11 files changed, 1265 insertions, 160 deletions

diff --git a/docs/plugins/gst-plugins-good-plugins.signals b/docs/plugins/gst-plugins-good-plugins.signals
index 3db17e912..44bbddad1 100644
--- a/docs/plugins/gst-plugins-good-plugins.signals
+++ b/docs/plugins/gst-plugins-good-plugins.signals
@@ -375,6 +375,14 @@ guint  arg1
 </SIGNAL>
 
 <SIGNAL>
+<NAME>GstRtpBin::on-bundled-ssrc</NAME>
+<RETURNS>guint</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
 <NAME>GstRtpJitterBuffer::clear-pt-map</NAME>
 <RETURNS>void</RETURNS>
 <FLAGS>la</FLAGS>
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 648adb9f8..f58de0116 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -53,6 +53,13 @@
  * 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
@@ -276,6 +283,8 @@ enum
   SIGNAL_ON_NEW_SENDER_SSRC,
   SIGNAL_ON_SENDER_SSRC_ACTIVE,
 
+  SIGNAL_ON_BUNDLED_SSRC,
+
   LAST_SIGNAL
 };
 
@@ -362,6 +371,14 @@ static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
 static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
 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);
+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);
 
 /* Manages the RTP stream for one SSRC.
  *
@@ -428,6 +445,12 @@ 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;
+
   GMutex lock;
 
   /* list of GstRtpBinStream */
@@ -629,6 +652,96 @@ 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->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)
@@ -649,6 +762,10 @@ create_session (GstRtpBin * rtpbin, gint 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->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
       (GDestroyNotify) gst_caps_unref);
   rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
@@ -696,6 +813,8 @@ 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_OBJECT_LOCK (rtpbin);
   target = GST_STATE_TARGET (rtpbin);
@@ -704,6 +823,8 @@ 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);
 
   return sess;
 
@@ -807,7 +928,7 @@ get_pt_map (GstRtpBinSession * session, guint pt)
   GValue ret = { 0 };
   GValue args[3] = { {0}, {0}, {0} };
 
-  GST_DEBUG ("searching pt %d in cache", pt);
+  GST_DEBUG ("searching pt %u in cache", pt);
 
   GST_RTP_SESSION_LOCK (session);
 
@@ -820,7 +941,7 @@ get_pt_map (GstRtpBinSession * session, guint pt)
 
   bin = session->bin;
 
-  GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
+  GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
 
   /* not in cache, send signal to request caps */
   g_value_init (&args[0], GST_TYPE_ELEMENT);
@@ -856,7 +977,7 @@ get_pt_map (GstRtpBinSession * session, guint pt)
   if (!caps)
     goto no_caps;
 
-  GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
+  GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
 
   /* store in cache, take additional ref */
   g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
@@ -947,7 +1068,7 @@ gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
   GstElement *ret = NULL;
 
   GST_RTP_BIN_LOCK (bin);
-  GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %d", session_id);
+  GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
   session = find_session_by_id (bin, (gint) session_id);
   if (session) {
     ret = gst_object_ref (session->session);
@@ -964,7 +1085,7 @@ gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
   GstRtpBinSession *session;
 
   GST_RTP_BIN_LOCK (bin);
-  GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
+  GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
       session_id);
   session = find_session_by_id (bin, (gint) session_id);
   if (session) {
@@ -2194,6 +2315,29 @@ 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",
@@ -3021,7 +3165,7 @@ new_payload_found (GstElement * element, guint pt, GstPad * pad,
 
   rtpbin = stream->bin;
 
-  GST_DEBUG ("new payload pad %d", pt);
+  GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
 
   GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
 
@@ -3078,7 +3222,7 @@ pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
 
   rtpbin = session->bin;
 
-  GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
+  GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
       session->id);
 
   caps = get_pt_map (session, pt);
@@ -3099,7 +3243,7 @@ static void
 payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
 {
   GST_DEBUG_OBJECT (session->bin,
-      "emiting signal for pt type changed to %d in session %d", pt,
+      "emiting signal for pt type changed to %u in session %u", pt,
       session->id);
 
   g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
@@ -3246,15 +3390,42 @@ no_stream:
   }
 }
 
-static gboolean
-complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
+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)
 {
-  gchar *gname;
   guint sessid = session->id;
   GstPad *recv_rtp_sink;
+  GstPad *funnel_src;
   GstElement *decoder;
-  GstElementClass *klass;
-  GstPadTemplate *templ;
+
+  g_assert (!session->recv_rtp_sink);
 
   /* get recv_rtp pad and store */
   session->recv_rtp_sink =
@@ -3265,6 +3436,9 @@ 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) {
@@ -3282,7 +3456,14 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
     if (decsrc == NULL)
       goto dec_src_failed;
 
-    ret = gst_pad_link (decsrc, session->recv_rtp_sink);
+    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);
+    }
     gst_object_unref (decsrc);
 
     if (ret != GST_PAD_LINK_OK)
@@ -3290,81 +3471,54 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
 
   } else {
     GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
-    recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
+    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");
+    }
   }
 
-  GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
-  klass = GST_ELEMENT_GET_CLASS (rtpbin);
-  gname = g_strdup_printf ("recv_rtp_sink_%u", sessid);
-  templ = gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u");
-  session->recv_rtp_sink_ghost =
-      gst_ghost_pad_new_from_template (gname, recv_rtp_sink, templ);
-  gst_object_unref (recv_rtp_sink);
-  gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
-  g_free (gname);
+  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 TRUE;
+  return recv_rtp_sink;
 
   /* ERRORS */
 pad_failed:
   {
     g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
-    return FALSE;
+    return NULL;
   }
 dec_sink_failed:
   {
-    g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
-    return FALSE;
+    g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
+    return NULL;
   }
 dec_src_failed:
   {
-    g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
     gst_object_unref (recv_rtp_sink);
-    return FALSE;
+    return NULL;
   }
 dec_link_failed:
   {
-    g_warning ("rtpbin: failed to link rtp decoder for session %d", sessid);
+    g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
     gst_object_unref (recv_rtp_sink);
-    return FALSE;
+    return NULL;
   }
 }
 
-/* Create a pad for receiving RTP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
-static GstPad *
-create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+static void
+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
+    guint sessid)
 {
-  guint sessid;
   GstElement *aux;
   GstPad *recv_rtp_src;
-  GstRtpBinSession *session;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
-    goto no_name;
-
-  GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
 
-  /* get or create session */
-  session = find_session_by_id (rtpbin, sessid);
-  if (!session) {
-    GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
-    /* create session now */
-    session = create_session (rtpbin, sessid);
-    if (session == NULL)
-      goto create_error;
-  }
-
-  /* check if pad was requested */
-  if (session->recv_rtp_sink_ghost != NULL)
-    return session->recv_rtp_sink_ghost;
-
-  /* setup the session sink pad */
-  if (!complete_session_sink (rtpbin, session))
-    goto session_sink_failed;
+  g_assert (!session->recv_rtp_src);
 
   session->recv_rtp_src =
       gst_element_get_static_pad (session->session, "recv_rtp_src");
@@ -3381,7 +3535,7 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
 
     GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
 
-    pname = g_strdup_printf ("sink_%d", sessid);
+    pname = g_strdup_printf ("sink_%u", sessid);
     auxsink = gst_element_get_static_pad (aux, pname);
     g_free (pname);
     if (auxsink == NULL)
@@ -3394,7 +3548,7 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
 
     /* this can be NULL when this AUX element is not to be linked to
      * an SSRC demuxer */
-    pname = g_strdup_printf ("src_%d", sessid);
+    pname = g_strdup_printf ("src_%u", sessid);
     recv_rtp_src = gst_element_get_static_pad (aux, pname);
     g_free (pname);
   } else {
@@ -3408,8 +3562,8 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
     sinkdpad = gst_element_get_static_pad (session->demux, "sink");
     GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
     gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
-    gst_object_unref (recv_rtp_src);
     gst_object_unref (sinkdpad);
+    gst_object_unref (recv_rtp_src);
 
     /* connect to the new-ssrc-pad signal of the SSRC demuxer */
     session->demux_newpad_sig = g_signal_connect (session->demux,
@@ -3417,6 +3571,71 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
     session->demux_padremoved_sig = g_signal_connect (session->demux,
         "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
   }
+
+  return;
+
+pad_failed:
+  {
+    g_warning ("rtpbin: failed to get session recv_rtp_src pad");
+    return;
+  }
+aux_sink_failed:
+  {
+    g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
+    return;
+  }
+aux_link_failed:
+  {
+    g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
+    return;
+  }
+}
+
+/* Create a pad for receiving RTP for the session in @name. Must be called with
+ * RTP_BIN_LOCK.
+ */
+static GstPad *
+create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+{
+  guint sessid;
+  GstRtpBinSession *session;
+  GstPad *recv_rtp_sink;
+
+  /* first get the session number */
+  if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
+    goto no_name;
+
+  GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
+
+  /* get or create session */
+  session = find_session_by_id (rtpbin, sessid);
+  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->recv_rtp_sink_ghost != NULL)
+    return session->recv_rtp_sink_ghost;
+
+  /* setup the session sink pad */
+  recv_rtp_sink = complete_session_sink (rtpbin, session, TRUE);
+  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);
+  gst_object_unref (recv_rtp_sink);
+  gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
+
+  complete_session_receiver (rtpbin, session, sessid);
+
   return session->recv_rtp_sink_ghost;
 
   /* ERRORS */
@@ -3435,21 +3654,6 @@ session_sink_failed:
     /* warning already done */
     return NULL;
   }
-pad_failed:
-  {
-    g_warning ("rtpbin: failed to get session recv_rtp_src pad");
-    return NULL;
-  }
-aux_sink_failed:
-  {
-    g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
-    return NULL;
-  }
-aux_link_failed:
-  {
-    g_warning ("rtpbin: failed to link AUX pad to session %d", sessid);
-    return NULL;
-  }
 }
 
 static void
@@ -3463,6 +3667,11 @@ 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;
@@ -3480,37 +3689,14 @@ remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
   }
 }
 
-/* Create a pad for receiving RTCP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
 static GstPad *
-create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
-    const gchar * name)
+complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
+    guint sessid, gboolean bundle_demuxer_needed)
 {
-  guint sessid;
   GstElement *decoder;
-  GstRtpBinSession *session;
-  GstPad *sinkdpad, *decsink;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
-    goto no_name;
-
-  GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
-
-  /* get or create the session */
-  session = find_session_by_id (rtpbin, sessid);
-  if (!session) {
-    GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
-    /* create session now */
-    session = create_session (rtpbin, sessid);
-    if (session == NULL)
-      goto create_error;
-  }
-
-  /* check if pad was requested */
-  if (session->recv_rtcp_sink_ghost != NULL)
-    return session->recv_rtcp_sink_ghost;
+  GstPad *sinkdpad;
+  GstPad *decsink = NULL;
+  GstPad *funnel_src;
 
   /* get recv_rtp pad and store */
   GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
@@ -3519,6 +3705,9 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
   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) {
@@ -3535,14 +3724,26 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
     if (decsrc == NULL)
       goto dec_src_failed;
 
-    ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
+    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);
+    }
     gst_object_unref (decsrc);
 
     if (ret != GST_PAD_LINK_OK)
       goto dec_link_failed;
   } else {
     GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
-    decsink = gst_object_ref (session->recv_rtcp_sink);
+    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");
+    }
   }
 
   /* get srcpad, link to SSRCDemux */
@@ -3556,26 +3757,12 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
   gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
   gst_object_unref (sinkdpad);
 
-  session->recv_rtcp_sink_ghost =
-      gst_ghost_pad_new_from_template (name, decsink, templ);
-  gst_object_unref (decsink);
-  gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
-      session->recv_rtcp_sink_ghost);
+  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 session->recv_rtcp_sink_ghost;
+  return decsink;
 
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("rtpbin: invalid name given");
-    return NULL;
-  }
-create_error:
-  {
-    /* create_session already warned */
-    return NULL;
-  }
 pad_failed:
   {
     g_warning ("rtpbin: failed to get session rtcp_sink pad");
@@ -3583,25 +3770,82 @@ pad_failed:
   }
 dec_sink_failed:
   {
-    g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
     return NULL;
   }
 dec_src_failed:
   {
-    g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
-    gst_object_unref (decsink);
-    return NULL;
+    g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
+    goto cleanup;
   }
 dec_link_failed:
   {
-    g_warning ("rtpbin: failed to link rtcp decoder for session %d", sessid);
-    gst_object_unref (decsink);
-    return NULL;
+    g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
+    goto cleanup;
   }
 src_pad_failed:
   {
     g_warning ("rtpbin: failed to get session sync_src pad");
-    gst_object_unref (decsink);
+  }
+
+cleanup:
+  gst_object_unref (decsink);
+  return NULL;
+}
+
+/* Create a pad for receiving RTCP for the session in @name. Must be called with
+ * RTP_BIN_LOCK.
+ */
+static GstPad *
+create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
+    const gchar * name)
+{
+  guint sessid;
+  GstRtpBinSession *session;
+  GstPad *decsink = NULL;
+
+  /* first get the session number */
+  if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
+    goto no_name;
+
+  GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
+
+  /* get or create the session */
+  session = find_session_by_id (rtpbin, sessid);
+  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->recv_rtcp_sink_ghost != NULL)
+    return session->recv_rtcp_sink_ghost;
+
+  decsink = complete_session_rtcp (rtpbin, session, sessid, TRUE);
+  if (!decsink)
+    goto create_error;
+
+  session->recv_rtcp_sink_ghost =
+      gst_ghost_pad_new_from_template (name, decsink, templ);
+  gst_object_unref (decsink);
+  gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
+      session->recv_rtcp_sink_ghost);
+
+  return session->recv_rtcp_sink_ghost;
+
+  /* ERRORS */
+no_name:
+  {
+    g_warning ("rtpbin: invalid name given");
+    return NULL;
+  }
+create_error:
+  {
+    /* create_session already warned */
     return NULL;
   }
 }
@@ -3651,7 +3895,7 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
     GstPadLinkReturn ret;
 
     GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
-    ename = g_strdup_printf ("rtp_src_%d", sessid);
+    ename = g_strdup_printf ("rtp_src_%u", sessid);
     encsrc = gst_element_get_static_pad (encoder, ename);
     g_free (ename);
 
@@ -3660,7 +3904,7 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
 
     send_rtp_src = encsrc;
 
-    ename = g_strdup_printf ("rtp_sink_%d", sessid);
+    ename = g_strdup_printf ("rtp_sink_%u", sessid);
     encsink = gst_element_get_static_pad (encoder, ename);
     g_free (ename);
     if (encsink == NULL)
@@ -3694,23 +3938,23 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
   /* ERRORS */
 no_srcpad:
   {
-    g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
     return FALSE;
   }
 enc_src_failed:
   {
-    g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
     return FALSE;
   }
 enc_sink_failed:
   {
-    g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
     gst_object_unref (send_rtp_src);
     return FALSE;
   }
 enc_link_failed:
   {
-    g_warning ("rtpbin: failed to link rtp encoder for session %d", sessid);
+    g_warning ("rtpbin: failed to link rtp encoder for session %u", sessid);
     gst_object_unref (send_rtp_src);
     return FALSE;
   }
@@ -3772,22 +4016,22 @@ create_error:
   }
 existing_session:
   {
-    g_warning ("rtpbin: session %d is already a sender", sessid);
+    g_warning ("rtpbin: session %u is already a sender", sessid);
     return FALSE;
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get session pad for session %u", sessid);
     return FALSE;
   }
 aux_link_failed:
   {
-    g_warning ("rtpbin: failed to link AUX for session %d", sessid);
+    g_warning ("rtpbin: failed to link AUX for session %u", sessid);
     return FALSE;
   }
 session_src_failed:
   {
-    g_warning ("rtpbin: failed to complete AUX for session %d", sessid);
+    g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
     return FALSE;
   }
 }
@@ -3847,7 +4091,7 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
     if (!setup_aux_sender (rtpbin, session, aux))
       goto aux_session_failed;
 
-    pname = g_strdup_printf ("sink_%d", sessid);
+    pname = g_strdup_printf ("sink_%u", sessid);
     send_rtp_sink = gst_element_get_static_pad (aux, pname);
     g_free (pname);
 
@@ -3887,27 +4131,27 @@ create_error:
   }
 existing_session:
   {
-    g_warning ("rtpbin: session %d is already in use", sessid);
+    g_warning ("rtpbin: session %u is already in use", sessid);
     return NULL;
   }
 aux_session_failed:
   {
-    g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
     return NULL;
   }
 aux_sink_failed:
   {
-    g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
     return NULL;
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get session pad for session %u", sessid);
     return NULL;
   }
 session_src_failed:
   {
-    g_warning ("rtpbin: failed to setup source pads for session %d", sessid);
+    g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
     return NULL;
   }
 }
@@ -3978,13 +4222,13 @@ create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
 
     GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
 
-    ename = g_strdup_printf ("rtcp_src_%d", sessid);
+    ename = g_strdup_printf ("rtcp_src_%u", sessid);
     encsrc = gst_element_get_static_pad (encoder, ename);
     g_free (ename);
     if (encsrc == NULL)
       goto enc_src_failed;
 
-    ename = g_strdup_printf ("rtcp_sink_%d", sessid);
+    ename = g_strdup_printf ("rtcp_sink_%u", sessid);
     encsink = gst_element_get_static_pad (encoder, ename);
     g_free (ename);
     if (encsink == NULL)
@@ -4021,23 +4265,23 @@ no_session:
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
     return NULL;
   }
 enc_src_failed:
   {
-    g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
     return NULL;
   }
 enc_sink_failed:
   {
-    g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
     gst_object_unref (encsrc);
     return NULL;
   }
 enc_link_failed:
   {
-    g_warning ("rtpbin: failed to link rtcp encoder for session %d", sessid);
+    g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
     gst_object_unref (encsrc);
     return NULL;
   }
diff --git a/gst/rtpmanager/gstrtpbin.h b/gst/rtpmanager/gstrtpbin.h
index fb13a4761..384b76df9 100644
--- a/gst/rtpmanager/gstrtpbin.h
+++ b/gst/rtpmanager/gstrtpbin.h
@@ -127,6 +127,8 @@ struct _GstRtpBinClass {
 
   void     (*on_new_sender_ssrc)      (GstRtpBin *rtpbin, guint session, guint32 ssrc);
   void     (*on_sender_ssrc_active)   (GstRtpBin *rtpbin, guint session, guint32 ssrc);
+
+  guint    (*on_bundled_ssrc)         (GstRtpBin *rtpbin, guint ssrc);
 };
 
 GType gst_rtp_bin_get_type (void);
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index aa40c705e..9c855ae5c 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -233,6 +233,7 @@ check_rtpmanager = \
 	elements/rtpaux \
 	elements/rtpbin \
 	elements/rtpbin_buffer_list \
+	elements/rtpbundle \
 	elements/rtpcollision \
 	elements/rtpjitterbuffer \
 	elements/rtpmux \
@@ -576,6 +577,9 @@ elements_rtpcollision_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_NET_LIBS) -lgstrtp-
 elements_rtpaux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_rtpaux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
 
+elements_rtpbundle_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
+elements_rtpbundle_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+
 # FIXME: configure should check for gdk-pixbuf not gtk
 # only need video.h header, not the lib
 elements_gdkpixbufsink_CFLAGS = \
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index ac1624ba0..de196b799 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -54,6 +54,7 @@ rtp-payloading
 rtpaux
 rtpbin
 rtpbin_buffer_list
+rtpbundle
 rtpcollision
 rtph261
 rtph263
diff --git a/tests/check/elements/rtpbundle.c b/tests/check/elements/rtpbundle.c
new file mode 100644
index 000000000..9b477e1a0
--- /dev/null
+++ b/tests/check/elements/rtpbundle.c
@@ -0,0 +1,390 @@
+/* GStreamer
+ *
+ * Copyright (C) 2016 Igalia S.L.
+ *   @author Philippe Normand <philn@igalia.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.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstconsistencychecker.h>
+#include <gst/check/gsttestclock.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+static GMainLoop *main_loop;
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+      GST_MESSAGE_SRC (message), message);
+
+  switch (message->type) {
+    case GST_MESSAGE_EOS:
+      g_main_loop_quit (main_loop);
+      break;
+    case GST_MESSAGE_WARNING:{
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_warning (message, &gerror, &debug);
+      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      break;
+    }
+    case GST_MESSAGE_ERROR:{
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_error (message, &gerror, &debug);
+      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      fail ("Error!");
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * element, GstPad * new_pad,
+    gpointer data)
+{
+  GstElement *pipeline = GST_ELEMENT (data);
+  gchar *pad_name = gst_pad_get_name (new_pad);
+
+  if (g_str_has_prefix (pad_name, "recv_rtp_src_")) {
+    GstCaps *caps = gst_pad_get_current_caps (new_pad);
+    GstStructure *s = gst_caps_get_structure (caps, 0);
+    const gchar *media_type = gst_structure_get_string (s, "media");
+    gchar *depayloader_name = g_strdup_printf ("%s_rtpdepayloader", media_type);
+    GstElement *rtpdepayloader =
+        gst_bin_get_by_name (GST_BIN (pipeline), depayloader_name);
+    GstPad *sinkpad;
+
+    g_free (depayloader_name);
+    fail_unless (rtpdepayloader != NULL, NULL);
+
+    sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+    gst_pad_link (new_pad, sinkpad);
+    gst_object_unref (sinkpad);
+    gst_object_unref (rtpdepayloader);
+
+    gst_caps_unref (caps);
+  }
+  g_free (pad_name);
+}
+
+static guint
+on_bundled_ssrc (GstElement * rtpbin, guint ssrc, gpointer user_data)
+{
+  static gboolean create_session = FALSE;
+  guint session_id = 0;
+
+  if (create_session) {
+    session_id = 1;
+  } else {
+    create_session = TRUE;
+    /* use existing session 0, a new session will be created for the next discovered bundled SSRC */
+  }
+  return session_id;
+}
+
+static GstCaps *
+on_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
+    gpointer user_data)
+{
+  GstCaps *caps = NULL;
+  if (pt == 96) {
+    caps =
+        gst_caps_from_string
+        ("application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000");
+  } else if (pt == 100) {
+    caps =
+        gst_caps_from_string
+        ("application/x-rtp,media=(string)video,encoding-name=(string)RAW,clock-rate=(int)90000,sampling=(string)\"YCbCr-4:2:0\",depth=(string)8,width=(string)320,height=(string)240");
+  }
+  return caps;
+}
+
+
+static GstElement *
+create_pipeline (gboolean send)
+{
+  GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder,
+      *audio_rtppayloader, *sendrtp_udpsink, *recv_rtp_udpsrc,
+      *send_rtcp_udpsink, *recv_rtcp_udpsrc, *sendrtcp_funnel, *sendrtp_funnel;
+  GstElement *audio_rtpdepayloader, *audio_decoder, *audio_sink;
+  GstElement *videosrc, *video_rtppayloader, *video_rtpdepayloader, *video_sink;
+  gboolean res;
+  GstPad *funnel_pad, *rtp_src_pad;
+  GstCaps *rtpcaps;
+  gint rtp_udp_port = 5001;
+  gint rtcp_udp_port = 5002;
+
+  pipeline = gst_pipeline_new (send ? "pipeline_send" : "pipeline_receive");
+
+  rtpbin =
+      gst_element_factory_make ("rtpbin",
+      send ? "rtpbin_send" : "rtpbin_receive");
+  g_object_set (rtpbin, "latency", 200, NULL);
+
+  if (!send) {
+    g_signal_connect (rtpbin, "on-bundled-ssrc",
+        G_CALLBACK (on_bundled_ssrc), NULL);
+    g_signal_connect (rtpbin, "request-pt-map",
+        G_CALLBACK (on_request_pt_map), NULL);
+  }
+
+  g_signal_connect (rtpbin, "pad-added",
+      G_CALLBACK (on_rtpbinreceive_pad_added), pipeline);
+
+  gst_bin_add (GST_BIN (pipeline), rtpbin);
+
+  if (send) {
+    audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
+    audio_encoder = gst_element_factory_make ("alawenc", NULL);
+    audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL);
+    g_object_set (audio_rtppayloader, "pt", 96, NULL);
+    g_object_set (audio_rtppayloader, "seqnum-offset", 1, NULL);
+
+    videosrc = gst_element_factory_make ("videotestsrc", NULL);
+    video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL);
+    g_object_set (video_rtppayloader, "pt", 100, "seqnum-offset", 1, NULL);
+
+    g_object_set (audiosrc, "num-buffers", 5, NULL);
+    g_object_set (videosrc, "num-buffers", 5, NULL);
+
+    /* muxed rtcp */
+    sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel");
+    send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+    g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL);
+    g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL);
+    g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL);
+    g_object_set (send_rtcp_udpsink, "async", FALSE, NULL);
+
+    /* outgoing bundled stream */
+    sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel");
+    sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL);
+    g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+    g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL);
+
+    gst_bin_add_many (GST_BIN (pipeline), audiosrc, audio_encoder,
+        audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink,
+        sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL);
+
+    res = gst_element_link (audiosrc, audio_encoder);
+    fail_unless (res == TRUE, NULL);
+    res = gst_element_link (audio_encoder, audio_rtppayloader);
+    fail_unless (res == TRUE, NULL);
+    res =
+        gst_element_link_pads_full (audio_rtppayloader, "src", rtpbin,
+        "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+
+    res = gst_element_link (videosrc, video_rtppayloader);
+    fail_unless (res == TRUE, NULL);
+    res =
+        gst_element_link_pads_full (video_rtppayloader, "src", rtpbin,
+        "send_rtp_sink_1", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+
+    res =
+        gst_element_link_pads_full (sendrtp_funnel, "src", sendrtp_udpsink,
+        "sink", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+
+    funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u");
+    rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
+    res = gst_pad_link (rtp_src_pad, funnel_pad);
+    gst_object_unref (funnel_pad);
+    gst_object_unref (rtp_src_pad);
+
+    funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u");
+    rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_1");
+    res = gst_pad_link (rtp_src_pad, funnel_pad);
+    gst_object_unref (funnel_pad);
+    gst_object_unref (rtp_src_pad);
+
+    res =
+        gst_element_link_pads_full (sendrtcp_funnel, "src", send_rtcp_udpsink,
+        "sink", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+
+    funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u");
+    rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
+    res =
+        gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING);
+    gst_object_unref (funnel_pad);
+    gst_object_unref (rtp_src_pad);
+
+    funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u");
+    rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1");
+    res =
+        gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING);
+    gst_object_unref (funnel_pad);
+    gst_object_unref (rtp_src_pad);
+
+  } else {
+    recv_rtp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+    g_object_set (recv_rtp_udpsrc, "port", rtp_udp_port, NULL);
+    rtpcaps = gst_caps_from_string ("application/x-rtp");
+    g_object_set (recv_rtp_udpsrc, "caps", rtpcaps, NULL);
+    gst_caps_unref (rtpcaps);
+
+    recv_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+    g_object_set (recv_rtcp_udpsrc, "port", rtcp_udp_port, NULL);
+
+    audio_rtpdepayloader =
+        gst_element_factory_make ("rtppcmadepay", "audio_rtpdepayloader");
+    audio_decoder = gst_element_factory_make ("alawdec", NULL);
+    audio_sink = gst_element_factory_make ("fakesink", NULL);
+    g_object_set (audio_sink, "sync", TRUE, NULL);
+
+    video_rtpdepayloader =
+        gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader");
+    video_sink = gst_element_factory_make ("fakesink", NULL);
+    g_object_set (video_sink, "sync", TRUE, NULL);
+
+    gst_bin_add_many (GST_BIN (pipeline), recv_rtp_udpsrc, recv_rtcp_udpsrc,
+        audio_rtpdepayloader, audio_decoder, audio_sink, video_rtpdepayloader,
+        video_sink, NULL);
+
+    res =
+        gst_element_link_pads_full (audio_rtpdepayloader, "src", audio_decoder,
+        "sink", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+    res = gst_element_link (audio_decoder, audio_sink);
+    fail_unless (res == TRUE, NULL);
+
+    res =
+        gst_element_link_pads_full (video_rtpdepayloader, "src", video_sink,
+        "sink", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+
+    /* request a single receiving RTP session. */
+    res =
+        gst_element_link_pads_full (recv_rtcp_udpsrc, "src", rtpbin,
+        "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+    res =
+        gst_element_link_pads_full (recv_rtp_udpsrc, "src", rtpbin,
+        "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+    fail_unless (res == TRUE, NULL);
+  }
+
+  return pipeline;
+}
+
+GST_START_TEST (test_simple_rtpbin_bundle)
+{
+  GstElement *send_pipeline, *recv_pipeline;
+  GstBus *send_bus, *recv_bus;
+  GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
+  GstElement *rtpbin_receive;
+  GObject *rtp_session;
+
+  main_loop = g_main_loop_new (NULL, FALSE);
+
+  send_pipeline = create_pipeline (TRUE);
+  recv_pipeline = create_pipeline (FALSE);
+
+  send_bus = gst_element_get_bus (send_pipeline);
+  gst_bus_add_signal_watch_full (send_bus, G_PRIORITY_HIGH);
+
+  g_signal_connect (send_bus, "message::error", (GCallback) message_received,
+      send_pipeline);
+  g_signal_connect (send_bus, "message::warning", (GCallback) message_received,
+      send_pipeline);
+  g_signal_connect (send_bus, "message::eos", (GCallback) message_received,
+      send_pipeline);
+
+  recv_bus = gst_element_get_bus (recv_pipeline);
+  gst_bus_add_signal_watch_full (recv_bus, G_PRIORITY_HIGH);
+
+  g_signal_connect (recv_bus, "message::error", (GCallback) message_received,
+      recv_pipeline);
+  g_signal_connect (recv_bus, "message::warning", (GCallback) message_received,
+      recv_pipeline);
+  g_signal_connect (recv_bus, "message::eos", (GCallback) message_received,
+      recv_pipeline);
+
+  state_res = gst_element_set_state (recv_pipeline, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  state_res = gst_element_set_state (send_pipeline, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  GST_INFO ("enter mainloop");
+  g_main_loop_run (main_loop);
+  GST_INFO ("exit mainloop");
+
+  rtpbin_receive =
+      gst_bin_get_by_name (GST_BIN (recv_pipeline), "rtpbin_receive");
+  fail_if (rtpbin_receive == NULL, NULL);
+
+  /* Check that 2 RTP sessions where created while only one was explicitely requested. */
+  g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 0,
+      &rtp_session);
+  fail_if (rtp_session == NULL, NULL);
+  g_object_unref (rtp_session);
+  g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 1,
+      &rtp_session);
+  fail_if (rtp_session == NULL, NULL);
+  g_object_unref (rtp_session);
+
+  gst_object_unref (rtpbin_receive);
+
+  state_res = gst_element_set_state (send_pipeline, GST_STATE_NULL);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  state_res = gst_element_set_state (recv_pipeline, GST_STATE_NULL);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  /* cleanup */
+  g_main_loop_unref (main_loop);
+
+  gst_bus_remove_signal_watch (send_bus);
+  gst_object_unref (send_bus);
+  gst_object_unref (send_pipeline);
+
+  gst_bus_remove_signal_watch (recv_bus);
+  gst_object_unref (recv_bus);
+  gst_object_unref (recv_pipeline);
+
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpbundle_suite (void)
+{
+  Suite *s = suite_create ("rtpbundle");
+  TCase *tc_chain = tcase_create ("general");
+
+  tcase_set_timeout (tc_chain, 10000);
+
+  suite_add_tcase (s, tc_chain);
+
+  tcase_add_test (tc_chain, test_simple_rtpbin_bundle);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtpbundle);
diff --git a/tests/check/meson.build b/tests/check/meson.build
index 4559e66c8..0f413777b 100644
--- a/tests/check/meson.build
+++ b/tests/check/meson.build
@@ -72,6 +72,7 @@ good_tests = [
   [ 'elements/rtpaux' ],
   [ 'elements/rtpbin' ],
   [ 'elements/rtpbin_buffer_list' ],
+  [ 'elements/rtpbundle' ],
   [ 'elements/rtpcollision' ],
   [ 'elements/rtpjitterbuffer' ],
   [ 'elements/rtpmux' ],
diff --git a/tests/examples/rtp/.gitignore b/tests/examples/rtp/.gitignore
index 02f551f12..8c3c6d3ff 100644
--- a/tests/examples/rtp/.gitignore
+++ b/tests/examples/rtp/.gitignore
@@ -2,3 +2,5 @@ client-PCMA
 server-alsasrc-PCMA
 client-rtpaux
 server-rtpaux
+client-rtpbundle
+server-rtpbundle
diff --git a/tests/examples/rtp/Makefile.am b/tests/examples/rtp/Makefile.am
index 40ab2c55e..4d7f7c1f8 100644
--- a/tests/examples/rtp/Makefile.am
+++ b/tests/examples/rtp/Makefile.am
@@ -1,5 +1,5 @@
 noinst_PROGRAMS = server-alsasrc-PCMA client-PCMA \
-	client-rtpaux server-rtpaux
+	client-rtpaux server-rtpaux client-rtpbundle server-rtpbundle
 
 # FIXME 0.11: ignore GValueArray warnings for now until this is sorted
 ERROR_CFLAGS=
@@ -12,6 +12,14 @@ client_rtpaux_SOURCES = client-rtpaux.c
 client_rtpaux_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
 client_rtpaux_LDADD = $(GST_LIBS)
 
+server_rtpbundle_SOURCES = server-rtpbundle.c
+server_rtpbundle_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+server_rtpbundle_LDADD = $(GST_LIBS)
+
+client_rtpbundle_SOURCES = client-rtpbundle.c
+client_rtpbundle_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+client_rtpbundle_LDADD = $(GST_LIBS)
+
 server_alsasrc_PCMA_SOURCES = server-alsasrc-PCMA.c
 server_alsasrc_PCMA_CFLAGS = $(GST_CFLAGS)
 server_alsasrc_PCMA_LDADD = $(GST_LIBS) $(LIBM)
diff --git a/tests/examples/rtp/client-rtpbundle.c b/tests/examples/rtp/client-rtpbundle.c
new file mode 100644
index 000000000..cd3a737f8
--- /dev/null
+++ b/tests/examples/rtp/client-rtpbundle.c
@@ -0,0 +1,266 @@
+/* GStreamer
+ * Copyright (C) 2016 Igalia S.L
+ *   @author Philippe Normand <philn@igalia.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.
+ */
+
+#include <gst/gst.h>
+
+/*
+ * RTP bundle receiver
+ *
+ * In this example we initially create one RTP session but the incoming RTP
+ * and RTCP streams actually bundle 2 different media type, one audio stream
+ * and one video stream. We are notified of the discovery of the streams by
+ * the on-bundled-ssrc rtpbin signal. In the handler we decide to assign the
+ * first SSRC to the (existing) audio session and the second SSRC to a new
+ * session (id: 1).
+ *
+ *             .-------.      .----------.        .-----------.    .-------.    .-------------.
+ *  RTP        |udpsrc |      | rtpbin   |        | pcmadepay |    |alawdec|    |autoaudiosink|
+ *  port=5001  |      src->recv_rtp_0 recv_rtp_0->sink       src->sink    src->sink           |
+ *             '-------'      |          |        '-----------'    '-------'    '-------------'
+ *                            |          |
+ *                            |          |     .-------.
+ *                            |          |     |udpsink|  RTCP
+ *                            |  send_rtcp_0->sink     | port=5003
+ *             .-------.      |          |     '-------' sync=false
+ *  RTCP       |udpsrc |      |          |               async=false
+ *  port=5002  |     src->recv_rtcp_0    |
+ *             '-------'      |          |
+ *                            |          |
+ *                            |          |        .---------.    .-------------.
+ *                            |          |        |vrawdepay|    |autovideosink|
+ *                            |       recv_rtp_1->sink     src->sink           |
+ *                            |          |        '---------'    '-------------'
+ *                            |          |
+ *                            |          |     .-------.
+ *                            |          |     |udpsink|  RTCP
+ *                            |  send_rtcp_1->sink     | port=5004
+ *                            |          |     '-------' sync=false
+ *                            |          |               async=false
+ *                            |          |
+ *                            '----------'
+ *
+ */
+
+static gboolean
+plug_video_rtcp_sender (gpointer user_data)
+{
+  gint send_video_rtcp_port = 5004;
+  GstElement *rtpbin = GST_ELEMENT_CAST (user_data);
+  GstElement *send_video_rtcp_udpsink;
+  GstElement *pipeline =
+      GST_ELEMENT_CAST (gst_object_get_parent (GST_OBJECT (rtpbin)));
+
+  send_video_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+  g_object_set (send_video_rtcp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (send_video_rtcp_udpsink, "port", send_video_rtcp_port, NULL);
+  g_object_set (send_video_rtcp_udpsink, "sync", FALSE, NULL);
+  g_object_set (send_video_rtcp_udpsink, "async", FALSE, NULL);
+  gst_bin_add (GST_BIN (pipeline), send_video_rtcp_udpsink);
+  gst_element_link_pads (rtpbin, "send_rtcp_src_1", send_video_rtcp_udpsink,
+      "sink");
+  gst_element_sync_state_with_parent (send_video_rtcp_udpsink);
+
+  gst_object_unref (pipeline);
+  gst_object_unref (rtpbin);
+  return G_SOURCE_REMOVE;
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * rtpbin, GstPad * new_pad,
+    gpointer data)
+{
+  GstElement *pipeline = GST_ELEMENT (data);
+  gchar *pad_name = gst_pad_get_name (new_pad);
+
+  if (g_str_has_prefix (pad_name, "recv_rtp_src_")) {
+    GstCaps *caps = gst_pad_get_current_caps (new_pad);
+    GstStructure *s = gst_caps_get_structure (caps, 0);
+    const gchar *media_type = gst_structure_get_string (s, "media");
+    gchar *depayloader_name = g_strdup_printf ("%s_rtpdepayloader", media_type);
+    GstElement *rtpdepayloader =
+        gst_bin_get_by_name (GST_BIN (pipeline), depayloader_name);
+    GstPad *sinkpad;
+
+    g_free (depayloader_name);
+
+    sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+    gst_pad_link (new_pad, sinkpad);
+    gst_object_unref (sinkpad);
+    gst_object_unref (rtpdepayloader);
+
+    gst_caps_unref (caps);
+
+    if (g_str_has_prefix (pad_name, "recv_rtp_src_1")) {
+      g_timeout_add (0, plug_video_rtcp_sender, gst_object_ref (rtpbin));
+    }
+  }
+  g_free (pad_name);
+}
+
+static guint
+on_bundled_ssrc (GstElement * rtpbin, guint ssrc, gpointer user_data)
+{
+  static gboolean create_session = FALSE;
+  guint session_id = 0;
+
+  if (create_session) {
+    session_id = 1;
+  } else {
+    create_session = TRUE;
+    /* use existing session 0, a new session will be created for the next discovered bundled SSRC */
+  }
+  return session_id;
+}
+
+static GstCaps *
+on_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
+    gpointer user_data)
+{
+  GstCaps *caps = NULL;
+  if (pt == 96) {
+    caps =
+        gst_caps_from_string
+        ("application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000");
+  } else if (pt == 100) {
+    caps =
+        gst_caps_from_string
+        ("application/x-rtp,media=(string)video,encoding-name=(string)RAW,clock-rate=(int)90000,sampling=(string)\"YCbCr-4:2:0\",depth=(string)8,width=(string)320,height=(string)240");
+  }
+  return caps;
+}
+
+static GstElement *
+create_pipeline (void)
+{
+  GstElement *pipeline, *rtpbin, *recv_rtp_udpsrc, *recv_rtcp_udpsrc,
+      *audio_rtpdepayloader, *audio_decoder, *audio_sink, *video_rtpdepayloader,
+      *video_sink, *send_audio_rtcp_udpsink;
+  GstCaps *rtpcaps;
+  gint rtp_udp_port = 5001;
+  gint rtcp_udp_port = 5002;
+  gint send_audio_rtcp_port = 5003;
+
+  pipeline = gst_pipeline_new (NULL);
+
+  rtpbin = gst_element_factory_make ("rtpbin", NULL);
+  g_object_set (rtpbin, "latency", 200, NULL);
+
+  g_signal_connect (rtpbin, "on-bundled-ssrc",
+      G_CALLBACK (on_bundled_ssrc), NULL);
+  g_signal_connect (rtpbin, "request-pt-map",
+      G_CALLBACK (on_request_pt_map), NULL);
+
+  g_signal_connect (rtpbin, "pad-added",
+      G_CALLBACK (on_rtpbinreceive_pad_added), pipeline);
+
+  gst_bin_add (GST_BIN (pipeline), rtpbin);
+
+  recv_rtp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+  g_object_set (recv_rtp_udpsrc, "port", rtp_udp_port, NULL);
+  rtpcaps = gst_caps_from_string ("application/x-rtp");
+  g_object_set (recv_rtp_udpsrc, "caps", rtpcaps, NULL);
+  gst_caps_unref (rtpcaps);
+
+  recv_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+  g_object_set (recv_rtcp_udpsrc, "port", rtcp_udp_port, NULL);
+
+  audio_rtpdepayloader =
+      gst_element_factory_make ("rtppcmadepay", "audio_rtpdepayloader");
+  audio_decoder = gst_element_factory_make ("alawdec", NULL);
+  audio_sink = gst_element_factory_make ("autoaudiosink", NULL);
+
+  video_rtpdepayloader =
+      gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader");
+  video_sink = gst_element_factory_make ("autovideosink", NULL);
+
+  gst_bin_add_many (GST_BIN (pipeline), recv_rtp_udpsrc, recv_rtcp_udpsrc,
+      audio_rtpdepayloader, audio_decoder, audio_sink, video_rtpdepayloader,
+      video_sink, NULL);
+
+  gst_element_link_pads (audio_rtpdepayloader, "src", audio_decoder, "sink");
+  gst_element_link (audio_decoder, audio_sink);
+
+  gst_element_link_pads (video_rtpdepayloader, "src", video_sink, "sink");
+
+  /* request a single receiving RTP session. */
+  gst_element_link_pads (recv_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0");
+  gst_element_link_pads (recv_rtp_udpsrc, "src", rtpbin, "recv_rtp_sink_0");
+
+  send_audio_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+  g_object_set (send_audio_rtcp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (send_audio_rtcp_udpsink, "port", send_audio_rtcp_port, NULL);
+  g_object_set (send_audio_rtcp_udpsink, "sync", FALSE, NULL);
+  g_object_set (send_audio_rtcp_udpsink, "async", FALSE, NULL);
+  gst_bin_add (GST_BIN (pipeline), send_audio_rtcp_udpsink);
+  gst_element_link_pads (rtpbin, "send_rtcp_src_0", send_audio_rtcp_udpsink,
+      "sink");
+
+  return pipeline;
+}
+
+/*
+ * Used to generate informative messages during pipeline startup
+ */
+static void
+cb_state (GstBus * bus, GstMessage * message, gpointer data)
+{
+  GstObject *pipe = GST_OBJECT (data);
+  GstState old, new, pending;
+  gst_message_parse_state_changed (message, &old, &new, &pending);
+  if (message->src == pipe) {
+    g_print ("Pipeline %s changed state from %s to %s\n",
+        GST_OBJECT_NAME (message->src),
+        gst_element_state_get_name (old), gst_element_state_get_name (new));
+    if (old == GST_STATE_PAUSED && new == GST_STATE_PLAYING)
+      GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (pipe), GST_DEBUG_GRAPH_SHOW_ALL,
+          GST_OBJECT_NAME (message->src));
+  }
+}
+
+int
+main (int argc, char **argv)
+{
+  GstElement *pipe;
+  GstBus *bus;
+  GMainLoop *loop;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  pipe = create_pipeline ();
+  bus = gst_element_get_bus (pipe);
+  g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
+  gst_bus_add_signal_watch (bus);
+  gst_object_unref (bus);
+
+  g_print ("starting server pipeline\n");
+  gst_element_set_state (pipe, GST_STATE_PLAYING);
+
+  g_main_loop_run (loop);
+
+  g_print ("stopping server pipeline\n");
+  gst_element_set_state (pipe, GST_STATE_NULL);
+
+  gst_object_unref (pipe);
+  g_main_loop_unref (loop);
+
+  return 0;
+}
diff --git a/tests/examples/rtp/server-rtpbundle.c b/tests/examples/rtp/server-rtpbundle.c
new file mode 100644
index 000000000..1f6d01bef
--- /dev/null
+++ b/tests/examples/rtp/server-rtpbundle.c
@@ -0,0 +1,179 @@
+/* GStreamer
+ * Copyright (C) 2016 Igalia S.L
+ *   @author Philippe Normand <philn@igalia.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.
+ */
+#include <gst/gst.h>
+
+/*
+ * An bundling RTP server
+ *  creates two sessions and streams audio on one, video on the other, with RTCP
+ *  on both sessions. The destination is 127.0.0.1.
+ *
+ *  The RTP streams are bundled to a single outgoing connection. Same for the RTCP streams.
+ *
+ *  .-------.    .-------.    .-------.      .------------.         .------.
+ *  |audiots|    |alawenc|    |pcmapay|      | rtpbin     |         |funnel|
+ *  |      src->sink    src->sink    src->send_rtp_0 send_rtp_0--->sink_0  |    .-------.
+ *  '-------'    '-------'    '-------'      |            |         |      |    |udpsink|
+ *                                           |            |         |     src->sink     |
+ *  .-------.               .---------.      |            |         |      |    '-------'
+ *  |videots|               | vrawpay |      |            |         |      |
+ *  |      src------------>sink      src->send_rtp_1 send_rtp_1--->sink_1  |
+ *  '-------'               '---------'      |            |         '------'
+ *                                           |            |
+ *                               .------.    |            |
+ *                               |udpsrc|    |            |         .------.
+ *                               |     src->recv_rtcp_0   |         |funnel|
+ *                               '------'    |       send_rtcp_0-->sink_0  |   .-------.
+ *                                           |            |         |      |   |udpsink|
+ *                               .------.    |            |         |    src->sink     |
+ *                               |udpsrc|    |            |         |      |   '-------'
+ *                               |     src->recv_rtcp_1   |         |      |
+ *                               '------'    |       send_rtcp_1-->sink_1  |
+ *                                           '------------'         '------'
+ *
+ */
+
+static GstElement *
+create_pipeline (void)
+{
+  GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder,
+      *audio_rtppayloader, *sendrtp_udpsink,
+      *send_rtcp_udpsink, *sendrtcp_funnel, *sendrtp_funnel;
+  GstElement *videosrc, *video_rtppayloader, *time_overlay;
+  gint rtp_udp_port = 5001;
+  gint rtcp_udp_port = 5002;
+  gint recv_audio_rtcp_port = 5003;
+  gint recv_video_rtcp_port = 5004;
+  GstElement *audio_rtcp_udpsrc, *video_rtcp_udpsrc;
+
+  pipeline = gst_pipeline_new (NULL);
+
+  rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+  audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
+  g_object_set (audiosrc, "is-live", TRUE, NULL);
+  audio_encoder = gst_element_factory_make ("alawenc", NULL);
+  audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL);
+  g_object_set (audio_rtppayloader, "pt", 96, NULL);
+
+  videosrc = gst_element_factory_make ("videotestsrc", NULL);
+  g_object_set (videosrc, "is-live", TRUE, NULL);
+  time_overlay = gst_element_factory_make ("timeoverlay", NULL);
+  video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL);
+  g_object_set (video_rtppayloader, "pt", 100, NULL);
+
+  /* muxed rtcp */
+  sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel");
+  send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+  g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL);
+  g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL);
+  g_object_set (send_rtcp_udpsink, "async", FALSE, NULL);
+
+  /* outgoing bundled stream */
+  sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel");
+  sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL);
+  g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL);
+  g_object_set (sendrtp_udpsink, "sync", FALSE, NULL);
+  g_object_set (sendrtp_udpsink, "async", FALSE, NULL);
+
+  gst_bin_add_many (GST_BIN (pipeline), rtpbin, audiosrc, audio_encoder,
+      audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink,
+      sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL);
+
+  if (time_overlay)
+    gst_bin_add (GST_BIN (pipeline), time_overlay);
+
+  gst_element_link_many (audiosrc, audio_encoder, audio_rtppayloader, NULL);
+  gst_element_link_pads (audio_rtppayloader, "src", rtpbin, "send_rtp_sink_0");
+
+  if (time_overlay) {
+    gst_element_link_many (videosrc, time_overlay, video_rtppayloader, NULL);
+  } else {
+    gst_element_link (videosrc, video_rtppayloader);
+  }
+
+  gst_element_link_pads (video_rtppayloader, "src", rtpbin, "send_rtp_sink_1");
+
+  gst_element_link_pads (sendrtp_funnel, "src", sendrtp_udpsink, "sink");
+  gst_element_link_pads (rtpbin, "send_rtp_src_0", sendrtp_funnel, "sink_%u");
+  gst_element_link_pads (rtpbin, "send_rtp_src_1", sendrtp_funnel, "sink_%u");
+  gst_element_link_pads (sendrtcp_funnel, "src", send_rtcp_udpsink, "sink");
+  gst_element_link_pads (rtpbin, "send_rtcp_src_0", sendrtcp_funnel, "sink_%u");
+  gst_element_link_pads (rtpbin, "send_rtcp_src_1", sendrtcp_funnel, "sink_%u");
+
+  audio_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+  g_object_set (audio_rtcp_udpsrc, "port", recv_audio_rtcp_port, NULL);
+  video_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+  g_object_set (video_rtcp_udpsrc, "port", recv_video_rtcp_port, NULL);
+  gst_bin_add_many (GST_BIN (pipeline), audio_rtcp_udpsrc, video_rtcp_udpsrc,
+      NULL);
+  gst_element_link_pads (audio_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0");
+  gst_element_link_pads (video_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_1");
+
+  return pipeline;
+}
+
+/*
+ * Used to generate informative messages during pipeline startup
+ */
+static void
+cb_state (GstBus * bus, GstMessage * message, gpointer data)
+{
+  GstObject *pipe = GST_OBJECT (data);
+  GstState old, new, pending;
+  gst_message_parse_state_changed (message, &old, &new, &pending);
+  if (message->src == pipe) {
+    g_print ("Pipeline %s changed state from %s to %s\n",
+        GST_OBJECT_NAME (message->src),
+        gst_element_state_get_name (old), gst_element_state_get_name (new));
+  }
+}
+
+int
+main (int argc, char **argv)
+{
+  GstElement *pipe;
+  GstBus *bus;
+  GMainLoop *loop;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  pipe = create_pipeline ();
+  bus = gst_element_get_bus (pipe);
+  g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
+  gst_bus_add_signal_watch (bus);
+  gst_object_unref (bus);
+
+  g_print ("starting server pipeline\n");
+  gst_element_set_state (pipe, GST_STATE_PLAYING);
+
+  g_main_loop_run (loop);
+
+  g_print ("stopping server pipeline\n");
+  gst_element_set_state (pipe, GST_STATE_NULL);
+
+  gst_object_unref (pipe);
+  g_main_loop_unref (loop);
+
+  return 0;
+}