summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorOlivier Crête <olivier.crete@collabora.com>2013-02-21 00:37:51 (GMT)
committerOlivier Crête <olivier.crete@collabora.com>2013-02-21 00:39:56 (GMT)
commitcf39e542b165ff56e047def621152f6502009085 (patch)
tree3158d654305635cb9ee7f904182110bc6593ee03
parent4bd2a209a1469de2584d6c357764384952a387ac (diff)
downloadgst-rtsp-server-misc-tests.tar.gz
gst-rtsp-server-misc-tests.tar.xz
tests: Add test to check selecting a port the server will send frommisc-tests
-rw-r--r--tests/check/gst/rtspserver.c105
1 files changed, 104 insertions, 1 deletions
diff --git a/tests/check/gst/rtspserver.c b/tests/check/gst/rtspserver.c
index 001450f..be51945 100644
--- a/tests/check/gst/rtspserver.c
+++ b/tests/check/gst/rtspserver.c
@@ -1212,6 +1212,109 @@ GST_START_TEST (test_play_disconnect)
GST_END_TEST;
+/* Only different with test_play is the specific ports selected */
+
+GST_START_TEST (test_play_specific_server_port)
+{
+ GstRTSPMountPoints *mounts;
+ gchar *service;
+ GstRTSPMediaFactory *factory;
+ GstRTSPAddressPool *pool;
+ GstRTSPConnection *conn;
+ GstSDPMessage *sdp_message = NULL;
+ const GstSDPMedia *sdp_media;
+ const gchar *video_control;
+ GstRTSPRange client_port;
+ gchar *session = NULL;
+ GstRTSPTransport *video_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
+ GSocketAddress *rtp_address, *rtcp_address;
+ guint16 rtp_port, rtcp_port;
+
+ mounts = gst_rtsp_server_get_mount_points (server);
+
+ factory = gst_rtsp_media_factory_new ();
+ pool = gst_rtsp_address_pool_new ();
+ gst_rtsp_address_pool_add_range_unicast (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
+ GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780);
+ gst_rtsp_media_factory_set_address_pool (factory, pool);
+ g_object_unref (pool);
+ gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " )");
+ gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+ g_object_unref (mounts);
+
+ /* set port */
+ test_port = get_unused_port (SOCK_STREAM);
+ service = g_strdup_printf ("%d", test_port);
+ gst_rtsp_server_set_service (server, service);
+ g_free (service);
+
+ /* attach to default main context */
+ source_id = gst_rtsp_server_attach (server, NULL);
+ fail_if (source_id == 0);
+
+ GST_DEBUG ("rtsp server listening on port %d", test_port);
+
+
+ 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) == 1);
+ sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+ video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+ get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+ /* do SETUP for video */
+ fail_unless (do_setup (conn, video_control, &client_port, &session,
+ &video_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);
+
+ receive_rtp (rtp_socket, &rtp_address);
+ receive_rtcp (rtcp_socket, &rtcp_address, 0);
+
+ fail_unless (G_IS_INET_SOCKET_ADDRESS (rtp_address));
+ fail_unless (G_IS_INET_SOCKET_ADDRESS (rtcp_address));
+ rtp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_address));
+ rtcp_port =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtcp_address));
+ fail_unless (rtp_port >= 7770 && rtp_port <= 7780 && rtp_port % 2 == 0);
+ fail_unless (rtcp_port >= 7770 && rtcp_port <= 7780 && rtcp_port % 2 == 1);
+ fail_unless (rtp_port + 1 == rtcp_port);
+
+ g_object_unref (rtp_address);
+ g_object_unref (rtcp_address);
+
+ /* send TEARDOWN request and check that we get 200 OK */
+ fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+ session) == GST_RTSP_STS_OK);
+
+ /* FIXME: The rtsp-server always disconnects the transport before
+ * sending the RTCP BYE
+ * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+ */
+
+ /* clean up and iterate so the clean-up can finish */
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
+ g_free (session);
+ gst_rtsp_transport_free (video_transport);
+ gst_sdp_message_free (sdp_message);
+ gst_rtsp_connection_free (conn);
+
+
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
static Suite *
rtspserver_suite (void)
{
@@ -1234,7 +1337,7 @@ rtspserver_suite (void)
tcase_add_test (tc, test_play_multithreaded_timeout_client);
tcase_add_test (tc, test_play_multithreaded_timeout_session);
tcase_add_test (tc, test_play_disconnect);
-
+ tcase_add_test (tc, test_play_specific_server_port);
return s;
}