diff --git a/API.md b/API.md index a231fdc..0be6628 100644 --- a/API.md +++ b/API.md @@ -1,461 +1,466 @@ ## 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). 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(oldToken, newToken) 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.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.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`. #### 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.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.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.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 '. diff --git a/lib/call.js b/lib/call.js index 5e61816..1cb4aa1 100644 --- a/lib/call.js +++ b/lib/call.js @@ -1,433 +1,444 @@ 'use strict'; import debug from 'debug'; import uuidv4 from 'uuid/v4'; 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._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._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 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.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 () => { utils.createLocalSdp(this._pc, 'answer', answerOptions) .then((sdp) => { DEBUG('Local SDP: %s', sdp); this._sendAnswer(sdp); }) .catch((reason) => { DEBUG(reason); this.terminate(); }); }, // failure (error) => { DEBUG('Error setting remote description: %s', error); this.terminate(); } ); } 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 = 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) { this._id = id; this._remoteIdentity = new utils.Identity(caller.uri, caller.display_name); this._incomingSdp = sdp; this._direction = 'incoming'; this._state = 'incoming'; + 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'); 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._pc.setRemoteDescription( new RTCSessionDescription({type: 'answer', sdp: sdp}), // success () => { 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 (error) => { DEBUG('Error accepting call: %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 (tracks.length != 0) { tempStream = new MediaStream(tracks); utils.closeMediaStream(tempStream); } } else { for (let stream of this._pc.getLocalStreams()) { 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 }; diff --git a/lib/utils.js b/lib/utils.js index c5f1a2d..1a5949e 100644 --- a/lib/utils.js +++ b/lib/utils.js @@ -1,140 +1,148 @@ 'use strict'; import transform from 'sdp-transform'; import attachMediaStream from 'attachmediastream'; class Identity { constructor(uri, displayName=null) { this._uri = uri; this._displayName = displayName; } get uri() { return this._uri; } get displayName() { return this._displayName; } toString() { if (!this._displayName) { return this._uri; } else { return `${this._displayName} <${this._uri}>`; } } } function createLocalSdp(pc, type, options) { if (type !== 'offer' && type !== 'answer') { return Promise.reject('type must be "offer" or "answer", but "' +type+ '" was given'); } let p = new Promise(function(resolve, reject) { let createFunc; if (type === 'offer' ) { createFunc = pc.createOffer; } else { createFunc = pc.createAnswer; } createFunc.call( pc, // success function(desc) { pc.setLocalDescription( desc, // success function() { resolve(mungeSdp(pc.localDescription.sdp)); }, // failure function(error) { reject('Error setting local description: ' + error); } ); }, // failure function(error) { reject('Error creating local SDP: ' + error); }, options ); }); return p; } function mungeSdp(sdp) { let parsedSdp = transform.parse(sdp); let h264payload = null; let hasProfileLevelId = false; // try to fix H264 support for (let media of parsedSdp.media) { if (media.type === 'video') { for (let rtp of media.rtp) { if (rtp.codec === 'H264') { h264payload = rtp.payload; break; } } if (h264payload !== null) { for (let fmtp of media.fmtp) { if (fmtp.payload === h264payload && fmtp.config.indexOf('profile-level-id') !== -1) { hasProfileLevelId = true; break; } } if (!hasProfileLevelId) { media.fmtp.push({payload: h264payload, config: 'profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1'}); } break; } } } // remove bogus rtcp-fb elements for (let media of parsedSdp.media) { let payloads = String(media.payloads).split(' '); if (media.rtcpFb) { media.rtcpFb = media.rtcpFb.filter((item) => { return payloads.indexOf(String(item.payload)) !== -1; }); } } return transform.write(parsedSdp); } +function getMediaDirections(sdp) { + const parsedSdp = transform.parse(sdp); + const directions = {}; + for (let media of parsedSdp.media) { + directions[media.type] = (directions[media.type] || []).concat(media.direction); + } + return directions; +} function closeMediaStream(stream) { if (!stream) { return; } // Latest spec states that MediaStream has no stop() method and instead must // call stop() on every MediaStreamTrack. if (MediaStreamTrack && MediaStreamTrack.prototype && MediaStreamTrack.prototype.stop) { if (stream.getTracks) { for (let track of stream.getTracks()) { track.stop(); } } else { for (let track of stream.getAudioTracks()) { track.stop(); } for (let track of stream.getVideoTracks()) { track.stop(); } } // Deprecated by the spec, but still in use. } else if (typeof stream.stop === 'function') { stream.stop(); } } -export default { Identity, createLocalSdp, mungeSdp, attachMediaStream, closeMediaStream }; +export default { Identity, createLocalSdp, mungeSdp, getMediaDirections, attachMediaStream, closeMediaStream};