Certain aspects of the disclosure are directed to customized communication lists for data communications systems using high-level programming. According to a specific example, a data communications server is configured to interface with a remotely-situated client entity using a first programming language. The data communications server includes a communication control engine that is configured to provide a private branch exchange (PBX) for the client entity. The communication control engine is configured and arranged to identify client-specific sets of control data derived from instructions written in a second programming language that is compatible with the first programming language, and to control communication routing by determining whether to route incoming data communications in response to a comparison of at least one feature of the incoming data communications telephone communications and the client-specific sets of control data.

A method of transmitting data determines a measure of consecutive packet loss in a network; a ratio of a number of data packets and a number of error correction packets is selected in dependence on the measure. A stream of data packets is generated, and a stream of error correction packets is generated in dependence on the stream of data packets such that the proportion of error correction packets generated to the data packets generated is commensurate with the selected ratio.

Processing of communications routed by an IPBX server are disclosed. At least one processing circuit is communicatively coupled to an Internet-Protocol Private Branch Exchange (IPBX) server that is configured and arranged to route calls for a plurality of agents in a communications/call center. The processing circuit is configured to receive communications event messages from the IPBX server for communications routed by the IPBX server, generate, during a communication to a first agent of the plurality of agents, a set of data metrics including communications summary metrics based on the communications event messages; and redirect, during the communication to the first agent, the communication to a second agent of the plurality of agents in response to the set of data metrics satisfying a set of criteria indicated in a policy.

Disclosed is a system for telephones by providing an improved and streamlined user experience and enhanced fail over mechanisms. A decentralized system managed through a web site which allows for continued operation even when the primary systems fail includes a mechanism for restoring the primary systems automatically when they become available again. Phones connect to two PBX systems at the same time, one local and one at a remote location. The two PBX systems synchronize configuration data and media files between them. The website can also be used to manage any number of systems allowing any size organization to manage every phone system in their organization from a single interface.

A method and system for triggering RF reconfiguration of wireless communication system. RSRP-SINR pairs determined respectively for locations in a region of interest are mapped to predicted MOS of voice communication per location, using a prediction engine trained based on actual correlated measurements of RSRP, SINR, and MOS. Locations having threshold low MOS and threshold low RSRP but not threshold low SINR are identified and clustered to identify an area where coverage strength should be increased in an effort to help improve voice-call quality. And locations having threshold low MOS and threshold low SINR but not threshold low RSRP are identified and clustered to identify an area where interference should be reduced in an effort to help improve voice-call quality. An engineering trouble ticket or other signal could then be generated to trigger associated network reconfiguration.

A channel-based voice call method, including: accepting a matching request of a client, acquiring a matching result of the client from a matching pool, and returning the matching result to the client; and acquiring a call matching acceptance instruction of the client, allocating a temporary channel and locking the temporary channel, and sending a successfully accepted matching message, a temporary channel number and a channel password to the client, such that the client enters the temporary channel based on the successfully accepted matching message, the temporary channel number and the channel password to make a call.

Apparatuses and methods concerning routing of calls in an IPBX server are disclosed. As an example, one apparatus includes an IPBX server configured to route VoIP calls for a plurality of end-users having respective VoIP-enabled circuit devices communicatively coupled to the IPBX server via a communications network. A call data processing circuit is communicatively coupled to the IPBX server. The call data processing circuit is configured to receive call event messages corresponding to a plurality of end-users from the IPBX server and generate call models from the call event messages. The call data processing circuit also generates call summary metrics from the call models. In response to an input query, the call data processing circuit evaluates the call summary metrics for a parameter of interest specified in the input query to generate data for the parameter of interest.

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 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.

A system and method is provided for calculating session initiation protocol (SIP) key performance indicators (KPIs) for multimedia communication in a network accessing network packets monitored over time. The method includes accessing network packets monitored over time, detecting a SIP response message of the monitored network packets, and accounting for the detected SIP response message in a time window of a series of consecutive time windows that includes the SIP response message's associated observed time. The SIP response message has an associated observed time at which the monitored SIP message was observed during monitoring. The method further includes pairing a SIP request message expected to be paired with the detected SIP response message with the detected SIP response message and accounting for the paired SIP request message in the time window with which the detected SIP response message is accounted. The method further includes calculating a KPI for the time window using the detected SIP response message and its paired SIP request message that are accounted for in the time window.

Embodiments described herein provide a dynamic voice over Internet Protocol (VoIP) audio quality management mechanism in real time, e.g., when a VoIP call is ongoing. Specifically, when a VoIP call has unsatisfactory audio quality, e.g., due to packet loss, jitter, etc., the dynamic VoIP audio quality management mechanism may redirect the VoIP traffic from the previous endpoint that initiates the VoIP session to a different endpoint within the same carrier. Upon the endpoint redirection, a new call leg is established, allowing re-negotiation or re-configuration of VoIP parameters. The re-negotiated or re-configured VoIP parameters may then be used to conduct the remainder of the VoIP call to improve the audio quality.