diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 964e06bfeb..5dac08b9d1 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -2026,7 +2026,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
   if (str)
     gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
 
-  if (!gst_rtsp_media_is_receive_only (media)) {
+  if (gst_rtsp_media_has_completed_sender (media)) {
     /* the scale and speed headers must always be added if they were present in
      * the request. however, even if they were not, we still add them if
      * applied_rate or rate deviate from the "normal", i.e. 1.0 */
diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index 4aaed7ce6e..35a9ec0cfd 100644
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -2562,7 +2562,10 @@ gst_rtsp_media_get_rates (GstRTSPMedia * media, gdouble * rate,
 {
   GstRTSPMediaPrivate *priv;
   GstRTSPStream *stream;
+  gdouble save_rate, save_applied_rate;
   gboolean result = TRUE;
+  gboolean first_stream = TRUE;
+  gint i;
 
   g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
 
@@ -2576,11 +2579,34 @@ gst_rtsp_media_get_rates (GstRTSPMedia * media, gdouble * rate,
   g_mutex_lock (&priv->lock);
 
   g_assert (priv->streams->len > 0);
-  stream = g_ptr_array_index (priv->streams, 0);
-  if (!gst_rtsp_stream_get_rates (stream, rate, applied_rate)) {
+  for (i = 0; i < priv->streams->len; i++) {
+    stream = g_ptr_array_index (priv->streams, i);
+    if (gst_rtsp_stream_is_complete (stream)) {
+      if (gst_rtsp_stream_get_rates (stream, rate, applied_rate)) {
+        if (first_stream) {
+          save_rate = *rate;
+          save_applied_rate = *applied_rate;
+          first_stream = FALSE;
+        } else {
+          if (save_rate != *rate || save_applied_rate != *applied_rate) {
+            /* diffrent rate or applied_rate, weird */
+            g_assert (FALSE);
+            result = FALSE;
+            break;
+          }
+        }
+      } else {
+        /* complete stream withot rate and applied_rate, weird */
+        g_assert (FALSE);
+        result = FALSE;
+        break;
+      }
+    }
+  }
+
+  if (!result) {
     GST_WARNING_OBJECT (media,
-        "failed to obtain rate and applied_rate from first stream");
-    result = FALSE;
+        "failed to obtain consistent rate and applied_rate");
   }
 
   g_mutex_unlock (&priv->lock);
@@ -4781,6 +4807,36 @@ gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
   return receive_only;
 }
 
+/**
+ * gst_rtsp_media_has_completed_sender:
+ *
+ * See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender().
+ *    
+ * Returns: whether @media has at least one complete sender stream.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_has_completed_sender (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  gboolean sender = FALSE;
+  guint i;
+
+  g_mutex_lock (&priv->lock);
+  for (i = 0; i < priv->streams->len; i++) {
+    GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+    if (gst_rtsp_stream_is_complete (stream))
+      if (gst_rtsp_stream_is_sender (stream) ||
+          !gst_rtsp_stream_is_receiver (stream)) {
+        sender = TRUE;
+        break;
+      }
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return sender;
+}
+
 /**
  * gst_rtsp_media_set_rate_control:
  *
diff --git a/gst/rtsp-server/rtsp-media.h b/gst/rtsp-server/rtsp-media.h
index 987a8ad017..5379f65e64 100644
--- a/gst/rtsp-server/rtsp-media.h
+++ b/gst/rtsp-server/rtsp-media.h
@@ -421,6 +421,9 @@ gboolean              gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GP
 GST_RTSP_SERVER_API
 gboolean              gst_rtsp_media_is_receive_only (GstRTSPMedia * media);
 
+GST_RTSP_SERVER_API
+gboolean              gst_rtsp_media_has_completed_sender (GstRTSPMedia * media);
+
 GST_RTSP_SERVER_API
 void                  gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled);
 
diff --git a/tests/check/gst/rtspserver.c b/tests/check/gst/rtspserver.c
index d342a436d3..f3bd5d0676 100644
--- a/tests/check/gst/rtspserver.c
+++ b/tests/check/gst/rtspserver.c
@@ -2401,6 +2401,78 @@ GST_START_TEST (test_multiple_transports)
 
 GST_END_TEST;
 
+GST_START_TEST (test_suspend_mode_reset_only_audio)
+{
+  GstRTSPMountPoints *mounts;
+  gchar *service;
+  GstRTSPMediaFactory *factory;
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *audio_control;
+  GstRTSPRange client_port;
+  gchar *session = NULL;
+  GstRTSPTransport *audio_transport = NULL;
+  GSocket *rtp_socket, *rtcp_socket;
+
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_suspend_mode (factory,
+      GST_RTSP_SUSPEND_MODE_RESET);
+  gst_rtsp_media_factory_set_launch (factory,
+      "( " VIDEO_PIPELINE "  " AUDIO_PIPELINE " )");
+  gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+  g_object_unref (mounts);
+
+  /* set port to any */
+  gst_rtsp_server_set_service (server, "0");
+
+  /* attach to default main context */
+  source_id = gst_rtsp_server_attach (server, NULL);
+  fail_if (source_id == 0);
+
+  /* get port */
+  service = gst_rtsp_server_get_service (server);
+  test_port = atoi (service);
+  fail_unless (test_port != 0);
+  g_free (service);
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+  /* do SETUP for audio */
+  fail_unless (do_setup (conn, audio_control, &client_port, &session,
+          &audio_transport) == GST_RTSP_STS_OK);
+
+  /* send PLAY request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          session) == GST_RTSP_STS_OK);
+
+  /* send TEARDOWN request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session) == GST_RTSP_STS_OK);
+
+  /* clean up and iterate so the clean-up can finish */
+  g_free (session);
+  gst_rtsp_transport_free (audio_transport);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
 static Suite *
 rtspserver_suite (void)
 {
@@ -2439,6 +2511,7 @@ rtspserver_suite (void)
   tcase_add_test (tc, test_announce_without_sdp);
   tcase_add_test (tc, test_record_tcp);
   tcase_add_test (tc, test_multiple_transports);
+  tcase_add_test (tc, test_suspend_mode_reset_only_audio);
 
   return s;
 }