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tests: Add test to check selecting a port the server will send from

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This commit is contained in:
Olivier Crête 2013-02-20 19:37:51 -05:00 committed by Wim Taymans
parent d3c70d4d51
commit 6a2238b2fb
1 changed files with 104 additions and 1 deletions
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105
tests/check/gst/rtspserver.c
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@ -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;
}