client: Don't reserve multicast address in the client setting case

When two multicast clients request specific transport
configurations, and "transport.client-settings" parameter is
set to true, it's wrong to actually require that these two
clients request the same multicast group.
Removed test_client_multicast_invalid_transport_specific test
cases as they wrongly require that the requested destination
address is supposed to be present in the address pool, also in
the case when "transport.client-settings" parameter is set to true.

Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa

https://bugzilla.gnome.org/show_bug.cgi?id=793441

diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 92135f2..01e63bf 100644 (file)
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -1990,17 +1990,16 @@ default_configure_client_transport (GstRTSPClient * client,
       goto error_allocating_ports;
 
     if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
-      GstRTSPAddress *addr = NULL;
+      /* FIXME: the address has been successfully allocated, however, in
+       * the use_client_settings case we need to verify that the allocated
+       * address is the one requested by the client and if this address is
+       * an allowed destination. Verifying this via the address pool in not
+       * the proper way as the address pool should only be used for choosing
+       * the server-selected address/port pairs. */
 
-      if (use_client_settings) {
-        /* the address has been successfully allocated, let's check if it's
-         * the one requested by the client */
-        addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
-            ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+      if (!use_client_settings) {
+        GstRTSPAddress *addr = NULL;
 
-        if (addr == NULL)
-          goto no_address;
-      } else {
         g_free (ct->destination);
         addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
         if (addr == NULL)
@@ -2009,9 +2008,8 @@ default_configure_client_transport (GstRTSPClient * client,
         ct->port.min = addr->port;
         ct->port.max = addr->port + addr->n_ports - 1;
         ct->ttl = addr->ttl;
+        gst_rtsp_address_free (addr);
       }
-
-      gst_rtsp_address_free (addr);
     } else {
       GstRTSPUrl *url;
 

diff --git a/tests/check/gst/client.c b/tests/check/gst/client.c
index 4adf3d1..ab95509 100644 (file)
--- a/tests/check/gst/client.c
+++ b/tests/check/gst/client.c
@@ -525,7 +525,6 @@ test_setup_response_200_multicast (GstRTSPClient * client,
   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);
   session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
   g_strfreev (session_hdr_params);
 
@@ -727,117 +726,6 @@ GST_START_TEST (test_client_multicast_ignore_transport_specific)
 
 GST_END_TEST;
 
-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;
-}
-
-GST_START_TEST (test_client_multicast_invalid_transport_specific)
-{
-  GstRTSPClient *client;
-  GstRTSPMessage request = { 0, };
-  gchar *str;
-  GstRTSPSessionPool *session_pool;
-  GstRTSPContext ctx = { NULL };
-
-  client = setup_multicast_client ();
-
-  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 a valid URI and multicast, but an invalid ip */
-  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=233.252.0.2;ttl=1;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);
-
-
-  /* simple SETUP with a valid URI and multicast, but an invalid prt */
-  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=1;port=6000-6001;");
-
-  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);
-
-
-  /* simple SETUP with a valid URI and multicast, but an invalid ttl */
-  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=2;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;
-
 GST_START_TEST (test_client_multicast_transport_specific)
 {
   GstRTSPClient *client;
@@ -859,7 +747,7 @@ 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\"";
 
-  /* simple SETUP with a valid URI and multicast, but an invalid ip */
+  /* simple SETUP with a valid URI */
   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);
@@ -1031,6 +919,156 @@ GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
 
 GST_END_TEST;
 
+static void
+mcast_transport_specific_two_clients (gboolean shared)
+{
+  GstRTSPClient *client, *client2;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+  GstRTSPSessionPool *session_pool;
+  GstRTSPContext ctx = { NULL };
+  GstRTSPContext ctx2 = { NULL };
+  GstRTSPMountPoints *mount_points;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAddressPool *address_pool;
+  GstRTSPThreadPool *thread_pool;
+  gchar *session_id1;
+
+  mount_points = gst_rtsp_mount_points_new ();
+  factory = gst_rtsp_media_factory_new ();
+  if (shared)
+    gst_rtsp_media_factory_set_shared (factory, TRUE);
+  gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5);
+  gst_rtsp_media_factory_set_launch (factory,
+      "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
+  address_pool = gst_rtsp_address_pool_new ();
+  fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+          "233.252.0.1", "233.252.0.1", 5000, 5001, 1));
+  gst_rtsp_media_factory_set_address_pool (factory, address_pool);
+  gst_rtsp_media_factory_add_role (factory, "user",
+      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+      "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+  session_pool = gst_rtsp_session_pool_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);
+
+  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);
+
+  expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+      "ttl=1;port=5000-5001;mode=\"PLAY\"";
+
+  /* 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_client_set_send_func (client, test_setup_response_200_multicast,
+      NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+  expected_transport = NULL;
+
+  /* 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);
+  gst_rtsp_context_pop_current (&ctx);
+  session_id1 = session_id;
+
+  /* second multicast client with transport specific request */
+  cseq = 0;
+  client2 = gst_rtsp_client_new ();
+  gst_rtsp_client_set_session_pool (client2, session_pool);
+  gst_rtsp_client_set_mount_points (client2, mount_points);
+  gst_rtsp_client_set_thread_pool (client2, thread_pool);
+
+  ctx2.client = client2;
+  ctx2.auth = gst_rtsp_auth_new ();
+  ctx2.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 (&ctx2);
+
+  expected_transport = "RTP/AVP;multicast;destination=233.252.0.2;"
+      "ttl=1;port=5002-5003;mode=\"PLAY\"";
+
+  /* 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_client_set_send_func (client2, test_setup_response_200_multicast,
+      NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client2,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+  expected_transport = NULL;
+
+  /* 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 (client2, test_response_200, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client2,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  send_teardown (client2);
+  gst_rtsp_context_pop_current (&ctx2);
+
+  gst_rtsp_context_push_current (&ctx);
+  session_id = session_id1;
+  send_teardown (client);
+  gst_rtsp_context_pop_current (&ctx);
+
+  teardown_client (client);
+  teardown_client (client2);
+  g_object_unref (ctx.auth);
+  g_object_unref (ctx2.auth);
+  gst_rtsp_token_unref (ctx.token);
+  gst_rtsp_token_unref (ctx2.token);
+  g_object_unref (mount_points);
+  g_object_unref (session_pool);
+  g_object_unref (address_pool);
+  g_object_unref (thread_pool);
+}
+
+/* test if two multicast clients can choose different transport settings */
+GST_START_TEST
+    (test_client_multicast_transport_specific_two_clients_shared_media) {
+  mcast_transport_specific_two_clients (TRUE);
+}
+
+GST_END_TEST;
+
 static Suite *
 rtspclient_suite (void)
 {
@@ -1046,12 +1084,13 @@ rtspclient_suite (void)
   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_multicast_invalid_transport_specific);
   tcase_add_test (tc, test_client_multicast_transport_specific);
   tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag);
   tcase_add_test (tc, test_client_sdp_with_bitrate_tag);
   tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags);
   tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
+  tcase_add_test (tc,
+      test_client_multicast_transport_specific_two_clients_shared_media);
 
   return s;
 }