2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 * @short_description: A client connection state
22 * @see_also: #GstRTSPServer, #GstRTSPThreadPool
24 * The client object handles the connection with a client for as long as a TCP
27 * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
28 * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
29 * #GstRTSPAuth and #GstRTSPThreadPool from the server.
31 * The client connection should be configured with the #GstRTSPConnection using
32 * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
33 * using gst_rtsp_client_attach(). From then on the client will handle requests
36 * Use gst_rtsp_client_session_filter() to iterate or modify all the
37 * #GstRTSPSession objects managed by the client object.
39 * Last reviewed on 2013-07-11 (1.0.0)
45 #include <gst/sdp/gstmikey.h>
47 #include "rtsp-client.h"
49 #include "rtsp-params.h"
51 #define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
52 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
55 * send_lock, lock, tunnels_lock
58 struct _GstRTSPClientPrivate
60 GMutex lock; /* protects everything else */
63 GstRTSPConnection *connection;
65 GMainContext *watch_context;
70 GstRTSPClientSendFunc send_func; /* protected by send_lock */
71 gpointer send_data; /* protected by send_lock */
72 GDestroyNotify send_notify; /* protected by send_lock */
74 GstRTSPSessionPool *session_pool;
75 gulong session_removed_id;
76 GstRTSPMountPoints *mount_points;
78 GstRTSPThreadPool *thread_pool;
80 /* used to cache the media in the last requested DESCRIBE so that
81 * we can pick it up in the next SETUP immediately */
85 GHashTable *transports;
87 guint sessions_cookie;
89 gboolean drop_backlog;
92 static GMutex tunnels_lock;
93 static GHashTable *tunnels; /* protected by tunnels_lock */
95 /* FIXME make this configurable. We don't want to do this yet because it will
96 * be superceeded by a cache object later */
97 #define WATCH_BACKLOG_SIZE 100
99 #define DEFAULT_SESSION_POOL NULL
100 #define DEFAULT_MOUNT_POINTS NULL
101 #define DEFAULT_DROP_BACKLOG TRUE
116 SIGNAL_OPTIONS_REQUEST,
117 SIGNAL_DESCRIBE_REQUEST,
118 SIGNAL_SETUP_REQUEST,
120 SIGNAL_PAUSE_REQUEST,
121 SIGNAL_TEARDOWN_REQUEST,
122 SIGNAL_SET_PARAMETER_REQUEST,
123 SIGNAL_GET_PARAMETER_REQUEST,
124 SIGNAL_HANDLE_RESPONSE,
129 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
130 #define GST_CAT_DEFAULT rtsp_client_debug
132 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
134 static void gst_rtsp_client_get_property (GObject * object, guint propid,
135 GValue * value, GParamSpec * pspec);
136 static void gst_rtsp_client_set_property (GObject * object, guint propid,
137 const GValue * value, GParamSpec * pspec);
138 static void gst_rtsp_client_finalize (GObject * obj);
140 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
141 static gboolean default_configure_client_media (GstRTSPClient * client,
142 GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
143 static gboolean default_configure_client_transport (GstRTSPClient * client,
144 GstRTSPContext * ctx, GstRTSPTransport * ct);
145 static GstRTSPResult default_params_set (GstRTSPClient * client,
146 GstRTSPContext * ctx);
147 static GstRTSPResult default_params_get (GstRTSPClient * client,
148 GstRTSPContext * ctx);
149 static gchar *default_make_path_from_uri (GstRTSPClient * client,
150 const GstRTSPUrl * uri);
151 static void client_session_removed (GstRTSPSessionPool * pool,
152 GstRTSPSession * session, GstRTSPClient * client);
154 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
157 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
159 GObjectClass *gobject_class;
161 g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
163 gobject_class = G_OBJECT_CLASS (klass);
165 gobject_class->get_property = gst_rtsp_client_get_property;
166 gobject_class->set_property = gst_rtsp_client_set_property;
167 gobject_class->finalize = gst_rtsp_client_finalize;
169 klass->create_sdp = create_sdp;
170 klass->configure_client_media = default_configure_client_media;
171 klass->configure_client_transport = default_configure_client_transport;
172 klass->params_set = default_params_set;
173 klass->params_get = default_params_get;
174 klass->make_path_from_uri = default_make_path_from_uri;
176 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
177 g_param_spec_object ("session-pool", "Session Pool",
178 "The session pool to use for client session",
179 GST_TYPE_RTSP_SESSION_POOL,
180 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
182 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
183 g_param_spec_object ("mount-points", "Mount Points",
184 "The mount points to use for client session",
185 GST_TYPE_RTSP_MOUNT_POINTS,
186 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
188 g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
189 g_param_spec_boolean ("drop-backlog", "Drop Backlog",
190 "Drop data when the backlog queue is full",
191 DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
193 gst_rtsp_client_signals[SIGNAL_CLOSED] =
194 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
195 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
196 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
198 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
199 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
200 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
201 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
203 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
204 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
205 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
206 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
207 GST_TYPE_RTSP_CONTEXT);
209 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
210 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
212 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
213 GST_TYPE_RTSP_CONTEXT);
215 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
216 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
217 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
218 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
219 GST_TYPE_RTSP_CONTEXT);
221 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
222 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
224 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
225 GST_TYPE_RTSP_CONTEXT);
227 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
228 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
229 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
230 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
231 GST_TYPE_RTSP_CONTEXT);
233 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
234 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
236 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
237 GST_TYPE_RTSP_CONTEXT);
239 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
240 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
241 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
242 set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
243 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
245 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
246 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
248 get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
249 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
251 gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
252 g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
253 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
254 handle_response), NULL, NULL, g_cclosure_marshal_generic,
255 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
258 * GstRTSPClient::send-message:
259 * @client: The RTSP client
260 * @session: (type GstRtspServer.RTSPSession): The session
261 * @message: (type GstRtsp.RTSPMessage): The message
263 gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
264 g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
265 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
266 G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
269 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
270 g_mutex_init (&tunnels_lock);
272 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
276 gst_rtsp_client_init (GstRTSPClient * client)
278 GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
282 g_mutex_init (&priv->lock);
283 g_mutex_init (&priv->send_lock);
284 g_mutex_init (&priv->watch_lock);
286 priv->drop_backlog = DEFAULT_DROP_BACKLOG;
287 priv->transports = g_hash_table_new (g_direct_hash, g_direct_equal);
290 static GstRTSPFilterResult
291 filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
294 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
296 return GST_RTSP_FILTER_REMOVE;
300 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
302 GstRTSPClientPrivate *priv = client->priv;
304 g_mutex_lock (&priv->lock);
305 /* check if we already know about this session */
306 if (g_list_find (priv->sessions, session) == NULL) {
307 GST_INFO ("watching session %p", session);
309 priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
310 priv->sessions_cookie++;
312 /* connect removed session handler, it will be disconnected when the last
313 * session gets removed */
314 if (priv->session_removed_id == 0)
315 priv->session_removed_id = g_signal_connect_data (priv->session_pool,
316 "session-removed", G_CALLBACK (client_session_removed),
317 g_object_ref (client), (GClosureNotify) g_object_unref, 0);
319 g_mutex_unlock (&priv->lock);
324 /* should be called with lock */
326 client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
329 GstRTSPClientPrivate *priv = client->priv;
331 GST_INFO ("client %p: unwatch session %p", client, session);
334 link = g_list_find (priv->sessions, session);
339 priv->sessions = g_list_delete_link (priv->sessions, link);
340 priv->sessions_cookie++;
342 /* if this was the last session, disconnect the handler.
343 * This will also drop the extra client ref */
344 if (!priv->sessions) {
345 g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
346 priv->session_removed_id = 0;
349 /* unlink all media managed in this session */
350 gst_rtsp_session_filter (session, filter_session_media, client);
352 /* remove the session */
353 g_object_unref (session);
356 static GstRTSPFilterResult
357 cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
360 return GST_RTSP_FILTER_REMOVE;
363 /* A client is finalized when the connection is broken */
365 gst_rtsp_client_finalize (GObject * obj)
367 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
368 GstRTSPClientPrivate *priv = client->priv;
370 GST_INFO ("finalize client %p", client);
373 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
374 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
377 g_source_destroy ((GSource *) priv->watch);
379 if (priv->watch_context)
380 g_main_context_unref (priv->watch_context);
382 /* all sessions should have been removed by now. We keep a ref to
383 * the client object for the session removed handler. The ref is
384 * dropped when the last session is removed from the list. */
385 g_assert (priv->sessions == NULL);
386 g_assert (priv->session_removed_id == 0);
388 g_hash_table_unref (priv->transports);
390 if (priv->connection)
391 gst_rtsp_connection_free (priv->connection);
392 if (priv->session_pool) {
393 g_object_unref (priv->session_pool);
395 if (priv->mount_points)
396 g_object_unref (priv->mount_points);
398 g_object_unref (priv->auth);
399 if (priv->thread_pool)
400 g_object_unref (priv->thread_pool);
405 gst_rtsp_media_unprepare (priv->media);
406 g_object_unref (priv->media);
409 g_free (priv->server_ip);
410 g_mutex_clear (&priv->lock);
411 g_mutex_clear (&priv->send_lock);
412 g_mutex_clear (&priv->watch_lock);
414 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
418 gst_rtsp_client_get_property (GObject * object, guint propid,
419 GValue * value, GParamSpec * pspec)
421 GstRTSPClient *client = GST_RTSP_CLIENT (object);
422 GstRTSPClientPrivate *priv = client->priv;
425 case PROP_SESSION_POOL:
426 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
428 case PROP_MOUNT_POINTS:
429 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
431 case PROP_DROP_BACKLOG:
432 g_value_set_boolean (value, priv->drop_backlog);
435 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
440 gst_rtsp_client_set_property (GObject * object, guint propid,
441 const GValue * value, GParamSpec * pspec)
443 GstRTSPClient *client = GST_RTSP_CLIENT (object);
444 GstRTSPClientPrivate *priv = client->priv;
447 case PROP_SESSION_POOL:
448 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
450 case PROP_MOUNT_POINTS:
451 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
453 case PROP_DROP_BACKLOG:
454 g_mutex_lock (&priv->lock);
455 priv->drop_backlog = g_value_get_boolean (value);
456 g_mutex_unlock (&priv->lock);
459 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
464 * gst_rtsp_client_new:
466 * Create a new #GstRTSPClient instance.
468 * Returns: (transfer full): a new #GstRTSPClient
471 gst_rtsp_client_new (void)
473 GstRTSPClient *result;
475 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
481 send_message (GstRTSPClient * client, GstRTSPContext * ctx,
482 GstRTSPMessage * message, gboolean close)
484 GstRTSPClientPrivate *priv = client->priv;
486 gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
487 "GStreamer RTSP server");
489 /* remove any previous header */
490 gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
492 /* add the new session header for new session ids */
494 gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
495 gst_rtsp_session_get_header (ctx->session));
498 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
499 gst_rtsp_message_dump (message);
503 gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
505 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
508 g_mutex_lock (&priv->send_lock);
510 priv->send_func (client, message, close, priv->send_data);
511 g_mutex_unlock (&priv->send_lock);
513 gst_rtsp_message_unset (message);
517 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
518 GstRTSPContext * ctx)
520 gst_rtsp_message_init_response (ctx->response, code,
521 gst_rtsp_status_as_text (code), ctx->request);
525 send_message (client, ctx, ctx->response, FALSE);
529 send_option_not_supported_response (GstRTSPClient * client,
530 GstRTSPContext * ctx, const gchar * unsupported_options)
532 GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
534 gst_rtsp_message_init_response (ctx->response, code,
535 gst_rtsp_status_as_text (code), ctx->request);
537 if (unsupported_options != NULL) {
538 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
539 unsupported_options);
544 send_message (client, ctx, ctx->response, FALSE);
548 paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
550 if (path1 == NULL || path2 == NULL)
553 if (strlen (path1) != len2)
556 if (strncmp (path1, path2, len2))
562 /* this function is called to initially find the media for the DESCRIBE request
563 * but is cached for when the same client (without breaking the connection) is
564 * doing a setup for the exact same url. */
565 static GstRTSPMedia *
566 find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
569 GstRTSPClientPrivate *priv = client->priv;
570 GstRTSPMediaFactory *factory;
574 /* find the longest matching factory for the uri first */
575 if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
579 ctx->factory = factory;
581 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
582 goto no_factory_access;
584 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
590 path_len = strlen (path);
592 if (!paths_are_equal (priv->path, path, path_len)) {
593 GstRTSPThread *thread;
595 /* remove any previously cached values before we try to construct a new
601 gst_rtsp_media_unprepare (priv->media);
602 g_object_unref (priv->media);
606 /* prepare the media and add it to the pipeline */
607 if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
612 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
613 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
617 /* prepare the media */
618 if (!(gst_rtsp_media_prepare (media, thread)))
621 /* now keep track of the uri and the media */
622 priv->path = g_strndup (path, path_len);
625 /* we have seen this path before, used cached media */
628 GST_INFO ("reusing cached media %p for path %s", media, priv->path);
631 g_object_unref (factory);
635 g_object_ref (media);
642 GST_ERROR ("client %p: no factory for path %s", client, path);
643 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
648 GST_ERROR ("client %p: not authorized to see factory path %s", client,
650 /* error reply is already sent */
655 GST_ERROR ("client %p: not authorized for factory path %s", client, path);
656 /* error reply is already sent */
661 GST_ERROR ("client %p: can't create media", client);
662 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
663 g_object_unref (factory);
669 GST_ERROR ("client %p: can't create thread", client);
670 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
671 g_object_unref (media);
673 g_object_unref (factory);
679 GST_ERROR ("client %p: can't prepare media", client);
680 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
681 g_object_unref (media);
683 g_object_unref (factory);
690 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
692 GstRTSPClientPrivate *priv = client->priv;
693 GstRTSPMessage message = { 0 };
698 gst_rtsp_message_init_data (&message, channel);
700 /* FIXME, need some sort of iovec RTSPMessage here */
701 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
704 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
706 g_mutex_lock (&priv->send_lock);
708 priv->send_func (client, &message, FALSE, priv->send_data);
709 g_mutex_unlock (&priv->send_lock);
711 gst_rtsp_message_steal_body (&message, &data, &usize);
712 gst_buffer_unmap (buffer, &map_info);
714 gst_rtsp_message_unset (&message);
720 * gst_rtsp_client_close:
721 * @client: a #GstRTSPClient
723 * Close the connection of @client and remove all media it was managing.
728 gst_rtsp_client_close (GstRTSPClient * client)
730 GstRTSPClientPrivate *priv = client->priv;
731 const gchar *tunnelid;
733 GST_DEBUG ("client %p: closing connection", client);
735 if (priv->connection) {
736 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
737 g_mutex_lock (&tunnels_lock);
738 /* remove from tunnelids */
739 g_hash_table_remove (tunnels, tunnelid);
740 g_mutex_unlock (&tunnels_lock);
742 gst_rtsp_connection_close (priv->connection);
745 /* connection is now closed, destroy the watch which will also cause the
746 * closed signal to be emitted */
748 GST_DEBUG ("client %p: destroying watch", client);
749 g_source_destroy ((GSource *) priv->watch);
751 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
756 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
761 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
763 path = g_strdup (uri->abspath);
769 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
771 GstRTSPClientPrivate *priv = client->priv;
772 GstRTSPClientClass *klass;
773 GstRTSPSession *session;
774 GstRTSPSessionMedia *sessmedia;
775 GstRTSPStatusCode code;
778 gboolean keep_session;
783 session = ctx->session;
788 klass = GST_RTSP_CLIENT_GET_CLASS (client);
789 path = klass->make_path_from_uri (client, ctx->uri);
791 /* get a handle to the configuration of the media in the session */
792 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
796 /* only aggregate control for now.. */
797 if (path[matched] != '\0')
802 ctx->sessmedia = sessmedia;
804 /* we emit the signal before closing the connection */
805 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
808 /* make sure we unblock the backlog and don't accept new messages
810 if (priv->watch != NULL)
811 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
813 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
815 /* allow messages again so that we can send the reply */
816 if (priv->watch != NULL)
817 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
819 /* unmanage the media in the session, returns false if all media session
821 keep_session = gst_rtsp_session_release_media (session, sessmedia);
823 /* construct the response now */
824 code = GST_RTSP_STS_OK;
825 gst_rtsp_message_init_response (ctx->response, code,
826 gst_rtsp_status_as_text (code), ctx->request);
828 send_message (client, ctx, ctx->response, TRUE);
831 /* remove the session */
832 gst_rtsp_session_pool_remove (priv->session_pool, session);
840 GST_ERROR ("client %p: no session", client);
841 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
846 GST_ERROR ("client %p: no uri supplied", client);
847 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
852 GST_ERROR ("client %p: no media for uri", client);
853 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
859 GST_ERROR ("client %p: no aggregate path %s", client, path);
860 send_generic_response (client,
861 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
868 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
872 res = gst_rtsp_params_set (client, ctx);
878 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
882 res = gst_rtsp_params_get (client, ctx);
888 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
894 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
895 if (res != GST_RTSP_OK)
899 /* no body, keep-alive request */
900 send_generic_response (client, GST_RTSP_STS_OK, ctx);
902 /* there is a body, handle the params */
903 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
904 if (res != GST_RTSP_OK)
907 send_message (client, ctx, ctx->response, FALSE);
910 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
918 GST_ERROR ("client %p: bad request", client);
919 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
925 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
931 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
932 if (res != GST_RTSP_OK)
936 /* no body, keep-alive request */
937 send_generic_response (client, GST_RTSP_STS_OK, ctx);
939 /* there is a body, handle the params */
940 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
941 if (res != GST_RTSP_OK)
944 send_message (client, ctx, ctx->response, FALSE);
947 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
955 GST_ERROR ("client %p: bad request", client);
956 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
962 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
964 GstRTSPSession *session;
965 GstRTSPClientClass *klass;
966 GstRTSPSessionMedia *sessmedia;
967 GstRTSPStatusCode code;
968 GstRTSPState rtspstate;
972 if (!(session = ctx->session))
978 klass = GST_RTSP_CLIENT_GET_CLASS (client);
979 path = klass->make_path_from_uri (client, ctx->uri);
981 /* get a handle to the configuration of the media in the session */
982 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
986 if (path[matched] != '\0')
991 ctx->sessmedia = sessmedia;
993 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
994 /* the session state must be playing or recording */
995 if (rtspstate != GST_RTSP_STATE_PLAYING &&
996 rtspstate != GST_RTSP_STATE_RECORDING)
999 /* then pause sending */
1000 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1002 /* construct the response now */
1003 code = GST_RTSP_STS_OK;
1004 gst_rtsp_message_init_response (ctx->response, code,
1005 gst_rtsp_status_as_text (code), ctx->request);
1007 send_message (client, ctx, ctx->response, FALSE);
1009 /* the state is now READY */
1010 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1012 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1019 GST_ERROR ("client %p: no seesion", client);
1020 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1025 GST_ERROR ("client %p: no uri supplied", client);
1026 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1031 GST_ERROR ("client %p: no media for uri", client);
1032 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1038 GST_ERROR ("client %p: no aggregate path %s", client, path);
1039 send_generic_response (client,
1040 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1046 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1047 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1053 /* convert @url and @path to a URL used as a content base for the factory
1054 * located at @path */
1056 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1062 /* check for trailing '/' and append one */
1063 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1068 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1070 result = gst_rtsp_url_get_request_uri (&tmp);
1071 g_free (tmp.abspath);
1077 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1079 GstRTSPSession *session;
1080 GstRTSPClientClass *klass;
1081 GstRTSPSessionMedia *sessmedia;
1082 GstRTSPMedia *media;
1083 GstRTSPStatusCode code;
1086 GstRTSPTimeRange *range;
1088 GstRTSPState rtspstate;
1089 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1090 gchar *path, *rtpinfo;
1093 if (!(session = ctx->session))
1096 if (!(uri = ctx->uri))
1099 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1100 path = klass->make_path_from_uri (client, uri);
1102 /* get a handle to the configuration of the media in the session */
1103 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1107 if (path[matched] != '\0')
1112 ctx->sessmedia = sessmedia;
1113 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1115 /* the session state must be playing or ready */
1116 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1117 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1120 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1121 if (!gst_rtsp_media_unsuspend (media))
1122 goto unsuspend_failed;
1124 /* parse the range header if we have one */
1125 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1126 if (res == GST_RTSP_OK) {
1127 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1128 /* we have a range, seek to the position */
1130 gst_rtsp_media_seek (media, range);
1131 gst_rtsp_range_free (range);
1135 /* grab RTPInfo from the media now */
1136 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1138 /* construct the response now */
1139 code = GST_RTSP_STS_OK;
1140 gst_rtsp_message_init_response (ctx->response, code,
1141 gst_rtsp_status_as_text (code), ctx->request);
1143 /* add the RTP-Info header */
1145 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1149 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1151 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1153 send_message (client, ctx, ctx->response, FALSE);
1155 /* start playing after sending the response */
1156 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1158 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1160 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1167 GST_ERROR ("client %p: no session", client);
1168 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1173 GST_ERROR ("client %p: no uri supplied", client);
1174 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1179 GST_ERROR ("client %p: media not found", client);
1180 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1185 GST_ERROR ("client %p: no aggregate path %s", client, path);
1186 send_generic_response (client,
1187 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1193 GST_ERROR ("client %p: not PLAYING or READY", client);
1194 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1200 GST_ERROR ("client %p: unsuspend failed", client);
1201 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1207 do_keepalive (GstRTSPSession * session)
1209 GST_INFO ("keep session %p alive", session);
1210 gst_rtsp_session_touch (session);
1213 /* parse @transport and return a valid transport in @tr. only transports
1214 * supported by @stream are returned. Returns FALSE if no valid transport
1217 parse_transport (const char *transport, GstRTSPStream * stream,
1218 GstRTSPTransport * tr)
1225 gst_rtsp_transport_init (tr);
1227 GST_DEBUG ("parsing transports %s", transport);
1229 transports = g_strsplit (transport, ",", 0);
1231 /* loop through the transports, try to parse */
1232 for (i = 0; transports[i]; i++) {
1233 res = gst_rtsp_transport_parse (transports[i], tr);
1234 if (res != GST_RTSP_OK) {
1235 /* no valid transport, search some more */
1236 GST_WARNING ("could not parse transport %s", transports[i]);
1240 /* we have a transport, see if it's supported */
1241 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1242 GST_WARNING ("unsupported transport %s", transports[i]);
1246 /* we have a valid transport */
1247 GST_INFO ("found valid transport %s", transports[i]);
1252 gst_rtsp_transport_init (tr);
1254 g_strfreev (transports);
1260 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1261 GstRTSPStream * stream, GstRTSPContext * ctx)
1263 GstRTSPMessage *request = ctx->request;
1264 gchar *blocksize_str;
1266 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1267 &blocksize_str, 0) == GST_RTSP_OK) {
1271 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1272 if (end == blocksize_str)
1275 /* we don't want to change the mtu when this media
1276 * can be shared because it impacts other clients */
1277 if (gst_rtsp_media_is_shared (media))
1280 if (blocksize > G_MAXUINT)
1281 blocksize = G_MAXUINT;
1283 gst_rtsp_stream_set_mtu (stream, blocksize);
1291 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1292 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1298 default_configure_client_transport (GstRTSPClient * client,
1299 GstRTSPContext * ctx, GstRTSPTransport * ct)
1301 GstRTSPClientPrivate *priv = client->priv;
1303 /* we have a valid transport now, set the destination of the client. */
1304 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1305 gboolean use_client_settings;
1307 use_client_settings =
1308 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1310 if (ct->destination && use_client_settings) {
1311 GstRTSPAddress *addr;
1313 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1314 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1319 gst_rtsp_address_free (addr);
1321 GstRTSPAddress *addr;
1322 GSocketFamily family;
1324 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1326 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1330 g_free (ct->destination);
1331 ct->destination = g_strdup (addr->address);
1332 ct->port.min = addr->port;
1333 ct->port.max = addr->port + addr->n_ports - 1;
1334 ct->ttl = addr->ttl;
1336 gst_rtsp_address_free (addr);
1341 url = gst_rtsp_connection_get_url (priv->connection);
1342 g_free (ct->destination);
1343 ct->destination = g_strdup (url->host);
1345 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1347 GSocketAddress *addr;
1349 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1350 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1351 /* our read port is the sender port of client */
1352 ct->client_port.min =
1353 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1354 g_object_unref (addr);
1356 if ((addr = g_socket_get_local_address (sock, NULL))) {
1357 ct->server_port.max =
1358 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1359 g_object_unref (addr);
1361 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1362 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1363 /* our write port is the receiver port of client */
1364 ct->client_port.max =
1365 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1366 g_object_unref (addr);
1368 if ((addr = g_socket_get_local_address (sock, NULL))) {
1369 ct->server_port.min =
1370 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1371 g_object_unref (addr);
1373 /* check if the client selected channels for TCP */
1374 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1375 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1385 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1390 static GstRTSPTransport *
1391 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1392 GstRTSPTransport * ct)
1394 GstRTSPTransport *st;
1396 GSocketFamily family;
1398 /* prepare the server transport */
1399 gst_rtsp_transport_new (&st);
1401 st->trans = ct->trans;
1402 st->profile = ct->profile;
1403 st->lower_transport = ct->lower_transport;
1405 addr = g_inet_address_new_from_string (ct->destination);
1408 GST_ERROR ("failed to get inet addr from client destination");
1409 family = G_SOCKET_FAMILY_IPV4;
1411 family = g_inet_address_get_family (addr);
1412 g_object_unref (addr);
1416 switch (st->lower_transport) {
1417 case GST_RTSP_LOWER_TRANS_UDP:
1418 st->client_port = ct->client_port;
1419 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1421 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1422 st->port = ct->port;
1423 st->destination = g_strdup (ct->destination);
1426 case GST_RTSP_LOWER_TRANS_TCP:
1427 st->interleaved = ct->interleaved;
1428 st->client_port = ct->client_port;
1429 st->server_port = ct->server_port;
1434 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1439 #define AES_128_KEY_LEN 16
1440 #define AES_256_KEY_LEN 32
1442 #define HMAC_32_KEY_LEN 4
1443 #define HMAC_80_KEY_LEN 10
1446 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1448 const gchar *srtp_cipher;
1449 const gchar *srtp_auth;
1450 const GstMIKEYPayload *sp;
1453 /* loop over Security policy until we find one containing policy */
1455 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1458 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1462 /* the default ciphers */
1463 srtp_cipher = "aes-128-icm";
1464 srtp_auth = "hmac-sha1-80";
1466 /* now override the defaults with what is in the Security Policy */
1470 /* collect all the params and go over them */
1471 len = gst_mikey_payload_sp_get_n_params (sp);
1472 for (i = 0; i < len; i++) {
1473 const GstMIKEYPayloadSPParam *param =
1474 gst_mikey_payload_sp_get_param (sp, i);
1476 switch (param->type) {
1477 case GST_MIKEY_SP_SRTP_ENC_ALG:
1478 switch (param->val[0]) {
1480 srtp_cipher = "null";
1484 srtp_cipher = "aes-128-icm";
1490 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1491 switch (param->val[0]) {
1492 case AES_128_KEY_LEN:
1493 srtp_cipher = "aes-128-icm";
1495 case AES_256_KEY_LEN:
1496 srtp_cipher = "aes-256-icm";
1502 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1503 switch (param->val[0]) {
1509 srtp_auth = "hmac-sha1-80";
1515 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1516 switch (param->val[0]) {
1517 case HMAC_32_KEY_LEN:
1518 srtp_auth = "hmac-sha1-32";
1520 case HMAC_80_KEY_LEN:
1521 srtp_auth = "hmac-sha1-80";
1527 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1529 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1536 /* now configure the SRTP parameters */
1537 gst_caps_set_simple (caps,
1538 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1539 "srtp-auth", G_TYPE_STRING, srtp_auth,
1540 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1541 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1547 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1548 guint8 * data, gsize size)
1550 GstMIKEYMessage *msg;
1552 GstCaps *caps = NULL;
1553 GstMIKEYPayloadKEMAC *kemac;
1554 const GstMIKEYPayloadKeyData *pkd;
1557 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1558 * set of Crypto Sessions protected with the same master key.
1559 * In the context of SRTP, an RTP and its RTCP stream is part of a
1561 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1564 /* we can only handle SRTP crypto sessions for now */
1565 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1566 goto invalid_map_type;
1568 /* get the number of crypto sessions. This maps SSRC to its
1569 * security parameters */
1570 n_cs = gst_mikey_message_get_n_cs (msg);
1572 goto no_crypto_sessions;
1574 /* we also need keys */
1575 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1576 (msg, GST_MIKEY_PT_KEMAC, 0)))
1579 /* we don't support encrypted keys */
1580 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1581 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1582 goto unsupported_encryption;
1584 /* get Key data sub-payload */
1585 pkd = (const GstMIKEYPayloadKeyData *)
1586 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1589 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1592 /* go over all crypto sessions and create the security policy for each
1594 for (i = 0; i < n_cs; i++) {
1595 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1597 caps = gst_caps_new_simple ("application/x-srtp",
1598 "ssrc", G_TYPE_UINT, map->ssrc,
1599 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1600 mikey_apply_policy (caps, msg, map->policy);
1602 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1603 gst_caps_unref (caps);
1605 gst_mikey_message_unref (msg);
1612 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1617 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1618 goto cleanup_message;
1622 GST_DEBUG_OBJECT (client, "no crypto sessions");
1623 goto cleanup_message;
1627 GST_DEBUG_OBJECT (client, "no keys found");
1628 goto cleanup_message;
1630 unsupported_encryption:
1632 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1633 goto cleanup_message;
1637 gst_mikey_message_unref (msg);
1642 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1645 strip_chars (gchar * str)
1652 if (!IS_STRIP_CHAR (str[len]))
1656 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1657 memmove (str, s, len + 1);
1660 /* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1661 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1664 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1669 specs = g_strsplit (keymgmt, ",", 0);
1670 for (i = 0; specs[i]; i++) {
1673 split = g_strsplit (specs[i], ";", 0);
1674 for (j = 0; split[j]; j++) {
1675 g_strstrip (split[j]);
1676 if (g_str_has_prefix (split[j], "prot=")) {
1677 g_strstrip (split[j] + 5);
1678 if (!g_str_equal (split[j] + 5, "mikey"))
1680 GST_DEBUG ("found mikey");
1681 } else if (g_str_has_prefix (split[j], "uri=")) {
1682 strip_chars (split[j] + 4);
1683 GST_DEBUG ("found uri '%s'", split[j] + 4);
1684 } else if (g_str_has_prefix (split[j], "data=")) {
1687 strip_chars (split[j] + 5);
1688 GST_DEBUG ("found data '%s'", split[j] + 5);
1689 data = g_base64_decode_inplace (split[j] + 5, &size);
1690 handle_mikey_data (client, ctx, data, size);
1698 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1700 GstRTSPClientPrivate *priv = client->priv;
1703 gchar *transport, *keymgmt;
1704 GstRTSPTransport *ct, *st;
1705 GstRTSPStatusCode code;
1706 GstRTSPSession *session;
1707 GstRTSPStreamTransport *trans;
1709 GstRTSPSessionMedia *sessmedia;
1710 GstRTSPMedia *media;
1711 GstRTSPStream *stream;
1712 GstRTSPState rtspstate;
1713 GstRTSPClientClass *klass;
1714 gchar *path, *control;
1716 gboolean new_session = FALSE;
1722 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1723 path = klass->make_path_from_uri (client, uri);
1725 /* parse the transport */
1727 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1729 if (res != GST_RTSP_OK)
1732 /* we create the session after parsing stuff so that we don't make
1733 * a session for malformed requests */
1734 if (priv->session_pool == NULL)
1737 session = ctx->session;
1740 g_object_ref (session);
1741 /* get a handle to the configuration of the media in the session, this can
1742 * return NULL if this is a new url to manage in this session. */
1743 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1745 /* we need a new media configuration in this session */
1749 /* we have no session media, find one and manage it */
1750 if (sessmedia == NULL) {
1751 /* get a handle to the configuration of the media in the session */
1752 media = find_media (client, ctx, path, &matched);
1754 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1755 g_object_ref (media);
1757 goto media_not_found;
1759 /* no media, not found then */
1761 goto media_not_found_no_reply;
1763 if (path[matched] == '\0')
1764 goto control_not_found;
1766 /* path is what matched. */
1767 path[matched] = '\0';
1768 /* control is remainder */
1769 control = &path[matched + 1];
1771 /* find the stream now using the control part */
1772 stream = gst_rtsp_media_find_stream (media, control);
1774 goto stream_not_found;
1776 /* now we have a uri identifying a valid media and stream */
1777 ctx->stream = stream;
1780 if (session == NULL) {
1781 /* create a session if this fails we probably reached our session limit or
1783 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1784 goto service_unavailable;
1786 /* make sure this client is closed when the session is closed */
1787 client_watch_session (client, session);
1790 /* signal new session */
1791 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1794 ctx->session = session;
1797 if (!klass->configure_client_media (client, media, stream, ctx))
1798 goto configure_media_failed_no_reply;
1800 gst_rtsp_transport_new (&ct);
1802 /* parse and find a usable supported transport */
1803 if (!parse_transport (transport, stream, ct))
1804 goto unsupported_transports;
1806 /* update the client transport */
1807 if (!klass->configure_client_transport (client, ctx, ct))
1808 goto unsupported_client_transport;
1810 /* parse the keymgmt */
1811 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1812 &keymgmt, 0) == GST_RTSP_OK) {
1813 if (!handle_keymgmt (client, ctx, keymgmt))
1817 if (sessmedia == NULL) {
1818 /* manage the media in our session now, if not done already */
1819 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1820 /* if we stil have no media, error */
1821 if (sessmedia == NULL)
1822 goto sessmedia_unavailable;
1824 g_object_unref (media);
1827 ctx->sessmedia = sessmedia;
1829 /* set in the session media transport */
1830 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1832 /* configure the url used to set this transport, this we will use when
1833 * generating the response for the PLAY request */
1834 gst_rtsp_stream_transport_set_url (trans, uri);
1835 /* configure keepalive for this transport */
1836 gst_rtsp_stream_transport_set_keepalive (trans,
1837 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1839 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1840 /* our callbacks to send data on this TCP connection */
1841 gst_rtsp_stream_transport_set_callbacks (trans,
1842 (GstRTSPSendFunc) do_send_data,
1843 (GstRTSPSendFunc) do_send_data, client, NULL);
1845 g_hash_table_insert (priv->transports,
1846 GINT_TO_POINTER (ct->interleaved.min), trans);
1847 g_hash_table_insert (priv->transports,
1848 GINT_TO_POINTER (ct->interleaved.max), trans);
1851 /* create and serialize the server transport */
1852 st = make_server_transport (client, ctx, ct);
1853 trans_str = gst_rtsp_transport_as_text (st);
1854 gst_rtsp_transport_free (st);
1856 /* construct the response now */
1857 code = GST_RTSP_STS_OK;
1858 gst_rtsp_message_init_response (ctx->response, code,
1859 gst_rtsp_status_as_text (code), ctx->request);
1861 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1865 send_message (client, ctx, ctx->response, FALSE);
1867 /* update the state */
1868 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1869 switch (rtspstate) {
1870 case GST_RTSP_STATE_PLAYING:
1871 case GST_RTSP_STATE_RECORDING:
1872 case GST_RTSP_STATE_READY:
1873 /* no state change */
1876 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1879 g_object_unref (session);
1882 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1889 GST_ERROR ("client %p: no uri", client);
1890 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1895 GST_ERROR ("client %p: no transport", client);
1896 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1901 GST_ERROR ("client %p: no session pool configured", client);
1902 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1905 media_not_found_no_reply:
1907 GST_ERROR ("client %p: media '%s' not found", client, path);
1908 /* error reply is already sent */
1913 GST_ERROR ("client %p: media '%s' not found", client, path);
1914 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1919 GST_ERROR ("client %p: no control in path '%s'", client, path);
1920 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1921 g_object_unref (media);
1926 GST_ERROR ("client %p: stream '%s' not found", client, control);
1927 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1928 g_object_unref (media);
1931 service_unavailable:
1933 GST_ERROR ("client %p: can't create session", client);
1934 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1935 g_object_unref (media);
1938 sessmedia_unavailable:
1940 GST_ERROR ("client %p: can't create session media", client);
1941 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1942 g_object_unref (media);
1943 goto cleanup_session;
1945 configure_media_failed_no_reply:
1947 GST_ERROR ("client %p: configure_media failed", client);
1948 /* error reply is already sent */
1949 goto cleanup_session;
1951 unsupported_transports:
1953 GST_ERROR ("client %p: unsupported transports", client);
1954 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1955 goto cleanup_transport;
1957 unsupported_client_transport:
1959 GST_ERROR ("client %p: unsupported client transport", client);
1960 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1961 goto cleanup_transport;
1965 GST_ERROR ("client %p: keymgmt error", client);
1966 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
1967 goto cleanup_transport;
1971 gst_rtsp_transport_free (ct);
1974 gst_rtsp_session_pool_remove (priv->session_pool, session);
1975 g_object_unref (session);
1982 static GstSDPMessage *
1983 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1985 GstRTSPClientPrivate *priv = client->priv;
1990 gst_sdp_message_new (&sdp);
1992 /* some standard things first */
1993 gst_sdp_message_set_version (sdp, "0");
2000 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
2003 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2004 gst_sdp_message_set_information (sdp, "rtsp-server");
2005 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2006 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2007 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2008 gst_sdp_message_add_attribute (sdp, "control", "*");
2010 info.is_ipv6 = priv->is_ipv6;
2011 info.server_ip = priv->server_ip;
2013 /* create an SDP for the media object */
2014 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2022 GST_ERROR ("client %p: could not create SDP", client);
2023 gst_sdp_message_free (sdp);
2028 /* for the describe we must generate an SDP */
2030 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2032 GstRTSPClientPrivate *priv = client->priv;
2037 GstRTSPMedia *media;
2038 GstRTSPClientClass *klass;
2040 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2045 /* check what kind of format is accepted, we don't really do anything with it
2046 * and always return SDP for now. */
2051 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2053 if (res == GST_RTSP_ENOTIMPL)
2056 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2060 if (!priv->mount_points)
2061 goto no_mount_points;
2063 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2066 /* find the media object for the uri */
2067 if (!(media = find_media (client, ctx, path, NULL)))
2070 /* create an SDP for the media object on this client */
2071 if (!(sdp = klass->create_sdp (client, media)))
2074 /* we suspend after the describe */
2075 gst_rtsp_media_suspend (media);
2076 g_object_unref (media);
2078 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2079 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2081 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2084 /* content base for some clients that might screw up creating the setup uri */
2085 str = make_base_url (client, ctx->uri, path);
2088 GST_INFO ("adding content-base: %s", str);
2089 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2091 /* add SDP to the response body */
2092 str = gst_sdp_message_as_text (sdp);
2093 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2094 gst_sdp_message_free (sdp);
2096 send_message (client, ctx, ctx->response, FALSE);
2098 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2106 GST_ERROR ("client %p: no uri", client);
2107 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2112 GST_ERROR ("client %p: no mount points configured", client);
2113 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2118 GST_ERROR ("client %p: can't find path for url", client);
2119 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2124 GST_ERROR ("client %p: no media", client);
2126 /* error reply is already sent */
2131 GST_ERROR ("client %p: can't create SDP", client);
2132 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2134 g_object_unref (media);
2140 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2142 GstRTSPMethod options;
2145 options = GST_RTSP_DESCRIBE |
2150 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2152 str = gst_rtsp_options_as_text (options);
2154 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2155 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2157 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2160 send_message (client, ctx, ctx->response, FALSE);
2162 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2168 /* remove duplicate and trailing '/' */
2170 sanitize_uri (GstRTSPUrl * uri)
2174 gboolean have_slash, prev_slash;
2176 s = d = uri->abspath;
2177 len = strlen (uri->abspath);
2181 for (i = 0; i < len; i++) {
2182 have_slash = s[i] == '/';
2184 if (!have_slash || !prev_slash)
2186 prev_slash = have_slash;
2188 len = d - uri->abspath;
2189 /* don't remove the first slash if that's the only thing left */
2190 if (len > 1 && *(d - 1) == '/')
2195 /* is called when the session is removed from its session pool. */
2197 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
2198 GstRTSPClient * client)
2200 GstRTSPClientPrivate *priv = client->priv;
2202 GST_INFO ("client %p: session %p removed", client, session);
2204 g_mutex_lock (&priv->lock);
2205 client_unwatch_session (client, session, NULL);
2206 g_mutex_unlock (&priv->lock);
2209 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2210 * and also returns a newly-allocated string of (comma-separated) unsupported
2211 * options in the unsupported_reqs variable .
2213 * There may be multiple Require headers, but we must send one single
2214 * Unsupported header with all the unsupported options as response. If
2215 * an incoming Require header contained a comma-separated list of options
2216 * GstRtspConnection will already have split that list up into multiple
2219 * TODO: allow the application to decide what features are supported
2222 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2225 GPtrArray *arr = NULL;
2231 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2233 if (res == GST_RTSP_ENOTIMPL)
2237 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2239 g_ptr_array_add (arr, g_strdup (reqs));
2243 /* if we don't have any Require headers at all, all is fine */
2247 /* otherwise we've now processed at all the Require headers */
2248 g_ptr_array_add (arr, NULL);
2250 /* for now we don't commit to supporting anything, so will just report
2251 * all of the required options as unsupported */
2252 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2254 g_ptr_array_unref (arr);
2259 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2261 GstRTSPClientPrivate *priv = client->priv;
2262 GstRTSPMethod method;
2263 const gchar *uristr;
2264 GstRTSPUrl *uri = NULL;
2265 GstRTSPVersion version;
2267 GstRTSPSession *session = NULL;
2268 GstRTSPContext sctx = { NULL }, *ctx;
2269 GstRTSPMessage response = { 0 };
2270 gchar *unsupported_reqs = NULL;
2273 if (!(ctx = gst_rtsp_context_get_current ())) {
2275 ctx->auth = priv->auth;
2276 gst_rtsp_context_push_current (ctx);
2279 ctx->conn = priv->connection;
2280 ctx->client = client;
2281 ctx->request = request;
2282 ctx->response = &response;
2284 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2285 gst_rtsp_message_dump (request);
2288 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2290 GST_INFO ("client %p: received a request %s %s %s", client,
2291 gst_rtsp_method_as_text (method), uristr,
2292 gst_rtsp_version_as_text (version));
2294 /* we can only handle 1.0 requests */
2295 if (version != GST_RTSP_VERSION_1_0)
2298 ctx->method = method;
2300 /* we always try to parse the url first */
2301 if (strcmp (uristr, "*") == 0) {
2302 /* special case where we have * as uri, keep uri = NULL */
2303 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2304 /* check if the uristr is an absolute path <=> scheme and host information
2308 scheme = g_uri_parse_scheme (uristr);
2309 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2310 gchar *absolute_uristr = NULL;
2312 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2313 if (priv->server_ip == NULL) {
2314 GST_WARNING_OBJECT (client, "host information missing");
2319 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2321 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2322 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2323 g_free (absolute_uristr);
2326 g_free (absolute_uristr);
2333 /* get the session if there is any */
2334 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2335 if (res == GST_RTSP_OK) {
2336 if (priv->session_pool == NULL)
2339 /* we had a session in the request, find it again */
2340 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2341 goto session_not_found;
2343 /* we add the session to the client list of watched sessions. When a session
2344 * disappears because it times out, we will be notified. If all sessions are
2345 * gone, we will close the connection */
2346 client_watch_session (client, session);
2349 /* sanitize the uri */
2353 ctx->session = session;
2355 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2356 goto not_authorized;
2358 /* handle any 'Require' headers */
2359 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2360 goto unsupported_requirement;
2362 /* now see what is asked and dispatch to a dedicated handler */
2364 case GST_RTSP_OPTIONS:
2365 handle_options_request (client, ctx);
2367 case GST_RTSP_DESCRIBE:
2368 handle_describe_request (client, ctx);
2370 case GST_RTSP_SETUP:
2371 handle_setup_request (client, ctx);
2374 handle_play_request (client, ctx);
2376 case GST_RTSP_PAUSE:
2377 handle_pause_request (client, ctx);
2379 case GST_RTSP_TEARDOWN:
2380 handle_teardown_request (client, ctx);
2382 case GST_RTSP_SET_PARAMETER:
2383 handle_set_param_request (client, ctx);
2385 case GST_RTSP_GET_PARAMETER:
2386 handle_get_param_request (client, ctx);
2388 case GST_RTSP_ANNOUNCE:
2389 case GST_RTSP_RECORD:
2390 case GST_RTSP_REDIRECT:
2391 goto not_implemented;
2392 case GST_RTSP_INVALID:
2399 gst_rtsp_context_pop_current (ctx);
2401 g_object_unref (session);
2403 gst_rtsp_url_free (uri);
2409 GST_ERROR ("client %p: version %d not supported", client, version);
2410 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2416 GST_ERROR ("client %p: bad request", client);
2417 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2422 GST_ERROR ("client %p: no pool configured", client);
2423 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2428 GST_ERROR ("client %p: session not found", client);
2429 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2434 GST_ERROR ("client %p: not allowed", client);
2435 /* error reply is already sent */
2438 unsupported_requirement:
2440 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2442 send_option_not_supported_response (client, ctx, unsupported_reqs);
2443 g_free (unsupported_reqs);
2448 GST_ERROR ("client %p: method %d not implemented", client, method);
2449 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2456 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2458 GstRTSPClientPrivate *priv = client->priv;
2460 GstRTSPSession *session = NULL;
2461 GstRTSPContext sctx = { NULL }, *ctx;
2464 if (!(ctx = gst_rtsp_context_get_current ())) {
2466 ctx->auth = priv->auth;
2467 gst_rtsp_context_push_current (ctx);
2470 ctx->conn = priv->connection;
2471 ctx->client = client;
2472 ctx->request = NULL;
2474 ctx->method = GST_RTSP_INVALID;
2475 ctx->response = response;
2477 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2478 gst_rtsp_message_dump (response);
2481 GST_INFO ("client %p: received a response", client);
2483 /* get the session if there is any */
2485 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2486 if (res == GST_RTSP_OK) {
2487 if (priv->session_pool == NULL)
2490 /* we had a session in the request, find it again */
2491 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2492 goto session_not_found;
2494 /* we add the session to the client list of watched sessions. When a session
2495 * disappears because it times out, we will be notified. If all sessions are
2496 * gone, we will close the connection */
2497 client_watch_session (client, session);
2500 ctx->session = session;
2502 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2507 gst_rtsp_context_pop_current (ctx);
2509 g_object_unref (session);
2514 GST_ERROR ("client %p: no pool configured", client);
2519 GST_ERROR ("client %p: session not found", client);
2525 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2527 GstRTSPClientPrivate *priv = client->priv;
2533 GstRTSPStreamTransport *trans;
2535 /* find the stream for this message */
2536 res = gst_rtsp_message_parse_data (message, &channel);
2537 if (res != GST_RTSP_OK)
2540 gst_rtsp_message_steal_body (message, &data, &size);
2542 buffer = gst_buffer_new_wrapped (data, size);
2544 trans = g_hash_table_lookup (priv->transports, GINT_TO_POINTER (channel));
2546 /* dispatch to the stream based on the channel number */
2547 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
2549 gst_buffer_unref (buffer);
2554 * gst_rtsp_client_set_session_pool:
2555 * @client: a #GstRTSPClient
2556 * @pool: (transfer none): a #GstRTSPSessionPool
2558 * Set @pool as the sessionpool for @client which it will use to find
2559 * or allocate sessions. the sessionpool is usually inherited from the server
2560 * that created the client but can be overridden later.
2563 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2564 GstRTSPSessionPool * pool)
2566 GstRTSPSessionPool *old;
2567 GstRTSPClientPrivate *priv;
2569 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2571 priv = client->priv;
2574 g_object_ref (pool);
2576 g_mutex_lock (&priv->lock);
2577 old = priv->session_pool;
2578 priv->session_pool = pool;
2580 if (priv->session_removed_id) {
2581 g_signal_handler_disconnect (old, priv->session_removed_id);
2582 priv->session_removed_id = 0;
2584 g_mutex_unlock (&priv->lock);
2586 /* FIXME, should remove all sessions from the old pool for this client */
2588 g_object_unref (old);
2592 * gst_rtsp_client_get_session_pool:
2593 * @client: a #GstRTSPClient
2595 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2597 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2599 GstRTSPSessionPool *
2600 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2602 GstRTSPClientPrivate *priv;
2603 GstRTSPSessionPool *result;
2605 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2607 priv = client->priv;
2609 g_mutex_lock (&priv->lock);
2610 if ((result = priv->session_pool))
2611 g_object_ref (result);
2612 g_mutex_unlock (&priv->lock);
2618 * gst_rtsp_client_set_mount_points:
2619 * @client: a #GstRTSPClient
2620 * @mounts: (transfer none): a #GstRTSPMountPoints
2622 * Set @mounts as the mount points for @client which it will use to map urls
2623 * to media streams. These mount points are usually inherited from the server that
2624 * created the client but can be overriden later.
2627 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2628 GstRTSPMountPoints * mounts)
2630 GstRTSPClientPrivate *priv;
2631 GstRTSPMountPoints *old;
2633 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2635 priv = client->priv;
2638 g_object_ref (mounts);
2640 g_mutex_lock (&priv->lock);
2641 old = priv->mount_points;
2642 priv->mount_points = mounts;
2643 g_mutex_unlock (&priv->lock);
2646 g_object_unref (old);
2650 * gst_rtsp_client_get_mount_points:
2651 * @client: a #GstRTSPClient
2653 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2655 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2657 GstRTSPMountPoints *
2658 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2660 GstRTSPClientPrivate *priv;
2661 GstRTSPMountPoints *result;
2663 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2665 priv = client->priv;
2667 g_mutex_lock (&priv->lock);
2668 if ((result = priv->mount_points))
2669 g_object_ref (result);
2670 g_mutex_unlock (&priv->lock);
2676 * gst_rtsp_client_set_auth:
2677 * @client: a #GstRTSPClient
2678 * @auth: (transfer none): a #GstRTSPAuth
2680 * configure @auth to be used as the authentication manager of @client.
2683 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2685 GstRTSPClientPrivate *priv;
2688 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2690 priv = client->priv;
2693 g_object_ref (auth);
2695 g_mutex_lock (&priv->lock);
2698 g_mutex_unlock (&priv->lock);
2701 g_object_unref (old);
2706 * gst_rtsp_client_get_auth:
2707 * @client: a #GstRTSPClient
2709 * Get the #GstRTSPAuth used as the authentication manager of @client.
2711 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2715 gst_rtsp_client_get_auth (GstRTSPClient * client)
2717 GstRTSPClientPrivate *priv;
2718 GstRTSPAuth *result;
2720 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2722 priv = client->priv;
2724 g_mutex_lock (&priv->lock);
2725 if ((result = priv->auth))
2726 g_object_ref (result);
2727 g_mutex_unlock (&priv->lock);
2733 * gst_rtsp_client_set_thread_pool:
2734 * @client: a #GstRTSPClient
2735 * @pool: (transfer none): a #GstRTSPThreadPool
2737 * configure @pool to be used as the thread pool of @client.
2740 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2741 GstRTSPThreadPool * pool)
2743 GstRTSPClientPrivate *priv;
2744 GstRTSPThreadPool *old;
2746 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2748 priv = client->priv;
2751 g_object_ref (pool);
2753 g_mutex_lock (&priv->lock);
2754 old = priv->thread_pool;
2755 priv->thread_pool = pool;
2756 g_mutex_unlock (&priv->lock);
2759 g_object_unref (old);
2763 * gst_rtsp_client_get_thread_pool:
2764 * @client: a #GstRTSPClient
2766 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2768 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2772 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2774 GstRTSPClientPrivate *priv;
2775 GstRTSPThreadPool *result;
2777 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2779 priv = client->priv;
2781 g_mutex_lock (&priv->lock);
2782 if ((result = priv->thread_pool))
2783 g_object_ref (result);
2784 g_mutex_unlock (&priv->lock);
2790 * gst_rtsp_client_set_connection:
2791 * @client: a #GstRTSPClient
2792 * @conn: (transfer full): a #GstRTSPConnection
2794 * Set the #GstRTSPConnection of @client. This function takes ownership of
2797 * Returns: %TRUE on success.
2800 gst_rtsp_client_set_connection (GstRTSPClient * client,
2801 GstRTSPConnection * conn)
2803 GstRTSPClientPrivate *priv;
2804 GSocket *read_socket;
2805 GSocketAddress *address;
2807 GError *error = NULL;
2809 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2810 g_return_val_if_fail (conn != NULL, FALSE);
2812 priv = client->priv;
2814 read_socket = gst_rtsp_connection_get_read_socket (conn);
2816 if (!(address = g_socket_get_local_address (read_socket, &error)))
2819 g_free (priv->server_ip);
2820 /* keep the original ip that the client connected to */
2821 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2822 GInetAddress *iaddr;
2824 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2826 /* socket might be ipv6 but adress still ipv4 */
2827 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2828 priv->server_ip = g_inet_address_to_string (iaddr);
2829 g_object_unref (address);
2831 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2832 priv->server_ip = g_strdup ("unknown");
2835 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2836 priv->server_ip, priv->is_ipv6);
2838 url = gst_rtsp_connection_get_url (conn);
2839 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2841 priv->connection = conn;
2848 GST_ERROR ("could not get local address %s", error->message);
2849 g_error_free (error);
2855 * gst_rtsp_client_get_connection:
2856 * @client: a #GstRTSPClient
2858 * Get the #GstRTSPConnection of @client.
2860 * Returns: (transfer none): the #GstRTSPConnection of @client.
2861 * The connection object returned remains valid until the client is freed.
2864 gst_rtsp_client_get_connection (GstRTSPClient * client)
2866 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2868 return client->priv->connection;
2872 * gst_rtsp_client_set_send_func:
2873 * @client: a #GstRTSPClient
2874 * @func: (scope notified): a #GstRTSPClientSendFunc
2875 * @user_data: (closure): user data passed to @func
2876 * @notify: (allow-none): called when @user_data is no longer in use
2878 * Set @func as the callback that will be called when a new message needs to be
2879 * sent to the client. @user_data is passed to @func and @notify is called when
2880 * @user_data is no longer in use.
2882 * By default, the client will send the messages on the #GstRTSPConnection that
2883 * was configured with gst_rtsp_client_attach() was called.
2886 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2887 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2889 GstRTSPClientPrivate *priv;
2890 GDestroyNotify old_notify;
2893 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2895 priv = client->priv;
2897 g_mutex_lock (&priv->send_lock);
2898 priv->send_func = func;
2899 old_notify = priv->send_notify;
2900 old_data = priv->send_data;
2901 priv->send_notify = notify;
2902 priv->send_data = user_data;
2903 g_mutex_unlock (&priv->send_lock);
2906 old_notify (old_data);
2910 * gst_rtsp_client_handle_message:
2911 * @client: a #GstRTSPClient
2912 * @message: (transfer none): an #GstRTSPMessage
2914 * Let the client handle @message.
2916 * Returns: a #GstRTSPResult.
2919 gst_rtsp_client_handle_message (GstRTSPClient * client,
2920 GstRTSPMessage * message)
2922 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2923 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2925 switch (message->type) {
2926 case GST_RTSP_MESSAGE_REQUEST:
2927 handle_request (client, message);
2929 case GST_RTSP_MESSAGE_RESPONSE:
2930 handle_response (client, message);
2932 case GST_RTSP_MESSAGE_DATA:
2933 handle_data (client, message);
2942 * gst_rtsp_client_send_message:
2943 * @client: a #GstRTSPClient
2944 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
2945 * the message to or %NULL
2946 * @message: (transfer none): The #GstRTSPMessage to send
2948 * Send a message message to the remote end. @message must be a
2949 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2952 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2953 GstRTSPMessage * message)
2955 GstRTSPContext sctx = { NULL }
2957 GstRTSPClientPrivate *priv;
2959 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2960 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2961 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
2962 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
2964 priv = client->priv;
2966 if (!(ctx = gst_rtsp_context_get_current ())) {
2968 ctx->auth = priv->auth;
2969 gst_rtsp_context_push_current (ctx);
2972 ctx->conn = priv->connection;
2973 ctx->client = client;
2974 ctx->session = session;
2976 send_message (client, ctx, message, FALSE);
2979 gst_rtsp_context_pop_current (ctx);
2984 static GstRTSPResult
2985 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
2986 gboolean close, gpointer user_data)
2988 GstRTSPClientPrivate *priv = client->priv;
2996 /* send the response and store the seq number so we can wait until it's
2997 * written to the client to close the connection */
2999 gst_rtsp_watch_send_message (priv->watch, message,
3000 close ? &priv->close_seq : NULL);
3001 if (ret == GST_RTSP_OK)
3004 if (ret != GST_RTSP_ENOMEM)
3008 if (priv->drop_backlog)
3011 /* queue was full, wait for more space */
3012 GST_DEBUG_OBJECT (client, "waiting for backlog");
3013 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3014 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3015 } while (ret != GST_RTSP_EINTR);
3022 GST_DEBUG_OBJECT (client, "got error %d", ret);
3027 static GstRTSPResult
3028 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3031 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3034 static GstRTSPResult
3035 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3037 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3038 GstRTSPClientPrivate *priv = client->priv;
3040 if (priv->close_seq && priv->close_seq == cseq) {
3041 GST_INFO ("client %p: send close message", client);
3042 priv->close_seq = 0;
3043 gst_rtsp_client_close (client);
3049 static GstRTSPResult
3050 closed (GstRTSPWatch * watch, gpointer user_data)
3052 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3053 GstRTSPClientPrivate *priv = client->priv;
3054 const gchar *tunnelid;
3056 GST_INFO ("client %p: connection closed", client);
3058 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3059 g_mutex_lock (&tunnels_lock);
3060 /* remove from tunnelids */
3061 g_hash_table_remove (tunnels, tunnelid);
3062 g_mutex_unlock (&tunnels_lock);
3065 gst_rtsp_watch_set_flushing (watch, TRUE);
3066 g_mutex_lock (&priv->watch_lock);
3067 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3068 g_mutex_unlock (&priv->watch_lock);
3073 static GstRTSPResult
3074 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3076 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3079 str = gst_rtsp_strresult (result);
3080 GST_INFO ("client %p: received an error %s", client, str);
3086 static GstRTSPResult
3087 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3088 GstRTSPMessage * message, guint id, gpointer user_data)
3090 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3093 str = gst_rtsp_strresult (result);
3095 ("client %p: error when handling message %p with id %d: %s",
3096 client, message, id, str);
3103 remember_tunnel (GstRTSPClient * client)
3105 GstRTSPClientPrivate *priv = client->priv;
3106 const gchar *tunnelid;
3108 /* store client in the pending tunnels */
3109 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3110 if (tunnelid == NULL)
3113 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3115 /* we can't have two clients connecting with the same tunnelid */
3116 g_mutex_lock (&tunnels_lock);
3117 if (g_hash_table_lookup (tunnels, tunnelid))
3118 goto tunnel_existed;
3120 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3121 g_mutex_unlock (&tunnels_lock);
3128 GST_ERROR ("client %p: no tunnelid provided", client);
3133 g_mutex_unlock (&tunnels_lock);
3134 GST_ERROR ("client %p: tunnel session %s already existed", client,
3140 static GstRTSPResult
3141 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3143 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3144 GstRTSPClientPrivate *priv = client->priv;
3146 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3149 /* ignore error, it'll only be a problem when the client does a POST again */
3150 remember_tunnel (client);
3156 handle_tunnel (GstRTSPClient * client)
3158 GstRTSPClientPrivate *priv = client->priv;
3159 GstRTSPClient *oclient;
3160 GstRTSPClientPrivate *opriv;
3161 const gchar *tunnelid;
3163 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3164 if (tunnelid == NULL)
3167 /* check for previous tunnel */
3168 g_mutex_lock (&tunnels_lock);
3169 oclient = g_hash_table_lookup (tunnels, tunnelid);
3171 if (oclient == NULL) {
3172 /* no previous tunnel, remember tunnel */
3173 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3174 g_mutex_unlock (&tunnels_lock);
3176 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3177 client, priv->connection);
3179 /* merge both tunnels into the first client */
3180 /* remove the old client from the table. ref before because removing it will
3181 * remove the ref to it. */
3182 g_object_ref (oclient);
3183 g_hash_table_remove (tunnels, tunnelid);
3184 g_mutex_unlock (&tunnels_lock);
3186 opriv = oclient->priv;
3188 g_mutex_lock (&opriv->watch_lock);
3189 if (opriv->watch == NULL)
3192 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3193 oclient, opriv->connection, priv->connection);
3195 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3196 gst_rtsp_watch_reset (priv->watch);
3197 gst_rtsp_watch_reset (opriv->watch);
3198 g_mutex_unlock (&opriv->watch_lock);
3199 g_object_unref (oclient);
3201 /* the old client owns the tunnel now, the new one will be freed */
3202 g_source_destroy ((GSource *) priv->watch);
3204 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3212 GST_ERROR ("client %p: no tunnelid provided", client);
3217 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3218 g_mutex_unlock (&opriv->watch_lock);
3219 g_object_unref (oclient);
3224 static GstRTSPStatusCode
3225 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3227 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3229 GST_INFO ("client %p: tunnel get (connection %p)", client,
3230 client->priv->connection);
3232 if (!handle_tunnel (client)) {
3233 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3236 return GST_RTSP_STS_OK;
3239 static GstRTSPResult
3240 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3242 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3244 GST_INFO ("client %p: tunnel post (connection %p)", client,
3245 client->priv->connection);
3247 if (!handle_tunnel (client)) {
3248 return GST_RTSP_ERROR;
3254 static GstRTSPResult
3255 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3256 GstRTSPMessage * response, gpointer user_data)
3258 GstRTSPClientClass *klass;
3260 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3261 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3263 if (klass->tunnel_http_response) {
3264 klass->tunnel_http_response (client, request, response);
3270 static GstRTSPWatchFuncs watch_funcs = {
3279 tunnel_http_response
3283 client_watch_notify (GstRTSPClient * client)
3285 GstRTSPClientPrivate *priv = client->priv;
3287 GST_INFO ("client %p: watch destroyed", client);
3289 g_main_context_unref (priv->watch_context);
3290 priv->watch_context = NULL;
3291 /* remove all sessions and so drop the extra client ref */
3292 gst_rtsp_client_session_filter (client, cleanup_session, NULL);
3293 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3294 g_object_unref (client);
3298 * gst_rtsp_client_attach:
3299 * @client: a #GstRTSPClient
3300 * @context: (allow-none): a #GMainContext
3302 * Attaches @client to @context. When the mainloop for @context is run, the
3303 * client will be dispatched. When @context is %NULL, the default context will be
3306 * This function should be called when the client properties and urls are fully
3307 * configured and the client is ready to start.
3309 * Returns: the ID (greater than 0) for the source within the GMainContext.
3312 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3314 GstRTSPClientPrivate *priv;
3317 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3318 priv = client->priv;
3319 g_return_val_if_fail (priv->connection != NULL, 0);
3320 g_return_val_if_fail (priv->watch == NULL, 0);
3322 /* make sure noone will free the context before the watch is destroyed */
3323 priv->watch_context = g_main_context_ref (context);
3325 /* create watch for the connection and attach */
3326 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3327 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3328 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3329 (GDestroyNotify) gst_rtsp_watch_unref);
3331 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
3333 GST_INFO ("client %p: attaching to context %p", client, context);
3334 res = gst_rtsp_watch_attach (priv->watch, context);
3340 * gst_rtsp_client_session_filter:
3341 * @client: a #GstRTSPClient
3342 * @func: (scope call) (allow-none): a callback
3343 * @user_data: user data passed to @func
3345 * Call @func for each session managed by @client. The result value of @func
3346 * determines what happens to the session. @func will be called with @client
3347 * locked so no further actions on @client can be performed from @func.
3349 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3352 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3354 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3355 * will also be added with an additional ref to the result #GList of this
3358 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3360 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3361 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3362 * element in the #GList should be unreffed before the list is freed.
3365 gst_rtsp_client_session_filter (GstRTSPClient * client,
3366 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3368 GstRTSPClientPrivate *priv;
3369 GList *result, *walk, *next;
3370 GHashTable *visited;
3373 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3375 priv = client->priv;
3379 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3381 g_mutex_lock (&priv->lock);
3383 cookie = priv->sessions_cookie;
3384 for (walk = priv->sessions; walk; walk = next) {
3385 GstRTSPSession *sess = walk->data;
3386 GstRTSPFilterResult res;
3389 next = g_list_next (walk);
3392 /* only visit each session once */
3393 if (g_hash_table_contains (visited, sess))
3396 g_hash_table_add (visited, g_object_ref (sess));
3397 g_mutex_unlock (&priv->lock);
3399 res = func (client, sess, user_data);
3401 g_mutex_lock (&priv->lock);
3403 res = GST_RTSP_FILTER_REF;
3405 changed = (cookie != priv->sessions_cookie);
3408 case GST_RTSP_FILTER_REMOVE:
3409 /* stop watching the session and pretend it went away, if the list was
3410 * changed, we can't use the current list position, try to see if we
3411 * still have the session */
3412 client_unwatch_session (client, sess, changed ? NULL : walk);
3413 cookie = priv->sessions_cookie;
3415 case GST_RTSP_FILTER_REF:
3416 result = g_list_prepend (result, g_object_ref (sess));
3418 case GST_RTSP_FILTER_KEEP:
3425 g_mutex_unlock (&priv->lock);
3428 g_hash_table_unref (visited);