Special Focus

TMC Names WebRTC Product of the Year Award Winners

By TMCnet News  |  May 06, 2015

INTERNET TELEPHONY magazine and its parent company, TMC (News - Alert), recognize some of the top movers in shakers in real-time communications with the 2015 WebRTC Product of the Year Awards.

“Congratulations to the winners of the 2015 WebRTC Product of the Year Award," said Erik Linask, group editorial director at TMC. “The recipients have displayed groundbreaking innovation in the WebRTC space and we look forward to seeing their future successes.”

THE WINNERS

3CX

3CX Phone (News - Alert) System 12.5

The 3CX Phone System 12.5 leverages WebRTC technology to offer click-to-call and video calling. With click-to-call, website visitors can make a free voice or video call directly to a business through the Internet browser, without the need to download any additional software. When a visitor clicks on the embedded call button, a VoIP call is initiated via 3CX Phone System directly to the business. With 3CX’s click-to-call function, businesses can also elevate a call to offer screen sharing so customers can be

shown new products or offered assistance, thereby providing a personalized customer service experience. Businesses using 3CX Phone System 12.5 can publish any number of call button links, enabling customers to be directed to the most relevant person. As inbound calls via the click-to-call function are free, businesses can reduce the cost of expensive 800 number calls.

AGNITY Inc.

ACONYX Unify

ACONYX Unify is a turnkey cloud-based mobile UC and collaboration solution that enables service providers to empower enterprise mobility. It supports voice, video, conferencing, presence and messaging capabilities. It is built on the foundation of ACONYX Communications Application Server and FlexYcharge, and offers a scalable, carrier-grade architecture that can be deployed across heterogeneous networks. And it is

optimized for service delivery over LTE and smartphone/tablet devices.

Amaryllo International Inc.

iCamPRO

The iCamPRO is a robotic home security camera that can actually see, hear, sense, and track moving objects, all while communicating with its owner. The 2 Megapixel lens and image sensor deliver extremely accurate pixel calculation and therefore a more accurate motion analysis for tracking moving objects. It stands at about 3 inches tall and is powered by a high-speed CPU with a patent-pending multi-sensor network. The multiple-sensor network embedded in the product can steer the camera in all directions. That means that no matter where the intruders appear in your home, iCamPRO can immediately rotate itself to capture the intruders in seconds. Also included in the box is an adjustable mount for use on a flat surface, wall, or ceiling.  

APEX Communications

APEX Video Conferencing (WebRTC)

The APEX Video Conferencing System is based on the APEX Service Delivery Platform with support for CIF, VGA, and 720p with H.264. This complete wideband/HD audio and HD videoconferencing platform seamlessly integrates with existing mobile/3G/4G (LTE) and IP/IMS converged networks, and is device agnostic. It incorporates the power of WebRTC, making videoconferencing available to anyone using a website. And it can be configured either as a single APEX SDP with an integrated media server, or an APEX SDP providing conferencing services through multiple media servers. With the OmniVox3D Application Server, servers are accessed across local or distributed networks, using SIP and Markup Language technology.

The display/layout options supported by the APEX Video Conferencing System include Voice Activated Switching and Continuous Presence (Tiled). With Voice Activated Switching, callers see the real-time video stream of the active speaker; Continuous Presence shows multiple sites continuously and supports multiple real-time video streams within a multi-pane display.

Altitude Digital Media – Brazil

WebMeeting Visit

This solution shortens the distance between you and your audience via live webcast, audio, video, slides and audience interactivity. Compatible with PCs, MacsiPads and Android devices, the WebMeeting solution includes all the features for disclosure and access control, and offers the ability to obtain accurate, real-time profiles of the audience. It is available live and on demand, offers support for Q&A sessions, polls, and surveys. And it enables the simultaneous translation of live communications and the interface in Portuguese, English, and Spanish.

BroadSoft (News - Alert)

BroadWorks WebRTC Server

The BroadWorks WebRTC Server is a core enabling component of BroadSoft’s WebRTC solution, which fully integrates a seamless communication experience, putting the power of unified communications in the hands of anyone with a WebRTC enabled web-browser. A key component of the BroadSoft WebRTC solution is access to BroadSoft Labs, a suite of tools allowing service providers and developers to create and customize WebRTC-enabled communication and collaboration applications in as little as 5 minutes. BroadSoft’s WebRTC-enabling tools simplify the process for service providers seeking to design and customize their own unified communications offers, which can increase service provider revenue opportunities, enhance the customer experience, and future proof investments as the technology evolves.

Comverse

Comverse WebRTC Gateway

Comverse WebRTC-GW enables real time audio/video/messaging communication directly from a web browser to any IMS/RCS/UC/VoIP network. It extends these communication services to the web domain, making them more available and reachable. Comverse WebRTC-GW takes a novel approach to web browser support, reducing the integration complexity and increasing solution security. It eliminates the need for telco protocols in the browser and isolates the web world from the telco world. The WebRTC-GW can be deployed in-network or in a cloud, and supports a wide range of enterprise and service provider use cases. For added security it isolates web credentials from telco credentials. It combines RCS and WebRTC functionalities, enabling innovative IP-based services. And it offers integrated billing for WebRTC services – both prepaid and postpaid.

Dialogic Inc.

PowerMedia XMS

The Dialogic PowerMedia XMS can do advanced mixing, recording, transcoding, and interworking capabilities in a high-load conferencing environment. The cloud- and NFV-ready, virtualized media processing software offers the functionality of a production-ready media server, MRF, and MCU interfaces for web-oriented and traditional media applications. It provides comprehensive audio, video, and contact center features. And it allows for reduced telecom application development time because it leverages proven MRF element and platform, and delivers an improved user experience because it offloads difficult media handling functions to network. PowerMedia products deliver on the IMS framework and MRF that Dialogic has been talking about for years and have finally started to make a huge impact.

Digium Inc.

Respoke

Respoke is a cloud platform that provides everything that’s needed to add communications to an app. It makes it easy to add secure, high-definition voice and video features to an app using just a few lines of JavaScript. It offers the ability to easily add individual and group chat messaging, voice and/or video calling, screen sharing, and file sharing to web applications, all without the need to install a cumbersome plug-in. Developers interface with Respoke using an intuitive JavaScript library, RESTful APIs, and mobile SDKs. A robust, scalable cloud infrastructure handles all of the complex back-end aspects of making real-time communications work. Respoke began as a virtual startup within Digium and is staffed with a mix of communications industry veterans and cloud computing experts.

FRAFOS

ABC WebRTC Gateway

The ABC WebRTC gateway is the missing piece that connects web clients to the SIP telephony infrastructure in a transparent manner. The gateway anchors signaling and media, and performs translation between different standards for WebRTC and SIP, particularly security, codecs, and signaling protocols. This software-based solution can be deployed as part of the ABC SBC or as a stand-alone solution. It also features GUI-based management, system and network monitoring, border security, SNMP V2 alarms and status information, and high availability using active/hot standby mode. And it provides the following WebRTC standards support: routing audio codec including G.711 and OPUS; transcoding of audio codec including OPUS to G.711; routing of video codec including VP8; media security using SRTP for secure real-time media transmission; exchange of security keys using DTLS and SDES; signaling security using TLS; SIP over WebSocket; and NAT traversal using STUN, ICE, and TURN.

GENBAND (News - Alert)

GENBAND SPiDR WebRTC Gateway

GENBAND's SPiDR WebRTC Gateway provides an intelligent bridge between traditional voice over Internet protocol networks and the open ecosystem of the Internet. SPiDR empowers network operators to deliver competitive applications over the Internet and unlocks new revenue potential from their existing wireless and fixed communication assets. It sits at the edge of the network and provides open, web-centric APIs that allow application developers to leverage the rich communications services of the telecommunications network – including voice, video, presence, shared address book, call history, instant messaging, and collaboration. SPiDR seamlessly and intelligently interworks both the signaling and media planes between the web and telecom worlds and is offered as part of the GENBAND solutions for service providers and enterprises.

SPiDR introduces WebRTC capabilities into GENBAND’s cloud-ready EXPERiUS solution, and offers GENCom for Web, a browser-based WebRTC soft-client that simplifies the deployment of communications services.

Hewlett-Packard Co.

HP WebRTC Gateway Controller

HP says it took an evolutionary approach to specific CSP requirements in introducing the HP Multimedia Services Environment WebRTC Gateway Controller. Its architecture facilitates integration between legacy services and WebRTC endpoints, generating a new set of use cases to enhance existing services, while offering telco functionalities to new digital WebRTC services. Examples include click-to-call, multidevice communication services, screen sharing, video sharing, call recording, audio and videoconferencing, voice mail access, and unified communication. For example, it’s possible to provide users with a single communication experience across the phone and web.

HireVue

HireVue Live

HireVue lets people tell their story and demonstrate their talents, enabling high-touch collaboration and insights – all at the speed, quality and consistency of digital. HireVue was recognized by Forbes as a top 10 company on America’s Most Promising Companies list, a top 50 company in Deloitte’s (News - Alert) Annual Fast 500, and as a top 500 fastest-growing private company by Inc. magazine. There are 500 HireVue customers including Ocean Spray Cranberries, Hilton Worldwide, UnitedHealth Group, Chipotle Mexican Grill and Discovery Communications. 

OnSIP

InstaPhone

InstaPhone, one of OnSIP’s WebRTC-based products, is a business-grade phone that presents workers with all of the features of a deskphone right in their browser. The app is capable of voice calling to and from any SIP phone, software phone, webpage, or the public switched telephone network. It also offers HD video calling and displays real-time caller data (such as what product the caller is viewing) when a call is initiated from a webpage. It offers an easy way for remote workers to use the OnSIP phone system. InstaPhone is accessible within a webpage and as an application in the Salesforce AppExchange. InstaPhone for Salesforce lives right in the Salesforce dashboard and adds to the standard features by displaying linked leads, opportunities, and accounts for each caller. Leveraging this integration and InstaPhone's custom data display, sales and support agents can engage in informed discussions to better close deals and solve problems.

Oracle Corp.

Oracle Communications WebRTC Session Controller

WebRTC allows users to seamlessly communicate in high definition video and/or voice with shared screen capabilities. While WebRTC promises a heightened communications experience and creates new opportunities for both CSPs and enterprises, there are several network challenges to overcome. Oracle addresses these challenges with a purpose-built WebRTC media/signaling engine and a client-side software development kit solution. The Oracle Communications WebRTC Session Controller bridges the web to the SIP/IMS network with secure client-network management, highly reliable fault resilient web to SIP session processing, and full WebRTC device to SIP network interoperability. It allows for application control and synchronization during network changes and browser page reloads, rapid application integration with existing systems, identity management between multiple devices and across web and telephony domains, border and application security to prevent attacks and service abuse, high capacity media handling for NAT traversal, encryption, and transcoding, and robust and dynamic interworking with existing infrastructure.

Sansay

WebSBC Click to Call

Sansay's WebSBC architecture delivers a carrier-class media engine and highly scalable WebRTC-SIP interconnect for revenue-generating WebRTC applications within service provider networks. WebSBC also eliminates the complexity and expense of dedicated WebRTC-SIP gateways and heavyweight browser clients with embedded SIP stacks. Instead, WebSBC encapsulates both functions into the application layer via the RAPID development environment. WebSBC is an evolution of the VSXi SBC, on which more than 350 service providers worldwide currently rely to maximize the performance, profitability and growth of their VoIP networks and services. WebSBC is available in both service provider hosted and platform-as-a service models.

Speekio A/S

Speekio PRO & Speekio HR

Speekio PRO enables users set up and manage teams of employees that need access to videoconferencing with internal or external users. It runs from a Speekio webpage or can be embedded directly into an organization’s website. Speekio HR optimizes the hiring process for recruiting companies and HR departments in large organizations. Speekio HR saves organizations valuable time by letting candidates fill in a web questionnaire and web interview before they get selected for the next step in the recruitment process.

Switch Communications

Switch.co

Switch.co sends calls and texts to all of a user’s devices – mobile phone, desktop, and even desk phone – at the same time. That way users can answer the device that’s most convenient at any time. The Switch.co mobile apps for iPhone and Android provide a business number for a personal device. That way the user can access his or her company directory, place calls, and send texts on the go. Meetings move with users from the office, to the car, and to the airport without any dropped connection.

Temasys Communications Pte Ltd.

Skylink

The Skylink Platform is a suite of applications and services that power real-time communications in websites and applications. It can use WebRTC to extend or enhance existing services or applications, and is ready to support online interaction and collaboration in completely new and different ways. This telco-grade, enterprise-ready infrastructure can scale up to hundreds of millions of users. Temasys has built its offering on top of Amazon Web Services, with best-of-breed components, optimized for cloud-based infrastructure, allowing Temasys to maximize availability and scalability while being extremely cost-effective.

TokBox

OpenTok

TokBox was founded in 2007 as a consumer video chat service. In November 2010 it launched OpenTok, a cloud platform for adding live video, voice and messaging to websites and mobile applications and in 2012 became the first company to launch support for WebRTC. The company’s scalable, customizable platform gives users the creative freedom to develop any real-time communication interaction, from one-to-one video chats to large-scale broadcasts. Having pioneered real-time communications, even before WebRTC, OpenTok has several years experience helping companies bring voice and video to real-world applications. The platform has been used by organizations across a wide range of industries, including companies like Mozilla, with its Firefox Hello!, Major League Baseball, Fluke, Esurance, Minerva Project, Bridgestone and Double Robotics.

UNIFY

Circuit

Circuit is a SaaS-based workspace solution that delivers voice, video, screen sharing, messaging, and file sharing in a single view. Circuit, the brand name of the effort previously referred to as Project Ansible, is a a WebRTC-based clientless platform that delivers the above-noted capabilities in an integrated way to drive productivity. As a result, people can focus on their work instead of focusing on the mode of communication to get the work done. This solution was created in collaboration with Frog Design, a German company that has produced award-winning designs for Apple, and worked with other top-shelf companies such as Disney and Sony.

Vocalcom

Vocalcom WebRTC Solution

The Vocalcom says its WebRTC solution is the first and only cloud contact center with zero on-premises hardware, software, and telephony infrastructure. This development allows contact center agents to use the Vocalcom cloud contact center solution agent desktop on Chromebooks through WebRTC-enabled interfaces. Within a single window, the solution puts the customer record front and center, with the most critical and recent information in clear view. The agent may then search easily for more details, as all of the customer’s interaction history – across all channels – is there in one place. Customers may then initiate contact with a company representative or customer service agent directly from the website without a need for installing an additional third-party application.

Voice4Net

Voice4Net RTC Client

RTC Client is a WebRTC-based contact center solution that allows for ease of customization and implementation. It provides a widget-based GUI and has drag-and-drop elements that can be used to populate the agent desktop as needed. And it can be integrated easily into legacy contact centers by dealers and integrators. Voice4Net is leveraging WebRTC in its solutions because with WebRTC, end customers can deploy state-of-the art features that provide a user experience that is purpose-built for each business, without requiring a full overhaul of the customer’s legacy solution. A browser-based framework requires far less configuration on the part of the developer, making it less expensive to integrate, while realizing a quicker time to market and more attainable ROI. A browser-based HTML5 solution can be deployed right over an existing infrastructure, preserving the end user’s legacy investment.

Vonage Holdings Corp.

Vonage Extensions/Mobile Inbound Calling

The Vonage Extensions App allows users to make and receive calls on their smartphones, plus get free Wi-Fi calling. They can answer an incoming call to their Vonage number as if they were at home. All calls to their Vonage number ring their home phone and the app. Users will know an incoming call is to their Vonage number or to their mobile number. Calls they make from the app will use their home Vonage number as their caller ID. The app uses the subscriber’s phone contact book to provide a caller ID, so he or she knows who is calling.

Web-call

Web-call Click to Call

Web-call Click to call service allows website visitors to call an organization free from any part of the world. This technology can be installed to a company's website with very easy and simple steps. Visitors of the website don't need to download any software or phone to make calls to the company. 




Edited by Stefania Viscusi
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