diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 9389160f3e..799b42e512 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -174,6 +174,7 @@ static void
 gst_rtsp_client_init (GstRTSPClient * client)
 {
   client->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
+  client->teardown_response_seq = 0;
 }
 
 static void
@@ -300,7 +301,7 @@ gst_rtsp_client_new (void)
 
 static void
 send_response (GstRTSPClient * client, GstRTSPSession * session,
-    GstRTSPMessage * response)
+    GstRTSPMessage * response, guint * id)
 {
   gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
       "GStreamer RTSP server");
@@ -318,7 +319,7 @@ send_response (GstRTSPClient * client, GstRTSPSession * session,
     gst_rtsp_message_dump (response);
   }
 
-  gst_rtsp_watch_send_message (client->watch, response, NULL);
+  gst_rtsp_watch_send_message (client->watch, response, id);
   gst_rtsp_message_unset (response);
 }
 
@@ -329,7 +330,7 @@ send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
   gst_rtsp_message_init_response (state->response, code,
       gst_rtsp_status_as_text (code), state->request);
 
-  send_response (client, NULL, state->response);
+  send_response (client, NULL, state->response, NULL);
 }
 
 static void
@@ -344,7 +345,7 @@ handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
     gst_rtsp_auth_setup_auth (auth, client, 0, state);
   }
 
-  send_response (client, state->session, state->response);
+  send_response (client, state->session, state->response, NULL);
 }
 
 
@@ -603,14 +604,15 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
   gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
       "close");
 
-  send_response (client, session, state->response);
+  /* send the response and store the seq number so we can wait until it's
+   * written to the client to close the connection */
+  send_response (client, session, state->response,
+      &client->teardown_response_seq);
 
   /* we emit the signal before closing the connection */
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
       0, state);
 
-  close_connection (client);
-
   return TRUE;
 
   /* ERRORS */
@@ -646,7 +648,7 @@ handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
     if (res != GST_RTSP_OK)
       goto bad_request;
 
-    send_response (client, state->session, state->response);
+    send_response (client, state->session, state->response, NULL);
   }
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
@@ -682,7 +684,7 @@ handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
     if (res != GST_RTSP_OK)
       goto bad_request;
 
-    send_response (client, state->session, state->response);
+    send_response (client, state->session, state->response, NULL);
   }
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
@@ -731,7 +733,7 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
   gst_rtsp_message_init_response (state->response, code,
       gst_rtsp_status_as_text (code), state->request);
 
-  send_response (client, session, state->response);
+  send_response (client, session, state->response, NULL);
 
   /* the state is now READY */
   media->state = GST_RTSP_STATE_READY;
@@ -853,7 +855,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
   str = gst_rtsp_media_get_range_string (media->media, TRUE);
   gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
 
-  send_response (client, session, state->response);
+  send_response (client, session, state->response, NULL);
 
   /* start playing after sending the request */
   gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
@@ -1179,7 +1181,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
       trans_str);
   g_free (trans_str);
 
-  send_response (client, session, state->response);
+  send_response (client, session, state->response, NULL);
 
   /* update the state */
   switch (sessmedia->state) {
@@ -1368,7 +1370,7 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
   gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
   gst_sdp_message_free (sdp);
 
-  send_response (client, state->session, state->response);
+  send_response (client, state->session, state->response, NULL);
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
       0, state);
@@ -1410,7 +1412,7 @@ handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
   gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
   g_free (str);
 
-  send_response (client, state->session, state->response);
+  send_response (client, state->session, state->response, NULL);
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
       0, state);
@@ -1891,11 +1893,13 @@ message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
 static GstRTSPResult
 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
 {
-  /* GstRTSPClient *client; */
+  GstRTSPClient *client;
 
-  /* client = GST_RTSP_CLIENT (user_data); */
-
-  /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
+  client = GST_RTSP_CLIENT (user_data);
+  if (client->teardown_response_seq && client->teardown_response_seq == cseq) {
+    client->teardown_response_seq = 0;
+    close_connection (client);
+  }
 
   return GST_RTSP_OK;
 }
diff --git a/gst/rtsp-server/rtsp-client.h b/gst/rtsp-server/rtsp-client.h
index 71a28adc9c..dd4e71230e 100644
--- a/gst/rtsp-server/rtsp-client.h
+++ b/gst/rtsp-server/rtsp-client.h
@@ -107,6 +107,8 @@ struct _GstRTSPClient {
 
   GList *transports;
   GList *sessions;
+
+  guint teardown_response_seq;
 };
 
 struct _GstRTSPClientClass {