Various methods, apparatuses/systems, and media for delivering a multimedia communication to a thin client device are provided. A processor receives multimedia communication data from a telephony device over a network and transfers the multimedia communication data to a switching device. The processor also establishes a connection between a browser, embedded within the thin client device, and a server. The processor delivers and terminates the multimedia communication data to the thin client device using the browser embedded within the thin client device after establishing the connection between the browser and the server.

Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a 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.

The method performed by a gateway between a LAN and a WAN that includes a platform of the VoIP telephone service, is characterized in that it includes a step of activating the VoIP telephone service on detecting a connected state of a terminal connected to the gateway.

Embodiments of the present invention are directed to a method, system and apparatus for causing a communication client to join a media-over-packet communication session. The method comprises receiving from a communication client an indication indicative of a desire of the communication client to establish a communication session, the communication client being registered in association with a user account. The method further comprises determining if there exists at least one active communication session established with at least one other communication client registered in association with the user account. Responsive to determining that there does exist at least one active communication session, the method further comprises causing the communication client to join at least one of the at least one active communication session.

A configurable, multi-function IP-based communication product has a touch screen display and common hardware set form factor that can be configured for use in any of a plurality of operational modes and can be ruggedized for use in harsh environments. The common hardware set form factor enables the IP-based communication product to be optimized for commercial use, that is, easily adapted for different applications and environments using a minimal number of housing configurations or features (e.g., tactile buttons or dials). The operational modes can include, but are not limited to, VoIP telephone, video streaming for video calls, intercom, alarm activation for a central public address/general alarm (PA/GA) system, a “serverless” page-party (SP2) system, voice product with FLASH, product with conference function, transit on-board communications device, and so on.

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.

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.

Apparatus and methods concerning simulation of call quality are disclosed. In an example embodiment, computing server is communicatively coupled to a server. The computing server is configured to receive a first set of data which may include audio of a call routed by the server. The computing server also includes a processing circuit configured to characterize a post-transmission quality state of the first set of data. The processing circuit is also configured to generate a second set of data including audio that is different from the audio of the call and data including characteristics indicative of the post-transmission quality state of a first set of 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.

A communication system and method for displaying text captions corresponding to voice communications between an assisted user's mobile wireless device and a separate hearing user's device includes at least one communication component configured to enable the appliance to communicate with a relay, a display, and a processor operably coupled to the at least one communication component and the display. The processor is configured to enable the assisted user to establish an association between the appliance and the mobile device, receive text originating at the relay, the text corresponding to a transcript of the hearing user's voice signal originating at the hearing user's device, and cause text captions corresponding to the received text to be displayed on the display.

A distributed voice applications system includes a voice services instruction rendering agent and at least one voice services software client that is configured to provide voice applications to an individual user. A management system may control and direct the voice services rendering agent to create voice applications that are personalized for individual users based on user characteristics, information about the environment in which the voice applications will be performed, prior user interactions and other information. The voice services software client may be resident on a local user device which includes a voice browser and speech recognition capabilities. The local device, voice services instruction rendering agent and the management system may be interconnected via a communications network.