fix: Fixes mute to stop forwarding stream.

This works only in the case when a streams are being forwarded (no transcoding).

Author: damencho

src/main/java/org/jitsi/jigasi/AudioModeration.java

@@ -281,8 +281,7 @@ else if (type.equalsIgnoreCase("muteRequest"))
                     {
                         this.raiseHand();
                     }
-                    // Send request to jicofo
-                    else if (this.requestAudioMuteByJicofo(bAudioMute))
+                    else if (this.mute(bAudioMute))
                     {
                         // Send response through sip, respondRemoteAudioMute
                         this.gatewaySession.sendJson(
@@ -342,14 +341,14 @@ void setChatRoomAudioMuted(boolean muted)
     }
 
     /**
-     * Request Jicofo on behalf to mute/unmute us.
+     * Mute/unmute us.
      *
      * @param bMuted <tt>true</tt> if request is to mute audio,
      * false otherwise
      * @return <tt>true</tt> if request succeeded, false
      * otherwise
      */
-    public boolean requestAudioMuteByJicofo(boolean bMuted)
+    public boolean mute(boolean bMuted)
     {
         ChatRoom mucRoom = this.jvbConference.getJvbRoom();
 
@@ -360,6 +359,13 @@ public boolean requestAudioMuteByJicofo(boolean bMuted)
             return false;
         }
 
+        this.gatewaySession.mute(bMuted);
+
+        if (!this.avModerationEnabled)
+        {
+            return true;
+        }
+
         StanzaCollector collector = null;
         try
         {
@@ -421,11 +427,11 @@ && isMutingSupported()
                 // in case of startAudioMuted, we want jicofo to stop the bridge from sending our audio
                 // a specific case for jigasi as it doesn't do local muting
                 // in case of av-moderation jicofo has done that already for us
-                this.requestAudioMuteByJicofo(true);
+                this.mute(true);
             }
 
             // inform the sip side that our state is muted (av moderation or not)
-            mute();
+            muteViaSIPInfo();
 
             // in case we reconnect start muted maybe no-longer set
             this.startAudioMuted = false;
@@ -438,7 +444,7 @@ && isMutingSupported()
      * When we receive confirmation for the announcement we will update
      * our presence status in the conference.
      */
-    public void mute()
+    public void muteViaSIPInfo()
     {
         if (!isMutingSupported())
             return;
@@ -601,7 +607,7 @@ private IQ handleMuteIq(MuteIq muteIq)
 
             if (doMute)
             {
-                mute();
+                muteViaSIPInfo();
             }
 
             return IQ.createResultIQ(muteIq);

src/main/java/org/jitsi/jigasi/SipCallKeepAliveTransformer.java renamed to src/main/java/org/jitsi/jigasi/SipCallTransformer.java

@@ -35,10 +35,11 @@
  * send hole punch packets. It can happen that the xmpp call is dropped and
  * the sip call participant is waiting then there will be no media ot RTCP to
  * keep alive the stream from the other side.
+ * Can be muted to stop forwarding packets coming from the sip side.
  *
  * @author Damian Minkov
  */
-public class SipCallKeepAliveTransformer
+public class SipCallTransformer
     extends SinglePacketTransformerAdapter
     implements TransformEngine
 {
@@ -76,11 +77,21 @@ public class SipCallKeepAliveTransformer
      */
     private final MediaStream stream;
 
+    /**
+     * If true we will mute the media, so we will not forward any media from the sip side.
+     */
+    private boolean mute = false;
+
+    /**
+     * Counter for the number of packets received.
+     * Used for passing through a packet while being muted, one on every 1000.
+     */
+    private long packetsCounter = 0;
+
     /**
      * Initializes a new {@link SsrcRewriter} instance.
      */
-    public SipCallKeepAliveTransformer(
-        CallPeerMediaHandler handler, MediaStream stream)
+    public SipCallTransformer(CallPeerMediaHandler handler, MediaStream stream)
     {
         super(RTPPacketPredicate.INSTANCE);
 
@@ -116,8 +127,15 @@ public RawPacket transform(RawPacket pkt)
     @Override
     public RawPacket reverseTransform(RawPacket pkt)
     {
+        packetsCounter++;
         seenSSRCs.add(pkt.getSSRCAsLong());
 
+        // if muted we want to pass one packet every 1000 packets
+        if (mute && packetsCounter % 1000 != 0)
+        {
+            return null;
+        }
+
         return pkt;
     }
 
@@ -141,6 +159,15 @@ public PacketTransformer getRTCPTransformer()
         return rtcpTransformer;
     }
 
+    /**
+     * Sets the mute state of this transformer.
+     * @param bMuted <tt>true</tt> to mute the media, <tt>false</tt> to unmute.
+     */
+    public void mute(boolean bMuted)
+    {
+        this.mute = bMuted;
+    }
+
     /**
      * Checks and filters RTCP.BYE.
      */
@@ -236,7 +263,7 @@ public void run()
                             RawPacket.FIXED_HEADER_SIZE + 1);
 
                         stream.injectPacket(
-                            packet, true, SipCallKeepAliveTransformer.this);
+                            packet, true, SipCallTransformer.this);
 
                         ts += 160;
                     }

src/main/java/org/jitsi/jigasi/SipGatewaySession.java

@@ -290,8 +290,9 @@ public class SipGatewaySession
      * from the sip direction. Skips forwarding RTCP traffic which is not
      * intended for that direction (particularly we had seen RTCP.BYE for
      * to cause media to stop (even when ssrc is not matching)).
+     * Used for muting the media stream.
      */
-    private SipCallKeepAliveTransformer transformerMonitor;
+    private SipCallTransformer callTransformer;
 
     /**
      * Whether we had sent indication that XMPP connection terminated and
@@ -716,10 +717,10 @@ private void sipCallEnded()
         if (peerStateListener != null)
             peerStateListener.unregister();
 
-        if (this.transformerMonitor != null)
+        if (this.callTransformer != null)
         {
-            this.transformerMonitor.dispose();
-            this.transformerMonitor = null;
+            this.callTransformer.dispose();
+            this.callTransformer = null;
         }
 
         this.soundNotificationManager.stop();
@@ -1155,11 +1156,16 @@ void notifyJvbRoomJoined()
             if (this.jvbConference.getAudioModeration() != null)
             {
                 // notify user that is muted
-                this.jvbConference.getAudioModeration().mute();
+                this.jvbConference.getAudioModeration().muteViaSIPInfo();
             }
         }
     }
 
+    public void mute(boolean bMuted)
+    {
+        callTransformer.mute(bMuted);
+    }
+
     /**
      * {@inheritDoc}
      */
@@ -1236,9 +1242,8 @@ private boolean addSipCallTransformer(CallPeer peer)
                 MediaStream stream = mediaHandler.getStream(MediaType.AUDIO);
                 if (stream != null)
                 {
-                    transformerMonitor = new SipCallKeepAliveTransformer(
-                        peerMedia.getMediaHandler(), stream);
-                    stream.setExternalTransformer(transformerMonitor);
+                    callTransformer = new SipCallTransformer(peerMedia.getMediaHandler(), stream);
+                    stream.setExternalTransformer(callTransformer);
                     return true;
                 }
             }