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	diff --git a/API.md b/API.md
	index 9d4f469..75da7df 100644
	--- a/API.md
	+++ b/API.md
	@@ -1,645 +1,646 @@

	## 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 supported options are "server" and optional object "userAgent". Where server 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'});

	If the optional userAgent object is given, it should contain:
	* `name` : string with the name of the application.
	* `version`: version string of the application.

	Example with userAgent:

	let connection = sylkrtc.createConnection({server: 'wss://1.2.3.4:8088/webrtcgateway/ws', userAgent: {name: 'Some Apllication', version: '0.99.9'}});


	#### 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.
	* `sanatizeHtml`: function to XSS sanitize html strings

	### 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). The callback will be called with an error object.

	Example:

	connection.removeAccount(account, function(error) {
	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(token, platform, device, silent, app)

	Set the current device token for this account. The device token is an opaque string usually provided by the Firebase SDK
	which SylkServer will inject with the other parameters as parameters into to contact header when a SIP account is registered.
	The parameter `silent` must be a boolean and all other parameters should be strings.


	### 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
	+ * early-media: the call has an session description before it is accepted
	+ * established: call media has been established, in case of early media this happens before accepted
	* 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.startScreensharing(newTrack)

	Start sharing a screen/window. `newTrack` should be a `RTCMediaStreamTrack` containing the screen/window. Internally it will call
	replace track with the keep flag enabled and it will set the state so it can be tracked.


	#### Call.stopScreensharing()

	Stop sharing a screen/window and restore the previousTrack.


	#### Call.replaceTrack(oldTrack, newTrack, keep=false, cb=null)

	Replace a local track inside a call. If the keep flag is set, it will store the replaced track internally so it
	can be used later. The callback will be called with a true value once the operation completes.


	#### Call.terminate()

	End the call.


	#### Call.getLocalStreams()

	Returns an array of local `RTCMediaStream` objects.


	#### Call.getRemoteStreams()

	Returns an array of remote `RTCMediaStream` objects.


	#### Call.getSenders()

	Returns an array of `RTCRtpSender` objects.


	#### Call.getReceivers()

	Returns an array of `RTCRtpReceiver` 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.callId

	Getter property which returns the call-id for this call. Note: this is the SIP Call-ID.


	#### Call.sharingScreen

	Getter property which returns the screen sharing state.


	#### 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).


	#### Call.remoteMediaDirections

	Getter property which returns an object with the directions of the remote streams. Note: this is related to the SDP "a=" direction attribute.


	### 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`.
	* roomConfigured: emitted when the room is configured by the server. A single argument is provided: an object with the
	`originator` of the message which is an `Identity` or string and a list of `activeParticipants`. The list contains
	instances of `Participant`.
	* fileSharing: emitted when a participant in the room shares files. A single argument is provided: a list of instances of `SharedFile`.
	* message: emitted when a message is received. A single argument is provided, an instance of `Message`.
	* sendingMessage: emitted when a message will be sent. A single argument is provided, an instance of `Message`.
	* composingIndication: emitted when somebody in the room is typing. A single argument is provided, an object with `refresh`, `sender`
	and `state`. The `sender` is an `Identity`.
	* muteAudio: emitted when a `Participant` requests to `muteAudioParticipants`.
	* raisedHands: emitted when a `Participant` raises or lower his hand. A single argument is provided: a list of `raisedHands`.
	The list contains instances of `Participant`.

	#### Conference.startScreensharing(newTrack)

	Start sharing a screen/window. `newTrack` should be a `RTCMediaStreamTrack` containing the screen/window. Internally it will call
	replace track with the keep flag enabled and it will set the state so it can be tracked.


	#### Conference.stopScreensharing()

	Stop sharing a screen/window and restore the previousTrack.


	#### Conference.sendMessage(message, type)

	Send a chat message to the conference. `message` should contain a string, `type` should contain the message content type like
	'text/plain', 'text/html', 'image/png'. The function returns an instance of `Message`.


	#### Conference.sendComposing(state)

	Send a composing indication to the conference. `state` should be either `active` or `idle`.


	#### Conference.replaceTrack(oldTrack, newTrack, keep=false, cb=null)

	Replace a local track inside the conference. If the keep flag is set, it will store the replaced track internally so it
	can be used later. The callback will be called with a true value once the operation completes.


	#### 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.getSenders()

	Returns an array of `RTCRtpSender` objects. The sender objects get the local tracks being published to the conference.


	#### Conference.getReceivers()

	Returns an array of `RTCRtpReceiver` objects. The receiver objects get the remote tracks published by all other
	participants in the conference.


	#### Conference.scaleLocalTrack(track, divider)

	Scale the given local video track by a given divider. Currently this function will not work, since browser support is lacking.


	#### Conference.configureRoom(participants, cb=null)

	Configure the room. `Participants` is a list with the publisher session ids of the new active participants. The active participants
	will get more bandwidth and the other participants will get a limited bandwidth. On success the roomConfigured event is emitted.

	The `cb` argument is a callback which will be called on an error with error as argument.


	#### Conference.muteAudioParticipants()

	Request muting for all participants. All participants in the room will get a `muteAudio` event from the server.


	#### Conference.toggleHand(participantSession)

	Raise/Lower your hand. An optional participant session can be provided, so the hand of this specific session is raised/lowered.
	Calling this function will trigger a `raisedHands` event to all participants in the room.


	#### Conference.participants

	Getter property which returns an array of `Participant` objects in the conference.


	#### Conference.activeParticipants

	Getter property for the Active Participants which returns an array of `Participant` objects in the conference.


	#### Conference.sharedFiles

	Getter property for the Shared Files which returns an array of `SharedFile` objects in the conference.


	#### Conference.messages

	Getter property for the Messages which returns an array of `Message` objects in the conference.


	#### Conference.raisedHands

	Getter property for the Raised Hands which returns an array of `Participant` objects.


	#### 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.sharingScreen

	Getter property which returns the screen sharing state.


	#### 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.publisherId

	Getter property which returns the participant's publisher session id.


	#### Participant.streams

	Getter property which returns the audio / video streams for this participant.


	#### Participant.videoPaused

	Getter property which returns true / false when the video subscription is paused / not paused


	#### Participant.getReceivers()

	Returns an array of `RTCRtpReceiver` objects. The receiver objects get the remote tracks published by the
	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(isRemoved=false)

	Stop receiving audio / video from this participant. The opposite of `Participant.attach()`. The isRemoved
	option needs to be true used when the participant has already left. This is the case when you receive the
	'participantLeft' event.


	#### Participant.pauseVideo()

	Stop receiving video from this participant. The opposite of `Participant.resumeVideo()`.


	#### Participant.resumeVideo()

	Resume receiving video from this participant. The opposite of `Participant.pauseVideo()`.


	### 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 <bob@biloxi.com>'.


	### SharedFile

	Object representing a shared file.


	#### SharedFile.filename

	The filename of the shared file


	#### SharedFile.filesize

	The filesize in bytes of the shared file


	#### SharedFile.uploader

	The `Identity` of the uploader.


	#### SharedFile.session

	The session UUID which was used to upload the file


	### Message

	Object representing a message.

	Events emitted:
	* stateChanged: indicates the message state has changed. Two arguments are provided: `oldState`, `newState`.
	`oldState` and `newState` indicate the previous and current state respectively. Possible states:
	* received: the message was received
	* pending: the message is pending delivery
	* delivered: the message has been delivered
	* failed: something went wrong, either it is not delivered, or it could not be sent


	#### Message.id

	Getter property for id the message


	#### Message.content

	Getter property for the content of the message. In case content type of the message is 'text/html', it will be sanatized.


	#### Message.content

	Getter property for the content type of the message.


	#### Message.sender

	Getter property for the `Identity` of the message sender.


	#### Message.timestamp

	Getter property for the `Date` object of the message.


	#### Message.type

	Getter property for the type of the message, it can be `normal` or `status`.


	#### Message.state

	Getter property for the state of the message. It can be `received`, `pending`, `delivered`, `failed`.

	diff --git a/lib/call.js b/lib/call.js
	index 01930c1..cf4f1cc 100644
	--- a/lib/call.js
	+++ b/lib/call.js
	@@ -1,508 +1,510 @@
	'use strict';

	import debug from 'debug';
	import { v4 as uuidv4 } from 'uuid';
	import utils from './utils';

	import { EventEmitter } from 'events';

	const DEBUG = debug('sylkrtc:Call');


	class Call extends EventEmitter {
	constructor(account) {
	super();
	this._account = account;
	this._id = null;
	this._callId = null;
	this._direction = null;
	this._pc = null;
	this._state = null;
	this._terminated = false;
	this._incomingSdp = null;
	this._remoteMediaDirections = {};
	this._localIdentity = new utils.Identity(account.id, account.displayName);
	this._remoteIdentity = null;
	this._remoteStreams = new MediaStream();
	this._localStreams = new MediaStream();
	this._previousTrack = null;
	this._sharingScreen = false;
	this._dtmfSender = null;
	this._delay_established = false; // set to true when we need to delay posting the state change to 'established'
	this._setup_in_progress = false; // set while we set the remote description and setup the peer copnnection

	// bind some handlers to this instance
	this._onDtmf = this._onDtmf.bind(this);
	}

	get account() {
	return this._account;
	}

	get id() {
	return this._id;
	}

	get callId() {
	return this._callId;
	}

	get sharingScreen() {
	return this._sharingScreen;
	}

	get direction() {
	return this._direction;
	}

	get state() {
	return this._state;
	}

	get localIdentity() {
	return this._localIdentity;
	}

	get remoteIdentity() {
	return this._remoteIdentity;
	}

	get remoteMediaDirections() {
	return this._remoteMediaDirections;
	}

	getLocalStreams() {
	if (this._pc !== null) {
	if (this._pc.getSenders) {
	this._pc.getSenders().forEach((e) => {
	if (e.track != null) {
	if (e.track.readyState !== "ended") {
	this._localStreams.addTrack(e.track);
	} else {
	this._localStreams.removeTrack(e.track);
	}
	}
	});
	return [this._localStreams];
	} else {
	return this._pc.getLocalStreams();
	}
	} else {
	return [];
	}
	}

	getRemoteStreams() {
	if (this._pc !== null) {
	if (this._pc.getReceivers) {
	this._pc.getReceivers().forEach((e) => {
	if (e.track.readyState !== "ended") {
	this._remoteStreams.addTrack(e.track);
	}
	});
	return [this._remoteStreams];
	} else {
	return this._pc.getRemoteStreams();
	}
	} else {
	return [];
	}
	}

	getSenders() {
	if (this._pc !== null) {
	return this._pc.getSenders();
	} else {
	return [];
	}
	}

	getReceivers() {
	if (this._pc !== null) {
	return this._pc.getReceivers();
	} else {
	return [];
	}
	}

	answer(options = {}) {
	if (this._state !== 'incoming') {
	throw new Error('Call is not in the incoming state: ' + this._state);
	}

	if (!options.localStream) {
	throw new Error('Missing localStream');
	}

	const pcConfig = options.pcConfig \|\| {iceServers:[]};
	const answerOptions = options.answerOptions;

	// Create the RTCPeerConnection
	this._initRTCPeerConnection(pcConfig);

	this._pc.addStream(options.localStream);
	this.emit('localStreamAdded', options.localStream);
	this._pc.setRemoteDescription(new RTCSessionDescription({type: 'offer', sdp: this._incomingSdp}))
	// success
	.then(() => {
	utils.createLocalSdp(this._pc, 'answer', answerOptions)
	.then((sdp) => {
	DEBUG('Local SDP: %s', sdp);
	this._sendAnswer(sdp);
	})
	.catch((reason) => {
	DEBUG(reason);
	this.terminate();
	});
	})
	// failure
	.catch((error) => {
	DEBUG('Error setting remote description: %s', error);
	this.terminate();
	});
	}

	startScreensharing(newTrack) {
	let oldTrack = this.getLocalStreams()[0].getVideoTracks()[0];
	this.replaceTrack(oldTrack, newTrack, true, (value) => {
	this._sharingScreen = value;
	});
	}

	stopScreensharing() {
	let oldTrack = this.getLocalStreams()[0].getVideoTracks()[0];
	this.replaceTrack(oldTrack, this._previousTrack);
	this._sharingScreen = false;
	}

	replaceTrack(oldTrack, newTrack, keep=false, cb=null) {
	let sender;
	for (sender of this._pc.getSenders()) {
	if (sender.track === oldTrack) {
	break;
	}
	}

	sender.replaceTrack(newTrack)
	.then(() => {
	if (keep) {
	this._previousTrack = oldTrack;
	} else {
	if (oldTrack) {
	oldTrack.stop();
	}
	if (newTrack === this._previousTrack) {
	this._previousTrack = null;
	}
	}

	if (oldTrack) {
	this._localStreams.removeTrack(oldTrack);
	}
	this._localStreams.addTrack(newTrack);

	if (cb) {
	cb(true);
	}
	}).catch((error)=> {
	DEBUG('Error replacing track: %s', error);
	});
	}

	terminate() {
	if (this._terminated) {
	return;
	}
	DEBUG('Terminating call');
	this._sendTerminate();
	}

	sendDtmf(tones, duration=100, interToneGap=70) {
	DEBUG('sendDtmf()');
	if (this._dtmfSender === null) {
	if (this._pc !== null) {
	let track = null;
	try {
	track = this._pc.getLocalStreams()[0].getAudioTracks()[0];
	} catch (e) {
	// ignore
	}
	if (track !== null) {
	DEBUG('Creating DTMF sender');
	this._dtmfSender = this._pc.createDTMFSender(track);
	if (this._dtmfSender) {
	this._dtmfSender.addEventListener('tonechange', this._onDtmf);
	}
	}
	}
	}
	if (this._dtmfSender) {
	DEBUG('Sending DTMF tones');
	this._dtmfSender.insertDTMF(tones, duration, interToneGap);
	}
	}

	// Private API

	_initOutgoing(uri, options={}) {
	if (uri.indexOf('@') === -1) {
	throw new Error('Invalid URI');
	}

	if (!options.localStream) {
	throw new Error('Missing localStream');
	}

	this._id = options.id \|\| uuidv4();
	this._direction = 'outgoing';
	this._remoteIdentity = new utils.Identity(uri);

	const pcConfig = options.pcConfig \|\| {iceServers:[]};
	const offerOptions = options.offerOptions;

	// Create the RTCPeerConnection
	this._initRTCPeerConnection(pcConfig);

	this._pc.addStream(options.localStream);
	this.emit('localStreamAdded', options.localStream);
	utils.createLocalSdp(this._pc, 'offer', offerOptions)
	.then((sdp) => {
	DEBUG('Local SDP: %s', sdp);
	this._sendCall(uri, sdp);
	})
	.catch((reason) => {
	DEBUG(reason);
	this._localTerminate(reason);
	});
	}

	_initIncoming(id, caller, sdp, callId) {
	this._id = id;
	this._remoteIdentity = new utils.Identity(caller.uri, caller.display_name);
	this._incomingSdp = sdp;
	this._direction = 'incoming';
	this._state = 'incoming';
	this._callId = callId;
	this._remoteMediaDirections = Object.assign(
	{audio: [], video:[]}, utils.getMediaDirections(sdp)
	);
	DEBUG('Remote SDP: %s', sdp);
	}

	_handleEvent(message) {
	DEBUG('Call event: %o', message);
	switch (message.event) {
	case 'state':
	let oldState = this._state;
	let newState = message.state;
	this._state = newState;

	- if (newState === 'accepted' && this._direction === 'outgoing') {
	- DEBUG('Call accepted');
	+ if ((newState === 'accepted' \|\| newState === 'early-media') && this._direction === 'outgoing') {
	+ DEBUG('Call accepted or early media');
	this.emit('stateChanged', oldState, newState, {});
	- const sdp = utils.mungeSdp(message.sdp);
	- DEBUG('Remote SDP: %s', sdp);
	- this._remoteMediaDirections = Object.assign(
	- {audio: [], video:[]}, utils.getMediaDirections(sdp)
	- );
	- this._setup_in_progress = true;
	- this._callId = message.call_id;
	- this._pc.setRemoteDescription(new RTCSessionDescription({type: 'answer', sdp: sdp}))
	- // success
	- .then(() => {
	- this._setup_in_progress = false;
	- if (!this._terminated) {
	- if (this._delay_established) {
	- oldState = this._state;
	- this._state = 'established';
	- DEBUG('Setting delayed established state!');
	- this.emit('stateChanged', oldState, this._state, {});
	- this._delay_established = false;
	+ if (message.sdp !== 'undefined') {
	+ const sdp = utils.mungeSdp(message.sdp);
	+ DEBUG('Remote SDP: %s', sdp);
	+ this._remoteMediaDirections = Object.assign(
	+ {audio: [], video:[]}, utils.getMediaDirections(sdp)
	+ );
	+ this._setup_in_progress = true;
	+ this._callId = message.call_id;
	+ this._pc.setRemoteDescription(new RTCSessionDescription({type: 'answer', sdp: sdp}))
	+ // success
	+ .then(() => {
	+ this._setup_in_progress = false;
	+ if (!this._terminated) {
	+ if (this._delay_established) {
	+ oldState = this._state;
	+ this._state = 'established';
	+ DEBUG('Setting delayed established state!');
	+ this.emit('stateChanged', oldState, this._state, {});
	+ this._delay_established = false;
	+ }
	}
	- }
	- })
	- // failure
	- .catch((error) => {
	- DEBUG('Error accepting call: %s', error);
	- this.terminate();
	- });
	+ })
	+ // failure
	+ .catch((error) => {
	+ DEBUG('Error accepting call or early media: %s', error);
	+ this.terminate();
	+ });
	+ }
	} else if (newState === 'established' && this._direction === 'outgoing') {
	if (this._setup_in_progress) {
	this._delay_established = true;
	} else {
	this.emit('stateChanged', oldState, newState, {});
	}
	} else if (newState === 'terminated') {
	this.emit('stateChanged', oldState, newState, {reason: message.reason});
	this._terminated = true;
	this._account._calls.delete(this.id);
	this._closeRTCPeerConnection();
	} else {
	this.emit('stateChanged', oldState, newState, {});
	}
	break;
	default:
	break;
	}
	}

	_initRTCPeerConnection(pcConfig) {
	if (this._pc !== null) {
	throw new Error('RTCPeerConnection already initialized');
	}

	this._pc = new RTCPeerConnection(pcConfig);
	this._pc.addEventListener('addstream', (event) => {
	DEBUG('Stream added');
	this.emit('streamAdded', event.stream);
	});
	this._pc.addEventListener('icecandidate', (event) => {
	if (event.candidate !== null) {
	DEBUG('New ICE candidate %o', event.candidate);
	} else {
	DEBUG('ICE candidate gathering finished');
	}
	this._sendTrickle(event.candidate);
	});
	}

	_sendRequest(req, cb) {
	this._account._sendRequest(req, cb);
	}

	_sendCall(uri, sdp) {
	const req = {
	sylkrtc: 'session-create',
	account: this.account.id,
	session: this.id,
	uri: uri,
	sdp: sdp
	};
	this._sendRequest(req, (error) => {
	if (error) {
	DEBUG('Call error: %s', error);
	this._localTerminate(error);
	}
	});
	}

	_sendTerminate() {
	const req = {
	sylkrtc: 'session-terminate',
	session: this.id
	};
	this._sendRequest(req, (error) => {
	if (error) {
	DEBUG('Error terminating call: %s', error);
	this._localTerminate(error);
	}
	});
	setTimeout(() => {
	if (!this._terminated) {
	DEBUG('Timeout terminating call');
	this._localTerminate('200 OK');
	}
	this._terminated = true;
	}, 150);
	}

	_sendTrickle(candidate) {
	const req = {
	sylkrtc: 'session-trickle',
	session: this.id,
	candidates: candidate !== null ? [candidate] : [],
	};
	this._sendRequest(req, (error) => {
	if (error) {
	DEBUG('Trickle error: %s', error);
	this._localTerminate(error);
	}
	});
	}

	_sendAnswer(sdp) {
	const req = {
	sylkrtc: 'session-answer',
	session: this.id,
	sdp: sdp
	};
	this._sendRequest(req, (error) => {
	if (error) {
	DEBUG('Answer error: %s', error);
	this.terminate();
	}
	});
	}

	_closeRTCPeerConnection() {
	DEBUG('Closing RTCPeerConnection');
	if (this._pc !== null) {
	let tempStream;
	if (this._pc.getSenders) {
	let tracks = [];
	for (let track of this._pc.getSenders()) {
	if (track.track != null ) {
	tracks = tracks.concat(track.track);
	}
	if (this._previousTrack !== null) {
	tracks = tracks.concat(this._previousTrack);
	}
	}
	if (tracks.length !== 0) {
	tempStream = new MediaStream(tracks);
	utils.closeMediaStream(tempStream);
	}
	} else {
	for (let stream of this._pc.getLocalStreams()) {
	if (this._previousTrack !== null) {
	stream = stream.concat(this._previousTrack);
	}
	utils.closeMediaStream(stream);
	}
	}

	if (this._pc.getReceivers) {
	let tracks = [];
	for (let track of this._pc.getReceivers()) {
	tracks = tracks.concat(track.track);
	}
	tempStream = new MediaStream(tracks);
	utils.closeMediaStream(tempStream);
	} else {
	for (let stream of this._pc.getRemoteStreams()) {
	utils.closeMediaStream(stream);
	}
	}
	this._pc.close();
	this._pc = null;
	if (this._dtmfSender !== null) {
	this._dtmfSender.removeEventListener('tonechange', this._onDtmf);
	this._dtmfSender = null;
	}
	}
	}

	_localTerminate(error) {
	if (this._terminated) {
	return;
	}
	DEBUG('Local terminate');
	this._account._calls.delete(this.id);
	this._terminated = true;
	const oldState = this._state;
	const newState = 'terminated';
	const data = {
	reason: error.toString()
	};
	this._closeRTCPeerConnection();
	this.emit('stateChanged', oldState, newState, data);
	}

	_onDtmf(event) {
	DEBUG('Sent DTMF tone %s', event.tone);
	this.emit('dtmfToneSent', event.tone);
	}
	}


	export { Call };

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