diff --git a/API.md b/API.md index c45269a..e0f306c 100644 --- a/API.md +++ b/API.md @@ -1,411 +1,414 @@ ## 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, 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.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.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.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.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()`. #### 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/conference.js b/lib/conference.js index bf94cdf..93c4297 100644 --- a/lib/conference.js +++ b/lib/conference.js @@ -1,563 +1,567 @@ 'use strict'; import debug from 'debug'; import uuidv4 from 'uuid/v4'; import utils from './utils'; import { EventEmitter } from 'events'; const DEBUG = debug('sylkrtc:Conference'); class Participant extends EventEmitter { constructor(publisherId, identity, conference) { super(); this._id = uuidv4(); this._publisherId = publisherId; this._identity = identity; this._conference = conference; this._state = null; this._pc = null; this._videoSubscriptionPaused = false; this._audioSubscriptionPaused = false; this._videoPublishingPaused = false; this._audioPublishingPaused = false; } get id() { return this._id; } + get publisherId() { + return this._publisherId; + } + get identity() { return this._identity; } get conference() { return this._conference; } get videoPaused() { return this._videoSubscriptionPaused; } get state() { return this._state; } get streams() { if (this._pc !== null) { return this._pc.getRemoteStreams(); } else { return []; } } attach() { if (this._state !== null) { return; } this._setState('progress'); this._sendAttach(); } detach() { if (this._state !== null) { this._sendDetach(); } } pauseVideo() { this._sendUpdate({video: false}); this._videoSubscriptionPaused = true; } resumeVideo() { this._sendUpdate({video: true}); this._videoSubscriptionPaused = false; } _setState(newState) { const oldState = this._state; this._state = newState; DEBUG(`Participant ${this.id} state change: ${oldState} -> ${newState}`); this.emit('stateChanged', oldState, newState); } _handleOffer(offerSdp) { DEBUG('Handling SDP for participant offer: %s', offerSdp); // Create the RTCPeerConnection const pcConfig = this.conference._pcConfig; const pc = new RTCPeerConnection(pcConfig); pc.addEventListener('addstream', (event) => { DEBUG('Stream added'); this.emit('streamAdded', event.stream); }); 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); }); this._pc = pc; // no need for a local stream since we are only going to receive media here pc.setRemoteDescription( new RTCSessionDescription({type: 'offer', sdp: offerSdp}), // success () => { utils.createLocalSdp(pc, 'answer') .then((sdp) => { DEBUG('Local SDP: %s', sdp); this._sendAnswer(sdp); }) .catch((reason) => { DEBUG(reason); this._close(); }); }, // failure (error) => { DEBUG('Error setting remote description: %s', error); this._close(); } ); } _sendAttach() { const req = { sylkrtc: 'videoroom-ctl', session: this.conference.id, option: 'feed-attach', feed_attach: { session: this.id, publisher: this._publisherId } }; DEBUG('Sending request: %o', req); this.conference._sendRequest(req, (error) => { if (error) { DEBUG('Error attaching to participant %s: %s', this._publisherId, error); } }); } _sendDetach() { const req = { sylkrtc: 'videoroom-ctl', session: this.conference.id, option: 'feed-detach', feed_detach: { session: this.id } }; DEBUG('Sending request: %o', req); this.conference._sendRequest(req, (error) => { if (error) { DEBUG('Error detaching to participant %s: %s', this._publisherId, error); } this._close(); }); } _sendTrickle(candidate) { const req = { sylkrtc: 'videoroom-ctl', session: this.conference.id, option: 'trickle', trickle: { session: this.id, candidates: candidate !== null ? [candidate] : [] } }; this.conference._sendRequest(req, null); } _sendAnswer(sdp) { const req = { sylkrtc: 'videoroom-ctl', session: this.conference.id, option: 'feed-answer', feed_answer: { session: this.id, sdp: sdp } }; this.conference._sendRequest(req, (error) => { if (error) { DEBUG('Answer error: %s', error); this._close(); } }); } _sendUpdate(options = {}) { const req = { sylkrtc: 'videoroom-ctl', session: this.id, option: 'update', update: {} }; req.update = Object.assign({}, options); DEBUG('Sending update participant request %o', req); this.conference._sendRequest(req, (error) => { if (error) { DEBUG('Answer error: %s', error); } }); } _close() { DEBUG('Closing Participant RTCPeerConnection'); if (this._pc !== null) { for (let stream of this._pc.getLocalStreams()) { utils.closeMediaStream(stream); } for (let stream of this._pc.getRemoteStreams()) { utils.closeMediaStream(stream); } this._pc.close(); this._pc = null; this._setState(null); } } } class ConferenceCall extends EventEmitter { constructor(account) { super(); this._account = account; this._id = null; this._pc = null; this._participants = new Map(); this._terminated = false; this._state = null; this._localIdentity = new utils.Identity(account.id, account.displayName); this._remoteIdentity = null; this._pcConfig = null; // saved on initialize, used later for subscriptions } get account() { return this._account; } get id() { return this._id; } get direction() { // make this object API compatible with `Call` return 'outgoing'; } get state() { return this._state; } get localIdentity() { return this._localIdentity; } get remoteIdentity() { return this._remoteIdentity; } get participants() { return Array.from(this._participants.values()); } getLocalStreams() { if (this._pc !== null) { return this._pc.getLocalStreams(); } else { return []; } } getRemoteStreams() { let streams = []; for (let participant of this._participants) { streams = streams.concat(participant.streams); } return streams; } scaleLocalTrack(oldTrack, divider) { DEBUG('Scaling track by %d', divider); let sender; for (sender of this._pc.getSenders()) { if (sender.track === oldTrack) { DEBUG('Found sender to modify track %o', sender); break; } } sender.setParameters({encodings: [{scaleResolutionDownBy: divider}]}) .then(() => { DEBUG("Scale set to %o", divider); DEBUG('Active encodings %o', sender.getParameters().encodings); }) .catch((error) => { DEBUG('Error %o', error) }); } terminate() { if (this._terminated) { return; } DEBUG('Terminating conference'); this._sendTerminate(); } inviteParticipants(ps) { if (this._terminated) { return; } if (!Array.isArray(ps) || ps.length === 0) { return; } DEBUG('Inviting participants: %o', ps); const req = { sylkrtc: 'videoroom-ctl', session: this.id, option: 'invite-participants', invite_participants: { participants: ps } }; this._sendRequest(req, null); } // Private API _initialize(uri, options={}) { if (this._id !== null) { throw new Error('Already initialized'); } if (uri.indexOf('@') === -1) { throw new Error('Invalid URI'); } if (!options.localStream) { throw new Error('Missing localStream'); } this._id = uuidv4(); this._remoteIdentity = new utils.Identity(uri); options = Object.assign({}, options); const pcConfig = options.pcConfig || {iceServers:[]}; this._pcConfig = pcConfig; this._initialParticipants = options.initialParticipants || []; const offerOptions = options.offerOptions || {}; // only send audio / video through the publisher connection offerOptions.offerToReceiveAudio = false; offerOptions.offerToReceiveVideo = false; delete offerOptions.mandatory; // Create the RTCPeerConnection this._pc = new RTCPeerConnection(pcConfig); 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); }); this._pc.addStream(options.localStream); this.emit('localStreamAdded', options.localStream); DEBUG('Offer options: %o', offerOptions); utils.createLocalSdp(this._pc, 'offer', offerOptions) .then((sdp) => { DEBUG('Local SDP: %s', sdp); this._sendJoin(sdp); }) .catch((reason) => { this._localTerminate(reason); }); } _handleEvent(message) { DEBUG('Conference event: %o', message); switch (message.event) { case 'state': const oldState = this._state; const newState = message.data.state; this._state = newState; let data = {}; let participant; if (newState === 'accepted') { let sdp = utils.mungeSdp(message.data.sdp); DEBUG('Remote SDP: %s', sdp); this._pc.setRemoteDescription( new RTCSessionDescription({type: 'answer', sdp: sdp}), // success () => { DEBUG('Conference accepted'); this.emit('stateChanged', oldState, newState, data); if (this._initialParticipants.length > 0 ) { setTimeout(() => { this.inviteParticipants(this._initialParticipants); }, 50); } }, // failure (error) => { DEBUG('Error processing conference accept: %s', error); this.terminate(); } ); } else { if (newState === 'terminated') { data.reason = message.data.reason; this._terminated = true; this._close(); } this.emit('stateChanged', oldState, newState, data); } break; case 'initial_publishers': // this comes between 'accepted' and 'established' states for (let p of message.data.publishers) { participant = new Participant(p.id, new utils.Identity(p.uri, p.display_name), this); this._participants.set(participant.id, participant); } break; case 'publishers_joined': for (let p of message.data.publishers) { DEBUG('Participant joined: %o', p); participant = new Participant(p.id, new utils.Identity(p.uri, p.display_name), this); this._participants.set(participant.id, participant); this.emit('participantJoined', participant); } break; case 'publishers_left': for (let pId of message.data.publishers) { for (participant of this._participants.values()) { if (pId === participant._publisherId) { this._participants.delete(participant.id); this.emit('participantLeft', participant); } } } break; case 'feed_attached': participant = this._participants.get(message.data.subscription); if (participant) { participant._handleOffer(message.data.sdp); } break; case 'feed_established': participant = this._participants.get(message.data.subscription); if (participant) { participant._setState('established'); } break; default: break; } } _sendJoin(sdp) { const req = { sylkrtc: 'videoroom-join', account: this.account.id, session: this.id, uri: this.remoteIdentity.uri, sdp: sdp }; DEBUG('Sending request: %o', req); this._sendRequest(req, (error) => { if (error) { this._localTerminate(error); } }); } _sendTerminate() { const req = { sylkrtc: 'videoroom-terminate', session: this.id }; this._sendRequest(req, (error) => { if (error) { DEBUG('Error terminating conference: %s', error); this._localTerminate(error); } }); setTimeout(() => { if (!this._terminated) { DEBUG('Timeout terminating call'); this._localTerminate(''); } this._terminated = true; }, 150); } _sendTrickle(candidate) { const req = { sylkrtc: 'videoroom-ctl', session: this.id, option: 'trickle', trickle: { candidates: candidate !== null ? [candidate] : [] } }; this._sendRequest(req, null); } _sendRequest(req, cb) { this._account._sendRequest(req, cb); } _close() { DEBUG('Closing RTCPeerConnection'); if (this._pc !== null) { for (let stream of this._pc.getLocalStreams()) { utils.closeMediaStream(stream); } for (let stream of this._pc.getRemoteStreams()) { utils.closeMediaStream(stream); } this._pc.close(); this._pc = null; } const participants = this.participants; this._participants = []; for (let p of participants) { p._close(); } } _localTerminate(reason) { if (this._terminated) { return; } DEBUG(`Local terminate, reason: ${reason}`); this._account._confCalls.delete(this.id); this._terminated = true; const oldState = this._state; const newState = 'terminated'; const data = { reason: reason.toString() }; this._close(); this.emit('stateChanged', oldState, newState, data); } } export { ConferenceCall };