diff --git a/API.md b/API.md index 3834657..a17bfd1 100644 --- a/API.md +++ b/API.md @@ -1,386 +1,391 @@ ## 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.utils Helper module with utility functions. * `attachMediaStream`: function to easily attach a media stream to an element. It reexports [attachmediastream](https://github.com/otalk/attachMediaStream). * `closeMediaStream`: function to close the given media stream. ### 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*. An optional *displayName* can be set. The account won't be registered, it will just be created. Optionally *realm* can be passed, which will be used instead of the domain for the HA1 calculation. 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.reconnect() Starts reconnecting immediately if the state was 'disconnected'; #### 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, which is an `Identity` object. * **conferenceInvite**: emitted when someone invites us to join a conference. A `data` object is provided, which contains an `originator` attribute indicating who invited us, and a `room` attribute indicating what conference we have been invited to. #### 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). * offerOptions: `RTCOfferOptions`. [Reference](http://w3c.github.io/webrtc-pc/#idl-def-RTCOfferOptions). * localStream: user provided local media stream (acquired with `getUserMedia` TODO). Example: const call = account.call('3333@sip2sip.info', {localStream: stream}); #### Account.joinConference(uri, options={}) Join (or create in case it doesn't exist) a multi-party video conference at the given URI. Supported options: * pcConfig: configuration options for `RTCPeerConnection`. [Reference](http://w3c.github.io/webrtc-pc/#configuration). * offerOptions: `RTCOfferOptions`. [Reference](http://w3c.github.io/webrtc-pc/#idl-def-RTCOfferOptions). * localStream: user provided local media stream (acquired with `getUserMedia` TODO). Example: const conf = account.joinConference('test123@conference.sip2sip.info', {localStream: stream}); #### Account.id Getter property returning the account ID. #### Account.displayName Getter property returning the account display name. #### Account.password Getter property returning the HA1 password for the account. #### Account.registrationState getter property returning the current registration state. +#### Account.setDeviceToken(oldToken, newTokenn) + +Set the current device token for this account. The device token is an opaque string usually provided by the Firebase SDK +which SylkServer can use to send push notifications. + ### 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 * **dtmfToneSent**: emitted when one of the tones passed to `sendDtmf` is actually sent. An empty tone indicates all tones have finished playing. #### Call.answer(options={}) Answer an incoming call. Supported options: * pcConfig: configuration options for `RTCPeerConnection`. [Reference](http://w3c.github.io/webrtc-pc/#configuration). * answerOptions: `RTCAnswerOptions`. [Reference](http://w3c.github.io/webrtc-pc/#idl-def-RTCAnswerOptions). * localStream: user provided local media stream (acquired with `getUserMedia` TODO). #### Call.terminate() End the call. #### Call.getLocalStreams() Returns an array of *local* `RTCMediaStream` objects. #### Call.getRemoteStreams() Returns an array of *remote* `RTCMediaStream` objects. #### Call.sendDtmf(tones, duration=100, interToneGap=70) Sends the given DTMF tones over the active audio stream track. **Note**: This feature requires browser support for `RTCPeerConnection.createDTMFSender`. #### 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. (See the `Identity` object). #### Call.remoteIdentity Getter property which returns the remote identity. (See the `Identity` object). ### Conference Object representing a multi-party audio/video conference. Events emitted: * **localStreamAdded**: emitted when the local stream is added to the call. A single argument is provided: the stream itself. * **stateChanged**: indicates the conference 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 conference has ended * accepted: the initial offer has been accepted * progress: initial state * established: conference has been established and media is flowing * **participantJoined**: emitted when a participant joined the conference. A single argument is provided: an instance of `Participant`. Note that this event is only emitted when new participants join, `Conference.participants` should be checked upon the initial join to check what participants are already in the conference. * **participantLeft**: emitted when a participant leaves the conference. A single argument is provided: an instance of `Participant`. #### Conference.getLocalStreams() Returns an array of *local* `RTCMediaStream` objects. These are the streams being published to the conference. #### Conference.getRemoteStreams() Returns an array of *remote* `RTCMediaStream` objects. These are the streams published by all other participants in the conference. #### Conference.participants Getter property which returns an array of `Participant` objects in the conference. #### Conference.account Getter property which returns the `Account` object associated with this conference. #### Conference.id Getter property which returns the ID for this conference. Note: this is not related to the URI. #### Conference.direction Dummy property always returning "outgoing", in order to provide the same API as `Call`. #### Conference.state Getter property which returns the conference state. #### Conference.localIdentity Getter property which returns the local identity. (See the `Identity` object). This will always be built from the account. #### Conference.remoteIdentity Getter property which returns the remote identity. (See the `Identity` object). This will always be built from the remote URI. ### Participant Object representing another user connected to the same conference. Events emitted: * **streamAdded**: emitted when a remote stream is added. A single argument is provided: the stream itself. * **stateChanged**: indicates the participant 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: * null: initial state * progress: the participant is being attached to, this will happen as a result to `Participant.attach` * established: media is flowing from this participant #### Participant.id Getter property which returns the ID for this participant. Note this an abstract ID. #### Participant.state Getter property which returns the participant state. #### Participant.identity Getter property which returns the participant's identity. (See the `Identity` object). #### Participant.streams Getter property which returns the audio / video streams for this participant. #### Participant.attach() Start receiving audio / video from this participant. Once attached the participant's state will switch to 'established' and its audio /video stream(s) will be available in `Participant.streams`. If a participant is not attached to, no audio or video will be received from them. #### Participant.detach() Stop receiving audio / video from this participant. The opposite of `Participant.attach()`. ### Identity Object representing the identity of the caller / callee. #### Identity.uri SIP URI, without the 'sip:' prefix. #### Identity.displayName Display name assiciated with the identity. Set to '' if absent. #### Identity.toString() Function returning a string representation of the identity. It can take 2 forms depending on the availability of the display name: 'bob@biloxi.com' or 'Bob '. diff --git a/lib/account.js b/lib/account.js index 36096ee..f330742 100644 --- a/lib/account.js +++ b/lib/account.js @@ -1,150 +1,165 @@ 'use strict'; import debug from 'debug'; import md5 from 'blueimp-md5'; import transform from 'sdp-transform'; import utils from './utils'; import { EventEmitter } from 'events'; import { Call } from './call'; import { ConferenceCall } from './conference'; const DEBUG = debug('sylkrtc:Account'); class Account extends EventEmitter { constructor(options, connection) { if (options.account.indexOf('@') === -1) { throw new Error('Invalid account id specified'); } super(); const id = options.account; const [username, domain] = id.split('@'); this._id = id; this._displayName = options.displayName; this._password = md5(username + ':' + (options.realm || domain)+ ':' + options.password); this._connection = connection; this._registrationState = null; this._calls = new Map(); this._confCalls = new Map(); } get id() { return this._id; } get password() { return this._password; } get displayName() { return this._displayName; } get registrationState() { return this._registrationState; } register() { const req = { sylkrtc: 'account-register', account: this._id }; this._sendRequest(req, (error) => { if (error) { DEBUG('Register error: %s', error); const oldState = this._registrationState; const newState = 'failed'; const data = {reason: error.toString()}; this._registrationState = newState; this.emit('registrationStateChanged', oldState, newState, data); } }); } unregister() { const req = { sylkrtc: 'account-unregister', account: this._id, }; this._sendRequest(req, (error) => { if (error) { DEBUG('Unregister error: %s', error); } const oldState = this._registrationState; const newState = null; this._registrationState = newState; this.emit('registrationStateChanged', oldState, newState, {}); }); } call(uri, options={}) { const callObj = new Call(this); callObj._initOutgoing(uri, options); this._calls.set(callObj.id, callObj); this.emit('outgoingCall', callObj); return callObj; } joinConference(uri, options={}) { const confCall = new ConferenceCall(this); confCall._initialize(uri, options); this._confCalls.set(confCall.id, confCall); this.emit('conferenceCall', confCall); return confCall; } + setDeviceToken(oldToken, newToken) { + DEBUG('Setting device token: %s -> %s', oldToken, newToken); + const req = { + sylkrtc: 'account-devicetoken', + account: this._id, + old_token: oldToken, + new_token: newToken + }; + this._sendRequest(req, (error) => { + if (error) { + DEBUG('Error setting device token: %s', error); + } + }); + } + // Private API _handleEvent(message) { DEBUG('Received account event: %s', message.event); const data = {}; switch (message.event) { case 'registration_state': const oldState = this._registrationState; const newState = message.data.state; this._registrationState = newState; if (newState === 'failed') { data.reason = message.data.reason; } this.emit('registrationStateChanged', oldState, newState, data); break; case 'incoming_session': let call = new Call(this); call._initIncoming(message.session, message.data.originator, message.data.sdp); this._calls.set(call.id, call); // see what media types are offered const mediaTypes = { audio: false, video: false }; const parsedSdp = transform.parse(message.data.sdp); for (let media of parsedSdp.media) { if (media.type === 'audio' && media.port !== 0 && media.direction === 'sendrecv') { mediaTypes.audio = true; } else if (media.type === 'video' && media.port !== 0 && media.direction === 'sendrecv') { mediaTypes.video = true; } } DEBUG('Incoming call from %s with media types: %o', message.data.originator.uri, mediaTypes); this.emit('incomingCall', call, mediaTypes); break; case 'missed_session': data.originator = new utils.Identity(message.data.originator.uri, message.data.originator.display_name); this.emit('missedCall', data); break; case 'conference_invite': data.originator = new utils.Identity(message.data.originator.uri, message.data.originator.display_name); data.room = message.data.room; this.emit('conferenceInvite', data); break; default: break; } } _sendRequest(req, cb) { this._connection._sendRequest(req, cb); } } export { Account };