate comfort noise may be programmed to be a balanced representation of all background noises.

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: 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: A system enables conferencing of multiple participants within a single voice communication session. A replay module records conference audio during the conference and, at appropriate times, provides this buffered audio to selected participants of the conference.

Abstract: A system enables conferencing of multiple participants within a single voice communication session. A replay module records conference audio during the conference and, at appropriate times, provides this buffered audio to selected participants of the conference.

Abstract: A locally configurable audio IP network appliance provides audio feedback to the installer while in a configuration mode. The appliance includes a normal audio output module, that generates a signal for the speaker driver, which in turn drives a speaker. The appliance also includes a switch that permits, during configuration, its maintenance module to be coupled to the speaker driver instead of the normal audio output module. The maintenance module includes prompts that generate suitable utterances when applied to the speaker. The installer listens to the utterances, while interactively pushing the buttons of a numeric keypad such as a DTMF keypad. A method is for the appliance to present the installer with an option to change the settings of configuration parameters. Presentation is made by announcement through the speaker. The appliance then waits for inputs, such as the keystrokes. When valid inputs are received, then the appliance implements the inputted settings.

Abstract: A method and system for real-time monitoring voice mail during an active call includes generating a duplicate of a calling party voice stream being recorded by voice mail for a called party on a call with a previously connected party. The duplicate calling party voice stream is combined with a previously connected party voice stream to generate a combined voice stream for the called party. The combined party stream is played to the called party during the call.

Abstract: A communication system includes a server, a call manager, and a plurality of network endpoints. The network endpoints are assigned to network regions to allow selection of a proper compression/decompression algorithm (codec) for a call between the network endpoints. In one embodiment, the server receives a request for a network address from a network endpoint, and communicates the network address and a network location parameter identifying a network region to the endpoint. In another embodiment, the call manager receives a call request from a network endpoint and determines a network region for the endpoint based on the endpoint's network address. In yet another embodiment, an endpoint determines its own network region based on its network address.

Abstract: Devices, software and methods are provided for generating aggregate comfort noise for teleconferencing over IP networks. A transcoding component includes a decoder for decoding streams of packets. A summing component has a summer with summing inputs to receive the decoded streams of packets. The summing component has at least one silence flag input, and an additional signaling path is used by the transcoding component to signal to the silence flag input if any of the decoded streams of packets includes a silence identification packet. In another embodiment, the summing component may or may not include the silence flag input, but the device includes an aggregate comfort noise generation component. The aggregate comfort noise may be programmed to be a balanced representation of all background noises.