fix(connectionstatus) Increase the rtc mute timeout for p2p.

Increase the RTC mute timeout from 500ms to 2500ms for p2p connections. This fixes an issue with Chrome tab sharing where the application keeps switching between the avatar and the share contnuously because of a chrome bug https://bugs.chromium.org/p/chromium/issues/detail?id=1258034

JitsiConference.js

             this.rtc,
             this,
             {
-                // Both these options are not public API, leaving it here only
-                // as an entry point through config for tuning up purposes.
-                // Default values should be adjusted as soon as optimal values
-                // are discovered.
+                // These options are not public API, leaving it here only as an entry point through config for tuning
+                // up purposes. Default values should be adjusted as soon as optimal values are discovered.
+                p2pRtcMuteTimeout: config._p2pConnStatusRtcMuteTimeout,
                 rtcMuteTimeout: config._peerConnStatusRtcMuteTimeout,
                 outOfLastNTimeout: config._peerConnStatusOutOfLastNTimeout
             });

modules/connectivity/ParticipantConnectionStatus.js

 const logger = getLogger(__filename);
 
 /**
- * Default value of 500 milliseconds for
- * {@link ParticipantConnectionStatus.outOfLastNTimeout}.
+ * Default value of 500 milliseconds for {@link ParticipantConnectionStatus.outOfLastNTimeout}.
  *
  * @type {number}
  */
 const DEFAULT_NOT_IN_LAST_N_TIMEOUT = 500;
 
 /**
- * Default value of 2000 milliseconds for
- * {@link ParticipantConnectionStatus.rtcMuteTimeout}.
+ * Default value of 2500 milliseconds for {@link ParticipantConnectionStatus.p2pRtcMuteTimeout}.
+ */
+const DEFAULT_P2P_RTC_MUTE_TIMEOUT = 2500;
+
+/**
+ * Default value of 10000 milliseconds for {@link ParticipantConnectionStatus.rtcMuteTimeout}.
  *
  * @type {number}
  */
      * @param {RTC} rtc the RTC service instance
      * @param {JitsiConference} conference parent conference instance
      * @param {Object} options
+     * @param {number} [options.p2pRtcMuteTimeout=2500] custom value for
+     * {@link ParticipantConnectionStatus.p2pRtcMuteTimeout}.
      * @param {number} [options.rtcMuteTimeout=2000] custom value for
      * {@link ParticipantConnectionStatus.rtcMuteTimeout}.
      * @param {number} [options.outOfLastNTimeout=500] custom value for
             = typeof options.outOfLastNTimeout === 'number'
                 ? options.outOfLastNTimeout : DEFAULT_NOT_IN_LAST_N_TIMEOUT;
 
+        /**
+         * How long we are going to wait for the corresponding signaling mute event after the RTC video track muted
+         * event is fired on the Media stream, before the connection interrupted is fired. The default value is
+         * {@link DEFAULT_P2P_RTC_MUTE_TIMEOUT}.
+         *
+         * @type {number} amount of time in milliseconds.
+         */
+        this.p2pRtcMuteTimeout = typeof options.p2pRtcMuteTimeout === 'number'
+            ? options.p2pRtcMuteTimeout : DEFAULT_P2P_RTC_MUTE_TIMEOUT;
+
         /**
          * How long we're going to wait after the RTC video track muted event
          * for the corresponding signalling mute event, before the connection
      */
     _getVideoFrozenTimeout(id) {
         return this.rtc.isInLastN(id)
-            ? this.rtcMuteTimeout : this.outOfLastNTimeout;
+            ? this.rtcMuteTimeout
+            : this.conference.isP2PActive() ? this.p2pRtcMuteTimeout : this.outOfLastNTimeout;
     }
 
     /**