All communications via shout/whisper are sent over the default data channel and emitted by the LioWebRTC instance as events. You can create your own custom listeners suited for whatever purpose you'd like.
### Data channels only
Disable webcam/mic streaming, and only enable data channel.
```js
constwebrtc=newLioWebRTC({
// The url for your signaling server
url:'https://sandbox.simplewebrtc.com:443/',
dataOnly:true
});
```
### Audio and data channels only
Great for voice chatting.
```js
constwebrtc=newLioWebRTC({
// The url for your signaling server
url:'https://sandbox.simplewebrtc.com:443/',
autoRequestMedia:true,
media:{
video:false,
audio:true
}
});
```
## Example
...
...
@@ -212,17 +232,14 @@ export default Party;
`new LioWebRTC(options)`
-`object options`
-`string url` - url for signaling server.
-`bool debug` - *optional* flag to set the instance to debug mode
-`[string|DomElement] localVideoEl` - ID or Element to contain the local video
element
-`bool autoRequestMedia` - *optional(=true)* option to automatically request
-`string url` - url for your socket.io signaling server.
-`bool debug` - *optional* logs all webrtc events
-`[string|DomElement|Ref] localVideoEl` - Can be a ref, DOM element, or ID of the local video
`connect` event, with the unique id for the session.
`'receivedPeerData', type, payload, peer` - emitted when a peer sends data via `shout` or `whisper`
-`type` a label, usually a string, that describes the payload
-`payload` any kind of data sent by the peer, usually an object
-`peer` the object representing the peer and its peer connection
`'createdPeer', peer` - emitted three times:
- when joining a room with existing peers, once for each peer
- when a new peer joins a joined room
- when sharing screen, once for each peer
`'createdPeer', peer` - this will be emitted when:
- joining a room with existing peers, once for each peer
- a new peer joins a joined room
- sharing screen, once for each peer
-`peer` - the object representing the peer and underlying peer connection
-`peer` - the object representing the peer and its peer connection
`'stunservers', [...args]` - emitted when the signaling connection emits the
same event
`'stunservers', [...args]` - emitted when the signaling server emits this event.
`'turnservers', [...args]` - emitted when the signaling connection emits the
same event
`'turnservers', [...args]` - emitted when the signaling server emits this event.
`'localScreenAdded', el` - emitted after triggering the start of screen sharing
...
...
@@ -309,57 +326,52 @@ ending all peers, and stopping the local screen stream
### Methods
`createRoom(name, callback)` - emits the `create` event on the connection with
`name` and (if provided) invokes `callback` on response
`createRoom(name, callback)` - emits the `create` event and optionally invokes `callback` on response
`joinRoom(name, callback)` - joins the conference in room `name`. Callback is
`joinRoom(name, callback)` - joins the room `name`. Callback is
invoked with `callback(err, roomDescription)` where `roomDescription` is yielded
by the connection on the `join` event. See [signalmaster](https://github.com/andyet/signalmaster) for more details.
`startLocalVideo()` - starts the local media with the `media` options provided
in the config passed to the constructor
by the connection on the `join` event. See [signalmaster](https://github.com/andyet/signalmaster) for details about rooms.
`testReadiness()` - tests that the connection is ready and that (if media is
enabled) streams have started
`startLocalVideo()` - starts the local video or audio streams with the `media` options provided
in the config
`mute()` - mutes the local audio stream for all peers (pauses sending audio)
`mute()` - mutes the local audio stream to your peers (stops sending audio in the WebRTC audio channel)
`unmute()` - unmutes local audio stream for all peers (resumes sending audio)
`unmute()` - unmutes the audio stream to your peers (resumes sending audio in the WebRTC audio channel)
`pauseVideo()` - pauses sending video to peers
`pauseVideo()` - pauses the video stream to your peers (stops sending video in the WebRTC video channel)
`resumeVideo()` - resumes sending video to all peers
`resumeVideo()` - resumes the video stream to your peers (resumes sending video in the WebRTC video channel)
`pause()` - pauses sending audio and video to all peers
`pause()` - pauses both video and audio streams to your peers
`resume()` - resumes sending audio and video to all peers
`resume()` - resumes sending video and audio to your peers
`sendToAll(messageType, payload)` - broadcasts a message to all peers in the
room via the signaling channel (websocket)
room via the signaling server
-`string messageLabel` - The event label that be broadcasted via the signaling server
-`string messageType` - The event label that be broadcasted via the signaling server
-`object payload` - an arbitrary value or object to send to peers
`sendDirectlyToAll(messageType, payload, channel)` - broadcasts a message
to all peers in the room via a dataChannel
to all peers in the room via a data channel
-`string messageType` - the event label that peers will listen for
-`object payload` - an arbitrary value or object to send to peers
-`string channel` - (optional) the label for the dataChannel to send on
-`object payload` - an arbitrary value or object
-`string channel` - (optional) the name of the data channel
`shout(messageType, payload)` - broadcasts a message
to all peers in the room via the default data channel
-`string messageType` - A value that represents the classification of the payload
-`object payload` - an arbitrary value or object to send to peers
`whisper(peer, messageType, payload)` - sends a message to a single peer in the room
`whisper(peer, messageType, payload)` - sends a message to a single peer in the room via the default data channel
-`string messageType` - A value that represents the classification of the payload
-`object payload` - an arbitrary value or object to send to peers
`getPeers(sessionId, type)` - returns all peers by `sessionId` and/or `type`
`shareScreen(callback)` - initiates screen capture request to browser, then
adds the stream to the conference
`shareScreen(callback)` - initiates screen capture request to browser, then streams the video to peers in the room
`getLocalScreen()` - returns the local screen stream
...
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@@ -381,16 +393,8 @@ DOM and perform other setup
`handlePeerStreamRemoved(peer)` - used internally to remove the video container
from the DOM and emit `videoRemoved`
`getDomId(peer)` - used internally to get the DOM id associated with a peer
`getEl(idOrEl)` - helper used internally to get an element where `idOrEl` is
either an element, or an id of an element
`getLocalVideoContainer()` - used internally to get the container that will hold
the local video element
`getDomId(peer)` - get the DOM id associated with a peer's media stream. In JSX, you will need to set the ID of the peer's video element to this value.
`getRemoteVideoContainer()` - used internally to get the container that holds
the remote video elements
...
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@@ -399,14 +403,17 @@ the remote video elements
### Connection
For signaling, LioWebRTC uses [socket.io](http://socket.io/) to
communicate with the signaling server. The connection object comes with these methods:
communicate with the signaling server, and returns a connection object. The connection object comes with the following methods:
-`on(ev, fn)` - A method to invoke `fn` when event `ev` is triggered
-`emit()` - A method to send/emit arbitrary arguments on the connection
-`getSessionId()` - A method to get a unique session Id for the connection
-`disconnect()` - A method to disconnect the connection
-`on(ev, fn)` - a method to set a listener
-`emit()` - send/emit arbitrary events on the connection
-`getSessionId()` - returns the session ID of the connection
-`disconnect()` - disconnect from the signaling server (closes the websocket)
### Signaling Server
LioWebRTC uses the signaling server provided for testing purposes by SimpleWebRTC.
You will need to set up your own [signalmaster](https://github.com/andyet/signalmaster) server, and pass in your server's url when creating a new instance of LioWebRTC.
In production, you will need to set up your own [signalmaster](https://github.com/andyet/signalmaster) server, and pass in your server's url when creating an instance of LioWebRTC. To start your signalmaster server in production mode using PM2, do the following:
