Flashphoner Web Call Server Enhancement through the WebRTC-Technology

The largest developers in the sphere of web-technologies (among which are Mozila, Google, Cisco and others) have been recently actively engaged in the process of in-browser audio and video communications improvement. The WebRTC-technology has been found to be an optimal solution for these purposes. What features make WebRTC that particular? The answers are provided in the article below.

It is the right time for WebRTC

After Google acquired the Global IP Solution in winter of 2010, the interest of web developers to online web-based communication technologies grew manifold. They got focused on the idea to create a new technology enabling high-quality audio and video media direct from-a-browser transmission in an easy but effective way, without the need to use any additional software.

Thus WebRTC possesses certain advantages in comparison with other technologies of that type, since it doesn’t require the use of Adobe Flash Player (RTMP) or other plugins for the development of on-website communication services such as “cliсk-to-call”, “web-phone”, videochats, etc. An open-source code and Javascript/HTML5 API make the technology available to developers today.

WebRTC Advantages

WebRTC enables developers to create business-applications designed for media streaming (including for mobile devices) of a high quality, ensuring reliable browser-to-browser (peer-to-peer) communications. Though it is necessary to admit that WebRTC has been available in beta-version browsers only (in Chrome for Android, in particular) up to now, and won’t function on IOS Safari without additional applications installed at all. It all made developers to work out new solutions to improve WebRTC functionality and reliability, which took them about six months. The innovations were introduced in Firefox and Google Chrome, besides a new WebRTC-based Chrome Beta for Android-based platforms was released by the Google company in March of 2013, while opening new horizons for web-communications not available earlier even with the use of the Flash.

Flashphoner Web Call Server Enhancement through WebRTC

The new technology has inspired Flashphoner developers to create an updated Web Call Server 3.0 software platform version based on WebRTC which possesses a significant advantage as compared to its previous versions. While being an integral part of a browser itself, this software doesn’t require any additional plugins to be installed any longer (like it used to be in a conventional Adobe Flash Player browser technology earlier for example). Though the technology is used only within last Google Chrome versions today encompassing about 35% of all browsers installed, Flashphoner experts believe that WebRTC still has all chances for expansion due to its competitive edges in comparison with analogous products.

The Flashphoner team has developed its own Web Call Server software version compatible with both RTMFP (Adobe Flash Player) and WebRTC, which can be integrated into any website ensuring that a browser installed on a user’s computer shall allow to make audio- or video-calls from a website. This new solution from the Flashphoner company is designed for SIP и VoIP-communications directly from a browser, ensuring high-quality multimedia data streaming through SAVPF (SRTP + RTCP), also with the help of G.711 and VP8 codecs.

Being based on Flash and Javascript, this product has an open-source code from client’s part. It allows for any changes to be introduced if necessary, as well as enables to adapt the solution to corporate needs and integrate it with Asterisk server, CRM or ERP systems. One should use WebRTC through Asterisk server in prior Web Call Server 3.0 software versions, since Asterisk server can be SAVPF-supported, provided that proper configurations have been introduced. The software is planned to be adjusted for integration possibility with any VoIP-servers SIP proxy and PBX that support a standard RTP/AVP profile for media traffic, in the short run.

In case it is impossible to integrate WebRTC with Asterisk for some reason, a previous 2.1 Flash-based software version compatible with different PBX systems and SIP devices is recommended for use by developers.

The software installation is to be performed in three steps.

    1. Download the latest program version from the website
    1. Perform a server platform element installation on a separate Linux-server and configure a SIP-account therein (http://docs.flashphoner.com/display/WCS/Quick+installation)
    1. Configure SAVPF support on the corporate Asterisk server (http://docs.flashphoner.com/display/WCS/WebCallServer+WebRTC+with+Asterisk).
      As soon as the system is configured, the use of basic program widgets (“Web-Phone”, “Click to Call” and others) with installation of a preferred version into your website shall be enabled.

Technology Testing

A testing web-softphone of WCS 3.0 platform (http://demo.flashphoner.com/WCS-3.0/297/PhoneJS.html) is recommended for testing of WebRTC-based audio and video media streams transmission and establishing of calls directly from a browser (Google Chrome). Testing may be performed on an analogous browser softphone that may be installed on any website, and it is important that SIP-account configurations of a VoIP provider coincide with the ones described on image:http://flashphoner.com/img/articles/webrtc_technology_in_action_with_web_call_server/sip_video_phone.png

Microphone and camera must be connected properly and allowed to be used by selecting the “Accept” button before making test-calls, as it is shown on image:


It will be possible to make browser-to-browser calls within Google Chrome due to the use of WebRTC technology by Web Call Server. Adobe Flash Player-based RTMFP shall be used for other types of calls at that.

The Flashphoner website provides a detailed description of Web Call Server features and principles of its use, together with a possibility to download and install a free trial of the product.