diff --git a/README.md b/README.md index f887dbd..7667ad5 100644 --- a/README.md +++ b/README.md @@ -1,235 +1,235 @@ # sylkrtc.js JavaScript library implementing the API for communicating with [SylkServer's](http://sylkserver.com) WebRTC gateway application. ## Building Grab the source code using Darcs or Git and install the dependencies: cd sylkrtc ./configure Build the development release (not minified): make Build a minified version: make min ## Development Auto-building the library as changes are made: make watch ### Debugging sylkrtc uses the [debug](https://github.com/visionmedia/debug) library for easy debugging. By default debugging is disabled. In order to enable sylkrtc debug type the following in the browser JavaScript console: sylkrtc.debug.enable('sylkrtc*'); Then refresh the page. ## API The entrypoint to the library is the `sylkrtc` object. Several objects (`Connection`, `Account` and `Call`) inherit from Node's `EventEmitter` class, you may want to check [its documentation](https://nodejs.org/api/events.html). ### sylkrtc The main entrypoint to the library. It exposes the main function to connect to SylkServer and some utility functions for general use. #### sylkrtc.createConnection(options={}) Creates a `sylkrtc` connection towards a SylkServer instance. The only supported option (at the moment) is "server", which should point to the WebSocket endpoint of the WebRTC gateway application. Example: `wss://1.2.3.4:8088/webrtcgateway/ws`. It returns a `Connection` object. Example: let connection = sylkrtc.createConnection({server: 'wss://1.2.3.4:8088/webrtcgateway/ws'}); #### sylkrtc.debug [debug](https://github.com/visionmedia/debug) module, exposed. Used for debugging, with the 'sylkrtc' prefix. #### sylkrtc.rtcninja [rtcninja](https://github.com/eface2face/rtcninja.js) module, exposed. Used for accessing WebRTC APIs and dealing with platform differences. #### sylkrtc.closeMediaStream(stream) Helper function to close the given `stream`. When a local media stream is closed the camera is stopped in case it was active, for example. Note: when a `Call` is terminated all streams will be automatically closed. ### Connection Object representing the interaction with SylkServer. Multiple connections can be created with `sylkrtc.createConnection`, but typically only one is needed. Reconnecting in case the connection is interrupted is taken care of automatically. Events emitted: * **stateChanged**: indicates the WebSocket connection state has changed. Two arguments are provided: `oldState` and `newState`, the old connection state and the new connection state, respectively. Possible state values are: null, connecting, connected, ready, disconnected and closed. If the connection is involuntarily interrupted the state will transition to disconnected and the connection will be retried. Once the closed state is set, as a result of the user calling Connection.close(), the connection can no longer be used or reconnected. #### Connection.addAccount(options={}, cb=null) Configures an `Account` to be used through `sylkrtc`. 2 options are required: *account* (the account ID) and *password*. The account won't be registered, it will just be created. The *password* won't be stored or transmitted as given, the HA1 hash (as used in [Digest access authentication](https://en.wikipedia.org/wiki/Digest_access_authentication)) is created and used instead. The `cb` argument is a callback which will be called with an error and the account object itself. Example: connection.addAccount({account: saghul@sip2sip.info, password: 1234}, function(error, account) { if (error) { console.log('Error adding account!' + account); } else { console.log('Account added!'); } }); #### Connection.removeAccount(account, cb=null) Removes the given account. The callback will be called once the operation completes (it cannot fail). Example: connection.removeAccount(account, function() { console('Account removed!'); }); #### Connection.close() Close the connection with SylkServer. All accounts will be unbound. #### Connection.state Getter property returning the current connection state. ### Account Object representing a SIP account which will be used for making / receiving calls. Events emitted: * **registrationStateChanged**: indicates the SIP registration state has changed. Three arguments are provided: `oldState`, `newState` and `data`. `oldState` and `newState` represent the old registration state and the new registration state, respectively, and `data` is a generic per-state data object. Possible states: * null: registration hasn't started or it has ended * registering: registration is in progress * registered * failed: registration failed, the `data` object will contain a 'reason' property. * **outgoingCall**: emitted when an outgoing call is made. A single argument is provided: the `Call` object. * **incomingCall**: emitted when an incoming call is received. Two arguments are provided: the `Call` object and a `mediaTypes` object, which has 2 boolean properties: `audio` and `video`, indicating if those media types were present in the initial SDP. * **missedCall**: emitted when an incoming call is missed. A `data` object is provided, which contains an `originator` attribute. #### Account.register() Start the SIP registration process for the account. Progress will be reported via the *registrationStateChanged* event. Note: it's not necessary to be registered to make an outgoing call. #### Account.unregister() Unregister the account. Progress will be reported via the *registrationStateChanged* event. #### Account.call(uri, options={}) Start an outgoing call. Supported options: * pcConfig: configuration options for `RTCPeerConnection`. [Reference](http://w3c.github.io/webrtc-pc/#configuration). * mediaConstraints: constraints to be used when getting the local user media. [Reference](http://www.w3.org/TR/mediacapture-streams/#mediastreamconstraints). * offerOptions: `RTCOfferOptions`. [Reference](http://w3c.github.io/webrtc-pc/#idl-def-RTCOfferOptions). * localStream: if specified, it will be used by sylkrtc instead of using `getUserMedia`. Example: let call = account.call('3333@sip2sip.info', {mediaConstraints: {audio: true, video: false}}); #### Account.id Getter property returning the account ID. #### Account.password Getter property returning the HA1 password for the account. #### Account.registrationState getter property returning the current registration state. ### Call Object representing a audio/video call. Signalling is done using SIP underneath. Events emitted: * **localStreamAdded**: emitted when the local stream is added to the call. A single argument is provided: the stream itself. * **streamAdded**: emitted when a remote stream is added to the call. A single argument is provided: the stream itself. * **stateChanged**: indicates the call state has changed. Three arguments are provided: `oldState`, `newState` and `data`. `oldState` and `newState` indicate the previous and current state respectively, and `data` is a generic per-state data object. Possible states: * terminated: the call has ended (the `data` object contains a `reason` attribute) * accepted: the call has been accepted (either locally or remotely) * incoming: initial state for incoming calls * progress: initial state for outgoing calls * established: call media has been established #### Call.answer(options={}) Answer an incoming call. Supported options: * pcConfig: configuration options for `RTCPeerConnection`. [Reference](http://w3c.github.io/webrtc-pc/#configuration). * mediaConstraints: constraints to be used when getting the local user media. [Reference](http://www.w3.org/TR/mediacapture-streams/#mediastreamconstraints). * answerOptions: `RTCAnswerOptions`. [Reference](http://w3c.github.io/webrtc-pc/#idl-def-RTCAnswerOptions). * localStream: if specified, it will be used by sylkrtc instead of using `getUserMedia`. #### Call.terminate() End the call. #### Call.getLocalStreams() Returns an array of *local* `RTCMediaStream` objects. #### Call.getRemoteStreams() Returns an array of *remote* `RTCMediaStream` objects. #### Call.account Getter property which returns the `Account` object associated with this call. #### Call.id Getter property which returns the ID for this call. Note: this is not related to the SIP Call-ID header. #### Call.direction Getter property which returns the call direction: "incoming" or "outgoing". Note: this is not related to the SDP "a=" direction attribute. #### Call.state Getter property which returns the call state. #### Call.localIdentity Getter property which returns the local identity URI (SIP URI). #### Call.remoteIdentity Getter property which returns the remote identity URI (SIP URI). ## License MIT. See the `LICENSE` file in this directory. ## Credits -Special thanks to NLNET http://nlnet.nl for sponsoring most of the efforts behind this project. +Special thanks to [NLnet](http://nlnet.nl) for sponsoring most of the efforts behind this project.