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 */
62 GstRTSPConnection *connection;
68 GstRTSPClientSendFunc send_func; /* protected by send_lock */
69 gpointer send_data; /* protected by send_lock */
70 GDestroyNotify send_notify; /* protected by send_lock */
72 GstRTSPSessionPool *session_pool;
73 GstRTSPMountPoints *mount_points;
75 GstRTSPThreadPool *thread_pool;
77 /* used to cache the media in the last requested DESCRIBE so that
78 * we can pick it up in the next SETUP immediately */
85 gboolean drop_backlog;
88 static GMutex tunnels_lock;
89 static GHashTable *tunnels; /* protected by tunnels_lock */
91 #define DEFAULT_SESSION_POOL NULL
92 #define DEFAULT_MOUNT_POINTS NULL
93 #define DEFAULT_DROP_BACKLOG TRUE
108 SIGNAL_OPTIONS_REQUEST,
109 SIGNAL_DESCRIBE_REQUEST,
110 SIGNAL_SETUP_REQUEST,
112 SIGNAL_PAUSE_REQUEST,
113 SIGNAL_TEARDOWN_REQUEST,
114 SIGNAL_SET_PARAMETER_REQUEST,
115 SIGNAL_GET_PARAMETER_REQUEST,
116 SIGNAL_HANDLE_RESPONSE,
121 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
122 #define GST_CAT_DEFAULT rtsp_client_debug
124 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
126 static void gst_rtsp_client_get_property (GObject * object, guint propid,
127 GValue * value, GParamSpec * pspec);
128 static void gst_rtsp_client_set_property (GObject * object, guint propid,
129 const GValue * value, GParamSpec * pspec);
130 static void gst_rtsp_client_finalize (GObject * obj);
132 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
133 static void client_session_finalized (GstRTSPClient * client,
134 GstRTSPSession * session);
135 static void unlink_session_transports (GstRTSPClient * client,
136 GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
137 static gboolean default_configure_client_media (GstRTSPClient * client,
138 GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
139 static gboolean default_configure_client_transport (GstRTSPClient * client,
140 GstRTSPContext * ctx, GstRTSPTransport * ct);
141 static GstRTSPResult default_params_set (GstRTSPClient * client,
142 GstRTSPContext * ctx);
143 static GstRTSPResult default_params_get (GstRTSPClient * client,
144 GstRTSPContext * ctx);
145 static gchar *default_make_path_from_uri (GstRTSPClient * client,
146 const GstRTSPUrl * uri);
148 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
151 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
153 GObjectClass *gobject_class;
155 g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
157 gobject_class = G_OBJECT_CLASS (klass);
159 gobject_class->get_property = gst_rtsp_client_get_property;
160 gobject_class->set_property = gst_rtsp_client_set_property;
161 gobject_class->finalize = gst_rtsp_client_finalize;
163 klass->create_sdp = create_sdp;
164 klass->configure_client_media = default_configure_client_media;
165 klass->configure_client_transport = default_configure_client_transport;
166 klass->params_set = default_params_set;
167 klass->params_get = default_params_get;
168 klass->make_path_from_uri = default_make_path_from_uri;
170 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
171 g_param_spec_object ("session-pool", "Session Pool",
172 "The session pool to use for client session",
173 GST_TYPE_RTSP_SESSION_POOL,
174 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
176 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
177 g_param_spec_object ("mount-points", "Mount Points",
178 "The mount points to use for client session",
179 GST_TYPE_RTSP_MOUNT_POINTS,
180 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
182 g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
183 g_param_spec_boolean ("drop-backlog", "Drop Backlog",
184 "Drop data when the backlog queue is full",
185 DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
187 gst_rtsp_client_signals[SIGNAL_CLOSED] =
188 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
189 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
190 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
192 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
193 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
194 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
195 g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
197 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
198 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
199 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
200 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
203 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
204 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
205 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
206 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
209 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
210 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
212 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
215 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
216 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
217 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
218 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
221 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
222 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
224 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
227 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
228 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
229 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
230 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
233 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
234 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
236 set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
237 G_TYPE_NONE, 1, G_TYPE_POINTER);
239 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
240 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
241 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
242 get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
243 G_TYPE_NONE, 1, G_TYPE_POINTER);
245 gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
246 g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
248 handle_response), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
249 G_TYPE_NONE, 1, G_TYPE_POINTER);
251 gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
252 g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
253 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
254 G_TYPE_NONE, 2, G_TYPE_POINTER, G_TYPE_POINTER);
257 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
258 g_mutex_init (&tunnels_lock);
260 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
264 gst_rtsp_client_init (GstRTSPClient * client)
266 GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
270 g_mutex_init (&priv->lock);
271 g_mutex_init (&priv->send_lock);
273 priv->drop_backlog = DEFAULT_DROP_BACKLOG;
276 static GstRTSPFilterResult
277 filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
280 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
282 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
283 unlink_session_transports (client, sess, sessmedia);
285 /* unmanage the media in the session */
286 return GST_RTSP_FILTER_REMOVE;
290 client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
292 /* unlink all media managed in this session */
293 gst_rtsp_session_filter (session, filter_session, client);
297 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
299 GstRTSPClientPrivate *priv = client->priv;
302 for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
303 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
305 /* we already know about this session */
306 if (msession == session)
310 GST_INFO ("watching session %p", session);
312 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
314 priv->sessions = g_list_prepend (priv->sessions, session);
318 client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session)
320 GstRTSPClientPrivate *priv = client->priv;
322 GST_INFO ("unwatching session %p", session);
324 g_object_weak_unref (G_OBJECT (session),
325 (GWeakNotify) client_session_finalized, client);
326 priv->sessions = g_list_remove (priv->sessions, session);
330 client_cleanup_session (GstRTSPClient * client, GstRTSPSession * session)
332 g_object_weak_unref (G_OBJECT (session),
333 (GWeakNotify) client_session_finalized, client);
334 client_unlink_session (client, session);
338 client_cleanup_sessions (GstRTSPClient * client)
340 GstRTSPClientPrivate *priv = client->priv;
343 /* remove weak-ref from sessions */
344 for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
345 client_cleanup_session (client, (GstRTSPSession *) sessions->data);
347 g_list_free (priv->sessions);
348 priv->sessions = NULL;
351 /* A client is finalized when the connection is broken */
353 gst_rtsp_client_finalize (GObject * obj)
355 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
356 GstRTSPClientPrivate *priv = client->priv;
358 GST_INFO ("finalize client %p", client);
361 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
362 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
365 g_source_destroy ((GSource *) priv->watch);
367 client_cleanup_sessions (client);
369 if (priv->connection)
370 gst_rtsp_connection_free (priv->connection);
371 if (priv->session_pool)
372 g_object_unref (priv->session_pool);
373 if (priv->mount_points)
374 g_object_unref (priv->mount_points);
376 g_object_unref (priv->auth);
377 if (priv->thread_pool)
378 g_object_unref (priv->thread_pool);
383 gst_rtsp_media_unprepare (priv->media);
384 g_object_unref (priv->media);
387 g_free (priv->server_ip);
388 g_mutex_clear (&priv->lock);
389 g_mutex_clear (&priv->send_lock);
391 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
395 gst_rtsp_client_get_property (GObject * object, guint propid,
396 GValue * value, GParamSpec * pspec)
398 GstRTSPClient *client = GST_RTSP_CLIENT (object);
399 GstRTSPClientPrivate *priv = client->priv;
402 case PROP_SESSION_POOL:
403 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
405 case PROP_MOUNT_POINTS:
406 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
408 case PROP_DROP_BACKLOG:
409 g_value_set_boolean (value, priv->drop_backlog);
412 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
417 gst_rtsp_client_set_property (GObject * object, guint propid,
418 const GValue * value, GParamSpec * pspec)
420 GstRTSPClient *client = GST_RTSP_CLIENT (object);
421 GstRTSPClientPrivate *priv = client->priv;
424 case PROP_SESSION_POOL:
425 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
427 case PROP_MOUNT_POINTS:
428 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
430 case PROP_DROP_BACKLOG:
431 g_mutex_lock (&priv->lock);
432 priv->drop_backlog = g_value_get_boolean (value);
433 g_mutex_unlock (&priv->lock);
436 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
441 * gst_rtsp_client_new:
443 * Create a new #GstRTSPClient instance.
445 * Returns: (transfer full): a new #GstRTSPClient
448 gst_rtsp_client_new (void)
450 GstRTSPClient *result;
452 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
458 send_message (GstRTSPClient * client, GstRTSPContext * ctx,
459 GstRTSPMessage * message, gboolean close)
461 GstRTSPClientPrivate *priv = client->priv;
463 gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
464 "GStreamer RTSP server");
466 /* remove any previous header */
467 gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
469 /* add the new session header for new session ids */
471 gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
472 gst_rtsp_session_get_header (ctx->session));
475 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
476 gst_rtsp_message_dump (message);
480 gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
482 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
485 g_mutex_lock (&priv->send_lock);
487 priv->send_func (client, message, close, priv->send_data);
488 g_mutex_unlock (&priv->send_lock);
490 gst_rtsp_message_unset (message);
494 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
495 GstRTSPContext * ctx)
497 gst_rtsp_message_init_response (ctx->response, code,
498 gst_rtsp_status_as_text (code), ctx->request);
502 send_message (client, ctx, ctx->response, FALSE);
506 send_option_not_supported_response (GstRTSPClient * client,
507 GstRTSPContext * ctx, const gchar * unsupported_options)
509 GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
511 gst_rtsp_message_init_response (ctx->response, code,
512 gst_rtsp_status_as_text (code), ctx->request);
514 if (unsupported_options != NULL) {
515 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
516 unsupported_options);
521 send_message (client, ctx, ctx->response, FALSE);
525 paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
527 if (path1 == NULL || path2 == NULL)
530 if (strlen (path1) != len2)
533 if (strncmp (path1, path2, len2))
539 /* this function is called to initially find the media for the DESCRIBE request
540 * but is cached for when the same client (without breaking the connection) is
541 * doing a setup for the exact same url. */
542 static GstRTSPMedia *
543 find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
546 GstRTSPClientPrivate *priv = client->priv;
547 GstRTSPMediaFactory *factory;
551 /* find the longest matching factory for the uri first */
552 if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
556 ctx->factory = factory;
558 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
559 goto no_factory_access;
561 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
567 path_len = strlen (path);
569 if (!paths_are_equal (priv->path, path, path_len)) {
570 GstRTSPThread *thread;
572 /* remove any previously cached values before we try to construct a new
578 gst_rtsp_media_unprepare (priv->media);
579 g_object_unref (priv->media);
583 /* prepare the media and add it to the pipeline */
584 if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
589 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
590 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
594 /* prepare the media */
595 if (!(gst_rtsp_media_prepare (media, thread)))
598 /* now keep track of the uri and the media */
599 priv->path = g_strndup (path, path_len);
602 /* we have seen this path before, used cached media */
605 GST_INFO ("reusing cached media %p for path %s", media, priv->path);
608 g_object_unref (factory);
612 g_object_ref (media);
619 GST_ERROR ("client %p: no factory for path %s", client, path);
620 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
625 GST_ERROR ("client %p: not authorized to see factory path %s", client,
627 /* error reply is already sent */
632 GST_ERROR ("client %p: not authorized for factory path %s", client, path);
633 /* error reply is already sent */
638 GST_ERROR ("client %p: can't create media", client);
639 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
640 g_object_unref (factory);
646 GST_ERROR ("client %p: can't create thread", client);
647 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
648 g_object_unref (media);
650 g_object_unref (factory);
656 GST_ERROR ("client %p: can't prepare media", client);
657 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
658 g_object_unref (media);
660 g_object_unref (factory);
667 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
669 GstRTSPClientPrivate *priv = client->priv;
670 GstRTSPMessage message = { 0 };
675 gst_rtsp_message_init_data (&message, channel);
677 /* FIXME, need some sort of iovec RTSPMessage here */
678 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
681 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
683 g_mutex_lock (&priv->send_lock);
685 priv->send_func (client, &message, FALSE, priv->send_data);
686 g_mutex_unlock (&priv->send_lock);
688 gst_rtsp_message_steal_body (&message, &data, &usize);
689 gst_buffer_unmap (buffer, &map_info);
691 gst_rtsp_message_unset (&message);
697 link_transport (GstRTSPClient * client, GstRTSPSession * session,
698 GstRTSPStreamTransport * trans)
700 GstRTSPClientPrivate *priv = client->priv;
702 GST_DEBUG ("client %p: linking transport %p", client, trans);
704 gst_rtsp_stream_transport_set_callbacks (trans,
705 (GstRTSPSendFunc) do_send_data,
706 (GstRTSPSendFunc) do_send_data, client, NULL);
708 priv->transports = g_list_prepend (priv->transports, trans);
710 /* make sure our session can't expire */
711 gst_rtsp_session_prevent_expire (session);
715 link_session_transports (GstRTSPClient * client, GstRTSPSession * session,
716 GstRTSPSessionMedia * sessmedia)
721 gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
722 for (i = 0; i < n_streams; i++) {
723 GstRTSPStreamTransport *trans;
724 const GstRTSPTransport *tr;
726 /* get the transport, if there is no transport configured, skip this stream */
727 trans = gst_rtsp_session_media_get_transport (sessmedia, i);
731 tr = gst_rtsp_stream_transport_get_transport (trans);
733 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
734 /* for TCP, link the stream to the TCP connection of the client */
735 link_transport (client, session, trans);
741 unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
742 GstRTSPStreamTransport * trans)
744 GstRTSPClientPrivate *priv = client->priv;
746 GST_DEBUG ("client %p: unlinking transport %p", client, trans);
748 gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
750 priv->transports = g_list_remove (priv->transports, trans);
752 /* our session can now expire */
753 gst_rtsp_session_allow_expire (session);
757 unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
758 GstRTSPSessionMedia * sessmedia)
763 gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
764 for (i = 0; i < n_streams; i++) {
765 GstRTSPStreamTransport *trans;
766 const GstRTSPTransport *tr;
768 /* get the transport, if there is no transport configured, skip this stream */
769 trans = gst_rtsp_session_media_get_transport (sessmedia, i);
773 tr = gst_rtsp_stream_transport_get_transport (trans);
775 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
776 /* for TCP, unlink the stream from the TCP connection of the client */
777 unlink_transport (client, session, trans);
783 close_connection (GstRTSPClient * client)
785 GstRTSPClientPrivate *priv = client->priv;
786 const gchar *tunnelid;
788 GST_DEBUG ("client %p: closing connection", client);
790 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
791 g_mutex_lock (&tunnels_lock);
792 /* remove from tunnelids */
793 g_hash_table_remove (tunnels, tunnelid);
794 g_mutex_unlock (&tunnels_lock);
797 gst_rtsp_connection_close (priv->connection);
801 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
806 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
808 path = g_strdup (uri->abspath);
814 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
816 GstRTSPClientPrivate *priv = client->priv;
817 GstRTSPClientClass *klass;
818 GstRTSPSession *session;
819 GstRTSPSessionMedia *sessmedia;
820 GstRTSPStatusCode code;
827 session = ctx->session;
832 klass = GST_RTSP_CLIENT_GET_CLASS (client);
833 path = klass->make_path_from_uri (client, ctx->uri);
835 /* get a handle to the configuration of the media in the session */
836 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
840 /* only aggregate control for now.. */
841 if (path[matched] != '\0')
846 ctx->sessmedia = sessmedia;
848 /* we emit the signal before closing the connection */
849 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
852 /* make sure we unblock the backlog and don't accept new messages
854 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
856 /* unlink the all TCP callbacks */
857 unlink_session_transports (client, session, sessmedia);
859 /* remove the session from the watched sessions */
860 client_unwatch_session (client, session);
862 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
864 /* allow messages again so that we can send the reply */
865 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
867 /* unmanage the media in the session, returns false if all media session
869 if (!gst_rtsp_session_release_media (session, sessmedia)) {
870 /* remove the session */
871 gst_rtsp_session_pool_remove (priv->session_pool, session);
873 /* construct the response now */
874 code = GST_RTSP_STS_OK;
875 gst_rtsp_message_init_response (ctx->response, code,
876 gst_rtsp_status_as_text (code), ctx->request);
878 send_message (client, ctx, ctx->response, TRUE);
885 GST_ERROR ("client %p: no session", client);
886 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
891 GST_ERROR ("client %p: no uri supplied", client);
892 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
897 GST_ERROR ("client %p: no media for uri", client);
898 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
904 GST_ERROR ("client %p: no aggregate path %s", client, path);
905 send_generic_response (client,
906 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
913 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
917 res = gst_rtsp_params_set (client, ctx);
923 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
927 res = gst_rtsp_params_get (client, ctx);
933 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
939 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
940 if (res != GST_RTSP_OK)
944 /* no body, keep-alive request */
945 send_generic_response (client, GST_RTSP_STS_OK, ctx);
947 /* there is a body, handle the params */
948 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
949 if (res != GST_RTSP_OK)
952 send_message (client, ctx, ctx->response, FALSE);
955 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
963 GST_ERROR ("client %p: bad request", client);
964 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
970 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
976 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
977 if (res != GST_RTSP_OK)
981 /* no body, keep-alive request */
982 send_generic_response (client, GST_RTSP_STS_OK, ctx);
984 /* there is a body, handle the params */
985 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
986 if (res != GST_RTSP_OK)
989 send_message (client, ctx, ctx->response, FALSE);
992 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
1000 GST_ERROR ("client %p: bad request", client);
1001 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1007 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
1009 GstRTSPSession *session;
1010 GstRTSPClientClass *klass;
1011 GstRTSPSessionMedia *sessmedia;
1012 GstRTSPStatusCode code;
1013 GstRTSPState rtspstate;
1017 if (!(session = ctx->session))
1023 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1024 path = klass->make_path_from_uri (client, ctx->uri);
1026 /* get a handle to the configuration of the media in the session */
1027 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1031 if (path[matched] != '\0')
1036 ctx->sessmedia = sessmedia;
1038 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1039 /* the session state must be playing or recording */
1040 if (rtspstate != GST_RTSP_STATE_PLAYING &&
1041 rtspstate != GST_RTSP_STATE_RECORDING)
1044 /* unlink the all TCP callbacks */
1045 unlink_session_transports (client, session, sessmedia);
1047 /* then pause sending */
1048 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1050 /* construct the response now */
1051 code = GST_RTSP_STS_OK;
1052 gst_rtsp_message_init_response (ctx->response, code,
1053 gst_rtsp_status_as_text (code), ctx->request);
1055 send_message (client, ctx, ctx->response, FALSE);
1057 /* the state is now READY */
1058 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1060 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1067 GST_ERROR ("client %p: no seesion", client);
1068 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1073 GST_ERROR ("client %p: no uri supplied", client);
1074 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1079 GST_ERROR ("client %p: no media for uri", client);
1080 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1086 GST_ERROR ("client %p: no aggregate path %s", client, path);
1087 send_generic_response (client,
1088 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1094 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1095 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1101 /* convert @url and @path to a URL used as a content base for the factory
1102 * located at @path */
1104 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1110 /* check for trailing '/' and append one */
1111 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1116 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1118 result = gst_rtsp_url_get_request_uri (&tmp);
1119 g_free (tmp.abspath);
1125 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1127 GstRTSPSession *session;
1128 GstRTSPClientClass *klass;
1129 GstRTSPSessionMedia *sessmedia;
1130 GstRTSPMedia *media;
1131 GstRTSPStatusCode code;
1134 GstRTSPTimeRange *range;
1136 GstRTSPState rtspstate;
1137 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1138 gchar *path, *rtpinfo;
1141 if (!(session = ctx->session))
1144 if (!(uri = ctx->uri))
1147 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1148 path = klass->make_path_from_uri (client, uri);
1150 /* get a handle to the configuration of the media in the session */
1151 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1155 if (path[matched] != '\0')
1160 ctx->sessmedia = sessmedia;
1161 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1163 /* the session state must be playing or ready */
1164 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1165 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1168 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1169 if (!gst_rtsp_media_unsuspend (media))
1170 goto unsuspend_failed;
1172 /* parse the range header if we have one */
1173 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1174 if (res == GST_RTSP_OK) {
1175 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1176 /* we have a range, seek to the position */
1178 gst_rtsp_media_seek (media, range);
1179 gst_rtsp_range_free (range);
1183 /* link the all TCP callbacks */
1184 link_session_transports (client, session, sessmedia);
1186 /* grab RTPInfo from the media now */
1187 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1189 /* construct the response now */
1190 code = GST_RTSP_STS_OK;
1191 gst_rtsp_message_init_response (ctx->response, code,
1192 gst_rtsp_status_as_text (code), ctx->request);
1194 /* add the RTP-Info header */
1196 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1200 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1202 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1204 send_message (client, ctx, ctx->response, FALSE);
1206 /* start playing after sending the response */
1207 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1209 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1211 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1218 GST_ERROR ("client %p: no session", client);
1219 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1224 GST_ERROR ("client %p: no uri supplied", client);
1225 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1230 GST_ERROR ("client %p: media not found", client);
1231 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1236 GST_ERROR ("client %p: no aggregate path %s", client, path);
1237 send_generic_response (client,
1238 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1244 GST_ERROR ("client %p: not PLAYING or READY", client);
1245 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1251 GST_ERROR ("client %p: unsuspend failed", client);
1252 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1258 do_keepalive (GstRTSPSession * session)
1260 GST_INFO ("keep session %p alive", session);
1261 gst_rtsp_session_touch (session);
1264 /* parse @transport and return a valid transport in @tr. only transports
1265 * supported by @stream are returned. Returns FALSE if no valid transport
1268 parse_transport (const char *transport, GstRTSPStream * stream,
1269 GstRTSPTransport * tr)
1276 gst_rtsp_transport_init (tr);
1278 GST_DEBUG ("parsing transports %s", transport);
1280 transports = g_strsplit (transport, ",", 0);
1282 /* loop through the transports, try to parse */
1283 for (i = 0; transports[i]; i++) {
1284 res = gst_rtsp_transport_parse (transports[i], tr);
1285 if (res != GST_RTSP_OK) {
1286 /* no valid transport, search some more */
1287 GST_WARNING ("could not parse transport %s", transports[i]);
1291 /* we have a transport, see if it's supported */
1292 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1293 GST_WARNING ("unsupported transport %s", transports[i]);
1297 /* we have a valid transport */
1298 GST_INFO ("found valid transport %s", transports[i]);
1303 gst_rtsp_transport_init (tr);
1305 g_strfreev (transports);
1311 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1312 GstRTSPStream * stream, GstRTSPContext * ctx)
1314 GstRTSPMessage *request = ctx->request;
1315 gchar *blocksize_str;
1317 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1318 &blocksize_str, 0) == GST_RTSP_OK) {
1322 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1323 if (end == blocksize_str)
1326 /* we don't want to change the mtu when this media
1327 * can be shared because it impacts other clients */
1328 if (gst_rtsp_media_is_shared (media))
1331 if (blocksize > G_MAXUINT)
1332 blocksize = G_MAXUINT;
1334 gst_rtsp_stream_set_mtu (stream, blocksize);
1342 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1343 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1349 default_configure_client_transport (GstRTSPClient * client,
1350 GstRTSPContext * ctx, GstRTSPTransport * ct)
1352 GstRTSPClientPrivate *priv = client->priv;
1354 /* we have a valid transport now, set the destination of the client. */
1355 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1356 gboolean use_client_settings;
1358 use_client_settings =
1359 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1361 if (ct->destination && use_client_settings) {
1362 GstRTSPAddress *addr;
1364 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1365 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1370 gst_rtsp_address_free (addr);
1372 GstRTSPAddress *addr;
1373 GSocketFamily family;
1375 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1377 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1381 g_free (ct->destination);
1382 ct->destination = g_strdup (addr->address);
1383 ct->port.min = addr->port;
1384 ct->port.max = addr->port + addr->n_ports - 1;
1385 ct->ttl = addr->ttl;
1387 gst_rtsp_address_free (addr);
1392 url = gst_rtsp_connection_get_url (priv->connection);
1393 g_free (ct->destination);
1394 ct->destination = g_strdup (url->host);
1396 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1398 GSocketAddress *addr;
1400 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1401 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1402 /* our read port is the sender port of client */
1403 ct->client_port.min =
1404 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1405 g_object_unref (addr);
1407 if ((addr = g_socket_get_local_address (sock, NULL))) {
1408 ct->server_port.max =
1409 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1410 g_object_unref (addr);
1412 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1413 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1414 /* our write port is the receiver port of client */
1415 ct->client_port.max =
1416 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1417 g_object_unref (addr);
1419 if ((addr = g_socket_get_local_address (sock, NULL))) {
1420 ct->server_port.min =
1421 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1422 g_object_unref (addr);
1424 /* check if the client selected channels for TCP */
1425 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1426 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1436 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1441 static GstRTSPTransport *
1442 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1443 GstRTSPTransport * ct)
1445 GstRTSPTransport *st;
1447 GSocketFamily family;
1449 /* prepare the server transport */
1450 gst_rtsp_transport_new (&st);
1452 st->trans = ct->trans;
1453 st->profile = ct->profile;
1454 st->lower_transport = ct->lower_transport;
1456 addr = g_inet_address_new_from_string (ct->destination);
1459 GST_ERROR ("failed to get inet addr from client destination");
1460 family = G_SOCKET_FAMILY_IPV4;
1462 family = g_inet_address_get_family (addr);
1463 g_object_unref (addr);
1467 switch (st->lower_transport) {
1468 case GST_RTSP_LOWER_TRANS_UDP:
1469 st->client_port = ct->client_port;
1470 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1472 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1473 st->port = ct->port;
1474 st->destination = g_strdup (ct->destination);
1477 case GST_RTSP_LOWER_TRANS_TCP:
1478 st->interleaved = ct->interleaved;
1479 st->client_port = ct->client_port;
1480 st->server_port = ct->server_port;
1485 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1491 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1493 const gchar *srtp_cipher;
1494 const gchar *srtp_auth;
1495 const GstMIKEYPayload *sp;
1498 /* loop over Security policy until we find one containing policy */
1500 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1503 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1507 /* the default ciphers */
1508 srtp_cipher = "aes-128-icm";
1509 srtp_auth = "hmac-sha1-80";
1511 /* now override the defaults with what is in the Security Policy */
1515 /* collect all the params and go over them */
1516 len = gst_mikey_payload_sp_get_n_params (sp);
1517 for (i = 0; i < len; i++) {
1518 const GstMIKEYPayloadSPParam *param =
1519 gst_mikey_payload_sp_get_param (sp, i);
1521 switch (param->type) {
1522 case GST_MIKEY_SP_SRTP_ENC_ALG:
1523 switch (param->val[0]) {
1525 srtp_cipher = "null";
1529 srtp_cipher = "aes-128-icm";
1535 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1536 switch (param->val[0]) {
1542 srtp_auth = "hmac-sha1-80";
1548 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1550 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1557 /* now configure the SRTP parameters */
1558 gst_caps_set_simple (caps,
1559 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1560 "srtp-auth", G_TYPE_STRING, srtp_auth,
1561 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1562 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1568 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1569 guint8 * data, gsize size)
1571 GstMIKEYMessage *msg;
1573 GstCaps *caps = NULL;
1574 GstMIKEYPayloadKEMAC *kemac;
1575 const GstMIKEYPayloadKeyData *pkd;
1578 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1579 * set of Crypto Sessions protected with the same master key.
1580 * In the context of SRTP, an RTP and its RTCP stream is part of a
1582 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1585 /* we can only handle SRTP crypto sessions for now */
1586 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1587 goto invalid_map_type;
1589 /* get the number of crypto sessions. This maps SSRC to its
1590 * security parameters */
1591 n_cs = gst_mikey_message_get_n_cs (msg);
1593 goto no_crypto_sessions;
1595 /* we also need keys */
1596 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1597 (msg, GST_MIKEY_PT_KEMAC, 0)))
1600 /* we don't support encrypted keys */
1601 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1602 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1603 goto unsupported_encryption;
1605 /* get Key data sub-payload */
1606 pkd = (const GstMIKEYPayloadKeyData *)
1607 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1610 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1613 /* go over all crypto sessions and create the security policy for each
1615 for (i = 0; i < n_cs; i++) {
1616 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1618 caps = gst_caps_new_simple ("application/x-srtp",
1619 "ssrc", G_TYPE_UINT, map->ssrc,
1620 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1621 mikey_apply_policy (caps, msg, map->policy);
1623 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1624 gst_caps_unref (caps);
1626 gst_mikey_message_free (msg);
1633 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1638 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1639 goto cleanup_message;
1643 GST_DEBUG_OBJECT (client, "no crypto sessions");
1644 goto cleanup_message;
1648 GST_DEBUG_OBJECT (client, "no keys found");
1649 goto cleanup_message;
1651 unsupported_encryption:
1653 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1654 goto cleanup_message;
1658 gst_mikey_message_free (msg);
1663 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1666 strip_chars (gchar * str)
1673 if (!IS_STRIP_CHAR (str[len]))
1677 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1678 memmove (str, s, len + 1);
1682 * KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1683 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1686 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1691 specs = g_strsplit (keymgmt, ",", 0);
1692 for (i = 0; specs[i]; i++) {
1695 split = g_strsplit (specs[i], ";", 0);
1696 for (j = 0; split[j]; j++) {
1697 g_strstrip (split[j]);
1698 if (g_str_has_prefix (split[j], "prot=")) {
1699 g_strstrip (split[j] + 5);
1700 if (!g_str_equal (split[j] + 5, "mikey"))
1702 GST_DEBUG ("found mikey");
1703 } else if (g_str_has_prefix (split[j], "uri=")) {
1704 strip_chars (split[j] + 4);
1705 GST_DEBUG ("found uri '%s'", split[j] + 4);
1706 } else if (g_str_has_prefix (split[j], "data=")) {
1709 strip_chars (split[j] + 5);
1710 GST_DEBUG ("found data '%s'", split[j] + 5);
1711 data = g_base64_decode_inplace (split[j] + 5, &size);
1712 handle_mikey_data (client, ctx, data, size);
1720 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1722 GstRTSPClientPrivate *priv = client->priv;
1725 gchar *transport, *keymgmt;
1726 GstRTSPTransport *ct, *st;
1727 GstRTSPStatusCode code;
1728 GstRTSPSession *session;
1729 GstRTSPStreamTransport *trans;
1731 GstRTSPSessionMedia *sessmedia;
1732 GstRTSPMedia *media;
1733 GstRTSPStream *stream;
1734 GstRTSPState rtspstate;
1735 GstRTSPClientClass *klass;
1736 gchar *path, *control;
1743 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1744 path = klass->make_path_from_uri (client, uri);
1746 /* parse the transport */
1748 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1750 if (res != GST_RTSP_OK)
1753 /* we create the session after parsing stuff so that we don't make
1754 * a session for malformed requests */
1755 if (priv->session_pool == NULL)
1758 session = ctx->session;
1761 g_object_ref (session);
1762 /* get a handle to the configuration of the media in the session, this can
1763 * return NULL if this is a new url to manage in this session. */
1764 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1766 /* we need a new media configuration in this session */
1770 /* we have no session media, find one and manage it */
1771 if (sessmedia == NULL) {
1772 /* get a handle to the configuration of the media in the session */
1773 media = find_media (client, ctx, path, &matched);
1775 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1776 g_object_ref (media);
1778 goto media_not_found;
1780 /* no media, not found then */
1782 goto media_not_found_no_reply;
1784 if (path[matched] == '\0')
1785 goto control_not_found;
1787 /* path is what matched. */
1788 path[matched] = '\0';
1789 /* control is remainder */
1790 control = &path[matched + 1];
1792 /* find the stream now using the control part */
1793 stream = gst_rtsp_media_find_stream (media, control);
1795 goto stream_not_found;
1797 /* now we have a uri identifying a valid media and stream */
1798 ctx->stream = stream;
1801 if (session == NULL) {
1802 /* create a session if this fails we probably reached our session limit or
1804 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1805 goto service_unavailable;
1807 /* make sure this client is closed when the session is closed */
1808 client_watch_session (client, session);
1810 /* signal new session */
1811 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1814 ctx->session = session;
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 if (!klass->configure_client_media (client, media, stream, ctx))
1830 goto configure_media_failed_no_reply;
1832 gst_rtsp_transport_new (&ct);
1834 /* parse and find a usable supported transport */
1835 if (!parse_transport (transport, stream, ct))
1836 goto unsupported_transports;
1838 /* update the client transport */
1839 if (!klass->configure_client_transport (client, ctx, ct))
1840 goto unsupported_client_transport;
1842 /* parse the keymgmt */
1843 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1844 &keymgmt, 0) == GST_RTSP_OK) {
1845 if (!handle_keymgmt (client, ctx, keymgmt))
1849 /* set in the session media transport */
1850 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1852 /* configure the url used to set this transport, this we will use when
1853 * generating the response for the PLAY request */
1854 gst_rtsp_stream_transport_set_url (trans, uri);
1856 /* configure keepalive for this transport */
1857 gst_rtsp_stream_transport_set_keepalive (trans,
1858 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1860 /* create and serialize the server transport */
1861 st = make_server_transport (client, ctx, ct);
1862 trans_str = gst_rtsp_transport_as_text (st);
1863 gst_rtsp_transport_free (st);
1865 /* construct the response now */
1866 code = GST_RTSP_STS_OK;
1867 gst_rtsp_message_init_response (ctx->response, code,
1868 gst_rtsp_status_as_text (code), ctx->request);
1870 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1874 send_message (client, ctx, ctx->response, FALSE);
1876 /* update the state */
1877 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1878 switch (rtspstate) {
1879 case GST_RTSP_STATE_PLAYING:
1880 case GST_RTSP_STATE_RECORDING:
1881 case GST_RTSP_STATE_READY:
1882 /* no state change */
1885 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1888 g_object_unref (session);
1891 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1898 GST_ERROR ("client %p: no uri", client);
1899 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1904 GST_ERROR ("client %p: no transport", client);
1905 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1910 GST_ERROR ("client %p: no session pool configured", client);
1911 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1914 media_not_found_no_reply:
1916 GST_ERROR ("client %p: media '%s' not found", client, path);
1917 /* error reply is already sent */
1922 GST_ERROR ("client %p: media '%s' not found", client, path);
1923 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1928 GST_ERROR ("client %p: no control in path '%s'", client, path);
1929 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1930 g_object_unref (media);
1935 GST_ERROR ("client %p: stream '%s' not found", client, control);
1936 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1937 g_object_unref (media);
1940 service_unavailable:
1942 GST_ERROR ("client %p: can't create session", client);
1943 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1944 g_object_unref (media);
1947 sessmedia_unavailable:
1949 GST_ERROR ("client %p: can't create session media", client);
1950 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1951 g_object_unref (media);
1952 goto cleanup_session;
1954 configure_media_failed_no_reply:
1956 GST_ERROR ("client %p: configure_media failed", client);
1957 /* error reply is already sent */
1958 goto cleanup_session;
1960 unsupported_transports:
1962 GST_ERROR ("client %p: unsupported transports", client);
1963 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1964 goto cleanup_transport;
1966 unsupported_client_transport:
1968 GST_ERROR ("client %p: unsupported client transport", client);
1969 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1970 goto cleanup_transport;
1974 GST_ERROR ("client %p: keymgmt error", client);
1975 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
1976 goto cleanup_transport;
1980 gst_rtsp_transport_free (ct);
1982 g_object_unref (session);
1989 static GstSDPMessage *
1990 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1992 GstRTSPClientPrivate *priv = client->priv;
1997 gst_sdp_message_new (&sdp);
1999 /* some standard things first */
2000 gst_sdp_message_set_version (sdp, "0");
2007 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
2010 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2011 gst_sdp_message_set_information (sdp, "rtsp-server");
2012 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2013 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2014 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2015 gst_sdp_message_add_attribute (sdp, "control", "*");
2017 info.is_ipv6 = priv->is_ipv6;
2018 info.server_ip = priv->server_ip;
2020 /* create an SDP for the media object */
2021 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2029 GST_ERROR ("client %p: could not create SDP", client);
2030 gst_sdp_message_free (sdp);
2035 /* for the describe we must generate an SDP */
2037 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2039 GstRTSPClientPrivate *priv = client->priv;
2044 GstRTSPMedia *media;
2045 GstRTSPClientClass *klass;
2047 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2052 /* check what kind of format is accepted, we don't really do anything with it
2053 * and always return SDP for now. */
2058 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2060 if (res == GST_RTSP_ENOTIMPL)
2063 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2067 if (!priv->mount_points)
2068 goto no_mount_points;
2070 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2073 /* find the media object for the uri */
2074 if (!(media = find_media (client, ctx, path, NULL)))
2077 /* create an SDP for the media object on this client */
2078 if (!(sdp = klass->create_sdp (client, media)))
2081 /* we suspend after the describe */
2082 gst_rtsp_media_suspend (media);
2083 g_object_unref (media);
2085 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2086 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2088 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2091 /* content base for some clients that might screw up creating the setup uri */
2092 str = make_base_url (client, ctx->uri, path);
2095 GST_INFO ("adding content-base: %s", str);
2096 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2098 /* add SDP to the response body */
2099 str = gst_sdp_message_as_text (sdp);
2100 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2101 gst_sdp_message_free (sdp);
2103 send_message (client, ctx, ctx->response, FALSE);
2105 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2113 GST_ERROR ("client %p: no uri", client);
2114 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2119 GST_ERROR ("client %p: no mount points configured", client);
2120 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2125 GST_ERROR ("client %p: can't find path for url", client);
2126 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2131 GST_ERROR ("client %p: no media", client);
2133 /* error reply is already sent */
2138 GST_ERROR ("client %p: can't create SDP", client);
2139 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2141 g_object_unref (media);
2147 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2149 GstRTSPMethod options;
2152 options = GST_RTSP_DESCRIBE |
2157 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2159 str = gst_rtsp_options_as_text (options);
2161 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2162 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2164 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2167 send_message (client, ctx, ctx->response, FALSE);
2169 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2175 /* remove duplicate and trailing '/' */
2177 sanitize_uri (GstRTSPUrl * uri)
2181 gboolean have_slash, prev_slash;
2183 s = d = uri->abspath;
2184 len = strlen (uri->abspath);
2188 for (i = 0; i < len; i++) {
2189 have_slash = s[i] == '/';
2191 if (!have_slash || !prev_slash)
2193 prev_slash = have_slash;
2195 len = d - uri->abspath;
2196 /* don't remove the first slash if that's the only thing left */
2197 if (len > 1 && *(d - 1) == '/')
2203 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
2205 GstRTSPClientPrivate *priv = client->priv;
2207 GST_INFO ("client %p: session %p finished", client, session);
2209 /* unlink all media managed in this session */
2210 client_unlink_session (client, session);
2212 /* remove the session */
2213 if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
2214 GST_INFO ("client %p: all sessions finalized, close the connection",
2216 close_connection (client);
2220 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2221 * and also returns a newly-allocated string of (comma-separated) unsupported
2222 * options in the unsupported_reqs variable .
2224 * There may be multiple Require headers, but we must send one single
2225 * Unsupported header with all the unsupported options as response. If
2226 * an incoming Require header contained a comma-separated list of options
2227 * GstRtspConnection will already have split that list up into multiple
2230 * TODO: allow the application to decide what features are supported
2233 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2236 GPtrArray *arr = NULL;
2242 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2244 if (res == GST_RTSP_ENOTIMPL)
2248 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2250 g_ptr_array_add (arr, g_strdup (reqs));
2254 /* if we don't have any Require headers at all, all is fine */
2258 /* otherwise we've now processed at all the Require headers */
2259 g_ptr_array_add (arr, NULL);
2261 /* for now we don't commit to supporting anything, so will just report
2262 * all of the required options as unsupported */
2263 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2265 g_ptr_array_unref (arr);
2270 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2272 GstRTSPClientPrivate *priv = client->priv;
2273 GstRTSPMethod method;
2274 const gchar *uristr;
2275 GstRTSPUrl *uri = NULL;
2276 GstRTSPVersion version;
2278 GstRTSPSession *session = NULL;
2279 GstRTSPContext sctx = { NULL }, *ctx;
2280 GstRTSPMessage response = { 0 };
2281 gchar *unsupported_reqs = NULL;
2284 if (!(ctx = gst_rtsp_context_get_current ())) {
2286 ctx->auth = priv->auth;
2287 gst_rtsp_context_push_current (ctx);
2290 ctx->conn = priv->connection;
2291 ctx->client = client;
2292 ctx->request = request;
2293 ctx->response = &response;
2295 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2296 gst_rtsp_message_dump (request);
2299 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2301 GST_INFO ("client %p: received a request %s %s %s", client,
2302 gst_rtsp_method_as_text (method), uristr,
2303 gst_rtsp_version_as_text (version));
2305 /* we can only handle 1.0 requests */
2306 if (version != GST_RTSP_VERSION_1_0)
2309 ctx->method = method;
2311 /* we always try to parse the url first */
2312 if (strcmp (uristr, "*") == 0) {
2313 /* special case where we have * as uri, keep uri = NULL */
2314 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2315 /* check if the uristr is an absolute path <=> scheme and host information
2319 scheme = g_uri_parse_scheme (uristr);
2320 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2321 gchar *absolute_uristr = NULL;
2323 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2324 if (priv->server_ip == NULL) {
2325 GST_WARNING_OBJECT (client, "host information missing");
2330 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2332 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2333 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2334 g_free (absolute_uristr);
2337 g_free (absolute_uristr);
2344 /* get the session if there is any */
2345 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2346 if (res == GST_RTSP_OK) {
2347 if (priv->session_pool == NULL)
2350 /* we had a session in the request, find it again */
2351 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2352 goto session_not_found;
2354 /* we add the session to the client list of watched sessions. When a session
2355 * disappears because it times out, we will be notified. If all sessions are
2356 * gone, we will close the connection */
2357 client_watch_session (client, session);
2360 /* sanitize the uri */
2364 ctx->session = session;
2366 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2367 goto not_authorized;
2369 /* handle any 'Require' headers */
2370 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2371 goto unsupported_requirement;
2373 /* now see what is asked and dispatch to a dedicated handler */
2375 case GST_RTSP_OPTIONS:
2376 handle_options_request (client, ctx);
2378 case GST_RTSP_DESCRIBE:
2379 handle_describe_request (client, ctx);
2381 case GST_RTSP_SETUP:
2382 handle_setup_request (client, ctx);
2385 handle_play_request (client, ctx);
2387 case GST_RTSP_PAUSE:
2388 handle_pause_request (client, ctx);
2390 case GST_RTSP_TEARDOWN:
2391 handle_teardown_request (client, ctx);
2393 case GST_RTSP_SET_PARAMETER:
2394 handle_set_param_request (client, ctx);
2396 case GST_RTSP_GET_PARAMETER:
2397 handle_get_param_request (client, ctx);
2399 case GST_RTSP_ANNOUNCE:
2400 case GST_RTSP_RECORD:
2401 case GST_RTSP_REDIRECT:
2402 goto not_implemented;
2403 case GST_RTSP_INVALID:
2410 gst_rtsp_context_pop_current (ctx);
2412 g_object_unref (session);
2414 gst_rtsp_url_free (uri);
2420 GST_ERROR ("client %p: version %d not supported", client, version);
2421 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2427 GST_ERROR ("client %p: bad request", client);
2428 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2433 GST_ERROR ("client %p: no pool configured", client);
2434 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2439 GST_ERROR ("client %p: session not found", client);
2440 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2445 GST_ERROR ("client %p: not allowed", client);
2446 /* error reply is already sent */
2449 unsupported_requirement:
2451 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2453 send_option_not_supported_response (client, ctx, unsupported_reqs);
2454 g_free (unsupported_reqs);
2459 GST_ERROR ("client %p: method %d not implemented", client, method);
2460 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2467 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2469 GstRTSPClientPrivate *priv = client->priv;
2471 GstRTSPSession *session = NULL;
2472 GstRTSPContext sctx = { NULL }, *ctx;
2475 if (!(ctx = gst_rtsp_context_get_current ())) {
2477 ctx->auth = priv->auth;
2478 gst_rtsp_context_push_current (ctx);
2481 ctx->conn = priv->connection;
2482 ctx->client = client;
2483 ctx->request = NULL;
2485 ctx->method = GST_RTSP_INVALID;
2486 ctx->response = response;
2488 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2489 gst_rtsp_message_dump (response);
2492 GST_INFO ("client %p: received a response", client);
2494 /* get the session if there is any */
2496 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2497 if (res == GST_RTSP_OK) {
2498 if (priv->session_pool == NULL)
2501 /* we had a session in the request, find it again */
2502 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2503 goto session_not_found;
2505 /* we add the session to the client list of watched sessions. When a session
2506 * disappears because it times out, we will be notified. If all sessions are
2507 * gone, we will close the connection */
2508 client_watch_session (client, session);
2511 ctx->session = session;
2513 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2518 gst_rtsp_context_pop_current (ctx);
2520 g_object_unref (session);
2525 GST_ERROR ("client %p: no pool configured", client);
2530 GST_ERROR ("client %p: session not found", client);
2536 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2538 GstRTSPClientPrivate *priv = client->priv;
2547 /* find the stream for this message */
2548 res = gst_rtsp_message_parse_data (message, &channel);
2549 if (res != GST_RTSP_OK)
2552 gst_rtsp_message_steal_body (message, &data, &size);
2554 buffer = gst_buffer_new_wrapped (data, size);
2557 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2558 GstRTSPStreamTransport *trans;
2559 GstRTSPStream *stream;
2560 const GstRTSPTransport *tr;
2564 tr = gst_rtsp_stream_transport_get_transport (trans);
2565 stream = gst_rtsp_stream_transport_get_stream (trans);
2567 /* check for TCP transport */
2568 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
2569 /* dispatch to the stream based on the channel number */
2570 if (tr->interleaved.min == channel) {
2571 gst_rtsp_stream_recv_rtp (stream, buffer);
2574 } else if (tr->interleaved.max == channel) {
2575 gst_rtsp_stream_recv_rtcp (stream, buffer);
2582 gst_buffer_unref (buffer);
2586 * gst_rtsp_client_set_session_pool:
2587 * @client: a #GstRTSPClient
2588 * @pool: (transfer none): a #GstRTSPSessionPool
2590 * Set @pool as the sessionpool for @client which it will use to find
2591 * or allocate sessions. the sessionpool is usually inherited from the server
2592 * that created the client but can be overridden later.
2595 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2596 GstRTSPSessionPool * pool)
2598 GstRTSPSessionPool *old;
2599 GstRTSPClientPrivate *priv;
2601 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2603 priv = client->priv;
2606 g_object_ref (pool);
2608 g_mutex_lock (&priv->lock);
2609 old = priv->session_pool;
2610 priv->session_pool = pool;
2611 g_mutex_unlock (&priv->lock);
2614 g_object_unref (old);
2618 * gst_rtsp_client_get_session_pool:
2619 * @client: a #GstRTSPClient
2621 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2623 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2625 GstRTSPSessionPool *
2626 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2628 GstRTSPClientPrivate *priv;
2629 GstRTSPSessionPool *result;
2631 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2633 priv = client->priv;
2635 g_mutex_lock (&priv->lock);
2636 if ((result = priv->session_pool))
2637 g_object_ref (result);
2638 g_mutex_unlock (&priv->lock);
2644 * gst_rtsp_client_set_mount_points:
2645 * @client: a #GstRTSPClient
2646 * @mounts: (transfer none): a #GstRTSPMountPoints
2648 * Set @mounts as the mount points for @client which it will use to map urls
2649 * to media streams. These mount points are usually inherited from the server that
2650 * created the client but can be overriden later.
2653 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2654 GstRTSPMountPoints * mounts)
2656 GstRTSPClientPrivate *priv;
2657 GstRTSPMountPoints *old;
2659 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2661 priv = client->priv;
2664 g_object_ref (mounts);
2666 g_mutex_lock (&priv->lock);
2667 old = priv->mount_points;
2668 priv->mount_points = mounts;
2669 g_mutex_unlock (&priv->lock);
2672 g_object_unref (old);
2676 * gst_rtsp_client_get_mount_points:
2677 * @client: a #GstRTSPClient
2679 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2681 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2683 GstRTSPMountPoints *
2684 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2686 GstRTSPClientPrivate *priv;
2687 GstRTSPMountPoints *result;
2689 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2691 priv = client->priv;
2693 g_mutex_lock (&priv->lock);
2694 if ((result = priv->mount_points))
2695 g_object_ref (result);
2696 g_mutex_unlock (&priv->lock);
2702 * gst_rtsp_client_set_auth:
2703 * @client: a #GstRTSPClient
2704 * @auth: (transfer none): a #GstRTSPAuth
2706 * configure @auth to be used as the authentication manager of @client.
2709 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2711 GstRTSPClientPrivate *priv;
2714 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2716 priv = client->priv;
2719 g_object_ref (auth);
2721 g_mutex_lock (&priv->lock);
2724 g_mutex_unlock (&priv->lock);
2727 g_object_unref (old);
2732 * gst_rtsp_client_get_auth:
2733 * @client: a #GstRTSPClient
2735 * Get the #GstRTSPAuth used as the authentication manager of @client.
2737 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2741 gst_rtsp_client_get_auth (GstRTSPClient * client)
2743 GstRTSPClientPrivate *priv;
2744 GstRTSPAuth *result;
2746 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2748 priv = client->priv;
2750 g_mutex_lock (&priv->lock);
2751 if ((result = priv->auth))
2752 g_object_ref (result);
2753 g_mutex_unlock (&priv->lock);
2759 * gst_rtsp_client_set_thread_pool:
2760 * @client: a #GstRTSPClient
2761 * @pool: (transfer none): a #GstRTSPThreadPool
2763 * configure @pool to be used as the thread pool of @client.
2766 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2767 GstRTSPThreadPool * pool)
2769 GstRTSPClientPrivate *priv;
2770 GstRTSPThreadPool *old;
2772 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2774 priv = client->priv;
2777 g_object_ref (pool);
2779 g_mutex_lock (&priv->lock);
2780 old = priv->thread_pool;
2781 priv->thread_pool = pool;
2782 g_mutex_unlock (&priv->lock);
2785 g_object_unref (old);
2789 * gst_rtsp_client_get_thread_pool:
2790 * @client: a #GstRTSPClient
2792 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2794 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2798 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2800 GstRTSPClientPrivate *priv;
2801 GstRTSPThreadPool *result;
2803 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2805 priv = client->priv;
2807 g_mutex_lock (&priv->lock);
2808 if ((result = priv->thread_pool))
2809 g_object_ref (result);
2810 g_mutex_unlock (&priv->lock);
2816 * gst_rtsp_client_set_connection:
2817 * @client: a #GstRTSPClient
2818 * @conn: (transfer full): a #GstRTSPConnection
2820 * Set the #GstRTSPConnection of @client. This function takes ownership of
2823 * Returns: %TRUE on success.
2826 gst_rtsp_client_set_connection (GstRTSPClient * client,
2827 GstRTSPConnection * conn)
2829 GstRTSPClientPrivate *priv;
2830 GSocket *read_socket;
2831 GSocketAddress *address;
2833 GError *error = NULL;
2835 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2836 g_return_val_if_fail (conn != NULL, FALSE);
2838 priv = client->priv;
2840 read_socket = gst_rtsp_connection_get_read_socket (conn);
2842 if (!(address = g_socket_get_local_address (read_socket, &error)))
2845 g_free (priv->server_ip);
2846 /* keep the original ip that the client connected to */
2847 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2848 GInetAddress *iaddr;
2850 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2852 /* socket might be ipv6 but adress still ipv4 */
2853 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2854 priv->server_ip = g_inet_address_to_string (iaddr);
2855 g_object_unref (address);
2857 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2858 priv->server_ip = g_strdup ("unknown");
2861 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2862 priv->server_ip, priv->is_ipv6);
2864 url = gst_rtsp_connection_get_url (conn);
2865 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2867 priv->connection = conn;
2874 GST_ERROR ("could not get local address %s", error->message);
2875 g_error_free (error);
2881 * gst_rtsp_client_get_connection:
2882 * @client: a #GstRTSPClient
2884 * Get the #GstRTSPConnection of @client.
2886 * Returns: (transfer none): the #GstRTSPConnection of @client.
2887 * The connection object returned remains valid until the client is freed.
2890 gst_rtsp_client_get_connection (GstRTSPClient * client)
2892 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2894 return client->priv->connection;
2898 * gst_rtsp_client_set_send_func:
2899 * @client: a #GstRTSPClient
2900 * @func: (scope notified): a #GstRTSPClientSendFunc
2901 * @user_data: (closure): user data passed to @func
2902 * @notify: (allow-none): called when @user_data is no longer in use
2904 * Set @func as the callback that will be called when a new message needs to be
2905 * sent to the client. @user_data is passed to @func and @notify is called when
2906 * @user_data is no longer in use.
2908 * By default, the client will send the messages on the #GstRTSPConnection that
2909 * was configured with gst_rtsp_client_attach() was called.
2912 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2913 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2915 GstRTSPClientPrivate *priv;
2916 GDestroyNotify old_notify;
2919 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2921 priv = client->priv;
2923 g_mutex_lock (&priv->send_lock);
2924 priv->send_func = func;
2925 old_notify = priv->send_notify;
2926 old_data = priv->send_data;
2927 priv->send_notify = notify;
2928 priv->send_data = user_data;
2929 g_mutex_unlock (&priv->send_lock);
2932 old_notify (old_data);
2936 * gst_rtsp_client_handle_message:
2937 * @client: a #GstRTSPClient
2938 * @message: (transfer none): an #GstRTSPMessage
2940 * Let the client handle @message.
2942 * Returns: a #GstRTSPResult.
2945 gst_rtsp_client_handle_message (GstRTSPClient * client,
2946 GstRTSPMessage * message)
2948 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2949 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2951 switch (message->type) {
2952 case GST_RTSP_MESSAGE_REQUEST:
2953 handle_request (client, message);
2955 case GST_RTSP_MESSAGE_RESPONSE:
2956 handle_response (client, message);
2958 case GST_RTSP_MESSAGE_DATA:
2959 handle_data (client, message);
2968 * gst_rtsp_client_send_message:
2969 * @client: a #GstRTSPClient
2970 * @session: (transfer none): a #GstRTSPSession to send the message to or %NULL
2971 * @message: (transfer none): The #GstRTSPMessage to send
2973 * Send a message message to the remote end. @message must be a
2974 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2977 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2978 GstRTSPMessage * message)
2980 GstRTSPContext sctx = { NULL }
2982 GstRTSPClientPrivate *priv;
2984 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2985 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2986 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
2987 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
2989 priv = client->priv;
2991 if (!(ctx = gst_rtsp_context_get_current ())) {
2993 ctx->auth = priv->auth;
2994 gst_rtsp_context_push_current (ctx);
2997 ctx->conn = priv->connection;
2998 ctx->client = client;
2999 ctx->session = session;
3001 send_message (client, ctx, message, FALSE);
3004 gst_rtsp_context_pop_current (ctx);
3009 static GstRTSPResult
3010 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
3011 gboolean close, gpointer user_data)
3013 GstRTSPClientPrivate *priv = client->priv;
3021 /* send the response and store the seq number so we can wait until it's
3022 * written to the client to close the connection */
3024 gst_rtsp_watch_send_message (priv->watch, message,
3025 close ? &priv->close_seq : NULL);
3026 if (ret == GST_RTSP_OK)
3029 if (ret != GST_RTSP_ENOMEM)
3033 if (priv->drop_backlog)
3036 /* queue was full, wait for more space */
3037 GST_DEBUG_OBJECT (client, "waiting for backlog");
3038 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3039 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3040 } while (ret != GST_RTSP_EINTR);
3047 GST_DEBUG_OBJECT (client, "got error %d", ret);
3052 static GstRTSPResult
3053 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3056 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3059 static GstRTSPResult
3060 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3062 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3063 GstRTSPClientPrivate *priv = client->priv;
3065 if (priv->close_seq && priv->close_seq == cseq) {
3066 priv->close_seq = 0;
3067 close_connection (client);
3073 static GstRTSPResult
3074 closed (GstRTSPWatch * watch, gpointer user_data)
3076 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3077 GstRTSPClientPrivate *priv = client->priv;
3078 const gchar *tunnelid;
3080 GST_INFO ("client %p: connection closed", client);
3082 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3083 g_mutex_lock (&tunnels_lock);
3084 /* remove from tunnelids */
3085 g_hash_table_remove (tunnels, tunnelid);
3086 g_mutex_unlock (&tunnels_lock);
3089 gst_rtsp_watch_set_flushing (watch, TRUE);
3090 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3095 static GstRTSPResult
3096 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3098 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3101 str = gst_rtsp_strresult (result);
3102 GST_INFO ("client %p: received an error %s", client, str);
3108 static GstRTSPResult
3109 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3110 GstRTSPMessage * message, guint id, gpointer user_data)
3112 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3115 str = gst_rtsp_strresult (result);
3117 ("client %p: error when handling message %p with id %d: %s",
3118 client, message, id, str);
3125 remember_tunnel (GstRTSPClient * client)
3127 GstRTSPClientPrivate *priv = client->priv;
3128 const gchar *tunnelid;
3130 /* store client in the pending tunnels */
3131 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3132 if (tunnelid == NULL)
3135 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3137 /* we can't have two clients connecting with the same tunnelid */
3138 g_mutex_lock (&tunnels_lock);
3139 if (g_hash_table_lookup (tunnels, tunnelid))
3140 goto tunnel_existed;
3142 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3143 g_mutex_unlock (&tunnels_lock);
3150 GST_ERROR ("client %p: no tunnelid provided", client);
3155 g_mutex_unlock (&tunnels_lock);
3156 GST_ERROR ("client %p: tunnel session %s already existed", client,
3162 static GstRTSPResult
3163 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3165 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3166 GstRTSPClientPrivate *priv = client->priv;
3168 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3171 /* ignore error, it'll only be a problem when the client does a POST again */
3172 remember_tunnel (client);
3178 handle_tunnel (GstRTSPClient * client)
3180 GstRTSPClientPrivate *priv = client->priv;
3181 GstRTSPClient *oclient;
3182 GstRTSPClientPrivate *opriv;
3183 const gchar *tunnelid;
3185 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3186 if (tunnelid == NULL)
3189 /* check for previous tunnel */
3190 g_mutex_lock (&tunnels_lock);
3191 oclient = g_hash_table_lookup (tunnels, tunnelid);
3193 if (oclient == NULL) {
3194 /* no previous tunnel, remember tunnel */
3195 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3196 g_mutex_unlock (&tunnels_lock);
3198 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3199 client, priv->connection);
3201 /* merge both tunnels into the first client */
3202 /* remove the old client from the table. ref before because removing it will
3203 * remove the ref to it. */
3204 g_object_ref (oclient);
3205 g_hash_table_remove (tunnels, tunnelid);
3206 g_mutex_unlock (&tunnels_lock);
3208 opriv = oclient->priv;
3210 if (opriv->watch == NULL)
3213 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3214 oclient, opriv->connection, priv->connection);
3216 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3217 gst_rtsp_watch_reset (priv->watch);
3218 gst_rtsp_watch_reset (opriv->watch);
3219 g_object_unref (oclient);
3221 /* the old client owns the tunnel now, the new one will be freed */
3222 g_source_destroy ((GSource *) priv->watch);
3224 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3232 GST_ERROR ("client %p: no tunnelid provided", client);
3237 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3238 g_object_unref (oclient);
3243 static GstRTSPStatusCode
3244 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3246 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3248 GST_INFO ("client %p: tunnel get (connection %p)", client,
3249 client->priv->connection);
3251 if (!handle_tunnel (client)) {
3252 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3255 return GST_RTSP_STS_OK;
3258 static GstRTSPResult
3259 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3261 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3263 GST_INFO ("client %p: tunnel post (connection %p)", client,
3264 client->priv->connection);
3266 if (!handle_tunnel (client)) {
3267 return GST_RTSP_ERROR;
3273 static GstRTSPResult
3274 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3275 GstRTSPMessage * response, gpointer user_data)
3277 GstRTSPClientClass *klass;
3279 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3280 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3282 if (klass->tunnel_http_response) {
3283 klass->tunnel_http_response (client, request, response);
3289 static GstRTSPWatchFuncs watch_funcs = {
3298 tunnel_http_response
3302 client_watch_notify (GstRTSPClient * client)
3304 GstRTSPClientPrivate *priv = client->priv;
3306 GST_INFO ("client %p: watch destroyed", client);
3308 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3309 g_object_unref (client);
3313 * gst_rtsp_client_attach:
3314 * @client: a #GstRTSPClient
3315 * @context: (allow-none): a #GMainContext
3317 * Attaches @client to @context. When the mainloop for @context is run, the
3318 * client will be dispatched. When @context is %NULL, the default context will be
3321 * This function should be called when the client properties and urls are fully
3322 * configured and the client is ready to start.
3324 * Returns: the ID (greater than 0) for the source within the GMainContext.
3327 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3329 GstRTSPClientPrivate *priv;
3332 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3333 priv = client->priv;
3334 g_return_val_if_fail (priv->connection != NULL, 0);
3335 g_return_val_if_fail (priv->watch == NULL, 0);
3337 /* create watch for the connection and attach */
3338 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3339 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3340 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3341 (GDestroyNotify) gst_rtsp_watch_unref);
3343 /* FIXME make this configurable. We don't want to do this yet because it will
3344 * be superceeded by a cache object later */
3345 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
3347 GST_INFO ("attaching to context %p", context);
3348 res = gst_rtsp_watch_attach (priv->watch, context);
3354 * gst_rtsp_client_session_filter:
3355 * @client: a #GstRTSPClient
3356 * @func: (scope call) (allow-none): a callback
3357 * @user_data: user data passed to @func
3359 * Call @func for each session managed by @client. The result value of @func
3360 * determines what happens to the session. @func will be called with @client
3361 * locked so no further actions on @client can be performed from @func.
3363 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3366 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3368 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3369 * will also be added with an additional ref to the result #GList of this
3372 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3374 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3375 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3376 * element in the #GList should be unreffed before the list is freed.
3379 gst_rtsp_client_session_filter (GstRTSPClient * client,
3380 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3382 GstRTSPClientPrivate *priv;
3383 GList *result, *walk, *next;
3385 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3387 priv = client->priv;
3391 g_mutex_lock (&priv->lock);
3392 for (walk = priv->sessions; walk; walk = next) {
3393 GstRTSPSession *sess = walk->data;
3394 GstRTSPFilterResult res;
3396 next = g_list_next (walk);
3399 res = func (client, sess, user_data);
3401 res = GST_RTSP_FILTER_REF;
3404 case GST_RTSP_FILTER_REMOVE:
3405 /* stop watching the session and pretent it went away */
3406 client_cleanup_session (client, sess);
3408 case GST_RTSP_FILTER_REF:
3409 result = g_list_prepend (result, g_object_ref (sess));
3411 case GST_RTSP_FILTER_KEEP:
3416 g_mutex_unlock (&priv->lock);