feat: implement debug output for file and websocket

packages/js/README.md

@@ -163,15 +163,16 @@ const call = client.newCall({
   // Destination is required and can be a phone number or SIP URI
   destinationNumber: '18004377950',
   callerNumber: '‬155531234567',
+  debug: true // Default is false,
+  debugOutput: 'socket' // Possible values are 'socket' | 'file'
 });
 
-// Start the gathering of data
-call.startDebugger();
-
-// Stop the gathering of data
-call.stopDebugger();
+// The debug dump is set to be sent to telnyx by default, if you want to save the debug data to disk
+// You can change the debugOutput option to 'file'
 ```
 
+
+
 ---
 
 ## Examples

packages/js/src/Modules/Verto/messages/WebRTCStatsMessage.ts

@@ -0,0 +1,14 @@
+import BaseMessage from './BaseMessage';
+import { v4 } from 'uuid';
+export class WebRTCStatsMessage extends BaseMessage {
+  constructor(data: any) {
+    super();
+    const reportId = v4();
+    this.buildRequest({
+      type: 'debug_final_report',
+      debug_final_report: btoa(JSON.stringify(data)),
+      debug_report_id: reportId,
+    });
+    console.log(`WebRTC Debug Report generated: ${reportId}`);
+  }
+}

packages/js/src/Modules/Verto/util/interfaces.ts

@@ -15,11 +15,14 @@ export interface IVertoOptions {
   env?: Environment;
   iceServers?: RTCIceServer[];
   /**
-   * autoReconnect: Determine if the SDK has to re-connect automatically when detecting a gateway connection failure. 
+   * autoReconnect: Determine if the SDK has to re-connect automatically when detecting a gateway connection failure.
    * This is set to`true` as default
    * @type {boolean}
    */
   autoReconnect?: boolean;
+
+  debug?: boolean;
+  debugOutput?: 'socket' | 'file';
 }
 export interface SubscribeParams {
   channels?: string[];

packages/js/src/Modules/Verto/webrtc/BaseCall.ts

@@ -51,6 +51,7 @@ import { MCULayoutEventHandler } from './LayoutHandler';
 import Call from './Call';
 import pkg from '../../../../package.json';
 import { WebRTCStats } from '@peermetrics/webrtc-stats';
+import { WebRTCStatsMessage } from '../messages/WebRTCStatsMessage';
 const SDK_VERSION = pkg.version;
 
 /**
@@ -296,7 +297,6 @@ export default abstract class BaseCall implements IWebRTCCall {
     this.direction = Direction.Outbound;
     this.peer = new Peer(PeerType.Offer, this.options);
     await this.peer.iceGatheringComplete.promise;
-    console.log('icegatheringcompleted')
     this._onIceSdp(this.peer.instance.localDescription);
   }
 
@@ -374,12 +374,30 @@ export default abstract class BaseCall implements IWebRTCCall {
     this.setState(State.Hangup);
 
     const _close = () => {
-      this.peer ? this.peer.instance.close() : null;
-      this.setState(State.Destroy);
+      const debugData = this.peer?.close();
+      if (!debugData) {
+        return this.setState(State.Destroy);
+      }
+      if (this.options.debugOutput === 'socket') {
+        this._execute(new WebRTCStatsMessage(debugData));
+      }
+      if (this.options.debugOutput === 'file') {
+        // Save to file
+        const blob = new Blob([JSON.stringify(debugData)], {
+          type: 'application/json',
+        });
+        const downloadUrl = URL.createObjectURL(blob);
+        const link = document.createElement('a');
+        link.href = downloadUrl;
+        link.download = `call_${this.options.id}.json`;
+        link.click();
+        URL.revokeObjectURL(downloadUrl);
+      }
+      return this.setState(State.Destroy);
     };
 
     this.stopRingtone();
-    this.stopDebugger();  
+    this.stopDebugger();
     if (execute) {
       const bye = new Bye({
         sessid: this.session.sessionid,
@@ -1470,7 +1488,6 @@ export default abstract class BaseCall implements IWebRTCCall {
         this.hangup();
       });
     console.timeEnd(TIME_CALL_INVITE);
-
   }
 
   private _onIce(event: RTCPeerConnectionIceEvent) {

packages/js/src/Modules/Verto/webrtc/Peer.ts

@@ -16,22 +16,18 @@ import {
   RTCPeerConnection,
   streamIsValid,
 } from '../util/webrtc';
-import {
-  DeferredPromise,
-  deferredPromise,
-  isFunction,
-} from '../util/helpers';
+import { DeferredPromise, deferredPromise, isFunction } from '../util/helpers';
 import { IVertoCallOptions } from './interfaces';
 import { trigger } from '../services/Handler';
-
+import { WebRTCStats } from '@peermetrics/webrtc-stats';
 /**
  * @ignore Hide in docs output
  */
 export default class Peer {
   public instance: RTCPeerConnection;
   public iceGatheringComplete: DeferredPromise<boolean>;
   public onSdpReadyTwice: Function = null;
-
+  private _webrtcStats: WebRTCStats;
   private _constraints: {
     offerToReceiveAudio: boolean;
     offerToReceiveVideo: boolean;
@@ -56,6 +52,18 @@ export default class Peer {
     this.createPeerConnection = this.createPeerConnection.bind(this);
     this.iceGatheringComplete = deferredPromise({ debounceTime: 100 });
 
+    if (this.options.debug) {
+      this._webrtcStats = new WebRTCStats({
+        getStatsInterval: 1000,
+        rawStats: false,
+        statsObject: false,
+        filteredStats: false,
+        remote: true,
+        wrapGetUserMedia: true,
+        debug: false,
+        logLevel: 'warn',
+      });
+    }
     this._init();
   }
 
@@ -144,7 +152,13 @@ export default class Peer {
   };
   private async createPeerConnection() {
     this.instance = RTCPeerConnection(this._config());
-
+    if (this.options.debug) {
+      this._webrtcStats?.addConnection({
+        pc: this.instance,
+        peerId: this.options.id,
+        connectionId: this.options.id,
+      });
+    }
     this.instance.onsignalingstatechange = this.handleSignalingStateChangeEvent;
     this.instance.onnegotiationneeded = this.handleNegotiationNeededEvent;
     this.instance.ontrack = this.handleTrackEvent;
@@ -233,6 +247,7 @@ export default class Peer {
     } else {
       this.startNegotiation();
     }
+
     this._logTransceivers();
   }
 
@@ -386,4 +401,14 @@ export default class Peer {
     logger.info('RTC config', config);
     return config;
   }
+
+  public close() {
+    let data = null;
+    if (this._webrtcStats) {
+      data = this._webrtcStats.getTimeline('stats');
+      this._webrtcStats.destroy();
+    }
+    this.instance.close();
+    return data;
+  }
 }

packages/js/src/Modules/Verto/webrtc/VertoHandler.ts

@@ -76,6 +76,8 @@ class VertoHandler {
         callerNumber: params.callee_id_number,
         attach,
         mediaSettings: params.mediaSettings,
+        debug: session.options.debug ?? false,
+        debugOutput: session.options.debugOutput ?? 'socket',
       };
 
       if (params.telnyx_call_control_id) {

packages/js/src/Modules/Verto/webrtc/constants.ts

@@ -50,6 +50,8 @@ export const DEFAULT_CALL_OPTIONS: IVertoCallOptions = {
   audio: true,
   video: false,
   useStereo: false,
+  debug: false,
+  debugOutput: 'socket',
   attach: false,
   screenShare: false,
   userVariables: {},

packages/js/src/Modules/Verto/webrtc/interfaces.ts

@@ -45,6 +45,8 @@ export interface IVertoCallOptions {
   negotiateVideo?: boolean;
   mediaSettings?: IMediaSettings;
   customHeaders?: Array<{ name: string; value: string }>;
+  debug?: boolean;
+  debugOutput?: 'socket' | 'file';
 }
 
 export interface IStatsBinding {