FONE API’s international & flexibly scalable platform, low latency media relay, and smart call control allows you to pay for only what you use without the need to establish server-side elements. Major enterprises can experience the security & efficiency that they need from FONE API’s WebRTC client.
Easily embed communications into web-based enterprise tools like the CRM & create an exceptional user experience. Whether call centers or Hosted PBX, enterprise users are empowered to communicate from within their main web interface, as opposed to connecting to a phone interface. Interaction can take place with the most vital details available and deliver a more optimized customer experience.
The foundation in developing more efficient communications incorporated into the browser.
The international cloud infrastructure of FONE API covers all items necessary to provide service via a browser's WebRTC capabilities, without needing new infrastructure. With Interconnect, you can utilize enterprise-grade network links to gain end-to-end service quality.
Call other browser-based users or those on mobile client endpoints, SIP endpoints, or any PSTN telephone number. WebRTC calls may be controlled via programs, recorded, or even conferenced using simple API calls.
Route & connect calls worldwide through smart global-media routing that boosts service quality, no matter where web users are. FONE API intelligently knows how media is streamed or relayed between callers to enhance call quality and reduce latency and influence of public Internet.
The differences in WebRTC support between browsers can be easily managed by FONEAPI, ensuring you don’t get concerned on Chrome and Firefox-specific API calls.
You can record and store all web client calls in the cloud. Recordings may be transcribed anytime to preserve and make client calls easy to find.
Get all client users in one conference, as well as SIP endpoints and PSTN phone numbers in one line of code.
FoneAPI handles all inbound and outbound call logic, giving you flexible control.
Voice out text to callers in more than 20 languages and accents.
Conveniently create smart on-hold experiences for your clients.
Discover the foundations of the scalable and convenient FONE API.
For Fone API Clients - Starting a phone call allows the browser to create a request to FONE API telling you how to route the call to other browsers, devices, or other carrier networks.
The call logic is based on a number of foneAPIML verbs such as to voice out to the caller and to take dialpad input from the user. The browser connects to FONE API via WebRTC then out to the carrier network.
Businesses are utilizing FONE API Web Client as their agent's primary "phone" to give a comprehensive contextual, efficient, and integrated experience for the agent whose company may be building personalized contact center capabilities integrated with their CRM or those wanting to create a full-featured offering.
WebRTC capabilities can conveniently be added to companies with existing SIP infrastructure via FONE API’s web client which provides the WebRTC capabilities. All calls are routed to the current IP infrastructure over an SIP connection.
Companies developing hosted PBX offerings or virtual receptionists have set up FONE API Web Client into products as the primary phone device for enterprise users, delivering total operating flexibility at an entry with minimal cost.
The kiosk mode helps companies provide real-time interactive communications such as car rentals or sports equipment. Interactive WebRTC activated kiosks deliver a low-cost approach to make your customer interaction become more consistent while maximizing efficiency among your staff.