chore(2.2.15-beta.0): version updated to 2.2.15 beta.0

plivo · Sep 20, 2024 · 6499d24 · 6499d24
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diff --git a/CHANGELOG.md b/CHANGELOG.md
@@ -5,6 +5,26 @@ All notable GA release changes to this project will be documented in this file.
 The format is based on [Keep a Changelog](http://keepachangelog.com/en/1.0.0/)
 and this project adheres to [Semantic Versioning](http://semver.org/spec/v2.0.0.html).
 
+## v2.2.15-beta.0 (released@ 20-09-2024)
+
+**Bug Fixes**
+* Fixed the error occurring when answer() and hangup() are called simultaneously.
+
+## v2.2.14 (released@ 19-09-2024)
+
+**Feature**
+* A new event named `onCallConnected` has been introduced, which is triggered when the PSTN callee starts ringing.
+
+## v2.2.13 (released@ 22-08-2024)
+
+**Bug Fixes**
+* Removed unnecessary dependency.
+
+## v2.2.12 (released@ 24-07-2024)
+
+**Bug Fixes**
+* Renamed `DOMError` to `DOMException` in the underlying JSSIP library to support latest Typescript versions.
+
 ## v2.2.12-beta.0 (released@ 08-07-2024)
 
 **Features**

diff --git a/index.d.ts b/index.d.ts
@@ -150,7 +150,7 @@ declare module 'plivo-browser-sdk/client' {
                 */
             timeDiff: number;
             /**
-                * Holds the incoming or outgoing JSSIP RTCSession(WebRTC media session)
+                * Holds the incoming or outgoing JsSIP RTCSession(WebRTC media session)
                 * @private
                 */
             callSession: null | RTCSession;
@@ -169,7 +169,7 @@ declare module 'plivo-browser-sdk/client' {
                 * speech when the user speaks on mute
                 * @private
                 */
-            speechRecognition: SpeechRecognition;
+            speechRecognition: any;
             /**
                 * Holds the loggerUtil instance which keeps the
                 * value of username and sipCallID to attached to each log
@@ -369,7 +369,7 @@ declare module 'plivo-browser-sdk/client' {
                 */
             statsSocket: null | StatsSocket;
             /**
-                * Contains available audio devices.This is done for backward compatiblity
+                * Contains available audio devices.This is done for backward compatibility
                 * @private
                 */
             audioDevDic: any;
@@ -834,7 +834,7 @@ declare module 'plivo-browser-sdk/managers/callSession' {
                 */
             stirShakenState: string;
             /**
-                * Identifier generated by JSSIP when a new RTCSession is created for the call
+                * Identifier generated by JsSIP when a new RTCSession is created for the call
                 * @private
                 */
             sipCallID: string | null;
@@ -864,7 +864,7 @@ declare module 'plivo-browser-sdk/managers/callSession' {
                 */
             isCallTerminatedDuringRinging: boolean;
             /**
-                * Holds the current status of speechrecgnition
+                * Holds the current status of speech recognition
                 * @private
                 */
             speech_state: string;
@@ -1256,7 +1256,7 @@ declare module 'plivo-browser-sdk/managers/workerManager' {
         /**
          * Start the network check timer.
          * @param {number} networkCheckInterval - time interval at which the timer is to be executed.
-         * @param {any} callback - callback to be trigerred when the timer executes.
+         * @param {any} callback - callback to be triggered when the timer executes.
          */
         startNetworkCheckTimer: (networkCheckInterval: number, callback: any, responseCallback: any) => void;
         /**
@@ -1633,12 +1633,12 @@ declare module 'plivo-browser-sdk/stats/rtpStats' {
                 */
             xcallUUID: string;
             /**
-                * Identifier generated by JSSIP when a new RTCSession is created for the call
+                * Identifier generated by JsSIP when a new RTCSession is created for the call
                 * @private
                 */
             callUUID: string;
             /**
-                * Identifier generated by JSSIP when a new RTCSession is created for the call
+                * Identifier generated by JsSIP when a new RTCSession is created for the call
                 * @private
                 */
             corelationId: string;
@@ -1763,6 +1763,7 @@ declare module 'plivo-browser-sdk/stats/nonRTPStats' {
             sdkVersionMajor: number;
             sdkVersionMinor: number;
             sdkVersionPatch: number;
+            sdkVersionPre: any;
             devicePlatform: string;
             deviceOs: string;
             setupOptions: ConfiguationOptions;
@@ -1791,6 +1792,7 @@ declare module 'plivo-browser-sdk/stats/nonRTPStats' {
             sdkVersionMajor: number;
             sdkVersionMinor: number;
             sdkVersionPatch: number;
+            sdkVersionPre: any;
             clientName: string;
             devicePlatform: string;
             deviceOs: string;
@@ -1813,6 +1815,7 @@ declare module 'plivo-browser-sdk/stats/nonRTPStats' {
             sdkVersionMajor: number;
             sdkVersionMinor: number;
             sdkVersionPatch: number;
+            sdkVersionPre: any;
             clientName: string;
             devicePlatform: string;
             deviceOs: string;
@@ -1905,11 +1908,11 @@ declare module 'plivo-browser-sdk/stats/nonRTPStats' {
 
 declare module 'plivo-browser-sdk/media/audioLevel' {
     /**
-        * Analyse the audio level for a stream(local/remote).
+        * Analyze the audio level for a stream(local/remote).
         */
     export class AudioLevel {
             /**
-                * Audio volume in decibles
+                * Audio volume in decibels
                 * @private
                 */
             volumeLevel: number;
@@ -1934,7 +1937,7 @@ declare module 'plivo-browser-sdk/media/audioLevel' {
                 */
             sourceNode: MediaStreamAudioSourceNode;
             /**
-                * Get audio level in decibles.
+                * Get audio level in decibels.
                 */
             getAudioLevel: () => number;
             /**

diff --git a/lib/client.ts b/lib/client.ts
@@ -198,7 +198,7 @@ export class Client extends EventEmitter {
   timeDiff: number;
 
   /**
-   * Holds the incoming or outgoing JSSIP RTCSession(WebRTC media session)
+   * Holds the incoming or outgoing JsSIP RTCSession(WebRTC media session)
    * @private
    */
   callSession: null | RTCSession;
@@ -220,7 +220,7 @@ export class Client extends EventEmitter {
    * speech when the user speaks on mute
    * @private
    */
-  speechRecognition: SpeechRecognition;
+  speechRecognition: any;
 
   /**
    * Holds the loggerUtil instance which keeps the
@@ -463,7 +463,7 @@ export class Client extends EventEmitter {
   statsSocket: null | StatsSocket;
 
   /**
-   * Contains available audio devices.This is done for backward compatiblity
+   * Contains available audio devices.This is done for backward compatibility
    * @private
    */
   audioDevDic: any;
@@ -910,6 +910,7 @@ export class Client extends EventEmitter {
       captureSDKCrashOnly: _options.captureSDKCrashOnly,
       permOnClick: _options.permOnClick,
       stopAutoRegisterOnConnect: _options.stopAutoRegisterOnConnect,
+      registrationRefreshTimer: _options.registrationRefreshTimer,
     };
     Plivo.log.info(`${C.LOGCAT.INIT} | Plivo SDK initialized successfully with options:- `, JSON.stringify(data), `in ${Plivo.log.level()} mode`);
     // instantiates event emitter
@@ -1525,7 +1526,7 @@ export class Client extends EventEmitter {
     this.loggerUtil.setSipCallID(incomingCall.sipCallID ?? "");
     Plivo.log.debug(`${LOGCAT.CALL} | reject - ${incomingCall.callUUID}`);
     if (incomingCall.session && incomingCall.session.isEstablished()) {
-      Plivo.log.warn(`${LOGCAT.CALL} |call already answerd, please use hangup() method`);
+      Plivo.log.warn(`${LOGCAT.CALL} | Call already answered, please use hangup() method`);
       return false;
     }
     if (incomingCall) {

diff --git a/lib/managers/account.ts b/lib/managers/account.ts
@@ -441,7 +441,7 @@ class Account {
     // NOTE: This event is not documented by JsSIP.
     // Should be used to just record the timestamp of invite received only
     // Do not have any other logic here
-    // Invite Server Trasaction(ist) gives us the incoming invite timestamp.
+    // Invite Server Transaction(ist) gives us the incoming invite timestamp.
     if (evt.transaction.type === 'ist') {
       const callID = evt.transaction.request.getHeader('Call-ID') ?? "";
       Plivo.log.info(`${C.LOGCAT.LOGIN} | new Transaction created, incoming call received callUUID: ${callID}`);
@@ -576,7 +576,7 @@ class Account {
           this._tiggerNetworkChangeEvent(fetchIpCount);
         }, fetchIpCount * 200);
       } else {
-        Plivo.log.warn(`${C.LOGCAT.NETWORK_CHANGE} | Could not retreive ipaddress`);
+        Plivo.log.warn(`${C.LOGCAT.NETWORK_CHANGE} | Could not retrieve IP address`);
         fetchIpCount = 0;
       }
     });

diff --git a/lib/managers/callSession.ts b/lib/managers/callSession.ts
@@ -135,7 +135,7 @@ export class CallSession {
   stirShakenState: string;
 
   /**
-   * Identifier generated by JSSIP when a new RTCSession is created for the call
+   * Identifier generated by JsSIP when a new RTCSession is created for the call
    * @private
    */
   sipCallID: string | null;
@@ -171,7 +171,7 @@ export class CallSession {
   isCallTerminatedDuringRinging: boolean;
 
   /**
-   * Holds the current status of speechrecgnition
+   * Holds the current status of speech recognition
    * @private
    */
   speech_state: string;

diff --git a/lib/managers/incomingCall.ts b/lib/managers/incomingCall.ts
@@ -98,7 +98,8 @@ const updateSessionInfo = (evt: UserAgentNewRtcSessionEvent, call: CallSession):
   if (call.callUUID) {
     cs.incomingInvites.set(call.callUUID, call);
     cs.lastIncomingCall = call;
-    // Storing these info for backward compatiblity, in case user used these non-document variables.
+    // Storing these info for backward compatibility,
+    // In case user used these non-document variables.
     if (!cs.options.allowMultipleIncomingCalls) {
       cs.callSession = call.session;
       cs.callUUID = call.callUUID;
@@ -345,6 +346,9 @@ const newDTMF = (evt: SessionNewDtmfEvent): void => {
  */
 const newInfo = (evt: SessionNewInfoEvent): void => {
   Plivo.log.info(`${LOGCAT.CALL} | Incoming Call | ${evt.originator} INFO packet with body : ${evt.info.body}`);
+  if (cs._currentSession && evt.info.body === "remote-party-ringing") {
+    cs.emit('onCallConnected', cs._currentSession.getCallInfo(evt.originator));
+  }
   if (evt.info.body === 'no-remote-audio') {
     cs.emit('remoteAudioStatus', false);
   }
@@ -593,7 +597,11 @@ export const answerIncomingCall = function (
       Plivo.log.debug(`${LOGCAT.CALL} | CallUUID is ${cs.callUUID}`);
       cs.callDirection = cs._currentSession.direction;
       getAnswerOptions().then((options) => {
-        curIncomingCall.session.answer(options);
+        if (cs.getCurrentSession()) {
+          curIncomingCall.session.answer(options);
+        } else {
+          Plivo.log.info(`${LOGCAT.CALL} | No Call session exists to answer the incoming call`);
+        }
       });
     } catch (err) {
       Plivo.log.error(`${LOGCAT.CALL} | error in answering incoming call : `, err.message);

diff --git a/lib/managers/outgoingCall.ts b/lib/managers/outgoingCall.ts
@@ -207,7 +207,7 @@ const onSDP = (evt: SessionSdpEvent): void => {
  * @param {SessionProgressEvent} evt - rtcsession progress information
  */
 const handleProgressTone = (evt: SessionProgressEvent): void => {
-  Plivo.log.debug(`${LOGCAT.CALL} | ringback tone enabled : ${cs.ringToneBackFlag} audioPaused: ${cs.connectToneView.paused}`);
+  Plivo.log.debug(`${LOGCAT.CALL} | Ringback Tone Enabled : ${cs.ringToneBackFlag} | Audio Paused: ${cs.connectToneView.paused}`);
   if (!cs.connectToneView.paused) {
     stopAudio(CONNECT_TONE_ELEMENT_ID);
   }
@@ -237,7 +237,6 @@ const OnProgress = (evt: SessionProgressEvent): void => {
     cs._currentSession.onRinging(cs);
     const callUUID = evt.response.getHeader('X-Calluuid');
     cs._currentSession.setCallUUID(callUUID);
-    Plivo.log.info(`${LOGCAT.CALL} | Outgoing call Ringing`);
     cs._currentSession.setState(cs._currentSession.STATE.RINGING);
     cs.callUUID = callUUID;
     Plivo.log.info(`${LOGCAT.CALL} | Outgoing call Ringing, CallUUID: ${cs.callUUID}`);
@@ -265,7 +264,7 @@ const OnProgress = (evt: SessionProgressEvent): void => {
       cs.mute();
     }
   } else {
-    Plivo.log.debug(`${LOGCAT.CALL} | inavlid response received, while call in progress`);
+    Plivo.log.debug(`${LOGCAT.CALL} | Invalid response received, while call in progress`);
   }
 };
 
@@ -423,6 +422,9 @@ const newDTMF = (evt: SessionNewDtmfEvent): void => {
  */
 const newInfo = (evt: SessionNewInfoEvent): void => {
   Plivo.log.info(`${LOGCAT.CALL} | Outgoing Call | ${evt.originator} INFO packet with body : ${evt.info.body}`);
+  if (cs._currentSession && evt.info.body === "remote-party-ringing") {
+    cs.emit('onCallConnected', cs._currentSession.getCallInfo(evt.originator));
+  }
   if (evt.info.body === 'no-remote-audio') {
     cs.emit('remoteAudioStatus', false);
   }

diff --git a/lib/managers/util.ts b/lib/managers/util.ts
@@ -9,6 +9,7 @@ import {
   getSDKVersion,
   getOS,
   SemverParserVersion,
+  getSdkVersionPre,
 } from '../utils/device';
 import {
   sendEvents,
@@ -61,6 +62,7 @@ const getSummaryEvent = async function (client: Client): Promise<SummaryEvent> {
     sdkVersionMajor: sdkVersionParse.major,
     sdkVersionMinor: sdkVersionParse.minor,
     sdkVersionPatch: sdkVersionParse.patch,
+    sdkVersionPre: getSdkVersionPre(sdkVersionParse),
     clientName: getBrowserDetails().browser,
     devicePlatform: navigator.platform,
     deviceOs,
@@ -72,7 +74,6 @@ const getSummaryEvent = async function (client: Client): Promise<SummaryEvent> {
     isAudioDeviceToggled: client.deviceToggledInCurrentSession,
     isNetworkChanged: client.networkChangeInCurrentSession,
     jsFramework: client.jsFramework,
-
   };
   if (client._currentSession) {
     summaryEvent.signalling = client._currentSession.signallingInfo;
@@ -219,7 +220,7 @@ const applyStatsSettings = function (): boolean {
 };
 
 /**
- * Initialise callstats storage.
+ * Initialize callstats storage.
  */
 export const callStart = function (): void {
   const client: Client = this;
@@ -470,7 +471,7 @@ const calcConnStage = function (obj: string[]): string {
 export const clearSessionInfo = function (session: CallSession): void {
   const client: Client = this;
   if (session === client._currentSession) {
-    // audio element clearence
+    // audio element clearance
     if (client.remoteView) {
       client.remoteView.pause();
     }
@@ -653,7 +654,7 @@ export const addCallstatsIOFabric = function (
 };
 
 /**
- * Send error to stats wahen media error occurs.
+ * Send error to stats when media error occurs.
  * @param {SessionFailedEvent} evt - rtcsession failed information
  * @param {CallSession} callSession - call session information
  */

diff --git a/lib/managers/workerManager.ts b/lib/managers/workerManager.ts
@@ -85,7 +85,7 @@ export class WorkerManager {
   /**
   * Start the network check timer.
   * @param {number} networkCheckInterval - time interval at which the timer is to be executed.
-  * @param {any} callback - callback to be trigerred when the timer executes.
+  * @param {any} callback - callback to be triggered when the timer executes.
   */
   startNetworkCheckTimer = (networkCheckInterval: number, callback: any, responseCallback: any) => {
     Plivo.log.debug(`${LOGCAT.NIMBUS} | starting the network check timer`);