Example methods and apparatus for providing voice alignment are described. One example method including: obtaining an original voice and a test voice, the test voice is a voice generated after the original voice is transmitted over a communications network; performing loss detection and/or discontinuity detection on the test voice, the loss detection is used to determine whether the test voice has a voice loss compared with the original voice, and the discontinuity detection is used to determine whether the test voice has voice discontinuity compared with the original voice; and aligning the test voice with the original voice based on a result of the loss detection and/or the discontinuity detection, to obtain an aligned original voice and an aligned test voice, the result of the loss detection and/or the discontinuity detection is used to indicate a manner of aligning the test voice with the original voice.

The disclosed system identifies international calling performance issues of a wireless telecommunication network. The system receives network traffic data for international calls including information about call attempts to a country. The system categorizes the country into a major category and a minor category based on the call attempts information. For a subset of countries, and for each key performance indicator in a subset of selected key performance indicators, the system monitors performance using an anomaly detection model to identify an anomaly in network performance, determines an actual value of the key performance indicator for the detected anomaly, and computes a variation value of the determined actual value based on a predicted range of values. The system ranks countries using the computed variation values, to indicate problematic parts of the wireless telecommunication network.

A voice call quality information generation method and an apparatus of a server for use in a wireless communication system are provided. The voice call quality information generation method includes receiving log information from at least one base station and generating voice call quality information using muting information generated based on the log information.

The present invention relates to systems and methods for network labeling in order to enhance real time data transfers. A network for a real time data transfer is identified and predictive models for network performance are compared against to determine if the network is suitable for the data transfer. If so, then the real time data transfer may be completed as expected. However, if the network is predicted to be unsuitable for transmission an alternate means for connection may be suggested. The alternate suggestion may include delaying the data transfer until the network is expected to be in better conditions, connecting to another access point in the network, or switching to another network entirely. During the data transfer, the quality of the network is monitored in order to update the predictive models for the network's quality. Identifiers for the network may be utilized to keep track of the networks. Network signal strength, signal pollution and time may also be tracked in order to identify patterns in the network's performance.

Methods and systems for facilitating a communication session receive a sound signal and determine at least one contributing source to the received sound signal from among a plurality of communication endpoints, wherein each respective communication endpoint of the plurality of communication endpoints is associated with a corresponding participant in the communication session and includes a microphone and loudspeaker. Based on a determination of multiple communication endpoints being co-located, one or more embodiments perform at least one of processing the received sound signal by selectively altering the received sound signal, or initiating an instruction to inhibit further contribution, to the sound signal, from at least one of the co-located multiple communication endpoints.

Responsive to concurrently recording on a plurality of recorders a telephonic interaction constructing a plurality of playlists, and according to faults identified while recording the plurality of playlists evaluating a quantitative quality score for each playlist in the plurality of playlists and presenting the playlists in a prioritized order according to the scores thereof.

A set of user pools can be determined based on location data associated with each device in an audio/video (A/V) conference. A key active user can be determined for each user pool of the set of user pools based on valid audio signals received from each device within each user pool. A determination can be made whether there is feedback within each user pool. Responsive to determining feedback in at least one user pool, speakers of devices within the at least one user pool can be disconnected except for the key active user device within each respective user pool.

Communication systems and communication methods are provided, by which load of a communication network when measuring a voice communication quality can be suppressed. A test signal for voice communication quality evaluation is transmitted to a communication terminal apparatus via a communication network, and a received voice data encoded by a predetermined scheme for the test signal received with the communication terminal apparatus is received from the communication terminal apparatus. A received voice signal is generated from the received voice data received from the communication terminal apparatus, and a evaluation value of voice communication quality is calculated by comparing the generated received voice signal with the test signal before transmitting.

A method and a device for improving audio processing performance are provided. In some embodiments, the method includes: acquiring audio data in an audio call in a terminal; processing the audio data by utilizing an audio processing algorithm, to obtain a characteristic parameter of the audio data; analyzing the processing quality of the audio data according to the characteristic parameter of the audio data; and if the processing quality of the audio data does not reach a preset quality standard, optimizing the audio processing algorithm by utilizing optimization data matching the terminal.

Example embodiments disclosed herein relate to an evaluation of perceptual delay impact on a conversation. A method of evaluating perceptual delay impact on a conversation is disclosed. The method includes detecting a conversational event between a first participant and a second participant, the conversational event including a plurality of speech segments; in response to the conversational event being detected, obtaining at least one conversational parameter for the first and second participants, the at least one conversational parameter indicating a relationship between the plurality of speech segments in the conversational event; and determining a quality of the conversation according to a model based on the at least one conversational parameter, the model characterizing the quality with the at least one conversational parameter. A method of obtaining a model for evaluating perceptual delay impact on a conversation is also disclosed. Corresponding systems and computer program products are also disclosed.