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WebRTC Signalling Server
Attention: We have migrated our documentation to our new platform, Ant Media Resources. Please follow this link for the latest and up-to-date documentation.
NOTE: We have updated our documentation. This page is outdated. You can access updated version from the sidebar menu.
Ant Media Server can also be used as a signalling server for peer to peer connections. WebSocket is used for connection between peers and Ant Media Server.
In order to use Ant Media Server as a WebSocket you just need to use an app that provides this feature. If you do not know, how to do that drop an email to contact at antmedia dot io
Of course there is a JavaScript SDK in order to make using signalling server straight forward. There is a sample peer.html file in the sample app, you can also try it to understand how to use JavaScript SDK
JavaScript SDK is so easy, just create WebRTCAdaptor
object and call join(roomName)
function.
Let's see how to make it step by step
<head>
...
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
<script src="js/webrtc_adaptor.js" ></script>
...
</head>
<video id="localVideo" autoplay muted width="480"></video>
<video id="remoteVideo" autoplay controls width="480"></video>
First video tag is for local video and the second video tag for remote video.
<script>
var pc_config = null;
var sdpConstraints =
{
OfferToReceiveAudio : true,
OfferToReceiveVideo : true
};
var mediaConstraints = {
video: true,
audio: true
};
var webRTCAdaptor = new WebRTCAdaptor({
websocket_url:"ws://" + location.hostname + ":8081/WebRTCApp4", // url of the WebSocket Signalling Server
mediaConstraints: mediaConstraints,
peerconnection_config: pc_config,
sdp_constraints: sdpConstraints,
localVideoId: "localVideo", // id of the local video tag
remoteVideoId: "remoteVideo", // id of the remote video tag
callback: function(info) { // *success callback function*
if (info == "initialized")
{
// it is called with this parameter when it connects to
// signalling server and everything is ok
console.log("initialized");
}
else if (info == "joined")
{
// it is called with this parameter when it joins a room
console.log("joined");
}
else if (info == "leaved")
{
// it is called with this parameter when it leaves from room
console.log("leaved");
}
},
callbackError: function(error) {
// error callback function it is called when an error occurs
//some of the possible errors, NotFoundError, SecurityError,PermissionDeniedError
console.log("error callback: " + error);
alert(error);
}
});
</script>
In order to create a connection between peers, each peer should join the same room by calling join(roomName)
function of
the WebRTCAdaptor. When there are two peers in the same room, signalling starts automatically and peer to peer connection
is established.
For instance, you can call join function in success callback function when the info parameter is having value "initialized"
if (info == "initialized")
{
// it is called with this parameter when it connects to
// signalling server and everything is ok
console.log("initialized");
webRTCAdaptor.join("room1");
}
According to code above, when peer.html file is opened by two peers, they will joined the "room1" and peer to peer connection will be established.
As shown above, main object is WebRTCAdaptor so that let's look at its functions
-
join(roomName)
:Lets peer join to a room specified in the parameter, if operation is successfull then callback function is called with info parameter having "joined" value. When there are two people in the same room, signalling starts automatically and peer to peer connection is established
-
leave()
:Lets peer leave the room it joined previously. If operation is successfull then callback function is called with info parameter having "leaved" value
-
turnOnLocalCamera()
:Lets the local camera turn on and add the video stream to peer connection
-
turnOffLocalCamera()
:Lets the local camera turn off and remove the video stream from peer connection
-
unmuteLocalMic()
:Lets the local mic unmute and add the audio stream to peer connection
-
muteLocalMic()
:Lets the local mic mute and remove the audio stream from peer connection
Please take a look at the WebRTCApp4/peer.html file in order to see How JavaScript SDK can be used
- Introduction
- Quick Start
- Installation
- Publishing Live Streams
- Playing Live Streams
- Conference Call
- Peer to Peer Call
- Adaptive Bitrate(Multi-Bitrate) Streaming
- Data Channel
- Video on Demand Streaming
- Simulcasting to Social Media Channels
- Clustering & Scaling
- Monitor Ant Media Servers with Apache Kafka and Grafana
- WebRTC SDKs
- Security
- Integration with your Project
- Advanced
- WebRTC Load Testing
- TURN Servers
- AWS Wavelength Deployment
- Multi-Tenancy Support
- Monitor Ant Media Server with Datadog
- Clustering in Alibaba
- Playlist
- Kubernetes
- Time based One Time Password
- Kubernetes Autoscaling
- Kubernetes Ingress
- How to Install Ant Media Server on EKS
- Release Tests
- Spaceport Volumetric Video
- WebRTC Viewers Info
- Webhook Authentication for Publishing Streams
- Recording Streams
- How to Update Ant Media Server with Cloudformation
- How to Install Ant Media Server on GKE
- Ant Media Server on Docker Swarm
- Developer Quick Start
- Recording HLS, MP4 and how to recover
- Re-streaming update
- Git Branching
- UML Diagrams