At&t Mobility II LLC

At&t Intellectual Property I, LP

Jeffrey A Krinsky

James Gordon Beattie, Jr

Haywood S Peitzer

Falguni Sarkar

Bahman Khosravi-Sichani

Jennifer Yates

Leslie Paul Leonard

Linh Nguyen

In one embodiment, a method includes identifying, by a packet analyzer, one or more silence packets within a network and initiating, by the packet analyzer, a replacement of the one or more silence packets with one or more performance silence packets. The one or more performance silence packets are transmitted between a first node of the network and a second node of the network during a silence period. The method further includes receiving, by the packet analyzer, information associated with the one or more performance silence packets; and analyzing, by the packet analyzer, a connection between the first node of the network and the second node of the network using the information associated with the one or more performance silence packets.

Systems and methods for analyzing performance silence packets

A radio access network system is described that determines a signal metric associated with a user equipment or device. The user equipment device can implement an altered transmission policy. The altered transmission policy can alter a strength of transmissions by increasing power consumption per transmission, increasing a length of timer per transmission, and altering other parameters of transmissions. The altered transmission policy can also alter an error correction policy. The error correction policy can indicate that error correction transmissions are to be decreased. The altered transmission policy can be implemented until the signal metric changes to a more desirable level.

Latency reduction and range extension system for radio networks

In one example, the present disclosure describes a device, computer-readable medium, and method for improving media quality at the network edge by encoding the media into a digital stream that is carried, end-to-end, to a peer edge device. For instance, in one example, a method includes initiating, by a first computing device, a connection to a second computing device, selecting, by the first computing device, a codec for encoding data into a data stream, wherein the codec is selected such that the data stream can be decoded by the second computing device without being transcoded by an intermediary, encoding, by the first computing device, the data into the data stream using the codec, and sending, by the first computing device, the data stream to the second computing device.

Codec selection for end-to-end communication without intermediate transcoding

Aspects of the subject disclosure may include, for example, a method including receiving advertisements from a remote advertising content provider and pre-loading the advertisements at a communication device. An advertisement is selected for presentation at a presentation device, based on a user's advertising viewing history, the duration of the advertisement, the device location, the device environment, the user's demographic profile, or the user's activities. The presentation device presents the advertisement responsive to the trigger message. The effectiveness of the advertisement is determined, and the advertisement viewing history is updated; the advertisement is selected based in part on the effectiveness of a past presentation of the advertisement. Other embodiments are disclosed.

Method and apparatus for managing advertisements

Performance sensitive selection of audio signal data is disclosed. A first rank value for the processed audio signal data can be based on a level of fidelity to an input audio signal. A second rank value for the processed audio signal data can be based on a performance metric of a processing scheme for processing the input audio signal into the processed audio signal. A third rank for the processed audio signal data can be based on an indication of a level of satisfaction for performance of the processed audio signal by a receiving device. A fourth rank for the processed audio signal data can be based on a network condition for a network conveying the processed audio signal to a device for performing the processed audio signal. Selection of the processed audio signal data can be based on the first, second, third, fourth rank, or combinations thereof.

Performance sensitive audio signal selection

In a particular implementation, a method includes dynamically adjusting a size of a buffer of a receiving terminal based on a comparison of a signal quality metric associated with a wireless communication channel between the receiving terminal and a transmitting terminal to a signal quality threshold. The method also includes storing a first set of data packets of a plurality of data packets in the buffer. The first set of data packets are received at the receiving terminal from the transmitting terminal. The method further includes outputting reconstructed speech based on the first set of data packets and replacement packets that are generated based at least in part on the first set of data packets. The replacement packets are associated with a set of unreceived data packets of the plurality of data packets.

Dynamic jitter buffer size adjustment

A method includes determining, at a processor of an automated attendant system, whether a customer service representative is available to speak with a caller. The method also includes scheduling a call back time in response to a determination that the customer service representative is not available to speak with the caller. The call back time indicates a time where the customer service representative will call the caller. The method further includes providing the call back time to the caller and receiving an availability flag from the caller at the call back time. The availability flag indicates whether the caller is available to receive a call from the customer service representative. The method also includes providing a call back signal to the customer service representative based on the availability flag. The call back signal indicates to the customer service representative whether to call the caller at the call back time.

Enhanced automated attendant system

A method includes storing data packets in a buffer at a receiving terminal. The data packets are received by the receiving terminal from a transmitting terminal. The method also includes determining a dropped packet rate at the receiving terminal, a packet transmission latency between the receiving terminal and the transmitting terminal, a signal quality at the receiving terminal, and a use application at the receiving terminal. The method further includes dynamically adjusting a size of the buffer based on the dropped packet rate, based on the packet transmission latency, based on the signal quality, and based on the use application.

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Woodinville, WA, United States of America

Jeffrey A Krinsky