Devices and methods are provided that permit interoperability between Radio over Internet Protocol (RoIP) devices and land mobile radio (LMR) devices. In one device according to the invention, a networked radio includes a wireless network transceiver, an LMR signal connector, and an analog-to-digital converter/digital-to-analog converter (ADC/DAC). A computer processor that inputs data from or outputs data to the ADC/DAC includes a sound architecture and a server. The server includes a digital audio buffer, a voice-activity detector, and a daemon. The daemon is configured to (1) collect data on one or more variables affecting a best-choice determination for transmitting a communications signal intended for retransmission to one or more LMR devices, and (2) make the best-choice determination based on the data collected.

A system and method for presenting caller ID information related to a caption assisted telephone call incorporating a first party Network appliance, a first party telephone service/terminal, and a relay linkable to the first party Network appliance via the Internet.

A telecommunication terminal that integrated with a wireless access point is provided. According to one embodiment, a telecommunication terminal includes a local area network (LAN) port, a processor, an Internet Protocol (IP) phone unit, a wireless access point unit and a housing. The LAN port is connectable to an enterprise computer network via an Ethernet cable. The processor runs a host operating system (OS). The IP phone unit is implemented as an application that is loaded and run within the host OS. The wireless access point unit facilitates access to the enterprise computer network by wireless devices within a coverage area of the wireless access point unit by providing wireless connections to the wireless devices and is implemented as an application that is loaded and run within the host OS. The housing encloses the processor and the LAN port.

Examples are disclosed for placing an outbound telephony call using a mobile telephony device as a proxy to make the call on behalf of a smart speaker device. At a communications server, it is determined whether a mobile telephony device is in proximity of a smart speaker device. When the mobile telephony device is in proximity of a smart speaker device and attempts to place a telephony call, the communications server establishes a communications link between the communications server and the smart speaker device. The communications server may then dial the telephone number sent by the mobile telephony device and establish a communications link between the communications server and a device associated with the dialed telephone number. The communications server may then join the communications link between the communications server and the smart speaker device with the communications link between the communications server and a device associated with the dialed telephone number to create a communications session.

A software application enables a mobile device to make and receive voice calls, video calls, and text messages over a data network using VoIP. The application may run in the background and intercept outgoing and incoming calls. When the outgoing or incoming call is to/from another user of the application, the application may launch an appropriate screen and make/receive the call or text via Voice-over-Internet Protocol (VoIP). Otherwise, the call or text functionality may be performed by native functionality of the mobile device via a GSM network.

Transmission mode selection operations include determining that a first wireless device out of a plurality of wireless devices connected to an access node qualifies for a beamforming transmission mode, determining that a pair of wireless devices out of the plurality of wireless devices qualifies for a multi-user multiple-input multiple-output (MU-MIMO) transmission mode, comparing a first total throughput of the access node using the beamforming transmission mode with a second total throughput of the access node using the MU-MIMO transmission mode, and selecting the downlink transmission mode from the first total throughput and the second total throughput that has a highest total throughput.

A system is provided that includes a first network interface for a first network type and a second network interface for a second network type that is different from the first network type. The system also includes at least one processor configured to cause the system to perform operations. The operations may include obtaining, from the first network interface, audio from a communication session with a remote device established over the first network and obtaining an indication of a communication device available to participate in the communication session and direct audio obtained from the communication session to a remote transcription system. The operations may also include directing the audio to the second network interface for transmission to the communication device, obtaining transcript data from the remote transcription system based on the audio, and directing the transcript data to the second network interface for transmission to the communication device.

The present disclosure relates to an adaptive jitter buffer for buffering audio data received via a network. The adaptive jitter buffer comprises an adaptive audio sample buffer, which comprises an adaptive resampler that receives a number of audio samples of the audio data and that outputs a first number of audio samples, which are resampled from the received number of audio samples according to a resampling factor, an audio sample buffer that buffers audio samples, wherein the outputted first number of audio samples are written to the audio sample buffer during an input access event and a second number of audio samples are read from the audio sample buffer during an output access event, and an audio sample buffer fill quantity controller that controls a fill quantity of the audio sample buffer based on controlling the resampling factor of the adaptive resampler.

Interactive services are enhanced by intercepting, in a telecommunications network, a session initiation protocol message used to initiate a communication. The session initiation protocol message is intercepted prior to terminating at a recipient computer configured to provide interactive services for the communication. Based on predetermined rules, supplemental information is determined to provide to the recipient computer of the communication. The supplemental information is inserted into the session initiation protocol message prior to the terminating at the recipient computer. The session initiation protocol message with the supplemental information is routed to the recipient computer that provides the interactive services for the communication. The interactive services are provided based at least in part on the supplemental information.

Apparatus and methods concerning simulation of call quality are disclosed. In an example embodiment, computing server is communicatively coupled to a VoIP server. The computing server includes a communication circuit configured to receive a first set of VoIP data including audio of a VoIP call routed by the VoIP server. The computing server also includes a processing circuit configured to characterize a post-transmission quality state of the first set of VoIP data. The processing circuit is also configured to generate a second set of VoIP data including audio that is different from the audio of the VoIP call and data including characteristics indicative of the post-transmission quality state of a first set of VoIP data. The processing circuitry may configured to use the second set of data to provide security, protect the confidentiality and privacy, and/or monitor changes of behavior/quality for different audio CODECs, encryption, bit-rate, etc.