@nethesis/phone-island 0.7.79 → 0.7.81

package/dist/App.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react"),t=require("./components/Events.js"),s=require("./components/Socket.js"),r=require("./components/WebRTC.js"),o=require("./components/Island.js"),n=require("./components/RestAPI.js");require("./node_modules/react-redux/es/index.js");var a=require("./store/index.js");require("./node_modules/@fortawesome/react-fontawesome/index.es.js"),require("./node_modules/framer-motion/dist/framer-motion.js");var u=require("./node_modules/js-base64/base64.mjs.js");require("./node_modules/react-tooltip/dist/react-tooltip.min.css.js");var d=require("./node_modules/react-redux/es/components/Provider.js");function i(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}require("./node_modules/hoist-non-react-statics/dist/hoist-non-react-statics.cjs.js"),require("./node_modules/react-redux/node_modules/react-is/index.js"),require("./node_modules/react-redux/es/components/Context.js");var l=i(e),m=function(e){var i=e.dataConfig,m=e.showAlways,c=void 0!==m&&m,f=u.Base64.atob(i||"").split(":"),j=f[0],p=f[1],q=f[2],x=f[3],_=f[4];return l.default.createElement(l.default.Fragment,null,l.default.createElement(d.default,{store:a.store},l.default.createElement(r.WebRTC,{hostName:j,sipExten:x,sipSecret:_},l.default.createElement(n.RestAPI,{hostName:j,username:p,authToken:q},l.default.createElement(s.Socket,{hostName:j,username:p,authToken:q},l.default.createElement(t.Events,null,l.default.createElement(o.Island,{showAlways:c})))))))};m.displayName="PhoneIsland",exports.PhoneIsland=m;
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react"),t=require("./components/Events.js"),s=require("./components/Socket.js"),r=require("./components/WebRTC.js"),o=require("./components/Island.js"),n=require("./components/RestAPI.js");require("./node_modules/react-redux/es/index.js");var a=require("./store/index.js");require("./node_modules/@fortawesome/react-fontawesome/index.es.js"),require("./node_modules/framer-motion/dist/framer-motion.js");var u=require("./node_modules/js-base64/base64.mjs.js");require("./node_modules/react-tooltip/dist/react-tooltip.min.css.js");var d=require("./node_modules/react-redux/es/components/Provider.js");function i(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}require("./node_modules/hoist-non-react-statics/dist/hoist-non-react-statics.cjs.js"),require("./node_modules/react-redux/node_modules/react-is/index.js"),require("./node_modules/react-redux/es/components/Context.js");var l=i(e),m=function(e){var i=e.dataConfig,m=e.showAlways,c=void 0!==m&&m,f=u.Base64.atob(i||"").split(":"),j=f[0],p=f[1],q=f[2],x=f[3],_=f[4],h=f[5],v=f[6];return l.default.createElement(l.default.Fragment,null,l.default.createElement(d.default,{store:a.store},l.default.createElement(r.WebRTC,{hostName:j,sipExten:x,sipSecret:_,sipHost:h,sipPort:v},l.default.createElement(n.RestAPI,{hostName:j,username:p,authToken:q},l.default.createElement(s.Socket,{hostName:j,username:p,authToken:q},l.default.createElement(t.Events,{sipHost:h},l.default.createElement(o.Island,{showAlways:c})))))))};m.displayName="PhoneIsland",exports.PhoneIsland=m;
 //# sourceMappingURL=App.js.map

package/dist/App.js.map

@@ -1 +1 @@
-{"version":3,"file":"App.js","sources":["../src/App.tsx"],"sourcesContent":["import React, { type FC } from 'react'\nimport { Events, Socket, WebRTC, Island, RestAPI } from './components'\nimport { Provider } from 'react-redux'\nimport { store } from './store'\nimport { Base64 } from 'js-base64'\n\nimport 'react-tooltip/dist/react-tooltip.css'\n\ninterface PhoneIslandProps {\n  dataConfig: string\n  showAlways?: boolean\n}\n\nexport const PhoneIsland: FC<PhoneIslandProps> = ({ dataConfig, showAlways = false }) => {\n  const CONFIG: string[] = Base64.atob(dataConfig || '').split(':')\n  const HOST_NAME: string = CONFIG[0]\n  const USERNAME: string = CONFIG[1]\n  const AUTH_TOKEN: string = CONFIG[2]\n  const SIP_EXTEN: string = CONFIG[3]\n  const SIP_SECRET: string = CONFIG[4]\n\n  return (\n    <>\n      <Provider store={store}>\n        <WebRTC hostName={HOST_NAME} sipExten={SIP_EXTEN} sipSecret={SIP_SECRET}>\n          <RestAPI hostName={HOST_NAME} username={USERNAME} authToken={AUTH_TOKEN}>\n            <Socket hostName={HOST_NAME} username={USERNAME} authToken={AUTH_TOKEN}>\n              <Events>\n                <Island showAlways={showAlways} />\n              </Events>\n            </Socket>\n          </RestAPI>\n        </WebRTC>\n      </Provider>\n    </>\n  )\n}\n\nPhoneIsland.displayName = 'PhoneIsland'\n"],"names":["PhoneIsland","_a","dataConfig","_b","showAlways","CONFIG","Base64","atob","split","HOST_NAME","USERNAME","AUTH_TOKEN","SIP_EXTEN","SIP_SECRET","React","createElement","Fragment","Provider","store","WebRTC","hostName","sipExten","sipSecret","RestAPI","username","authToken","Socket","Events","Island","displayName"],"mappings":"w9BAaaA,EAAoC,SAACC,GAAE,IAAAC,eAAYC,EAAAF,EAAAG,WAAAA,OAAU,IAAAD,GAAQA,EAC1EE,EAAmBC,EAAMA,OAACC,KAAKL,GAAc,IAAIM,MAAM,KACvDC,EAAoBJ,EAAO,GAC3BK,EAAmBL,EAAO,GAC1BM,EAAqBN,EAAO,GAC5BO,EAAoBP,EAAO,GAC3BQ,EAAqBR,EAAO,GAElC,OACES,UAAAC,cAAAD,EAAA,QAAAE,SAAA,KACEF,EAAAA,QAAAC,cAACE,EAAAA,QAAQ,CAACC,MAAOA,EAAAA,OACfJ,UAAAC,cAACI,EAAMA,OAAA,CAACC,SAAUX,EAAWY,SAAUT,EAAWU,UAAWT,GAC3DC,UAAAC,cAACQ,EAAOA,QAAA,CAACH,SAAUX,EAAWe,SAAUd,EAAUe,UAAWd,GAC3DG,UAAAC,cAACW,EAAMA,OAAA,CAACN,SAAUX,EAAWe,SAAUd,EAAUe,UAAWd,GAC1DG,UAAAC,cAACY,EAAAA,OAAM,KACLb,UAAAC,cAACa,EAAMA,OAAA,CAACxB,WAAYA,SAQpC,EAEAJ,EAAY6B,YAAc"}
+{"version":3,"file":"App.js","sources":["../src/App.tsx"],"sourcesContent":["import React, { type FC } from 'react'\nimport { Events, Socket, WebRTC, Island, RestAPI } from './components'\nimport { Provider } from 'react-redux'\nimport { store } from './store'\nimport { Base64 } from 'js-base64'\n\nimport 'react-tooltip/dist/react-tooltip.css'\n\ninterface PhoneIslandProps {\n  dataConfig: string\n  showAlways?: boolean\n}\n\nexport const PhoneIsland: FC<PhoneIslandProps> = ({ dataConfig, showAlways = false }) => {\n  const CONFIG: string[] = Base64.atob(dataConfig || '').split(':')\n  const HOST_NAME: string = CONFIG[0]\n  const USERNAME: string = CONFIG[1]\n  const AUTH_TOKEN: string = CONFIG[2]\n  const SIP_EXTEN: string = CONFIG[3]\n  const SIP_SECRET: string = CONFIG[4]\n  const SIP_HOST: string = CONFIG[5]\n  const SIP_PORT: string = CONFIG[6]\n\n  return (\n    <>\n      <Provider store={store}>\n        <WebRTC\n          hostName={HOST_NAME}\n          sipExten={SIP_EXTEN}\n          sipSecret={SIP_SECRET}\n          sipHost={SIP_HOST}\n          sipPort={SIP_PORT}\n        >\n          <RestAPI hostName={HOST_NAME} username={USERNAME} authToken={AUTH_TOKEN}>\n            <Socket hostName={HOST_NAME} username={USERNAME} authToken={AUTH_TOKEN}>\n              <Events sipHost={SIP_HOST}>\n                <Island showAlways={showAlways} />\n              </Events>\n            </Socket>\n          </RestAPI>\n        </WebRTC>\n      </Provider>\n    </>\n  )\n}\n\nPhoneIsland.displayName = 'PhoneIsland'\n"],"names":["PhoneIsland","_a","dataConfig","_b","showAlways","CONFIG","Base64","atob","split","HOST_NAME","USERNAME","AUTH_TOKEN","SIP_EXTEN","SIP_SECRET","SIP_HOST","SIP_PORT","React","createElement","Fragment","Provider","store","WebRTC","hostName","sipExten","sipSecret","sipHost","sipPort","RestAPI","username","authToken","Socket","Events","Island","displayName"],"mappings":"w9BAaaA,EAAoC,SAACC,GAAE,IAAAC,eAAYC,EAAAF,EAAAG,WAAAA,OAAU,IAAAD,GAAQA,EAC1EE,EAAmBC,EAAMA,OAACC,KAAKL,GAAc,IAAIM,MAAM,KACvDC,EAAoBJ,EAAO,GAC3BK,EAAmBL,EAAO,GAC1BM,EAAqBN,EAAO,GAC5BO,EAAoBP,EAAO,GAC3BQ,EAAqBR,EAAO,GAC5BS,EAAmBT,EAAO,GAC1BU,EAAmBV,EAAO,GAEhC,OACEW,UAAAC,cAAAD,EAAA,QAAAE,SAAA,KACEF,EAAAA,QAAAC,cAACE,EAAAA,QAAQ,CAACC,MAAOA,EAAAA,OACfJ,EAAAA,QAACC,cAAAI,EAAAA,QACCC,SAAUb,EACVc,SAAUX,EACVY,UAAWX,EACXY,QAASX,EACTY,QAASX,GAETC,UAAAC,cAACU,EAAOA,QAAA,CAACL,SAAUb,EAAWmB,SAAUlB,EAAUmB,UAAWlB,GAC3DK,UAAAC,cAACa,EAAMA,OAAA,CAACR,SAAUb,EAAWmB,SAAUlB,EAAUmB,UAAWlB,GAC1DK,EAAAA,QAAAC,cAACc,EAAAA,OAAM,CAACN,QAASX,GACfE,UAAAC,cAACe,EAAMA,OAAA,CAAC5B,WAAYA,SAQpC,EAEAJ,EAAYiC,YAAc"}

package/dist/components/Events.d.ts

@@ -1,6 +1,7 @@
 import { type ReactNode, FC } from 'react';
 interface EventsProps {
     children: ReactNode;
+    sipHost: string;
 }
 export declare const Events: FC<EventsProps>;
 export {};

package/dist/components/Events.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react"),t=require("../events/CallEvents.js"),r=require("../events/PlayerEvents.js");function l(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}var n=l(e);exports.Events=function(e){var l=e.children;return n.default.createElement(n.default.Fragment,null,n.default.createElement(r.PlayerEvents,null),n.default.createElement(t.CallEvents,null),l)};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react"),t=require("../events/CallEvents.js"),r=require("../events/PlayerEvents.js");function l(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}var n=l(e);exports.Events=function(e){var l=e.sipHost,a=e.children;return n.default.createElement(n.default.Fragment,null,n.default.createElement(r.PlayerEvents,null),n.default.createElement(t.CallEvents,{sipHost:l}),a)};
 //# sourceMappingURL=Events.js.map

package/dist/components/Events.js.map

@@ -1 +1 @@
-{"version":3,"file":"Events.js","sources":["../../src/components/Events.tsx"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport React, { type ReactNode, FC } from 'react'\nimport { CallEvents, PlayerEvents } from '../events'\n\ninterface EventsProps {\n  children: ReactNode\n}\n\nexport const Events: FC<EventsProps> = ({ children }) => {\n  return (\n    <>\n      {<PlayerEvents />}\n      {<CallEvents />}\n      {children}\n    </>\n  )\n}\n"],"names":["_a","children","React","createElement","Fragment","PlayerEvents","CallEvents"],"mappings":"yQAUuC,SAACA,GAAE,IAAAC,EAAQD,EAAAC,SAChD,OACEC,UAAAC,cAAAD,EAAA,QAAAE,SAAA,KACGF,UAAAC,cAACE,EAAYA,aAAG,MAChBH,UAAAC,cAACG,EAAUA,WAAG,MACdL,EAGP"}
+{"version":3,"file":"Events.js","sources":["../../src/components/Events.tsx"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport React, { type ReactNode, FC } from 'react'\nimport { CallEvents, PlayerEvents } from '../events'\n\ninterface EventsProps {\n  children: ReactNode\n  sipHost: string\n}\n\nexport const Events: FC<EventsProps> = ({ sipHost, children }) => {\n  return (\n    <>\n      {<PlayerEvents />}\n      {<CallEvents sipHost={sipHost} />}\n      {children}\n    </>\n  )\n}\n"],"names":["_a","sipHost","children","React","createElement","Fragment","PlayerEvents","CallEvents"],"mappings":"yQAWuC,SAACA,OAAEC,EAAOD,EAAAC,QAAEC,EAAQF,EAAAE,SACzD,OACEC,UAAAC,cAAAD,EAAA,QAAAE,SAAA,KACGF,UAAAC,cAACE,EAAYA,aAAG,MAChBH,EAAAA,QAAAC,cAACG,EAAAA,WAAU,CAACN,QAASA,IACrBC,EAGP"}

package/dist/components/WebRTC.d.ts

@@ -4,6 +4,8 @@ interface WebRTCProps {
     sipExten: string;
     sipSecret: string;
     hostName: string;
+    sipHost: string;
+    sipPort: string;
 }
 export declare const WebRTC: FC<WebRTCProps>;
 export {};

package/dist/components/WebRTC.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react");require("../node_modules/react-redux/es/index.js");var t=require("../node_modules/webrtc-adapter/src/js/adapter_core.js"),r=require("../lib/webrtc/janus.js"),a=require("../lib/webrtc/messages.js"),n=require("../store/index.js"),o=require("../lib/devices/devices.js"),i=require("../lib/phone/call.js"),c=require("../lib/webrtc/connection.js"),s=require("../static/outgoing_ringtone.js"),l=require("../node_modules/react-redux/es/hooks/useDispatch.js");function u(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}var g=u(e),d=r.default;exports.WebRTC=function(r){var u=r.hostName,p=r.sipExten,b=r.sipSecret,f=r.children,m=l.useDispatch(),w=e.useRef(null);return e.useEffect((function(){o.checkMediaPermissions()}),[]),e.useEffect((function(){var e;return d.init({debug:"all",dependencies:d.useDefaultDependencies({adapter:t.default}),callback:function(){var e=new d({server:"https://".concat(u,"/janus"),success:function(){e.attach&&e.attach({plugin:"janus.plugin.sip",opaqueId:"sebastian_"+(new Date).getTime(),success:function(e){e&&(m.webrtc.updateWebRTC({sipcall:e}),a.register(p,b),e&&d.log&&d.log("SIP plugin attached! ("+e.getPlugin()+", id = )"))},error:function(e){d.error&&(d.error("  -- Error attaching plugin..."),d.error(e))},consentDialog:function(e){d.log&&d.log("janus consentDialog (on: ".concat(e,")"))},webrtcState:function(e){d.log&&d.log("Janus says our WebRTC PeerConnection is "+(e?"up":"down")+" now")},iceState:function(e){n.store.getState().webrtc.sipcall&&d.log&&d.log('ICE state of PeerConnection of handle has changed to "'.concat(e,'"'))},mediaState:function(e,t){d.log&&d.log("Janus "+(t?"started":"stopped")+" receiving our "+e)},slowLink:function(e,t){e?d.warn&&d.warn("SLOW link: several missing packets from janus (".concat(t,")")):d.warn&&d.warn("SLOW link: janus is not receiving all your packets (".concat(t,")"))},onmessage:function(e,t){var r=n.store.getState().webrtc.sipcall;d.debug&&(d.debug(" ::: Got a message :::"),d.debug(JSON.stringify(e)));var o=e.error;if(null==o||null==o){var c=e.result;if(null!=c&&void 0!==c.event&&null!==c.event)switch(c.event){case"registration_failed":d.error&&d.error("Registration failed: "+c.code+" "+c.reason);break;case"unregistered":d.log&&d.log("Successfully un-registered as "+c.username+"!");break;case"registered":d.log&&d.log("Successfully registered as "+c.username+"!"),n.store.getState().webrtc.registered||n.store.dispatch.webrtc.updateWebRTC({registered:!0}),m.webrtc.updateLastActivity((new Date).getTime());break;case"registering":d.log&&d.log("janus registering");break;case"calling":m.currentCall.checkOutgoingUpdate({outgoingWebRTC:!0}),m.webrtc.updateLastActivity((new Date).getTime());break;case"ringing":n.store.getState().player.audioPlayerPlaying||m.player.updateStartAudioPlayer({src:s.default,loop:!0}),m.webrtc.updateLastActivity((new Date).getTime());break;case"progress":d.log&&d.log("There's early media from "+c.username+", wairing for the call!"),null!=t&&a.handleRemote(t),m.webrtc.updateLastActivity((new Date).getTime());break;case"incomingcall":m.webrtc.updateWebRTC({jsepGlobal:t}),m.currentCall.checkIncomingUpdatePlay({incomingWebRTC:!0}),d.log&&d.log("Incoming call from "+c.username+"!"),m.webrtc.updateLastActivity((new Date).getTime());break;case"accepted":d.log&&d.log(c.username+" accepted the call!"),t&&a.handleRemote(t),m.currentCall.checkAcceptedUpdate({acceptedWebRTC:!0}),m.currentCall.updateCurrentCall({incoming:!1,incomingWebRTC:!1}),n.store.dispatch.player.stopAudioPlayer(),m.webrtc.updateLastActivity((new Date).getTime());break;case"hangup":i.hangupCurrentCall(),r.hangup(),d.log&&d.log("Call hung up ("+c.code+" "+c.reason+")!"),m.webrtc.updateLastActivity((new Date).getTime())}}else n.store.getState().webrtc.registered?(r&&r.hangup(),n.store.dispatch.player.stopAudioPlayer()):d.log&&d.log("User is not registered")},onlocalstream:function(e){d.debug&&(d.debug(" ::: Got a local stream :::"),d.debug(e))},onremotestream:function(e){d.debug&&d.debug(" ::: Got a remote stream :::"),n.store.dispatch.player.stopAudioPlayer();var t=n.store.getState().player.remoteAudio,r=n.store.getState().player.remoteVideo,a=e.getAudioTracks(),o=e.getVideoTracks();if(d.attachMediaStream){if(a&&a.length>0){var i=new MediaStream(a);t&&t.current&&d.attachMediaStream(t.current,i),n.store.dispatch.webrtc.updateRemoteAudioStream(i)}else console.warn("No audio tracks on remote stream");if(o&&o.length>0){var c=new MediaStream(o);r&&r.current&&d.attachMediaStream(r.current,c)}else console.warn("No video tracks on remote stream")}},oncleanup:function(){d.log&&d.log(" ::: janus Got a cleanup notification :::")},detached:function(){d.warn&&d.warn("SIP plugin handle detached from the plugin itself")}})},error:function(e){d.log&&d.log("error",e),m.alerts.setAlert("webrtc_down")},destroyed:function(){m.webrtc.updateWebRTC({destroyed:!0}),m.alerts.setAlert("webrtc_down")}})}}),e=n.store.getState().webrtc.CHECK_INTERVAL_TIME,w.current||(w.current=setInterval((function(){return c.webrtcCheck((function(){a.register(p,b)}))}),e)),function(){a.unregister(),clearInterval(w.current)}}),[]),g.default.createElement(g.default.Fragment,null,f)};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react");require("../node_modules/react-redux/es/index.js");var t=require("../node_modules/webrtc-adapter/src/js/adapter_core.js"),r=require("../lib/webrtc/janus.js"),a=require("../lib/webrtc/messages.js"),n=require("../store/index.js"),o=require("../lib/devices/devices.js"),s=require("../lib/phone/call.js"),i=require("../lib/webrtc/connection.js"),c=require("../static/outgoing_ringtone.js"),l=require("../node_modules/react-redux/es/hooks/useDispatch.js");function u(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}var g=u(e),d=r.default;exports.WebRTC=function(r){var u=r.hostName,p=r.sipExten,b=r.sipSecret,f=r.children,m=r.sipHost,w=r.sipPort,h=l.useDispatch(),v=e.useRef(null);return e.useEffect((function(){o.checkMediaPermissions()}),[]),e.useEffect((function(){var e;return d.init({debug:"all",dependencies:d.useDefaultDependencies({adapter:t.default}),callback:function(){var e=new d({server:"https://".concat(u,"/janus"),success:function(){e.attach&&e.attach({plugin:"janus.plugin.sip",opaqueId:"sebastian_"+(new Date).getTime(),success:function(e){e&&(h.webrtc.updateWebRTC({sipcall:e}),a.register({sipExten:p,sipSecret:b,sipHost:m,sipPort:w}),e&&d.log&&d.log("SIP plugin attached! ("+e.getPlugin()+", id = )"))},error:function(e){d.error&&(d.error("  -- Error attaching plugin..."),d.error(e))},consentDialog:function(e){d.log&&d.log("janus consentDialog (on: ".concat(e,")"))},webrtcState:function(e){d.log&&d.log("Janus says our WebRTC PeerConnection is "+(e?"up":"down")+" now")},iceState:function(e){n.store.getState().webrtc.sipcall&&d.log&&d.log('ICE state of PeerConnection of handle has changed to "'.concat(e,'"'))},mediaState:function(e,t){d.log&&d.log("Janus "+(t?"started":"stopped")+" receiving our "+e)},slowLink:function(e,t){e?d.warn&&d.warn("SLOW link: several missing packets from janus (".concat(t,")")):d.warn&&d.warn("SLOW link: janus is not receiving all your packets (".concat(t,")"))},onmessage:function(e,t){var r=n.store.getState().webrtc.sipcall;d.debug&&(d.debug(" ::: Got a message :::"),d.debug(JSON.stringify(e)));var o=e.error;if(null==o||null==o){var i=e.result;if(null!=i&&void 0!==i.event&&null!==i.event)switch(i.event){case"registration_failed":d.error&&d.error("Registration failed: "+i.code+" "+i.reason);break;case"unregistered":d.log&&d.log("Successfully un-registered as "+i.username+"!");break;case"registered":d.log&&d.log("Successfully registered as "+i.username+"!"),n.store.getState().webrtc.registered||n.store.dispatch.webrtc.updateWebRTC({registered:!0}),h.webrtc.updateLastActivity((new Date).getTime());break;case"registering":d.log&&d.log("janus registering");break;case"calling":h.currentCall.checkOutgoingUpdate({outgoingWebRTC:!0}),h.webrtc.updateLastActivity((new Date).getTime());break;case"ringing":n.store.getState().player.audioPlayerPlaying||h.player.updateStartAudioPlayer({src:c.default,loop:!0}),h.webrtc.updateLastActivity((new Date).getTime());break;case"progress":d.log&&d.log("There's early media from "+i.username+", wairing for the call!"),null!=t&&a.handleRemote(t),h.webrtc.updateLastActivity((new Date).getTime());break;case"incomingcall":h.webrtc.updateWebRTC({jsepGlobal:t}),h.currentCall.checkIncomingUpdatePlay({incomingWebRTC:!0}),d.log&&d.log("Incoming call from "+i.username+"!"),h.webrtc.updateLastActivity((new Date).getTime());break;case"accepted":d.log&&d.log(i.username+" accepted the call!"),t&&a.handleRemote(t),h.currentCall.checkAcceptedUpdate({acceptedWebRTC:!0}),h.currentCall.updateCurrentCall({incoming:!1,incomingWebRTC:!1}),n.store.dispatch.player.stopAudioPlayer(),h.webrtc.updateLastActivity((new Date).getTime());break;case"hangup":s.hangupCurrentCall(),r.hangup(),d.log&&d.log("Call hung up ("+i.code+" "+i.reason+")!"),h.webrtc.updateLastActivity((new Date).getTime())}}else n.store.getState().webrtc.registered?(r&&r.hangup(),n.store.dispatch.player.stopAudioPlayer()):d.log&&d.log("User is not registered")},onlocalstream:function(e){d.debug&&(d.debug(" ::: Got a local stream :::"),d.debug(e))},onremotestream:function(e){d.debug&&d.debug(" ::: Got a remote stream :::"),n.store.dispatch.player.stopAudioPlayer();var t=n.store.getState().player.remoteAudio,r=n.store.getState().player.remoteVideo,a=e.getAudioTracks(),o=e.getVideoTracks();if(d.attachMediaStream){if(a&&a.length>0){var s=new MediaStream(a);t&&t.current&&d.attachMediaStream(t.current,s),n.store.dispatch.webrtc.updateRemoteAudioStream(s)}else console.warn("No audio tracks on remote stream");if(o&&o.length>0){var i=new MediaStream(o);r&&r.current&&d.attachMediaStream(r.current,i)}else console.warn("No video tracks on remote stream")}},oncleanup:function(){d.log&&d.log(" ::: janus Got a cleanup notification :::")},detached:function(){d.warn&&d.warn("SIP plugin handle detached from the plugin itself")}})},error:function(e){d.log&&d.log("error",e),h.alerts.setAlert("webrtc_down")},destroyed:function(){h.webrtc.updateWebRTC({destroyed:!0}),h.alerts.setAlert("webrtc_down")}})}}),e=n.store.getState().webrtc.CHECK_INTERVAL_TIME,v.current||(v.current=setInterval((function(){return i.webrtcCheck((function(){a.register({sipExten:p,sipSecret:b,sipHost:m,sipPort:w})}))}),e)),function(){a.unregister(),clearInterval(v.current)}}),[]),g.default.createElement(g.default.Fragment,null,f)};
 //# sourceMappingURL=WebRTC.js.map

package/dist/components/WebRTC.js.map

@@ -1 +1 @@
-{"version":3,"file":"WebRTC.js","sources":["../../src/components/WebRTC.tsx"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport React, { type ReactNode, FC, useEffect, useState, useRef } from 'react'\nimport { useDispatch } from 'react-redux'\nimport { Dispatch } from '../store'\nimport adapter from 'webrtc-adapter'\nimport JanusLib from '../lib/webrtc/janus.js'\nimport type { JanusTypes } from '../types'\nimport { register, unregister, handleRemote } from '../lib/webrtc/messages'\nimport { store, type RootState } from '../store'\nimport { checkMediaPermissions } from '../lib/devices/devices'\nimport { hangupCurrentCall } from '../lib/phone/call'\nimport { webrtcCheck } from '../lib/webrtc/connection'\nimport outgoingRingtone from '../static/outgoing_ringtone'\n\ninterface WebRTCProps {\n  children: ReactNode\n  sipExten: string\n  sipSecret: string\n  hostName: string\n}\n\nconst Janus: JanusTypes = JanusLib\n\nexport const WebRTC: FC<WebRTCProps> = ({ hostName, sipExten, sipSecret, children }) => {\n  const dispatch = useDispatch<Dispatch>()\n\n  // Initialize janus check interval id\n  const janusCheckInterval = useRef<any>(null)\n\n  // check audio and video permissions\n  useEffect(() => {\n    checkMediaPermissions()\n  }, [])\n\n  useEffect(() => {\n    const setupDeps = () =>\n      Janus.useDefaultDependencies({\n        adapter,\n      })\n\n    function initWebRTC() {\n      Janus.init({\n        debug: 'all',\n        dependencies: setupDeps(),\n        callback: function () {\n          const janusInstance = new Janus({\n            server: `https://${hostName}/janus`,\n            success: () => {\n              if (janusInstance.attach) {\n                // Use Janus Sip Plugin\n                janusInstance.attach({\n                  plugin: 'janus.plugin.sip',\n                  opaqueId: 'sebastian' + '_' + new Date().getTime(),\n                  success: function (pluginHandle) {\n                    // Set sipcall to the store\n                    if (pluginHandle) {\n                      dispatch.webrtc.updateWebRTC({\n                        sipcall: pluginHandle,\n                      })\n                      // Register the extension to the server\n                      register(sipExten, sipSecret)\n                      if (pluginHandle) {\n                        if (Janus.log)\n                          Janus.log(\n                            'SIP plugin attached! (' + pluginHandle.getPlugin() + ', id = ' + ')',\n                          )\n                      }\n                    }\n                  },\n                  error: function (error) {\n                    if (Janus.error) {\n                      Janus.error('  -- Error attaching plugin...')\n                      Janus.error(error)\n                    }\n                    // reject()\n                  },\n                  consentDialog: function (on) {\n                    if (Janus.log) Janus.log(`janus consentDialog (on: ${on})`)\n                  },\n                  webrtcState: function (on) {\n                    if (Janus.log)\n                      Janus.log(\n                        'Janus says our WebRTC PeerConnection is ' + (on ? 'up' : 'down') + ' now',\n                      )\n                  },\n                  iceState: function (newState) {\n                    const { sipcall }: { sipcall: any } = store.getState().webrtc\n\n                    if (sipcall) {\n                      if (Janus.log)\n                        Janus.log(\n                          `ICE state of PeerConnection of handle has changed to \"${newState}\"`,\n                        )\n                    }\n                  },\n                  mediaState: function (medium, on) {\n                    if (Janus.log)\n                      Janus.log(\n                        'Janus ' + (on ? 'started' : 'stopped') + ' receiving our ' + medium,\n                      )\n                  },\n                  slowLink: function (uplink, count) {\n                    if (uplink) {\n                      if (Janus.warn)\n                        Janus.warn(`SLOW link: several missing packets from janus (${count})`)\n                    } else {\n                      if (Janus.warn)\n                        Janus.warn(`SLOW link: janus is not receiving all your packets (${count})`)\n                    }\n                  },\n                  onmessage: function (msg, jsep) {\n                    const { sipcall }: { sipcall: any } = store.getState().webrtc\n\n                    if (Janus.debug) {\n                      Janus.debug(' ::: Got a message :::')\n                      Janus.debug(JSON.stringify(msg))\n                    }\n\n                    // Handle errors in message\n                    var error = msg['error']\n                    if (error != null && error != undefined) {\n                      if (!store.getState().webrtc.registered) {\n                        if (Janus.log) Janus.log('User is not registered')\n                      } else {\n                        // Reset status\n                        sipcall && sipcall.hangup()\n\n                        // Stop the local audio element ringing\n                        store.dispatch.player.stopAudioPlayer()\n                      }\n                      return\n                    }\n                    // Manage events\n                    var result = msg['result']\n                    if (\n                      result !== null &&\n                      result !== undefined &&\n                      result['event'] !== undefined &&\n                      result['event'] !== null\n                    ) {\n                      // get event\n                      var event = result['event']\n\n                      //switch event\n                      switch (event) {\n                        case 'registration_failed':\n                          if (Janus.error)\n                            Janus.error(\n                              'Registration failed: ' + result['code'] + ' ' + result['reason'],\n                            )\n                          break\n\n                        case 'unregistered':\n                          if (Janus.log)\n                            Janus.log('Successfully un-registered as ' + result['username'] + '!')\n                          // registered = false\n                          break\n\n                        case 'registered':\n                          if (Janus.log)\n                            Janus.log('Successfully registered as ' + result['username'] + '!')\n                          if (!store.getState().webrtc.registered) {\n                            store.dispatch.webrtc.updateWebRTC({\n                              registered: true,\n                            })\n                          }\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'registering':\n                          if (Janus.log) Janus.log('janus registering')\n                          break\n\n                        // This event arrive on outgoing call start\n                        case 'calling':\n                          // Number and display name are updated inside socket\n                          dispatch.currentCall.checkOutgoingUpdate({\n                            outgoingWebRTC: true,\n                          })\n\n                          // Update webrtc last activity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        // After an outgoing call start on 180 code, it means\n                        // ...that the local outgoing ringtone must be player\n                        case 'ringing':\n                          const { audioPlayerPlaying } = store.getState().player\n\n                          // Check if the local audio is already playing and start playing\n                          if (!audioPlayerPlaying) {\n                            // Update audio player and start playing\n                            dispatch.player.updateStartAudioPlayer({\n                              src: outgoingRingtone,\n                              loop: true,\n                            })\n                          }\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        // After an outgoing call start on 183 code, it means\n                        // ...that the outgoing ringtone arrives from the stream\n                        // ...playing the local outgoing ringtone isn't needed\n                        case 'progress':\n                          if (Janus.log)\n                            Janus.log(\n                              \"There's early media from \" +\n                                result['username'] +\n                                ', wairing for the call!',\n                            )\n                          if (jsep !== null && jsep !== undefined) {\n                            handleRemote(jsep)\n                          }\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'incomingcall':\n                          dispatch.webrtc.updateWebRTC({ jsepGlobal: jsep })\n\n                          // Number and display name are updated inside socket\n                          dispatch.currentCall.checkIncomingUpdatePlay({\n                            incomingWebRTC: true,\n                          })\n\n                          if (Janus.log) Janus.log('Incoming call from ' + result['username'] + '!')\n                          // Update the webrtc last activity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'accepted':\n                          if (Janus.log) Janus.log(result['username'] + ' accepted the call!')\n                          if (jsep) {\n                            handleRemote(jsep)\n                          }\n                          // Set current call accepted\n                          dispatch.currentCall.checkAcceptedUpdate({\n                            acceptedWebRTC: true,\n                          })\n                          // Set incoming value to false\n                          dispatch.currentCall.updateCurrentCall({\n                            incoming: false,\n                            incomingWebRTC: false,\n                          })\n\n                          // Stop the local audio element ringing\n                          store.dispatch.player.stopAudioPlayer()\n\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'hangup':\n                          hangupCurrentCall()\n                          sipcall.hangup()\n                          // if (\n                          //   result['code'] === 486 &&\n                          //   result['event'] === 'hangup' &&\n                          //   result['reason'] === 'Busy Here'\n                          // ) {\n                          //   dispatch.player.updateAudioSource({\n                          //     src: busyRingtone,\n                          //   })\n                          //   dispatch.player.playAudio()\n                          // }\n                          if (Janus.log)\n                            Janus.log(\n                              'Call hung up (' + result['code'] + ' ' + result['reason'] + ')!',\n                            )\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          // stopScreenSharingI()\n                          break\n\n                        default:\n                          break\n                      }\n                    }\n                  },\n                  onlocalstream: function (stream) {\n                    // const localVideoElement = store.getState().player.localVideo\n                    if (Janus.debug) {\n                      Janus.debug(' ::: Got a local stream :::')\n                      Janus.debug(stream)\n                    }\n                    // if (Janus.attachMediaStream) Janus.attachMediaStream(localVideoElement, stream)\n                    /* IS VIDEO ENABLED ? */\n                    // var videoTracks = stream.getVideoTracks()\n                    /* */\n                  },\n                  onremotestream: function (stream: MediaStream) {\n                    if (Janus.debug) {\n                      Janus.debug(' ::: Got a remote stream :::')\n                    }\n                    // Stop the local audio element ringing\n                    store.dispatch.player.stopAudioPlayer()\n\n                    // Get remote audio and video elements\n                    const remoteAudioElement = store.getState().player.remoteAudio\n                    const remoteVideoElement = store.getState().player.remoteVideo\n\n                    // Get audio and video from stream\n                    const audioTracks: MediaStreamTrack[] = stream.getAudioTracks()\n                    const videoTracks: MediaStreamTrack[] = stream.getVideoTracks()\n\n                    if (Janus.attachMediaStream) {\n                      // Initialize the new media stream for remote audio\n                      if (audioTracks && audioTracks.length > 0) {\n                        const audioStream: MediaStream = new MediaStream(audioTracks)\n                        remoteAudioElement &&\n                          remoteAudioElement.current &&\n                          Janus.attachMediaStream(remoteAudioElement.current, audioStream)\n\n                        // Save the new audio stream to the store\n                        store.dispatch.webrtc.updateRemoteAudioStream(audioStream)\n                      } else {\n                        console.warn('No audio tracks on remote stream')\n                      }\n                      // Initialize the new media stream for remote video\n                      if (videoTracks && videoTracks.length > 0) {\n                        const videoStream: MediaStream = new MediaStream(videoTracks)\n                        remoteVideoElement &&\n                          remoteVideoElement.current &&\n                          Janus.attachMediaStream(remoteVideoElement.current, videoStream)\n                      } else {\n                        console.warn('No video tracks on remote stream')\n                      }\n                    }\n                  },\n                  oncleanup: function () {\n                    if (Janus.log) Janus.log(' ::: janus Got a cleanup notification :::')\n                  },\n                  detached: function () {\n                    if (Janus.warn) Janus.warn('SIP plugin handle detached from the plugin itself')\n                  },\n                })\n              }\n            },\n            error: (err: any) => {\n              if (Janus.log) Janus.log('error', err)\n              // Activate webrtc connection alert\n              dispatch.alerts.setAlert('webrtc_down')\n            },\n            destroyed: () => {\n              // Set webrtc destroyed status\n              dispatch.webrtc.updateWebRTC({\n                destroyed: true,\n              })\n              // Activate webrtc connection alert\n              dispatch.alerts.setAlert('webrtc_down')\n            },\n          })\n        },\n      })\n    }\n\n    // Initializes the webrtc registration check interval\n    function startWebrtcCheck() {\n      const { CHECK_INTERVAL_TIME } = store.getState().webrtc\n      if (!janusCheckInterval.current) {\n        // Initialize the interval that check the webrtc\n        janusCheckInterval.current = setInterval(\n          () =>\n            webrtcCheck(() => {\n              // Do the register as callback of webrtc check\n              register(sipExten, sipSecret)\n            }),\n          CHECK_INTERVAL_TIME,\n        )\n      }\n    }\n\n    // Start webrtc initialization and handlers\n    initWebRTC()\n    // Start the check of webrtc activity\n    startWebrtcCheck()\n\n    return () => {\n      // Unregister from janus\n      unregister()\n      // Stop Janus check interval\n      clearInterval(janusCheckInterval.current)\n    }\n  }, [])\n\n  return 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+{"version":3,"file":"WebRTC.js","sources":["../../src/components/WebRTC.tsx"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport React, { type ReactNode, FC, useEffect, useState, useRef } from 'react'\nimport { useDispatch } from 'react-redux'\nimport { Dispatch } from '../store'\nimport adapter from 'webrtc-adapter'\nimport JanusLib from '../lib/webrtc/janus.js'\nimport type { JanusTypes } from '../types'\nimport { register, unregister, handleRemote } from '../lib/webrtc/messages'\nimport { store, type RootState } from '../store'\nimport { checkMediaPermissions } from '../lib/devices/devices'\nimport { hangupCurrentCall } from '../lib/phone/call'\nimport { webrtcCheck } from '../lib/webrtc/connection'\nimport outgoingRingtone from '../static/outgoing_ringtone'\n\ninterface WebRTCProps {\n  children: ReactNode\n  sipExten: string\n  sipSecret: string\n  hostName: string\n  sipHost: string\n  sipPort: string\n}\n\nconst Janus: JanusTypes = JanusLib\n\nexport const WebRTC: FC<WebRTCProps> = ({\n  hostName,\n  sipExten,\n  sipSecret,\n  children,\n  sipHost,\n  sipPort,\n}) => {\n  const dispatch = useDispatch<Dispatch>()\n\n  // Initialize janus check interval id\n  const janusCheckInterval = useRef<any>(null)\n\n  // check audio and video permissions\n  useEffect(() => {\n    checkMediaPermissions()\n  }, [])\n\n  useEffect(() => {\n    const setupDeps = () =>\n      Janus.useDefaultDependencies({\n        adapter,\n      })\n\n    function initWebRTC() {\n      Janus.init({\n        debug: 'all',\n        dependencies: setupDeps(),\n        callback: function () {\n          const janusInstance = new Janus({\n            server: `https://${hostName}/janus`,\n            success: () => {\n              if (janusInstance.attach) {\n                // Use Janus Sip Plugin\n                janusInstance.attach({\n                  plugin: 'janus.plugin.sip',\n                  opaqueId: 'sebastian' + '_' + new Date().getTime(),\n                  success: function (pluginHandle) {\n                    // Set sipcall to the store\n                    if (pluginHandle) {\n                      dispatch.webrtc.updateWebRTC({\n                        sipcall: pluginHandle,\n                      })\n                      // Register the extension to the server\n                      register({ sipExten, sipSecret, sipHost, sipPort })\n                      if (pluginHandle) {\n                        if (Janus.log)\n                          Janus.log(\n                            'SIP plugin attached! (' + pluginHandle.getPlugin() + ', id = ' + ')',\n                          )\n                      }\n                    }\n                  },\n                  error: function (error) {\n                    if (Janus.error) {\n                      Janus.error('  -- Error attaching plugin...')\n                      Janus.error(error)\n                    }\n                    // reject()\n                  },\n                  consentDialog: function (on) {\n                    if (Janus.log) Janus.log(`janus consentDialog (on: ${on})`)\n                  },\n                  webrtcState: function (on) {\n                    if (Janus.log)\n                      Janus.log(\n                        'Janus says our WebRTC PeerConnection is ' + (on ? 'up' : 'down') + ' now',\n                      )\n                  },\n                  iceState: function (newState) {\n                    const { sipcall }: { sipcall: any } = store.getState().webrtc\n\n                    if (sipcall) {\n                      if (Janus.log)\n                        Janus.log(\n                          `ICE state of PeerConnection of handle has changed to \"${newState}\"`,\n                        )\n                    }\n                  },\n                  mediaState: function (medium, on) {\n                    if (Janus.log)\n                      Janus.log(\n                        'Janus ' + (on ? 'started' : 'stopped') + ' receiving our ' + medium,\n                      )\n                  },\n                  slowLink: function (uplink, count) {\n                    if (uplink) {\n                      if (Janus.warn)\n                        Janus.warn(`SLOW link: several missing packets from janus (${count})`)\n                    } else {\n                      if (Janus.warn)\n                        Janus.warn(`SLOW link: janus is not receiving all your packets (${count})`)\n                    }\n                  },\n                  onmessage: function (msg, jsep) {\n                    const { sipcall }: { sipcall: any } = store.getState().webrtc\n\n                    if (Janus.debug) {\n                      Janus.debug(' ::: Got a message :::')\n                      Janus.debug(JSON.stringify(msg))\n                    }\n\n                    // Handle errors in message\n                    var error = msg['error']\n                    if (error != null && error != undefined) {\n                      if (!store.getState().webrtc.registered) {\n                        if (Janus.log) Janus.log('User is not registered')\n                      } else {\n                        // Reset status\n                        sipcall && sipcall.hangup()\n\n                        // Stop the local audio element ringing\n                        store.dispatch.player.stopAudioPlayer()\n                      }\n                      return\n                    }\n                    // Manage events\n                    var result = msg['result']\n                    if (\n                      result !== null &&\n                      result !== undefined &&\n                      result['event'] !== undefined &&\n                      result['event'] !== null\n                    ) {\n                      // get event\n                      var event = result['event']\n\n                      //switch event\n                      switch (event) {\n                        case 'registration_failed':\n                          if (Janus.error)\n                            Janus.error(\n                              'Registration failed: ' + result['code'] + ' ' + result['reason'],\n                            )\n                          break\n\n                        case 'unregistered':\n                          if (Janus.log)\n                            Janus.log('Successfully un-registered as ' + result['username'] + '!')\n                          // registered = false\n                          break\n\n                        case 'registered':\n                          if (Janus.log)\n                            Janus.log('Successfully registered as ' + result['username'] + '!')\n                          if (!store.getState().webrtc.registered) {\n                            store.dispatch.webrtc.updateWebRTC({\n                              registered: true,\n                            })\n                          }\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'registering':\n                          if (Janus.log) Janus.log('janus registering')\n                          break\n\n                        // This event arrive on outgoing call start\n                        case 'calling':\n                          // Number and display name are updated inside socket\n                          dispatch.currentCall.checkOutgoingUpdate({\n                            outgoingWebRTC: true,\n                          })\n\n                          // Update webrtc last activity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        // After an outgoing call start on 180 code, it means\n                        // ...that the local outgoing ringtone must be player\n                        case 'ringing':\n                          const { audioPlayerPlaying } = store.getState().player\n\n                          // Check if the local audio is already playing and start playing\n                          if (!audioPlayerPlaying) {\n                            // Update audio player and start playing\n                            dispatch.player.updateStartAudioPlayer({\n                              src: outgoingRingtone,\n                              loop: true,\n                            })\n                          }\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        // After an outgoing call start on 183 code, it means\n                        // ...that the outgoing ringtone arrives from the stream\n                        // ...playing the local outgoing ringtone isn't needed\n                        case 'progress':\n                          if (Janus.log)\n                            Janus.log(\n                              \"There's early media from \" +\n                                result['username'] +\n                                ', wairing for the call!',\n                            )\n                          if (jsep !== null && jsep !== undefined) {\n                            handleRemote(jsep)\n                          }\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'incomingcall':\n                          dispatch.webrtc.updateWebRTC({ jsepGlobal: jsep })\n\n                          // Number and display name are updated inside socket\n                          dispatch.currentCall.checkIncomingUpdatePlay({\n                            incomingWebRTC: true,\n                          })\n\n                          if (Janus.log) Janus.log('Incoming call from ' + result['username'] + '!')\n                          // Update the webrtc last activity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'accepted':\n                          if (Janus.log) Janus.log(result['username'] + ' accepted the call!')\n                          if (jsep) {\n                            handleRemote(jsep)\n                          }\n                          // Set current call accepted\n                          dispatch.currentCall.checkAcceptedUpdate({\n                            acceptedWebRTC: true,\n                          })\n                          // Set incoming value to false\n                          dispatch.currentCall.updateCurrentCall({\n                            incoming: false,\n                            incomingWebRTC: false,\n                          })\n\n                          // Stop the local audio element ringing\n                          store.dispatch.player.stopAudioPlayer()\n\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          break\n\n                        case 'hangup':\n                          hangupCurrentCall()\n                          sipcall.hangup()\n                          // if (\n                          //   result['code'] === 486 &&\n                          //   result['event'] === 'hangup' &&\n                          //   result['reason'] === 'Busy Here'\n                          // ) {\n                          //   dispatch.player.updateAudioSource({\n                          //     src: busyRingtone,\n                          //   })\n                          //   dispatch.player.playAudio()\n                          // }\n                          if (Janus.log)\n                            Janus.log(\n                              'Call hung up (' + result['code'] + ' ' + result['reason'] + ')!',\n                            )\n                          // Update webrtc lastActivity time\n                          dispatch.webrtc.updateLastActivity(new Date().getTime())\n                          // stopScreenSharingI()\n                          break\n\n                        default:\n                          break\n                      }\n                    }\n                  },\n                  onlocalstream: function (stream) {\n                    // const localVideoElement = store.getState().player.localVideo\n                    if (Janus.debug) {\n                      Janus.debug(' ::: Got a local stream :::')\n                      Janus.debug(stream)\n                    }\n                    // if (Janus.attachMediaStream) Janus.attachMediaStream(localVideoElement, stream)\n                    /* IS VIDEO ENABLED ? */\n                    // var videoTracks = stream.getVideoTracks()\n                    /* */\n                  },\n                  onremotestream: function (stream: MediaStream) {\n                    if (Janus.debug) {\n                      Janus.debug(' ::: Got a remote stream :::')\n                    }\n                    // Stop the local audio element ringing\n                    store.dispatch.player.stopAudioPlayer()\n\n                    // Get remote audio and video elements\n                    const remoteAudioElement = store.getState().player.remoteAudio\n                    const remoteVideoElement = store.getState().player.remoteVideo\n\n                    // Get audio and video from stream\n                    const audioTracks: MediaStreamTrack[] = stream.getAudioTracks()\n                    const videoTracks: MediaStreamTrack[] = stream.getVideoTracks()\n\n                    if (Janus.attachMediaStream) {\n                      // Initialize the new media stream for remote audio\n                      if (audioTracks && audioTracks.length > 0) {\n                        const audioStream: MediaStream = new MediaStream(audioTracks)\n                        remoteAudioElement &&\n                          remoteAudioElement.current &&\n                          Janus.attachMediaStream(remoteAudioElement.current, audioStream)\n\n                        // Save the new audio stream to the store\n                        store.dispatch.webrtc.updateRemoteAudioStream(audioStream)\n                      } else {\n                        console.warn('No audio tracks on remote stream')\n                      }\n                      // Initialize the new media stream for remote video\n                      if (videoTracks && videoTracks.length > 0) {\n                        const videoStream: MediaStream = new MediaStream(videoTracks)\n                        remoteVideoElement &&\n                          remoteVideoElement.current &&\n                          Janus.attachMediaStream(remoteVideoElement.current, videoStream)\n                      } else {\n                        console.warn('No video tracks on remote stream')\n                      }\n                    }\n                  },\n                  oncleanup: function () {\n                    if (Janus.log) Janus.log(' ::: janus Got a cleanup notification :::')\n                  },\n                  detached: function () {\n                    if (Janus.warn) Janus.warn('SIP plugin handle detached from the plugin itself')\n                  },\n                })\n              }\n            },\n            error: (err: any) => {\n              if (Janus.log) Janus.log('error', err)\n              // Activate webrtc connection alert\n              dispatch.alerts.setAlert('webrtc_down')\n            },\n            destroyed: () => {\n              // Set webrtc destroyed status\n              dispatch.webrtc.updateWebRTC({\n                destroyed: true,\n              })\n              // Activate webrtc connection alert\n              dispatch.alerts.setAlert('webrtc_down')\n            },\n          })\n        },\n      })\n    }\n\n    // Initializes the webrtc registration check interval\n    function startWebrtcCheck() {\n      const { CHECK_INTERVAL_TIME } = store.getState().webrtc\n      if (!janusCheckInterval.current) {\n        // Initialize the interval that check the webrtc\n        janusCheckInterval.current = setInterval(\n          () =>\n            webrtcCheck(() => {\n              // Do the register as callback of webrtc check\n              register({ sipExten, sipSecret, sipHost, sipPort })\n            }),\n          CHECK_INTERVAL_TIME,\n        )\n      }\n    }\n\n    // Start webrtc initialization and handlers\n    initWebRTC()\n    // Start the check of webrtc activity\n    startWebrtcCheck()\n\n    return () => {\n      // Unregister from janus\n      unregister()\n      // Stop Janus check interval\n      clearInterval(janusCheckInterval.current)\n    }\n  }, [])\n\n  return 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package/dist/events/CallEvents.d.ts

@@ -1,5 +1,7 @@
 import { FC } from 'react';
-export declare const CallEvents: FC;
+export declare const CallEvents: FC<{
+    sipHost: string;
+}>;
 /**
  * Dispatch the phone-island-outgoing-call-started event
  */

package/dist/events/CallEvents.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react"),t=require("../utils/customHooks/useEventListener.js"),r=require("../utils/genericFunctions/eventDispatch.js");require("../node_modules/react-redux/es/index.js");var s=require("../lib/phone/call.js"),n=require("../node_modules/react-redux/es/hooks/useSelector.js");function u(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}require("../node_modules/react-redux/es/components/Context.js"),require("../node_modules/hoist-non-react-statics/dist/hoist-non-react-statics.cjs.js"),require("../node_modules/react-redux/node_modules/react-is/index.js");var o=u(e);exports.CallEvents=function(){return n.useSelector((function(e){return e.webrtc})).sipcall,t.useEventListener("phone-island-call-start",(function(e){var t=e.number.replace(/\s/g,"");s.callNumber(t)})),o.default.createElement(o.default.Fragment,null)},exports.dispatchOutgoingCallStarted=function(e,t){void 0===e&&(e=""),void 0===t&&(t="");var s={name:e,number:t};r.eventDispatch("phone-island-outgoing-call-started",s)};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("react"),t=require("../utils/customHooks/useEventListener.js"),r=require("../utils/genericFunctions/eventDispatch.js");require("../node_modules/react-redux/es/index.js");var s=require("../lib/phone/call.js"),n=require("../node_modules/react-redux/es/hooks/useSelector.js");function o(e){return e&&"object"==typeof e&&"default"in e?e:{default:e}}require("../node_modules/react-redux/es/components/Context.js"),require("../node_modules/hoist-non-react-statics/dist/hoist-non-react-statics.cjs.js"),require("../node_modules/react-redux/node_modules/react-is/index.js");var u=o(e);exports.CallEvents=function(e){var r=e.sipHost;return n.useSelector((function(e){return e.webrtc})).sipcall,t.useEventListener("phone-island-call-start",(function(e){var t=e.number.replace(/\s/g,"");s.callNumber(t,r)})),u.default.createElement(u.default.Fragment,null)},exports.dispatchOutgoingCallStarted=function(e,t){void 0===e&&(e=""),void 0===t&&(t="");var s={name:e,number:t};r.eventDispatch("phone-island-outgoing-call-started",s)};
 //# sourceMappingURL=CallEvents.js.map

package/dist/events/CallEvents.js.map

@@ -1 +1 @@
-{"version":3,"file":"CallEvents.js","sources":["../../src/events/CallEvents.tsx"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport React, { FC } from 'react'\nimport { useEventListener, eventDispatch } from '../utils'\nimport { useSelector } from 'react-redux'\nimport { RootState } from '../store'\nimport { callNumber } from '../lib/phone/call'\n\nexport const CallEvents: FC = () => {\n  const { sipcall }: any = useSelector((state: RootState) => state.webrtc)\n  /**\n   * Event listner for phone-island-call-start event\n   */\n  useEventListener('phone-island-call-start', (data: CallStartTypes) => {\n    const number = data.number.replace(/\\s/g, '')\n    callNumber(number)\n  })\n  return <></>\n}\n\n// !TODO add phone-island-outgoing-call-started phone-island-outgoing-call-ended\n\n/**\n * Dispatch the phone-island-outgoing-call-started event\n */\nexport function dispatchOutgoingCallStarted(name: string = '', number: string = '') {\n  const data: OutgoingCallStartedTypes = {\n    name,\n    number,\n  }\n  eventDispatch('phone-island-outgoing-call-started', data)\n}\n\nexport interface OutgoingCallStartedTypes {\n  name: string\n  number: string\n}\n\ninterface CallStartTypes {\n  number: string\n}\n"],"names":["useSelector","state","webrtc","useEventListener","data","number","replace","callNumber","React","name","eventDispatch"],"mappings":"sqBAS8B,WAS5B,OARyBA,EAAWA,aAAC,SAACC,GAAqB,OAAAA,EAAMC,MAAM,YAIvEC,mBAAiB,2BAA2B,SAACC,GAC3C,IAAMC,EAASD,EAAKC,OAAOC,QAAQ,MAAO,IAC1CC,EAAUA,WAACF,EACb,IACOG,gDACT,sCAOgB,SAA4BC,EAAmBJ,QAAnB,IAAAI,IAAAA,EAAiB,SAAE,IAAAJ,IAAAA,EAAmB,IAChF,IAAMD,EAAiC,CACrCK,KAAIA,EACJJ,OAAMA,GAERK,gBAAc,qCAAsCN,EACtD"}
+{"version":3,"file":"CallEvents.js","sources":["../../src/events/CallEvents.tsx"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport React, { FC } from 'react'\nimport { useEventListener, eventDispatch } from '../utils'\nimport { useSelector } from 'react-redux'\nimport { RootState } from '../store'\nimport { callNumber } from '../lib/phone/call'\n\nexport const CallEvents: FC<{ sipHost: string }> = ({ sipHost }) => {\n  const { sipcall }: any = useSelector((state: RootState) => state.webrtc)\n  /**\n   * Event listner for phone-island-call-start event\n   */\n  useEventListener('phone-island-call-start', (data: CallStartTypes) => {\n    const number = data.number.replace(/\\s/g, '')\n    callNumber(number, sipHost)\n  })\n  return <></>\n}\n\n// !TODO add phone-island-outgoing-call-started phone-island-outgoing-call-ended\n\n/**\n * Dispatch the phone-island-outgoing-call-started event\n */\nexport function dispatchOutgoingCallStarted(name: string = '', number: string = '') {\n  const data: OutgoingCallStartedTypes = {\n    name,\n    number,\n  }\n  eventDispatch('phone-island-outgoing-call-started', data)\n}\n\nexport interface OutgoingCallStartedTypes {\n  name: string\n  number: string\n}\n\ninterface CallStartTypes {\n  number: string\n}\n"],"names":["_a","sipHost","useSelector","state","webrtc","useEventListener","data","number","replace","callNumber","React","name","eventDispatch"],"mappings":"sqBASmD,SAACA,GAAE,IAAAC,EAAOD,EAAAC,QAS3D,OARyBC,EAAWA,aAAC,SAACC,GAAqB,OAAAA,EAAMC,MAAM,YAIvEC,mBAAiB,2BAA2B,SAACC,GAC3C,IAAMC,EAASD,EAAKC,OAAOC,QAAQ,MAAO,IAC1CC,aAAWF,EAAQN,EACrB,IACOS,gDACT,sCAOgB,SAA4BC,EAAmBJ,QAAnB,IAAAI,IAAAA,EAAiB,SAAE,IAAAJ,IAAAA,EAAmB,IAChF,IAAMD,EAAiC,CACrCK,KAAIA,EACJJ,OAAMA,GAERK,gBAAc,qCAAsCN,EACtD"}

package/dist/lib/phone/call.d.ts

@@ -2,7 +2,7 @@
  * Starts a call to a number
  * @param number The number string
  */
-export declare function callNumber(number: string): void;
+export declare function callNumber(number: string, sipHost: string): void;
 /**
  * Starts a call to a SIP URI
  *

package/dist/lib/phone/call.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../../node_modules/tslib/tslib.es6.js"),t=require("../devices/devices.js"),r=require("../webrtc/janus.js"),a=require("../webrtc/messages.js"),n=require("../../store/index.js"),o=require("../user/default_device.js"),s=require("../../services/astproxy.js"),u=require("../../static/dtmf/index.js");function i(n){var o=this;t.getSupportedDevices((function(){return e.__awaiter(o,void 0,void 0,(function(){return e.__generator(this,(function(e){switch(e.label){case 0:return r.default.log("This is a SIP call"),[4,a.call(n,{audio:{mandatory:{echoCancellation:!1,googEchoCancellation:!1,googAutoGainControl:!1,googAutoGainControl2:!1,googNoiseSuppression:!1,googHighpassFilter:!1,googTypingNoiseDetection:!1}},audioSend:!0,audioRecv:!0,videoSend:!1,videoRecv:!1})];case 1:return e.sent(),[2]}}))}))}))}exports.answerIncomingCall=function(){o.isWebRTC()&&a.answerWebRTC()},exports.attendedTransfer=function(t){return e.__awaiter(this,void 0,void 0,(function(){var r,a;return e.__generator(this,(function(e){switch(e.label){case 0:return r=n.store.getState().currentCall.conversationId,a=n.store.getState().currentUser.default_device,r&&(null==a?void 0:a.id)&&t?[4,s.attendedTransfer({convid:r,to:t,endpointId:a.id})]:[3,2];case 1:return[2,e.sent()];case 2:return[2]}}))}))},exports.callNumber=function(e){i("sip:".concat(e,"@").concat("127.0.0.1"))},exports.callSipURI=i,exports.hangupCurrentCall=function(){var e=n.store.getState().currentCall,t=e.outgoing,r=e.accepted;t||r?a.hangup():a.decline(),n.store.dispatch.player.stopAudioPlayer(),n.store.dispatch.currentCall.reset()},exports.muteCurrentCall=function(){o.isWebRTC()&&a.muteWebRTC()&&n.store.dispatch.currentCall.updateCurrentCall({muted:!0})},exports.pauseCurrentCall=function(){o.isWebRTC()&&a.pauseWebRTC()&&(n.store.dispatch.currentCall.updateCurrentCall({paused:!0}),n.store.dispatch.player.pauseRemoteAudio())},exports.playDtmfAudio=function(e){"*"===e&&(e="star"),"#"===e&&(e="pound"),n.store.dispatch.player.updateStartAudioPlayer({src:u.default["dtmf_".concat(e)]})},exports.unmuteCurrentCall=function(){o.isWebRTC()&&a.unmuteWebRTC()&&n.store.dispatch.currentCall.updateCurrentCall({muted:!1})},exports.unpauseCurrentCall=function(){o.isWebRTC()&&a.unpauseWebRTC()&&(n.store.dispatch.currentCall.updateCurrentCall({paused:!1}),n.store.dispatch.player.playRemoteAudio())};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../../node_modules/tslib/tslib.es6.js"),t=require("../devices/devices.js"),r=require("../webrtc/janus.js"),a=require("../webrtc/messages.js"),n=require("../../store/index.js"),o=require("../user/default_device.js"),s=require("../../services/astproxy.js"),u=require("../../static/dtmf/index.js");function i(n){var o=this;t.getSupportedDevices((function(){return e.__awaiter(o,void 0,void 0,(function(){return e.__generator(this,(function(e){switch(e.label){case 0:return r.default.log("This is a SIP call"),[4,a.call(n,{audio:{mandatory:{echoCancellation:!1,googEchoCancellation:!1,googAutoGainControl:!1,googAutoGainControl2:!1,googNoiseSuppression:!1,googHighpassFilter:!1,googTypingNoiseDetection:!1}},audioSend:!0,audioRecv:!0,videoSend:!1,videoRecv:!1})];case 1:return e.sent(),[2]}}))}))}))}exports.answerIncomingCall=function(){o.isWebRTC()&&a.answerWebRTC()},exports.attendedTransfer=function(t){return e.__awaiter(this,void 0,void 0,(function(){var r,a;return e.__generator(this,(function(e){switch(e.label){case 0:return r=n.store.getState().currentCall.conversationId,a=n.store.getState().currentUser.default_device,r&&(null==a?void 0:a.id)&&t?[4,s.attendedTransfer({convid:r,to:t,endpointId:a.id})]:[3,2];case 1:return[2,e.sent()];case 2:return[2]}}))}))},exports.callNumber=function(e,t){i("sip:".concat(e,"@").concat(t))},exports.callSipURI=i,exports.hangupCurrentCall=function(){var e=n.store.getState().currentCall,t=e.outgoing,r=e.accepted;t||r?a.hangup():a.decline(),n.store.dispatch.player.stopAudioPlayer(),n.store.dispatch.currentCall.reset()},exports.muteCurrentCall=function(){o.isWebRTC()&&a.muteWebRTC()&&n.store.dispatch.currentCall.updateCurrentCall({muted:!0})},exports.pauseCurrentCall=function(){o.isWebRTC()&&a.pauseWebRTC()&&(n.store.dispatch.currentCall.updateCurrentCall({paused:!0}),n.store.dispatch.player.pauseRemoteAudio())},exports.playDtmfAudio=function(e){"*"===e&&(e="star"),"#"===e&&(e="pound"),n.store.dispatch.player.updateStartAudioPlayer({src:u.default["dtmf_".concat(e)]})},exports.unmuteCurrentCall=function(){o.isWebRTC()&&a.unmuteWebRTC()&&n.store.dispatch.currentCall.updateCurrentCall({muted:!1})},exports.unpauseCurrentCall=function(){o.isWebRTC()&&a.unpauseWebRTC()&&(n.store.dispatch.currentCall.updateCurrentCall({paused:!1}),n.store.dispatch.player.playRemoteAudio())};
 //# sourceMappingURL=call.js.map

package/dist/lib/phone/call.js.map

@@ -1 +1 @@
-{"version":3,"file":"call.js","sources":["../../../src/lib/phone/call.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport { getSupportedDevices } from '../devices/devices'\nimport Janus from '../webrtc/janus'\nimport {\n  call,\n  hangup,\n  decline,\n  answerWebRTC,\n  muteWebRTC,\n  unmuteWebRTC,\n  pauseWebRTC,\n  unpauseWebRTC,\n} from '../webrtc/messages'\nimport { store } from '../../store'\nimport { isWebRTC } from '../user/default_device'\nimport {\n  blindTransfer as blindTransferRequest,\n  attendedTransfer as attendedTransferRequest,\n} from '../../services/astproxy'\nimport dtmfAudios from '../../static/dtmf'\n\n/**\n * Starts a call to a number\n * @param number The number string\n */\nexport function callNumber(number: string) {\n  const BASE_HOST_URL: string = '127.0.0.1'\n  const sipURI = `sip:${number}@${BASE_HOST_URL}`\n  callSipURI(sipURI)\n}\n\n/**\n * Starts a call to a SIP URI\n *\n * @param 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store.dispatch.player.stopAudioPlayer()\n  store.dispatch.currentCall.reset()\n}\n\n/**\n * Mute the current call\n */\nexport function muteCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const muted = muteWebRTC()\n    if (muted) {\n      store.dispatch.currentCall.updateCurrentCall({\n        muted: true,\n      })\n    }\n  }\n}\n\n/**\n * Unmute the current call\n */\nexport function unmuteCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const unmuted = unmuteWebRTC()\n    if (unmuted) {\n      store.dispatch.currentCall.updateCurrentCall({\n        muted: false,\n      })\n    }\n  }\n}\n\n/**\n * Pause the current call\n */\nexport function pauseCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const paused = pauseWebRTC()\n    if (paused) {\n      store.dispatch.currentCall.updateCurrentCall({\n        paused: true,\n      })\n      // Pause remote audio\n      store.dispatch.player.pauseRemoteAudio()\n    }\n  }\n}\n\n/**\n * Unpause the current call\n */\nexport function unpauseCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const unpaused = unpauseWebRTC()\n    if (unpaused) {\n      store.dispatch.currentCall.updateCurrentCall({\n        paused: false,\n      })\n      // Play remote audio\n      store.dispatch.player.playRemoteAudio()\n    }\n  }\n}\n\n/**\n * Transfer the current call through a blind transfer (not in use)\n */\nexport async function blindTransfer(number: string) {\n  // Retrieve current conversation info\n  const { conversationId } = store.getState().currentCall\n  const { default_device } = store.getState().currentUser\n  // Transfer the call through blind transfer\n  if (conversationId && default_device?.id && number) {\n    return await blindTransferRequest({\n      convid: conversationId,\n      to: number,\n      endpointId: default_device.id,\n    })\n  }\n}\n\n/**\n * 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+{"version":3,"file":"call.js","sources":["../../../src/lib/phone/call.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport { getSupportedDevices } from '../devices/devices'\nimport Janus from '../webrtc/janus'\nimport {\n  call,\n  hangup,\n  decline,\n  answerWebRTC,\n  muteWebRTC,\n  unmuteWebRTC,\n  pauseWebRTC,\n  unpauseWebRTC,\n} from '../webrtc/messages'\nimport { store } from '../../store'\nimport { isWebRTC } from '../user/default_device'\nimport {\n  blindTransfer as blindTransferRequest,\n  attendedTransfer as attendedTransferRequest,\n} from '../../services/astproxy'\nimport dtmfAudios from '../../static/dtmf'\n\n/**\n * Starts a call to a number\n * @param number The number string\n */\nexport function callNumber(number: string, sipHost: string) {\n  const sipURI = `sip:${number}@${sipHost}`\n  callSipURI(sipURI)\n}\n\n/**\n * Starts a call to a SIP URI\n *\n * @param sipURI The SIP URI string\n */\nexport 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store.dispatch.currentCall.reset()\n}\n\n/**\n * Mute the current call\n */\nexport function muteCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const muted = muteWebRTC()\n    if (muted) {\n      store.dispatch.currentCall.updateCurrentCall({\n        muted: true,\n      })\n    }\n  }\n}\n\n/**\n * Unmute the current call\n */\nexport function unmuteCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const unmuted = unmuteWebRTC()\n    if (unmuted) {\n      store.dispatch.currentCall.updateCurrentCall({\n        muted: false,\n      })\n    }\n  }\n}\n\n/**\n * Pause the current call\n */\nexport function pauseCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const paused = pauseWebRTC()\n    if (paused) {\n      store.dispatch.currentCall.updateCurrentCall({\n        paused: true,\n      })\n      // Pause remote audio\n      store.dispatch.player.pauseRemoteAudio()\n    }\n  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package/dist/lib/webrtc/messages.d.ts

@@ -1,4 +1,9 @@
-export declare function register(sipExten: string, sipSecret: string): void;
+export declare function register({ sipExten, sipSecret, sipHost, sipPort, }: {
+    sipExten: string;
+    sipSecret: string;
+    sipHost: string;
+    sipPort: string;
+}): void;
 export declare function answerWebRTC(): void;
 export declare function decline(): void;
 export declare function hangup(): void;

package/dist/lib/webrtc/messages.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("./janus.js"),r=require("../../store/index.js"),t=require("../../node_modules/webrtc-adapter/src/js/adapter_core.js");exports.answerWebRTC=function(){var t=r.store.getState().webrtc,s=t.sipcall,a=t.jsepGlobal;s&&a&&s.createAnswer({jsep:a,media:{audio:!0,videoSend:!1,videoRecv:!1},success:function(e){s.send({message:{request:"accept"},jsep:e})},error:function(r){e.default.error("WebRTC error:",r),s.send({message:{request:"decline",code:480}})}})},exports.call=function(t,s){return new Promise((function(a,n){var o=r.store.getState().webrtc.sipcall;t&&s&&o.createOffer({media:s,success:function(r){e.default.debug("Got SDP!"),e.default.debug(r),o.send({message:{request:"call",uri:t},jsep:r}),a(!0)},error:function(r){e.default.error("ebRTC error...",r),e.default.error("WebRTC error call on createOffer: ",r),n(!1)}})}))},exports.decline=function(){var e=r.store.getState().webrtc.sipcall;e&&e.send({message:{request:"decline"}})},exports.handleRemote=function(e){var t=r.store.getState().webrtc.sipcall;t&&t.handleRemoteJsep({jsep:e,error:function(){t.send({message:{request:"hangup"}}),t.hangup()}})},exports.hangup=function(){var e=r.store.getState().webrtc.sipcall;e&&e.send({message:{request:"hangup"}})},exports.muteWebRTC=function(){var e=r.store.getState().webrtc.sipcall;return e.muteAudio(),e.isAudioMuted()},exports.pauseWebRTC=function(){var e=r.store.getState().webrtc.sipcall;try{return e.send({message:{request:"hold"}}),!0}catch(e){return console.error(e),!1}},exports.register=function(e,t){var s=r.store.getState().webrtc.sipcall,a=r.store.getState().currentUser.name;s&&s.send({message:{request:"register",username:"sip:"+e+"@127.0.0.1",display_name:a||"",secret:t,proxy:"sip:127.0.0.1:5060",sips:!1,refresh:!1}})},exports.sendDTMF=function(e){var s=r.store.getState().webrtc.sipcall;"chrome"===t.default.browserDetails.browser?s.dtmf({dtmf:{tones:e}}):s.send({message:{request:"dtmf_info",digit:e}})},exports.unmuteWebRTC=function(){var e=r.store.getState().webrtc.sipcall;return e.unmuteAudio(),!e.isAudioMuted()},exports.unpauseWebRTC=function(){var e=r.store.getState().webrtc.sipcall;try{return e.send({message:{request:"unhold"}}),!0}catch(e){return console.error(e),!1}},exports.unregister=function(){var e=r.store.getState().webrtc.sipcall;e&&e.send({message:{request:"unregister"}})};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("./janus.js"),t=require("../../store/index.js"),r=require("../../node_modules/webrtc-adapter/src/js/adapter_core.js");exports.answerWebRTC=function(){var r=t.store.getState().webrtc,s=r.sipcall,o=r.jsepGlobal;s&&o&&s.createAnswer({jsep:o,media:{audio:!0,videoSend:!1,videoRecv:!1},success:function(e){s.send({message:{request:"accept"},jsep:e})},error:function(t){e.default.error("WebRTC error:",t),s.send({message:{request:"decline",code:480}})}})},exports.call=function(r,s){return new Promise((function(o,n){var a=t.store.getState().webrtc.sipcall;r&&s&&a.createOffer({media:s,success:function(t){e.default.debug("Got SDP!"),e.default.debug(t),a.send({message:{request:"call",uri:r},jsep:t}),o(!0)},error:function(t){e.default.error("ebRTC error...",t),e.default.error("WebRTC error call on createOffer: ",t),n(!1)}})}))},exports.decline=function(){var e=t.store.getState().webrtc.sipcall;e&&e.send({message:{request:"decline"}})},exports.handleRemote=function(e){var r=t.store.getState().webrtc.sipcall;r&&r.handleRemoteJsep({jsep:e,error:function(){r.send({message:{request:"hangup"}}),r.hangup()}})},exports.hangup=function(){var e=t.store.getState().webrtc.sipcall;e&&e.send({message:{request:"hangup"}})},exports.muteWebRTC=function(){var e=t.store.getState().webrtc.sipcall;return e.muteAudio(),e.isAudioMuted()},exports.pauseWebRTC=function(){var e=t.store.getState().webrtc.sipcall;try{return e.send({message:{request:"hold"}}),!0}catch(e){return console.error(e),!1}},exports.register=function(e){var r=e.sipExten,s=e.sipSecret,o=e.sipHost,n=e.sipPort,a=t.store.getState().webrtc.sipcall,c=t.store.getState().currentUser.name;a&&a.send({message:{request:"register",username:"sip:".concat(r,"@").concat(o),display_name:c||"",secret:s,proxy:"sip:".concat(o,":").concat(n),outbound_proxy:"sip:".concat(o,":").concat(n),sips:!1,refresh:!1}})},exports.sendDTMF=function(e){var s=t.store.getState().webrtc.sipcall;"chrome"===r.default.browserDetails.browser?s.dtmf({dtmf:{tones:e}}):s.send({message:{request:"dtmf_info",digit:e}})},exports.unmuteWebRTC=function(){var e=t.store.getState().webrtc.sipcall;return e.unmuteAudio(),!e.isAudioMuted()},exports.unpauseWebRTC=function(){var e=t.store.getState().webrtc.sipcall;try{return e.send({message:{request:"unhold"}}),!0}catch(e){return console.error(e),!1}},exports.unregister=function(){var e=t.store.getState().webrtc.sipcall;e&&e.send({message:{request:"unregister"}})};
 //# sourceMappingURL=messages.js.map

package/dist/lib/webrtc/messages.js.map

@@ -1 +1 @@
-{"version":3,"file":"messages.js","sources":["../../../src/lib/webrtc/messages.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport Janus from './janus'\nimport { store } from '../../store'\nimport adapter from 'webrtc-adapter'\n\nexport function register(sipExten: string, sipSecret: string) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  const { name } = store.getState().currentUser\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'register',\n        username: 'sip:' + sipExten + '@' + '127.0.0.1',\n        display_name: name || '',\n        secret: sipSecret,\n        proxy: 'sip:' + '127.0.0.1' + ':5060',\n        sips: false,\n        refresh: false,\n      },\n    })\n  }\n}\n\nexport function answerWebRTC() {\n  const { sipcall, jsepGlobal }: { sipcall: any; jsepGlobal: any } = store.getState().webrtc\n  if (sipcall && jsepGlobal) {\n    sipcall.createAnswer({\n      jsep: jsepGlobal,\n      media: {\n        audio: true,\n        videoSend: false,\n        videoRecv: false,\n      },\n      success: (jsep) => {\n        sipcall.send({\n          message: {\n            request: 'accept',\n          },\n          jsep: jsep,\n        })\n      },\n      error: (error) => {\n        // @ts-ignore\n        Janus.error('WebRTC error:', error)\n        sipcall.send({\n          message: {\n            request: 'decline',\n            code: 480,\n          },\n        })\n      },\n    })\n  }\n}\n\nexport function decline() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'decline',\n      },\n    })\n  }\n}\n\nexport function hangup() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'hangup',\n      },\n    })\n  }\n}\n\nexport function unregister() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'unregister',\n      },\n    })\n  }\n}\n\nexport function handleRemote(jsep) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.handleRemoteJsep({\n      jsep: jsep,\n      error: function () {\n        var hangup = {\n          request: 'hangup',\n        }\n        sipcall.send({\n          message: hangup,\n        })\n        sipcall.hangup()\n      },\n    })\n  }\n}\n\n/**\n * Starts a call to a sipuri\n *\n * @param sipURI The sipuri to be called\n * @param mediaObj The object containing the media constrains\n * @returns\n */\nexport function call(sipURI: string, mediaObj: object) {\n  return new Promise((resolve, reject) => {\n    const { sipcall }: { sipcall: any } = store.getState().webrtc\n    if (sipURI && mediaObj) {\n      sipcall.createOffer({\n        media: mediaObj,\n        success: function (jsep: any) {\n          // @ts-ignore\n          Janus.debug('Got SDP!')\n          // @ts-ignore\n          Janus.debug(jsep)\n          sipcall.send({\n            message: {\n              request: 'call',\n              uri: sipURI,\n            },\n            jsep: jsep,\n          })\n          resolve(true)\n        },\n        error: function (error) {\n          // @ts-ignore\n          Janus.error('ebRTC error...', error)\n          // @ts-ignore\n          Janus.error('WebRTC error call on createOffer: ', error)\n          reject(false)\n        },\n      })\n    }\n  })\n}\n\n/**\n * Mute current call so the counterpart can't listen the current user\n * @returns The muted status\n */\nexport function muteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Uset the janus library functions to mute call\n  sipcall.muteAudio()\n  return sipcall.isAudioMuted()\n}\n\n/**\n * Unmute current call so the counterpart can listen the current user\n * @returns The muted status\n */\nexport function unmuteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Use the janus library functions to unmute call\n  sipcall.unmuteAudio()\n  return !sipcall.isAudioMuted()\n}\n\n/**\n * Pause current call so the counterpart listens the pause ringtone\n * @returns The true if no errors occurs\n */\nexport function pauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'hold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Unpause the current call so the counterpart listens the current user\n * @returns The true if no errors occurs\n */\nexport function unpauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'unhold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Send DTMF messages to Janus\n */\nexport function sendDTMF(key: string) {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (adapter.browserDetails.browser === 'chrome') {\n    sipcall.dtmf({\n      dtmf: {\n        tones: key,\n      },\n    })\n  } else {\n    sipcall.send({\n      message: {\n        request: 'dtmf_info',\n        digit: key,\n      },\n    })\n  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+{"version":3,"file":"messages.js","sources":["../../../src/lib/webrtc/messages.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport Janus from './janus'\nimport { store } from '../../store'\nimport adapter from 'webrtc-adapter'\n\nexport function register({\n  sipExten,\n  sipSecret,\n  sipHost,\n  sipPort,\n}: {\n  sipExten: string\n  sipSecret: string\n  sipHost: string\n  sipPort: string\n}) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  const { name } = store.getState().currentUser\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'register',\n        username: `sip:${sipExten}@${sipHost}`,\n        display_name: name || '',\n        secret: sipSecret,\n        proxy: `sip:${sipHost}:${sipPort}`,\n        outbound_proxy: `sip:${sipHost}:${sipPort}`,\n        sips: false,\n        refresh: false,\n      },\n    })\n  }\n}\n\nexport function answerWebRTC() {\n  const { sipcall, jsepGlobal }: { sipcall: any; jsepGlobal: any } = store.getState().webrtc\n  if (sipcall && jsepGlobal) {\n    sipcall.createAnswer({\n      jsep: jsepGlobal,\n      media: {\n        audio: true,\n        videoSend: false,\n        videoRecv: false,\n      },\n      success: (jsep) => {\n        sipcall.send({\n          message: {\n            request: 'accept',\n          },\n          jsep: jsep,\n        })\n      },\n      error: (error) => {\n        // @ts-ignore\n        Janus.error('WebRTC error:', error)\n        sipcall.send({\n          message: {\n            request: 'decline',\n            code: 480,\n          },\n        })\n      },\n    })\n  }\n}\n\nexport function decline() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'decline',\n      },\n    })\n  }\n}\n\nexport function hangup() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'hangup',\n      },\n    })\n  }\n}\n\nexport function unregister() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'unregister',\n      },\n    })\n  }\n}\n\nexport function handleRemote(jsep) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.handleRemoteJsep({\n      jsep: jsep,\n      error: function () {\n        var hangup = {\n          request: 'hangup',\n        }\n        sipcall.send({\n          message: hangup,\n        })\n        sipcall.hangup()\n      },\n    })\n  }\n}\n\n/**\n * Starts a call to a sipuri\n *\n * @param sipURI The sipuri to be called\n * @param mediaObj The object containing the media constrains\n * @returns\n */\nexport function call(sipURI: string, mediaObj: object) {\n  return new Promise((resolve, reject) => {\n    const { sipcall }: { sipcall: any } = store.getState().webrtc\n    if (sipURI && mediaObj) {\n      sipcall.createOffer({\n        media: mediaObj,\n        success: function (jsep: any) {\n          // @ts-ignore\n          Janus.debug('Got SDP!')\n          // @ts-ignore\n          Janus.debug(jsep)\n          sipcall.send({\n            message: {\n              request: 'call',\n              uri: sipURI,\n            },\n            jsep: jsep,\n          })\n          resolve(true)\n        },\n        error: function (error) {\n          // @ts-ignore\n          Janus.error('ebRTC error...', error)\n          // @ts-ignore\n          Janus.error('WebRTC error call on createOffer: ', error)\n          reject(false)\n        },\n      })\n    }\n  })\n}\n\n/**\n * Mute current call so the counterpart can't listen the current user\n * @returns The muted status\n */\nexport function muteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Uset the janus library functions to mute call\n  sipcall.muteAudio()\n  return sipcall.isAudioMuted()\n}\n\n/**\n * Unmute current call so the counterpart can listen the current user\n * @returns The muted status\n */\nexport function unmuteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Use the janus library functions to unmute call\n  sipcall.unmuteAudio()\n  return !sipcall.isAudioMuted()\n}\n\n/**\n * Pause current call so the counterpart listens the pause ringtone\n * @returns The true if no errors occurs\n */\nexport function pauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'hold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Unpause the current call so the counterpart listens the current user\n * 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package/package.json

@@ -1,7 +1,7 @@
 {
   "name": "@nethesis/phone-island",
   "author": "Nethesis",
-  "version": "0.7.79",
+  "version": "0.7.81",
   "description": "NethVoice CTI Phone Island",
   "keywords": [
     "nethserver",