Adds start silent config, to disable send/receive audio.

JitsiConferenceEventManager.js

@@ -644,15 +644,19 @@ JitsiConferenceEventManager.prototype.setupStatisticsListeners = function() {
         conference.eventEmitter.emit(
             JitsiConferenceEvents.BEFORE_STATISTICS_DISPOSED);
     });
-    conference.statistics.addByteSentStatsListener((tpc, stats) => {
-        conference.getLocalTracks(MediaType.AUDIO).forEach(track => {
-            const ssrc = tpc.getLocalSSRC(track);
 
-            if (!ssrc || !stats.hasOwnProperty(ssrc)) {
-                return;
-            }
+    // if we are in startSilent mode we will not be sending/receiving so nothing to detect
+    if (!conference.options.config.startSilent) {
+        conference.statistics.addByteSentStatsListener((tpc, stats) => {
+            conference.getLocalTracks(MediaType.AUDIO).forEach(track => {
+                const ssrc = tpc.getLocalSSRC(track);
+
+                if (!ssrc || !stats.hasOwnProperty(ssrc)) {
+                    return;
+                }
 
-            track._onByteSentStatsReceived(tpc, stats[ssrc]);
+                track._onByteSentStatsReceived(tpc, stats[ssrc]);
+            });
         });
-    });
+    }
 };

doc/API.md

@@ -225,7 +225,8 @@ This objects represents the server connection. You can create new ```JitsiConnec
         6. enableTalkWhileMuted - boolean property. Enables/disables talk while muted detection, by default the value is false/disabled.
         7. ignoreStartMuted - ignores start muted events coming from jicofo.
         8. enableStatsID - enables sending callStatsUsername as stats-id in presence, jicofo and videobridge will use it as endpointID to report stats
-        9. enableDisplayNameInStats - enables sending the users display name, if set, to callstats as alias of the endpointID stats 
+        9. enableDisplayNameInStats - enables sending the users display name, if set, to callstats as alias of the endpointID stats
+        10. startSilent - enables silent mode, will mark audio as inactive will not send/receive audio
 
         **NOTE: if 4 and 5 are set the library is going to send events to callstats. Otherwise the callstats integration will be disabled.**
 

modules/RTC/RTC.js

@@ -494,6 +494,7 @@ export default class RTC extends Listenable {
      *      disabled by removing it from the SDP.
      * @param {boolean} options.preferH264 If set to 'true' H264 will be
      *      preferred over other video codecs.
+     * @param {boolean} options.startSilent If set to 'true' no audio will be sent or received.
      * @return {TraceablePeerConnection}
      */
     createPeerConnection(signaling, iceConfig, isP2P, options) {

modules/RTC/TraceablePeerConnection.js

@@ -52,6 +52,7 @@ const SIM_LAYER_RIDS = [ SIM_LAYER_1_RID, SIM_LAYER_2_RID, SIM_LAYER_3_RID ];
  * @param {boolean} options.enableLayerSuspension if set to 'true', we will
  * cap the video send bitrate when we are told we have not been selected by
  * any endpoints (and therefore the non-thumbnail streams are not in use).
+ * @param {boolean} options.startSilent If set to 'true' no audio will be sent or received.
  *
  * FIXME: initially the purpose of TraceablePeerConnection was to be able to
  * debug the peer connection. Since many other responsibilities have been added
@@ -77,7 +78,7 @@ export default function TraceablePeerConnection(
      * @type {boolean}
      * @private
      */
-    this.audioTransferActive = true;
+    this.audioTransferActive = !(options.startSilent === true);
 
     /**
      * Indicates whether or not this peer connection instance is actively
@@ -1166,8 +1167,9 @@ function replaceDefaultUnifiedPlanMsid(ssrcLines = []) {
 /**
  * Makes sure that both audio and video directions are configured as 'sendrecv'.
  * @param {Object} localDescription the SDP object as defined by WebRTC.
+ * @param {object} options <tt>TracablePeerConnection</tt> config options.
  */
-const enforceSendRecv = function(localDescription) {
+const enforceSendRecv = function(localDescription, options) {
     if (!localDescription) {
         throw new Error('No local description passed in.');
     }
@@ -1177,7 +1179,12 @@ const enforceSendRecv = function(localDescription) {
     let changed = false;
 
     if (audioMedia && audioMedia.direction !== 'sendrecv') {
-        audioMedia.direction = 'sendrecv';
+        if (options.startSilent) {
+            audioMedia.direction = 'inactive';
+        } else {
+            audioMedia.direction = 'sendrecv';
+        }
+
         changed = true;
     }
 
@@ -1298,7 +1305,7 @@ const getters = {
         // Note that the description we set in chrome does have the accurate
         // direction (e.g. 'recvonly'), since that is technically what is
         // happening (check setLocalDescription impl).
-        desc = enforceSendRecv(desc);
+        desc = enforceSendRecv(desc, this.options);
 
         // See the method's doc for more info about this transformation.
         desc = this.localSdpMunger.transformStreamIdentifiers(desc);
@@ -1810,7 +1817,7 @@ TraceablePeerConnection.prototype.setLocalDescription = function(description) {
  * disabled the SDP audio media direction in the local SDP will be adjusted to
  * 'inactive' which means that no data will be sent nor accepted, but
  * the connection should be kept alive.
- * @param {boolean} active <tt>true</tt> to enable video media transmission or
+ * @param {boolean} active <tt>true</tt> to enable audio media transmission or
  * <tt>false</tt> to disable. If the value is not a boolean the call will have
  * no effect.
  * @return {boolean} <tt>true</tt> if the value has changed and sRD/sLD cycle

modules/xmpp/JingleSessionPC.js

@@ -304,6 +304,10 @@ export default class JingleSessionPC extends JingleSession {
             pcOptions.enableLayerSuspension = options.enableLayerSuspension;
         }
 
+        if (options.startSilent) {
+            pcOptions.startSilent = true;
+        }
+
         this.peerconnection
             = this.rtc.createPeerConnection(
                     this.signalingLayer,