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 if (!gst_rtsp_media_seek (media, range)) {
1169 gst_rtsp_range_free (range);
1172 gst_rtsp_range_free (range);
1176 /* grab RTPInfo from the media now */
1177 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1179 /* construct the response now */
1180 code = GST_RTSP_STS_OK;
1181 gst_rtsp_message_init_response (ctx->response, code,
1182 gst_rtsp_status_as_text (code), ctx->request);
1184 /* add the RTP-Info header */
1186 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1190 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1192 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1194 send_message (client, ctx, ctx->response, FALSE);
1196 /* start playing after sending the response */
1197 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1199 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1201 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1208 GST_ERROR ("client %p: no session", client);
1209 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1214 GST_ERROR ("client %p: no uri supplied", client);
1215 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1220 GST_ERROR ("client %p: media not found", client);
1221 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1226 GST_ERROR ("client %p: no aggregate path %s", client, path);
1227 send_generic_response (client,
1228 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1234 GST_ERROR ("client %p: not PLAYING or READY", client);
1235 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1241 GST_ERROR ("client %p: unsuspend failed", client);
1242 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1247 GST_ERROR ("client %p: seek failed", client);
1248 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1253 GST_ERROR ("client %p: media does not support PLAY", client);
1254 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
1260 do_keepalive (GstRTSPSession * session)
1262 GST_INFO ("keep session %p alive", session);
1263 gst_rtsp_session_touch (session);
1266 /* parse @transport and return a valid transport in @tr. only transports
1267 * supported by @stream are returned. Returns FALSE if no valid transport
1270 parse_transport (const char *transport, GstRTSPStream * stream,
1271 GstRTSPTransport * tr)
1278 gst_rtsp_transport_init (tr);
1280 GST_DEBUG ("parsing transports %s", transport);
1282 transports = g_strsplit (transport, ",", 0);
1284 /* loop through the transports, try to parse */
1285 for (i = 0; transports[i]; i++) {
1286 res = gst_rtsp_transport_parse (transports[i], tr);
1287 if (res != GST_RTSP_OK) {
1288 /* no valid transport, search some more */
1289 GST_WARNING ("could not parse transport %s", transports[i]);
1293 /* we have a transport, see if it's supported */
1294 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1295 GST_WARNING ("unsupported transport %s", transports[i]);
1299 /* we have a valid transport */
1300 GST_INFO ("found valid transport %s", transports[i]);
1305 gst_rtsp_transport_init (tr);
1307 g_strfreev (transports);
1313 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1314 GstRTSPStream * stream, GstRTSPContext * ctx)
1316 GstRTSPMessage *request = ctx->request;
1317 gchar *blocksize_str;
1319 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1320 &blocksize_str, 0) == GST_RTSP_OK) {
1324 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1325 if (end == blocksize_str)
1328 /* we don't want to change the mtu when this media
1329 * can be shared because it impacts other clients */
1330 if (gst_rtsp_media_is_shared (media))
1333 if (blocksize > G_MAXUINT)
1334 blocksize = G_MAXUINT;
1336 gst_rtsp_stream_set_mtu (stream, blocksize);
1344 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1345 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1351 default_configure_client_transport (GstRTSPClient * client,
1352 GstRTSPContext * ctx, GstRTSPTransport * ct)
1354 GstRTSPClientPrivate *priv = client->priv;
1356 /* we have a valid transport now, set the destination of the client. */
1357 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1358 gboolean use_client_settings;
1360 use_client_settings =
1361 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1363 if (ct->destination && use_client_settings) {
1364 GstRTSPAddress *addr;
1366 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1367 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1372 gst_rtsp_address_free (addr);
1374 GstRTSPAddress *addr;
1375 GSocketFamily family;
1377 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1379 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1383 g_free (ct->destination);
1384 ct->destination = g_strdup (addr->address);
1385 ct->port.min = addr->port;
1386 ct->port.max = addr->port + addr->n_ports - 1;
1387 ct->ttl = addr->ttl;
1389 gst_rtsp_address_free (addr);
1394 url = gst_rtsp_connection_get_url (priv->connection);
1395 g_free (ct->destination);
1396 ct->destination = g_strdup (url->host);
1398 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1400 GSocketAddress *addr;
1402 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1403 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1404 /* our read port is the sender port of client */
1405 ct->client_port.min =
1406 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1407 g_object_unref (addr);
1409 if ((addr = g_socket_get_local_address (sock, NULL))) {
1410 ct->server_port.max =
1411 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1412 g_object_unref (addr);
1414 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1415 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1416 /* our write port is the receiver port of client */
1417 ct->client_port.max =
1418 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1419 g_object_unref (addr);
1421 if ((addr = g_socket_get_local_address (sock, NULL))) {
1422 ct->server_port.min =
1423 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1424 g_object_unref (addr);
1426 /* check if the client selected channels for TCP */
1427 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1428 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1438 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1443 static GstRTSPTransport *
1444 make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
1445 GstRTSPContext * ctx, GstRTSPTransport * ct)
1447 GstRTSPTransport *st;
1449 GSocketFamily family;
1451 /* prepare the server transport */
1452 gst_rtsp_transport_new (&st);
1454 st->trans = ct->trans;
1455 st->profile = ct->profile;
1456 st->lower_transport = ct->lower_transport;
1457 st->mode_play = ct->mode_play;
1458 st->mode_record = ct->mode_record;
1460 addr = g_inet_address_new_from_string (ct->destination);
1463 GST_ERROR ("failed to get inet addr from client destination");
1464 family = G_SOCKET_FAMILY_IPV4;
1466 family = g_inet_address_get_family (addr);
1467 g_object_unref (addr);
1471 switch (st->lower_transport) {
1472 case GST_RTSP_LOWER_TRANS_UDP:
1473 st->client_port = ct->client_port;
1474 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1476 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1477 st->port = ct->port;
1478 st->destination = g_strdup (ct->destination);
1481 case GST_RTSP_LOWER_TRANS_TCP:
1482 st->interleaved = ct->interleaved;
1483 st->client_port = ct->client_port;
1484 st->server_port = ct->server_port;
1489 if ((gst_rtsp_media_get_transport_mode (media) &
1490 GST_RTSP_TRANSPORT_MODE_PLAY))
1491 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1496 #define AES_128_KEY_LEN 16
1497 #define AES_256_KEY_LEN 32
1499 #define HMAC_32_KEY_LEN 4
1500 #define HMAC_80_KEY_LEN 10
1503 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1505 const gchar *srtp_cipher;
1506 const gchar *srtp_auth;
1507 const GstMIKEYPayload *sp;
1510 /* loop over Security policy until we find one containing policy */
1512 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1515 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1519 /* the default ciphers */
1520 srtp_cipher = "aes-128-icm";
1521 srtp_auth = "hmac-sha1-80";
1523 /* now override the defaults with what is in the Security Policy */
1527 /* collect all the params and go over them */
1528 len = gst_mikey_payload_sp_get_n_params (sp);
1529 for (i = 0; i < len; i++) {
1530 const GstMIKEYPayloadSPParam *param =
1531 gst_mikey_payload_sp_get_param (sp, i);
1533 switch (param->type) {
1534 case GST_MIKEY_SP_SRTP_ENC_ALG:
1535 switch (param->val[0]) {
1537 srtp_cipher = "null";
1541 srtp_cipher = "aes-128-icm";
1547 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1548 switch (param->val[0]) {
1549 case AES_128_KEY_LEN:
1550 srtp_cipher = "aes-128-icm";
1552 case AES_256_KEY_LEN:
1553 srtp_cipher = "aes-256-icm";
1559 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1560 switch (param->val[0]) {
1566 srtp_auth = "hmac-sha1-80";
1572 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1573 switch (param->val[0]) {
1574 case HMAC_32_KEY_LEN:
1575 srtp_auth = "hmac-sha1-32";
1577 case HMAC_80_KEY_LEN:
1578 srtp_auth = "hmac-sha1-80";
1584 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1586 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1593 /* now configure the SRTP parameters */
1594 gst_caps_set_simple (caps,
1595 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1596 "srtp-auth", G_TYPE_STRING, srtp_auth,
1597 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1598 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1604 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1605 guint8 * data, gsize size)
1607 GstMIKEYMessage *msg;
1609 GstCaps *caps = NULL;
1610 GstMIKEYPayloadKEMAC *kemac;
1611 const GstMIKEYPayloadKeyData *pkd;
1614 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1615 * set of Crypto Sessions protected with the same master key.
1616 * In the context of SRTP, an RTP and its RTCP stream is part of a
1618 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1621 /* we can only handle SRTP crypto sessions for now */
1622 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1623 goto invalid_map_type;
1625 /* get the number of crypto sessions. This maps SSRC to its
1626 * security parameters */
1627 n_cs = gst_mikey_message_get_n_cs (msg);
1629 goto no_crypto_sessions;
1631 /* we also need keys */
1632 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1633 (msg, GST_MIKEY_PT_KEMAC, 0)))
1636 /* we don't support encrypted keys */
1637 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1638 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1639 goto unsupported_encryption;
1641 /* get Key data sub-payload */
1642 pkd = (const GstMIKEYPayloadKeyData *)
1643 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1646 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1649 /* go over all crypto sessions and create the security policy for each
1651 for (i = 0; i < n_cs; i++) {
1652 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1654 caps = gst_caps_new_simple ("application/x-srtp",
1655 "ssrc", G_TYPE_UINT, map->ssrc,
1656 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1657 mikey_apply_policy (caps, msg, map->policy);
1659 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1660 gst_caps_unref (caps);
1662 gst_mikey_message_unref (msg);
1663 gst_buffer_unref (key);
1670 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1675 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1676 goto cleanup_message;
1680 GST_DEBUG_OBJECT (client, "no crypto sessions");
1681 goto cleanup_message;
1685 GST_DEBUG_OBJECT (client, "no keys found");
1686 goto cleanup_message;
1688 unsupported_encryption:
1690 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1691 goto cleanup_message;
1695 gst_mikey_message_unref (msg);
1700 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1703 strip_chars (gchar * str)
1710 if (!IS_STRIP_CHAR (str[len]))
1714 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1715 memmove (str, s, len + 1);
1718 /* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1719 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1722 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1727 specs = g_strsplit (keymgmt, ",", 0);
1728 for (i = 0; specs[i]; i++) {
1731 split = g_strsplit (specs[i], ";", 0);
1732 for (j = 0; split[j]; j++) {
1733 g_strstrip (split[j]);
1734 if (g_str_has_prefix (split[j], "prot=")) {
1735 g_strstrip (split[j] + 5);
1736 if (!g_str_equal (split[j] + 5, "mikey"))
1738 GST_DEBUG ("found mikey");
1739 } else if (g_str_has_prefix (split[j], "uri=")) {
1740 strip_chars (split[j] + 4);
1741 GST_DEBUG ("found uri '%s'", split[j] + 4);
1742 } else if (g_str_has_prefix (split[j], "data=")) {
1745 strip_chars (split[j] + 5);
1746 GST_DEBUG ("found data '%s'", split[j] + 5);
1747 data = g_base64_decode_inplace (split[j] + 5, &size);
1748 handle_mikey_data (client, ctx, data, size);
1758 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1760 GstRTSPClientPrivate *priv = client->priv;
1763 gchar *transport, *keymgmt;
1764 GstRTSPTransport *ct, *st;
1765 GstRTSPStatusCode code;
1766 GstRTSPSession *session;
1767 GstRTSPStreamTransport *trans;
1769 GstRTSPSessionMedia *sessmedia;
1770 GstRTSPMedia *media;
1771 GstRTSPStream *stream;
1772 GstRTSPState rtspstate;
1773 GstRTSPClientClass *klass;
1774 gchar *path, *control = NULL;
1776 gboolean new_session = FALSE;
1782 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1783 path = klass->make_path_from_uri (client, uri);
1785 /* parse the transport */
1787 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1789 if (res != GST_RTSP_OK)
1792 /* we create the session after parsing stuff so that we don't make
1793 * a session for malformed requests */
1794 if (priv->session_pool == NULL)
1797 session = ctx->session;
1800 g_object_ref (session);
1801 /* get a handle to the configuration of the media in the session, this can
1802 * return NULL if this is a new url to manage in this session. */
1803 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1805 /* we need a new media configuration in this session */
1809 /* we have no session media, find one and manage it */
1810 if (sessmedia == NULL) {
1811 /* get a handle to the configuration of the media in the session */
1812 media = find_media (client, ctx, path, &matched);
1814 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1815 g_object_ref (media);
1817 goto media_not_found;
1819 /* no media, not found then */
1821 goto media_not_found_no_reply;
1823 if (path[matched] == '\0') {
1824 if (gst_rtsp_media_n_streams (media) == 1) {
1825 stream = gst_rtsp_media_get_stream (media, 0);
1827 goto control_not_found;
1830 /* path is what matched. */
1831 path[matched] = '\0';
1832 /* control is remainder */
1833 control = &path[matched + 1];
1835 /* find the stream now using the control part */
1836 stream = gst_rtsp_media_find_stream (media, control);
1840 goto stream_not_found;
1842 /* now we have a uri identifying a valid media and stream */
1843 ctx->stream = stream;
1846 if (session == NULL) {
1847 /* create a session if this fails we probably reached our session limit or
1849 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1850 goto service_unavailable;
1852 /* make sure this client is closed when the session is closed */
1853 client_watch_session (client, session);
1856 /* signal new session */
1857 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1860 ctx->session = session;
1863 if (!klass->configure_client_media (client, media, stream, ctx))
1864 goto configure_media_failed_no_reply;
1866 gst_rtsp_transport_new (&ct);
1868 /* parse and find a usable supported transport */
1869 if (!parse_transport (transport, stream, ct))
1870 goto unsupported_transports;
1873 && !(gst_rtsp_media_get_transport_mode (media) &
1874 GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
1875 && !(gst_rtsp_media_get_transport_mode (media) &
1876 GST_RTSP_TRANSPORT_MODE_RECORD)))
1877 goto unsupported_mode;
1879 /* parse the keymgmt */
1880 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1881 &keymgmt, 0) == GST_RTSP_OK) {
1882 if (!handle_keymgmt (client, ctx, keymgmt))
1886 if (sessmedia == NULL) {
1887 /* manage the media in our session now, if not done already */
1888 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1889 /* if we stil have no media, error */
1890 if (sessmedia == NULL)
1891 goto sessmedia_unavailable;
1893 /* don't cache media anymore */
1894 clean_cached_media (client, FALSE);
1896 g_object_unref (media);
1899 ctx->sessmedia = sessmedia;
1901 /* update the client transport */
1902 if (!klass->configure_client_transport (client, ctx, ct))
1903 goto unsupported_client_transport;
1905 /* set in the session media transport */
1906 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1910 /* configure the url used to set this transport, this we will use when
1911 * generating the response for the PLAY request */
1912 gst_rtsp_stream_transport_set_url (trans, uri);
1913 /* configure keepalive for this transport */
1914 gst_rtsp_stream_transport_set_keepalive (trans,
1915 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1917 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1918 /* our callbacks to send data on this TCP connection */
1919 gst_rtsp_stream_transport_set_callbacks (trans,
1920 (GstRTSPSendFunc) do_send_data,
1921 (GstRTSPSendFunc) do_send_data, client, NULL);
1923 g_hash_table_insert (priv->transports,
1924 GINT_TO_POINTER (ct->interleaved.min), trans);
1925 g_object_ref (trans);
1926 g_hash_table_insert (priv->transports,
1927 GINT_TO_POINTER (ct->interleaved.max), trans);
1928 g_object_ref (trans);
1931 /* create and serialize the server transport */
1932 st = make_server_transport (client, media, ctx, ct);
1933 trans_str = gst_rtsp_transport_as_text (st);
1934 gst_rtsp_transport_free (st);
1936 /* construct the response now */
1937 code = GST_RTSP_STS_OK;
1938 gst_rtsp_message_init_response (ctx->response, code,
1939 gst_rtsp_status_as_text (code), ctx->request);
1941 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1945 send_message (client, ctx, ctx->response, FALSE);
1947 /* update the state */
1948 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1949 switch (rtspstate) {
1950 case GST_RTSP_STATE_PLAYING:
1951 case GST_RTSP_STATE_RECORDING:
1952 case GST_RTSP_STATE_READY:
1953 /* no state change */
1956 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1959 g_object_unref (session);
1962 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1969 GST_ERROR ("client %p: no uri", client);
1970 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1975 GST_ERROR ("client %p: no transport", client);
1976 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1981 GST_ERROR ("client %p: no session pool configured", client);
1982 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1985 media_not_found_no_reply:
1987 GST_ERROR ("client %p: media '%s' not found", client, path);
1988 /* error reply is already sent */
1993 GST_ERROR ("client %p: media '%s' not found", client, path);
1994 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1999 GST_ERROR ("client %p: no control in path '%s'", client, path);
2000 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2001 g_object_unref (media);
2006 GST_ERROR ("client %p: stream '%s' not found", client,
2007 GST_STR_NULL (control));
2008 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2009 g_object_unref (media);
2012 service_unavailable:
2014 GST_ERROR ("client %p: can't create session", client);
2015 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2016 g_object_unref (media);
2019 sessmedia_unavailable:
2021 GST_ERROR ("client %p: can't create session media", client);
2022 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2023 g_object_unref (media);
2024 goto cleanup_session;
2026 configure_media_failed_no_reply:
2028 GST_ERROR ("client %p: configure_media failed", client);
2029 /* error reply is already sent */
2030 goto cleanup_session;
2032 unsupported_transports:
2034 GST_ERROR ("client %p: unsupported transports", client);
2035 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2036 goto cleanup_transport;
2038 unsupported_client_transport:
2040 GST_ERROR ("client %p: unsupported client transport", client);
2041 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2042 goto cleanup_transport;
2046 GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
2047 "mode play: %d, mode record: %d)", client,
2048 ! !(gst_rtsp_media_get_transport_mode (media) &
2049 GST_RTSP_TRANSPORT_MODE_PLAY),
2050 ! !(gst_rtsp_media_get_transport_mode (media) &
2051 GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
2052 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2053 goto cleanup_transport;
2057 GST_ERROR ("client %p: keymgmt error", client);
2058 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
2059 goto cleanup_transport;
2063 gst_rtsp_transport_free (ct);
2066 gst_rtsp_session_pool_remove (priv->session_pool, session);
2067 g_object_unref (session);
2074 static GstSDPMessage *
2075 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
2077 GstRTSPClientPrivate *priv = client->priv;
2081 guint64 session_id_tmp;
2082 gchar session_id[21];
2084 gst_sdp_message_new (&sdp);
2086 /* some standard things first */
2087 gst_sdp_message_set_version (sdp, "0");
2094 session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
2095 g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
2098 gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
2101 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2102 gst_sdp_message_set_information (sdp, "rtsp-server");
2103 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2104 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2105 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2106 gst_sdp_message_add_attribute (sdp, "control", "*");
2108 info.is_ipv6 = priv->is_ipv6;
2109 info.server_ip = priv->server_ip;
2111 /* create an SDP for the media object */
2112 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2120 GST_ERROR ("client %p: could not create SDP", client);
2121 gst_sdp_message_free (sdp);
2126 /* for the describe we must generate an SDP */
2128 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2130 GstRTSPClientPrivate *priv = client->priv;
2135 GstRTSPMedia *media;
2136 GstRTSPClientClass *klass;
2138 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2143 /* check what kind of format is accepted, we don't really do anything with it
2144 * and always return SDP for now. */
2149 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2151 if (res == GST_RTSP_ENOTIMPL)
2154 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2158 if (!priv->mount_points)
2159 goto no_mount_points;
2161 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2164 /* find the media object for the uri */
2165 if (!(media = find_media (client, ctx, path, NULL)))
2168 if (!(gst_rtsp_media_get_transport_mode (media) &
2169 GST_RTSP_TRANSPORT_MODE_PLAY))
2170 goto unsupported_mode;
2172 /* create an SDP for the media object on this client */
2173 if (!(sdp = klass->create_sdp (client, media)))
2176 /* we suspend after the describe */
2177 gst_rtsp_media_suspend (media);
2178 g_object_unref (media);
2180 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2181 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2183 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2186 /* content base for some clients that might screw up creating the setup uri */
2187 str = make_base_url (client, ctx->uri, path);
2190 GST_INFO ("adding content-base: %s", str);
2191 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2193 /* add SDP to the response body */
2194 str = gst_sdp_message_as_text (sdp);
2195 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2196 gst_sdp_message_free (sdp);
2198 send_message (client, ctx, ctx->response, FALSE);
2200 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2208 GST_ERROR ("client %p: no uri", client);
2209 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2214 GST_ERROR ("client %p: no mount points configured", client);
2215 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2220 GST_ERROR ("client %p: can't find path for url", client);
2221 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2226 GST_ERROR ("client %p: no media", client);
2228 /* error reply is already sent */
2233 GST_ERROR ("client %p: media does not support DESCRIBE", client);
2234 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2236 g_object_unref (media);
2241 GST_ERROR ("client %p: can't create SDP", client);
2242 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2244 g_object_unref (media);
2250 handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
2251 GstSDPMessage * sdp)
2253 GstRTSPClientPrivate *priv = client->priv;
2254 GstRTSPThread *thread;
2256 /* create an SDP for the media object */
2257 if (!gst_rtsp_media_handle_sdp (media, sdp))
2260 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
2261 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
2265 /* prepare the media */
2266 if (!gst_rtsp_media_prepare (media, thread))
2274 GST_ERROR ("client %p: could not handle SDP", client);
2279 GST_ERROR ("client %p: can't create thread", client);
2284 GST_ERROR ("client %p: can't prepare media", client);
2290 handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
2292 GstRTSPClientPrivate *priv = client->priv;
2293 GstRTSPClientClass *klass;
2296 GstRTSPMedia *media;
2297 gchar *path, *cont = NULL;
2301 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2306 if (!priv->mount_points)
2307 goto no_mount_points;
2309 /* check if reply is SDP */
2310 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
2312 /* could not be set but since the request returned OK, we assume it
2313 * was SDP, else check it. */
2315 if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
2316 goto wrong_content_type;
2319 /* get message body and parse as SDP */
2320 gst_rtsp_message_get_body (ctx->request, &data, &size);
2321 if (data == NULL || size == 0)
2324 GST_DEBUG ("client %p: parse SDP...", client);
2325 gst_sdp_message_new (&sdp);
2326 sres = gst_sdp_message_parse_buffer (data, size, sdp);
2327 if (sres != GST_SDP_OK)
2328 goto sdp_parse_failed;
2330 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2333 /* find the media object for the uri */
2334 if (!(media = find_media (client, ctx, path, NULL)))
2337 if (!(gst_rtsp_media_get_transport_mode (media) &
2338 GST_RTSP_TRANSPORT_MODE_RECORD))
2339 goto unsupported_mode;
2341 /* Tell client subclass about the media */
2342 if (!klass->handle_sdp (client, ctx, media, sdp))
2345 /* we suspend after the announce */
2346 gst_rtsp_media_suspend (media);
2347 g_object_unref (media);
2349 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2350 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2352 send_message (client, ctx, ctx->response, FALSE);
2354 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
2357 gst_sdp_message_free (sdp);
2363 GST_ERROR ("client %p: no uri", client);
2364 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2369 GST_ERROR ("client %p: no mount points configured", client);
2370 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2375 GST_ERROR ("client %p: can't find path for url", client);
2376 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2377 gst_sdp_message_free (sdp);
2382 GST_ERROR ("client %p: unknown content type", client);
2383 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2388 GST_ERROR ("client %p: can't find SDP message", client);
2389 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2394 GST_ERROR ("client %p: failed to parse SDP message", client);
2395 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2396 gst_sdp_message_free (sdp);
2401 GST_ERROR ("client %p: no media", client);
2403 /* error reply is already sent */
2404 gst_sdp_message_free (sdp);
2409 GST_ERROR ("client %p: media does not support ANNOUNCE", client);
2410 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2412 g_object_unref (media);
2413 gst_sdp_message_free (sdp);
2418 GST_ERROR ("client %p: can't handle SDP", client);
2419 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
2421 g_object_unref (media);
2422 gst_sdp_message_free (sdp);
2428 handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
2430 GstRTSPSession *session;
2431 GstRTSPClientClass *klass;
2432 GstRTSPSessionMedia *sessmedia;
2433 GstRTSPMedia *media;
2435 GstRTSPState rtspstate;
2439 if (!(session = ctx->session))
2442 if (!(uri = ctx->uri))
2445 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2446 path = klass->make_path_from_uri (client, uri);
2448 /* get a handle to the configuration of the media in the session */
2449 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
2453 if (path[matched] != '\0')
2458 ctx->sessmedia = sessmedia;
2459 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
2461 if (!(gst_rtsp_media_get_transport_mode (media) &
2462 GST_RTSP_TRANSPORT_MODE_RECORD))
2463 goto unsupported_mode;
2465 /* the session state must be playing or ready */
2466 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
2467 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
2470 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
2471 if (!gst_rtsp_media_unsuspend (media))
2472 goto unsuspend_failed;
2474 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2475 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2477 send_message (client, ctx, ctx->response, FALSE);
2479 /* start playing after sending the response */
2480 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
2482 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
2484 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
2492 GST_ERROR ("client %p: no session", client);
2493 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2498 GST_ERROR ("client %p: no uri supplied", client);
2499 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2504 GST_ERROR ("client %p: media not found", client);
2505 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2510 GST_ERROR ("client %p: no aggregate path %s", client, path);
2511 send_generic_response (client,
2512 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
2518 GST_ERROR ("client %p: media does not support RECORD", client);
2519 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2524 GST_ERROR ("client %p: not PLAYING or READY", client);
2525 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
2531 GST_ERROR ("client %p: unsuspend failed", client);
2532 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2538 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2540 GstRTSPMethod options;
2543 options = GST_RTSP_DESCRIBE |
2548 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2550 str = gst_rtsp_options_as_text (options);
2552 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2553 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2555 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2558 send_message (client, ctx, ctx->response, FALSE);
2560 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2566 /* remove duplicate and trailing '/' */
2568 sanitize_uri (GstRTSPUrl * uri)
2572 gboolean have_slash, prev_slash;
2574 s = d = uri->abspath;
2575 len = strlen (uri->abspath);
2579 for (i = 0; i < len; i++) {
2580 have_slash = s[i] == '/';
2582 if (!have_slash || !prev_slash)
2584 prev_slash = have_slash;
2586 len = d - uri->abspath;
2587 /* don't remove the first slash if that's the only thing left */
2588 if (len > 1 && *(d - 1) == '/')
2593 /* is called when the session is removed from its session pool. */
2595 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
2596 GstRTSPClient * client)
2598 GstRTSPClientPrivate *priv = client->priv;
2600 GST_INFO ("client %p: session %p removed", client, session);
2602 g_mutex_lock (&priv->lock);
2603 if (priv->watch != NULL)
2604 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2605 client_unwatch_session (client, session, NULL);
2606 if (priv->watch != NULL)
2607 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2608 g_mutex_unlock (&priv->lock);
2611 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2612 * and also returns a newly-allocated string of (comma-separated) unsupported
2613 * options in the unsupported_reqs variable .
2615 * There may be multiple Require headers, but we must send one single
2616 * Unsupported header with all the unsupported options as response. If
2617 * an incoming Require header contained a comma-separated list of options
2618 * GstRtspConnection will already have split that list up into multiple
2621 * TODO: allow the application to decide what features are supported
2624 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2627 GPtrArray *arr = NULL;
2633 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2635 if (res == GST_RTSP_ENOTIMPL)
2639 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2641 g_ptr_array_add (arr, g_strdup (reqs));
2645 /* if we don't have any Require headers at all, all is fine */
2649 /* otherwise we've now processed at all the Require headers */
2650 g_ptr_array_add (arr, NULL);
2652 /* for now we don't commit to supporting anything, so will just report
2653 * all of the required options as unsupported */
2654 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2656 g_ptr_array_unref (arr);
2661 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2663 GstRTSPClientPrivate *priv = client->priv;
2664 GstRTSPMethod method;
2665 const gchar *uristr;
2666 GstRTSPUrl *uri = NULL;
2667 GstRTSPVersion version;
2669 GstRTSPSession *session = NULL;
2670 GstRTSPContext sctx = { NULL }, *ctx;
2671 GstRTSPMessage response = { 0 };
2672 gchar *unsupported_reqs = NULL;
2675 if (!(ctx = gst_rtsp_context_get_current ())) {
2677 ctx->auth = priv->auth;
2678 gst_rtsp_context_push_current (ctx);
2681 ctx->conn = priv->connection;
2682 ctx->client = client;
2683 ctx->request = request;
2684 ctx->response = &response;
2686 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2687 gst_rtsp_message_dump (request);
2690 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2692 GST_INFO ("client %p: received a request %s %s %s", client,
2693 gst_rtsp_method_as_text (method), uristr,
2694 gst_rtsp_version_as_text (version));
2696 /* we can only handle 1.0 requests */
2697 if (version != GST_RTSP_VERSION_1_0)
2700 ctx->method = method;
2702 /* we always try to parse the url first */
2703 if (strcmp (uristr, "*") == 0) {
2704 /* special case where we have * as uri, keep uri = NULL */
2705 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2706 /* check if the uristr is an absolute path <=> scheme and host information
2710 scheme = g_uri_parse_scheme (uristr);
2711 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2712 gchar *absolute_uristr = NULL;
2714 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2715 if (priv->server_ip == NULL) {
2716 GST_WARNING_OBJECT (client, "host information missing");
2721 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2723 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2724 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2725 g_free (absolute_uristr);
2728 g_free (absolute_uristr);
2735 /* get the session if there is any */
2736 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2737 if (res == GST_RTSP_OK) {
2738 if (priv->session_pool == NULL)
2741 /* we had a session in the request, find it again */
2742 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2743 goto session_not_found;
2745 /* we add the session to the client list of watched sessions. When a session
2746 * disappears because it times out, we will be notified. If all sessions are
2747 * gone, we will close the connection */
2748 client_watch_session (client, session);
2751 /* sanitize the uri */
2755 ctx->session = session;
2757 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2758 goto not_authorized;
2760 /* handle any 'Require' headers */
2761 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2762 goto unsupported_requirement;
2764 /* the backlog must be unlimited while processing requests.
2765 * the causes of this are two cases of deadlocks while streaming over TCP:
2767 * 1. consider the scenario where the media pipeline's streaming thread
2768 * is blocking in the appsink (taking the appsink's preroll lock) because
2769 * the backlog is full. when a PAUSE request is received by the RTSP
2770 * client thread then the the state of the session media ought to change
2771 * to PAUSED. while most elements in the pipeline can change state this
2772 * can never happen for the appsink since its preroll lock is taken by
2775 * 2. consider the scenario where the media pipeline's streaming thread
2776 * is blocking in the appsink new_sample callback (taking the send lock
2777 * in RTSP client) because the backlog is full. when e.g. a GET request
2778 * is received by the RTSP client thread then a response ought to be sent
2779 * but this can never happen since it requires taking the send lock
2780 * already taken by another thread.
2782 * the reason that the backlog is never emptied is that the source used
2783 * for dequeing messages from the backlog is never dispatched because it
2784 * is attached to the same mainloop as the source receving RTSP requests and
2785 * therefore run by the RTSP client thread which is alreayd blocking.
2787 * without significant changes the easiest way to cope with this is to
2788 * not block indefinitely when the backlog is full, but rather let the
2789 * backlog grow in size. this in effect means that there can not be any
2790 * upper boundary on its size.
2792 if (priv->watch != NULL)
2793 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2795 /* now see what is asked and dispatch to a dedicated handler */
2797 case GST_RTSP_OPTIONS:
2798 handle_options_request (client, ctx);
2800 case GST_RTSP_DESCRIBE:
2801 handle_describe_request (client, ctx);
2803 case GST_RTSP_SETUP:
2804 handle_setup_request (client, ctx);
2807 handle_play_request (client, ctx);
2809 case GST_RTSP_PAUSE:
2810 handle_pause_request (client, ctx);
2812 case GST_RTSP_TEARDOWN:
2813 handle_teardown_request (client, ctx);
2815 case GST_RTSP_SET_PARAMETER:
2816 handle_set_param_request (client, ctx);
2818 case GST_RTSP_GET_PARAMETER:
2819 handle_get_param_request (client, ctx);
2821 case GST_RTSP_ANNOUNCE:
2822 handle_announce_request (client, ctx);
2824 case GST_RTSP_RECORD:
2825 handle_record_request (client, ctx);
2827 case GST_RTSP_REDIRECT:
2828 if (priv->watch != NULL)
2829 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2830 goto not_implemented;
2831 case GST_RTSP_INVALID:
2833 if (priv->watch != NULL)
2834 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2838 if (priv->watch != NULL)
2839 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2843 gst_rtsp_context_pop_current (ctx);
2845 g_object_unref (session);
2847 gst_rtsp_url_free (uri);
2853 GST_ERROR ("client %p: version %d not supported", client, version);
2854 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2860 GST_ERROR ("client %p: bad request", client);
2861 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2866 GST_ERROR ("client %p: no pool configured", client);
2867 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2872 GST_ERROR ("client %p: session not found", client);
2873 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2878 GST_ERROR ("client %p: not allowed", client);
2879 /* error reply is already sent */
2882 unsupported_requirement:
2884 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2886 send_option_not_supported_response (client, ctx, unsupported_reqs);
2887 g_free (unsupported_reqs);
2892 GST_ERROR ("client %p: method %d not implemented", client, method);
2893 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2900 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2902 GstRTSPClientPrivate *priv = client->priv;
2904 GstRTSPSession *session = NULL;
2905 GstRTSPContext sctx = { NULL }, *ctx;
2908 if (!(ctx = gst_rtsp_context_get_current ())) {
2910 ctx->auth = priv->auth;
2911 gst_rtsp_context_push_current (ctx);
2914 ctx->conn = priv->connection;
2915 ctx->client = client;
2916 ctx->request = NULL;
2918 ctx->method = GST_RTSP_INVALID;
2919 ctx->response = response;
2921 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2922 gst_rtsp_message_dump (response);
2925 GST_INFO ("client %p: received a response", client);
2927 /* get the session if there is any */
2929 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2930 if (res == GST_RTSP_OK) {
2931 if (priv->session_pool == NULL)
2934 /* we had a session in the request, find it again */
2935 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2936 goto session_not_found;
2938 /* we add the session to the client list of watched sessions. When a session
2939 * disappears because it times out, we will be notified. If all sessions are
2940 * gone, we will close the connection */
2941 client_watch_session (client, session);
2944 ctx->session = session;
2946 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2951 gst_rtsp_context_pop_current (ctx);
2953 g_object_unref (session);
2958 GST_ERROR ("client %p: no pool configured", client);
2963 GST_ERROR ("client %p: session not found", client);
2969 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2971 GstRTSPClientPrivate *priv = client->priv;
2977 GstRTSPStreamTransport *trans;
2979 /* find the stream for this message */
2980 res = gst_rtsp_message_parse_data (message, &channel);
2981 if (res != GST_RTSP_OK)
2984 gst_rtsp_message_get_body (message, &data, &size);
2986 goto invalid_length;
2988 gst_rtsp_message_steal_body (message, &data, &size);
2990 /* Strip trailing \0 (which GstRTSPConnection adds) */
2993 buffer = gst_buffer_new_wrapped (data, size);
2996 g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
2998 /* dispatch to the stream based on the channel number */
2999 GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
3000 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
3002 GST_DEBUG_OBJECT (client, "received %u bytes of data for "
3003 "unknown channel %u", size, channel);
3004 gst_buffer_unref (buffer);
3012 GST_DEBUG ("client %p: Short message received, ignoring", client);
3018 * gst_rtsp_client_set_session_pool:
3019 * @client: a #GstRTSPClient
3020 * @pool: (transfer none): a #GstRTSPSessionPool
3022 * Set @pool as the sessionpool for @client which it will use to find
3023 * or allocate sessions. the sessionpool is usually inherited from the server
3024 * that created the client but can be overridden later.
3027 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
3028 GstRTSPSessionPool * pool)
3030 GstRTSPSessionPool *old;
3031 GstRTSPClientPrivate *priv;
3033 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3035 priv = client->priv;
3038 g_object_ref (pool);
3040 g_mutex_lock (&priv->lock);
3041 old = priv->session_pool;
3042 priv->session_pool = pool;
3044 if (priv->session_removed_id) {
3045 g_signal_handler_disconnect (old, priv->session_removed_id);
3046 priv->session_removed_id = 0;
3048 g_mutex_unlock (&priv->lock);
3050 /* FIXME, should remove all sessions from the old pool for this client */
3052 g_object_unref (old);
3056 * gst_rtsp_client_get_session_pool:
3057 * @client: a #GstRTSPClient
3059 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
3061 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
3063 GstRTSPSessionPool *
3064 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
3066 GstRTSPClientPrivate *priv;
3067 GstRTSPSessionPool *result;
3069 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3071 priv = client->priv;
3073 g_mutex_lock (&priv->lock);
3074 if ((result = priv->session_pool))
3075 g_object_ref (result);
3076 g_mutex_unlock (&priv->lock);
3082 * gst_rtsp_client_set_mount_points:
3083 * @client: a #GstRTSPClient
3084 * @mounts: (transfer none): a #GstRTSPMountPoints
3086 * Set @mounts as the mount points for @client which it will use to map urls
3087 * to media streams. These mount points are usually inherited from the server that
3088 * created the client but can be overriden later.
3091 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
3092 GstRTSPMountPoints * mounts)
3094 GstRTSPClientPrivate *priv;
3095 GstRTSPMountPoints *old;
3097 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3099 priv = client->priv;
3102 g_object_ref (mounts);
3104 g_mutex_lock (&priv->lock);
3105 old = priv->mount_points;
3106 priv->mount_points = mounts;
3107 g_mutex_unlock (&priv->lock);
3110 g_object_unref (old);
3114 * gst_rtsp_client_get_mount_points:
3115 * @client: a #GstRTSPClient
3117 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
3119 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
3121 GstRTSPMountPoints *
3122 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
3124 GstRTSPClientPrivate *priv;
3125 GstRTSPMountPoints *result;
3127 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3129 priv = client->priv;
3131 g_mutex_lock (&priv->lock);
3132 if ((result = priv->mount_points))
3133 g_object_ref (result);
3134 g_mutex_unlock (&priv->lock);
3140 * gst_rtsp_client_set_auth:
3141 * @client: a #GstRTSPClient
3142 * @auth: (transfer none): a #GstRTSPAuth
3144 * configure @auth to be used as the authentication manager of @client.
3147 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
3149 GstRTSPClientPrivate *priv;
3152 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3154 priv = client->priv;
3157 g_object_ref (auth);
3159 g_mutex_lock (&priv->lock);
3162 g_mutex_unlock (&priv->lock);
3165 g_object_unref (old);
3170 * gst_rtsp_client_get_auth:
3171 * @client: a #GstRTSPClient
3173 * Get the #GstRTSPAuth used as the authentication manager of @client.
3175 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
3179 gst_rtsp_client_get_auth (GstRTSPClient * client)
3181 GstRTSPClientPrivate *priv;
3182 GstRTSPAuth *result;
3184 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3186 priv = client->priv;
3188 g_mutex_lock (&priv->lock);
3189 if ((result = priv->auth))
3190 g_object_ref (result);
3191 g_mutex_unlock (&priv->lock);
3197 * gst_rtsp_client_set_thread_pool:
3198 * @client: a #GstRTSPClient
3199 * @pool: (transfer none): a #GstRTSPThreadPool
3201 * configure @pool to be used as the thread pool of @client.
3204 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
3205 GstRTSPThreadPool * pool)
3207 GstRTSPClientPrivate *priv;
3208 GstRTSPThreadPool *old;
3210 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3212 priv = client->priv;
3215 g_object_ref (pool);
3217 g_mutex_lock (&priv->lock);
3218 old = priv->thread_pool;
3219 priv->thread_pool = pool;
3220 g_mutex_unlock (&priv->lock);
3223 g_object_unref (old);
3227 * gst_rtsp_client_get_thread_pool:
3228 * @client: a #GstRTSPClient
3230 * Get the #GstRTSPThreadPool used as the thread pool of @client.
3232 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
3236 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
3238 GstRTSPClientPrivate *priv;
3239 GstRTSPThreadPool *result;
3241 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3243 priv = client->priv;
3245 g_mutex_lock (&priv->lock);
3246 if ((result = priv->thread_pool))
3247 g_object_ref (result);
3248 g_mutex_unlock (&priv->lock);
3254 * gst_rtsp_client_set_connection:
3255 * @client: a #GstRTSPClient
3256 * @conn: (transfer full): a #GstRTSPConnection
3258 * Set the #GstRTSPConnection of @client. This function takes ownership of
3261 * Returns: %TRUE on success.
3264 gst_rtsp_client_set_connection (GstRTSPClient * client,
3265 GstRTSPConnection * conn)
3267 GstRTSPClientPrivate *priv;
3268 GSocket *read_socket;
3269 GSocketAddress *address;
3271 GError *error = NULL;
3273 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
3274 g_return_val_if_fail (conn != NULL, FALSE);
3276 priv = client->priv;
3278 read_socket = gst_rtsp_connection_get_read_socket (conn);
3280 if (!(address = g_socket_get_local_address (read_socket, &error)))
3283 g_free (priv->server_ip);
3284 /* keep the original ip that the client connected to */
3285 if (G_IS_INET_SOCKET_ADDRESS (address)) {
3286 GInetAddress *iaddr;
3288 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
3290 /* socket might be ipv6 but adress still ipv4 */
3291 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
3292 priv->server_ip = g_inet_address_to_string (iaddr);
3293 g_object_unref (address);
3295 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
3296 priv->server_ip = g_strdup ("unknown");
3299 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
3300 priv->server_ip, priv->is_ipv6);
3302 url = gst_rtsp_connection_get_url (conn);
3303 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
3305 priv->connection = conn;
3312 GST_ERROR ("could not get local address %s", error->message);
3313 g_error_free (error);
3319 * gst_rtsp_client_get_connection:
3320 * @client: a #GstRTSPClient
3322 * Get the #GstRTSPConnection of @client.
3324 * Returns: (transfer none): the #GstRTSPConnection of @client.
3325 * The connection object returned remains valid until the client is freed.
3328 gst_rtsp_client_get_connection (GstRTSPClient * client)
3330 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3332 return client->priv->connection;
3336 * gst_rtsp_client_set_send_func:
3337 * @client: a #GstRTSPClient
3338 * @func: (scope notified): a #GstRTSPClientSendFunc
3339 * @user_data: (closure): user data passed to @func
3340 * @notify: (allow-none): called when @user_data is no longer in use
3342 * Set @func as the callback that will be called when a new message needs to be
3343 * sent to the client. @user_data is passed to @func and @notify is called when
3344 * @user_data is no longer in use.
3346 * By default, the client will send the messages on the #GstRTSPConnection that
3347 * was configured with gst_rtsp_client_attach() was called.
3350 gst_rtsp_client_set_send_func (GstRTSPClient * client,
3351 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
3353 GstRTSPClientPrivate *priv;
3354 GDestroyNotify old_notify;
3357 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3359 priv = client->priv;
3361 g_mutex_lock (&priv->send_lock);
3362 priv->send_func = func;
3363 old_notify = priv->send_notify;
3364 old_data = priv->send_data;
3365 priv->send_notify = notify;
3366 priv->send_data = user_data;
3367 g_mutex_unlock (&priv->send_lock);
3370 old_notify (old_data);
3374 * gst_rtsp_client_handle_message:
3375 * @client: a #GstRTSPClient
3376 * @message: (transfer none): an #GstRTSPMessage
3378 * Let the client handle @message.
3380 * Returns: a #GstRTSPResult.
3383 gst_rtsp_client_handle_message (GstRTSPClient * client,
3384 GstRTSPMessage * message)
3386 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
3387 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
3389 switch (message->type) {
3390 case GST_RTSP_MESSAGE_REQUEST:
3391 handle_request (client, message);
3393 case GST_RTSP_MESSAGE_RESPONSE:
3394 handle_response (client, message);
3396 case GST_RTSP_MESSAGE_DATA:
3397 handle_data (client, message);
3406 * gst_rtsp_client_send_message:
3407 * @client: a #GstRTSPClient
3408 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
3409 * the message to or %NULL
3410 * @message: (transfer none): The #GstRTSPMessage to send
3412 * Send a message message to the remote end. @message must be a
3413 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
3416 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
3417 GstRTSPMessage * message)
3419 GstRTSPContext sctx = { NULL }
3421 GstRTSPClientPrivate *priv;
3423 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
3424 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
3425 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
3426 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
3428 priv = client->priv;
3430 if (!(ctx = gst_rtsp_context_get_current ())) {
3432 ctx->auth = priv->auth;
3433 gst_rtsp_context_push_current (ctx);
3436 ctx->conn = priv->connection;
3437 ctx->client = client;
3438 ctx->session = session;
3440 send_message (client, ctx, message, FALSE);
3443 gst_rtsp_context_pop_current (ctx);
3448 static GstRTSPResult
3449 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
3450 gboolean close, gpointer user_data)
3452 GstRTSPClientPrivate *priv = client->priv;
3460 /* send the response and store the seq number so we can wait until it's
3461 * written to the client to close the connection */
3463 gst_rtsp_watch_send_message (priv->watch, message,
3464 close ? &priv->close_seq : NULL);
3465 if (ret == GST_RTSP_OK)
3468 if (ret != GST_RTSP_ENOMEM)
3472 if (priv->drop_backlog)
3475 /* queue was full, wait for more space */
3476 GST_DEBUG_OBJECT (client, "waiting for backlog");
3477 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3478 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3479 } while (ret != GST_RTSP_EINTR);
3486 GST_DEBUG_OBJECT (client, "got error %d", ret);
3491 static GstRTSPResult
3492 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3495 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3498 static GstRTSPResult
3499 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3501 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3502 GstRTSPClientPrivate *priv = client->priv;
3504 if (priv->close_seq && priv->close_seq == cseq) {
3505 GST_INFO ("client %p: send close message", client);
3506 priv->close_seq = 0;
3507 gst_rtsp_client_close (client);
3513 static GstRTSPResult
3514 closed (GstRTSPWatch * watch, gpointer user_data)
3516 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3517 GstRTSPClientPrivate *priv = client->priv;
3518 const gchar *tunnelid;
3520 GST_INFO ("client %p: connection closed", client);
3522 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3523 g_mutex_lock (&tunnels_lock);
3524 /* remove from tunnelids */
3525 g_hash_table_remove (tunnels, tunnelid);
3526 g_mutex_unlock (&tunnels_lock);
3529 gst_rtsp_watch_set_flushing (watch, TRUE);
3530 g_mutex_lock (&priv->watch_lock);
3531 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3532 g_mutex_unlock (&priv->watch_lock);
3537 static GstRTSPResult
3538 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3540 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3543 str = gst_rtsp_strresult (result);
3544 GST_INFO ("client %p: received an error %s", client, str);
3550 static GstRTSPResult
3551 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3552 GstRTSPMessage * message, guint id, gpointer user_data)
3554 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3557 str = gst_rtsp_strresult (result);
3559 ("client %p: error when handling message %p with id %d: %s",
3560 client, message, id, str);
3567 remember_tunnel (GstRTSPClient * client)
3569 GstRTSPClientPrivate *priv = client->priv;
3570 const gchar *tunnelid;
3572 /* store client in the pending tunnels */
3573 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3574 if (tunnelid == NULL)
3577 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3579 /* we can't have two clients connecting with the same tunnelid */
3580 g_mutex_lock (&tunnels_lock);
3581 if (g_hash_table_lookup (tunnels, tunnelid))
3582 goto tunnel_existed;
3584 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3585 g_mutex_unlock (&tunnels_lock);
3592 GST_ERROR ("client %p: no tunnelid provided", client);
3597 g_mutex_unlock (&tunnels_lock);
3598 GST_ERROR ("client %p: tunnel session %s already existed", client,
3604 static GstRTSPResult
3605 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3607 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3608 GstRTSPClientPrivate *priv = client->priv;
3610 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3613 /* ignore error, it'll only be a problem when the client does a POST again */
3614 remember_tunnel (client);
3620 handle_tunnel (GstRTSPClient * client)
3622 GstRTSPClientPrivate *priv = client->priv;
3623 GstRTSPClient *oclient;
3624 GstRTSPClientPrivate *opriv;
3625 const gchar *tunnelid;
3627 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3628 if (tunnelid == NULL)
3631 /* check for previous tunnel */
3632 g_mutex_lock (&tunnels_lock);
3633 oclient = g_hash_table_lookup (tunnels, tunnelid);
3635 if (oclient == NULL) {
3636 /* no previous tunnel, remember tunnel */
3637 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3638 g_mutex_unlock (&tunnels_lock);
3640 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3641 client, priv->connection);
3643 /* merge both tunnels into the first client */
3644 /* remove the old client from the table. ref before because removing it will
3645 * remove the ref to it. */
3646 g_object_ref (oclient);
3647 g_hash_table_remove (tunnels, tunnelid);
3648 g_mutex_unlock (&tunnels_lock);
3650 opriv = oclient->priv;
3652 g_mutex_lock (&opriv->watch_lock);
3653 if (opriv->watch == NULL)
3656 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3657 oclient, opriv->connection, priv->connection);
3659 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3660 gst_rtsp_watch_reset (priv->watch);
3661 gst_rtsp_watch_reset (opriv->watch);
3662 g_mutex_unlock (&opriv->watch_lock);
3663 g_object_unref (oclient);
3665 /* the old client owns the tunnel now, the new one will be freed */
3666 g_source_destroy ((GSource *) priv->watch);
3668 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3676 GST_ERROR ("client %p: no tunnelid provided", client);
3681 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3682 g_mutex_unlock (&opriv->watch_lock);
3683 g_object_unref (oclient);
3688 static GstRTSPStatusCode
3689 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3691 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3693 GST_INFO ("client %p: tunnel get (connection %p)", client,
3694 client->priv->connection);
3696 if (!handle_tunnel (client)) {
3697 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3700 return GST_RTSP_STS_OK;
3703 static GstRTSPResult
3704 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3706 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3708 GST_INFO ("client %p: tunnel post (connection %p)", client,
3709 client->priv->connection);
3711 if (!handle_tunnel (client)) {
3712 return GST_RTSP_ERROR;
3718 static GstRTSPResult
3719 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3720 GstRTSPMessage * response, gpointer user_data)
3722 GstRTSPClientClass *klass;
3724 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3725 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3727 if (klass->tunnel_http_response) {
3728 klass->tunnel_http_response (client, request, response);
3734 static GstRTSPWatchFuncs watch_funcs = {
3743 tunnel_http_response
3747 client_watch_notify (GstRTSPClient * client)
3749 GstRTSPClientPrivate *priv = client->priv;
3751 GST_INFO ("client %p: watch destroyed", client);
3753 /* remove all sessions and so drop the extra client ref */
3754 gst_rtsp_client_session_filter (client, cleanup_session, NULL);
3755 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3756 g_object_unref (client);
3760 * gst_rtsp_client_attach:
3761 * @client: a #GstRTSPClient
3762 * @context: (allow-none): a #GMainContext
3764 * Attaches @client to @context. When the mainloop for @context is run, the
3765 * client will be dispatched. When @context is %NULL, the default context will be
3768 * This function should be called when the client properties and urls are fully
3769 * configured and the client is ready to start.
3771 * Returns: the ID (greater than 0) for the source within the GMainContext.
3774 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3776 GstRTSPClientPrivate *priv;
3779 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3780 priv = client->priv;
3781 g_return_val_if_fail (priv->connection != NULL, 0);
3782 g_return_val_if_fail (priv->watch == NULL, 0);
3784 /* make sure noone will free the context before the watch is destroyed */
3785 priv->watch_context = g_main_context_ref (context);
3787 /* create watch for the connection and attach */
3788 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3789 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3790 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3791 (GDestroyNotify) gst_rtsp_watch_unref);
3793 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
3795 GST_INFO ("client %p: attaching to context %p", client, context);
3796 res = gst_rtsp_watch_attach (priv->watch, context);
3802 * gst_rtsp_client_session_filter:
3803 * @client: a #GstRTSPClient
3804 * @func: (scope call) (allow-none): a callback
3805 * @user_data: user data passed to @func
3807 * Call @func for each session managed by @client. The result value of @func
3808 * determines what happens to the session. @func will be called with @client
3809 * locked so no further actions on @client can be performed from @func.
3811 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3814 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3816 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3817 * will also be added with an additional ref to the result #GList of this
3820 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3822 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3823 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3824 * element in the #GList should be unreffed before the list is freed.
3827 gst_rtsp_client_session_filter (GstRTSPClient * client,
3828 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3830 GstRTSPClientPrivate *priv;
3831 GList *result, *walk, *next;
3832 GHashTable *visited;
3835 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3837 priv = client->priv;
3841 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3843 g_mutex_lock (&priv->lock);
3845 cookie = priv->sessions_cookie;
3846 for (walk = priv->sessions; walk; walk = next) {
3847 GstRTSPSession *sess = walk->data;
3848 GstRTSPFilterResult res;
3851 next = g_list_next (walk);
3854 /* only visit each session once */
3855 if (g_hash_table_contains (visited, sess))
3858 g_hash_table_add (visited, g_object_ref (sess));
3859 g_mutex_unlock (&priv->lock);
3861 res = func (client, sess, user_data);
3863 g_mutex_lock (&priv->lock);
3865 res = GST_RTSP_FILTER_REF;
3867 changed = (cookie != priv->sessions_cookie);
3870 case GST_RTSP_FILTER_REMOVE:
3871 /* stop watching the session and pretend it went away, if the list was
3872 * changed, we can't use the current list position, try to see if we
3873 * still have the session */
3874 client_unwatch_session (client, sess, changed ? NULL : walk);
3875 cookie = priv->sessions_cookie;
3877 case GST_RTSP_FILTER_REF:
3878 result = g_list_prepend (result, g_object_ref (sess));
3880 case GST_RTSP_FILTER_KEEP:
3887 g_mutex_unlock (&priv->lock);
3890 g_hash_table_unref (visited);