WebRTC: Video telephony without a browser plugin
Data Channel
WebRTC can do more than just transfer media streams. The data()
function in Listing 14 creates a data channel for transmitting text messages that works like a WebSocket. Line 3 uses the createDataChannel()
method to open a channel for the current peer-to-peer connection in the peer object, pc
. The ondatachannel
event in line 5 occurs when a data channel is opened. The config()
function in line 11 defines the callback function that is triggered when a text message arrives, and the send()
method in line 14 sends messages over the existing connection.
Listing 14
Data Channel
Chrome Can Do
The sample application for Firefox can be ported for the Google Chrome browser with little effort. To port the application to Chrome, just change the manufacturer prefix moz
to webkit
in the examples. The expressions you need to change could just as easily be outsourced into functions that call the appropriate function, including a manufacturer prefix for each browser. Note that the examples only work in Chrome when called via the HTTP protocol, such as http://localhost:6655/mediastream.html.
WebRTC can also connect multiple browsers. The topology of peer-to-peer connections is endless, but a computer quickly reaches its limits when asked to play multiple media streams. WebRTC brings native real-time communication to the browser. Two browsers are needed to build a peer-to-peer connection across the Internet. WebRTC also works around NAT or firewalls. With its data channel, WebRTC not only ensures a bidirectional exchange of application data but also transfers media streams.
The direct connection between browsers saves time and data traffic. This opens up opportunities for many new HTML5 applications to web developers – for example, video chats or distributed applications, such as a networked memory game.
Infos
- WebRTC: http://www.webrtc.org
- UC-RTC: http://html5labs.interoperabilitybridges.com/cu-rtc-web/cu-rtc-web.htm
- Rtcweb status pages: http://tools.ietf.org/wg/rtcweb/draft-ietf-rtcweb-overview/
- WebRTC 1.0 W3C editor's draft: http://dev.w3.org/2011/webrtc/editor/webrtc.html
- WebRTC MWC phone demo: http://www.youtube.com/watch?v=rWPZZeXK6g4
- ICE: http://en.wikipedia.org/wiki/Interactive_Connectivity_Establishment
- STUN: http://en.wikipedia.org/wiki/STUN
- TURN: http://en.wikipedia.org/wiki/Traversal_Using_Relays_around_NAT
- getUserMedia(): http://dev.w3.org/2011/webrtc/editor/getusermedia.html
- Listings for this article: ftp://ftp.linux-magazin.com/pub/listings/magazine/154
- SDP format: http://tools.ietf.org/html/rfc2327
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