Abstract: Techniques are provided in which an audio signal for transmission to a receiving device is acquired at a network device. The audio signal is analyzed for an audio feature to be suppressed or enhanced during playback of the audio signal at the receiving device. The audio feature is detected based on the analysis. The audio signal is encoded for transmission over a network to the receiving device. The encoded audio signal is transmitted to the receiving device. A packet is generated comprising an audio feature descriptor indicating where in the audio signal the audio feature is located to enable the receiving device to suppress or enhance the audio feature during playback of the audio signal. The packet comprising the audio feature descriptor is transmitted to the receiving device.

Abstract: Techniques are provided in which an audio signal for transmission to a receiving device is acquired at a network device. The audio signal is analyzed for an audio feature to be suppressed or enhanced during playback of the audio signal at the receiving device. The audio feature is detected based on the analysis. The audio signal is encoded for transmission over a network to the receiving device. The encoded audio signal is transmitted to the receiving device. A packet is generated comprising an audio feature descriptor indicating where in the audio signal the audio feature is located to enable the receiving device to suppress or enhance the audio feature during playback of the audio signal. The packet comprising the audio feature descriptor is transmitted to the receiving device.

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 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 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 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: 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: 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: 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: 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 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.