@wazo/sdk 1.0.0-alpha.1 → 1.0.0-alpha.2
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 +6 -6
- package/esm/__tests__/__voice__/call.test.js +44 -2
- package/esm/domain/Phone/WebRTCPhone.js +6 -7
- package/esm/simple/Phone.js +1 -0
- package/esm/state-machine/call-state-machine.js +1 -1
- package/esm/types/domain/Phone/WebRTCPhone.d.ts.map +1 -1
- package/esm/types/simple/Phone.d.ts.map +1 -1
- package/esm/types/state-machine/utils.d.ts +1 -1
- package/esm/types/state-machine/utils.d.ts.map +1 -1
- package/esm/types/utils/sdp.d.ts +2 -0
- package/esm/types/utils/sdp.d.ts.map +1 -1
- package/esm/types/utils/webrtc.d.ts +1 -0
- package/esm/types/utils/webrtc.d.ts.map +1 -1
- package/esm/types/voice/call.d.ts +19 -11
- package/esm/types/voice/call.d.ts.map +1 -1
- package/esm/types/voice/softphone.d.ts +18 -13
- package/esm/types/voice/softphone.d.ts.map +1 -1
- package/esm/utils/sdp.js +13 -0
- package/esm/utils/webrtc.js +20 -0
- package/esm/voice/call.js +90 -30
- package/esm/voice/softphone.js +94 -23
- package/lib/__tests__/__voice__/call.test.js +43 -1
- package/lib/domain/Phone/WebRTCPhone.js +5 -6
- package/lib/simple/Phone.js +1 -0
- package/lib/state-machine/call-state-machine.js +1 -1
- package/lib/types/domain/Phone/WebRTCPhone.d.ts.map +1 -1
- package/lib/types/simple/Phone.d.ts.map +1 -1
- package/lib/types/state-machine/utils.d.ts +1 -1
- package/lib/types/state-machine/utils.d.ts.map +1 -1
- package/lib/types/utils/sdp.d.ts +2 -0
- package/lib/types/utils/sdp.d.ts.map +1 -1
- package/lib/types/utils/webrtc.d.ts +1 -0
- package/lib/types/utils/webrtc.d.ts.map +1 -1
- package/lib/types/voice/call.d.ts +19 -11
- package/lib/types/voice/call.d.ts.map +1 -1
- package/lib/types/voice/softphone.d.ts +18 -13
- package/lib/types/voice/softphone.d.ts.map +1 -1
- package/lib/utils/sdp.js +16 -1
- package/lib/utils/webrtc.js +22 -1
- package/lib/voice/call.js +87 -28
- package/lib/voice/softphone.js +95 -23
- package/package.json +1 -1
- package/src/__tests__/__voice__/call.test.ts +47 -2
- package/src/domain/Phone/WebRTCPhone.ts +9 -9
- package/src/simple/Phone.ts +1 -0
- package/src/state-machine/call-state-machine.ts +1 -1
- package/src/state-machine/utils.ts +2 -2
- package/src/utils/sdp.ts +16 -0
- package/src/utils/webrtc.ts +23 -1
- package/src/voice/call.ts +118 -40
- package/src/voice/softphone.ts +116 -30
- package/src/web-rtc-client.ts +1 -1
package/src/voice/softphone.ts
CHANGED
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@@ -10,7 +10,6 @@ import InvalidState from '../domain/InvalidState';
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import Wazo from '..';
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import softphoneStateMachine, { type SoftphoneActorRef, Actions, type StateTypes, States, ActionTypes } from '../state-machine/softphone-state-machine';
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-
import { Actions as CallActions } from '../state-machine/call-state-machine';
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import { WebRtcConfig, SipCall } from '../domain/types';
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import WazoWebRTCClient from '../web-rtc-client';
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import IssueReporter from '../service/IssueReporter';
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@@ -19,12 +18,17 @@ import Call from './call';
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import { assertCan, can, getState, waitUntilState } from '../state-machine/utils';
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export const EVENT_INCOMING = 'incoming';
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export const EVENT_OUTGOING = 'outgoing';
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export const EVENT_HEARTBEAT = 'heartbeat';
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export const EVENT_HEARTBEAT_TIMEOUT = 'heartbeatTimeout';
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export const EVENT_REGISTERED = 'registered';
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export const EVENT_UNREGISTERED = 'unregistered';
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export const EVENT_DISCONNECTED = 'disconnected';
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export const EVENT_VIDEO_INPUT_CHANGE = 'videoInputChange';
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export const EVENT_TERMINATE_SOUND = 'terminateSound';
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export const EVENT_PLAY_HANGUP_SOUND = 'playHangupSound';
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export const EVENT_PLAY_RING_SOUND = 'playRingSound';
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export const EVENT_PLAY_INBOUND_CALL_SOUND = 'playInboundCallSound';
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export const EVENT_ON_MESSAGE = 'message';
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export const EVENT_ON_CHAT = 'phone/ON_CHAT';
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export const EVENT_ON_SIGNAL = 'phone/ON_SIGNAL';
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@@ -53,8 +57,6 @@ export class Softphone extends EventEmitter {
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audioRingVolume: number;
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eventEmitter: EventEmitter;
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-
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calls: Call[];
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client: WazoWebRTCClient;
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@@ -69,6 +71,8 @@ export class Softphone extends EventEmitter {
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shouldSendReinvite: boolean;
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ringingEnabled = true;
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constructor() {
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super();
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@@ -179,16 +183,16 @@ export class Softphone extends EventEmitter {
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return Promise.resolve(null);
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}
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if (this.
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throw new InvalidState(`Invalid state ${States.UNREGISTERED}`, States.UNREGISTERED);
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}
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if (this.getState() !== States.REGISTERING) {
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if (this.state === States.REGISTERING) {
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logger.info('softphone - calling when registering, waiting for Softphone to be registered ...');
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await waitUntilState(this.softphoneActor, States.REGISTERED);
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}
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if (this.state !== States.UNREGISTERED) {
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throw new InvalidState(`Invalid state ${States.UNREGISTERED}`, States.UNREGISTERED);
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}
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// Hold other calls
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this.holdAllCalls();
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@@ -198,6 +202,8 @@ export class Softphone extends EventEmitter {
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sipCall = this.client.call(options.params.To, options.withCamera, options.audioOnly, options.conference, options) as SipCall;
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const call = new Call(sipCall, this);
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this.emit(EVENT_OUTGOING, call);
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return call;
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} catch (error: any) {
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logger.warn('softphone - call, error', { message: error.message, stack: error.stack });
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@@ -209,6 +215,69 @@ export class Softphone extends EventEmitter {
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this.calls = this.calls.filter((call: Call) => call.id !== callToRemove.id);
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}
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onCallTerminated(call: Call) {
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logger.info('softphone - on call terminated', { id: call.id });
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const currentCall = this.getCurrentCall();
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const isCurrentCall = currentCall?.id === call.id;
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const isRinging = call.isRinging();
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const isCurrentIncomingCall = isCurrentCall && isRinging;
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const incomingCalls = this.getIncomingCalls();
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const nbIncomingCalls = incomingCalls.length;
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// Send an event to terminate ringing sound
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if (isCurrentIncomingCall) {
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setTimeout(() => {
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// Avoid race condition when the other is calling and hanging up immediately
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this.emit(EVENT_TERMINATE_SOUND, call, 'call terminated');
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}, 5);
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}
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// If the terminated call was an incoming call, we have to re-trigger if it's was the first incoming call
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// Otherwise, we have to re-trigger an incoming call event if another call is incoming
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const shouldRetrigger = isCurrentCall ? nbIncomingCalls : isCurrentIncomingCall;
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// Re-trigger incoming call event for remaining incoming calls
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logger.info('Softphone - check to re-trigger incoming call', { shouldRetrigger, nbIncomingCalls });
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if (shouldRetrigger && nbIncomingCalls > 1) {
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const nextCall = incomingCalls.find(c => c.id !== call.id);
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// Avoid race condition
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setTimeout(() => {
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if (this.ringingEnabled) {
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if (currentCall) {
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this.emit(EVENT_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, nextCall);
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} else {
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this.emit(EVENT_PLAY_INBOUND_CALL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, nextCall);
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}
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}
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this.emit(EVENT_INCOMING, nextCall);
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}, 100);
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}
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// @ts-ignore: does not exist
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if (!call.sipCall.isCanceled) {
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setTimeout(() => {
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this.emit(EVENT_PLAY_HANGUP_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, call);
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}, 10);
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}
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this.client.onCallEnded(call.sipCall);
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this.removeCall(call);
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}
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enableRinging(): Promise<void> | void {
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logger.info('softphone - enable ringing');
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this.ringingEnabled = true;
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}
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disableRinging(): Promise<void> | void {
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logger.info('softphone - disable ringing');
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this.ringingEnabled = false;
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}
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holdAllCalls(exceptCall: Call | null = null) {
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this.calls.filter(call => !call.isHeld() && call.id === exceptCall?.id).forEach(call => call.hold());
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}
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@@ -278,29 +347,43 @@ export class Softphone extends EventEmitter {
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this.audioRingVolume = volume;
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}
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getCalls(): Call[] {
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return this.calls;
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}
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getCurrentCall(): Call | undefined {
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return this.calls.find(call => call.isEstablished());
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}
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return this.
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getIncomingCalls(): Call[] {
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return this.calls.filter(call => call.isRinging());
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}
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get state(): StateTypes {
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return getState(this.softphoneActor) as StateTypes;
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}
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getUserAgent() {
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return this.client?.config?.userAgentString || 'softphone';
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}
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isRegistered() {
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return this.state === States.REGISTERED;
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}
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_bindEvents(): void {
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this.client.on(this.client.INVITE, (sipSession: SipCall) => {
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const call = new Call(sipSession, this);
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logger.info('softphone - on invite', { callId: call.id, number: call.number });
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call.onCallIncoming();
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this.calls.push(call);
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this.
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this.emit(EVENT_INCOMING, call);
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if (!this.getCurrentCall()) {
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if (this.ringingEnabled) {
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this.emit(EVENT_PLAY_RING_SOUND, this.audioRingDeviceId, this.audioRingVolume, call);
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}
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} else {
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this.emit(EVENT_PLAY_INBOUND_CALL_SOUND, this.audioOutputDeviceId, this.audioOutputVolume, call);
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}
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});
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this.client.on(this.client.ON_REINVITE, (sipCall: SipCall, request: IncomingRequestMessage, updatedCalleeName: string, updatedNumber: string, hadRemoteVideo: boolean) => {
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});
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this.client.on(this.client.ACCEPTED, async (sipCall: SipCall) => {
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const call = this._getCall(sipCall);
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this.emit(EVENT_TERMINATE_SOUND, call, 'call accepted');
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this._onCallEvent(sipCall, 'onAccepted');
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if (this.audioOutputDeviceId) {
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this.client.on(this.client.UNREGISTERED, () => {
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logger.info('softphone - unregistered');
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this.
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this.emit(EVENT_UNREGISTERED);
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this._sendAction(Actions.UNREGISTERED);
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});
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this.client.on(this.client.ON_DISCONNECTED, () => {
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logger.info('softphone - disconnected');
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this.
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this.emit(EVENT_DISCONNECTED);
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this._sendAction(Actions.TRANSPORT_CLOSED);
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});
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this.stopHeartbeat();
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this._sendAction(Actions.REGISTER_DONE);
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this.
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this.emit(EVENT_REGISTERED);
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// If the phone registered with a current callSession (eg: when switching network):
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// send a reinvite to renegociate ICE with new IP
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@@ -396,7 +482,7 @@ export class Softphone extends EventEmitter {
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this.client.on(this.client.DISCONNECTED, () => {
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logger.info('softphone - disconnected');
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this.emit(EVENT_DISCONNECTED);
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// Do not trigger heartbeat if already running
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if (!this.client.hasHeartbeat()) {
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@@ -408,33 +494,33 @@ export class Softphone extends EventEmitter {
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});
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this.client.on('onVideoInputChange', stream => {
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this.emit(EVENT_VIDEO_INPUT_CHANGE, stream);
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});
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this.client.on(this.client.MESSAGE, (message: Message) => {
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this._onMessage(message);
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this.emit(EVENT_ON_MESSAGE, message);
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});
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this.client.setOnHeartbeatTimeout(() => {
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this.emit(EVENT_HEARTBEAT_TIMEOUT);
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});
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this.client.setOnHeartbeatCallback(() => {
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this.
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this.emit(EVENT_HEARTBEAT);
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});
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}
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_sendAction(action: ActionTypes) {
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private _sendAction(action: ActionTypes) {
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this.softphoneActor.send({ type: action });
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}
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_getCall(sipCall: SipCall): Call | undefined {
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private _getCall(sipCall: SipCall): Call | undefined {
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return this.calls.find(call => call.id === sipCall.id);
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}
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_onCallEvent(sipCall: SipCall, eventName: string, ...args: any[]) {
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|
+
private _onCallEvent(sipCall: SipCall, eventName: string, ...args: any[]) {
|
|
438
524
|
logger.warn('softphone - received call event', { eventName, sipId: sipCall.id });
|
|
439
525
|
|
|
440
526
|
const call = this._getCall(sipCall);
|
|
@@ -447,7 +533,7 @@ export class Softphone extends EventEmitter {
|
|
|
447
533
|
call[eventName](...args);
|
|
448
534
|
}
|
|
449
535
|
|
|
450
|
-
_onMessage(message: Message & { method?: string, body?: string }): void {
|
|
536
|
+
private _onMessage(message: Message & { method?: string, body?: string }): void {
|
|
451
537
|
if (!message || message.method !== 'MESSAGE') {
|
|
452
538
|
return;
|
|
453
539
|
}
|
|
@@ -467,12 +553,12 @@ export class Softphone extends EventEmitter {
|
|
|
467
553
|
|
|
468
554
|
switch (type) {
|
|
469
555
|
case MESSAGE_TYPE_CHAT:
|
|
470
|
-
this.
|
|
556
|
+
this.emit(EVENT_ON_CHAT, content);
|
|
471
557
|
break;
|
|
472
558
|
|
|
473
559
|
case MESSAGE_TYPE_SIGNAL:
|
|
474
560
|
{
|
|
475
|
-
this.
|
|
561
|
+
this.emit(EVENT_ON_SIGNAL, content);
|
|
476
562
|
break;
|
|
477
563
|
}
|
|
478
564
|
|
package/src/web-rtc-client.ts
CHANGED
|
@@ -31,7 +31,7 @@ import IssueReporter from './service/IssueReporter';
|
|
|
31
31
|
import Heartbeat from './utils/Heartbeat';
|
|
32
32
|
import { getSipSessionId, makeURI, replaceLocalIpModifier, SIP_ID_LENGTH } from './utils/sdp';
|
|
33
33
|
import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, IncomingResponse, PeerConnection, WazoSession, WazoTransport } from './domain/types';
|
|
34
|
-
import { cleanupStream, createUaOptions, fetchNetworkStats, getAudioConstraints, getLocalStream, getLocalTracks, getLocalVideoStream, getMediaConfiguration, getPeerConnection, getRemoteStream, getRemoteTracks, getRemoteVideoStream, getRemoteVideoStreamFromPc,
|
|
34
|
+
import { cleanupStream, createUaOptions, fetchNetworkStats, getAudioConstraints, getLocalStream, getLocalTracks, getLocalVideoStream, getMediaConfiguration, getPeerConnection, getRemoteStream, getRemoteTracks, getRemoteVideoStream, getRemoteVideoStreamFromPc, getVideoConstraints, hasALocalVideoTrack, hasARemoteVideoTrack, hasAVideoTrack, hasLocalVideo, hasRemoteVideo, hasVideoTrack, isAudioOnly, isVideoRemotelyHeld, isWeb, sessionWantsToDoVideo, setLocalMediaStream, toggleAudio, toggleVideo, updateRemoteStream } from './utils/webrtc';
|
|
35
35
|
|
|
36
36
|
const SEND_STATS_DELAY = 5000;
|
|
37
37
|
const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
|