@wazo/sdk 0.43.5 → 1.0.0-alpha.0
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/dist/wazo-sdk.js +3 -3
- package/esm/__tests__/__voice__/call.test.js +72 -0
- package/esm/__tests__/__voice__/softphone.test.js +66 -0
- package/esm/api/auth.js +1 -5
- package/esm/checker/checks/webrtc.js +2 -1
- package/esm/domain/InvalidState.js +8 -0
- package/esm/domain/InvalidStateTransition.js +10 -0
- package/esm/domain/Phone/WebRTCPhone.js +33 -35
- package/esm/simple/Auth.js +12 -6
- package/esm/simple/Phone.js +2 -1
- package/esm/simple/__tests__/Auth.test.js +64 -8
- package/esm/simple/index.js +8 -2
- package/esm/simple/room/Room.js +2 -1
- package/esm/simple/room/SipRoom.js +2 -1
- package/esm/state-machine/call-state-machine.js +114 -0
- package/esm/state-machine/softphone-state-machine.js +39 -0
- package/esm/state-machine/utils.js +40 -0
- package/esm/types/__tests__/__voice__/call.test.d.ts +2 -0
- package/esm/types/__tests__/__voice__/call.test.d.ts.map +1 -0
- package/esm/types/__tests__/__voice__/softphone.test.d.ts +2 -0
- package/esm/types/__tests__/__voice__/softphone.test.d.ts.map +1 -0
- package/esm/types/api/auth.d.ts.map +1 -1
- package/esm/types/checker/checks/webrtc.d.ts.map +1 -1
- package/esm/types/domain/InvalidState.d.ts +6 -0
- package/esm/types/domain/InvalidState.d.ts.map +1 -0
- package/esm/types/domain/InvalidStateTransition.d.ts +7 -0
- package/esm/types/domain/InvalidStateTransition.d.ts.map +1 -0
- package/esm/types/domain/Phone/WebRTCPhone.d.ts +0 -1
- package/esm/types/domain/Phone/WebRTCPhone.d.ts.map +1 -1
- package/esm/types/domain/types.d.ts +4 -0
- package/esm/types/domain/types.d.ts.map +1 -1
- package/esm/types/simple/Auth.d.ts.map +1 -1
- package/esm/types/simple/Phone.d.ts.map +1 -1
- package/esm/types/simple/index.d.ts +7 -2
- package/esm/types/simple/index.d.ts.map +1 -1
- package/esm/types/simple/room/Room.d.ts.map +1 -1
- package/esm/types/simple/room/SipRoom.d.ts +1 -1
- package/esm/types/simple/room/SipRoom.d.ts.map +1 -1
- package/esm/types/state-machine/call-state-machine.d.ts +43 -0
- package/esm/types/state-machine/call-state-machine.d.ts.map +1 -0
- package/esm/types/state-machine/softphone-state-machine.d.ts +19 -0
- package/esm/types/state-machine/softphone-state-machine.d.ts.map +1 -0
- package/esm/types/state-machine/utils.d.ts +9 -0
- package/esm/types/state-machine/utils.d.ts.map +1 -0
- package/esm/types/utils/api-requester.d.ts.map +1 -1
- package/esm/types/utils/sdp.d.ts +6 -0
- package/esm/types/utils/sdp.d.ts.map +1 -1
- package/esm/types/utils/sip-logger.d.ts +4 -0
- package/esm/types/utils/sip-logger.d.ts.map +1 -0
- package/esm/types/utils/webrtc.d.ts +40 -0
- package/esm/types/utils/webrtc.d.ts.map +1 -1
- package/esm/types/voice/call.d.ts +110 -0
- package/esm/types/voice/call.d.ts.map +1 -0
- package/esm/types/voice/softphone.d.ts +77 -0
- package/esm/types/voice/softphone.d.ts.map +1 -0
- package/esm/types/web-rtc-client.d.ts +1 -36
- package/esm/types/web-rtc-client.d.ts.map +1 -1
- package/esm/utils/__tests__/api-requester.test.js +0 -17
- package/esm/utils/api-requester.js +1 -9
- package/esm/utils/sdp.js +26 -0
- package/esm/utils/sip-logger.js +32 -0
- package/esm/utils/webrtc.js +465 -0
- package/esm/voice/call.js +527 -0
- package/esm/voice/softphone.js +347 -0
- package/esm/web-rtc-client.js +86 -584
- package/lib/__tests__/__voice__/call.test.js +86 -0
- package/lib/__tests__/__voice__/softphone.test.js +80 -0
- package/lib/api/auth.js +1 -5
- package/lib/checker/checks/webrtc.js +2 -1
- package/lib/domain/InvalidState.js +9 -0
- package/lib/domain/InvalidStateTransition.js +10 -0
- package/lib/domain/Phone/WebRTCPhone.js +33 -35
- package/lib/simple/Auth.js +12 -6
- package/lib/simple/Phone.js +2 -1
- package/lib/simple/__tests__/Auth.test.js +64 -11
- package/lib/simple/index.js +6 -0
- package/lib/simple/room/Room.js +4 -3
- package/lib/simple/room/SipRoom.js +2 -1
- package/lib/state-machine/call-state-machine.js +120 -0
- package/lib/state-machine/softphone-state-machine.js +42 -0
- package/lib/state-machine/utils.js +52 -0
- package/lib/types/__tests__/__voice__/call.test.d.ts +2 -0
- package/lib/types/__tests__/__voice__/call.test.d.ts.map +1 -0
- package/lib/types/__tests__/__voice__/softphone.test.d.ts +2 -0
- package/lib/types/__tests__/__voice__/softphone.test.d.ts.map +1 -0
- package/lib/types/api/auth.d.ts.map +1 -1
- package/lib/types/checker/checks/webrtc.d.ts.map +1 -1
- package/lib/types/domain/InvalidState.d.ts +6 -0
- package/lib/types/domain/InvalidState.d.ts.map +1 -0
- package/lib/types/domain/InvalidStateTransition.d.ts +7 -0
- package/lib/types/domain/InvalidStateTransition.d.ts.map +1 -0
- package/lib/types/domain/Phone/WebRTCPhone.d.ts +0 -1
- package/lib/types/domain/Phone/WebRTCPhone.d.ts.map +1 -1
- package/lib/types/domain/types.d.ts +4 -0
- package/lib/types/domain/types.d.ts.map +1 -1
- package/lib/types/simple/Auth.d.ts.map +1 -1
- package/lib/types/simple/Phone.d.ts.map +1 -1
- package/lib/types/simple/index.d.ts +7 -2
- package/lib/types/simple/index.d.ts.map +1 -1
- package/lib/types/simple/room/Room.d.ts.map +1 -1
- package/lib/types/simple/room/SipRoom.d.ts +1 -1
- package/lib/types/simple/room/SipRoom.d.ts.map +1 -1
- package/lib/types/state-machine/call-state-machine.d.ts +43 -0
- package/lib/types/state-machine/call-state-machine.d.ts.map +1 -0
- package/lib/types/state-machine/softphone-state-machine.d.ts +19 -0
- package/lib/types/state-machine/softphone-state-machine.d.ts.map +1 -0
- package/lib/types/state-machine/utils.d.ts +9 -0
- package/lib/types/state-machine/utils.d.ts.map +1 -0
- package/lib/types/utils/api-requester.d.ts.map +1 -1
- package/lib/types/utils/sdp.d.ts +6 -0
- package/lib/types/utils/sdp.d.ts.map +1 -1
- package/lib/types/utils/sip-logger.d.ts +4 -0
- package/lib/types/utils/sip-logger.d.ts.map +1 -0
- package/lib/types/utils/webrtc.d.ts +40 -0
- package/lib/types/utils/webrtc.d.ts.map +1 -1
- package/lib/types/voice/call.d.ts +110 -0
- package/lib/types/voice/call.d.ts.map +1 -0
- package/lib/types/voice/softphone.d.ts +77 -0
- package/lib/types/voice/softphone.d.ts.map +1 -0
- package/lib/types/web-rtc-client.d.ts +1 -36
- package/lib/types/web-rtc-client.d.ts.map +1 -1
- package/lib/utils/__tests__/api-requester.test.js +0 -17
- package/lib/utils/api-requester.js +1 -9
- package/lib/utils/sdp.js +27 -1
- package/lib/utils/sip-logger.js +37 -0
- package/lib/utils/webrtc.js +521 -1
- package/lib/voice/call.js +577 -0
- package/lib/voice/softphone.js +375 -0
- package/lib/web-rtc-client.js +89 -601
- package/package.json +3 -2
- package/src/__tests__/__voice__/call.test.ts +90 -0
- package/src/__tests__/__voice__/softphone.test.ts +82 -0
- package/src/api/auth.ts +2 -13
- package/src/checker/checks/webrtc.ts +2 -1
- package/src/domain/InvalidState.ts +13 -0
- package/src/domain/InvalidStateTransition.ts +17 -0
- package/src/domain/Phone/WebRTCPhone.ts +34 -36
- package/src/domain/types.ts +6 -1
- package/src/simple/Auth.ts +13 -6
- package/src/simple/Phone.ts +2 -1
- package/src/simple/__tests__/Auth.test.ts +87 -9
- package/src/simple/index.ts +9 -2
- package/src/simple/room/Room.ts +2 -1
- package/src/simple/room/SipRoom.ts +2 -1
- package/src/state-machine/call-state-machine.ts +131 -0
- package/src/state-machine/softphone-state-machine.ts +49 -0
- package/src/state-machine/utils.ts +53 -0
- package/src/utils/__tests__/api-requester.test.ts +0 -20
- package/src/utils/api-requester.ts +1 -9
- package/src/utils/sdp.ts +36 -0
- package/src/utils/sip-logger.ts +40 -0
- package/src/utils/webrtc.ts +571 -0
- package/src/voice/call.ts +713 -0
- package/src/voice/softphone.ts +485 -0
- package/src/web-rtc-client.ts +90 -690
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Object.defineProperty(exports, "__esModule", { value: true });
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exports.MESSAGE_TRACK_UPDATED = exports.MESSAGE_TYPE_SIGNAL = exports.MESSAGE_TYPE_CHAT = exports.EVENT_TERMINATE_SOUND = exports.EVENT_CAMERA_DISABLED = exports.EVENT_CAMERA_RESUMED = exports.EVENT_WEB_RTC_STATS = exports.EVENT_SHARE_SCREEN_ENDED = exports.EVENT_SHARE_SCREEN_STARTED = exports.EVENT_SHARE_SCREEN_ENDING = exports.EVENT_REMOVE_STREAM = exports.EVENT_VIDEO_STREAM = exports.EVENT_AUDIO_STREAM = exports.EVENT_TRACK = exports.EVENT_REINVITE = exports.EVENT_CALL_RESUMED = exports.EVENT_CALL_HELD = exports.EVENT_CALL_UN_MUTED = exports.EVENT_CALL_MUTED = exports.EVENT_CALL_ENDED = exports.EVENT_CALL_TERMINATING = exports.EVENT_CALL_REJECTED = exports.EVENT_CALL_ERROR = exports.EVENT_EARLY_MEDIA = exports.EVENT_PROGRESS = exports.EVENT_CALL_CANCELED = exports.EVENT_CALL_ACCEPTED = void 0;
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/* eslint-disable no-underscore-dangle */
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const events_1 = require("events");
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const xstate_1 = require("xstate");
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const invitation_1 = require("sip.js/lib/api/invitation");
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const session_state_1 = require("sip.js/lib/api/session-state");
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const call_state_machine_1 = __importStar(require("../state-machine/call-state-machine"));
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const IssueReporter_1 = __importDefault(require("../service/IssueReporter"));
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const sdp_1 = require("../utils/sdp");
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const utils_1 = require("../state-machine/utils");
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const webrtc_1 = require("../utils/webrtc");
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const logger = IssueReporter_1.default.loggerFor('call');
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// Events
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exports.EVENT_CALL_ACCEPTED = 'accepted';
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exports.EVENT_CALL_CANCELED = 'canceled';
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exports.EVENT_PROGRESS = 'progress';
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exports.EVENT_EARLY_MEDIA = 'earlyMedia';
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exports.EVENT_CALL_ERROR = 'error';
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exports.EVENT_CALL_REJECTED = 'rejected';
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exports.EVENT_CALL_TERMINATING = 'terminating';
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exports.EVENT_CALL_ENDED = 'ended';
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exports.EVENT_CALL_MUTED = 'muted';
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exports.EVENT_CALL_UN_MUTED = 'unMuted';
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exports.EVENT_CALL_HELD = 'held';
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exports.EVENT_CALL_RESUMED = 'resumed';
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exports.EVENT_REINVITE = 'reinvite';
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exports.EVENT_TRACK = 'track';
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exports.EVENT_AUDIO_STREAM = 'onAudioStream';
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exports.EVENT_VIDEO_STREAM = 'onVideoStream';
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exports.EVENT_REMOVE_STREAM = 'onRemoveStream';
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exports.EVENT_SHARE_SCREEN_ENDING = 'onScreenShareEnding';
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exports.EVENT_SHARE_SCREEN_STARTED = 'onScreenShareStarted';
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exports.EVENT_SHARE_SCREEN_ENDED = 'onScreenShareEnded';
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exports.EVENT_WEB_RTC_STATS = 'webRtcStats';
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exports.EVENT_CAMERA_RESUMED = 'onCameraResumed';
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exports.EVENT_CAMERA_DISABLED = 'onCameraDisabled';
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exports.EVENT_TERMINATE_SOUND = 'terminateSound';
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exports.MESSAGE_TYPE_CHAT = 'message/TYPE_CHAT';
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exports.MESSAGE_TYPE_SIGNAL = 'message/TYPE_SIGNAL';
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exports.MESSAGE_TRACK_UPDATED = 'onTrackUpdated';
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class Call extends events_1.EventEmitter {
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constructor(sipCall, phone) {
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super();
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this.shouldSendReinvite = false;
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this.callActor = (0, xstate_1.createActor)(call_state_machine_1.default);
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this.callActor.start();
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this.creationTime = new Date();
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this.phone = phone;
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this.sipCall = sipCall;
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this._bindEvents();
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}
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accept() {
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return __awaiter(this, arguments, void 0, function* (options = {}) {
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(0, utils_1.assertCan)(this.callActor, call_state_machine_1.Actions.ACCEPT);
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logger.info('call - accept', Object.assign({ id: this.id }, options));
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this._sendAction(call_state_machine_1.Actions.ACCEPT);
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this.phone.holdAllCalls(this);
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this.shouldSendReinvite = false;
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this.answerTime = new Date();
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return this.phone.client.answer(this.sipCall, options.withCamera).then(() => {
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logger.info('call - accepted', Object.assign({ id: this.id }, options));
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return this.id;
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}).catch(e => {
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logger.error(e);
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this.hangup();
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throw e;
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});
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});
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}
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reject() {
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return __awaiter(this, void 0, void 0, function* () {
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(0, utils_1.assertCan)(this.callActor, call_state_machine_1.Actions.REJECT);
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logger.info('call - reject', { id: this.id });
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this.eventEmitter.emit(exports.EVENT_TERMINATE_SOUND, 'call rejected locally');
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this.shouldSendReinvite = false;
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return this.hangup();
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});
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}
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hangup() {
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return __awaiter(this, void 0, void 0, function* () {
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(0, utils_1.assertCan)(this.callActor, call_state_machine_1.Actions.HANGUP);
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logger.info('call - hangup', { id: this.id });
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yield this.phone.client.hangup(this.sipCall);
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// Remove this call from softphone's calls
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this.phone.removeCall(this);
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return true;
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});
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}
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forceCancel() {
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try {
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this.sipCall.outgoingInviteRequest.cancel();
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}
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catch (e) {
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logger.info('call - force cancel, error', e);
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}
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}
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startConference(withCalls) {
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// @TODO
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// const participants = [this, ...withCalls].reduce((acc: Record<string, any>, call: Call) => {
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// acc[call.talkingToIds[0]] = call;
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// return acc;
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// }, {});
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// const adHocConference = new AdHocAPIConference({
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// phone: this.phone,
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// host,
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// participants,
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// });
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// return adHocConference.start();
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}
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mute() {
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(0, utils_1.assertCan)(this.callActor, call_state_machine_1.EstablishedActions.MUTE);
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logger.info('call - mute', { id: this.id });
|
|
151
|
+
this._sendAction(call_state_machine_1.EstablishedActions.MUTE);
|
|
152
|
+
this.phone.client.mute(this.sipCall);
|
|
153
|
+
this.eventEmitter.emit(exports.EVENT_CALL_MUTED);
|
|
154
|
+
}
|
|
155
|
+
unMute() {
|
|
156
|
+
(0, utils_1.assertCan)(this.callActor, call_state_machine_1.EstablishedActions.UN_MUTE);
|
|
157
|
+
logger.info('call - un-mute', { id: this.id });
|
|
158
|
+
this._sendAction(call_state_machine_1.EstablishedActions.UN_MUTE);
|
|
159
|
+
this.phone.client.unmute(this.sipCall);
|
|
160
|
+
this.eventEmitter.emit(exports.EVENT_CALL_UN_MUTED);
|
|
161
|
+
}
|
|
162
|
+
hold() {
|
|
163
|
+
return __awaiter(this, void 0, void 0, function* () {
|
|
164
|
+
(0, utils_1.assertCan)(this.callActor, call_state_machine_1.EstablishedActions.HOLD);
|
|
165
|
+
logger.info('call - hold', { id: this.id });
|
|
166
|
+
this._sendAction(call_state_machine_1.EstablishedActions.HOLD);
|
|
167
|
+
const hasVideo = (0, webrtc_1.hasLocalVideo)(this.sipCall);
|
|
168
|
+
// Stop screenshare if needed
|
|
169
|
+
if (this.currentScreenShare && this.currentScreenShare.sipSessionId === this.id) {
|
|
170
|
+
yield this.stopScreenSharing(false);
|
|
171
|
+
}
|
|
172
|
+
// Downgrade to audio if needed
|
|
173
|
+
if (hasVideo) {
|
|
174
|
+
yield (0, webrtc_1.downgradeToAudio)(this.sipCall);
|
|
175
|
+
}
|
|
176
|
+
const promise = this.phone.client.hold(this.sipCall, this.isConference(), hasVideo);
|
|
177
|
+
this.eventEmitter.emit(exports.EVENT_CALL_HELD);
|
|
178
|
+
return promise;
|
|
179
|
+
});
|
|
180
|
+
}
|
|
181
|
+
resume() {
|
|
182
|
+
return __awaiter(this, void 0, void 0, function* () {
|
|
183
|
+
(0, utils_1.assertCan)(this.callActor, call_state_machine_1.EstablishedActions.RESUME);
|
|
184
|
+
this._sendAction(call_state_machine_1.EstablishedActions.RESUME);
|
|
185
|
+
const isConference = this.isConference();
|
|
186
|
+
const hasVideo = (0, webrtc_1.hasLocalVideo)(this.sipCall);
|
|
187
|
+
const wasScreensharing = this.currentScreenShare && this.currentScreenShare.sipSessionId === this.id;
|
|
188
|
+
const wasDesktop = this.currentScreenShare && this.currentScreenShare.desktop;
|
|
189
|
+
logger.info('call - resume', { id: this.id, isConference, hasVideo, wasScreensharing, wasDesktop });
|
|
190
|
+
const promise = this.phone.client.unhold(this.sipCall, isConference);
|
|
191
|
+
if (hasVideo) {
|
|
192
|
+
const constraints = {
|
|
193
|
+
audio: false,
|
|
194
|
+
video: true,
|
|
195
|
+
screen: wasScreensharing,
|
|
196
|
+
desktop: wasDesktop,
|
|
197
|
+
};
|
|
198
|
+
yield (0, webrtc_1.upgradeToVideo)(this.sipCall, constraints, isConference);
|
|
199
|
+
}
|
|
200
|
+
const onScreenSharing = (stream) => {
|
|
201
|
+
const hadVideo = this.currentScreenShare && this.currentScreenShare.hadVideo;
|
|
202
|
+
if (stream) {
|
|
203
|
+
this._onScreenSharing(stream, hadVideo, wasDesktop);
|
|
204
|
+
}
|
|
205
|
+
};
|
|
206
|
+
this.eventEmitter.emit(exports.EVENT_CALL_RESUMED);
|
|
207
|
+
return promise.then(() => {
|
|
208
|
+
const stream = this.getLocalVideoStream();
|
|
209
|
+
if (wasScreensharing) {
|
|
210
|
+
onScreenSharing(this.getLocalVideoStream());
|
|
211
|
+
}
|
|
212
|
+
return stream;
|
|
213
|
+
});
|
|
214
|
+
});
|
|
215
|
+
}
|
|
216
|
+
turnCameraOn() {
|
|
217
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
218
|
+
logger.info('call - turn camera on', { id: this.id });
|
|
219
|
+
(0, webrtc_1.toggleVideo)(this.sipCall, true);
|
|
220
|
+
this.eventEmitter.emit(exports.EVENT_CAMERA_RESUMED);
|
|
221
|
+
}
|
|
222
|
+
turnCameraOff() {
|
|
223
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
224
|
+
logger.info('call - turn camera off', { id: this.id });
|
|
225
|
+
(0, webrtc_1.toggleVideo)(this.sipCall, false);
|
|
226
|
+
this.eventEmitter.emit(exports.EVENT_CAMERA_DISABLED);
|
|
227
|
+
}
|
|
228
|
+
isHeld() {
|
|
229
|
+
return (0, utils_1.getState)(this.callActor) === call_state_machine_1.EstablishedStates.HELD;
|
|
230
|
+
}
|
|
231
|
+
isEstablished() {
|
|
232
|
+
return (0, utils_1.getState)(this.callActor) === call_state_machine_1.States.ESTABLISHED;
|
|
233
|
+
}
|
|
234
|
+
isConference() {
|
|
235
|
+
var _a;
|
|
236
|
+
return (_a = (0, webrtc_1.getPeerConnection)(this.sipCall)) === null || _a === void 0 ? void 0 : _a.sfu;
|
|
237
|
+
}
|
|
238
|
+
transfer(target) {
|
|
239
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
240
|
+
logger.info('call - transfer', { id: this.id, target });
|
|
241
|
+
this.phone.client.transfer(this.sipCall, target);
|
|
242
|
+
}
|
|
243
|
+
atxfer() {
|
|
244
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
245
|
+
logger.info('call - atxfer', { id: this.id });
|
|
246
|
+
return this.phone.client.atxfer(this.sipCall);
|
|
247
|
+
}
|
|
248
|
+
sendDigits(digits) {
|
|
249
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
250
|
+
logger.info('call - send digits', { id: this.id, digits });
|
|
251
|
+
this.phone.client.sendDTMF(this.sipCall, digits);
|
|
252
|
+
}
|
|
253
|
+
startNetworkMonitoring(interval = 1000) {
|
|
254
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
255
|
+
logger.info('call - start network monitoring', { id: this.id });
|
|
256
|
+
return this.phone.client.startNetworkMonitoring(this.sipCall, interval);
|
|
257
|
+
}
|
|
258
|
+
stopNetworkMonitoring() {
|
|
259
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
260
|
+
logger.info('call - stop network monitoring', { id: this.id });
|
|
261
|
+
return this.phone.client.stopNetworkMonitoring(this.sipCall);
|
|
262
|
+
}
|
|
263
|
+
getStats() {
|
|
264
|
+
return __awaiter(this, void 0, void 0, function* () {
|
|
265
|
+
return (0, webrtc_1.getWebRtcStats)(this.sipCall);
|
|
266
|
+
});
|
|
267
|
+
}
|
|
268
|
+
sendReinvite() {
|
|
269
|
+
return __awaiter(this, arguments, void 0, function* (options = {}) {
|
|
270
|
+
var _a, _b;
|
|
271
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
272
|
+
logger.info('call - send reinvite', Object.assign({ id: this.id }, options));
|
|
273
|
+
const shouldScreenShare = (_a = options.constraints) === null || _a === void 0 ? void 0 : _a.screen;
|
|
274
|
+
const isUpgrade = shouldScreenShare || ((_b = options.constraints) === null || _b === void 0 ? void 0 : _b.video);
|
|
275
|
+
// Downgrade
|
|
276
|
+
if (options.constraints && !isUpgrade) {
|
|
277
|
+
// No reinvite needed
|
|
278
|
+
this._downgradeToAudio();
|
|
279
|
+
return;
|
|
280
|
+
}
|
|
281
|
+
if (isUpgrade) {
|
|
282
|
+
const shouldReinvite = yield this._upgradeToVideo(options);
|
|
283
|
+
if (!shouldReinvite) {
|
|
284
|
+
return;
|
|
285
|
+
}
|
|
286
|
+
}
|
|
287
|
+
return this.phone.client.reinvite(this.sipCall, options.constraints, options.conference, options.audioOnly, options.iceRestart);
|
|
288
|
+
});
|
|
289
|
+
}
|
|
290
|
+
sendMessage(body, contentType = 'text/plain') {
|
|
291
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
292
|
+
logger.info('call - send message', { id: this.id, body });
|
|
293
|
+
return this.phone.client.sendMessage(this.sipCall, body, contentType);
|
|
294
|
+
}
|
|
295
|
+
sendChat(content) {
|
|
296
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
297
|
+
logger.info('call - send chat', { id: this.id, content });
|
|
298
|
+
this.sendMessage(JSON.stringify({ type: exports.MESSAGE_TYPE_CHAT, content }), 'application/json');
|
|
299
|
+
}
|
|
300
|
+
sendSignal(content) {
|
|
301
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
302
|
+
logger.info('call - send signal', { id: this.id, content });
|
|
303
|
+
this.sendMessage(JSON.stringify({ type: exports.MESSAGE_TYPE_SIGNAL, content }), 'application/json');
|
|
304
|
+
}
|
|
305
|
+
startScreenSharing(constraints) {
|
|
306
|
+
return __awaiter(this, void 0, void 0, function* () {
|
|
307
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
308
|
+
logger.info('call - start screen sharing', { constraints, id: this.id });
|
|
309
|
+
const screenShareStream = yield (0, webrtc_1.getStreamFromConstraints)(constraints);
|
|
310
|
+
if (!screenShareStream) {
|
|
311
|
+
throw new Error(`Can't create media stream for screensharing with: ${JSON.stringify(constraints)}`);
|
|
312
|
+
}
|
|
313
|
+
const screenTrack = screenShareStream.getVideoTracks()[0];
|
|
314
|
+
const pc = (0, webrtc_1.getPeerConnection)(this.sipCall);
|
|
315
|
+
const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
|
|
316
|
+
const localStream = (0, webrtc_1.getLocalStream)(this.sipCall);
|
|
317
|
+
const videoTrack = localStream ? localStream.getTracks().find(track => track.kind === 'video') : null;
|
|
318
|
+
const hadVideo = !!videoTrack;
|
|
319
|
+
// Stop local video tracks
|
|
320
|
+
if (videoTrack && localStream) {
|
|
321
|
+
videoTrack.enabled = false;
|
|
322
|
+
videoTrack.stop();
|
|
323
|
+
localStream.removeTrack(videoTrack);
|
|
324
|
+
}
|
|
325
|
+
if (localStream) {
|
|
326
|
+
localStream.addTrack(screenTrack);
|
|
327
|
+
}
|
|
328
|
+
if (sender) {
|
|
329
|
+
sender.replaceTrack(screenTrack);
|
|
330
|
+
}
|
|
331
|
+
this._onScreenSharing(screenShareStream, hadVideo, constraints.desktop);
|
|
332
|
+
return screenShareStream;
|
|
333
|
+
});
|
|
334
|
+
}
|
|
335
|
+
stopScreenSharing() {
|
|
336
|
+
return __awaiter(this, arguments, void 0, function* (restoreLocalStream = true) {
|
|
337
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
338
|
+
if (!this.currentScreenShare) {
|
|
339
|
+
return;
|
|
340
|
+
}
|
|
341
|
+
logger.info('call - stop screen sharing', { restoreLocalStream });
|
|
342
|
+
let reinvited = null;
|
|
343
|
+
try {
|
|
344
|
+
if (this.currentScreenShare.stream) {
|
|
345
|
+
this.currentScreenShare.stream.getTracks()
|
|
346
|
+
.filter((track) => track.enabled && track.kind === 'video')
|
|
347
|
+
.forEach((track) => track.stop());
|
|
348
|
+
}
|
|
349
|
+
if (restoreLocalStream) {
|
|
350
|
+
const conference = this.isConference();
|
|
351
|
+
// When stopping screenshare and we had video before that, we have to re-upgrade
|
|
352
|
+
// When upgrading directly to screenshare (eg: we don't have a videoLocalStream to replace)
|
|
353
|
+
// We have to downgrade to audio.
|
|
354
|
+
const screenshareStopped = this.currentScreenShare.hadVideo;
|
|
355
|
+
const constraints = {
|
|
356
|
+
audio: false,
|
|
357
|
+
video: screenshareStopped,
|
|
358
|
+
};
|
|
359
|
+
// We have to remove the video sender to be able to re-upgrade to video in `sendReinvite` below
|
|
360
|
+
this._downgradeToAudio(false);
|
|
361
|
+
// Wait for the track to be removed, unless the video sender won't have a null track in `_upgradeToVideo`.
|
|
362
|
+
yield new Promise(resolve => setTimeout(resolve, 500));
|
|
363
|
+
reinvited = yield this.sendReinvite({ constraints, conference });
|
|
364
|
+
}
|
|
365
|
+
}
|
|
366
|
+
catch (e) {
|
|
367
|
+
console.warn(e);
|
|
368
|
+
}
|
|
369
|
+
this.eventEmitter.emit(exports.EVENT_SHARE_SCREEN_ENDED);
|
|
370
|
+
this.currentScreenShare = null;
|
|
371
|
+
return reinvited;
|
|
372
|
+
});
|
|
373
|
+
}
|
|
374
|
+
onScreenShareReinvite(response, desktop) {
|
|
375
|
+
const localStream = (0, webrtc_1.getLocalStream)(this.sipCall);
|
|
376
|
+
logger.info('call - updrading directly in screensharing mode', { id: this.id, tracks: localStream === null || localStream === void 0 ? void 0 : localStream.getTracks() });
|
|
377
|
+
this._onScreenSharing(localStream, false, desktop);
|
|
378
|
+
}
|
|
379
|
+
onAccepted() {
|
|
380
|
+
this._sendAction(call_state_machine_1.Actions.REMOTLY_ACCEPTED);
|
|
381
|
+
this.emit(exports.EVENT_CALL_ACCEPTED);
|
|
382
|
+
}
|
|
383
|
+
onProgress() {
|
|
384
|
+
this.emit(exports.EVENT_PROGRESS);
|
|
385
|
+
}
|
|
386
|
+
onEarlyMedia() {
|
|
387
|
+
this.emit(exports.EVENT_EARLY_MEDIA);
|
|
388
|
+
}
|
|
389
|
+
onTrack(event) {
|
|
390
|
+
this.emit(exports.EVENT_TRACK, event);
|
|
391
|
+
}
|
|
392
|
+
onError(error) {
|
|
393
|
+
this.emit(exports.EVENT_CALL_ERROR, error);
|
|
394
|
+
}
|
|
395
|
+
onRejected() {
|
|
396
|
+
this.endTime = new Date();
|
|
397
|
+
this.emit(exports.EVENT_CALL_REJECTED);
|
|
398
|
+
this.phone.removeCall(this);
|
|
399
|
+
}
|
|
400
|
+
onNetworkStats(stats, previousStats) {
|
|
401
|
+
this.emit(exports.EVENT_WEB_RTC_STATS, stats, previousStats);
|
|
402
|
+
}
|
|
403
|
+
getRemoteStream() {
|
|
404
|
+
return (0, webrtc_1.getRemoteStream)(this.sipCall);
|
|
405
|
+
}
|
|
406
|
+
getRemoteVideoStream() {
|
|
407
|
+
return (0, webrtc_1.getRemoteVideoStream)(this.sipCall);
|
|
408
|
+
}
|
|
409
|
+
hasRemoteVideo() {
|
|
410
|
+
return (0, webrtc_1.hasRemoteVideo)(this.sipCall);
|
|
411
|
+
}
|
|
412
|
+
hasLocalVideo() {
|
|
413
|
+
return (0, webrtc_1.hasLocalVideo)(this.sipCall);
|
|
414
|
+
}
|
|
415
|
+
getLocalStream() {
|
|
416
|
+
return (0, webrtc_1.getLocalStream)(this.sipCall);
|
|
417
|
+
}
|
|
418
|
+
getLocalVideoStream() {
|
|
419
|
+
return (0, webrtc_1.getLocalVideoStream)(this.sipCall);
|
|
420
|
+
}
|
|
421
|
+
hasALocalVideoTrack() {
|
|
422
|
+
return (0, webrtc_1.hasALocalVideoTrack)(this.sipCall);
|
|
423
|
+
}
|
|
424
|
+
isVideoRemotelyHeld() {
|
|
425
|
+
return (0, webrtc_1.isVideoRemotelyHeld)(this.sipCall);
|
|
426
|
+
}
|
|
427
|
+
onReinvite(request, updatedCalleeName, updatedNumber, hadRemoteVideo) {
|
|
428
|
+
this.emit(exports.EVENT_REINVITE, request, updatedCalleeName, updatedNumber, hadRemoteVideo);
|
|
429
|
+
}
|
|
430
|
+
get id() {
|
|
431
|
+
return (0, sdp_1.getSipSessionId)(this.sipCall);
|
|
432
|
+
}
|
|
433
|
+
get number() {
|
|
434
|
+
const identity = this.sipCall.remoteIdentity || this.sipCall.assertedIdentity;
|
|
435
|
+
// @ts-ignore: private
|
|
436
|
+
return identity ? identity.uri._normal.user : null;
|
|
437
|
+
}
|
|
438
|
+
_bindEvents() {
|
|
439
|
+
if (this.sipCall instanceof invitation_1.Invitation) {
|
|
440
|
+
// Monkey patch to know when canceled with the CANCEL message
|
|
441
|
+
const onCancel = this.sipCall._onCancel.bind(this.sipCall);
|
|
442
|
+
this.sipCall._onCancel = (message) => {
|
|
443
|
+
logger.trace('call - on sip session canceled', { callId: message.callId, id: this.id });
|
|
444
|
+
onCancel(message);
|
|
445
|
+
const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
|
|
446
|
+
this.eventEmitter.emit(exports.EVENT_CALL_CANCELED, elsewhere);
|
|
447
|
+
};
|
|
448
|
+
}
|
|
449
|
+
this.sipCall.stateChange.addListener((newState) => {
|
|
450
|
+
switch (newState) {
|
|
451
|
+
case session_state_1.SessionState.Terminating:
|
|
452
|
+
logger.info('call - terminating', { id: this.id });
|
|
453
|
+
this.eventEmitter.emit(exports.EVENT_CALL_TERMINATING);
|
|
454
|
+
break;
|
|
455
|
+
case session_state_1.SessionState.Terminated:
|
|
456
|
+
{
|
|
457
|
+
logger.info('call - terminated', { id: this.id });
|
|
458
|
+
// Should be called before `_onCallTerminated` or the callCount will not decrement...
|
|
459
|
+
this.endTime = new Date();
|
|
460
|
+
return this.eventEmitter.emit(exports.EVENT_CALL_ENDED);
|
|
461
|
+
}
|
|
462
|
+
default:
|
|
463
|
+
break;
|
|
464
|
+
}
|
|
465
|
+
});
|
|
466
|
+
if (!this.sipCall.sessionDescriptionHandler) {
|
|
467
|
+
return;
|
|
468
|
+
}
|
|
469
|
+
// Video events
|
|
470
|
+
const sdh = this.sipCall.sessionDescriptionHandler;
|
|
471
|
+
const peerConnection = sdh.peerConnection;
|
|
472
|
+
if (!peerConnection) {
|
|
473
|
+
logger.warn('call - no peer connection', { id: this.id });
|
|
474
|
+
return;
|
|
475
|
+
}
|
|
476
|
+
peerConnection.ontrack = (rawEvent) => {
|
|
477
|
+
const event = rawEvent;
|
|
478
|
+
const [stream] = event.streams;
|
|
479
|
+
const kind = event && event.track && event.track.kind;
|
|
480
|
+
logger.info('call - on track stream called on the peer connection', {
|
|
481
|
+
id: this.id,
|
|
482
|
+
streamId: stream ? stream.id : null,
|
|
483
|
+
tracks: stream ? stream.getTracks() : null,
|
|
484
|
+
kind,
|
|
485
|
+
});
|
|
486
|
+
if (kind === 'audio') {
|
|
487
|
+
return this.eventEmitter.emit(exports.EVENT_AUDIO_STREAM, stream);
|
|
488
|
+
}
|
|
489
|
+
return this.eventEmitter.emit(exports.EVENT_VIDEO_STREAM, stream, event.track.id, event);
|
|
490
|
+
};
|
|
491
|
+
peerConnection.onremovestream = (event) => {
|
|
492
|
+
logger.info('call - on remove stream called on the peer connection', {
|
|
493
|
+
id: event.stream.id,
|
|
494
|
+
tracks: event.stream.getTracks(),
|
|
495
|
+
});
|
|
496
|
+
this.eventEmitter.emit(exports.EVENT_REMOVE_STREAM, event.stream);
|
|
497
|
+
};
|
|
498
|
+
// Client events
|
|
499
|
+
}
|
|
500
|
+
_downgradeToAudio(withMessage = true) {
|
|
501
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
502
|
+
logger.info('call - downgrade to audio', { id: this.id, withMessage });
|
|
503
|
+
(0, webrtc_1.downgradeToAudio)(this.sipCall);
|
|
504
|
+
if (withMessage) {
|
|
505
|
+
this._sendReinviteMessage(false);
|
|
506
|
+
}
|
|
507
|
+
return true;
|
|
508
|
+
}
|
|
509
|
+
// Returns true if we need to send a re-INVITE request
|
|
510
|
+
_upgradeToVideo() {
|
|
511
|
+
return __awaiter(this, arguments, void 0, function* (options = {}) {
|
|
512
|
+
var _a, _b;
|
|
513
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
514
|
+
logger.info('call - upgrade to video', Object.assign({ id: this.id }, options));
|
|
515
|
+
const shouldScreenShare = (_a = options.constraints) === null || _a === void 0 ? void 0 : _a.screen;
|
|
516
|
+
const desktop = (_b = options.constraints) === null || _b === void 0 ? void 0 : _b.desktop;
|
|
517
|
+
const sdhOptions = this.sipCall.sessionDescriptionHandlerOptionsReInvite;
|
|
518
|
+
const wasAudioOnly = sdhOptions === null || sdhOptions === void 0 ? void 0 : sdhOptions.audioOnly;
|
|
519
|
+
const newStream = yield (0, webrtc_1.upgradeToVideo)(this.sipCall, options.constraints || {}, !!options.conference);
|
|
520
|
+
// If no stream is returned, it means we have to reinvite
|
|
521
|
+
if (!newStream) {
|
|
522
|
+
return true;
|
|
523
|
+
}
|
|
524
|
+
this._sendReinviteMessage(true);
|
|
525
|
+
if (shouldScreenShare) {
|
|
526
|
+
this._onScreenSharing(newStream, false, desktop);
|
|
527
|
+
}
|
|
528
|
+
// We have to reinvite to change the direction on the bundle when upgrading from an audioOnly conference
|
|
529
|
+
if (options.conference && wasAudioOnly) {
|
|
530
|
+
return true;
|
|
531
|
+
}
|
|
532
|
+
// No reinvite needed here
|
|
533
|
+
return false;
|
|
534
|
+
});
|
|
535
|
+
}
|
|
536
|
+
_sendReinviteMessage(isUpgrade) {
|
|
537
|
+
(0, utils_1.assertState)(this.callActor, call_state_machine_1.States.ESTABLISHED);
|
|
538
|
+
logger.info('call - sending reinvite message', { id: this.id, isUpgrade });
|
|
539
|
+
// Have to send the message after a delay due to latency to update the remote peer
|
|
540
|
+
setTimeout(() => {
|
|
541
|
+
this.sendMessage(JSON.stringify({
|
|
542
|
+
type: exports.MESSAGE_TYPE_SIGNAL,
|
|
543
|
+
content: {
|
|
544
|
+
type: exports.MESSAGE_TRACK_UPDATED,
|
|
545
|
+
update: isUpgrade ? 'upgrade' : 'downgrade',
|
|
546
|
+
sipCallId: this.id,
|
|
547
|
+
callId: this.apiId,
|
|
548
|
+
number: this.number,
|
|
549
|
+
},
|
|
550
|
+
}));
|
|
551
|
+
}, 2500);
|
|
552
|
+
}
|
|
553
|
+
_onScreenSharing(screenStream, hadVideo, desktop) {
|
|
554
|
+
const screenTrack = screenStream.getVideoTracks()[0];
|
|
555
|
+
const pc = this.phone.client.getPeerConnection(this.id);
|
|
556
|
+
const sender = pc && pc.getSenders && pc.getSenders().find((s) => s && s.track && s.track.kind === 'video');
|
|
557
|
+
logger.info('call - on screensharing', { hadVideo, id: this.id, screenTrack });
|
|
558
|
+
if (screenTrack) {
|
|
559
|
+
screenTrack.onended = () => {
|
|
560
|
+
logger.info('call - on screenshare ended', { hadVideo, id: this.id, screenTrack });
|
|
561
|
+
this.eventEmitter.emit(exports.EVENT_SHARE_SCREEN_ENDING);
|
|
562
|
+
};
|
|
563
|
+
}
|
|
564
|
+
this.currentScreenShare = {
|
|
565
|
+
stream: screenStream,
|
|
566
|
+
hadVideo,
|
|
567
|
+
sender,
|
|
568
|
+
desktop,
|
|
569
|
+
};
|
|
570
|
+
(0, webrtc_1.setLocalMediaStream)(this.sipCall, screenStream);
|
|
571
|
+
this.eventEmitter.emit(exports.EVENT_SHARE_SCREEN_STARTED, screenStream);
|
|
572
|
+
}
|
|
573
|
+
_sendAction(action) {
|
|
574
|
+
this.callActor.send({ type: action });
|
|
575
|
+
}
|
|
576
|
+
}
|
|
577
|
+
exports.default = Call;
|