X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtpmanager%2Fgstrtpbin.c;h=fbed41415536c4759bdd13c4d7bc1140794e52dd;hb=f05e14ba316d67b9cdf1440d9b2129da03625a30;hp=dd83eb4df326300a6f703445392a76b36cf3366a;hpb=5986dc287c2e53b0a07498835d39ab1c2c47b939;p=platform%2Fupstream%2Fgst-plugins-good.git

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index dd83eb4..fbed414 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -1,21 +1,21 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
+ /* GStreamer
+  * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+  *
+  * This library is free software; you can redistribute it and/or
+  * modify it under the terms of the GNU Library General Public
+  * License as published by the Free Software Foundation; either
+  * version 2 of the License, or (at your option) any later version.
+  *
+  * This library is distributed in the hope that it will be useful,
+  * but WITHOUT ANY WARRANTY; without even the implied warranty of
+  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+  * Library General Public License for more details.
+  *
+  * You should have received a copy of the GNU Library General Public
+  * License along with this library; if not, write to the
+  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+  * Boston, MA 02110-1301, USA.
+  */
 
 /**
  * SECTION:element-rtpbin
@@ -53,13 +53,6 @@
  * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
  * send_rtp_src_\%u pad after updating its internal state.
  *
- * #GstRtpBin can also demultiplex incoming bundled streams. The first
- * #GstRtpSession will have a #GstRtpSsrcDemux element splitting the streams
- * based on their SSRC and potentially dispatched to a different #GstRtpSession.
- * Because retransmission SSRCs need to be merged with the corresponding media
- * stream the #GstRtpBin::on-bundled-ssrc signal is emitted so that the
- * application can find out to which session the SSRC belongs.
- *
  * The session manager needs the clock-rate of the payload types it is handling
  * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
  * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
@@ -204,9 +197,6 @@ static GstStaticPadTemplate rtpbin_send_rtp_src_template =
     GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
     );
 
-#define GST_RTP_BIN_GET_PRIVATE(obj)  \
-   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
-
 #define GST_RTP_BIN_LOCK(bin)   g_mutex_lock (&(bin)->priv->bin_lock)
 #define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
 
@@ -230,6 +220,10 @@ G_STMT_START {                                   \
 #define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin)         \
   GST_RTP_BIN_DYN_UNLOCK (bin);                  \
 
+/* Minimum time offset to apply. This compensates for rounding errors in NTP to
+ * RTP timestamp conversions */
+#define MIN_TS_OFFSET (4 * GST_MSECOND)
+
 struct _GstRtpBinPrivate
 {
   GMutex bin_lock;
@@ -258,6 +252,8 @@ enum
   SIGNAL_RESET_SYNC,
   SIGNAL_GET_SESSION,
   SIGNAL_GET_INTERNAL_SESSION,
+  SIGNAL_GET_STORAGE,
+  SIGNAL_GET_INTERNAL_STORAGE,
 
   SIGNAL_ON_NEW_SSRC,
   SIGNAL_ON_SSRC_COLLISION,
@@ -275,7 +271,11 @@ enum
   SIGNAL_REQUEST_RTCP_ENCODER,
   SIGNAL_REQUEST_RTCP_DECODER,
 
+  SIGNAL_REQUEST_FEC_DECODER,
+  SIGNAL_REQUEST_FEC_ENCODER,
+
   SIGNAL_NEW_JITTERBUFFER,
+  SIGNAL_NEW_STORAGE,
 
   SIGNAL_REQUEST_AUX_SENDER,
   SIGNAL_REQUEST_AUX_RECEIVER,
@@ -309,6 +309,8 @@ enum
 #define DEFAULT_MAX_MISORDER_TIME    2000
 #define DEFAULT_RFC7273_SYNC         FALSE
 #define DEFAULT_MAX_STREAMS          G_MAXUINT
+#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
+#define DEFAULT_MAX_TS_OFFSET        G_GINT64_CONSTANT(3000000000)
 
 enum
 {
@@ -333,7 +335,9 @@ enum
   PROP_MAX_DROPOUT_TIME,
   PROP_MAX_MISORDER_TIME,
   PROP_RFC7273_SYNC,
-  PROP_MAX_STREAMS
+  PROP_MAX_STREAMS,
+  PROP_MAX_TS_OFFSET_ADJUSTMENT,
+  PROP_MAX_TS_OFFSET,
 };
 
 #define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
@@ -373,12 +377,12 @@ static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
 static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
 static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
 static GstPad *complete_session_sink (GstRtpBin * rtpbin,
-    GstRtpBinSession * session, gboolean bundle_demuxer_needed);
+    GstRtpBinSession * session);
 static void
 complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
     guint sessid);
 static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
-    GstRtpBinSession * session, guint sessid, gboolean bundle_demuxer_needed);
+    GstRtpBinSession * session, guint sessid);
 
 /* Manages the RTP stream for one SSRC.
  *
@@ -433,6 +437,9 @@ struct _GstRtpBinStream
  * there they are pushed into an SSRC demuxer that splits the stream based on
  * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
  * the GstRtpBinStream above).
+ *
+ * Before the SSRC demuxer, a storage element may be inserted for the purpose
+ * of Forward Error Correction.
  */
 struct _GstRtpBinSession
 {
@@ -447,11 +454,8 @@ struct _GstRtpBinSession
   gulong demux_newpad_sig;
   gulong demux_padremoved_sig;
 
-  /* Bundling support */
-  GstElement *rtp_funnel;
-  GstElement *rtcp_funnel;
-  GstElement *bundle_demux;
-  gulong bundle_demux_newpad_sig;
+  /* Fec support */
+  GstElement *storage;
 
   GMutex lock;
 
@@ -473,7 +477,6 @@ struct _GstRtpBinSession
   GstPad *sync_src;
   GstPad *send_rtp_sink;
   GstPad *send_rtp_sink_ghost;
-  GstPad *send_rtp_src;
   GstPad *send_rtp_src_ghost;
   GstPad *send_rtcp_src;
   GstPad *send_rtcp_src_ghost;
@@ -654,108 +657,13 @@ ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
   GST_RTP_BIN_UNLOCK (rtpbin);
 }
 
-static void
-new_bundled_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
-    GstRtpBinSession * session)
-{
-  GValue result = G_VALUE_INIT;
-  GValue params[2] = { G_VALUE_INIT, G_VALUE_INIT };
-  guint session_id = 0;
-  GstRtpBinSession *target_session = NULL;
-  GstRtpBin *rtpbin = session->bin;
-  gchar *name;
-  GstPad *src_pad;
-  GstPad *recv_rtp_sink = NULL;
-  GstPad *recv_rtcp_sink = NULL;
-  GstPadLinkReturn ret;
-
-  GST_RTP_BIN_DYN_LOCK (rtpbin);
-  GST_DEBUG_OBJECT (rtpbin, "new bundled SSRC pad %08x, %s:%s", ssrc,
-      GST_DEBUG_PAD_NAME (pad));
-
-  g_value_init (&result, G_TYPE_UINT);
-  g_value_init (&params[0], GST_TYPE_ELEMENT);
-  g_value_set_object (&params[0], rtpbin);
-  g_value_init (&params[1], G_TYPE_UINT);
-  g_value_set_uint (&params[1], ssrc);
-
-  g_signal_emitv (params,
-      gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, &result);
-  g_value_unset (&params[0]);
-
-  session_id = g_value_get_uint (&result);
-  if (session_id == 0) {
-    target_session = session;
-  } else {
-    target_session = find_session_by_id (rtpbin, (gint) session_id);
-    if (!target_session) {
-      target_session = create_session (rtpbin, session_id);
-    }
-    if (!target_session) {
-      /* create_session() warned already */
-      GST_RTP_BIN_DYN_UNLOCK (rtpbin);
-      return;
-    }
-
-    if (!target_session->recv_rtp_sink) {
-      recv_rtp_sink = complete_session_sink (rtpbin, target_session, FALSE);
-    }
-
-    if (!target_session->recv_rtp_src)
-      complete_session_receiver (rtpbin, target_session, session_id);
-
-    if (!target_session->recv_rtcp_sink) {
-      recv_rtcp_sink =
-          complete_session_rtcp (rtpbin, target_session, session_id, FALSE);
-    }
-  }
-
-  GST_DEBUG_OBJECT (rtpbin, "Assigning bundled ssrc %u to session %u", ssrc,
-      session_id);
-
-  if (!recv_rtp_sink) {
-    recv_rtp_sink =
-        gst_element_get_request_pad (target_session->rtp_funnel, "sink_%u");
-  }
-
-  if (!recv_rtcp_sink) {
-    recv_rtcp_sink =
-        gst_element_get_request_pad (target_session->rtcp_funnel, "sink_%u");
-  }
-
-  name = g_strdup_printf ("src_%u", ssrc);
-  src_pad = gst_element_get_static_pad (element, name);
-  ret = gst_pad_link (src_pad, recv_rtp_sink);
-  g_free (name);
-  gst_object_unref (src_pad);
-  gst_object_unref (recv_rtp_sink);
-  if (ret != GST_PAD_LINK_OK) {
-    g_warning
-        ("rtpbin: failed to link bundle demuxer to receive rtp funnel for session %u",
-        session_id);
-  }
-
-  name = g_strdup_printf ("rtcp_src_%u", ssrc);
-  src_pad = gst_element_get_static_pad (element, name);
-  gst_pad_link (src_pad, recv_rtcp_sink);
-  g_free (name);
-  gst_object_unref (src_pad);
-  gst_object_unref (recv_rtcp_sink);
-  if (ret != GST_PAD_LINK_OK) {
-    g_warning
-        ("rtpbin: failed to link bundle demuxer to receive rtcp sink pad for session %u",
-        session_id);
-  }
-
-  GST_RTP_BIN_DYN_UNLOCK (rtpbin);
-}
-
 /* create a session with the given id.  Must be called with RTP_BIN_LOCK */
 static GstRtpBinSession *
 create_session (GstRtpBin * rtpbin, gint id)
 {
   GstRtpBinSession *sess;
   GstElement *session, *demux;
+  GstElement *storage = NULL;
   GstState target;
 
   if (!(session = gst_element_factory_make ("rtpsession", NULL)))
@@ -764,15 +672,22 @@ create_session (GstRtpBin * rtpbin, gint id)
   if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
     goto no_demux;
 
+  if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
+    goto no_storage;
+
+  /* need to sink the storage or otherwise signal handlers from bindings will
+   * take ownership of it and we don't own it anymore */
+  gst_object_ref_sink (storage);
+  g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
+      id);
+
   sess = g_new0 (GstRtpBinSession, 1);
   g_mutex_init (&sess->lock);
   sess->id = id;
   sess->bin = rtpbin;
   sess->session = session;
   sess->demux = demux;
-
-  sess->rtp_funnel = gst_element_factory_make ("funnel", NULL);
-  sess->rtcp_funnel = gst_element_factory_make ("funnel", NULL);
+  sess->storage = storage;
 
   sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
       (GDestroyNotify) gst_caps_unref);
@@ -821,8 +736,11 @@ create_session (GstRtpBin * rtpbin, gint id)
 
   gst_bin_add (GST_BIN_CAST (rtpbin), session);
   gst_bin_add (GST_BIN_CAST (rtpbin), demux);
-  gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtp_funnel);
-  gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtcp_funnel);
+  gst_bin_add (GST_BIN_CAST (rtpbin), storage);
+
+  /* unref the storage again, the bin has a reference now and
+   * we don't need it anymore */
+  gst_object_unref (storage);
 
   GST_OBJECT_LOCK (rtpbin);
   target = GST_STATE_TARGET (rtpbin);
@@ -831,8 +749,7 @@ create_session (GstRtpBin * rtpbin, gint id)
   /* change state only to what's needed */
   gst_element_set_state (demux, target);
   gst_element_set_state (session, target);
-  gst_element_set_state (sess->rtp_funnel, target);
-  gst_element_set_state (sess->rtcp_funnel, target);
+  gst_element_set_state (storage, target);
 
   return sess;
 
@@ -848,6 +765,13 @@ no_demux:
     g_warning ("rtpbin: could not create rtpssrcdemux element");
     return NULL;
   }
+no_storage:
+  {
+    gst_object_unref (session);
+    gst_object_unref (demux);
+    g_warning ("rtpbin: could not create rtpstorage element");
+    return NULL;
+  }
 }
 
 static gboolean
@@ -859,6 +783,10 @@ bin_manage_element (GstRtpBin * bin, GstElement * element)
     GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
   } else {
     GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
+
+    if (g_object_is_floating (element))
+      element = gst_object_ref_sink (element);
+
     if (!gst_bin_add (GST_BIN_CAST (bin), element))
       goto add_failed;
     if (!gst_element_sync_state_with_parent (element))
@@ -874,6 +802,7 @@ bin_manage_element (GstRtpBin * bin, GstElement * element)
 add_failed:
   {
     GST_WARNING_OBJECT (bin, "unable to add element");
+    gst_object_unref (element);
     return FALSE;
   }
 }
@@ -888,10 +817,13 @@ remove_bin_element (GstElement * element, GstRtpBin * bin)
   if (find) {
     priv->elements = g_list_delete_link (priv->elements, find);
 
-    if (!g_list_find (priv->elements, element))
+    if (!g_list_find (priv->elements, element)) {
+      gst_element_set_locked_state (element, TRUE);
       gst_bin_remove (GST_BIN_CAST (bin), element);
-    else
-      gst_object_unref (element);
+      gst_element_set_state (element, GST_STATE_NULL);
+    }
+
+    gst_object_unref (element);
   }
 }
 
@@ -903,9 +835,11 @@ free_session (GstRtpBinSession * sess, GstRtpBin * bin)
 
   gst_element_set_locked_state (sess->demux, TRUE);
   gst_element_set_locked_state (sess->session, TRUE);
+  gst_element_set_locked_state (sess->storage, TRUE);
 
   gst_element_set_state (sess->demux, GST_STATE_NULL);
   gst_element_set_state (sess->session, GST_STATE_NULL);
+  gst_element_set_state (sess->storage, GST_STATE_NULL);
 
   remove_recv_rtp (bin, sess);
   remove_recv_rtcp (bin, sess);
@@ -914,6 +848,7 @@ free_session (GstRtpBinSession * sess, GstRtpBin * bin)
 
   gst_bin_remove (GST_BIN_CAST (bin), sess->session);
   gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
+  gst_bin_remove (GST_BIN_CAST (bin), sess->storage);
 
   g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
   g_slist_free (sess->elements);
@@ -1106,6 +1041,43 @@ gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
 }
 
 static GstElement *
+gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
+{
+  GstRtpBinSession *session;
+  GstElement *res = NULL;
+
+  GST_RTP_BIN_LOCK (bin);
+  GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
+      session_id);
+  session = find_session_by_id (bin, (gint) session_id);
+  if (session && session->storage) {
+    res = gst_object_ref (session->storage);
+  }
+  GST_RTP_BIN_UNLOCK (bin);
+
+  return res;
+}
+
+static GObject *
+gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
+{
+  GObject *internal_storage = NULL;
+  GstRtpBinSession *session;
+
+  GST_RTP_BIN_LOCK (bin);
+  GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
+      session_id);
+  session = find_session_by_id (bin, (gint) session_id);
+  if (session && session->storage) {
+    g_object_get (session->storage, "internal-storage", &internal_storage,
+        NULL);
+  }
+  GST_RTP_BIN_UNLOCK (bin);
+
+  return internal_storage;
+}
+
+static GstElement *
 gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
 {
   GST_DEBUG_OBJECT (bin, "return NULL encoder");
@@ -1261,7 +1233,8 @@ get_current_times (GstRtpBin * bin, GstClockTime * running_time,
 
 static void
 stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
-    gint64 ts_offset, gboolean check)
+    gint64 ts_offset, gint64 max_ts_offset, gint64 min_ts_offset,
+    gboolean allow_positive_ts_offset)
 {
   gint64 prev_ts_offset;
 
@@ -1277,19 +1250,25 @@ stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
         "ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
         ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
 
-    if (check) {
-      /* only change diff when it changed more than 4 milliseconds. This
-       * compensates for rounding errors in NTP to RTP timestamp
-       * conversions */
-      if (ABS (diff) < 4 * GST_MSECOND) {
-        GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
+    /* ignore minor offsets */
+    if (ABS (diff) < min_ts_offset) {
+      GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
+      return;
+    }
+
+    /* sanity check offset */
+    if (max_ts_offset > 0) {
+      if (ts_offset > 0 && !allow_positive_ts_offset) {
+        GST_DEBUG_OBJECT (bin,
+            "offset is positive (clocks are out of sync), ignoring");
         return;
       }
-      if (ABS (diff) > (3 * GST_SECOND)) {
-        GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
+      if (ABS (ts_offset) > max_ts_offset) {
+        GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
         return;
       }
     }
+
     g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
   }
   GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
@@ -1421,13 +1400,17 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
         "local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
         local_ntpnstime, ntpnstime);
     GST_DEBUG_OBJECT (bin,
+        "local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
+        G_GUINT64_FORMAT, local_running_time, running_time);
+    GST_DEBUG_OBJECT (bin,
         "NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
         rtdiff);
 
     /* combine to get the final diff to apply to the running_time */
     stream->rt_delta = rtdiff - ntpdiff;
 
-    stream_set_ts_offset (bin, stream, stream->rt_delta, FALSE);
+    stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
+        0, FALSE);
   } else {
     gint64 min, rtp_min, clock_base = stream->clock_base;
     gboolean all_sync, use_rtp;
@@ -1579,7 +1562,8 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
       else
         ts_offset = ostream->rt_delta - min;
 
-      stream_set_ts_offset (bin, ostream, ts_offset, TRUE);
+      stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
+          MIN_TS_OFFSET, TRUE);
     }
   }
   gst_rtp_bin_send_sync_event (stream);
@@ -1723,9 +1707,10 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
   if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
     goto no_jitterbuffer;
 
-  if (!rtpbin->ignore_pt)
+  if (!rtpbin->ignore_pt) {
     if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
       goto no_demux;
+  }
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
   if (session->bin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
     if (!(queue2 = gst_element_factory_make ("queue2", NULL)))
@@ -1768,6 +1753,8 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
   g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
       "max-misorder-time", rtpbin->max_misorder_time, NULL);
   g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
+  g_object_set (buffer, "max-ts-offset-adjustment",
+      rtpbin->max_ts_offset_adjustment, NULL);
 
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
   /* configure queue2 to use live buffering */
@@ -1777,7 +1764,9 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
     g_object_set (queue2, "buffer-mode", GST_BUFFERING_LIVE, NULL);
   }
 #endif
-
+  /* need to sink the jitterbufer or otherwise signal handlers from bindings will
+   * take ownership of it and we don't own it anymore */
+  gst_object_ref_sink (buffer);
   g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
       buffer, session->id, ssrc);
 
@@ -1791,6 +1780,10 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
 
   gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
 
+  /* unref the jitterbuffer again, the bin has a reference now and
+   * we don't need it anymore */
+  gst_object_unref (buffer);
+
   /* link stuff */
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
   if (queue2) {
@@ -1800,10 +1793,10 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
       gst_element_link_pads_full (queue2, "src", demux, "sink",
           GST_PAD_LINK_CHECK_NOTHING);
     }
-  } else if (demux)
-     gst_element_link_pads_full (buffer, "src", demux, "sink",
-         GST_PAD_LINK_CHECK_NOTHING);
-
+  } else if (demux) {
+    gst_element_link_pads_full (buffer, "src", demux, "sink",
+        GST_PAD_LINK_CHECK_NOTHING);
+  }
 #else
   if (demux)
     gst_element_link_pads_full (buffer, "src", demux, "sink",
@@ -1942,7 +1935,7 @@ static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
 static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
 
 #define gst_rtp_bin_parent_class parent_class
-G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
+G_DEFINE_TYPE_WITH_PRIVATE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
 
 static gboolean
 _gst_element_accumulator (GSignalInvocationHint * ihint,
@@ -1987,8 +1980,6 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
   gstelement_class = (GstElementClass *) klass;
   gstbin_class = (GstBinClass *) klass;
 
-  g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
-
   gobject_class->dispose = gst_rtp_bin_dispose;
   gobject_class->finalize = gst_rtp_bin_finalize;
   gobject_class->set_property = gst_rtp_bin_set_property;
@@ -2089,6 +2080,36 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
       RTP_TYPE_SESSION, 1, G_TYPE_UINT);
 
   /**
+   * GstRtpBin::get-internal-storage:
+   * @rtpbin: the object which received the signal
+   * @id: the session id
+   *
+   * Request the internal RTPStorage object as #GObject in session @id.
+   *
+   * Since: 1.14
+   */
+  gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
+      g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+          get_internal_storage), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_OBJECT, 1, G_TYPE_UINT);
+
+  /**
+   * GstRtpBin::get-storage:
+   * @rtpbin: the object which received the signal
+   * @id: the session id
+   *
+   * Request the RTPStorage element as #GObject in session @id.
+   *
+   * Since: 1.16
+   */
+  gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
+      g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+          get_storage), NULL, NULL, g_cclosure_marshal_generic,
+      GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+  /**
    * GstRtpBin::on-new-ssrc:
    * @rtpbin: the object which received the signal
    * @session: the session
@@ -2312,6 +2333,23 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
       G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
 
   /**
+   * GstRtpBin::new-storage:
+   * @rtpbin: the object which received the signal
+   * @storage: the new storage
+   * @session: the session
+   *
+   * Notify that a new @storage was created for @session.
+   * This signal can, for example, be used to configure @storage.
+   *
+   * Since: 1.14
+   */
+  gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
+      g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+          new_storage), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
+
+  /**
    * GstRtpBin::request-aux-sender:
    * @rtpbin: the object which received the signal
    * @session: the session
@@ -2328,6 +2366,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
           request_aux_sender), _gst_element_accumulator, NULL,
       g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
   /**
    * GstRtpBin::request-aux-receiver:
    * @rtpbin: the object which received the signal
@@ -2345,6 +2384,43 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
           request_aux_receiver), _gst_element_accumulator, NULL,
       g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+  /**
+   * GstRtpBin::request-fec-decoder:
+   * @rtpbin: the object which received the signal
+   * @session: the session index
+   *
+   * Request a FEC decoder element for the given @session. The element
+   * will be added to the bin after the pt demuxer.
+   *
+   * If no handler is connected, no FEC decoder will be used.
+   *
+   * Since: 1.14
+   */
+  gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
+      g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+          request_fec_decoder), _gst_element_accumulator, NULL,
+      g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+  /**
+   * GstRtpBin::request-fec-encoder:
+   * @rtpbin: the object which received the signal
+   * @session: the session index
+   *
+   * Request a FEC encoder element for the given @session. The element
+   * will be added to the bin after the RTPSession.
+   *
+   * If no handler is connected, no FEC encoder will be used.
+   *
+   * Since: 1.14
+   */
+  gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
+      g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+          request_fec_encoder), _gst_element_accumulator, NULL,
+      g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
   /**
    * GstRtpBin::on-new-sender-ssrc:
    * @rtpbin: the object which received the signal
@@ -2376,29 +2452,6 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
           on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
       G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
 
-
-  /**
-   * GstRtpBin::on-bundled-ssrc:
-   * @rtpbin: the object which received the signal
-   * @ssrc: the bundled SSRC
-   *
-   * Notify of a new incoming bundled SSRC. If no handler is connected to the
-   * signal then the #GstRtpSession created for the recv_rtp_sink_\%u
-   * request pad will be managing this new SSRC. However if there is a handler
-   * connected then the application can decided to dispatch this new stream to
-   * another session by providing its ID as return value of the handler. This
-   * can be particularly useful to keep retransmission SSRCs grouped with the
-   * session for which they handle retransmission.
-   *
-   * Since: 1.12
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC] =
-      g_signal_new ("on-bundled-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
-          on_bundled_ssrc), NULL, NULL,
-      g_cclosure_marshal_generic, G_TYPE_UINT, 1, G_TYPE_UINT);
-
-
   g_object_class_install_property (gobject_class, PROP_SDES,
       g_param_spec_boxed ("sdes", "SDES",
           "The SDES items of this session",
@@ -2545,6 +2598,40 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
           0, G_MAXUINT, DEFAULT_MAX_STREAMS,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstRtpBin:max-ts-offset-adjustment:
+   *
+   * Syncing time stamps to NTP time adds a time offset. This parameter
+   * specifies the maximum number of nanoseconds per frame that this time offset
+   * may be adjusted with. This is used to avoid sudden large changes to time
+   * stamps.
+   *
+   * Since: 1.14
+   */
+  g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
+      g_param_spec_uint64 ("max-ts-offset-adjustment",
+          "Max Timestamp Offset Adjustment",
+          "The maximum number of nanoseconds per frame that time stamp offsets "
+          "may be adjusted (0 = no limit).", 0, G_MAXUINT64,
+          DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
+          G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRtpBin:max-ts-offset:
+   *
+   * Used to set an upper limit of how large a time offset may be. This
+   * is used to protect against unrealistic values as a result of either
+   * client,server or clock issues.
+   *
+   * Since: 1.14
+   */
+  g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
+      g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
+          "The maximum absolute value of the time offset in (nanoseconds). "
+          "Note, if the ntp-sync parameter is set the default value is "
+          "changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
   gstelement_class->request_new_pad =
       GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
@@ -2578,6 +2665,9 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
   klass->get_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
   klass->get_internal_session =
       GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
+  klass->get_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
+  klass->get_internal_storage =
+      GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
   klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
   klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
   klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
@@ -2591,7 +2681,7 @@ gst_rtp_bin_init (GstRtpBin * rtpbin)
 {
   gchar *cname;
 
-  rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
+  rtpbin->priv = gst_rtp_bin_get_instance_private (rtpbin);
   g_mutex_init (&rtpbin->priv->bin_lock);
   g_mutex_init (&rtpbin->priv->dyn_lock);
 
@@ -2616,6 +2706,9 @@ gst_rtp_bin_init (GstRtpBin * rtpbin)
   rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
   rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
   rtpbin->max_streams = DEFAULT_MAX_STREAMS;
+  rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
+  rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
+  rtpbin->max_ts_offset_is_set = FALSE;
 
   /* some default SDES entries */
   cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
@@ -2731,6 +2824,15 @@ gst_rtp_bin_set_property (GObject * object, guint prop_id,
       break;
     case PROP_NTP_SYNC:
       rtpbin->ntp_sync = g_value_get_boolean (value);
+      /* The default value of max_ts_offset depends on ntp_sync. If user
+       * hasn't set it then change default value */
+      if (!rtpbin->max_ts_offset_is_set) {
+        if (rtpbin->ntp_sync) {
+          rtpbin->max_ts_offset = 0;
+        } else {
+          rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
+        }
+      }
       break;
     case PROP_RTCP_SYNC:
       g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
@@ -2840,6 +2942,15 @@ gst_rtp_bin_set_property (GObject * object, guint prop_id,
     case PROP_MAX_STREAMS:
       rtpbin->max_streams = g_value_get_uint (value);
       break;
+    case PROP_MAX_TS_OFFSET_ADJUSTMENT:
+      rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
+      gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+          "max-ts-offset-adjustment", value);
+      break;
+    case PROP_MAX_TS_OFFSET:
+      rtpbin->max_ts_offset = g_value_get_int64 (value);
+      rtpbin->max_ts_offset_is_set = TRUE;
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -2928,6 +3039,12 @@ gst_rtp_bin_get_property (GObject * object, guint prop_id,
     case PROP_MAX_STREAMS:
       g_value_set_uint (value, rtpbin->max_streams);
       break;
+    case PROP_MAX_TS_OFFSET_ADJUSTMENT:
+      g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
+      break;
+    case PROP_MAX_TS_OFFSET:
+      g_value_set_int64 (value, rtpbin->max_ts_offset);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -3011,39 +3128,35 @@ gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
               GstStructure *caps_structure;
               const gchar *caps_str_media = NULL;
               temp_pad_src = gst_element_get_static_pad (stream->buffer, "src");
-              temp_caps_src = gst_pad_get_current_caps(temp_pad_src);
-              GST_DEBUG_OBJECT (bin, "stream %p percent %d : temp_caps_src=%"GST_PTR_FORMAT, stream,stream->percent,temp_caps_src);
-              if (temp_caps_src)
-              {
+              temp_caps_src = gst_pad_get_current_caps (temp_pad_src);
+              GST_DEBUG_OBJECT (bin,
+                  "stream %p percent %d : temp_caps_src=%" GST_PTR_FORMAT,
+                  stream, stream->percent, temp_caps_src);
+              if (temp_caps_src) {
                 caps_structure = gst_caps_get_structure (temp_caps_src, 0);
-                caps_str_media = gst_structure_get_string (caps_structure, "media");
-                if (caps_str_media != NULL)
-                {
-                  if ((strcmp(caps_str_media,"video") != 0)&&(strcmp(caps_str_media,"audio") != 0))
-                  {
-                    GST_DEBUG_OBJECT (bin, "Non Audio/Video Stream.. ignoring the same !!");
-                    gst_caps_unref( temp_caps_src );
-                    gst_object_unref( temp_pad_src );
+                caps_str_media =
+                    gst_structure_get_string (caps_structure, "media");
+                if (caps_str_media != NULL) {
+                  if ((strcmp (caps_str_media, "video") != 0)
+                      && (strcmp (caps_str_media, "audio") != 0)) {
+                    GST_DEBUG_OBJECT (bin,
+                        "Non Audio/Video Stream.. ignoring the same !!");
+                    gst_caps_unref (temp_caps_src);
+                    gst_object_unref (temp_pad_src);
                     continue;
-                  }
-                  else if(stream->percent >= 100)
-                  {
+                  } else if (stream->percent >= 100) {
                     /* Most of the time buffering icon displays in rtsp playback.
-                      Optimizing the buffering updation code. Whenever any stream percentage
-                      reaches 100 do not post buffering messages.*/
-                    if(stream->prev_percent < 100)
-                    {
+                       Optimizing the buffering updation code. Whenever any stream percentage
+                       reaches 100 do not post buffering messages. */
+                    if (stream->prev_percent < 100)
                       buffering_flag = TRUE;
-                    }
                     else
-                    {
                       update_buffering_status = FALSE;
-                    }
                   }
                 }
-                gst_caps_unref( temp_caps_src );
+                gst_caps_unref (temp_caps_src);
               }
-              gst_object_unref( temp_pad_src );
+              gst_object_unref (temp_pad_src);
 #else
               GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
                   stream->percent);
@@ -3102,18 +3215,14 @@ gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
         gst_message_unref (message);
 
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
-      if (rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
-      {
-	  if(update_buffering_status==FALSE)
-	  {
-	  	break;
-	  }
-	  if(buffering_flag)
-	  {
-	  	min_percent=100;
-	  	GST_DEBUG_OBJECT (bin, "forcefully change min_percent to 100!!!");
-	  }
-      }
+        if (rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
+          if (update_buffering_status == FALSE)
+            break;
+          if (buffering_flag) {
+            min_percent = 100;
+            GST_DEBUG_OBJECT (bin, "forcefully change min_percent to 100!!!");
+          }
+        }
 #endif
         /* make a new buffering message with the min value */
         message =
@@ -3297,21 +3406,46 @@ copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
   return TRUE;
 }
 
-/* a new pad (SSRC) was created in @session. This signal is emited from the
- * payload demuxer. */
 static void
-new_payload_found (GstElement * element, guint pt, GstPad * pad,
-    GstRtpBinStream * stream)
+expose_recv_src_pad (GstRtpBin * rtpbin, GstPad * pad, GstRtpBinStream * stream,
+    guint8 pt)
 {
-  GstRtpBin *rtpbin;
   GstElementClass *klass;
   GstPadTemplate *templ;
   gchar *padname;
   GstPad *gpad;
 
-  rtpbin = stream->bin;
+  gst_object_ref (pad);
 
-  GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
+  if (stream->session->storage) {
+    GstElement *fec_decoder =
+        session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
+
+    if (fec_decoder) {
+      GstPad *sinkpad, *srcpad;
+      GstPadLinkReturn ret;
+
+      sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
+
+      if (!sinkpad)
+        goto fec_decoder_sink_failed;
+
+      ret = gst_pad_link (pad, sinkpad);
+      gst_object_unref (sinkpad);
+
+      if (ret != GST_PAD_LINK_OK)
+        goto fec_decoder_link_failed;
+
+      srcpad = gst_element_get_static_pad (fec_decoder, "src");
+
+      if (!srcpad)
+        goto fec_decoder_src_failed;
+
+      gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
+      gst_object_unref (pad);
+      pad = srcpad;
+    }
+  }
 
   GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
 
@@ -3330,15 +3464,51 @@ new_payload_found (GstElement * element, guint pt, GstPad * pad,
   gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
   gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
 
+done:
+  gst_object_unref (pad);
+
   return;
 
 shutdown:
   {
     GST_DEBUG ("ignoring, we are shutting down");
-    return;
+    goto done;
+  }
+fec_decoder_sink_failed:
+  {
+    g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
+        stream->session->id);
+    goto done;
+  }
+fec_decoder_src_failed:
+  {
+    g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
+        stream->session->id);
+    goto done;
+  }
+fec_decoder_link_failed:
+  {
+    g_warning ("rtpbin: failed to link fec decoder for session %u",
+        stream->session->id);
+    goto done;
   }
 }
 
+/* a new pad (SSRC) was created in @session. This signal is emited from the
+ * payload demuxer. */
+static void
+new_payload_found (GstElement * element, guint pt, GstPad * pad,
+    GstRtpBinStream * stream)
+{
+  GstRtpBin *rtpbin;
+
+  rtpbin = stream->bin;
+
+  GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
+
+  expose_recv_src_pad (rtpbin, pad, stream, pt);
+}
+
 static void
 payload_pad_removed (GstElement * element, GstPad * pad,
     GstRtpBinStream * stream)
@@ -3385,6 +3555,36 @@ no_caps:
   }
 }
 
+static GstCaps *
+ptdemux_pt_map_requested (GstElement * element, guint pt,
+    GstRtpBinSession * session)
+{
+  GstCaps *ret = pt_map_requested (element, pt, session);
+
+  if (ret && gst_caps_get_size (ret) == 1) {
+    const GstStructure *s = gst_caps_get_structure (ret, 0);
+    gboolean is_fec;
+
+    if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
+      GValue v = G_VALUE_INIT;
+      GValue v2 = G_VALUE_INIT;
+
+      GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
+          session->id);
+      g_value_init (&v, GST_TYPE_ARRAY);
+      g_value_init (&v2, G_TYPE_INT);
+      g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
+      g_value_set_int (&v2, pt);
+      gst_value_array_append_value (&v, &v2);
+      g_value_unset (&v2);
+      g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
+      g_value_unset (&v);
+    }
+  }
+
+  return ret;
+}
+
 static void
 payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
 {
@@ -3396,7 +3596,7 @@ payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
       0, session->id, pt);
 }
 
-/* emited when caps changed for the session */
+/* emitted when caps changed for the session */
 static void
 caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
 {
@@ -3484,41 +3684,31 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
     stream->demux_padremoved_sig = g_signal_connect (stream->demux,
         "pad-removed", (GCallback) payload_pad_removed, stream);
 
-    /* connect to the request-pt-map signal. This signal will be emited by the
+    /* connect to the request-pt-map signal. This signal will be emitted by the
      * demuxer so that it can apply a proper caps on the buffers for the
      * depayloaders. */
     stream->demux_ptreq_sig = g_signal_connect (stream->demux,
-        "request-pt-map", (GCallback) pt_map_requested, session);
+        "request-pt-map", (GCallback) ptdemux_pt_map_requested, session);
     /* connect to the  signal so it can be forwarded. */
     stream->demux_ptchange_sig = g_signal_connect (stream->demux,
         "payload-type-change", (GCallback) payload_type_change, session);
+
+    GST_RTP_SESSION_UNLOCK (session);
+    GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
   } else {
     /* add rtpjitterbuffer src pad to pads */
-    GstElementClass *klass;
-    GstPadTemplate *templ;
-    gchar *padname;
-    GstPad *gpad, *pad;
+    GstPad *pad;
 
     pad = gst_element_get_static_pad (stream->buffer, "src");
 
-    /* ghost the pad to the parent */
-    klass = GST_ELEMENT_GET_CLASS (rtpbin);
-    templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
-    padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
-        stream->session->id, stream->ssrc, 255);
-    gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
-    g_free (padname);
+    GST_RTP_SESSION_UNLOCK (session);
+    GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
 
-    gst_pad_set_active (gpad, TRUE);
-    gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
-    gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
+    expose_recv_src_pad (rtpbin, pad, stream, 255);
 
     gst_object_unref (pad);
   }
 
-  GST_RTP_SESSION_UNLOCK (session);
-  GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
-
   return;
 
   /* ERRORS */
@@ -3536,39 +3726,11 @@ no_stream:
   }
 }
 
-static void
-session_maybe_create_bundle_demuxer (GstRtpBinSession * session)
-{
-  GstRtpBin *rtpbin;
-
-  if (session->bundle_demux)
-    return;
-
-  rtpbin = session->bin;
-  if (g_signal_has_handler_pending (rtpbin,
-          gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, TRUE)) {
-    GST_DEBUG_OBJECT (rtpbin, "Adding a bundle SSRC demuxer to session %u",
-        session->id);
-    session->bundle_demux = gst_element_factory_make ("rtpssrcdemux", NULL);
-    session->bundle_demux_newpad_sig = g_signal_connect (session->bundle_demux,
-        "new-ssrc-pad", (GCallback) new_bundled_ssrc_pad_found, session);
-
-    gst_bin_add (GST_BIN_CAST (rtpbin), session->bundle_demux);
-    gst_element_sync_state_with_parent (session->bundle_demux);
-  } else {
-    GST_DEBUG_OBJECT (rtpbin,
-        "No handler for the on-bundled-ssrc signal so no need for a bundle SSRC demuxer in session %u",
-        session->id);
-  }
-}
-
 static GstPad *
-complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
-    gboolean bundle_demuxer_needed)
+complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
 {
   guint sessid = session->id;
   GstPad *recv_rtp_sink;
-  GstPad *funnel_src;
   GstElement *decoder;
 
   g_assert (!session->recv_rtp_sink);
@@ -3582,9 +3744,6 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
   g_signal_connect (session->recv_rtp_sink, "notify::caps",
       (GCallback) caps_changed, session);
 
-  if (bundle_demuxer_needed)
-    session_maybe_create_bundle_demuxer (session);
-
   GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
   decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
   if (decoder) {
@@ -3602,14 +3761,8 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
     if (decsrc == NULL)
       goto dec_src_failed;
 
-    if (session->bundle_demux) {
-      GstPad *demux_sink;
-      demux_sink = gst_element_get_static_pad (session->bundle_demux, "sink");
-      ret = gst_pad_link (decsrc, demux_sink);
-      gst_object_unref (demux_sink);
-    } else {
-      ret = gst_pad_link (decsrc, session->recv_rtp_sink);
-    }
+    ret = gst_pad_link (decsrc, session->recv_rtp_sink);
+
     gst_object_unref (decsrc);
 
     if (ret != GST_PAD_LINK_OK)
@@ -3617,19 +3770,9 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
 
   } else {
     GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
-    if (session->bundle_demux) {
-      recv_rtp_sink =
-          gst_element_get_static_pad (session->bundle_demux, "sink");
-    } else {
-      recv_rtp_sink =
-          gst_element_get_request_pad (session->rtp_funnel, "sink_%u");
-    }
+    recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
   }
 
-  funnel_src = gst_element_get_static_pad (session->rtp_funnel, "src");
-  gst_pad_link (funnel_src, session->recv_rtp_sink);
-  gst_object_unref (funnel_src);
-
   return recv_rtp_sink;
 
   /* ERRORS */
@@ -3671,8 +3814,7 @@ complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
   if (session->recv_rtp_src == NULL)
     goto pad_failed;
 
-  /* find out if we need AUX elements or if we can go into the SSRC demuxer
-   * directly */
+  /* find out if we need AUX elements */
   aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
   if (aux) {
     gchar *pname;
@@ -3692,8 +3834,7 @@ complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
     if (ret != GST_PAD_LINK_OK)
       goto aux_link_failed;
 
-    /* this can be NULL when this AUX element is not to be linked to
-     * an SSRC demuxer */
+    /* this can be NULL when this AUX element is not to be linked any further */
     pname = g_strdup_printf ("src_%u", sessid);
     recv_rtp_src = gst_element_get_static_pad (aux, pname);
     g_free (pname);
@@ -3701,6 +3842,22 @@ complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
     recv_rtp_src = gst_object_ref (session->recv_rtp_src);
   }
 
+  /* Add a storage element if needed */
+  if (recv_rtp_src && session->storage) {
+    GstPadLinkReturn ret;
+    GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
+
+    ret = gst_pad_link (recv_rtp_src, sinkpad);
+
+    gst_object_unref (sinkpad);
+    gst_object_unref (recv_rtp_src);
+
+    if (ret != GST_PAD_LINK_OK)
+      goto storage_link_failed;
+
+    recv_rtp_src = gst_element_get_static_pad (session->storage, "src");
+  }
+
   if (recv_rtp_src) {
     GstPad *sinkdpad;
 
@@ -3735,6 +3892,11 @@ aux_link_failed:
     g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
     return;
   }
+storage_link_failed:
+  {
+    g_warning ("rtpbin: failed to link storage");
+    return;
+  }
 }
 
 /* Create a pad for receiving RTP for the session in @name. Must be called with
@@ -3768,11 +3930,10 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
     return session->recv_rtp_sink_ghost;
 
   /* setup the session sink pad */
-  recv_rtp_sink = complete_session_sink (rtpbin, session, TRUE);
+  recv_rtp_sink = complete_session_sink (rtpbin, session);
   if (!recv_rtp_sink)
     goto session_sink_failed;
 
-
   GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
   session->recv_rtp_sink_ghost =
       gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
@@ -3813,11 +3974,6 @@ remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
     g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
     session->demux_padremoved_sig = 0;
   }
-  if (session->bundle_demux_newpad_sig) {
-    g_signal_handler_disconnect (session->bundle_demux,
-        session->bundle_demux_newpad_sig);
-    session->bundle_demux_newpad_sig = 0;
-  }
   if (session->recv_rtp_src) {
     gst_object_unref (session->recv_rtp_src);
     session->recv_rtp_src = NULL;
@@ -3837,12 +3993,11 @@ remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
 
 static GstPad *
 complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
-    guint sessid, gboolean bundle_demuxer_needed)
+    guint sessid)
 {
   GstElement *decoder;
   GstPad *sinkdpad;
   GstPad *decsink = NULL;
-  GstPad *funnel_src;
 
   /* get recv_rtp pad and store */
   GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
@@ -3851,9 +4006,6 @@ complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
   if (session->recv_rtcp_sink == NULL)
     goto pad_failed;
 
-  if (bundle_demuxer_needed)
-    session_maybe_create_bundle_demuxer (session);
-
   GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
   decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
   if (decoder) {
@@ -3870,26 +4022,15 @@ complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
     if (decsrc == NULL)
       goto dec_src_failed;
 
-    if (session->bundle_demux) {
-      GstPad *demux_sink;
-      demux_sink =
-          gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
-      ret = gst_pad_link (decsrc, demux_sink);
-      gst_object_unref (demux_sink);
-    } else {
-      ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
-    }
+    ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
+
     gst_object_unref (decsrc);
 
     if (ret != GST_PAD_LINK_OK)
       goto dec_link_failed;
   } else {
     GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
-    if (session->bundle_demux) {
-      decsink = gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
-    } else {
-      decsink = gst_element_get_request_pad (session->rtcp_funnel, "sink_%u");
-    }
+    decsink = gst_object_ref (session->recv_rtcp_sink);
   }
 
   /* get srcpad, link to SSRCDemux */
@@ -3903,10 +4044,6 @@ complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
   gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
   gst_object_unref (sinkdpad);
 
-  funnel_src = gst_element_get_static_pad (session->rtcp_funnel, "src");
-  gst_pad_link (funnel_src, session->recv_rtcp_sink);
-  gst_object_unref (funnel_src);
-
   return decsink;
 
 pad_failed:
@@ -3970,7 +4107,7 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
   if (session->recv_rtcp_sink_ghost != NULL)
     return session->recv_rtcp_sink_ghost;
 
-  decsink = complete_session_rtcp (rtpbin, session, sessid, TRUE);
+  decsink = complete_session_rtcp (rtpbin, session, sessid);
   if (!decsink)
     goto create_error;
 
@@ -4026,11 +4163,12 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
   GstElement *encoder;
   GstElementClass *klass;
   GstPadTemplate *templ;
+  gboolean ret = FALSE;
 
   /* get srcpad */
-  session->send_rtp_src =
-      gst_element_get_static_pad (session->session, "send_rtp_src");
-  if (session->send_rtp_src == NULL)
+  send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
+
+  if (send_rtp_src == NULL)
     goto no_srcpad;
 
   GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
@@ -4048,22 +4186,22 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
     if (encsrc == NULL)
       goto enc_src_failed;
 
-    send_rtp_src = encsrc;
-
     ename = g_strdup_printf ("rtp_sink_%u", sessid);
     encsink = gst_element_get_static_pad (encoder, ename);
     g_free (ename);
     if (encsink == NULL)
       goto enc_sink_failed;
 
-    ret = gst_pad_link (session->send_rtp_src, encsink);
+    ret = gst_pad_link (send_rtp_src, encsink);
     gst_object_unref (encsink);
+    gst_object_unref (send_rtp_src);
+
+    send_rtp_src = encsrc;
 
     if (ret != GST_PAD_LINK_OK)
       goto enc_link_failed;
   } else {
     GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
-    send_rtp_src = gst_object_ref (session->send_rtp_src);
   }
 
   /* ghost the new source pad */
@@ -4072,37 +4210,43 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
   templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
   session->send_rtp_src_ghost =
       gst_ghost_pad_new_from_template (gname, send_rtp_src, templ);
-  gst_object_unref (send_rtp_src);
   gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
   gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
       session->send_rtp_src_ghost);
   gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
   g_free (gname);
 
-  return TRUE;
+  ret = TRUE;
+
+done:
+  if (send_rtp_src)
+    gst_object_unref (send_rtp_src);
+
+  return ret;
 
   /* ERRORS */
 no_srcpad:
   {
     g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
-    return FALSE;
+    goto done;
   }
 enc_src_failed:
   {
-    g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
-    return FALSE;
+    g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+        " src pad for session %u", encoder, sessid);
+    goto done;
   }
 enc_sink_failed:
   {
-    g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
-    gst_object_unref (send_rtp_src);
-    return FALSE;
+    g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+        " sink pad for session %u", encoder, sessid);
+    goto done;
   }
 enc_link_failed:
   {
-    g_warning ("rtpbin: failed to link rtp encoder for session %u", sessid);
-    gst_object_unref (send_rtp_src);
-    return FALSE;
+    g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+        encoder, sessid);
+    goto done;
   }
 }
 
@@ -4162,8 +4306,9 @@ create_error:
   }
 existing_session:
   {
-    g_warning ("rtpbin: session %u is already a sender", sessid);
-    return FALSE;
+    GST_DEBUG_OBJECT (rtpbin,
+        "skipping src_%i setup, since it is already configured.", sessid);
+    return TRUE;
   }
 pad_failed:
   {
@@ -4207,6 +4352,8 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   guint sessid;
   GstPad *send_rtp_sink;
   GstElement *aux;
+  GstElement *encoder;
+  GstElement *prev = NULL;
   GstRtpBinSession *session;
 
   /* first get the session number */
@@ -4230,19 +4377,47 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   if (session->send_rtp_sink != NULL)
     goto existing_session;
 
+  encoder = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
+
+  if (encoder) {
+    GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
+
+    send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
+
+    if (!send_rtp_sink)
+      goto enc_sink_failed;
+
+    prev = encoder;
+  }
+
   GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
   aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
   if (aux) {
+    GstPad *sinkpad;
     GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
     if (!setup_aux_sender (rtpbin, session, aux))
       goto aux_session_failed;
 
     pname = g_strdup_printf ("sink_%u", sessid);
-    send_rtp_sink = gst_element_get_static_pad (aux, pname);
+    sinkpad = gst_element_get_static_pad (aux, pname);
     g_free (pname);
 
-    if (send_rtp_sink == NULL)
+    if (sinkpad == NULL)
       goto aux_sink_failed;
+
+    if (!prev) {
+      send_rtp_sink = sinkpad;
+    } else {
+      GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+      GstPadLinkReturn ret;
+
+      ret = gst_pad_link (srcpad, sinkpad);
+      gst_object_unref (srcpad);
+      if (ret != GST_PAD_LINK_OK) {
+        goto aux_link_failed;
+      }
+    }
+    prev = aux;
   } else {
     /* get send_rtp pad and store */
     session->send_rtp_sink =
@@ -4253,7 +4428,17 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
     if (!complete_session_src (rtpbin, session))
       goto session_src_failed;
 
-    send_rtp_sink = gst_object_ref (session->send_rtp_sink);
+    if (!prev) {
+      send_rtp_sink = gst_object_ref (session->send_rtp_sink);
+    } else {
+      GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+      GstPadLinkReturn ret;
+
+      ret = gst_pad_link (srcpad, session->send_rtp_sink);
+      gst_object_unref (srcpad);
+      if (ret != GST_PAD_LINK_OK)
+        goto session_link_failed;
+    }
   }
 
   session->send_rtp_sink_ghost =
@@ -4290,6 +4475,12 @@ aux_sink_failed:
     g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
     return NULL;
   }
+aux_link_failed:
+  {
+    g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+        aux, sessid);
+    return NULL;
+  }
 pad_failed:
   {
     g_warning ("rtpbin: failed to get session pad for session %u", sessid);
@@ -4300,6 +4491,18 @@ session_src_failed:
     g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
     return NULL;
   }
+session_link_failed:
+  {
+    g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+        session, sessid);
+    return NULL;
+  }
+enc_sink_failed:
+  {
+    g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+        " sink pad for session %u", encoder, sessid);
+    return NULL;
+  }
 }
 
 static void
@@ -4311,10 +4514,6 @@ remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
         session->send_rtp_src_ghost);
     session->send_rtp_src_ghost = NULL;
   }
-  if (session->send_rtp_src) {
-    gst_object_unref (session->send_rtp_src);
-    session->send_rtp_src = NULL;
-  }
   if (session->send_rtp_sink) {
     gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
         session->send_rtp_sink);
@@ -4333,7 +4532,8 @@ remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
  * RTP_BIN_LOCK.
  */
 static GstPad *
-create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+create_send_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
+    const gchar * name)
 {
   guint sessid;
   GstPad *encsrc;
@@ -4346,8 +4546,13 @@ create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
 
   /* get or create session */
   session = find_session_by_id (rtpbin, sessid);
-  if (!session)
-    goto no_session;
+  if (!session) {
+    GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
+    /* create session now */
+    session = create_session (rtpbin, sessid);
+    if (session == NULL)
+      goto create_error;
+  }
 
   /* check if pad was requested */
   if (session->send_rtcp_src_ghost != NULL)
@@ -4404,9 +4609,9 @@ no_name:
     g_warning ("rtpbin: invalid name given");
     return NULL;
   }
-no_session:
+create_error:
   {
-    g_warning ("rtpbin: session with id %d does not exist", sessid);
+    /* create_session already warned */
     return NULL;
   }
 pad_failed:
@@ -4548,7 +4753,7 @@ gst_rtp_bin_request_new_pad (GstElement * element,
     result = create_send_rtp (rtpbin, templ, pad_name);
   } else if (templ == gst_element_class_get_pad_template (klass,
           "send_rtcp_src_%u")) {
-    result = create_rtcp (rtpbin, templ, pad_name);
+    result = create_send_rtcp (rtpbin, templ, pad_name);
   } else
     goto wrong_template;