A method and system are provided for opportunistically initiating a multimedia call between two or more devices such as smart speakers at a time when users of the respective smart speakers are available to participate in a call. A server system receives an instruction from a first smart speaker associated with a first user to establish a real-time communication session with a second smart speaker. In response to that instruction, the server system monitors availability of both the first user and the second user at their respective smart speakers. On determining that both the first user and the second user are available at the same time, the server system initiates the real-time communication session between the first smart speaker and the second smart speaker.

According to one embodiment, a system includes a base station wirelessly connected to a first handset and a second handset, the base station including a first communication port configured to create a wireless connection to a first mobile telephone and a second mobile telephone via Bluetooth, where the first mobile telephone uses a first mobile telephone network and the second mobile telephone uses a second mobile telephone network.

A method and system are provided for opportunistically initiating a multimedia call between two or more devices such as smart speakers at a time when users of the respective smart speakers are available to participate in a call. A server system receives an instruction from a first smart speaker associated with a first user to establish a real-time communication session with a second smart speaker. In response to that instruction, the server system monitors availability of both the first user and the second user at their respective smart speakers. On determining that both the first user and the second user are available at the same time, the server system initiates the real-time communication session between the first smart speaker and the second smart speaker.

The present invention provides an IP telephone and the like capable of preventing a loss of the beginning part of a talk. A communication apparatus according to the present invention includes: a memory (110) configured to store communication data with a communication destination; and a control unit (120) configured to transmit and receive the communication data to and from the communication destination in each first duration and store the transmitted and received communication data in the memory (110), then output, after a communication session with an external recording apparatus that records the communication data is established, the communication data stored in the memory (110) to the external recording apparatus in each second duration that is shorter than the first duration.

Methods and apparatus for deriving and/or generating, user presence information from communication signals monitored by a SBC or PRI Gateway and aggregating and/or distributing the user presence information. An exemplary method embodiment includes the steps of storing a plurality of user profile records, each of the user profile records mapping contact information for a user to one or more user identifiers corresponding to the user; receiving first signaling information; determining a first user corresponding to the first signaling information by comparing at least a first portion of the information contained in the first signaling information to at least a first portion of the information contained in the plurality of user profiles; and generating presence status information for the first user and communicating the first presence status information to a presence server as presence status information corresponding to a first user identifier, said first user identifier corresponding to said first user.

An audio bridge unit is provided for connecting a conventional analog page/party station in an analog page/party system to IP-based page/party stations that employ multicast Voice over Internet Protocol (VoIP) technology to support page/party communications between analog page/party stations and the VoIP page/party stations. The audio bridge unit converts page and/or party audio line(s) to multicast channel(s). The audio bridge unit analog interfaces for its respective page and/or party lines, and each analog interface is assigned to an independent multicast channel on which it will transmit and receive full duplex audio between an analog page/party station and one or more IP-based page/party stations. The audio bridge unit provides the ability to configure the multicast group, port and Time to Live (TTL) setting for each of the analog channels associated with the page and/or party lines.

A Radio Over Internet Protocol (RoIP) device can be configured to provide automatic determination of input gain, output gain, external power, and COR signaling settings for an audio port via a guided step-by-step user experience. According to some embodiments, input settings can be calculated based on the user speaking into the device and output settings are configured by the user adjusting the volume of predetermined audio being played. According to some embodiments, users can then label the settings, so that they can be copied to other audio ports using the same type of radio. Among other things, the embodiments provided herein increase the speed and ease with which the optimum audio port settings can be configured on a RoIP gateway device

Blocking of undesirable voice over internet protocol (VOIP) communications is disclosed. A communication screening service initiates operations to block a threat posed by a VOIP communication upon receiving the communication from a gateway server. The communication may include an audio/video conversation and/or an audio/video conference. Next, metadata and content of the communication is analyzed to detect a threat, such as a scamming scheme and/or a phishing scheme, from a sender of the communication. A rejection of the communication is generated to disrupt the threat associated with the communication. The rejection is transmitted to the gateway server to prompt the gateway server to block the communication.

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format.

A terminal can have at least one audio interface for at least one microphone, at least one audio interface for at least one speaker, an operator control module and at least one interface for connecting to at least one telephony- and/or IP-telephony-compatible terminal. The terminal can be configured as a media hub for the at least one telephony- or IP-telephony-compatible terminal. The terminal can have a media processing unit for mixing and a media hub controller or controlling media streams received by the at least one telephony- or IP-telephony-compatible terminal and the at least one telephony- or IP-telephony-compatible terminal can be operated through the operator control module of the telecommunication terminal. A method for controlling media streams through a media hub controlled integrated into a telecommunication terminal is also disclosed.