@wazo/sdk 0.37.21 → 0.37.25

package/src/lib/WazoSessionDescriptionHandler.js

@@ -12,11 +12,9 @@ import { SessionDescriptionHandler }
 import { SessionDescriptionHandlerOptions }
   from 'sip.js/lib/platform/web/session-description-handler/session-description-handler-options';
 import IssueReporter from '../service/IssueReporter';
-import { areCandidateValid, fixSdp, parseCandidate } from '../utils/sdp';
+import { fixSdp, parseCandidate } from '../utils/sdp';
 
 const wazoLogger = IssueReporter ? IssueReporter.loggerFor('webrtc-sdh') : console;
-const MAX_WAIT_FOR_ICE_TRIES = 20;
-const WAIT_FOR_ICE_TIMEOUT = 500;
 
 // Customized mediaStreamFactory allowing to send screensharing stream directory when upgrading
 export const wazoMediaStreamFactory = (constraints: Object): Promise<MediaStream> => {
@@ -206,38 +204,6 @@ class WazoSessionDescriptionHandler extends SessionDescriptionHandler {
     this._peerConnection = undefined;
   }
 
-  async _waitForValidGatheredIce(): Object {
-    let tries = 0;
-
-    while (!areCandidateValid(this.gatheredCandidates) && tries < MAX_WAIT_FOR_ICE_TRIES) {
-      wazoLogger.trace('SessionDescriptionHandler._waitForValidGatheredIce, waiting for ice', {
-        tries,
-        max: MAX_WAIT_FOR_ICE_TRIES,
-      });
-
-      tries++;
-
-      // eslint-disable-next-line no-await-in-loop
-      await new Promise(resolve => setTimeout(resolve, WAIT_FOR_ICE_TIMEOUT));
-    }
-
-    if (tries >= MAX_WAIT_FOR_ICE_TRIES) {
-      const errorMsg = 'No valid candidates found, can\'t answer the call';
-      const error = new Error(errorMsg);
-      wazoLogger.error(errorMsg, { tries, max: MAX_WAIT_FOR_ICE_TRIES });
-
-      // Emit an error, because sip.js catches the exception and switch to status Terminated.
-      this.eventEmitter.emit('error', error);
-
-      wazoLogger.error('No valid candidates found', { tries, candidates: JSON.stringify(this.gatheredCandidates) });
-    }
-
-    // eslint-disable-next-line
-    wazoLogger.trace('Found valid candidates', { tries, candidates: JSON.stringify(this.gatheredCandidates) });
-
-    return this.getLocalSessionDescription();
-  }
-
   // Overridden to send `inactive` in conference
   updateDirection(options?: SessionDescriptionHandlerOptions, isConference: boolean = false,
     audioOnly: boolean = false): Promise<void> {

package/src/web-rtc-client.js

@@ -453,7 +453,7 @@ export default class WebRTCClient extends Emitter {
   }
 
   call(number: string, enableVideo?: boolean, audioOnly: boolean = false, conference: boolean = false): Session {
-    logger.info('sdk webrtc creating call', { number, enableVideo, audioOnly });
+    logger.info('sdk webrtc creating call', { number, enableVideo, audioOnly, conference });
     const inviterOptions: Object = {
       sessionDescriptionHandlerOptionsReInvite: {
         conference,
@@ -1529,6 +1529,7 @@ export default class WebRTCClient extends Emitter {
     const screenSharing = constraints && 'screen' in constraints ? constraints.screen : false;
     const isDesktop = constraints && 'desktop' in constraints ? constraints.desktop : false;
     const withAudio = constraints && 'audio' in constraints ? constraints.audio : true;
+    const mandatoryVideo = constraints && typeof constraints.video === 'object' ? constraints.video.mandatory : {};
 
     logger.info('Retrieving media a configuration', { enableVideo, screenSharing, isDesktop, withAudio, constraints });
 
@@ -1536,7 +1537,8 @@ export default class WebRTCClient extends Emitter {
       constraints: {
         // Exact constraint are not supported with `getDisplayMedia` and we must have a video=false in desktop screenshare
         audio: screenSharing ? !isDesktop : (withAudio ? this._getAudioConstraints() : false),
-        video: screenSharing ? (isDesktop ? ({ mandatory: { chromeMediaSource: 'desktop' } }) : { cursor: 'always' })
+        video: screenSharing ? (isDesktop
+          ? ({ mandatory: { chromeMediaSource: 'desktop', ...(mandatoryVideo || {}) } }) : { cursor: 'always' })
           : this._getVideoConstraints(enableVideo),
         screen: screenSharing,
         desktop: isDesktop,
@@ -1871,11 +1873,14 @@ export default class WebRTCClient extends Emitter {
 
     session.sessionDescriptionHandler.remoteMediaStream.onaddtrack = onTrack;
 
-    pc.oniceconnectionstatechange = () => {
-      if (pc.iceConnectionState === 'disconnected') {
-        this.eventEmitter.emit(ON_DISCONNECTED);
-      }
-    };
+    if (pc) {
+      pc.oniceconnectionstatechange = () => {
+        logger.info('on ice connection state changed', { state: pc.iceConnectionState });
+        if (pc.iceConnectionState === 'disconnected') {
+          this.eventEmitter.emit(ON_DISCONNECTED);
+        }
+      };
+    }
 
     if (withEvent) {
       this.eventEmitter.emit(ACCEPTED, session, sessionDialog);
@@ -2085,13 +2090,17 @@ export default class WebRTCClient extends Emitter {
     const nbStats = this.sessionNetworkStats[sessionId].length;
     const lastNetworkStats = this.sessionNetworkStats[sessionId][nbStats - 1];
     const lastAudioSent = lastNetworkStats ? lastNetworkStats.audioBytesSent : 0;
+    const lastAudioContentSent = lastNetworkStats ? lastNetworkStats.audioContentSent : 0;
     const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
     const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 0;
+    const lastAudioContentReceived = lastNetworkStats ? lastNetworkStats.audioContentReceived : 0;
     const lastVideoReceived = lastNetworkStats ? lastNetworkStats.videoBytesReceived : 0;
     const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
     const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
-    let audioBytesSent = 0;
-    let audioBytesReceived = 0;
+    let audioBytesSent = 0; // content + header
+    let audioBytesReceived = 0; // content + header
+    let audioContentSent = 0; // useful to detect blank call
+    let audioContentReceived = 0;
     let videoBytesSent = 0;
     let videoBytesReceived = 0;
     let transportSent = 0;
@@ -2100,25 +2109,27 @@ export default class WebRTCClient extends Emitter {
 
     stats.forEach(report => {
       if (report.type === 'outbound-rtp' && report.kind === 'audio') {
-        audioBytesSent += report.bytesSent + report.headerBytesSent;
+        audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
+        audioContentSent += Number(report.bytesSent);
       }
       if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
-        audioBytesSent += report.bytesSent;
+        audioBytesSent += Number(report.bytesSent);
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
-        videoBytesSent += report.bytesSent + report.headerBytesSent;
+        videoBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
       }
       if (report.type === 'inbound-rtp' && report.kind === 'audio') {
-        packetsLost += report.packetsLost;
-        networkStats.packetsReceived = report.packetsReceived;
-        audioBytesReceived += report.bytesReceived + report.headerBytesReceived;
+        packetsLost += Number(report.packetsLost);
+        networkStats.packetsReceived = Number(report.packetsReceived);
+        audioBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
+        audioContentReceived += Number(report.bytesReceived);
       }
       if (report.type === 'inbound-rtp' && report.kind === 'video') {
-        videoBytesReceived += report.bytesReceived + report.headerBytesReceived;
+        videoBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
       }
       if (report.type === 'outbound-rtp' && report.kind === 'video') {
         if ('framesPerSecond' in report) {
-          networkStats.framesPerSecond = report.framesPerSecond;
+          networkStats.framesPerSecond = Number(report.framesPerSecond);
         }
         if ('framerateMean' in report) {
           // framerateMean is only available in FF
@@ -2126,24 +2137,31 @@ export default class WebRTCClient extends Emitter {
         }
       }
       if (report.type === 'remote-inbound-rtp' && report.kind === 'audio') {
-        packetsLost += report.packetsLost;
-        networkStats.roundTripTime = report.roundTripTime;
-        networkStats.jitter = report.jitter;
+        packetsLost += Number(report.packetsLost);
+        networkStats.roundTripTime = Number(report.roundTripTime);
+        networkStats.jitter = Number(report.jitter);
       }
       if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
-        packetsLost += report.packetsLost;
+        packetsLost += Number(report.packetsLost);
       }
       if (report.type === 'transport') {
-        transportSent += report.bytesSent;
-        transportReceived += report.bytesReceived;
+        transportSent += Number(report.bytesSent);
+        transportReceived += Number(report.bytesReceived);
       }
     });
 
     networkStats.packetsLost = packetsLost;
+
     networkStats.totalAudioBytesSent = audioBytesSent;
     networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
     networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
     networkStats.audioBytesReceived = audioBytesReceived;
+
+    networkStats.totalAudioContentSent = audioContentSent;
+    networkStats.audioContentSent = audioContentSent - lastAudioContentSent;
+    networkStats.totalAudioBytesReceived = audioContentReceived - lastAudioContentReceived;
+    networkStats.audioContentReceived = audioContentReceived;
+
     networkStats.totalVideoBytesSent = videoBytesSent;
     networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
     networkStats.totalVideoBytesReceived = videoBytesReceived;