2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: A client connection state
24 * @see_also: #GstRTSPServer, #GstRTSPThreadPool
26 * The client object handles the connection with a client for as long as a TCP
29 * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
30 * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
31 * #GstRTSPAuth and #GstRTSPThreadPool from the server.
33 * The client connection should be configured with the #GstRTSPConnection using
34 * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
35 * using gst_rtsp_client_attach(). From then on the client will handle requests
38 * Use gst_rtsp_client_session_filter() to iterate or modify all the
39 * #GstRTSPSession objects managed by the client object.
41 * Last reviewed on 2013-07-11 (1.0.0)
47 #include <gst/sdp/gstmikey.h>
49 #include "rtsp-client.h"
51 #include "rtsp-params.h"
53 #define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
54 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
57 * send_lock, lock, tunnels_lock
60 struct _GstRTSPClientPrivate
62 GMutex lock; /* protects everything else */
65 GstRTSPConnection *connection;
67 GMainContext *watch_context;
72 GstRTSPClientSendFunc send_func; /* protected by send_lock */
73 gpointer send_data; /* protected by send_lock */
74 GDestroyNotify send_notify; /* protected by send_lock */
76 GstRTSPSessionPool *session_pool;
77 gulong session_removed_id;
78 GstRTSPMountPoints *mount_points;
80 GstRTSPThreadPool *thread_pool;
82 /* used to cache the media in the last requested DESCRIBE so that
83 * we can pick it up in the next SETUP immediately */
87 GHashTable *transports;
89 guint sessions_cookie;
91 gboolean drop_backlog;
94 static GMutex tunnels_lock;
95 static GHashTable *tunnels; /* protected by tunnels_lock */
97 /* FIXME make this configurable. We don't want to do this yet because it will
98 * be superceeded by a cache object later */
99 #define WATCH_BACKLOG_SIZE 100
101 #define DEFAULT_SESSION_POOL NULL
102 #define DEFAULT_MOUNT_POINTS NULL
103 #define DEFAULT_DROP_BACKLOG TRUE
118 SIGNAL_OPTIONS_REQUEST,
119 SIGNAL_DESCRIBE_REQUEST,
120 SIGNAL_SETUP_REQUEST,
122 SIGNAL_PAUSE_REQUEST,
123 SIGNAL_TEARDOWN_REQUEST,
124 SIGNAL_SET_PARAMETER_REQUEST,
125 SIGNAL_GET_PARAMETER_REQUEST,
126 SIGNAL_HANDLE_RESPONSE,
128 SIGNAL_ANNOUNCE_REQUEST,
129 SIGNAL_RECORD_REQUEST,
133 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
134 #define GST_CAT_DEFAULT rtsp_client_debug
136 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
138 static void gst_rtsp_client_get_property (GObject * object, guint propid,
139 GValue * value, GParamSpec * pspec);
140 static void gst_rtsp_client_set_property (GObject * object, guint propid,
141 const GValue * value, GParamSpec * pspec);
142 static void gst_rtsp_client_finalize (GObject * obj);
144 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
145 static gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
146 GstRTSPMedia * media, GstSDPMessage * sdp);
147 static gboolean default_configure_client_media (GstRTSPClient * client,
148 GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
149 static gboolean default_configure_client_transport (GstRTSPClient * client,
150 GstRTSPContext * ctx, GstRTSPTransport * ct);
151 static GstRTSPResult default_params_set (GstRTSPClient * client,
152 GstRTSPContext * ctx);
153 static GstRTSPResult default_params_get (GstRTSPClient * client,
154 GstRTSPContext * ctx);
155 static gchar *default_make_path_from_uri (GstRTSPClient * client,
156 const GstRTSPUrl * uri);
157 static void client_session_removed (GstRTSPSessionPool * pool,
158 GstRTSPSession * session, GstRTSPClient * client);
160 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
163 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
165 GObjectClass *gobject_class;
167 g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
169 gobject_class = G_OBJECT_CLASS (klass);
171 gobject_class->get_property = gst_rtsp_client_get_property;
172 gobject_class->set_property = gst_rtsp_client_set_property;
173 gobject_class->finalize = gst_rtsp_client_finalize;
175 klass->create_sdp = create_sdp;
176 klass->handle_sdp = handle_sdp;
177 klass->configure_client_media = default_configure_client_media;
178 klass->configure_client_transport = default_configure_client_transport;
179 klass->params_set = default_params_set;
180 klass->params_get = default_params_get;
181 klass->make_path_from_uri = default_make_path_from_uri;
183 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
184 g_param_spec_object ("session-pool", "Session Pool",
185 "The session pool to use for client session",
186 GST_TYPE_RTSP_SESSION_POOL,
187 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
189 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
190 g_param_spec_object ("mount-points", "Mount Points",
191 "The mount points to use for client session",
192 GST_TYPE_RTSP_MOUNT_POINTS,
193 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
195 g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
196 g_param_spec_boolean ("drop-backlog", "Drop Backlog",
197 "Drop data when the backlog queue is full",
198 DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
200 gst_rtsp_client_signals[SIGNAL_CLOSED] =
201 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
202 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
203 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
205 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
206 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
207 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
208 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
210 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
211 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
212 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
213 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
214 GST_TYPE_RTSP_CONTEXT);
216 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
217 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
218 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
219 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
220 GST_TYPE_RTSP_CONTEXT);
222 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
223 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
224 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
225 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
226 GST_TYPE_RTSP_CONTEXT);
228 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
229 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
230 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
231 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
232 GST_TYPE_RTSP_CONTEXT);
234 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
235 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
236 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
237 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
238 GST_TYPE_RTSP_CONTEXT);
240 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
241 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
242 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
243 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
244 GST_TYPE_RTSP_CONTEXT);
246 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
247 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
248 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
249 set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
250 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
252 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
253 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
254 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
255 get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
256 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
258 gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
259 g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
260 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
261 handle_response), NULL, NULL, g_cclosure_marshal_generic,
262 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
265 * GstRTSPClient::send-message:
266 * @client: The RTSP client
267 * @session: (type GstRtspServer.RTSPSession): The session
268 * @message: (type GstRtsp.RTSPMessage): The message
270 gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
271 g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
272 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
273 send_message), NULL, NULL, g_cclosure_marshal_generic,
274 G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
276 gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST] =
277 g_signal_new ("announce-request", G_TYPE_FROM_CLASS (klass),
278 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, announce_request),
279 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
280 GST_TYPE_RTSP_CONTEXT);
282 gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST] =
283 g_signal_new ("record-request", G_TYPE_FROM_CLASS (klass),
284 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, record_request),
285 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
286 GST_TYPE_RTSP_CONTEXT);
289 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
290 g_mutex_init (&tunnels_lock);
292 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
296 gst_rtsp_client_init (GstRTSPClient * client)
298 GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
302 g_mutex_init (&priv->lock);
303 g_mutex_init (&priv->send_lock);
304 g_mutex_init (&priv->watch_lock);
306 priv->drop_backlog = DEFAULT_DROP_BACKLOG;
308 g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
312 static GstRTSPFilterResult
313 filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
316 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
318 return GST_RTSP_FILTER_REMOVE;
322 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
324 GstRTSPClientPrivate *priv = client->priv;
326 g_mutex_lock (&priv->lock);
327 /* check if we already know about this session */
328 if (g_list_find (priv->sessions, session) == NULL) {
329 GST_INFO ("watching session %p", session);
331 priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
332 priv->sessions_cookie++;
334 /* connect removed session handler, it will be disconnected when the last
335 * session gets removed */
336 if (priv->session_removed_id == 0)
337 priv->session_removed_id = g_signal_connect_data (priv->session_pool,
338 "session-removed", G_CALLBACK (client_session_removed),
339 g_object_ref (client), (GClosureNotify) g_object_unref, 0);
341 g_mutex_unlock (&priv->lock);
346 /* should be called with lock */
348 client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
351 GstRTSPClientPrivate *priv = client->priv;
353 GST_INFO ("client %p: unwatch session %p", client, session);
356 link = g_list_find (priv->sessions, session);
361 priv->sessions = g_list_delete_link (priv->sessions, link);
362 priv->sessions_cookie++;
364 /* if this was the last session, disconnect the handler.
365 * This will also drop the extra client ref */
366 if (!priv->sessions) {
367 g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
368 priv->session_removed_id = 0;
371 /* remove the session */
372 g_object_unref (session);
375 static GstRTSPFilterResult
376 cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
379 /* unlink all media managed in this session. This needs to happen
380 * without the client lock, so we really want to do it here. */
381 gst_rtsp_session_filter (sess, filter_session_media, client);
383 return GST_RTSP_FILTER_REMOVE;
387 clean_cached_media (GstRTSPClient * client, gboolean unprepare)
389 GstRTSPClientPrivate *priv = client->priv;
397 gst_rtsp_media_unprepare (priv->media);
398 g_object_unref (priv->media);
403 /* A client is finalized when the connection is broken */
405 gst_rtsp_client_finalize (GObject * obj)
407 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
408 GstRTSPClientPrivate *priv = client->priv;
410 GST_INFO ("finalize client %p", client);
413 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
414 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
417 g_source_destroy ((GSource *) priv->watch);
419 if (priv->watch_context)
420 g_main_context_unref (priv->watch_context);
422 /* all sessions should have been removed by now. We keep a ref to
423 * the client object for the session removed handler. The ref is
424 * dropped when the last session is removed from the list. */
425 g_assert (priv->sessions == NULL);
426 g_assert (priv->session_removed_id == 0);
428 g_hash_table_unref (priv->transports);
430 if (priv->connection)
431 gst_rtsp_connection_free (priv->connection);
432 if (priv->session_pool) {
433 g_object_unref (priv->session_pool);
435 if (priv->mount_points)
436 g_object_unref (priv->mount_points);
438 g_object_unref (priv->auth);
439 if (priv->thread_pool)
440 g_object_unref (priv->thread_pool);
442 clean_cached_media (client, TRUE);
444 g_free (priv->server_ip);
445 g_mutex_clear (&priv->lock);
446 g_mutex_clear (&priv->send_lock);
447 g_mutex_clear (&priv->watch_lock);
449 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
453 gst_rtsp_client_get_property (GObject * object, guint propid,
454 GValue * value, GParamSpec * pspec)
456 GstRTSPClient *client = GST_RTSP_CLIENT (object);
457 GstRTSPClientPrivate *priv = client->priv;
460 case PROP_SESSION_POOL:
461 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
463 case PROP_MOUNT_POINTS:
464 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
466 case PROP_DROP_BACKLOG:
467 g_value_set_boolean (value, priv->drop_backlog);
470 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
475 gst_rtsp_client_set_property (GObject * object, guint propid,
476 const GValue * value, GParamSpec * pspec)
478 GstRTSPClient *client = GST_RTSP_CLIENT (object);
479 GstRTSPClientPrivate *priv = client->priv;
482 case PROP_SESSION_POOL:
483 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
485 case PROP_MOUNT_POINTS:
486 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
488 case PROP_DROP_BACKLOG:
489 g_mutex_lock (&priv->lock);
490 priv->drop_backlog = g_value_get_boolean (value);
491 g_mutex_unlock (&priv->lock);
494 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
499 * gst_rtsp_client_new:
501 * Create a new #GstRTSPClient instance.
503 * Returns: (transfer full): a new #GstRTSPClient
506 gst_rtsp_client_new (void)
508 GstRTSPClient *result;
510 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
516 send_message (GstRTSPClient * client, GstRTSPContext * ctx,
517 GstRTSPMessage * message, gboolean close)
519 GstRTSPClientPrivate *priv = client->priv;
521 gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
522 "GStreamer RTSP server");
524 /* remove any previous header */
525 gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
527 /* add the new session header for new session ids */
529 gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
530 gst_rtsp_session_get_header (ctx->session));
533 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
534 gst_rtsp_message_dump (message);
538 gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
540 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
543 g_mutex_lock (&priv->send_lock);
545 priv->send_func (client, message, close, priv->send_data);
546 g_mutex_unlock (&priv->send_lock);
548 gst_rtsp_message_unset (message);
552 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
553 GstRTSPContext * ctx)
555 gst_rtsp_message_init_response (ctx->response, code,
556 gst_rtsp_status_as_text (code), ctx->request);
560 send_message (client, ctx, ctx->response, FALSE);
564 send_option_not_supported_response (GstRTSPClient * client,
565 GstRTSPContext * ctx, const gchar * unsupported_options)
567 GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
569 gst_rtsp_message_init_response (ctx->response, code,
570 gst_rtsp_status_as_text (code), ctx->request);
572 if (unsupported_options != NULL) {
573 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
574 unsupported_options);
579 send_message (client, ctx, ctx->response, FALSE);
583 paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
585 if (path1 == NULL || path2 == NULL)
588 if (strlen (path1) != len2)
591 if (strncmp (path1, path2, len2))
597 /* this function is called to initially find the media for the DESCRIBE request
598 * but is cached for when the same client (without breaking the connection) is
599 * doing a setup for the exact same url. */
600 static GstRTSPMedia *
601 find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
604 GstRTSPClientPrivate *priv = client->priv;
605 GstRTSPMediaFactory *factory;
609 /* find the longest matching factory for the uri first */
610 if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
614 ctx->factory = factory;
616 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
617 goto no_factory_access;
619 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
625 path_len = strlen (path);
627 if (!paths_are_equal (priv->path, path, path_len)) {
628 /* remove any previously cached values before we try to construct a new
630 clean_cached_media (client, TRUE);
632 /* prepare the media and add it to the pipeline */
633 if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
638 if (!(gst_rtsp_media_get_transport_mode (media) &
639 GST_RTSP_TRANSPORT_MODE_RECORD)) {
640 GstRTSPThread *thread;
642 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
643 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
647 /* prepare the media */
648 if (!gst_rtsp_media_prepare (media, thread))
652 /* now keep track of the uri and the media */
653 priv->path = g_strndup (path, path_len);
656 /* we have seen this path before, used cached media */
659 GST_INFO ("reusing cached media %p for path %s", media, priv->path);
662 g_object_unref (factory);
666 g_object_ref (media);
673 GST_ERROR ("client %p: no factory for path %s", client, path);
674 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
679 GST_ERROR ("client %p: not authorized to see factory path %s", client,
681 /* error reply is already sent */
686 GST_ERROR ("client %p: not authorized for factory path %s", client, path);
687 /* error reply is already sent */
692 GST_ERROR ("client %p: can't create media", client);
693 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
694 g_object_unref (factory);
700 GST_ERROR ("client %p: can't create thread", client);
701 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
702 g_object_unref (media);
704 g_object_unref (factory);
710 GST_ERROR ("client %p: can't prepare media", client);
711 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
712 g_object_unref (media);
714 g_object_unref (factory);
721 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
723 GstRTSPClientPrivate *priv = client->priv;
724 GstRTSPMessage message = { 0 };
725 GstRTSPResult res = GST_RTSP_OK;
730 gst_rtsp_message_init_data (&message, channel);
732 /* FIXME, need some sort of iovec RTSPMessage here */
733 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
736 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
738 g_mutex_lock (&priv->send_lock);
740 res = priv->send_func (client, &message, FALSE, priv->send_data);
741 g_mutex_unlock (&priv->send_lock);
743 gst_rtsp_message_steal_body (&message, &data, &usize);
744 gst_buffer_unmap (buffer, &map_info);
746 gst_rtsp_message_unset (&message);
748 return res == GST_RTSP_OK;
752 * gst_rtsp_client_close:
753 * @client: a #GstRTSPClient
755 * Close the connection of @client and remove all media it was managing.
760 gst_rtsp_client_close (GstRTSPClient * client)
762 GstRTSPClientPrivate *priv = client->priv;
763 const gchar *tunnelid;
765 GST_DEBUG ("client %p: closing connection", client);
767 if (priv->connection) {
768 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
769 g_mutex_lock (&tunnels_lock);
770 /* remove from tunnelids */
771 g_hash_table_remove (tunnels, tunnelid);
772 g_mutex_unlock (&tunnels_lock);
774 gst_rtsp_connection_close (priv->connection);
777 /* connection is now closed, destroy the watch which will also cause the
778 * closed signal to be emitted */
780 GST_DEBUG ("client %p: destroying watch", client);
781 g_source_destroy ((GSource *) priv->watch);
783 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
784 g_main_context_unref (priv->watch_context);
785 priv->watch_context = NULL;
790 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
795 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
797 path = g_strdup (uri->abspath);
803 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
805 GstRTSPClientPrivate *priv = client->priv;
806 GstRTSPClientClass *klass;
807 GstRTSPSession *session;
808 GstRTSPSessionMedia *sessmedia;
809 GstRTSPStatusCode code;
812 gboolean keep_session;
817 session = ctx->session;
822 klass = GST_RTSP_CLIENT_GET_CLASS (client);
823 path = klass->make_path_from_uri (client, ctx->uri);
825 /* get a handle to the configuration of the media in the session */
826 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
830 /* only aggregate control for now.. */
831 if (path[matched] != '\0')
836 ctx->sessmedia = sessmedia;
838 /* we emit the signal before closing the connection */
839 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
842 /* make sure we unblock the backlog and don't accept new messages
844 if (priv->watch != NULL)
845 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
847 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
849 /* allow messages again so that we can send the reply */
850 if (priv->watch != NULL)
851 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
853 /* unmanage the media in the session, returns false if all media session
855 keep_session = gst_rtsp_session_release_media (session, sessmedia);
857 /* construct the response now */
858 code = GST_RTSP_STS_OK;
859 gst_rtsp_message_init_response (ctx->response, code,
860 gst_rtsp_status_as_text (code), ctx->request);
862 send_message (client, ctx, ctx->response, TRUE);
865 /* remove the session */
866 gst_rtsp_session_pool_remove (priv->session_pool, session);
874 GST_ERROR ("client %p: no session", client);
875 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
880 GST_ERROR ("client %p: no uri supplied", client);
881 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
886 GST_ERROR ("client %p: no media for uri", client);
887 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
893 GST_ERROR ("client %p: no aggregate path %s", client, path);
894 send_generic_response (client,
895 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
902 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
906 res = gst_rtsp_params_set (client, ctx);
912 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
916 res = gst_rtsp_params_get (client, ctx);
922 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
928 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
929 if (res != GST_RTSP_OK)
933 /* no body, keep-alive request */
934 send_generic_response (client, GST_RTSP_STS_OK, ctx);
936 /* there is a body, handle the params */
937 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
938 if (res != GST_RTSP_OK)
941 send_message (client, ctx, ctx->response, FALSE);
944 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
952 GST_ERROR ("client %p: bad request", client);
953 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
959 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
965 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
966 if (res != GST_RTSP_OK)
970 /* no body, keep-alive request */
971 send_generic_response (client, GST_RTSP_STS_OK, ctx);
973 /* there is a body, handle the params */
974 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
975 if (res != GST_RTSP_OK)
978 send_message (client, ctx, ctx->response, FALSE);
981 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
989 GST_ERROR ("client %p: bad request", client);
990 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
996 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
998 GstRTSPSession *session;
999 GstRTSPClientClass *klass;
1000 GstRTSPSessionMedia *sessmedia;
1001 GstRTSPStatusCode code;
1002 GstRTSPState rtspstate;
1006 if (!(session = ctx->session))
1012 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1013 path = klass->make_path_from_uri (client, ctx->uri);
1015 /* get a handle to the configuration of the media in the session */
1016 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1020 if (path[matched] != '\0')
1025 ctx->sessmedia = sessmedia;
1027 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1028 /* the session state must be playing or recording */
1029 if (rtspstate != GST_RTSP_STATE_PLAYING &&
1030 rtspstate != GST_RTSP_STATE_RECORDING)
1033 /* then pause sending */
1034 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1036 /* construct the response now */
1037 code = GST_RTSP_STS_OK;
1038 gst_rtsp_message_init_response (ctx->response, code,
1039 gst_rtsp_status_as_text (code), ctx->request);
1041 send_message (client, ctx, ctx->response, FALSE);
1043 /* the state is now READY */
1044 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1046 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1053 GST_ERROR ("client %p: no seesion", client);
1054 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1059 GST_ERROR ("client %p: no uri supplied", client);
1060 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1065 GST_ERROR ("client %p: no media for uri", client);
1066 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1072 GST_ERROR ("client %p: no aggregate path %s", client, path);
1073 send_generic_response (client,
1074 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1080 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1081 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1087 /* convert @url and @path to a URL used as a content base for the factory
1088 * located at @path */
1090 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1096 /* check for trailing '/' and append one */
1097 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1102 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1104 result = gst_rtsp_url_get_request_uri (&tmp);
1105 g_free (tmp.abspath);
1111 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1113 GstRTSPSession *session;
1114 GstRTSPClientClass *klass;
1115 GstRTSPSessionMedia *sessmedia;
1116 GstRTSPMedia *media;
1117 GstRTSPStatusCode code;
1120 GstRTSPTimeRange *range;
1122 GstRTSPState rtspstate;
1123 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1124 gchar *path, *rtpinfo;
1127 if (!(session = ctx->session))
1130 if (!(uri = ctx->uri))
1133 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1134 path = klass->make_path_from_uri (client, uri);
1136 /* get a handle to the configuration of the media in the session */
1137 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1141 if (path[matched] != '\0')
1146 ctx->sessmedia = sessmedia;
1147 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1149 if (!(gst_rtsp_media_get_transport_mode (media) &
1150 GST_RTSP_TRANSPORT_MODE_PLAY))
1151 goto unsupported_mode;
1153 /* the session state must be playing or ready */
1154 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1155 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1158 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1159 if (!gst_rtsp_media_unsuspend (media))
1160 goto unsuspend_failed;
1162 /* parse the range header if we have one */
1163 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1164 if (res == GST_RTSP_OK) {
1165 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1166 /* we have a range, seek to the position */
1168 gst_rtsp_media_seek (media, range);
1169 gst_rtsp_range_free (range);
1173 /* grab RTPInfo from the media now */
1174 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1176 /* construct the response now */
1177 code = GST_RTSP_STS_OK;
1178 gst_rtsp_message_init_response (ctx->response, code,
1179 gst_rtsp_status_as_text (code), ctx->request);
1181 /* add the RTP-Info header */
1183 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1187 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1189 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1191 send_message (client, ctx, ctx->response, FALSE);
1193 /* start playing after sending the response */
1194 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1196 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1198 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1205 GST_ERROR ("client %p: no session", client);
1206 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1211 GST_ERROR ("client %p: no uri supplied", client);
1212 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1217 GST_ERROR ("client %p: media not found", client);
1218 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1223 GST_ERROR ("client %p: no aggregate path %s", client, path);
1224 send_generic_response (client,
1225 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1231 GST_ERROR ("client %p: not PLAYING or READY", client);
1232 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1238 GST_ERROR ("client %p: unsuspend failed", client);
1239 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1244 GST_ERROR ("client %p: media does not support PLAY", client);
1245 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
1251 do_keepalive (GstRTSPSession * session)
1253 GST_INFO ("keep session %p alive", session);
1254 gst_rtsp_session_touch (session);
1257 /* parse @transport and return a valid transport in @tr. only transports
1258 * supported by @stream are returned. Returns FALSE if no valid transport
1261 parse_transport (const char *transport, GstRTSPStream * stream,
1262 GstRTSPTransport * tr)
1269 gst_rtsp_transport_init (tr);
1271 GST_DEBUG ("parsing transports %s", transport);
1273 transports = g_strsplit (transport, ",", 0);
1275 /* loop through the transports, try to parse */
1276 for (i = 0; transports[i]; i++) {
1277 res = gst_rtsp_transport_parse (transports[i], tr);
1278 if (res != GST_RTSP_OK) {
1279 /* no valid transport, search some more */
1280 GST_WARNING ("could not parse transport %s", transports[i]);
1284 /* we have a transport, see if it's supported */
1285 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1286 GST_WARNING ("unsupported transport %s", transports[i]);
1290 /* we have a valid transport */
1291 GST_INFO ("found valid transport %s", transports[i]);
1296 gst_rtsp_transport_init (tr);
1298 g_strfreev (transports);
1304 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1305 GstRTSPStream * stream, GstRTSPContext * ctx)
1307 GstRTSPMessage *request = ctx->request;
1308 gchar *blocksize_str;
1310 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1311 &blocksize_str, 0) == GST_RTSP_OK) {
1315 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1316 if (end == blocksize_str)
1319 /* we don't want to change the mtu when this media
1320 * can be shared because it impacts other clients */
1321 if (gst_rtsp_media_is_shared (media))
1324 if (blocksize > G_MAXUINT)
1325 blocksize = G_MAXUINT;
1327 gst_rtsp_stream_set_mtu (stream, blocksize);
1335 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1336 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1342 default_configure_client_transport (GstRTSPClient * client,
1343 GstRTSPContext * ctx, GstRTSPTransport * ct)
1345 GstRTSPClientPrivate *priv = client->priv;
1347 /* we have a valid transport now, set the destination of the client. */
1348 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1349 gboolean use_client_settings;
1351 use_client_settings =
1352 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1354 if (ct->destination && use_client_settings) {
1355 GstRTSPAddress *addr;
1357 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1358 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1363 gst_rtsp_address_free (addr);
1365 GstRTSPAddress *addr;
1366 GSocketFamily family;
1368 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1370 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1374 g_free (ct->destination);
1375 ct->destination = g_strdup (addr->address);
1376 ct->port.min = addr->port;
1377 ct->port.max = addr->port + addr->n_ports - 1;
1378 ct->ttl = addr->ttl;
1380 gst_rtsp_address_free (addr);
1385 url = gst_rtsp_connection_get_url (priv->connection);
1386 g_free (ct->destination);
1387 ct->destination = g_strdup (url->host);
1389 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1391 GSocketAddress *addr;
1393 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1394 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1395 /* our read port is the sender port of client */
1396 ct->client_port.min =
1397 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1398 g_object_unref (addr);
1400 if ((addr = g_socket_get_local_address (sock, NULL))) {
1401 ct->server_port.max =
1402 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1403 g_object_unref (addr);
1405 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1406 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1407 /* our write port is the receiver port of client */
1408 ct->client_port.max =
1409 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1410 g_object_unref (addr);
1412 if ((addr = g_socket_get_local_address (sock, NULL))) {
1413 ct->server_port.min =
1414 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1415 g_object_unref (addr);
1417 /* check if the client selected channels for TCP */
1418 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1419 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1429 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1434 static GstRTSPTransport *
1435 make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
1436 GstRTSPContext * ctx, GstRTSPTransport * ct)
1438 GstRTSPTransport *st;
1440 GSocketFamily family;
1442 /* prepare the server transport */
1443 gst_rtsp_transport_new (&st);
1445 st->trans = ct->trans;
1446 st->profile = ct->profile;
1447 st->lower_transport = ct->lower_transport;
1448 st->mode_play = ct->mode_play;
1449 st->mode_record = ct->mode_record;
1451 addr = g_inet_address_new_from_string (ct->destination);
1454 GST_ERROR ("failed to get inet addr from client destination");
1455 family = G_SOCKET_FAMILY_IPV4;
1457 family = g_inet_address_get_family (addr);
1458 g_object_unref (addr);
1462 switch (st->lower_transport) {
1463 case GST_RTSP_LOWER_TRANS_UDP:
1464 st->client_port = ct->client_port;
1465 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1467 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1468 st->port = ct->port;
1469 st->destination = g_strdup (ct->destination);
1472 case GST_RTSP_LOWER_TRANS_TCP:
1473 st->interleaved = ct->interleaved;
1474 st->client_port = ct->client_port;
1475 st->server_port = ct->server_port;
1480 if ((gst_rtsp_media_get_transport_mode (media) &
1481 GST_RTSP_TRANSPORT_MODE_PLAY))
1482 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1487 #define AES_128_KEY_LEN 16
1488 #define AES_256_KEY_LEN 32
1490 #define HMAC_32_KEY_LEN 4
1491 #define HMAC_80_KEY_LEN 10
1494 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1496 const gchar *srtp_cipher;
1497 const gchar *srtp_auth;
1498 const GstMIKEYPayload *sp;
1501 /* loop over Security policy until we find one containing policy */
1503 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1506 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1510 /* the default ciphers */
1511 srtp_cipher = "aes-128-icm";
1512 srtp_auth = "hmac-sha1-80";
1514 /* now override the defaults with what is in the Security Policy */
1518 /* collect all the params and go over them */
1519 len = gst_mikey_payload_sp_get_n_params (sp);
1520 for (i = 0; i < len; i++) {
1521 const GstMIKEYPayloadSPParam *param =
1522 gst_mikey_payload_sp_get_param (sp, i);
1524 switch (param->type) {
1525 case GST_MIKEY_SP_SRTP_ENC_ALG:
1526 switch (param->val[0]) {
1528 srtp_cipher = "null";
1532 srtp_cipher = "aes-128-icm";
1538 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1539 switch (param->val[0]) {
1540 case AES_128_KEY_LEN:
1541 srtp_cipher = "aes-128-icm";
1543 case AES_256_KEY_LEN:
1544 srtp_cipher = "aes-256-icm";
1550 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1551 switch (param->val[0]) {
1557 srtp_auth = "hmac-sha1-80";
1563 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1564 switch (param->val[0]) {
1565 case HMAC_32_KEY_LEN:
1566 srtp_auth = "hmac-sha1-32";
1568 case HMAC_80_KEY_LEN:
1569 srtp_auth = "hmac-sha1-80";
1575 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1577 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1584 /* now configure the SRTP parameters */
1585 gst_caps_set_simple (caps,
1586 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1587 "srtp-auth", G_TYPE_STRING, srtp_auth,
1588 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1589 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1595 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1596 guint8 * data, gsize size)
1598 GstMIKEYMessage *msg;
1600 GstCaps *caps = NULL;
1601 GstMIKEYPayloadKEMAC *kemac;
1602 const GstMIKEYPayloadKeyData *pkd;
1605 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1606 * set of Crypto Sessions protected with the same master key.
1607 * In the context of SRTP, an RTP and its RTCP stream is part of a
1609 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1612 /* we can only handle SRTP crypto sessions for now */
1613 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1614 goto invalid_map_type;
1616 /* get the number of crypto sessions. This maps SSRC to its
1617 * security parameters */
1618 n_cs = gst_mikey_message_get_n_cs (msg);
1620 goto no_crypto_sessions;
1622 /* we also need keys */
1623 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1624 (msg, GST_MIKEY_PT_KEMAC, 0)))
1627 /* we don't support encrypted keys */
1628 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1629 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1630 goto unsupported_encryption;
1632 /* get Key data sub-payload */
1633 pkd = (const GstMIKEYPayloadKeyData *)
1634 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1637 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1640 /* go over all crypto sessions and create the security policy for each
1642 for (i = 0; i < n_cs; i++) {
1643 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1645 caps = gst_caps_new_simple ("application/x-srtp",
1646 "ssrc", G_TYPE_UINT, map->ssrc,
1647 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1648 mikey_apply_policy (caps, msg, map->policy);
1650 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1651 gst_caps_unref (caps);
1653 gst_mikey_message_unref (msg);
1654 gst_buffer_unref (key);
1661 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1666 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1667 goto cleanup_message;
1671 GST_DEBUG_OBJECT (client, "no crypto sessions");
1672 goto cleanup_message;
1676 GST_DEBUG_OBJECT (client, "no keys found");
1677 goto cleanup_message;
1679 unsupported_encryption:
1681 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1682 goto cleanup_message;
1686 gst_mikey_message_unref (msg);
1691 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1694 strip_chars (gchar * str)
1701 if (!IS_STRIP_CHAR (str[len]))
1705 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1706 memmove (str, s, len + 1);
1709 /* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1710 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1713 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1718 specs = g_strsplit (keymgmt, ",", 0);
1719 for (i = 0; specs[i]; i++) {
1722 split = g_strsplit (specs[i], ";", 0);
1723 for (j = 0; split[j]; j++) {
1724 g_strstrip (split[j]);
1725 if (g_str_has_prefix (split[j], "prot=")) {
1726 g_strstrip (split[j] + 5);
1727 if (!g_str_equal (split[j] + 5, "mikey"))
1729 GST_DEBUG ("found mikey");
1730 } else if (g_str_has_prefix (split[j], "uri=")) {
1731 strip_chars (split[j] + 4);
1732 GST_DEBUG ("found uri '%s'", split[j] + 4);
1733 } else if (g_str_has_prefix (split[j], "data=")) {
1736 strip_chars (split[j] + 5);
1737 GST_DEBUG ("found data '%s'", split[j] + 5);
1738 data = g_base64_decode_inplace (split[j] + 5, &size);
1739 handle_mikey_data (client, ctx, data, size);
1749 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1751 GstRTSPClientPrivate *priv = client->priv;
1754 gchar *transport, *keymgmt;
1755 GstRTSPTransport *ct, *st;
1756 GstRTSPStatusCode code;
1757 GstRTSPSession *session;
1758 GstRTSPStreamTransport *trans;
1760 GstRTSPSessionMedia *sessmedia;
1761 GstRTSPMedia *media;
1762 GstRTSPStream *stream;
1763 GstRTSPState rtspstate;
1764 GstRTSPClientClass *klass;
1765 gchar *path, *control = NULL;
1767 gboolean new_session = FALSE;
1773 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1774 path = klass->make_path_from_uri (client, uri);
1776 /* parse the transport */
1778 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1780 if (res != GST_RTSP_OK)
1783 /* we create the session after parsing stuff so that we don't make
1784 * a session for malformed requests */
1785 if (priv->session_pool == NULL)
1788 session = ctx->session;
1791 g_object_ref (session);
1792 /* get a handle to the configuration of the media in the session, this can
1793 * return NULL if this is a new url to manage in this session. */
1794 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1796 /* we need a new media configuration in this session */
1800 /* we have no session media, find one and manage it */
1801 if (sessmedia == NULL) {
1802 /* get a handle to the configuration of the media in the session */
1803 media = find_media (client, ctx, path, &matched);
1805 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1806 g_object_ref (media);
1808 goto media_not_found;
1810 /* no media, not found then */
1812 goto media_not_found_no_reply;
1814 if (path[matched] == '\0') {
1815 if (gst_rtsp_media_n_streams (media) == 1) {
1816 stream = gst_rtsp_media_get_stream (media, 0);
1818 goto control_not_found;
1821 /* path is what matched. */
1822 path[matched] = '\0';
1823 /* control is remainder */
1824 control = &path[matched + 1];
1826 /* find the stream now using the control part */
1827 stream = gst_rtsp_media_find_stream (media, control);
1831 goto stream_not_found;
1833 /* now we have a uri identifying a valid media and stream */
1834 ctx->stream = stream;
1837 if (session == NULL) {
1838 /* create a session if this fails we probably reached our session limit or
1840 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1841 goto service_unavailable;
1843 /* make sure this client is closed when the session is closed */
1844 client_watch_session (client, session);
1847 /* signal new session */
1848 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1851 ctx->session = session;
1854 if (!klass->configure_client_media (client, media, stream, ctx))
1855 goto configure_media_failed_no_reply;
1857 gst_rtsp_transport_new (&ct);
1859 /* parse and find a usable supported transport */
1860 if (!parse_transport (transport, stream, ct))
1861 goto unsupported_transports;
1864 && !(gst_rtsp_media_get_transport_mode (media) &
1865 GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
1866 && !(gst_rtsp_media_get_transport_mode (media) &
1867 GST_RTSP_TRANSPORT_MODE_RECORD)))
1868 goto unsupported_mode;
1870 /* parse the keymgmt */
1871 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1872 &keymgmt, 0) == GST_RTSP_OK) {
1873 if (!handle_keymgmt (client, ctx, keymgmt))
1877 if (sessmedia == NULL) {
1878 /* manage the media in our session now, if not done already */
1879 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1880 /* if we stil have no media, error */
1881 if (sessmedia == NULL)
1882 goto sessmedia_unavailable;
1884 /* don't cache media anymore */
1885 clean_cached_media (client, FALSE);
1887 g_object_unref (media);
1890 ctx->sessmedia = sessmedia;
1892 /* update the client transport */
1893 if (!klass->configure_client_transport (client, ctx, ct))
1894 goto unsupported_client_transport;
1896 /* set in the session media transport */
1897 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1901 /* configure the url used to set this transport, this we will use when
1902 * generating the response for the PLAY request */
1903 gst_rtsp_stream_transport_set_url (trans, uri);
1904 /* configure keepalive for this transport */
1905 gst_rtsp_stream_transport_set_keepalive (trans,
1906 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1908 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1909 /* our callbacks to send data on this TCP connection */
1910 gst_rtsp_stream_transport_set_callbacks (trans,
1911 (GstRTSPSendFunc) do_send_data,
1912 (GstRTSPSendFunc) do_send_data, client, NULL);
1914 g_hash_table_insert (priv->transports,
1915 GINT_TO_POINTER (ct->interleaved.min), trans);
1916 g_object_ref (trans);
1917 g_hash_table_insert (priv->transports,
1918 GINT_TO_POINTER (ct->interleaved.max), trans);
1919 g_object_ref (trans);
1922 /* create and serialize the server transport */
1923 st = make_server_transport (client, media, ctx, ct);
1924 trans_str = gst_rtsp_transport_as_text (st);
1925 gst_rtsp_transport_free (st);
1927 /* construct the response now */
1928 code = GST_RTSP_STS_OK;
1929 gst_rtsp_message_init_response (ctx->response, code,
1930 gst_rtsp_status_as_text (code), ctx->request);
1932 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1936 send_message (client, ctx, ctx->response, FALSE);
1938 /* update the state */
1939 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1940 switch (rtspstate) {
1941 case GST_RTSP_STATE_PLAYING:
1942 case GST_RTSP_STATE_RECORDING:
1943 case GST_RTSP_STATE_READY:
1944 /* no state change */
1947 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1950 g_object_unref (session);
1953 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1960 GST_ERROR ("client %p: no uri", client);
1961 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1966 GST_ERROR ("client %p: no transport", client);
1967 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1972 GST_ERROR ("client %p: no session pool configured", client);
1973 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1976 media_not_found_no_reply:
1978 GST_ERROR ("client %p: media '%s' not found", client, path);
1979 /* error reply is already sent */
1984 GST_ERROR ("client %p: media '%s' not found", client, path);
1985 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1990 GST_ERROR ("client %p: no control in path '%s'", client, path);
1991 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1992 g_object_unref (media);
1997 GST_ERROR ("client %p: stream '%s' not found", client,
1998 GST_STR_NULL (control));
1999 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2000 g_object_unref (media);
2003 service_unavailable:
2005 GST_ERROR ("client %p: can't create session", client);
2006 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2007 g_object_unref (media);
2010 sessmedia_unavailable:
2012 GST_ERROR ("client %p: can't create session media", client);
2013 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2014 g_object_unref (media);
2015 goto cleanup_session;
2017 configure_media_failed_no_reply:
2019 GST_ERROR ("client %p: configure_media failed", client);
2020 /* error reply is already sent */
2021 goto cleanup_session;
2023 unsupported_transports:
2025 GST_ERROR ("client %p: unsupported transports", client);
2026 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2027 goto cleanup_transport;
2029 unsupported_client_transport:
2031 GST_ERROR ("client %p: unsupported client transport", client);
2032 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2033 goto cleanup_transport;
2037 GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
2038 "mode play: %d, mode record: %d)", client,
2039 ! !(gst_rtsp_media_get_transport_mode (media) &
2040 GST_RTSP_TRANSPORT_MODE_PLAY),
2041 ! !(gst_rtsp_media_get_transport_mode (media) &
2042 GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
2043 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2044 goto cleanup_transport;
2048 GST_ERROR ("client %p: keymgmt error", client);
2049 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
2050 goto cleanup_transport;
2054 gst_rtsp_transport_free (ct);
2057 gst_rtsp_session_pool_remove (priv->session_pool, session);
2058 g_object_unref (session);
2065 static GstSDPMessage *
2066 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
2068 GstRTSPClientPrivate *priv = client->priv;
2072 guint64 session_id_tmp;
2073 gchar session_id[21];
2075 gst_sdp_message_new (&sdp);
2077 /* some standard things first */
2078 gst_sdp_message_set_version (sdp, "0");
2085 session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
2086 g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
2089 gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
2092 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2093 gst_sdp_message_set_information (sdp, "rtsp-server");
2094 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2095 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2096 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2097 gst_sdp_message_add_attribute (sdp, "control", "*");
2099 info.is_ipv6 = priv->is_ipv6;
2100 info.server_ip = priv->server_ip;
2102 /* create an SDP for the media object */
2103 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2111 GST_ERROR ("client %p: could not create SDP", client);
2112 gst_sdp_message_free (sdp);
2117 /* for the describe we must generate an SDP */
2119 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2121 GstRTSPClientPrivate *priv = client->priv;
2126 GstRTSPMedia *media;
2127 GstRTSPClientClass *klass;
2129 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2134 /* check what kind of format is accepted, we don't really do anything with it
2135 * and always return SDP for now. */
2140 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2142 if (res == GST_RTSP_ENOTIMPL)
2145 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2149 if (!priv->mount_points)
2150 goto no_mount_points;
2152 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2155 /* find the media object for the uri */
2156 if (!(media = find_media (client, ctx, path, NULL)))
2159 if (!(gst_rtsp_media_get_transport_mode (media) &
2160 GST_RTSP_TRANSPORT_MODE_PLAY))
2161 goto unsupported_mode;
2163 /* create an SDP for the media object on this client */
2164 if (!(sdp = klass->create_sdp (client, media)))
2167 /* we suspend after the describe */
2168 gst_rtsp_media_suspend (media);
2169 g_object_unref (media);
2171 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2172 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2174 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2177 /* content base for some clients that might screw up creating the setup uri */
2178 str = make_base_url (client, ctx->uri, path);
2181 GST_INFO ("adding content-base: %s", str);
2182 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2184 /* add SDP to the response body */
2185 str = gst_sdp_message_as_text (sdp);
2186 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2187 gst_sdp_message_free (sdp);
2189 send_message (client, ctx, ctx->response, FALSE);
2191 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2199 GST_ERROR ("client %p: no uri", client);
2200 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2205 GST_ERROR ("client %p: no mount points configured", client);
2206 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2211 GST_ERROR ("client %p: can't find path for url", client);
2212 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2217 GST_ERROR ("client %p: no media", client);
2219 /* error reply is already sent */
2224 GST_ERROR ("client %p: media does not support DESCRIBE", client);
2225 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2227 g_object_unref (media);
2232 GST_ERROR ("client %p: can't create SDP", client);
2233 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2235 g_object_unref (media);
2241 handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
2242 GstSDPMessage * sdp)
2244 GstRTSPClientPrivate *priv = client->priv;
2245 GstRTSPThread *thread;
2247 /* create an SDP for the media object */
2248 if (!gst_rtsp_media_handle_sdp (media, sdp))
2251 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
2252 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
2256 /* prepare the media */
2257 if (!gst_rtsp_media_prepare (media, thread))
2265 GST_ERROR ("client %p: could not handle SDP", client);
2270 GST_ERROR ("client %p: can't create thread", client);
2275 GST_ERROR ("client %p: can't prepare media", client);
2281 handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
2283 GstRTSPClientPrivate *priv = client->priv;
2284 GstRTSPClientClass *klass;
2287 GstRTSPMedia *media;
2288 gchar *path, *cont = NULL;
2292 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2297 if (!priv->mount_points)
2298 goto no_mount_points;
2300 /* check if reply is SDP */
2301 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
2303 /* could not be set but since the request returned OK, we assume it
2304 * was SDP, else check it. */
2306 if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
2307 goto wrong_content_type;
2310 /* get message body and parse as SDP */
2311 gst_rtsp_message_get_body (ctx->request, &data, &size);
2312 if (data == NULL || size == 0)
2315 GST_DEBUG ("client %p: parse SDP...", client);
2316 gst_sdp_message_new (&sdp);
2317 sres = gst_sdp_message_parse_buffer (data, size, sdp);
2318 if (sres != GST_SDP_OK)
2319 goto sdp_parse_failed;
2321 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2324 /* find the media object for the uri */
2325 if (!(media = find_media (client, ctx, path, NULL)))
2328 if (!(gst_rtsp_media_get_transport_mode (media) &
2329 GST_RTSP_TRANSPORT_MODE_RECORD))
2330 goto unsupported_mode;
2332 /* Tell client subclass about the media */
2333 if (!klass->handle_sdp (client, ctx, media, sdp))
2336 /* we suspend after the announce */
2337 gst_rtsp_media_suspend (media);
2338 g_object_unref (media);
2340 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2341 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2343 send_message (client, ctx, ctx->response, FALSE);
2345 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
2348 gst_sdp_message_free (sdp);
2354 GST_ERROR ("client %p: no uri", client);
2355 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2360 GST_ERROR ("client %p: no mount points configured", client);
2361 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2366 GST_ERROR ("client %p: can't find path for url", client);
2367 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2368 gst_sdp_message_free (sdp);
2373 GST_ERROR ("client %p: unknown content type", client);
2374 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2379 GST_ERROR ("client %p: can't find SDP message", client);
2380 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2385 GST_ERROR ("client %p: failed to parse SDP message", client);
2386 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2387 gst_sdp_message_free (sdp);
2392 GST_ERROR ("client %p: no media", client);
2394 /* error reply is already sent */
2395 gst_sdp_message_free (sdp);
2400 GST_ERROR ("client %p: media does not support ANNOUNCE", client);
2401 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2403 g_object_unref (media);
2404 gst_sdp_message_free (sdp);
2409 GST_ERROR ("client %p: can't handle SDP", client);
2410 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
2412 g_object_unref (media);
2413 gst_sdp_message_free (sdp);
2419 handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
2421 GstRTSPSession *session;
2422 GstRTSPClientClass *klass;
2423 GstRTSPSessionMedia *sessmedia;
2424 GstRTSPMedia *media;
2426 GstRTSPState rtspstate;
2430 if (!(session = ctx->session))
2433 if (!(uri = ctx->uri))
2436 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2437 path = klass->make_path_from_uri (client, uri);
2439 /* get a handle to the configuration of the media in the session */
2440 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
2444 if (path[matched] != '\0')
2449 ctx->sessmedia = sessmedia;
2450 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
2452 if (!(gst_rtsp_media_get_transport_mode (media) &
2453 GST_RTSP_TRANSPORT_MODE_RECORD))
2454 goto unsupported_mode;
2456 /* the session state must be playing or ready */
2457 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
2458 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
2461 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
2462 if (!gst_rtsp_media_unsuspend (media))
2463 goto unsuspend_failed;
2465 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2466 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2468 send_message (client, ctx, ctx->response, FALSE);
2470 /* start playing after sending the response */
2471 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
2473 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
2475 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
2483 GST_ERROR ("client %p: no session", client);
2484 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2489 GST_ERROR ("client %p: no uri supplied", client);
2490 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2495 GST_ERROR ("client %p: media not found", client);
2496 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2501 GST_ERROR ("client %p: no aggregate path %s", client, path);
2502 send_generic_response (client,
2503 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
2509 GST_ERROR ("client %p: media does not support RECORD", client);
2510 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2515 GST_ERROR ("client %p: not PLAYING or READY", client);
2516 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
2522 GST_ERROR ("client %p: unsuspend failed", client);
2523 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2529 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2531 GstRTSPMethod options;
2534 options = GST_RTSP_DESCRIBE |
2539 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2541 str = gst_rtsp_options_as_text (options);
2543 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2544 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2546 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2549 send_message (client, ctx, ctx->response, FALSE);
2551 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2557 /* remove duplicate and trailing '/' */
2559 sanitize_uri (GstRTSPUrl * uri)
2563 gboolean have_slash, prev_slash;
2565 s = d = uri->abspath;
2566 len = strlen (uri->abspath);
2570 for (i = 0; i < len; i++) {
2571 have_slash = s[i] == '/';
2573 if (!have_slash || !prev_slash)
2575 prev_slash = have_slash;
2577 len = d - uri->abspath;
2578 /* don't remove the first slash if that's the only thing left */
2579 if (len > 1 && *(d - 1) == '/')
2584 /* is called when the session is removed from its session pool. */
2586 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
2587 GstRTSPClient * client)
2589 GstRTSPClientPrivate *priv = client->priv;
2591 GST_INFO ("client %p: session %p removed", client, session);
2593 g_mutex_lock (&priv->lock);
2594 if (priv->watch != NULL)
2595 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2596 client_unwatch_session (client, session, NULL);
2597 if (priv->watch != NULL)
2598 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2599 g_mutex_unlock (&priv->lock);
2602 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2603 * and also returns a newly-allocated string of (comma-separated) unsupported
2604 * options in the unsupported_reqs variable .
2606 * There may be multiple Require headers, but we must send one single
2607 * Unsupported header with all the unsupported options as response. If
2608 * an incoming Require header contained a comma-separated list of options
2609 * GstRtspConnection will already have split that list up into multiple
2612 * TODO: allow the application to decide what features are supported
2615 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2618 GPtrArray *arr = NULL;
2624 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2626 if (res == GST_RTSP_ENOTIMPL)
2630 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2632 g_ptr_array_add (arr, g_strdup (reqs));
2636 /* if we don't have any Require headers at all, all is fine */
2640 /* otherwise we've now processed at all the Require headers */
2641 g_ptr_array_add (arr, NULL);
2643 /* for now we don't commit to supporting anything, so will just report
2644 * all of the required options as unsupported */
2645 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2647 g_ptr_array_unref (arr);
2652 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2654 GstRTSPClientPrivate *priv = client->priv;
2655 GstRTSPMethod method;
2656 const gchar *uristr;
2657 GstRTSPUrl *uri = NULL;
2658 GstRTSPVersion version;
2660 GstRTSPSession *session = NULL;
2661 GstRTSPContext sctx = { NULL }, *ctx;
2662 GstRTSPMessage response = { 0 };
2663 gchar *unsupported_reqs = NULL;
2666 if (!(ctx = gst_rtsp_context_get_current ())) {
2668 ctx->auth = priv->auth;
2669 gst_rtsp_context_push_current (ctx);
2672 ctx->conn = priv->connection;
2673 ctx->client = client;
2674 ctx->request = request;
2675 ctx->response = &response;
2677 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2678 gst_rtsp_message_dump (request);
2681 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2683 GST_INFO ("client %p: received a request %s %s %s", client,
2684 gst_rtsp_method_as_text (method), uristr,
2685 gst_rtsp_version_as_text (version));
2687 /* we can only handle 1.0 requests */
2688 if (version != GST_RTSP_VERSION_1_0)
2691 ctx->method = method;
2693 /* we always try to parse the url first */
2694 if (strcmp (uristr, "*") == 0) {
2695 /* special case where we have * as uri, keep uri = NULL */
2696 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2697 /* check if the uristr is an absolute path <=> scheme and host information
2701 scheme = g_uri_parse_scheme (uristr);
2702 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2703 gchar *absolute_uristr = NULL;
2705 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2706 if (priv->server_ip == NULL) {
2707 GST_WARNING_OBJECT (client, "host information missing");
2712 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2714 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2715 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2716 g_free (absolute_uristr);
2719 g_free (absolute_uristr);
2726 /* get the session if there is any */
2727 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2728 if (res == GST_RTSP_OK) {
2729 if (priv->session_pool == NULL)
2732 /* we had a session in the request, find it again */
2733 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2734 goto session_not_found;
2736 /* we add the session to the client list of watched sessions. When a session
2737 * disappears because it times out, we will be notified. If all sessions are
2738 * gone, we will close the connection */
2739 client_watch_session (client, session);
2742 /* sanitize the uri */
2746 ctx->session = session;
2748 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2749 goto not_authorized;
2751 /* handle any 'Require' headers */
2752 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2753 goto unsupported_requirement;
2755 /* the backlog must be unlimited while processing requests.
2756 * the causes of this are two cases of deadlocks while streaming over TCP:
2758 * 1. consider the scenario where the media pipeline's streaming thread
2759 * is blocking in the appsink (taking the appsink's preroll lock) because
2760 * the backlog is full. when a PAUSE request is received by the RTSP
2761 * client thread then the the state of the session media ought to change
2762 * to PAUSED. while most elements in the pipeline can change state this
2763 * can never happen for the appsink since its preroll lock is taken by
2766 * 2. consider the scenario where the media pipeline's streaming thread
2767 * is blocking in the appsink new_sample callback (taking the send lock
2768 * in RTSP client) because the backlog is full. when e.g. a GET request
2769 * is received by the RTSP client thread then a response ought to be sent
2770 * but this can never happen since it requires taking the send lock
2771 * already taken by another thread.
2773 * the reason that the backlog is never emptied is that the source used
2774 * for dequeing messages from the backlog is never dispatched because it
2775 * is attached to the same mainloop as the source receving RTSP requests and
2776 * therefore run by the RTSP client thread which is alreayd blocking.
2778 * without significant changes the easiest way to cope with this is to
2779 * not block indefinitely when the backlog is full, but rather let the
2780 * backlog grow in size. this in effect means that there can not be any
2781 * upper boundary on its size.
2783 if (priv->watch != NULL)
2784 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2786 /* now see what is asked and dispatch to a dedicated handler */
2788 case GST_RTSP_OPTIONS:
2789 handle_options_request (client, ctx);
2791 case GST_RTSP_DESCRIBE:
2792 handle_describe_request (client, ctx);
2794 case GST_RTSP_SETUP:
2795 handle_setup_request (client, ctx);
2798 handle_play_request (client, ctx);
2800 case GST_RTSP_PAUSE:
2801 handle_pause_request (client, ctx);
2803 case GST_RTSP_TEARDOWN:
2804 handle_teardown_request (client, ctx);
2806 case GST_RTSP_SET_PARAMETER:
2807 handle_set_param_request (client, ctx);
2809 case GST_RTSP_GET_PARAMETER:
2810 handle_get_param_request (client, ctx);
2812 case GST_RTSP_ANNOUNCE:
2813 handle_announce_request (client, ctx);
2815 case GST_RTSP_RECORD:
2816 handle_record_request (client, ctx);
2818 case GST_RTSP_REDIRECT:
2819 if (priv->watch != NULL)
2820 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2821 goto not_implemented;
2822 case GST_RTSP_INVALID:
2824 if (priv->watch != NULL)
2825 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2829 if (priv->watch != NULL)
2830 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2834 gst_rtsp_context_pop_current (ctx);
2836 g_object_unref (session);
2838 gst_rtsp_url_free (uri);
2844 GST_ERROR ("client %p: version %d not supported", client, version);
2845 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2851 GST_ERROR ("client %p: bad request", client);
2852 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2857 GST_ERROR ("client %p: no pool configured", client);
2858 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2863 GST_ERROR ("client %p: session not found", client);
2864 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2869 GST_ERROR ("client %p: not allowed", client);
2870 /* error reply is already sent */
2873 unsupported_requirement:
2875 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2877 send_option_not_supported_response (client, ctx, unsupported_reqs);
2878 g_free (unsupported_reqs);
2883 GST_ERROR ("client %p: method %d not implemented", client, method);
2884 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2891 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2893 GstRTSPClientPrivate *priv = client->priv;
2895 GstRTSPSession *session = NULL;
2896 GstRTSPContext sctx = { NULL }, *ctx;
2899 if (!(ctx = gst_rtsp_context_get_current ())) {
2901 ctx->auth = priv->auth;
2902 gst_rtsp_context_push_current (ctx);
2905 ctx->conn = priv->connection;
2906 ctx->client = client;
2907 ctx->request = NULL;
2909 ctx->method = GST_RTSP_INVALID;
2910 ctx->response = response;
2912 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2913 gst_rtsp_message_dump (response);
2916 GST_INFO ("client %p: received a response", client);
2918 /* get the session if there is any */
2920 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2921 if (res == GST_RTSP_OK) {
2922 if (priv->session_pool == NULL)
2925 /* we had a session in the request, find it again */
2926 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2927 goto session_not_found;
2929 /* we add the session to the client list of watched sessions. When a session
2930 * disappears because it times out, we will be notified. If all sessions are
2931 * gone, we will close the connection */
2932 client_watch_session (client, session);
2935 ctx->session = session;
2937 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2942 gst_rtsp_context_pop_current (ctx);
2944 g_object_unref (session);
2949 GST_ERROR ("client %p: no pool configured", client);
2954 GST_ERROR ("client %p: session not found", client);
2960 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2962 GstRTSPClientPrivate *priv = client->priv;
2968 GstRTSPStreamTransport *trans;
2970 /* find the stream for this message */
2971 res = gst_rtsp_message_parse_data (message, &channel);
2972 if (res != GST_RTSP_OK)
2975 gst_rtsp_message_get_body (message, &data, &size);
2977 goto invalid_length;
2979 gst_rtsp_message_steal_body (message, &data, &size);
2981 /* Strip trailing \0 (which GstRTSPConnection adds) */
2984 buffer = gst_buffer_new_wrapped (data, size);
2987 g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
2989 /* dispatch to the stream based on the channel number */
2990 GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
2991 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
2993 GST_DEBUG_OBJECT (client, "received %u bytes of data for "
2994 "unknown channel %u", size, channel);
2995 gst_buffer_unref (buffer);
3003 GST_DEBUG ("client %p: Short message received, ignoring", client);
3009 * gst_rtsp_client_set_session_pool:
3010 * @client: a #GstRTSPClient
3011 * @pool: (transfer none): a #GstRTSPSessionPool
3013 * Set @pool as the sessionpool for @client which it will use to find
3014 * or allocate sessions. the sessionpool is usually inherited from the server
3015 * that created the client but can be overridden later.
3018 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
3019 GstRTSPSessionPool * pool)
3021 GstRTSPSessionPool *old;
3022 GstRTSPClientPrivate *priv;
3024 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3026 priv = client->priv;
3029 g_object_ref (pool);
3031 g_mutex_lock (&priv->lock);
3032 old = priv->session_pool;
3033 priv->session_pool = pool;
3035 if (priv->session_removed_id) {
3036 g_signal_handler_disconnect (old, priv->session_removed_id);
3037 priv->session_removed_id = 0;
3039 g_mutex_unlock (&priv->lock);
3041 /* FIXME, should remove all sessions from the old pool for this client */
3043 g_object_unref (old);
3047 * gst_rtsp_client_get_session_pool:
3048 * @client: a #GstRTSPClient
3050 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
3052 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
3054 GstRTSPSessionPool *
3055 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
3057 GstRTSPClientPrivate *priv;
3058 GstRTSPSessionPool *result;
3060 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3062 priv = client->priv;
3064 g_mutex_lock (&priv->lock);
3065 if ((result = priv->session_pool))
3066 g_object_ref (result);
3067 g_mutex_unlock (&priv->lock);
3073 * gst_rtsp_client_set_mount_points:
3074 * @client: a #GstRTSPClient
3075 * @mounts: (transfer none): a #GstRTSPMountPoints
3077 * Set @mounts as the mount points for @client which it will use to map urls
3078 * to media streams. These mount points are usually inherited from the server that
3079 * created the client but can be overriden later.
3082 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
3083 GstRTSPMountPoints * mounts)
3085 GstRTSPClientPrivate *priv;
3086 GstRTSPMountPoints *old;
3088 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3090 priv = client->priv;
3093 g_object_ref (mounts);
3095 g_mutex_lock (&priv->lock);
3096 old = priv->mount_points;
3097 priv->mount_points = mounts;
3098 g_mutex_unlock (&priv->lock);
3101 g_object_unref (old);
3105 * gst_rtsp_client_get_mount_points:
3106 * @client: a #GstRTSPClient
3108 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
3110 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
3112 GstRTSPMountPoints *
3113 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
3115 GstRTSPClientPrivate *priv;
3116 GstRTSPMountPoints *result;
3118 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3120 priv = client->priv;
3122 g_mutex_lock (&priv->lock);
3123 if ((result = priv->mount_points))
3124 g_object_ref (result);
3125 g_mutex_unlock (&priv->lock);
3131 * gst_rtsp_client_set_auth:
3132 * @client: a #GstRTSPClient
3133 * @auth: (transfer none): a #GstRTSPAuth
3135 * configure @auth to be used as the authentication manager of @client.
3138 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
3140 GstRTSPClientPrivate *priv;
3143 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3145 priv = client->priv;
3148 g_object_ref (auth);
3150 g_mutex_lock (&priv->lock);
3153 g_mutex_unlock (&priv->lock);
3156 g_object_unref (old);
3161 * gst_rtsp_client_get_auth:
3162 * @client: a #GstRTSPClient
3164 * Get the #GstRTSPAuth used as the authentication manager of @client.
3166 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
3170 gst_rtsp_client_get_auth (GstRTSPClient * client)
3172 GstRTSPClientPrivate *priv;
3173 GstRTSPAuth *result;
3175 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3177 priv = client->priv;
3179 g_mutex_lock (&priv->lock);
3180 if ((result = priv->auth))
3181 g_object_ref (result);
3182 g_mutex_unlock (&priv->lock);
3188 * gst_rtsp_client_set_thread_pool:
3189 * @client: a #GstRTSPClient
3190 * @pool: (transfer none): a #GstRTSPThreadPool
3192 * configure @pool to be used as the thread pool of @client.
3195 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
3196 GstRTSPThreadPool * pool)
3198 GstRTSPClientPrivate *priv;
3199 GstRTSPThreadPool *old;
3201 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3203 priv = client->priv;
3206 g_object_ref (pool);
3208 g_mutex_lock (&priv->lock);
3209 old = priv->thread_pool;
3210 priv->thread_pool = pool;
3211 g_mutex_unlock (&priv->lock);
3214 g_object_unref (old);
3218 * gst_rtsp_client_get_thread_pool:
3219 * @client: a #GstRTSPClient
3221 * Get the #GstRTSPThreadPool used as the thread pool of @client.
3223 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
3227 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
3229 GstRTSPClientPrivate *priv;
3230 GstRTSPThreadPool *result;
3232 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3234 priv = client->priv;
3236 g_mutex_lock (&priv->lock);
3237 if ((result = priv->thread_pool))
3238 g_object_ref (result);
3239 g_mutex_unlock (&priv->lock);
3245 * gst_rtsp_client_set_connection:
3246 * @client: a #GstRTSPClient
3247 * @conn: (transfer full): a #GstRTSPConnection
3249 * Set the #GstRTSPConnection of @client. This function takes ownership of
3252 * Returns: %TRUE on success.
3255 gst_rtsp_client_set_connection (GstRTSPClient * client,
3256 GstRTSPConnection * conn)
3258 GstRTSPClientPrivate *priv;
3259 GSocket *read_socket;
3260 GSocketAddress *address;
3262 GError *error = NULL;
3264 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
3265 g_return_val_if_fail (conn != NULL, FALSE);
3267 priv = client->priv;
3269 read_socket = gst_rtsp_connection_get_read_socket (conn);
3271 if (!(address = g_socket_get_local_address (read_socket, &error)))
3274 g_free (priv->server_ip);
3275 /* keep the original ip that the client connected to */
3276 if (G_IS_INET_SOCKET_ADDRESS (address)) {
3277 GInetAddress *iaddr;
3279 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
3281 /* socket might be ipv6 but adress still ipv4 */
3282 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
3283 priv->server_ip = g_inet_address_to_string (iaddr);
3284 g_object_unref (address);
3286 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
3287 priv->server_ip = g_strdup ("unknown");
3290 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
3291 priv->server_ip, priv->is_ipv6);
3293 url = gst_rtsp_connection_get_url (conn);
3294 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
3296 priv->connection = conn;
3303 GST_ERROR ("could not get local address %s", error->message);
3304 g_error_free (error);
3310 * gst_rtsp_client_get_connection:
3311 * @client: a #GstRTSPClient
3313 * Get the #GstRTSPConnection of @client.
3315 * Returns: (transfer none): the #GstRTSPConnection of @client.
3316 * The connection object returned remains valid until the client is freed.
3319 gst_rtsp_client_get_connection (GstRTSPClient * client)
3321 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3323 return client->priv->connection;
3327 * gst_rtsp_client_set_send_func:
3328 * @client: a #GstRTSPClient
3329 * @func: (scope notified): a #GstRTSPClientSendFunc
3330 * @user_data: (closure): user data passed to @func
3331 * @notify: (allow-none): called when @user_data is no longer in use
3333 * Set @func as the callback that will be called when a new message needs to be
3334 * sent to the client. @user_data is passed to @func and @notify is called when
3335 * @user_data is no longer in use.
3337 * By default, the client will send the messages on the #GstRTSPConnection that
3338 * was configured with gst_rtsp_client_attach() was called.
3341 gst_rtsp_client_set_send_func (GstRTSPClient * client,
3342 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
3344 GstRTSPClientPrivate *priv;
3345 GDestroyNotify old_notify;
3348 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3350 priv = client->priv;
3352 g_mutex_lock (&priv->send_lock);
3353 priv->send_func = func;
3354 old_notify = priv->send_notify;
3355 old_data = priv->send_data;
3356 priv->send_notify = notify;
3357 priv->send_data = user_data;
3358 g_mutex_unlock (&priv->send_lock);
3361 old_notify (old_data);
3365 * gst_rtsp_client_handle_message:
3366 * @client: a #GstRTSPClient
3367 * @message: (transfer none): an #GstRTSPMessage
3369 * Let the client handle @message.
3371 * Returns: a #GstRTSPResult.
3374 gst_rtsp_client_handle_message (GstRTSPClient * client,
3375 GstRTSPMessage * message)
3377 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
3378 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
3380 switch (message->type) {
3381 case GST_RTSP_MESSAGE_REQUEST:
3382 handle_request (client, message);
3384 case GST_RTSP_MESSAGE_RESPONSE:
3385 handle_response (client, message);
3387 case GST_RTSP_MESSAGE_DATA:
3388 handle_data (client, message);
3397 * gst_rtsp_client_send_message:
3398 * @client: a #GstRTSPClient
3399 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
3400 * the message to or %NULL
3401 * @message: (transfer none): The #GstRTSPMessage to send
3403 * Send a message message to the remote end. @message must be a
3404 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
3407 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
3408 GstRTSPMessage * message)
3410 GstRTSPContext sctx = { NULL }
3412 GstRTSPClientPrivate *priv;
3414 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
3415 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
3416 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
3417 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
3419 priv = client->priv;
3421 if (!(ctx = gst_rtsp_context_get_current ())) {
3423 ctx->auth = priv->auth;
3424 gst_rtsp_context_push_current (ctx);
3427 ctx->conn = priv->connection;
3428 ctx->client = client;
3429 ctx->session = session;
3431 send_message (client, ctx, message, FALSE);
3434 gst_rtsp_context_pop_current (ctx);
3439 static GstRTSPResult
3440 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
3441 gboolean close, gpointer user_data)
3443 GstRTSPClientPrivate *priv = client->priv;
3451 /* send the response and store the seq number so we can wait until it's
3452 * written to the client to close the connection */
3454 gst_rtsp_watch_send_message (priv->watch, message,
3455 close ? &priv->close_seq : NULL);
3456 if (ret == GST_RTSP_OK)
3459 if (ret != GST_RTSP_ENOMEM)
3463 if (priv->drop_backlog)
3466 /* queue was full, wait for more space */
3467 GST_DEBUG_OBJECT (client, "waiting for backlog");
3468 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3469 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3470 } while (ret != GST_RTSP_EINTR);
3477 GST_DEBUG_OBJECT (client, "got error %d", ret);
3482 static GstRTSPResult
3483 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3486 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3489 static GstRTSPResult
3490 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3492 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3493 GstRTSPClientPrivate *priv = client->priv;
3495 if (priv->close_seq && priv->close_seq == cseq) {
3496 GST_INFO ("client %p: send close message", client);
3497 priv->close_seq = 0;
3498 gst_rtsp_client_close (client);
3504 static GstRTSPResult
3505 closed (GstRTSPWatch * watch, gpointer user_data)
3507 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3508 GstRTSPClientPrivate *priv = client->priv;
3509 const gchar *tunnelid;
3511 GST_INFO ("client %p: connection closed", client);
3513 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3514 g_mutex_lock (&tunnels_lock);
3515 /* remove from tunnelids */
3516 g_hash_table_remove (tunnels, tunnelid);
3517 g_mutex_unlock (&tunnels_lock);
3520 gst_rtsp_watch_set_flushing (watch, TRUE);
3521 g_mutex_lock (&priv->watch_lock);
3522 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3523 g_mutex_unlock (&priv->watch_lock);
3528 static GstRTSPResult
3529 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3531 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3534 str = gst_rtsp_strresult (result);
3535 GST_INFO ("client %p: received an error %s", client, str);
3541 static GstRTSPResult
3542 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3543 GstRTSPMessage * message, guint id, gpointer user_data)
3545 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3548 str = gst_rtsp_strresult (result);
3550 ("client %p: error when handling message %p with id %d: %s",
3551 client, message, id, str);
3558 remember_tunnel (GstRTSPClient * client)
3560 GstRTSPClientPrivate *priv = client->priv;
3561 const gchar *tunnelid;
3563 /* store client in the pending tunnels */
3564 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3565 if (tunnelid == NULL)
3568 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3570 /* we can't have two clients connecting with the same tunnelid */
3571 g_mutex_lock (&tunnels_lock);
3572 if (g_hash_table_lookup (tunnels, tunnelid))
3573 goto tunnel_existed;
3575 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3576 g_mutex_unlock (&tunnels_lock);
3583 GST_ERROR ("client %p: no tunnelid provided", client);
3588 g_mutex_unlock (&tunnels_lock);
3589 GST_ERROR ("client %p: tunnel session %s already existed", client,
3595 static GstRTSPResult
3596 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3598 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3599 GstRTSPClientPrivate *priv = client->priv;
3601 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3604 /* ignore error, it'll only be a problem when the client does a POST again */
3605 remember_tunnel (client);
3611 handle_tunnel (GstRTSPClient * client)
3613 GstRTSPClientPrivate *priv = client->priv;
3614 GstRTSPClient *oclient;
3615 GstRTSPClientPrivate *opriv;
3616 const gchar *tunnelid;
3618 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3619 if (tunnelid == NULL)
3622 /* check for previous tunnel */
3623 g_mutex_lock (&tunnels_lock);
3624 oclient = g_hash_table_lookup (tunnels, tunnelid);
3626 if (oclient == NULL) {
3627 /* no previous tunnel, remember tunnel */
3628 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3629 g_mutex_unlock (&tunnels_lock);
3631 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3632 client, priv->connection);
3634 /* merge both tunnels into the first client */
3635 /* remove the old client from the table. ref before because removing it will
3636 * remove the ref to it. */
3637 g_object_ref (oclient);
3638 g_hash_table_remove (tunnels, tunnelid);
3639 g_mutex_unlock (&tunnels_lock);
3641 opriv = oclient->priv;
3643 g_mutex_lock (&opriv->watch_lock);
3644 if (opriv->watch == NULL)
3647 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3648 oclient, opriv->connection, priv->connection);
3650 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3651 gst_rtsp_watch_reset (priv->watch);
3652 gst_rtsp_watch_reset (opriv->watch);
3653 g_mutex_unlock (&opriv->watch_lock);
3654 g_object_unref (oclient);
3656 /* the old client owns the tunnel now, the new one will be freed */
3657 g_source_destroy ((GSource *) priv->watch);
3659 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3667 GST_ERROR ("client %p: no tunnelid provided", client);
3672 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3673 g_mutex_unlock (&opriv->watch_lock);
3674 g_object_unref (oclient);
3679 static GstRTSPStatusCode
3680 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3682 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3684 GST_INFO ("client %p: tunnel get (connection %p)", client,
3685 client->priv->connection);
3687 if (!handle_tunnel (client)) {
3688 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3691 return GST_RTSP_STS_OK;
3694 static GstRTSPResult
3695 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3697 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3699 GST_INFO ("client %p: tunnel post (connection %p)", client,
3700 client->priv->connection);
3702 if (!handle_tunnel (client)) {
3703 return GST_RTSP_ERROR;
3709 static GstRTSPResult
3710 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3711 GstRTSPMessage * response, gpointer user_data)
3713 GstRTSPClientClass *klass;
3715 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3716 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3718 if (klass->tunnel_http_response) {
3719 klass->tunnel_http_response (client, request, response);
3725 static GstRTSPWatchFuncs watch_funcs = {
3734 tunnel_http_response
3738 client_watch_notify (GstRTSPClient * client)
3740 GstRTSPClientPrivate *priv = client->priv;
3742 GST_INFO ("client %p: watch destroyed", client);
3744 /* remove all sessions and so drop the extra client ref */
3745 gst_rtsp_client_session_filter (client, cleanup_session, NULL);
3746 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3747 g_object_unref (client);
3751 * gst_rtsp_client_attach:
3752 * @client: a #GstRTSPClient
3753 * @context: (allow-none): a #GMainContext
3755 * Attaches @client to @context. When the mainloop for @context is run, the
3756 * client will be dispatched. When @context is %NULL, the default context will be
3759 * This function should be called when the client properties and urls are fully
3760 * configured and the client is ready to start.
3762 * Returns: the ID (greater than 0) for the source within the GMainContext.
3765 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3767 GstRTSPClientPrivate *priv;
3770 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3771 priv = client->priv;
3772 g_return_val_if_fail (priv->connection != NULL, 0);
3773 g_return_val_if_fail (priv->watch == NULL, 0);
3775 /* make sure noone will free the context before the watch is destroyed */
3776 priv->watch_context = g_main_context_ref (context);
3778 /* create watch for the connection and attach */
3779 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3780 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3781 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3782 (GDestroyNotify) gst_rtsp_watch_unref);
3784 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
3786 GST_INFO ("client %p: attaching to context %p", client, context);
3787 res = gst_rtsp_watch_attach (priv->watch, context);
3793 * gst_rtsp_client_session_filter:
3794 * @client: a #GstRTSPClient
3795 * @func: (scope call) (allow-none): a callback
3796 * @user_data: user data passed to @func
3798 * Call @func for each session managed by @client. The result value of @func
3799 * determines what happens to the session. @func will be called with @client
3800 * locked so no further actions on @client can be performed from @func.
3802 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3805 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3807 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3808 * will also be added with an additional ref to the result #GList of this
3811 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3813 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3814 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3815 * element in the #GList should be unreffed before the list is freed.
3818 gst_rtsp_client_session_filter (GstRTSPClient * client,
3819 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3821 GstRTSPClientPrivate *priv;
3822 GList *result, *walk, *next;
3823 GHashTable *visited;
3826 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3828 priv = client->priv;
3832 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3834 g_mutex_lock (&priv->lock);
3836 cookie = priv->sessions_cookie;
3837 for (walk = priv->sessions; walk; walk = next) {
3838 GstRTSPSession *sess = walk->data;
3839 GstRTSPFilterResult res;
3842 next = g_list_next (walk);
3845 /* only visit each session once */
3846 if (g_hash_table_contains (visited, sess))
3849 g_hash_table_add (visited, g_object_ref (sess));
3850 g_mutex_unlock (&priv->lock);
3852 res = func (client, sess, user_data);
3854 g_mutex_lock (&priv->lock);
3856 res = GST_RTSP_FILTER_REF;
3858 changed = (cookie != priv->sessions_cookie);
3861 case GST_RTSP_FILTER_REMOVE:
3862 /* stop watching the session and pretend it went away, if the list was
3863 * changed, we can't use the current list position, try to see if we
3864 * still have the session */
3865 client_unwatch_session (client, sess, changed ? NULL : walk);
3866 cookie = priv->sessions_cookie;
3868 case GST_RTSP_FILTER_REF:
3869 result = g_list_prepend (result, g_object_ref (sess));
3871 case GST_RTSP_FILTER_KEEP:
3878 g_mutex_unlock (&priv->lock);
3881 g_hash_table_unref (visited);