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);
799 /* connection is now closed, destroy the watch which will also cause the
800 * closed signal to be emitted */
802 GST_DEBUG ("client %p: destroying watch", client);
803 g_source_destroy ((GSource *) priv->watch);
805 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
810 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
815 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
817 path = g_strdup (uri->abspath);
823 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
825 GstRTSPClientPrivate *priv = client->priv;
826 GstRTSPClientClass *klass;
827 GstRTSPSession *session;
828 GstRTSPSessionMedia *sessmedia;
829 GstRTSPStatusCode code;
836 session = ctx->session;
841 klass = GST_RTSP_CLIENT_GET_CLASS (client);
842 path = klass->make_path_from_uri (client, ctx->uri);
844 /* get a handle to the configuration of the media in the session */
845 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
849 /* only aggregate control for now.. */
850 if (path[matched] != '\0')
855 ctx->sessmedia = sessmedia;
857 /* we emit the signal before closing the connection */
858 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
861 /* make sure we unblock the backlog and don't accept new messages
863 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
865 /* unlink the all TCP callbacks */
866 unlink_session_transports (client, session, sessmedia);
868 /* remove the session from the watched sessions */
869 client_unwatch_session (client, session);
871 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
873 /* allow messages again so that we can send the reply */
874 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
876 /* unmanage the media in the session, returns false if all media session
878 if (!gst_rtsp_session_release_media (session, sessmedia)) {
879 /* remove the session */
880 gst_rtsp_session_pool_remove (priv->session_pool, session);
882 /* construct the response now */
883 code = GST_RTSP_STS_OK;
884 gst_rtsp_message_init_response (ctx->response, code,
885 gst_rtsp_status_as_text (code), ctx->request);
887 send_message (client, ctx, ctx->response, TRUE);
894 GST_ERROR ("client %p: no session", client);
895 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
900 GST_ERROR ("client %p: no uri supplied", client);
901 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
906 GST_ERROR ("client %p: no media for uri", client);
907 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
913 GST_ERROR ("client %p: no aggregate path %s", client, path);
914 send_generic_response (client,
915 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
922 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
926 res = gst_rtsp_params_set (client, ctx);
932 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
936 res = gst_rtsp_params_get (client, ctx);
942 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
948 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
949 if (res != GST_RTSP_OK)
953 /* no body, keep-alive request */
954 send_generic_response (client, GST_RTSP_STS_OK, ctx);
956 /* there is a body, handle the params */
957 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
958 if (res != GST_RTSP_OK)
961 send_message (client, ctx, ctx->response, FALSE);
964 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
972 GST_ERROR ("client %p: bad request", client);
973 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
979 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
985 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
986 if (res != GST_RTSP_OK)
990 /* no body, keep-alive request */
991 send_generic_response (client, GST_RTSP_STS_OK, ctx);
993 /* there is a body, handle the params */
994 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
995 if (res != GST_RTSP_OK)
998 send_message (client, ctx, ctx->response, FALSE);
1001 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
1009 GST_ERROR ("client %p: bad request", client);
1010 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1016 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
1018 GstRTSPSession *session;
1019 GstRTSPClientClass *klass;
1020 GstRTSPSessionMedia *sessmedia;
1021 GstRTSPStatusCode code;
1022 GstRTSPState rtspstate;
1026 if (!(session = ctx->session))
1032 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1033 path = klass->make_path_from_uri (client, ctx->uri);
1035 /* get a handle to the configuration of the media in the session */
1036 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1040 if (path[matched] != '\0')
1045 ctx->sessmedia = sessmedia;
1047 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1048 /* the session state must be playing or recording */
1049 if (rtspstate != GST_RTSP_STATE_PLAYING &&
1050 rtspstate != GST_RTSP_STATE_RECORDING)
1053 /* unlink the all TCP callbacks */
1054 unlink_session_transports (client, session, sessmedia);
1056 /* then pause sending */
1057 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1059 /* construct the response now */
1060 code = GST_RTSP_STS_OK;
1061 gst_rtsp_message_init_response (ctx->response, code,
1062 gst_rtsp_status_as_text (code), ctx->request);
1064 send_message (client, ctx, ctx->response, FALSE);
1066 /* the state is now READY */
1067 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1069 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1076 GST_ERROR ("client %p: no seesion", client);
1077 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1082 GST_ERROR ("client %p: no uri supplied", client);
1083 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1088 GST_ERROR ("client %p: no media for uri", client);
1089 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1095 GST_ERROR ("client %p: no aggregate path %s", client, path);
1096 send_generic_response (client,
1097 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1103 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1104 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1110 /* convert @url and @path to a URL used as a content base for the factory
1111 * located at @path */
1113 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1119 /* check for trailing '/' and append one */
1120 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1125 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1127 result = gst_rtsp_url_get_request_uri (&tmp);
1128 g_free (tmp.abspath);
1134 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1136 GstRTSPSession *session;
1137 GstRTSPClientClass *klass;
1138 GstRTSPSessionMedia *sessmedia;
1139 GstRTSPMedia *media;
1140 GstRTSPStatusCode code;
1143 GstRTSPTimeRange *range;
1145 GstRTSPState rtspstate;
1146 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1147 gchar *path, *rtpinfo;
1150 if (!(session = ctx->session))
1153 if (!(uri = ctx->uri))
1156 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1157 path = klass->make_path_from_uri (client, uri);
1159 /* get a handle to the configuration of the media in the session */
1160 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1164 if (path[matched] != '\0')
1169 ctx->sessmedia = sessmedia;
1170 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1172 /* the session state must be playing or ready */
1173 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1174 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1177 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1178 if (!gst_rtsp_media_unsuspend (media))
1179 goto unsuspend_failed;
1181 /* parse the range header if we have one */
1182 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1183 if (res == GST_RTSP_OK) {
1184 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1185 /* we have a range, seek to the position */
1187 gst_rtsp_media_seek (media, range);
1188 gst_rtsp_range_free (range);
1192 /* link the all TCP callbacks */
1193 link_session_transports (client, session, sessmedia);
1195 /* grab RTPInfo from the media now */
1196 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1198 /* construct the response now */
1199 code = GST_RTSP_STS_OK;
1200 gst_rtsp_message_init_response (ctx->response, code,
1201 gst_rtsp_status_as_text (code), ctx->request);
1203 /* add the RTP-Info header */
1205 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1209 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1211 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1213 send_message (client, ctx, ctx->response, FALSE);
1215 /* start playing after sending the response */
1216 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1218 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1220 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1227 GST_ERROR ("client %p: no session", client);
1228 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1233 GST_ERROR ("client %p: no uri supplied", client);
1234 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1239 GST_ERROR ("client %p: media not found", client);
1240 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1245 GST_ERROR ("client %p: no aggregate path %s", client, path);
1246 send_generic_response (client,
1247 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1253 GST_ERROR ("client %p: not PLAYING or READY", client);
1254 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1260 GST_ERROR ("client %p: unsuspend failed", client);
1261 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1267 do_keepalive (GstRTSPSession * session)
1269 GST_INFO ("keep session %p alive", session);
1270 gst_rtsp_session_touch (session);
1273 /* parse @transport and return a valid transport in @tr. only transports
1274 * supported by @stream are returned. Returns FALSE if no valid transport
1277 parse_transport (const char *transport, GstRTSPStream * stream,
1278 GstRTSPTransport * tr)
1285 gst_rtsp_transport_init (tr);
1287 GST_DEBUG ("parsing transports %s", transport);
1289 transports = g_strsplit (transport, ",", 0);
1291 /* loop through the transports, try to parse */
1292 for (i = 0; transports[i]; i++) {
1293 res = gst_rtsp_transport_parse (transports[i], tr);
1294 if (res != GST_RTSP_OK) {
1295 /* no valid transport, search some more */
1296 GST_WARNING ("could not parse transport %s", transports[i]);
1300 /* we have a transport, see if it's supported */
1301 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1302 GST_WARNING ("unsupported transport %s", transports[i]);
1306 /* we have a valid transport */
1307 GST_INFO ("found valid transport %s", transports[i]);
1312 gst_rtsp_transport_init (tr);
1314 g_strfreev (transports);
1320 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1321 GstRTSPStream * stream, GstRTSPContext * ctx)
1323 GstRTSPMessage *request = ctx->request;
1324 gchar *blocksize_str;
1326 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1327 &blocksize_str, 0) == GST_RTSP_OK) {
1331 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1332 if (end == blocksize_str)
1335 /* we don't want to change the mtu when this media
1336 * can be shared because it impacts other clients */
1337 if (gst_rtsp_media_is_shared (media))
1340 if (blocksize > G_MAXUINT)
1341 blocksize = G_MAXUINT;
1343 gst_rtsp_stream_set_mtu (stream, blocksize);
1351 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1352 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1358 default_configure_client_transport (GstRTSPClient * client,
1359 GstRTSPContext * ctx, GstRTSPTransport * ct)
1361 GstRTSPClientPrivate *priv = client->priv;
1363 /* we have a valid transport now, set the destination of the client. */
1364 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1365 gboolean use_client_settings;
1367 use_client_settings =
1368 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1370 if (ct->destination && use_client_settings) {
1371 GstRTSPAddress *addr;
1373 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1374 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1379 gst_rtsp_address_free (addr);
1381 GstRTSPAddress *addr;
1382 GSocketFamily family;
1384 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1386 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1390 g_free (ct->destination);
1391 ct->destination = g_strdup (addr->address);
1392 ct->port.min = addr->port;
1393 ct->port.max = addr->port + addr->n_ports - 1;
1394 ct->ttl = addr->ttl;
1396 gst_rtsp_address_free (addr);
1401 url = gst_rtsp_connection_get_url (priv->connection);
1402 g_free (ct->destination);
1403 ct->destination = g_strdup (url->host);
1405 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1407 GSocketAddress *addr;
1409 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1410 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1411 /* our read port is the sender port of client */
1412 ct->client_port.min =
1413 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1414 g_object_unref (addr);
1416 if ((addr = g_socket_get_local_address (sock, NULL))) {
1417 ct->server_port.max =
1418 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1419 g_object_unref (addr);
1421 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1422 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1423 /* our write port is the receiver port of client */
1424 ct->client_port.max =
1425 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1426 g_object_unref (addr);
1428 if ((addr = g_socket_get_local_address (sock, NULL))) {
1429 ct->server_port.min =
1430 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1431 g_object_unref (addr);
1433 /* check if the client selected channels for TCP */
1434 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1435 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1445 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1450 static GstRTSPTransport *
1451 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1452 GstRTSPTransport * ct)
1454 GstRTSPTransport *st;
1456 GSocketFamily family;
1458 /* prepare the server transport */
1459 gst_rtsp_transport_new (&st);
1461 st->trans = ct->trans;
1462 st->profile = ct->profile;
1463 st->lower_transport = ct->lower_transport;
1465 addr = g_inet_address_new_from_string (ct->destination);
1468 GST_ERROR ("failed to get inet addr from client destination");
1469 family = G_SOCKET_FAMILY_IPV4;
1471 family = g_inet_address_get_family (addr);
1472 g_object_unref (addr);
1476 switch (st->lower_transport) {
1477 case GST_RTSP_LOWER_TRANS_UDP:
1478 st->client_port = ct->client_port;
1479 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1481 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1482 st->port = ct->port;
1483 st->destination = g_strdup (ct->destination);
1486 case GST_RTSP_LOWER_TRANS_TCP:
1487 st->interleaved = ct->interleaved;
1488 st->client_port = ct->client_port;
1489 st->server_port = ct->server_port;
1494 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1500 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1502 const gchar *srtp_cipher;
1503 const gchar *srtp_auth;
1504 const GstMIKEYPayload *sp;
1507 /* loop over Security policy until we find one containing policy */
1509 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1512 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1516 /* the default ciphers */
1517 srtp_cipher = "aes-128-icm";
1518 srtp_auth = "hmac-sha1-80";
1520 /* now override the defaults with what is in the Security Policy */
1524 /* collect all the params and go over them */
1525 len = gst_mikey_payload_sp_get_n_params (sp);
1526 for (i = 0; i < len; i++) {
1527 const GstMIKEYPayloadSPParam *param =
1528 gst_mikey_payload_sp_get_param (sp, i);
1530 switch (param->type) {
1531 case GST_MIKEY_SP_SRTP_ENC_ALG:
1532 switch (param->val[0]) {
1534 srtp_cipher = "null";
1538 srtp_cipher = "aes-128-icm";
1544 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1545 switch (param->val[0]) {
1551 srtp_auth = "hmac-sha1-80";
1557 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1559 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1566 /* now configure the SRTP parameters */
1567 gst_caps_set_simple (caps,
1568 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1569 "srtp-auth", G_TYPE_STRING, srtp_auth,
1570 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1571 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1577 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1578 guint8 * data, gsize size)
1580 GstMIKEYMessage *msg;
1582 GstCaps *caps = NULL;
1583 GstMIKEYPayloadKEMAC *kemac;
1584 const GstMIKEYPayloadKeyData *pkd;
1587 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1588 * set of Crypto Sessions protected with the same master key.
1589 * In the context of SRTP, an RTP and its RTCP stream is part of a
1591 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1594 /* we can only handle SRTP crypto sessions for now */
1595 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1596 goto invalid_map_type;
1598 /* get the number of crypto sessions. This maps SSRC to its
1599 * security parameters */
1600 n_cs = gst_mikey_message_get_n_cs (msg);
1602 goto no_crypto_sessions;
1604 /* we also need keys */
1605 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1606 (msg, GST_MIKEY_PT_KEMAC, 0)))
1609 /* we don't support encrypted keys */
1610 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1611 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1612 goto unsupported_encryption;
1614 /* get Key data sub-payload */
1615 pkd = (const GstMIKEYPayloadKeyData *)
1616 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1619 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1622 /* go over all crypto sessions and create the security policy for each
1624 for (i = 0; i < n_cs; i++) {
1625 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1627 caps = gst_caps_new_simple ("application/x-srtp",
1628 "ssrc", G_TYPE_UINT, map->ssrc,
1629 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1630 mikey_apply_policy (caps, msg, map->policy);
1632 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1633 gst_caps_unref (caps);
1635 gst_mikey_message_free (msg);
1642 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1647 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1648 goto cleanup_message;
1652 GST_DEBUG_OBJECT (client, "no crypto sessions");
1653 goto cleanup_message;
1657 GST_DEBUG_OBJECT (client, "no keys found");
1658 goto cleanup_message;
1660 unsupported_encryption:
1662 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1663 goto cleanup_message;
1667 gst_mikey_message_free (msg);
1672 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1675 strip_chars (gchar * str)
1682 if (!IS_STRIP_CHAR (str[len]))
1686 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1687 memmove (str, s, len + 1);
1691 * KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1692 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1695 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1700 specs = g_strsplit (keymgmt, ",", 0);
1701 for (i = 0; specs[i]; i++) {
1704 split = g_strsplit (specs[i], ";", 0);
1705 for (j = 0; split[j]; j++) {
1706 g_strstrip (split[j]);
1707 if (g_str_has_prefix (split[j], "prot=")) {
1708 g_strstrip (split[j] + 5);
1709 if (!g_str_equal (split[j] + 5, "mikey"))
1711 GST_DEBUG ("found mikey");
1712 } else if (g_str_has_prefix (split[j], "uri=")) {
1713 strip_chars (split[j] + 4);
1714 GST_DEBUG ("found uri '%s'", split[j] + 4);
1715 } else if (g_str_has_prefix (split[j], "data=")) {
1718 strip_chars (split[j] + 5);
1719 GST_DEBUG ("found data '%s'", split[j] + 5);
1720 data = g_base64_decode_inplace (split[j] + 5, &size);
1721 handle_mikey_data (client, ctx, data, size);
1729 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1731 GstRTSPClientPrivate *priv = client->priv;
1734 gchar *transport, *keymgmt;
1735 GstRTSPTransport *ct, *st;
1736 GstRTSPStatusCode code;
1737 GstRTSPSession *session;
1738 GstRTSPStreamTransport *trans;
1740 GstRTSPSessionMedia *sessmedia;
1741 GstRTSPMedia *media;
1742 GstRTSPStream *stream;
1743 GstRTSPState rtspstate;
1744 GstRTSPClientClass *klass;
1745 gchar *path, *control;
1752 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1753 path = klass->make_path_from_uri (client, uri);
1755 /* parse the transport */
1757 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1759 if (res != GST_RTSP_OK)
1762 /* we create the session after parsing stuff so that we don't make
1763 * a session for malformed requests */
1764 if (priv->session_pool == NULL)
1767 session = ctx->session;
1770 g_object_ref (session);
1771 /* get a handle to the configuration of the media in the session, this can
1772 * return NULL if this is a new url to manage in this session. */
1773 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1775 /* we need a new media configuration in this session */
1779 /* we have no session media, find one and manage it */
1780 if (sessmedia == NULL) {
1781 /* get a handle to the configuration of the media in the session */
1782 media = find_media (client, ctx, path, &matched);
1784 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1785 g_object_ref (media);
1787 goto media_not_found;
1789 /* no media, not found then */
1791 goto media_not_found_no_reply;
1793 if (path[matched] == '\0')
1794 goto control_not_found;
1796 /* path is what matched. */
1797 path[matched] = '\0';
1798 /* control is remainder */
1799 control = &path[matched + 1];
1801 /* find the stream now using the control part */
1802 stream = gst_rtsp_media_find_stream (media, control);
1804 goto stream_not_found;
1806 /* now we have a uri identifying a valid media and stream */
1807 ctx->stream = stream;
1810 if (session == NULL) {
1811 /* create a session if this fails we probably reached our session limit or
1813 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1814 goto service_unavailable;
1816 /* make sure this client is closed when the session is closed */
1817 client_watch_session (client, session);
1819 /* signal new session */
1820 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1823 ctx->session = session;
1826 if (sessmedia == NULL) {
1827 /* manage the media in our session now, if not done already */
1828 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1829 /* if we stil have no media, error */
1830 if (sessmedia == NULL)
1831 goto sessmedia_unavailable;
1833 g_object_unref (media);
1836 ctx->sessmedia = sessmedia;
1838 if (!klass->configure_client_media (client, media, stream, ctx))
1839 goto configure_media_failed_no_reply;
1841 gst_rtsp_transport_new (&ct);
1843 /* parse and find a usable supported transport */
1844 if (!parse_transport (transport, stream, ct))
1845 goto unsupported_transports;
1847 /* update the client transport */
1848 if (!klass->configure_client_transport (client, ctx, ct))
1849 goto unsupported_client_transport;
1851 /* parse the keymgmt */
1852 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1853 &keymgmt, 0) == GST_RTSP_OK) {
1854 if (!handle_keymgmt (client, ctx, keymgmt))
1858 /* set in the session media transport */
1859 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1861 /* configure the url used to set this transport, this we will use when
1862 * generating the response for the PLAY request */
1863 gst_rtsp_stream_transport_set_url (trans, uri);
1865 /* configure keepalive for this transport */
1866 gst_rtsp_stream_transport_set_keepalive (trans,
1867 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1869 /* create and serialize the server transport */
1870 st = make_server_transport (client, ctx, ct);
1871 trans_str = gst_rtsp_transport_as_text (st);
1872 gst_rtsp_transport_free (st);
1874 /* construct the response now */
1875 code = GST_RTSP_STS_OK;
1876 gst_rtsp_message_init_response (ctx->response, code,
1877 gst_rtsp_status_as_text (code), ctx->request);
1879 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1883 send_message (client, ctx, ctx->response, FALSE);
1885 /* update the state */
1886 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1887 switch (rtspstate) {
1888 case GST_RTSP_STATE_PLAYING:
1889 case GST_RTSP_STATE_RECORDING:
1890 case GST_RTSP_STATE_READY:
1891 /* no state change */
1894 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1897 g_object_unref (session);
1900 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1907 GST_ERROR ("client %p: no uri", client);
1908 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1913 GST_ERROR ("client %p: no transport", client);
1914 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1919 GST_ERROR ("client %p: no session pool configured", client);
1920 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1923 media_not_found_no_reply:
1925 GST_ERROR ("client %p: media '%s' not found", client, path);
1926 /* error reply is already sent */
1931 GST_ERROR ("client %p: media '%s' not found", client, path);
1932 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1937 GST_ERROR ("client %p: no control in path '%s'", client, path);
1938 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1939 g_object_unref (media);
1944 GST_ERROR ("client %p: stream '%s' not found", client, control);
1945 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1946 g_object_unref (media);
1949 service_unavailable:
1951 GST_ERROR ("client %p: can't create session", client);
1952 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1953 g_object_unref (media);
1956 sessmedia_unavailable:
1958 GST_ERROR ("client %p: can't create session media", client);
1959 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1960 g_object_unref (media);
1961 goto cleanup_session;
1963 configure_media_failed_no_reply:
1965 GST_ERROR ("client %p: configure_media failed", client);
1966 /* error reply is already sent */
1967 goto cleanup_session;
1969 unsupported_transports:
1971 GST_ERROR ("client %p: unsupported transports", client);
1972 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1973 goto cleanup_transport;
1975 unsupported_client_transport:
1977 GST_ERROR ("client %p: unsupported client transport", client);
1978 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1979 goto cleanup_transport;
1983 GST_ERROR ("client %p: keymgmt error", client);
1984 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
1985 goto cleanup_transport;
1989 gst_rtsp_transport_free (ct);
1991 g_object_unref (session);
1998 static GstSDPMessage *
1999 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
2001 GstRTSPClientPrivate *priv = client->priv;
2006 gst_sdp_message_new (&sdp);
2008 /* some standard things first */
2009 gst_sdp_message_set_version (sdp, "0");
2016 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
2019 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2020 gst_sdp_message_set_information (sdp, "rtsp-server");
2021 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2022 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2023 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2024 gst_sdp_message_add_attribute (sdp, "control", "*");
2026 info.is_ipv6 = priv->is_ipv6;
2027 info.server_ip = priv->server_ip;
2029 /* create an SDP for the media object */
2030 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2038 GST_ERROR ("client %p: could not create SDP", client);
2039 gst_sdp_message_free (sdp);
2044 /* for the describe we must generate an SDP */
2046 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2048 GstRTSPClientPrivate *priv = client->priv;
2053 GstRTSPMedia *media;
2054 GstRTSPClientClass *klass;
2056 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2061 /* check what kind of format is accepted, we don't really do anything with it
2062 * and always return SDP for now. */
2067 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2069 if (res == GST_RTSP_ENOTIMPL)
2072 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2076 if (!priv->mount_points)
2077 goto no_mount_points;
2079 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2082 /* find the media object for the uri */
2083 if (!(media = find_media (client, ctx, path, NULL)))
2086 /* create an SDP for the media object on this client */
2087 if (!(sdp = klass->create_sdp (client, media)))
2090 /* we suspend after the describe */
2091 gst_rtsp_media_suspend (media);
2092 g_object_unref (media);
2094 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2095 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2097 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2100 /* content base for some clients that might screw up creating the setup uri */
2101 str = make_base_url (client, ctx->uri, path);
2104 GST_INFO ("adding content-base: %s", str);
2105 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2107 /* add SDP to the response body */
2108 str = gst_sdp_message_as_text (sdp);
2109 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2110 gst_sdp_message_free (sdp);
2112 send_message (client, ctx, ctx->response, FALSE);
2114 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2122 GST_ERROR ("client %p: no uri", client);
2123 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2128 GST_ERROR ("client %p: no mount points configured", client);
2129 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2134 GST_ERROR ("client %p: can't find path for url", client);
2135 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2140 GST_ERROR ("client %p: no media", client);
2142 /* error reply is already sent */
2147 GST_ERROR ("client %p: can't create SDP", client);
2148 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2150 g_object_unref (media);
2156 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2158 GstRTSPMethod options;
2161 options = GST_RTSP_DESCRIBE |
2166 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2168 str = gst_rtsp_options_as_text (options);
2170 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2171 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2173 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2176 send_message (client, ctx, ctx->response, FALSE);
2178 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2184 /* remove duplicate and trailing '/' */
2186 sanitize_uri (GstRTSPUrl * uri)
2190 gboolean have_slash, prev_slash;
2192 s = d = uri->abspath;
2193 len = strlen (uri->abspath);
2197 for (i = 0; i < len; i++) {
2198 have_slash = s[i] == '/';
2200 if (!have_slash || !prev_slash)
2202 prev_slash = have_slash;
2204 len = d - uri->abspath;
2205 /* don't remove the first slash if that's the only thing left */
2206 if (len > 1 && *(d - 1) == '/')
2212 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
2214 GstRTSPClientPrivate *priv = client->priv;
2216 GST_INFO ("client %p: session %p finished", client, session);
2218 /* unlink all media managed in this session */
2219 client_unlink_session (client, session);
2221 /* remove the session */
2222 if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
2223 GST_INFO ("client %p: all sessions finalized, close the connection",
2225 close_connection (client);
2229 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2230 * and also returns a newly-allocated string of (comma-separated) unsupported
2231 * options in the unsupported_reqs variable .
2233 * There may be multiple Require headers, but we must send one single
2234 * Unsupported header with all the unsupported options as response. If
2235 * an incoming Require header contained a comma-separated list of options
2236 * GstRtspConnection will already have split that list up into multiple
2239 * TODO: allow the application to decide what features are supported
2242 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2245 GPtrArray *arr = NULL;
2251 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2253 if (res == GST_RTSP_ENOTIMPL)
2257 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2259 g_ptr_array_add (arr, g_strdup (reqs));
2263 /* if we don't have any Require headers at all, all is fine */
2267 /* otherwise we've now processed at all the Require headers */
2268 g_ptr_array_add (arr, NULL);
2270 /* for now we don't commit to supporting anything, so will just report
2271 * all of the required options as unsupported */
2272 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2274 g_ptr_array_unref (arr);
2279 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2281 GstRTSPClientPrivate *priv = client->priv;
2282 GstRTSPMethod method;
2283 const gchar *uristr;
2284 GstRTSPUrl *uri = NULL;
2285 GstRTSPVersion version;
2287 GstRTSPSession *session = NULL;
2288 GstRTSPContext sctx = { NULL }, *ctx;
2289 GstRTSPMessage response = { 0 };
2290 gchar *unsupported_reqs = NULL;
2293 if (!(ctx = gst_rtsp_context_get_current ())) {
2295 ctx->auth = priv->auth;
2296 gst_rtsp_context_push_current (ctx);
2299 ctx->conn = priv->connection;
2300 ctx->client = client;
2301 ctx->request = request;
2302 ctx->response = &response;
2304 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2305 gst_rtsp_message_dump (request);
2308 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2310 GST_INFO ("client %p: received a request %s %s %s", client,
2311 gst_rtsp_method_as_text (method), uristr,
2312 gst_rtsp_version_as_text (version));
2314 /* we can only handle 1.0 requests */
2315 if (version != GST_RTSP_VERSION_1_0)
2318 ctx->method = method;
2320 /* we always try to parse the url first */
2321 if (strcmp (uristr, "*") == 0) {
2322 /* special case where we have * as uri, keep uri = NULL */
2323 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2324 /* check if the uristr is an absolute path <=> scheme and host information
2328 scheme = g_uri_parse_scheme (uristr);
2329 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2330 gchar *absolute_uristr = NULL;
2332 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2333 if (priv->server_ip == NULL) {
2334 GST_WARNING_OBJECT (client, "host information missing");
2339 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2341 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2342 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2343 g_free (absolute_uristr);
2346 g_free (absolute_uristr);
2353 /* get the session if there is any */
2354 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2355 if (res == GST_RTSP_OK) {
2356 if (priv->session_pool == NULL)
2359 /* we had a session in the request, find it again */
2360 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2361 goto session_not_found;
2363 /* we add the session to the client list of watched sessions. When a session
2364 * disappears because it times out, we will be notified. If all sessions are
2365 * gone, we will close the connection */
2366 client_watch_session (client, session);
2369 /* sanitize the uri */
2373 ctx->session = session;
2375 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2376 goto not_authorized;
2378 /* handle any 'Require' headers */
2379 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2380 goto unsupported_requirement;
2382 /* now see what is asked and dispatch to a dedicated handler */
2384 case GST_RTSP_OPTIONS:
2385 handle_options_request (client, ctx);
2387 case GST_RTSP_DESCRIBE:
2388 handle_describe_request (client, ctx);
2390 case GST_RTSP_SETUP:
2391 handle_setup_request (client, ctx);
2394 handle_play_request (client, ctx);
2396 case GST_RTSP_PAUSE:
2397 handle_pause_request (client, ctx);
2399 case GST_RTSP_TEARDOWN:
2400 handle_teardown_request (client, ctx);
2402 case GST_RTSP_SET_PARAMETER:
2403 handle_set_param_request (client, ctx);
2405 case GST_RTSP_GET_PARAMETER:
2406 handle_get_param_request (client, ctx);
2408 case GST_RTSP_ANNOUNCE:
2409 case GST_RTSP_RECORD:
2410 case GST_RTSP_REDIRECT:
2411 goto not_implemented;
2412 case GST_RTSP_INVALID:
2419 gst_rtsp_context_pop_current (ctx);
2421 g_object_unref (session);
2423 gst_rtsp_url_free (uri);
2429 GST_ERROR ("client %p: version %d not supported", client, version);
2430 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2436 GST_ERROR ("client %p: bad request", client);
2437 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2442 GST_ERROR ("client %p: no pool configured", client);
2443 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2448 GST_ERROR ("client %p: session not found", client);
2449 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2454 GST_ERROR ("client %p: not allowed", client);
2455 /* error reply is already sent */
2458 unsupported_requirement:
2460 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2462 send_option_not_supported_response (client, ctx, unsupported_reqs);
2463 g_free (unsupported_reqs);
2468 GST_ERROR ("client %p: method %d not implemented", client, method);
2469 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2476 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2478 GstRTSPClientPrivate *priv = client->priv;
2480 GstRTSPSession *session = NULL;
2481 GstRTSPContext sctx = { NULL }, *ctx;
2484 if (!(ctx = gst_rtsp_context_get_current ())) {
2486 ctx->auth = priv->auth;
2487 gst_rtsp_context_push_current (ctx);
2490 ctx->conn = priv->connection;
2491 ctx->client = client;
2492 ctx->request = NULL;
2494 ctx->method = GST_RTSP_INVALID;
2495 ctx->response = response;
2497 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2498 gst_rtsp_message_dump (response);
2501 GST_INFO ("client %p: received a response", client);
2503 /* get the session if there is any */
2505 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2506 if (res == GST_RTSP_OK) {
2507 if (priv->session_pool == NULL)
2510 /* we had a session in the request, find it again */
2511 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2512 goto session_not_found;
2514 /* we add the session to the client list of watched sessions. When a session
2515 * disappears because it times out, we will be notified. If all sessions are
2516 * gone, we will close the connection */
2517 client_watch_session (client, session);
2520 ctx->session = session;
2522 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2527 gst_rtsp_context_pop_current (ctx);
2529 g_object_unref (session);
2534 GST_ERROR ("client %p: no pool configured", client);
2539 GST_ERROR ("client %p: session not found", client);
2545 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2547 GstRTSPClientPrivate *priv = client->priv;
2556 /* find the stream for this message */
2557 res = gst_rtsp_message_parse_data (message, &channel);
2558 if (res != GST_RTSP_OK)
2561 gst_rtsp_message_steal_body (message, &data, &size);
2563 buffer = gst_buffer_new_wrapped (data, size);
2566 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2567 GstRTSPStreamTransport *trans;
2568 GstRTSPStream *stream;
2569 const GstRTSPTransport *tr;
2573 tr = gst_rtsp_stream_transport_get_transport (trans);
2574 stream = gst_rtsp_stream_transport_get_stream (trans);
2576 /* check for TCP transport */
2577 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
2578 /* dispatch to the stream based on the channel number */
2579 if (tr->interleaved.min == channel) {
2580 gst_rtsp_stream_recv_rtp (stream, buffer);
2583 } else if (tr->interleaved.max == channel) {
2584 gst_rtsp_stream_recv_rtcp (stream, buffer);
2591 gst_buffer_unref (buffer);
2595 * gst_rtsp_client_set_session_pool:
2596 * @client: a #GstRTSPClient
2597 * @pool: (transfer none): a #GstRTSPSessionPool
2599 * Set @pool as the sessionpool for @client which it will use to find
2600 * or allocate sessions. the sessionpool is usually inherited from the server
2601 * that created the client but can be overridden later.
2604 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2605 GstRTSPSessionPool * pool)
2607 GstRTSPSessionPool *old;
2608 GstRTSPClientPrivate *priv;
2610 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2612 priv = client->priv;
2615 g_object_ref (pool);
2617 g_mutex_lock (&priv->lock);
2618 old = priv->session_pool;
2619 priv->session_pool = pool;
2620 g_mutex_unlock (&priv->lock);
2623 g_object_unref (old);
2627 * gst_rtsp_client_get_session_pool:
2628 * @client: a #GstRTSPClient
2630 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2632 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2634 GstRTSPSessionPool *
2635 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2637 GstRTSPClientPrivate *priv;
2638 GstRTSPSessionPool *result;
2640 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2642 priv = client->priv;
2644 g_mutex_lock (&priv->lock);
2645 if ((result = priv->session_pool))
2646 g_object_ref (result);
2647 g_mutex_unlock (&priv->lock);
2653 * gst_rtsp_client_set_mount_points:
2654 * @client: a #GstRTSPClient
2655 * @mounts: (transfer none): a #GstRTSPMountPoints
2657 * Set @mounts as the mount points for @client which it will use to map urls
2658 * to media streams. These mount points are usually inherited from the server that
2659 * created the client but can be overriden later.
2662 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2663 GstRTSPMountPoints * mounts)
2665 GstRTSPClientPrivate *priv;
2666 GstRTSPMountPoints *old;
2668 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2670 priv = client->priv;
2673 g_object_ref (mounts);
2675 g_mutex_lock (&priv->lock);
2676 old = priv->mount_points;
2677 priv->mount_points = mounts;
2678 g_mutex_unlock (&priv->lock);
2681 g_object_unref (old);
2685 * gst_rtsp_client_get_mount_points:
2686 * @client: a #GstRTSPClient
2688 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2690 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2692 GstRTSPMountPoints *
2693 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2695 GstRTSPClientPrivate *priv;
2696 GstRTSPMountPoints *result;
2698 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2700 priv = client->priv;
2702 g_mutex_lock (&priv->lock);
2703 if ((result = priv->mount_points))
2704 g_object_ref (result);
2705 g_mutex_unlock (&priv->lock);
2711 * gst_rtsp_client_set_auth:
2712 * @client: a #GstRTSPClient
2713 * @auth: (transfer none): a #GstRTSPAuth
2715 * configure @auth to be used as the authentication manager of @client.
2718 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2720 GstRTSPClientPrivate *priv;
2723 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2725 priv = client->priv;
2728 g_object_ref (auth);
2730 g_mutex_lock (&priv->lock);
2733 g_mutex_unlock (&priv->lock);
2736 g_object_unref (old);
2741 * gst_rtsp_client_get_auth:
2742 * @client: a #GstRTSPClient
2744 * Get the #GstRTSPAuth used as the authentication manager of @client.
2746 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2750 gst_rtsp_client_get_auth (GstRTSPClient * client)
2752 GstRTSPClientPrivate *priv;
2753 GstRTSPAuth *result;
2755 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2757 priv = client->priv;
2759 g_mutex_lock (&priv->lock);
2760 if ((result = priv->auth))
2761 g_object_ref (result);
2762 g_mutex_unlock (&priv->lock);
2768 * gst_rtsp_client_set_thread_pool:
2769 * @client: a #GstRTSPClient
2770 * @pool: (transfer none): a #GstRTSPThreadPool
2772 * configure @pool to be used as the thread pool of @client.
2775 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2776 GstRTSPThreadPool * pool)
2778 GstRTSPClientPrivate *priv;
2779 GstRTSPThreadPool *old;
2781 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2783 priv = client->priv;
2786 g_object_ref (pool);
2788 g_mutex_lock (&priv->lock);
2789 old = priv->thread_pool;
2790 priv->thread_pool = pool;
2791 g_mutex_unlock (&priv->lock);
2794 g_object_unref (old);
2798 * gst_rtsp_client_get_thread_pool:
2799 * @client: a #GstRTSPClient
2801 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2803 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2807 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2809 GstRTSPClientPrivate *priv;
2810 GstRTSPThreadPool *result;
2812 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2814 priv = client->priv;
2816 g_mutex_lock (&priv->lock);
2817 if ((result = priv->thread_pool))
2818 g_object_ref (result);
2819 g_mutex_unlock (&priv->lock);
2825 * gst_rtsp_client_set_connection:
2826 * @client: a #GstRTSPClient
2827 * @conn: (transfer full): a #GstRTSPConnection
2829 * Set the #GstRTSPConnection of @client. This function takes ownership of
2832 * Returns: %TRUE on success.
2835 gst_rtsp_client_set_connection (GstRTSPClient * client,
2836 GstRTSPConnection * conn)
2838 GstRTSPClientPrivate *priv;
2839 GSocket *read_socket;
2840 GSocketAddress *address;
2842 GError *error = NULL;
2844 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2845 g_return_val_if_fail (conn != NULL, FALSE);
2847 priv = client->priv;
2849 read_socket = gst_rtsp_connection_get_read_socket (conn);
2851 if (!(address = g_socket_get_local_address (read_socket, &error)))
2854 g_free (priv->server_ip);
2855 /* keep the original ip that the client connected to */
2856 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2857 GInetAddress *iaddr;
2859 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2861 /* socket might be ipv6 but adress still ipv4 */
2862 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2863 priv->server_ip = g_inet_address_to_string (iaddr);
2864 g_object_unref (address);
2866 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2867 priv->server_ip = g_strdup ("unknown");
2870 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2871 priv->server_ip, priv->is_ipv6);
2873 url = gst_rtsp_connection_get_url (conn);
2874 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2876 priv->connection = conn;
2883 GST_ERROR ("could not get local address %s", error->message);
2884 g_error_free (error);
2890 * gst_rtsp_client_get_connection:
2891 * @client: a #GstRTSPClient
2893 * Get the #GstRTSPConnection of @client.
2895 * Returns: (transfer none): the #GstRTSPConnection of @client.
2896 * The connection object returned remains valid until the client is freed.
2899 gst_rtsp_client_get_connection (GstRTSPClient * client)
2901 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2903 return client->priv->connection;
2907 * gst_rtsp_client_set_send_func:
2908 * @client: a #GstRTSPClient
2909 * @func: (scope notified): a #GstRTSPClientSendFunc
2910 * @user_data: (closure): user data passed to @func
2911 * @notify: (allow-none): called when @user_data is no longer in use
2913 * Set @func as the callback that will be called when a new message needs to be
2914 * sent to the client. @user_data is passed to @func and @notify is called when
2915 * @user_data is no longer in use.
2917 * By default, the client will send the messages on the #GstRTSPConnection that
2918 * was configured with gst_rtsp_client_attach() was called.
2921 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2922 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2924 GstRTSPClientPrivate *priv;
2925 GDestroyNotify old_notify;
2928 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2930 priv = client->priv;
2932 g_mutex_lock (&priv->send_lock);
2933 priv->send_func = func;
2934 old_notify = priv->send_notify;
2935 old_data = priv->send_data;
2936 priv->send_notify = notify;
2937 priv->send_data = user_data;
2938 g_mutex_unlock (&priv->send_lock);
2941 old_notify (old_data);
2945 * gst_rtsp_client_handle_message:
2946 * @client: a #GstRTSPClient
2947 * @message: (transfer none): an #GstRTSPMessage
2949 * Let the client handle @message.
2951 * Returns: a #GstRTSPResult.
2954 gst_rtsp_client_handle_message (GstRTSPClient * client,
2955 GstRTSPMessage * message)
2957 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2958 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2960 switch (message->type) {
2961 case GST_RTSP_MESSAGE_REQUEST:
2962 handle_request (client, message);
2964 case GST_RTSP_MESSAGE_RESPONSE:
2965 handle_response (client, message);
2967 case GST_RTSP_MESSAGE_DATA:
2968 handle_data (client, message);
2977 * gst_rtsp_client_send_message:
2978 * @client: a #GstRTSPClient
2979 * @session: (transfer none): a #GstRTSPSession to send the message to or %NULL
2980 * @message: (transfer none): The #GstRTSPMessage to send
2982 * Send a message message to the remote end. @message must be a
2983 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2986 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2987 GstRTSPMessage * message)
2989 GstRTSPContext sctx = { NULL }
2991 GstRTSPClientPrivate *priv;
2993 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2994 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2995 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
2996 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
2998 priv = client->priv;
3000 if (!(ctx = gst_rtsp_context_get_current ())) {
3002 ctx->auth = priv->auth;
3003 gst_rtsp_context_push_current (ctx);
3006 ctx->conn = priv->connection;
3007 ctx->client = client;
3008 ctx->session = session;
3010 send_message (client, ctx, message, FALSE);
3013 gst_rtsp_context_pop_current (ctx);
3018 static GstRTSPResult
3019 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
3020 gboolean close, gpointer user_data)
3022 GstRTSPClientPrivate *priv = client->priv;
3030 /* send the response and store the seq number so we can wait until it's
3031 * written to the client to close the connection */
3033 gst_rtsp_watch_send_message (priv->watch, message,
3034 close ? &priv->close_seq : NULL);
3035 if (ret == GST_RTSP_OK)
3038 if (ret != GST_RTSP_ENOMEM)
3042 if (priv->drop_backlog)
3045 /* queue was full, wait for more space */
3046 GST_DEBUG_OBJECT (client, "waiting for backlog");
3047 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3048 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3049 } while (ret != GST_RTSP_EINTR);
3056 GST_DEBUG_OBJECT (client, "got error %d", ret);
3061 static GstRTSPResult
3062 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3065 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3068 static GstRTSPResult
3069 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3071 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3072 GstRTSPClientPrivate *priv = client->priv;
3074 if (priv->close_seq && priv->close_seq == cseq) {
3075 priv->close_seq = 0;
3076 close_connection (client);
3082 static GstRTSPResult
3083 closed (GstRTSPWatch * watch, gpointer user_data)
3085 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3086 GstRTSPClientPrivate *priv = client->priv;
3087 const gchar *tunnelid;
3089 GST_INFO ("client %p: connection closed", client);
3091 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3092 g_mutex_lock (&tunnels_lock);
3093 /* remove from tunnelids */
3094 g_hash_table_remove (tunnels, tunnelid);
3095 g_mutex_unlock (&tunnels_lock);
3098 gst_rtsp_watch_set_flushing (watch, TRUE);
3099 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3104 static GstRTSPResult
3105 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3107 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3110 str = gst_rtsp_strresult (result);
3111 GST_INFO ("client %p: received an error %s", client, str);
3117 static GstRTSPResult
3118 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3119 GstRTSPMessage * message, guint id, gpointer user_data)
3121 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3124 str = gst_rtsp_strresult (result);
3126 ("client %p: error when handling message %p with id %d: %s",
3127 client, message, id, str);
3134 remember_tunnel (GstRTSPClient * client)
3136 GstRTSPClientPrivate *priv = client->priv;
3137 const gchar *tunnelid;
3139 /* store client in the pending tunnels */
3140 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3141 if (tunnelid == NULL)
3144 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3146 /* we can't have two clients connecting with the same tunnelid */
3147 g_mutex_lock (&tunnels_lock);
3148 if (g_hash_table_lookup (tunnels, tunnelid))
3149 goto tunnel_existed;
3151 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3152 g_mutex_unlock (&tunnels_lock);
3159 GST_ERROR ("client %p: no tunnelid provided", client);
3164 g_mutex_unlock (&tunnels_lock);
3165 GST_ERROR ("client %p: tunnel session %s already existed", client,
3171 static GstRTSPResult
3172 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3174 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3175 GstRTSPClientPrivate *priv = client->priv;
3177 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3180 /* ignore error, it'll only be a problem when the client does a POST again */
3181 remember_tunnel (client);
3187 handle_tunnel (GstRTSPClient * client)
3189 GstRTSPClientPrivate *priv = client->priv;
3190 GstRTSPClient *oclient;
3191 GstRTSPClientPrivate *opriv;
3192 const gchar *tunnelid;
3194 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3195 if (tunnelid == NULL)
3198 /* check for previous tunnel */
3199 g_mutex_lock (&tunnels_lock);
3200 oclient = g_hash_table_lookup (tunnels, tunnelid);
3202 if (oclient == NULL) {
3203 /* no previous tunnel, remember tunnel */
3204 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3205 g_mutex_unlock (&tunnels_lock);
3207 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3208 client, priv->connection);
3210 /* merge both tunnels into the first client */
3211 /* remove the old client from the table. ref before because removing it will
3212 * remove the ref to it. */
3213 g_object_ref (oclient);
3214 g_hash_table_remove (tunnels, tunnelid);
3215 g_mutex_unlock (&tunnels_lock);
3217 opriv = oclient->priv;
3219 if (opriv->watch == NULL)
3222 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3223 oclient, opriv->connection, priv->connection);
3225 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3226 gst_rtsp_watch_reset (priv->watch);
3227 gst_rtsp_watch_reset (opriv->watch);
3228 g_object_unref (oclient);
3230 /* the old client owns the tunnel now, the new one will be freed */
3231 g_source_destroy ((GSource *) priv->watch);
3233 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3241 GST_ERROR ("client %p: no tunnelid provided", client);
3246 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3247 g_object_unref (oclient);
3252 static GstRTSPStatusCode
3253 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3255 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3257 GST_INFO ("client %p: tunnel get (connection %p)", client,
3258 client->priv->connection);
3260 if (!handle_tunnel (client)) {
3261 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3264 return GST_RTSP_STS_OK;
3267 static GstRTSPResult
3268 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3270 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3272 GST_INFO ("client %p: tunnel post (connection %p)", client,
3273 client->priv->connection);
3275 if (!handle_tunnel (client)) {
3276 return GST_RTSP_ERROR;
3282 static GstRTSPResult
3283 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3284 GstRTSPMessage * response, gpointer user_data)
3286 GstRTSPClientClass *klass;
3288 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3289 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3291 if (klass->tunnel_http_response) {
3292 klass->tunnel_http_response (client, request, response);
3298 static GstRTSPWatchFuncs watch_funcs = {
3307 tunnel_http_response
3311 client_watch_notify (GstRTSPClient * client)
3313 GstRTSPClientPrivate *priv = client->priv;
3315 GST_INFO ("client %p: watch destroyed", client);
3317 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3318 g_object_unref (client);
3322 * gst_rtsp_client_attach:
3323 * @client: a #GstRTSPClient
3324 * @context: (allow-none): a #GMainContext
3326 * Attaches @client to @context. When the mainloop for @context is run, the
3327 * client will be dispatched. When @context is %NULL, the default context will be
3330 * This function should be called when the client properties and urls are fully
3331 * configured and the client is ready to start.
3333 * Returns: the ID (greater than 0) for the source within the GMainContext.
3336 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3338 GstRTSPClientPrivate *priv;
3341 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3342 priv = client->priv;
3343 g_return_val_if_fail (priv->connection != NULL, 0);
3344 g_return_val_if_fail (priv->watch == NULL, 0);
3346 /* create watch for the connection and attach */
3347 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3348 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3349 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3350 (GDestroyNotify) gst_rtsp_watch_unref);
3352 /* FIXME make this configurable. We don't want to do this yet because it will
3353 * be superceeded by a cache object later */
3354 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
3356 GST_INFO ("attaching to context %p", context);
3357 res = gst_rtsp_watch_attach (priv->watch, context);
3363 * gst_rtsp_client_session_filter:
3364 * @client: a #GstRTSPClient
3365 * @func: (scope call) (allow-none): a callback
3366 * @user_data: user data passed to @func
3368 * Call @func for each session managed by @client. The result value of @func
3369 * determines what happens to the session. @func will be called with @client
3370 * locked so no further actions on @client can be performed from @func.
3372 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3375 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3377 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3378 * will also be added with an additional ref to the result #GList of this
3381 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3383 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3384 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3385 * element in the #GList should be unreffed before the list is freed.
3388 gst_rtsp_client_session_filter (GstRTSPClient * client,
3389 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3391 GstRTSPClientPrivate *priv;
3392 GList *result, *walk, *next;
3394 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3396 priv = client->priv;
3400 g_mutex_lock (&priv->lock);
3401 for (walk = priv->sessions; walk; walk = next) {
3402 GstRTSPSession *sess = walk->data;
3403 GstRTSPFilterResult res;
3405 next = g_list_next (walk);
3408 res = func (client, sess, user_data);
3410 res = GST_RTSP_FILTER_REF;
3413 case GST_RTSP_FILTER_REMOVE:
3414 /* stop watching the session and pretent it went away */
3415 client_cleanup_session (client, sess);
3417 case GST_RTSP_FILTER_REF:
3418 result = g_list_prepend (result, g_object_ref (sess));
3420 case GST_RTSP_FILTER_KEEP:
3425 g_mutex_unlock (&priv->lock);