#include <rtsp-client.h>
+#define VIDEO_PIPELINE "videotestsrc ! " \
+ "video/x-raw,width=352,height=288 ! " \
+ "rtpgstpay name=pay0 pt=96"
+#define AUDIO_PIPELINE "audiotestsrc ! " \
+ "audio/x-raw,rate=8000 ! " \
+ "rtpgstpay name=pay1 pt=97"
+
static gchar *session_id;
static gint cseq;
static guint expected_session_timeout = 60;
return TRUE;
}
+static void
+create_connection (GstRTSPConnection ** conn)
+{
+ GSocket *sock;
+ GError *error = NULL;
+
+ sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
+ G_SOCKET_PROTOCOL_TCP, &error);
+ g_assert_no_error (error);
+ fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1", 444,
+ NULL, conn) == GST_RTSP_OK);
+ g_object_unref (sock);
+}
+
static GstRTSPClient *
setup_client (const gchar * launch_line)
{
factory = gst_rtsp_media_factory_new ();
if (launch_line == NULL)
gst_rtsp_media_factory_set_launch (factory,
- "videotestsrc ! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96");
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
else
gst_rtsp_media_factory_set_launch (factory, launch_line);
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPConnection *conn;
- GSocket *sock;
- GError *error = NULL;
client = gst_rtsp_client_new ();
/* OPTIONS with an absolute path instead of an absolute url */
/* set host information */
- sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
- G_SOCKET_PROTOCOL_TCP, &error);
- g_assert_no_error (error);
- gst_rtsp_connection_create_from_socket (sock, "localhost", 444, NULL, &conn);
+ create_connection (&conn);
fail_unless (gst_rtsp_client_set_connection (client, conn));
- g_object_unref (sock);
-
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
static const gchar *expected_transport = NULL;
static gboolean
-test_setup_response_200_multicast (GstRTSPClient * client,
- GstRTSPMessage * response, gboolean close, gpointer user_data)
+test_setup_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
+ gchar *pattern;
GstRTSPSessionPool *session_pool;
GstRTSPSession *session;
gchar **session_hdr_params;
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT,
&str, 0) == GST_RTSP_OK);
- fail_unless_equals_string (str, expected_transport);
+ pattern = g_strdup_printf ("^%s$", expected_transport);
+ fail_unless (g_regex_match_simple (pattern, str, 0, 0),
+ "Transport '%s' doesn't match pattern '%s'", str, pattern);
+ g_free (pattern);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
&str, 0) == GST_RTSP_OK);
g_strfreev (session_hdr_params);
/* remember session id to be able to send teardown */
+ if (session_id)
+ g_free (session_id);
session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
fail_unless (session_id != NULL);
}
static gboolean
+test_setup_response_461 (GstRTSPClient * client,
+ GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+ GstRTSPStatusCode code;
+ const gchar *reason;
+ GstRTSPVersion version;
+ gchar *str;
+
+ fail_unless (expected_transport == NULL);
+
+ fail_unless (gst_rtsp_message_get_type (response) ==
+ GST_RTSP_MESSAGE_RESPONSE);
+
+ fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+ &version)
+ == GST_RTSP_OK);
+ fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT);
+ fail_unless (g_str_equal (reason, "Unsupported transport"));
+ fail_unless (version == GST_RTSP_VERSION_1_0);
+
+ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+ 0) == GST_RTSP_OK);
+ fail_unless (atoi (str) == cseq++);
+
+
+ return TRUE;
+}
+
+static gboolean
test_teardown_response_200 (GstRTSPClient * client,
GstRTSPMessage * response, gboolean close, gpointer user_data)
{
session_id = NULL;
}
+GST_START_TEST (test_setup_tcp)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ 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/TCP;unicast");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=0-1;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;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ /* test SETUP of a video stream with 0-1 as interleaved channels */
+ 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/TCP;unicast;interleaved=0-1");
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* test SETUP of an audio stream with *the same* interleaved channels.
+ * we expect the server to allocate new channel numbers */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=1") == 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/TCP;unicast;interleaved=0-1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=2-3;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;
+
static GstRTSPClient *
-setup_multicast_client (void)
+setup_multicast_client (guint max_ttl)
{
GstRTSPClient *client;
GstRTSPSessionPool *session_pool;
"media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
gst_rtsp_client_set_mount_points (client, mount_points);
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, max_ttl);
thread_pool = gst_rtsp_thread_pool_new ();
gst_rtsp_client_set_thread_pool (client, thread_pool);
GstRTSPMessage request = { 0, };
gchar *str;
- client = setup_multicast_client ();
+ client = setup_multicast_client (1);
/* simple SETUP for non-existing url */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
GstRTSPMessage request = { 0, };
gchar *str;
- client = setup_multicast_client ();
+ client = setup_multicast_client (1);
expected_session_timeout = 20;
g_signal_connect (G_OBJECT (client), "new-session",
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
GstRTSPMessage request = { 0, };
gchar *str;
- client = setup_multicast_client ();
+ client = setup_multicast_client (1);
/* simple SETUP with a valid URI and multicast and a specific dest,
* but ignore it */
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
GST_END_TEST;
-GST_START_TEST (test_client_multicast_transport_specific)
+static void
+multicast_transport_specific (void)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
GstRTSPSessionPool *session_pool;
GstRTSPContext ctx = { NULL };
- client = setup_multicast_client ();
+ client = setup_multicast_client (1);
ctx.client = client;
ctx.auth = gst_rtsp_auth_new ();
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
expected_transport);
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
- expected_transport = NULL;
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
session_pool = gst_rtsp_client_get_session_pool (client);
fail_unless (session_pool != NULL);
fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
g_object_unref (session_pool);
- send_teardown (client);
+ /* send PLAY request */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
+ "rtsp://localhost/test") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ send_teardown (client);
teardown_client (client);
g_object_unref (ctx.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_context_pop_current (&ctx);
}
+/* CASE: multicast address requested by the client exists in the address pool */
+GST_START_TEST (test_client_multicast_transport_specific)
+{
+ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ multicast_transport_specific ();
+ expected_transport = NULL;
+}
+
+GST_END_TEST;
+
+/* CASE: multicast address requested by the client does not exist in the address pool */
+GST_START_TEST (test_client_multicast_transport_specific_no_address_in_pool)
+{
+ expected_transport = "RTP/AVP;multicast;destination=234.252.0.3;"
+ "ttl=1;port=6000-6001;mode=\"PLAY\"";
+ multicast_transport_specific ();
+ expected_transport = NULL;
+}
+
GST_END_TEST;
static gboolean
GST_END_TEST;
static void
-mcast_transport_specific_two_clients (gboolean shared)
+mcast_transport_two_clients (gboolean shared, const gchar * transport1,
+ const gchar * expected_transport1, const gchar * addr1,
+ const gchar * transport2, const gchar * expected_transport2,
+ const gchar * addr2)
{
- GstRTSPClient *client, *client2;
+ GstRTSPClient *client1, *client2;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPSessionPool *session_pool;
GstRTSPAddressPool *address_pool;
GstRTSPThreadPool *thread_pool;
gchar *session_id1;
+ gchar *client_addr = NULL;
mount_points = gst_rtsp_mount_points_new ();
factory = gst_rtsp_media_factory_new ();
thread_pool = gst_rtsp_thread_pool_new ();
/* first multicast client with transport specific request */
- client = gst_rtsp_client_new ();
- gst_rtsp_client_set_session_pool (client, session_pool);
- gst_rtsp_client_set_mount_points (client, mount_points);
- gst_rtsp_client_set_thread_pool (client, thread_pool);
+ client1 = gst_rtsp_client_new ();
+ gst_rtsp_client_set_session_pool (client1, session_pool);
+ gst_rtsp_client_set_mount_points (client1, mount_points);
+ gst_rtsp_client_set_thread_pool (client1, thread_pool);
- ctx.client = client;
+ ctx.client = client1;
ctx.auth = gst_rtsp_auth_new ();
ctx.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
"user", NULL);
gst_rtsp_context_push_current (&ctx);
- expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
- "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ expected_transport = expected_transport1;
/* send SETUP request */
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_take_header (&request, GST_RTSP_HDR_CSEQ, str);
- gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
- expected_transport);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1);
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
- fail_unless (gst_rtsp_client_handle_message (client,
+ gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client1,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = NULL;
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
- gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
- fail_unless (gst_rtsp_client_handle_message (client,
+ gst_rtsp_client_set_send_func (client1, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client1,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
+
+ /* check address */
+ client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx.stream);
+ fail_if (client_addr == NULL);
+ fail_unless (g_str_equal (client_addr, addr1));
+ g_free (client_addr);
+
gst_rtsp_context_pop_current (&ctx);
- session_id1 = session_id;
+ session_id1 = g_strdup (session_id);
/* second multicast client with transport specific request */
cseq = 0;
"user", NULL);
gst_rtsp_context_push_current (&ctx2);
- expected_transport = "RTP/AVP;multicast;destination=233.252.0.2;"
- "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ expected_transport = expected_transport2;
/* send SETUP request */
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_take_header (&request, GST_RTSP_HDR_CSEQ, str);
- gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
- expected_transport);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2);
- gst_rtsp_client_set_send_func (client2, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client2,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
+ /* check addresses */
+ client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx2.stream);
+ fail_if (client_addr == NULL);
+ if (shared) {
+ if (g_str_equal (addr1, addr2)) {
+ fail_unless (g_str_equal (client_addr, addr1));
+ } else {
+ gchar *addr_str = g_strdup_printf ("%s,%s", addr2, addr1);
+ fail_unless (g_str_equal (client_addr, addr_str));
+ g_free (addr_str);
+ }
+ } else {
+ fail_unless (g_str_equal (client_addr, addr2));
+ }
+ g_free (client_addr);
+
send_teardown (client2);
gst_rtsp_context_pop_current (&ctx2);
gst_rtsp_context_push_current (&ctx);
session_id = session_id1;
- send_teardown (client);
+ send_teardown (client1);
gst_rtsp_context_pop_current (&ctx);
- teardown_client (client);
+ teardown_client (client1);
teardown_client (client2);
g_object_unref (ctx.auth);
g_object_unref (ctx2.auth);
g_object_unref (thread_pool);
}
-/* test if two multicast clients can choose different transport settings */
+/* test if two multicast clients can choose different transport settings
+ * CASE: media is shared */
GST_START_TEST
(test_client_multicast_transport_specific_two_clients_shared_media) {
- mcast_transport_specific_two_clients (TRUE);
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients can choose different transport settings
+ * CASE: media is not shared */
+GST_START_TEST (test_client_multicast_transport_specific_two_clients)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (FALSE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients can choose the same transport settings.
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_transport_specific_two_clients_shared_media_same_transport)
+{
+
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = transport_client_1;
+ const gchar *expected_transport_2 = expected_transport_1;
+ const gchar *addr_client_2 = addr_client_1;
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients get the same transport settings without
+ * requesting specific transport.
+ * CASE: media is shared */
+GST_START_TEST (test_client_multicast_two_clients_shared_media)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_1 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = transport_client_1;
+ const gchar *expected_transport_2 = expected_transport_1;
+ const gchar *addr_client_2 = addr_client_1;
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+
+/* test if two multicast clients get the different transport settings: the first client
+ * requests the specific transport configuration while the second client lets
+ * the server select the multicast address and the ports.
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_two_clients_first_specific_transport_shared_media) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = expected_transport_1;
+ const gchar *addr_client_2 = addr_client_1;
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+/* test if two multicast clients get the different transport settings: the first client lets
+ * the server select the multicast address and the ports while the second client requests
+ * the specific transport configuration.
+ * CASE: media is shared */
+GST_START_TEST
+ (test_client_multicast_two_clients_second_specific_transport_shared_media) {
+ const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
+ const gchar *expected_transport_1 =
+ "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=2;port=5004-5005;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5004";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+
+/* test if the maximum ttl multicast value is chosen by the server
+ * CASE: the first client provides the highest ttl value */
+GST_START_TEST (test_client_multicast_max_ttl_first_client)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=3;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 =
+ "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=3;port=5002-5003;mode=\"PLAY\"";
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+
+/* test if the maximum ttl multicast value is chosen by the server
+ * CASE: the second client provides the highest ttl value */
+GST_START_TEST (test_client_multicast_max_ttl_second_client)
+{
+ const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=2;port=5000-5001;mode=\"PLAY\"";
+ const gchar *expected_transport_1 = transport_client_1;
+ const gchar *addr_client_1 = "233.252.0.1:5000";
+
+ const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=4;port=5002-5003;mode=\"PLAY\"";
+ const gchar *expected_transport_2 = transport_client_2;
+ const gchar *addr_client_2 = "233.252.0.2:5002";
+
+ mcast_transport_two_clients (TRUE, transport_client_1,
+ expected_transport_1, addr_client_1, transport_client_2,
+ expected_transport_2, addr_client_2);
+}
+
+GST_END_TEST;
+GST_START_TEST (test_client_multicast_invalid_ttl)
+{
+ GstRTSPClient *client;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+ GstRTSPSessionPool *session_pool;
+ GstRTSPContext ctx = { NULL };
+
+ client = setup_multicast_client (3);
+
+ ctx.client = client;
+ ctx.auth = gst_rtsp_auth_new ();
+ ctx.token =
+ gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+ G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+ "user", NULL);
+ gst_rtsp_context_push_current (&ctx);
+
+ /* simple SETUP with an invalid ttl=0 */
+ 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_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP;multicast;destination=233.252.0.1;ttl=0;port=5000-5001;");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ session_pool = gst_rtsp_client_get_session_pool (client);
+ fail_unless (session_pool != NULL);
+ fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+ g_object_unref (session_pool);
+
+ teardown_client (client);
+ g_object_unref (ctx.auth);
+ gst_rtsp_token_unref (ctx.token);
+ gst_rtsp_context_pop_current (&ctx);
}
GST_END_TEST;
tcase_add_test (tc, test_request);
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_tcp_two_streams_same_channels);
tcase_add_test (tc, test_client_multicast_transport_404);
tcase_add_test (tc, test_client_multicast_transport);
tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
tcase_add_test (tc,
test_client_multicast_transport_specific_two_clients_shared_media);
+ tcase_add_test (tc, test_client_multicast_transport_specific_two_clients);
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_two_clients_shared_media_same_transport);
+ tcase_add_test (tc, test_client_multicast_two_clients_shared_media);
+ tcase_add_test (tc,
+ test_client_multicast_two_clients_first_specific_transport_shared_media);
+ tcase_add_test (tc,
+ test_client_multicast_two_clients_second_specific_transport_shared_media);
+ tcase_add_test (tc,
+ test_client_multicast_transport_specific_no_address_in_pool);
+ tcase_add_test (tc, test_client_multicast_max_ttl_first_client);
+ tcase_add_test (tc, test_client_multicast_max_ttl_second_client);
+ tcase_add_test (tc, test_client_multicast_invalid_ttl);
return s;
}
projects / platform / upstream / gstreamer.git / blobdiff