Method and apparatus for monitoring the status and location of personnel using a remote timer. In some embodiments, a method includes execution of a subject application (app) on a network accessible device. A countdown timer of a remote server is initiated to monitor a time interval, the remote server communicating with the network accessible device over a network. Geo positions of the network accessible device are monitored during the monitored time interval. The remote server sends a notification to a monitoring device responsive to at least one of the following events: the monitored time interval ends without the remote server receiving an authorized personal identification (PID) value; or the remote server receives an unauthorized PID value prior to the conclusion of the monitored time interval. In further embodiments, a notification is sent based on an out-of-bounds geo position of the network accessible device during the time interval.

Various embodiments include methods for determining when to resynchronize voice-over IP (VoIP) communications of the wireless device. Various embodiments may include storing one or more first call characteristics of the VoIP communications between the wireless device and a first base station, detecting whether the VoIP communications are transferred from the first base station to a second base station, analyzing one or more second call characteristics of the VoIP communications between the wireless device and the second base station, determining whether the one or more second call characteristics differ from the one or more first call characteristics, determining, within a total wait time, whether no active voice frames will be transmitted across an uplink, or no active voice frames will be provided to audio decoding, resynchronizing the VoIP communications in response to determining that the one or more second call characteristics differ from the one or more first call characteristics.

A congestion control method in an IP telephone system includes processing of acquiring the number of simultaneous calls on one line shared by a plurality of IP telephones, processing of acquiring lengths of call duration of respective calls, and communication parameter setting processing of setting communication parameters that contribute to communication quality of respective calls on the basis of the number of simultaneous calls and the call duration. The call duration of the first call is longer than the call duration of the second call. The communication parameter setting processing includes processing of setting communication parameters such that the communication quality of the first call is worse than the communication quality of the second call in a case where the number of simultaneous calls exceeds a threshold value.

Systems and methods are described herein for determining dropped call rates (DCR) for various communications networks, such as IP Multimedia Networks (IMS), which include Voice over LTE (VoLTE) networks. For example, the systems and methods utilize data (e.g. abnormal cause codes) generated by layers of the IMS networks, such as a Session Initiation Protocol (SIP) layer of the IMS network, when determining dropped call rates for IMS networks.

A service request for communication services for communication clients is received. In response, a communication service network is set up to support the communication services. Routing metadata is generated for each of the communication clients. The routing metadata is to be used by each of the communication clients for sharing service quality information with a respective peer communication client over a light-weight peer-to-peer (P2P) network. The routing metadata is downloaded to each of the communication clients. A communication client may exchange service signaling packets or service data packets over the communication service network. When the communication client determines that there is a problematic region in a bitstream received from the communication server, the communication client can request a peer communication client for a service quality information portion related to the problematic region.

A packet identifying method includes: connecting to an IP telephone device under test and querying to a current device information of the IP telephone device under test; determining if there is any instruction set in a database matched with the current device information, wherein the database includes a plurality of instruction sets, and each instruction set corresponds to existing device information of an IP telephone device; and executing a packet identifying procedure by using an executing instruction set.

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.

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.

In one embodiment, a device obtains call transcripts from an online application. The device detects cues within the call transcripts that are indicative of poor user experience. The device generates, based in part on the cues detected within the call transcripts, a model trained to predict poor user experience from network path telemetry for the online application. The device causes traffic for the online application to be routed along a particular network path, based on a prediction by the model.

Method and apparatus for monitoring the status and location of personnel using a remote timer. In some embodiments, a system is provided having a network accessible device with a controller, a memory and a native sensor. A subject application (app) is installed as native programming by an original manufacturer of the network accessible device and has access to the native sensor. A timer circuit of a remote server initiates a monitored elapsed time interval responsive to execution of the subject app, and sends a notification signal to a monitoring device responsive to a conclusion of the monitored elapsed time interval without having received a disable communication from the network accessible device. The disable communication is generated by a biometric input supplied by a user of the network accessible device via the native sensor, with the biometric input serving as a personal identification (PID) for the user.