rtsp-media-factory: expose API to disable RTCP

This is supported by the RFC, and can be useful on systems where
allocating two consecutive ports is problematic, and RTCP is not
necessary.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>

7 files changed, 232 insertions, 45 deletions

diff --git a/examples/test-launch.c b/examples/test-launch.c
index 1675591..b21df8e 100644
--- a/examples/test-launch.c
+++ b/examples/test-launch.c
@@ -22,12 +22,16 @@
 #include <gst/rtsp-server/rtsp-server.h>
 
 #define DEFAULT_RTSP_PORT "8554"
+#define DEFAULT_DISABLE_RTCP FALSE
 
 static char *port = (char *) DEFAULT_RTSP_PORT;
+static gboolean disable_rtcp = DEFAULT_DISABLE_RTCP;
 
 static GOptionEntry entries[] = {
   {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
       "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
+  {"disable-rtcp", '\0', 0, G_OPTION_ARG_NONE, &disable_rtcp,
+      "Whether RTCP should be disabled (default false)", NULL},
   {NULL}
 };
 
@@ -70,6 +74,7 @@ main (int argc, char *argv[])
   factory = gst_rtsp_media_factory_new ();
   gst_rtsp_media_factory_set_launch (factory, argv[1]);
   gst_rtsp_media_factory_set_shared (factory, TRUE);
+  gst_rtsp_media_factory_set_enable_rtcp (factory, !disable_rtcp);
 
   /* attach the test factory to the /test url */
   gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
diff --git a/gst/rtsp-server/rtsp-media-factory.c b/gst/rtsp-server/rtsp-media-factory.c
index a2d091d..5dd9dc0 100644
--- a/gst/rtsp-server/rtsp-media-factory.c
+++ b/gst/rtsp-server/rtsp-media-factory.c
@@ -41,6 +41,7 @@
 #include "config.h"
 #endif
 
+#include "rtsp-server-internal.h"
 #include "rtsp-media-factory.h"
 
 #define GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)       (&(GST_RTSP_MEDIA_FACTORY_CAST(f)->priv->lock))
@@ -65,6 +66,7 @@ struct _GstRTSPMediaFactoryPrivate
   gchar *multicast_iface;
   guint max_mcast_ttl;
   gboolean bind_mcast_address;
+  gboolean enable_rtcp;
 
   GstClockTime rtx_time;
   guint latency;
@@ -95,6 +97,7 @@ struct _GstRTSPMediaFactoryPrivate
 #define DEFAULT_STOP_ON_DISCONNECT TRUE
 #define DEFAULT_DO_RETRANSMISSION FALSE
 #define DEFAULT_DSCP_QOS        (-1)
+#define DEFAULT_ENABLE_RTCP     TRUE
 
 enum
 {
@@ -113,6 +116,7 @@ enum
   PROP_MAX_MCAST_TTL,
   PROP_BIND_MCAST_ADDRESS,
   PROP_DSCP_QOS,
+  PROP_ENABLE_RTCP,
   PROP_LAST
 };
 
@@ -247,6 +251,18 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
           DEFAULT_BIND_MCAST_ADDRESS,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstRTSPMediaFactory:enable-rtcp:
+   *
+   * Whether the created media should send and receive RTCP
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class, PROP_ENABLE_RTCP,
+      g_param_spec_boolean ("enable-rtcp", "Enable RTCP",
+          "Whether the created media should send and receive RTCP",
+          DEFAULT_ENABLE_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   g_object_class_install_property (gobject_class, PROP_DSCP_QOS,
       g_param_spec_int ("dscp-qos", "DSCP QoS",
           "The IP DSCP field to use", -1, 63,
@@ -295,6 +311,7 @@ gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
   priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
   priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
   priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+  priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
   priv->dscp_qos = DEFAULT_DSCP_QOS;
 
   g_mutex_init (&priv->lock);
@@ -381,6 +398,10 @@ gst_rtsp_media_factory_get_property (GObject * object, guint propid,
     case PROP_DSCP_QOS:
       g_value_set_int (value, gst_rtsp_media_factory_get_dscp_qos (factory));
       break;
+    case PROP_ENABLE_RTCP:
+      g_value_set_boolean (value,
+          gst_rtsp_media_factory_is_enable_rtcp (factory));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -442,6 +463,10 @@ gst_rtsp_media_factory_set_property (GObject * object, guint propid,
     case PROP_DSCP_QOS:
       gst_rtsp_media_factory_set_dscp_qos (factory, g_value_get_int (value));
       break;
+    case PROP_ENABLE_RTCP:
+      gst_rtsp_media_factory_set_enable_rtcp (factory,
+          g_value_get_boolean (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -1686,6 +1711,57 @@ gst_rtsp_media_factory_is_bind_mcast_address (GstRTSPMediaFactory * factory)
   return result;
 }
 
+/**
+ * gst_rtsp_media_factory_set_enable_rtcp:
+ * @factory: a #GstRTSPMediaFactory
+ * @enable: the new value
+ *
+ * Decide whether the created media should send and receive RTCP
+ *
+ * Since: 1.20
+ */
+void
+gst_rtsp_media_factory_set_enable_rtcp (GstRTSPMediaFactory * factory,
+    gboolean enable)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->enable_rtcp = enable;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_enable_rtcp:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Check if created media will send and receive RTCP
+ *
+ * Returns: %TRUE if created media will send and receive RTCP
+ *
+ * Since: 1.20
+ */
+gboolean
+gst_rtsp_media_factory_is_enable_rtcp (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = priv->enable_rtcp;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
 static gchar *
 default_gen_key (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
 {
@@ -1757,6 +1833,7 @@ default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
   GstElement *element, *pipeline;
   GstRTSPMediaFactoryClass *klass;
   GType media_gtype;
+  gboolean enable_rtcp;
 
   klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
 
@@ -1769,6 +1846,7 @@ default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
 
   GST_RTSP_MEDIA_FACTORY_LOCK (factory);
   media_gtype = factory->priv->media_gtype;
+  enable_rtcp = factory->priv->enable_rtcp;
   GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
 
   /* create a new empty media */
@@ -1776,6 +1854,9 @@ default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
       g_object_new (media_gtype, "element", element, "transport-mode",
       factory->priv->transport_mode, NULL);
 
+  /* We need to call this prior to collecting streams */
+  gst_rtsp_media_set_enable_rtcp (media, enable_rtcp);
+
   gst_rtsp_media_collect_streams (media);
 
   pipeline = klass->create_pipeline (factory, media);
diff --git a/gst/rtsp-server/rtsp-media-factory.h b/gst/rtsp-server/rtsp-media-factory.h
index dd08498..8e847fd 100644
--- a/gst/rtsp-server/rtsp-media-factory.h
+++ b/gst/rtsp-server/rtsp-media-factory.h
@@ -258,6 +258,13 @@ void                  gst_rtsp_media_factory_set_dscp_qos (GstRTSPMediaFactory *
 GST_RTSP_SERVER_API
 gint                  gst_rtsp_media_factory_get_dscp_qos (GstRTSPMediaFactory * factory);
 
+GST_RTSP_SERVER_API
+void                  gst_rtsp_media_factory_set_enable_rtcp (GstRTSPMediaFactory * factory,
+                                                              gboolean enable);
+
+GST_RTSP_SERVER_API
+gboolean              gst_rtsp_media_factory_is_enable_rtcp (GstRTSPMediaFactory * factory);
+
 /* creating the media from the factory and a url */
 
 GST_RTSP_SERVER_API
diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index eb9ead1..b22b95f 100644
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -112,6 +112,7 @@ struct _GstRTSPMediaPrivate
   gchar *multicast_iface;
   guint max_mcast_ttl;
   gboolean bind_mcast_address;
+  gboolean enable_rtcp;
   gboolean blocked;
   GstRTSPTransportMode transport_mode;
   gboolean stop_on_disconnect;
@@ -180,6 +181,7 @@ struct _GstRTSPMediaPrivate
 #define DEFAULT_MAX_MCAST_TTL   255
 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
 #define DEFAULT_DO_RATE_CONTROL TRUE
+#define DEFAULT_ENABLE_RTCP     TRUE
 
 #define DEFAULT_DO_RETRANSMISSION FALSE
 
@@ -492,6 +494,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
   priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
   priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
   priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+  priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
   priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
   priv->dscp_qos = DEFAULT_DSCP_QOS;
   priv->expected_async_done = FALSE;
@@ -2104,6 +2107,20 @@ gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
   return result;
 }
 
+void
+gst_rtsp_media_set_enable_rtcp (GstRTSPMedia * media, gboolean enable)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->enable_rtcp = enable;
+  g_mutex_unlock (&priv->lock);
+}
+
 static GList *
 _find_payload_types (GstRTSPMedia * media)
 {
@@ -2425,6 +2442,7 @@ gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
   gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
   gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
   gst_rtsp_stream_set_bind_mcast_address (stream, priv->bind_mcast_address);
+  gst_rtsp_stream_set_enable_rtcp (stream, priv->enable_rtcp);
   gst_rtsp_stream_set_profiles (stream, priv->profiles);
   gst_rtsp_stream_set_protocols (stream, priv->protocols);
   gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
diff --git a/gst/rtsp-server/rtsp-server-internal.h b/gst/rtsp-server/rtsp-server-internal.h
index 008b755..b5aaeff 100644
--- a/gst/rtsp-server/rtsp-server-internal.h
+++ b/gst/rtsp-server/rtsp-server-internal.h
@@ -58,6 +58,9 @@ gboolean                 gst_rtsp_stream_transport_check_back_pressure (GstRTSPS
 
 gboolean                 gst_rtsp_stream_is_tcp_receiver (GstRTSPStream * stream);
 
+void                     gst_rtsp_media_set_enable_rtcp (GstRTSPMedia *media, gboolean enable);
+void                     gst_rtsp_stream_set_enable_rtcp (GstRTSPStream *stream, gboolean enable);
+
 G_END_DECLS
 
 #endif /* __GST_RTSP_SERVER_INTERNAL_H__ */
diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index 7585a0c..3243e59 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -220,6 +220,9 @@ struct _GstRTSPStreamPrivate
   guint32 blocked_seqnum;
   guint32 blocked_rtptime;
   GstClockTime blocked_running_time;
+
+  /* Whether we should send and receive RTCP */
+  gboolean enable_rtcp;
 };
 
 #define DEFAULT_CONTROL         NULL
@@ -229,6 +232,7 @@ struct _GstRTSPStreamPrivate
 #define DEFAULT_MAX_MCAST_TTL   255
 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
 #define DEFAULT_DO_RATE_CONTROL TRUE
+#define DEFAULT_ENABLE_RTCP TRUE
 
 enum
 {
@@ -328,6 +332,7 @@ gst_rtsp_stream_init (GstRTSPStream * stream)
   priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
   priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
   priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
+  priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
 
   g_mutex_init (&priv->lock);
 
@@ -1422,6 +1427,7 @@ alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
 
   /* Start with random port */
   tmp_rtp = 0;
+  tmp_rtcp = 0;
 
   if (use_transport_settings) {
     if (!multicast)
@@ -1453,14 +1459,16 @@ alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
     }
   }
 
-  rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
-      G_SOCKET_PROTOCOL_UDP, NULL);
-  if (!rtcp_socket)
-    goto no_udp_protocol;
-  g_socket_set_multicast_loopback (rtcp_socket, FALSE);
+  if (priv->enable_rtcp) {
+    rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+        G_SOCKET_PROTOCOL_UDP, NULL);
+    if (!rtcp_socket)
+      goto no_udp_protocol;
+    g_socket_set_multicast_loopback (rtcp_socket, FALSE);
+  }
 
-  /* try to allocate 2 UDP ports, the RTP port should be an even
-   * number and the RTCP port should be the next (uneven) port */
+  /* try to allocate UDP ports, the RTP port should be an even
+   * number and the RTCP port (if enabled) should be the next (uneven) port */
 again:
 
   if (rtp_socket == NULL) {
@@ -1496,7 +1504,8 @@ again:
       if (*server_addr_out)
         addr = *server_addr_out;
       else
-        addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
+        addr = gst_rtsp_address_pool_acquire_address (pool, flags,
+            priv->enable_rtcp ? 2 : 1);
 
       if (addr == NULL)
         goto no_address;
@@ -1556,18 +1565,20 @@ again:
   g_object_unref (rtp_sockaddr);
 
   /* set port */
-  tmp_rtcp = tmp_rtp + 1;
+  if (priv->enable_rtcp) {
+    tmp_rtcp = tmp_rtp + 1;
 
-  rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
-  if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
-    GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
+    rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
+    if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
+      GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
+      g_object_unref (rtcp_sockaddr);
+      g_clear_object (&rtp_socket);
+      if (transport_settings_defined)
+        goto transport_settings_error;
+      goto again;
+    }
     g_object_unref (rtcp_sockaddr);
-    g_clear_object (&rtp_socket);
-    if (transport_settings_defined)
-      goto transport_settings_error;
-    goto again;
   }
-  g_object_unref (rtcp_sockaddr);
 
   if (!addr) {
     addr = g_slice_new0 (GstRTSPAddress);
@@ -1585,15 +1596,21 @@ again:
   if (multicast && (ct->ttl > 0) && (ct->ttl <= priv->max_mcast_ttl)) {
     GST_DEBUG ("setting mcast ttl to %d", ct->ttl);
     g_socket_set_multicast_ttl (rtp_socket, ct->ttl);
-    g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
+    if (rtcp_socket)
+      g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
   }
 
   socket_out[0] = rtp_socket;
   socket_out[1] = rtcp_socket;
   *server_addr_out = addr;
 
-  GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
-      addr->address, tmp_rtp, tmp_rtcp);
+  if (priv->enable_rtcp) {
+    GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
+        addr->address, tmp_rtp, tmp_rtcp);
+  } else {
+    GST_DEBUG_OBJECT (stream, "allocated address: %s and port: %d",
+        addr->address, tmp_rtp);
+  }
 
   g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
 
@@ -1922,14 +1939,18 @@ gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
   if (family == G_SOCKET_FAMILY_IPV4) {
     if (server_port && priv->server_addr_v4) {
       server_port->min = priv->server_addr_v4->port;
-      server_port->max =
-          priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
+      if (priv->enable_rtcp) {
+        server_port->max =
+            priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
+      }
     }
   } else {
     if (server_port && priv->server_addr_v6) {
       server_port->min = priv->server_addr_v6->port;
-      server_port->max =
-          priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
+      if (priv->enable_rtcp) {
+        server_port->max =
+            priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
+      }
     }
   }
   g_mutex_unlock (&priv->lock);
@@ -2262,6 +2283,16 @@ gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream)
   return result;
 }
 
+void
+gst_rtsp_stream_set_enable_rtcp (GstRTSPStream * stream, gboolean enable)
+{
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  g_mutex_lock (&stream->priv->lock);
+  stream->priv->enable_rtcp = enable;
+  g_mutex_unlock (&stream->priv->lock);
+}
+
 /* executed from streaming thread */
 static void
 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
@@ -3518,10 +3549,10 @@ create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
     g_object_set (priv->payloader, "onvif-no-rate-control",
         !priv->do_rate_control, NULL);
 
-  for (i = 0; i < 2; i++) {
+  for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
     gboolean link_tee = FALSE;
     /* For the sender we create this bit of pipeline for both
-     * RTP and RTCP.
+     * RTP and RTCP (when enabled).
      * Initially there will be only one active transport for
      * the stream, so the pipeline will look like this:
      *
@@ -3674,9 +3705,9 @@ create_receiver_part (GstRTSPStream * stream, const GstRTSPTransport *
       "RTP caps: %" GST_PTR_FORMAT " RTCP caps: %" GST_PTR_FORMAT, rtp_caps,
       rtcp_caps);
 
-  for (i = 0; i < 2; i++) {
+  for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
     /* For the receiver we create this bit of pipeline for both
-     * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
+     * RTP and RTCP (when enabled). We receive RTP/RTCP on appsrc and udpsrc
      * and it is all funneled into the rtpbin receive pad.
      *
      *
@@ -3938,12 +3969,15 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
     g_free (name);
   }
 
-  name = g_strdup_printf ("send_rtcp_src_%u", idx);
-  priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
-  g_free (name);
-  name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
-  priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
-  g_free (name);
+  if (priv->enable_rtcp) {
+    name = g_strdup_printf ("send_rtcp_src_%u", idx);
+    priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
+    g_free (name);
+
+    name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
+    priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
+    g_free (name);
+  }
 
   /* get the session */
   g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
@@ -4085,7 +4119,7 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
     priv->recv_rtp_src = NULL;
   }
 
-  for (i = 0; i < 2; i++) {
+  for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
     clear_element (bin, &priv->udpsrc_v4[i]);
     clear_element (bin, &priv->udpsrc_v6[i]);
     clear_element (bin, &priv->udpqueue[i]);
@@ -4115,9 +4149,11 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
     priv->send_src[0] = NULL;
   }
 
-  gst_element_release_request_pad (rtpbin, priv->send_src[1]);
-  gst_object_unref (priv->send_src[1]);
-  priv->send_src[1] = NULL;
+  if (priv->enable_rtcp) {
+    gst_element_release_request_pad (rtpbin, priv->send_src[1]);
+    gst_object_unref (priv->send_src[1]);
+    priv->send_src[1] = NULL;
+  }
 
   g_object_unref (priv->session);
   priv->session = NULL;
@@ -6206,8 +6242,8 @@ gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled)
   if (stream->priv->appsink[0])
     g_object_set (stream->priv->appsink[0], "sync", enabled, NULL);
   if (stream->priv->payloader
-      && g_object_class_find_property (G_OBJECT_GET_CLASS (stream->
-              priv->payloader), "onvif-no-rate-control"))
+      && g_object_class_find_property (G_OBJECT_GET_CLASS (stream->priv->
+              payloader), "onvif-no-rate-control"))
     g_object_set (stream->priv->payloader, "onvif-no-rate-control", !enabled,
         NULL);
   if (stream->priv->session) {
diff --git a/tests/check/gst/client.c b/tests/check/gst/client.c
index 6bf6544..d506f9a 100644
--- a/tests/check/gst/client.c
+++ b/tests/check/gst/client.c
@@ -206,7 +206,7 @@ create_connection (GstRTSPConnection ** conn)
 }
 
 static GstRTSPClient *
-setup_client (const gchar * launch_line)
+setup_client (const gchar * launch_line, gboolean enable_rtcp)
 {
   GstRTSPClient *client;
   GstRTSPSessionPool *session_pool;
@@ -227,6 +227,8 @@ setup_client (const gchar * launch_line)
   else
     gst_rtsp_media_factory_set_launch (factory, launch_line);
 
+  gst_rtsp_media_factory_set_enable_rtcp (factory, enable_rtcp);
+
   gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
   gst_rtsp_client_set_mount_points (client, mount_points);
 
@@ -515,7 +517,7 @@ GST_START_TEST (test_describe)
   g_object_unref (client);
 
   /* simple DESCRIBE for an existing url */
-  client = setup_client (NULL);
+  client = setup_client (NULL, TRUE);
   fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
           "rtsp://localhost/test") == GST_RTSP_OK);
   str = g_strdup_printf ("%d", cseq);
@@ -688,7 +690,7 @@ GST_START_TEST (test_setup_tcp)
   GstRTSPMessage request = { 0, };
   gchar *str;
 
-  client = setup_client (NULL);
+  client = setup_client (NULL, TRUE);
   create_connection (&conn);
   fail_unless (gst_rtsp_client_set_connection (client, conn));
 
@@ -714,6 +716,40 @@ GST_START_TEST (test_setup_tcp)
 
 GST_END_TEST;
 
+GST_START_TEST (test_setup_no_rtcp)
+{
+  GstRTSPClient *client;
+  GstRTSPConnection *conn;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  client = setup_client (NULL, FALSE);
+  create_connection (&conn);
+  fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;unicast;client_port=5000-5001");
+
+  gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+  /* We want to verify that server_port holds a single number, not a range */
+  expected_transport =
+      "RTP/AVP;unicast;client_port=5000-5001;server_port=[0-9]+;ssrc=.*;mode=\"PLAY\"";
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+
+  gst_rtsp_message_unset (&request);
+
+  send_teardown (client);
+  teardown_client (client);
+}
+
+GST_END_TEST;
+
 GST_START_TEST (test_setup_tcp_two_streams_same_channels)
 {
   GstRTSPClient *client;
@@ -721,7 +757,7 @@ GST_START_TEST (test_setup_tcp_two_streams_same_channels)
   GstRTSPMessage request = { 0, };
   gchar *str;
 
-  client = setup_client (NULL);
+  client = setup_client (NULL, TRUE);
   create_connection (&conn);
   fail_unless (gst_rtsp_client_set_connection (client, conn));
 
@@ -1044,7 +1080,7 @@ test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
   gchar *str;
 
   /* simple DESCRIBE for an existing url */
-  client = setup_client (launch_line);
+  client = setup_client (launch_line, TRUE);
   fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
           "rtsp://localhost/test") == GST_RTSP_OK);
   str = g_strdup_printf ("%d", cseq);
@@ -2044,6 +2080,7 @@ rtspclient_suite (void)
   tcase_add_test (tc, test_options);
   tcase_add_test (tc, test_describe);
   tcase_add_test (tc, test_setup_tcp);
+  tcase_add_test (tc, test_setup_no_rtcp);
   tcase_add_test (tc, test_setup_tcp_two_streams_same_channels);
   tcase_add_test (tc, test_client_multicast_transport_404);
   tcase_add_test (tc, test_client_multicast_transport);