This is documentation for Web Call Server 3.x

Take a look at our newest product Web Call Server 5

Web Call Server 5 product page

WebRTC

WebRTC support

Flashphoner Web Call Server supports HTML5 Websockets protocol and WebRTC since version 3.0

WebRTC support will work in browsers where WebRTC is installed as extension or natively supported.

Features list

- Chrome 26 or latest is supported
- Local browser-to browser calls (When both clients in the same LAN).
- Browser-Browser calls via external VoIP media server.
- G.711 audio codec and VP8 video codec
- DTMF SIP INFO
- Instant messaging SIP MESSAGE

Testing in the Chrome browser

If you test the application locally (not opening web page in browser from a web server).

You need to start Chrome as described below to garnt access to reading of the local config flashphoner.xml.

Close all Chrome tabs then start Chrome using command below:

chrome.exe --allow-file-access-from-files --enable-file-cookies

How to enable Websocket and WebRTC in the Flashphoner client

To enable WebRTC, you should edit flashphoner.xml config and set

<rtmp_server>ws://x.x.x.x:8080</rtmp_server>

, where x.x.x.x is IP address of Flashphoner Web Call Server

This settings mean that Flashphoner-client will use HTML5 Websocket connection to the server, use WebRTC for audio and video streaming and do not use Flash Player at all.

If you need to enable Flash Player support again, just change this back to 

<rtmp_server>rtmfp://x.x.x.x:1935</rtmp_server>

You can dinamically switch between tese settings depends on user browser.

For example, if user's browser is the latest Chrome browser with WebRTC support you can use ws:// URL to enable WebRTC and fallback to rtmfp:// for other browsers.

Quick start

1. Install Flashphoner Web Call Server 3.0.247

2. Install WCS-client 297

3. Change client flashphoner.xml file to enable HTML5 Websocket/WebRTC support.

<rtmp_server>ws://x.x.x.x:8080</rtmp_server>

4. Make sure the browser did not cache flashphoner.xml

5. Open page PhoneJS.html and try to make call via WebRTC.

Testing environment:

OpenSIPs, CommunigatePro

Chrome 26 Windows 7

Flashphoner Web Call Server 3.0 preview

Current version supports only Browser-Browser calls with latest Chrome browser via SIP proxy server. Asterisk is not a SIP proxy and it does not work with Asterisk for now.
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