Patent classifications
H04M7/0084
Voice quality assessment system
A new audio quality assessment system includes an assessment system running in a receiver system of a VoIP communication system. The new audio quality assessment system determines an accurate MOS of a VoIP call within a time window. The audio quality assessment system determines an effective PLC counter, a PLC impact factor, an effective AS counter, an AS impact factor, a network impact factor, a codec type of the received voice packets, a bitrate of the received voice packets, an initial MOS from a configured codec-bitrate MOS table, and determines the accurate MOS based on these data. The determined MOS is more accurate and efficiently obtained since it is based on efficiently collected statistics of the receiver system's modules and a pre-configured codec-bitrate MOS table.
Transmissions with common location dependent control information for machine type communication (MTC) data and mobile broadband user equipment (UE) data
A base station transmits a signal transmission to a mobile broadband (MBB) user equipment (UE) device and to a machine type communication (MTC) device that is in close proximity to the MBB UE device where the signal transmission includes control channel and a data channel. The control channel includes common location dependent control information that applies to the MTC device and the MBB UE device because they are at the same location.
Systems, methods, devices and arrangements for cost-effective routing
A variety of methods, systems, devices and arrangements are implemented for assessing and/or controlling call routing for Internet-based (e.g., VoIP/VioIP) calls. According to one such method, endpoint devices are used to monitor and/or assess the call-quality. The assessment is sent to a centralized server arrangement and call-routing is controlled therefrom. Endpoint devices employ a decentralized testing mechanism to further monitor and assess call quality. Aspects of call quality are analyzed and attributed to endpoint devices and/or local connections or networks to distinguish intermediate routing issues from local/endpoint issues.
Method, telephone, telecommunication system and device for controlling power consumption of a telephone
The invention relates to a method for controlling the power consumption of a telephone (10), (310). A power saving mode is automatically switched on or off for the telephone (10), (310), depending on: at least one given timespan, within which the telephone (10) is predicted not to be used or another terminal (380).
Technique for acquiring and correlating session-related information from an internet protocol multimedia subsystem
A technique for acquiring and correlating session-related information from an Internet Protocol Multimedia Subsystem, IMS, is described. The technique comprises the acquisition of control plane information from control plane signalling tapped at an IMS control entity, the acquisition of user plane information from user plane traffic tapped at an IMS user plane entity, and the acquisition of context information from tapped signalling between the IMS control entity and the IMS user plane entity. The acquired context information permits to correlate the control plane information and the user plane information acquired for a particular session.
COMPUTER-PROGRAMMED TELEPHONE-ENABLED DEVICES FOR PROCESSING AND MANAGING NUMEROUS SIMULTANEOUS VOICE CONVERSATIONS CONDUCTED BY AN INDIVIDUAL OVER A COMPUTER NETWORK AND COMPUTER METHODS OF IMPLEMENTING THEREOF
In some embodiments, the present invention provides for a computer-implemented method, including: causing, by a specifically programmed computer call management communication system, to transform, over a computer network, computing devices of users, into corresponding specialized call management devices, by having each computing device to execute a specialized call management client software application being in electronic communication with the specifically programmed computer call management communication system over the computer network by utilizing SIP; where the specialized call management client software application generates specialized graphical user interfaces configured to allow each user to concurrently initiate and maintain, over the computer network, a plurality of voice communications of distinct types with other users, by, for example, allowing each user to independently and dynamically divert, in real-time, any voice communication of any type to any audio device associated with a corresponding specialized call management device of such user.
CLOUD-BASED DEPLOYMENT OF COMMUNICATION SERVICES
A method, system and computer program product for cloud-based deployment of communication services. The method includes obtaining data related to an on premise communication system, and determining a multiplicity of tasks to be executed for migrating the on premise communication system to a cloud-based communication system. Each task from the multiplicity of tasks is associated with a type, a configuration, and an owner. Information of a multiplicity of user attempts to change the configuration during or following execution of some of the multiplicity of tasks of a particular type is collected. When the multiplicity of user attempts exceeding a predetermined amount, an owner of a task associated with the configuration is notified, and when approved by the owner, further tasks of the particular type yet to be executed are modified to conform to the configuration.
Analysis of data metrics in IPBX networks
Apparatuses and methods concerning routing of calls in an IPBX server are disclosed. As an example, one apparatus includes a first processing circuit communicatively coupled to an IPBX server. The IPBX server is configured to generate call event messages for the VoIP calls routed by the IPBX server. The first processing circuit is configured to generate call summary metrics from the call event messages. The call summary metrics and/or related data indicate respective sets of call data for participants of the calls routed by the IPBX server. A second processing circuit is configured to identify organizations associated with the participants of the calls. The second processing circuit aggregates the call related data to assess needs of the organization.
SITE LINK TESTER VIA UNIQUE PHONE EMULATION
Remote on-demand site link testing is provided. A site link tester (SLT) system includes an SLT connected to a customer's VoIP phone system. The SLT is configured to communicate with a front end client application operating remotely on a user's computing device. The packet-capture application receives instructions from the client application to perform a packet capture in association with the SLT's network interface and/or to operate as an emulated VoIP endpoint and conduct a test call (e.g., to confirm the customer's VoIP system's compliance with 911-associated legislation or to troubleshoot a VoIP issue). Results of the packet capture may be sent to the client application and analyzed for remotely diagnosing and troubleshooting VoIP-related problems. Using the SLT system, the technician is enabled to perform 911-associated legislation compliance and diagnose VoIP issues on-demand from a remote location, which can reduce or eliminate the need for a technician to be on-site.
Hybrid Cloud PBX
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.