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Voice Elements is a Microsoft Cloud Service, as well as a Calling Plan for Microsoft Teams. Voice Elements were released by Inventive Labs Corporation in 2008, based on their original CTI32 toolkit. Software developers who use C#, VB.NET, or Delphi use Voice Elements to write telephony-based applications, such as Interactive Voice Response systems, Voice dialers, Auto Attendants, Call centers, and more.
In addition to the Microsoft .NET Framework, Voice Elements supports the use of speech recognition and text-to-speech, [1] Dialogic TDM hardware, and the Inventive Labs HMP Elements SIP Platform. Applications built with Voice Elements are deployed via Voice over IP, via the Inventive Labs cloud hosting service, or by traditional TDM, such as T1, E1, or analog phone lines. [2]
Users of Voice Elements-based solutions interact by using Touch Tone (DTMF) input or with voice commands through speech recognition technology. In addition, developers may program with pre-recorded prompts or use text-to-speech.
Common applications that are built using Voice Elements include:
Typically, industries such as Health Care, Retail and Hospitality, and Financial Services use telephony applications to increase customer contact and automate tasks.
Visual Studio developers, by learning the Voice Elements classes, can create almost any voice application. Call logging and sample inbound and outbound applications are a part of the software package. Consulting services [4] are also available for planning, creating, or training IT staff to develop and manage custom solutions.
If deployed via SIP, Voice Elements developers may use the highly tuned Call Progress Analysis that is included in the Inventive Labs SIP Platform. Call Progress Analysis results inform the software if a person or machine answers a call, and are used in outbound dialing campaigns.
Voice Features
Voice Elements launched their WebRTC interface in August 2013, with their Voice Elements Platform 5. [7] It includes a simple API for creating browser-based, feature-rich WebRTC applications.
This new feature was premiered at the WebRTC Conference & Expo, Atlanta GA, June 25–27, 2013. [8]
WebRTC is a technology that allows you to use your browser simultaneously as a web browser and as softphone.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls for the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
Interactive voice response (IVR) is a technology that allows telephone users to interact with a computer-operated telephone system through the use of voice and DTMF tones input with a keypad. In telephony, IVR allows customers to interact with a company's host system via a telephone keypad or by speech recognition, after which services can be inquired about through the IVR dialogue. IVR systems can respond with pre-recorded or dynamically generated audio to further direct users on how to proceed. IVR systems deployed in the network are sized to handle large call volumes and also used for outbound calling as IVR systems are more intelligent than many predictive dialer systems.
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints such as customary telephone sets, destinations on the public switched telephone network (PSTN) and devices or services on voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
Call Control eXtensible Markup Language (CCXML) is an XML standard designed to provide asynchronous event-based telephony support to VoiceXML. Its current status is a W3C recommendation, adopted May 10, 2011. Whereas VoiceXML is designed to provide a Voice User Interface to a voice browser, CCXML is designed to inform the voice browser how to handle the telephony control of the voice channel. The two XML applications are wholly separate and are not required by each other to be implemented - however, they have been designed with interoperability in mind
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC, voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.
Linphone is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Linphone also provides the possibility to exchange instant messages. It has a simple multilanguage interface based on Qt for GUI and can also be run as a console-mode application on Linux.
A telephony system based on host media processing (HMP) is one that uses a general-purpose computer to process a telephony call’s media stream rather than using digital signal processors (DSPs) to perform the task. When telephony call streams started to be digitized for time-division-multiplexed (TDM) transport, processing of the media stream, to enhance it in some way, became common. For example, digital echo cancellers were added to long-haul circuits, and transport channels were shaped to improve modem performance. Then, in the mid-‘80s, computer-based systems that implemented messaging, for example, used DSPs to compress the audio for storage, and fax servers used DSPs to implement fax modems.
An IP PBX is a system that connects telephone extensions to the public switched telephone network (PSTN) and provides internal communication for a business. An IP PBX is a PBX system with IP connectivity and may provide additional audio, video, or instant messaging communication utilizing the TCP/IP protocol stack.
Voxeo Corporation was a technology company that specialized in providing development platforms for unified customer experience (self-service) and unified communications applications. Voxeo was headquartered in Orlando, Florida with main offices in Cologne, Germany; Beijing, China; London, UK and San Francisco, US.
Web-based VoIP is the integration of voice over IP technologies into the facilities and methodologies of the World-Wide Web. It enables digital communication sessions between Web users or between users of traditional telecommunication services.
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often, a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC or with a USB phone.
Global IP Solutions was a United States-based corporation that developed real-time voice and video processing software for IP networks, before it was acquired by Google in May 2010. The company delivered embedded software that enabled real-time communications capabilities for video and voice over IP (VoIP). GIPS was perhaps best known for developing the narrowband iLBC and wideband iSAC speech codecs.
Aculab is a privately held, UK-based limited company that was founded in 1978. It is a designer, developer and manufacturer that specialises in providing API-driven, enabling technology sub-systems for telecommunications related OEM products such as are used in fixed line PSTN, wireless and VoIP networks. Aculab's products are sold worldwide, primarily through direct sales and also via the reseller channel. Aculab's headquarters and R&D facilities are located in Milton Keynes, UK. It has a branch office in Norwood, Massachusetts, USA.
QuickFuse is a web-based telephony application editor and rapid application development platform. QuickFuse users build call flows by visually assembling modules from a library of building blocks that cover the functional requirements of interactive voice response (IVR), messaging, and telephony applications. QuickFuse uses speech recognition and text-to-speech technology and integrates with other systems through SOAP and REST APIs.
VaxTele SIP Server SDK is a complete development toolkit, which allows software vendors and Internet telephony service providers (ITSP) to develop SIP Server and (SIP) Session Initiation Protocol based VoIP systems for Microsoft Windows to install computer to computer voice chat, chat rooms, IVR systems, call center services, calling card services, dial/receive computer to PSTN and mobile phone calling services.
Voxbone S.A. is a global communication as a service (CaaS) company that is a wholly owned subsidiary of Bandwidth, Inc., with offices in locations including Brussels, London, San Francisco, Austin, Simi Valley, Dublin, Singapore and Iași. Voxbone became a part of Bandwidth on November 2, 2020.