HTML - Web RTC

Web RTC introduced by World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and P2P file sharing.

The web RTC implements three API's as shown below −

• MediaStream − get access to the user's camera and microphone.

• RTCPeerConnection − get access to audio or video calling facility.

• RTCDataChannel − get access to peer-to-peer communication.

MediaStream

The MediaStream represents synchronized streams of media, For an example, Click on HTML5 Video player in HTML5 demo section or else click here.

The above example contains stream.getAudioTracks() and stream.VideoTracks(). If there is no audio tracks, it returns an empty array and it will check video stream,if webcam connected, stream.getVideoTracks() returns an array of one MediaStreamTrack representing the stream from the webcam. A simple example is chat applications, a chat application gets stream from web camera, rear camera, microphone.

Sample code of MediaStream

function gotStream(stream) {
   window.AudioContext = window.AudioContext || window.webkitAudioContext;
   var audioContext = new AudioContext();
   
   // Create an AudioNode from the stream
   var mediaStreamSource = audioContext.createMediaStreamSource(stream);
   // Connect it to destination to hear yourself
   // or any other node for processing!
   mediaStreamSource.connect(audioContext.destination);
}
navigator.getUserMedia({audio:true}, gotStream);

Session Control, Network & Media Information

Web RTC required peer-to-peer communication between browsers. This mechanism required signaling, network information, session control and media information. Web developers can choose different mechanism to communicate between the browsers such as SIP or XMPP or any two way communications.

Sample code of createSignalingChannel()

var signalingChannel = createSignalingChannel();
var pc;
var configuration = ...;
// run start(true) to initiate a call
function start(isCaller) {
   pc = new RTCPeerConnection(configuration);
   // send any ice candidates to the other peer
   pc.onicecandidate = function (evt) {
      signalingChannel.send(JSON.stringify({ "candidate": evt.candidate }));
   };
   // once remote stream arrives, show it in the remote video element
   pc.onaddstream = function (evt) {
      remoteView.src = URL.createObjectURL(evt.stream);
   };
   // get the local stream, show it in the local video element and send it
   navigator.getUserMedia({ "audio": true, "video": true }, function (stream) {
      selfView.src = URL.createObjectURL(stream);
      pc.addStream(stream);
      if (isCaller)
         pc.createOffer(gotDescription);
      else
         pc.createAnswer(pc.remoteDescription, gotDescription);
         function gotDescription(desc) {
            pc.setLocalDescription(desc);
            signalingChannel.send(JSON.stringify({ "sdp": desc }));
         }
      });
   }
   signalingChannel.onmessage = function (evt) {
      if (!pc)
         start(false);
         var signal = JSON.parse(evt.data);
      if (signal.sdp)
         pc.setRemoteDescription(new RTCSessionDescription(signal.sdp));
      else
         pc.addIceCandidate(new RTCIceCandidate(signal.candidate));
};