Lucent Technologies Inc.

Alcatel Lucent

Alcatel-lucent USA Inc.

Nokia Solutions and Networks Oy

Ahmed A Tarraf

Xin Wang

Martin Howard Meyers

Mahmoud Sherif

Carl Francis Weaver

Doh-Suk Kim

Binshi Cao

Sherif Sherif

Kamakshi Sridhar

Yang Yang

Colin Leon Kahn

Edward Grinshpun

Mohammad Hossein Zarrabizadeh

Jonathan Dean Segel

Charles Payette

Donald J Youtkus

Philip Lamoureux

Jorge Rodriguez

Mahmoud R Sherif

Kevin G Tracy

Al-Nasir A Premji

Joseph P Hoffbeck

Jian Yi

Donald Joseph Youtkus

The present disclosure generally discloses an adaptive bitrate streaming support capability configured to support adaptive bitrate streaming of content. The adaptive bitrate streaming support capability is configured to support adaptive bitrate streaming of content via a wireless access network including an application scheduler and a wireless access device. The application scheduler receives an adaptive bitrate streaming flow and sends the adaptive bitrate streaming flow toward the wireless access device based on scheduling of the adaptive bitrate streaming flow at the application scheduler. The wireless access device determines feedback information associated with the adaptive bitrate streaming flow and provides the feedback information associated with the adaptive bitrate streaming flow to the application scheduler. The application scheduler determines scheduling of the adaptive bitrate streaming flow, for transmission toward the wireless access device, based on the feedback information associated with the adaptive bitrate streaming flow.

Quality-of-experience for adaptive bitrate streaming

A wireless communication system () includes backhaul congestion control. Whenever congestion is detected on the backhaul, a flow control message sets a window size to a reduced window size to reduce or eliminate packet drops and to facilitate alleviating the congestion condition. A disclosed example includes detecting backhaul congestion and responsively reducing an available backhaul receiver buffer space factor used for setting the window size for a subsequent backhaul transmission. In a disclosed example, the buffer space factor is gradually increased back toward an actual available buffer space as the congestion status improves.

Backhaul congestion control for wireless communication networks

In one embodiment, a plurality of queues are established corresponding to each of a plurality mobiles, respectively. Data packets associated with the plurality of mobiles are received at a network element. Each received data packet is assigned to a corresponding one of the plurality of queues. A determination operation is performed on a selected one of the plurality of queues to determine whether or not to send one or more packets of each queue from the network gateway element. The determination operation is performed based on a comparison between a packet size of a head packet and an adaptive transmission threshold value associated with the selected queue. The adaptive transmission threshold value is a value that is changed based on an amount data sent from the selected queue. The head data packet being an earliest received data packet in the selected queue.

Method of managing user traffic to prevent aggressive users from abusing network resources

The invention includes a method and apparatus adapted for controlling congestion in a Radio Access Network (RAN). In one embodiment, a method includes monitoring packets received at a first network element of the RAN for an indication of congestion, where the packets are associated with a quality-of-service class, and, in response to detecting an indication of congestion in at least one of the received packets, signaling congestion status information from the first network element of the RAN toward a second network element of the RAN that is adapted for applying congestion control for the quality-of-service class. In one embodiment, a method includes receiving, at a first network element of the RAN, from a second network element of the RAN, congestion status signaling indicative of a congestion condition associated with a quality-of-service class, and applying congestion control for the quality-of-service class in response to receiving the congestion status signaling indicative of the congestion condition.

Method and apparatus for providing congestion control in radio access networks

A video analysis technique includes correlating frames from a processed video with frames from a pre-processed, original video. A linear approximation of a relationship between the correlated frames is determined. A disclosed example includes determining a linear approximation that maximizes the number of processed video frames that fit into the linear approximation. The linear approximation and whether any frames do not fit within the linear approximation is then used to provide quality information for analyzing a quality of the processed video.

Video quality analysis using a linear approximation technique

A method and apparatus for use in suppressing acoustic echo in a target speech signal being transmitted through a packet-based communications network uses frame energy estimation applied to the target speech signal and to a reference speech signal. The method or apparatus estimates one or more reference speech energy levels in one or more reference packets based on one or more of the speech parameters generated by the speech encoding of the reference signal; estimates a target speech energy level in a target packet based on one or more of the speech parameters generated by the speech encoding of the target signal; compares the target speech energy level to one or more reference speech energy levels; and detects an echo in the target speech signal based on the comparison of the target speech energy level to the one or more reference speech energy levels.

Method and apparatus for the detection and suppression of echo in packet based communication networks using frame energy estimation

In a method for echo suppression or cancellation, a reference voice packet is selected from a plurality of reference voice packets based on at least one encoded voice parameter associated with each of the plurality of reference voice packets and the targeted voice packet. Echo in the targeted packet is suppressed or cancelled based on the selected reference voice packet.

Packet based echo cancellation and suppression

A method of determining at least an estimate of media latency for push-to-talk communications includes determining a first time associated with transmitting a first packet from a push-to-talk transceiver device. In a disclosed example, the same device receives a second packet that corresponds to the first packet and includes an indication of the first time, a second time associated with receipt of the first packet by another device such as a push-to-talk server and a third time associated with transmission of the second packet to the push-to-talk transceiver device. That same device determines a fourth time associated with receipt of the second packet. In a disclosed example, the push-to-talk transceiver device uses a relationship between the first time, second time, third time and fourth time for determining at least an estimate of media latency associated with the push-to-talk communication corresponding to the first and second packets. A plurality of latency test procedures are disclosed.

Determining latency associated with push-to-talk communications

This invention is directed toward improving the subjective voice quality of a mobile to mobile phone call having tandem vocoder processing by modifying the spectrum of the voice signal before it is processed by the second vocoder to compensate for digital distortion which is generated by the second vocoder. An adaptive filter can be used to modify the spectrum of the voice signal. With this invention, the voice quality of a call from a first mobile phone to a second mobile phone has a quality that is substantially similar to the voice quality of a call from a mobile phone to a desk phone.

Mobile to mobile digital wireless connection having enhanced voice quality

It has been recognized that in a wireless communication system certain frames of encoded speech data transmitted between a base station and a mobile unit, or between a mobile unit and a base station, are more critical than others. A frame may be determined to be erased by the receiving base station or mobile unit due to noise or interference over the wireless transmission medium. If an erased frame cannot be recreated from one or more preceding frames, then it is more critical than a frame that can be recreated by an extrapolation of data from one or more preceding frames. Accordingly, on a frame-by-frame basis, each frame in a sequence of frames is identified as being critical or non-critical. Each frame that is identified as being critical is then transmitted in a manner that is more robust than the manner in which non-critical frames are transmitted to decrease the likelihood that a receiver will determine that the frame is erased. In a CDMA system, a current frame is identified as being critical by forming a weighted sum of the differences between corresponding frame parameters that represent the current frame and frame parameters that represent a previous frame. That weighted sum is compared with a threshold. If the weighted sum exceeds the threshold, then the current frame is classified as being a critical frame and is transmitted at a higher output level than non-critical frames are transmitted.

Method and apparatus for improving the quality of speech signals transmitted over wireless communication facilities

A method for conveying information-carrying teletype signals over a wireless telecommunication system by encoding the information into a repetitive signal. Teletype information is derived from part of a received signal encoded as a repetitive signal which part has been determined not to have been adversely affected by the wireless telecommunication system.

Method for conveying TTY signals over wireless communication systems

An apparatus and method for conveying teletype signals over a wireless telecommunication system. When the apparatus receives a teletype signal, it derives information carried by said signal and represents said information as at least one message. The at least one message and/or the teletype signal is transmitted over the wireless telecommunication system. When the apparatus receives a signal from the wireless telecommunication system that it determines to be a teletype signal, the apparatus of the present invention decodes the teletype information from part of the received signal determined as not having been adversely affected by the telecommunication system.

Apparatus and method for conveying TTY signals over wireless telecommunication systems

Telecommunications processing is applied to a reference signal to generate a signal under test. A fidelity measure is generated characterizing the fidelity of the signal under test relative to the reference signal. A control signal is generated from the fidelity measure, where the control signal is used as a feedback signal to adjust the telecommunications processing. In one embodiment, the reference signal is a speech signal and the signal under test is a decoded speech signal generated by encoding, transmitting, and decoding the reference speech signal. The fidelity signal is an average mean opinion score (MOS) and the control signal is used to control the speech decoding processing. For example, the speech decoding processing may involve a speech decoder followed by a post filter, and the control signal is the cut-off frequency of the post filter. Such an embodiment may be applied to real-time speech processing to adjust the speech decoding processing in the receiver of a telecommunications system that includes a speech encoder in a transmitter and a communications network. Intermittently, the transmitter encodes reference signals that are also known to the receiver. The receiver performs the processing of the present invention to update its speech decoding processing to optimize speech quality.

Feedback control system for telecommunications systems

A fidelity measuring apparatus (103). The fidelity measuring apparatus (103) measures the fidelity of a system (101). A preprocessor (109) receives an input signal. The preprocessor (109) also receives an output signal from the system (101). The preprocessor (109) extracts a feature set from the output signal. An artificial intelligence tool (111), trained in the relationship between fidelity and the feature set, generates an output that is a measure of the fidelity of the system (101). A fidelity measuring apparatus (103b) includes a preprocessor (138) and two or more artificial intelligence tools (111-N). The artificial intelligence tools (111-N) are independently trained by input signals from different sources. An averaging circuit (140) averages the output of each artificial intelligence tool (111-N). The fidelity measuring apparatus (103b) operates to produce accurate fidelity measurements independent of the input signal, the source of the input signal and the technology of the system.

Apparatus and method for measuring the fidelity of a system

The present invention provides a method and apparatus for measuring at least one signal characteristic. Initially, a set of features is selected which characterize a signal. An intelligent system, such as a neural network, is trained in the relationship between feature sets and signal characteristics. The selected feature set is then extracted from a first input signal. The extracted feature set from the first signal is input to the trained intelligent system. The intelligent system creates an output signal based on the feature set extracted from the first input signal. This output signal is then used to characterize the input signal. In one embodiment, the invention assesses voice quality, typically as expressed in MOS scores, in a manner which accurately corresponds to the analysis of human evaluators. For voice signals processed by voice coders, the present invention provides a measurement technique which is independent of various voice coding algorithms and consistent for any given algorithm.

Growing community of inventors

Bayonne, NJ, United States of America

Ahmed A Tarraf