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.
23 #include "rtsp-client.h"
25 #include "rtsp-params.h"
27 #define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
28 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
31 * send_lock, lock, tunnels_lock
34 struct _GstRTSPClientPrivate
36 GMutex lock; /* protects everything else */
38 GstRTSPConnection *connection;
43 gboolean use_client_settings;
45 GstRTSPClientSendFunc send_func; /* protected by send_lock */
46 gpointer send_data; /* protected by send_lock */
47 GDestroyNotify send_notify; /* protected by send_lock */
49 GstRTSPSessionPool *session_pool;
50 GstRTSPMountPoints *mount_points;
60 static GMutex tunnels_lock;
61 static GHashTable *tunnels; /* protected by tunnels_lock */
63 #define DEFAULT_SESSION_POOL NULL
64 #define DEFAULT_MOUNT_POINTS NULL
65 #define DEFAULT_USE_CLIENT_SETTINGS FALSE
72 PROP_USE_CLIENT_SETTINGS,
80 SIGNAL_OPTIONS_REQUEST,
81 SIGNAL_DESCRIBE_REQUEST,
85 SIGNAL_TEARDOWN_REQUEST,
86 SIGNAL_SET_PARAMETER_REQUEST,
87 SIGNAL_GET_PARAMETER_REQUEST,
91 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
92 #define GST_CAT_DEFAULT rtsp_client_debug
94 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
96 static void gst_rtsp_client_get_property (GObject * object, guint propid,
97 GValue * value, GParamSpec * pspec);
98 static void gst_rtsp_client_set_property (GObject * object, guint propid,
99 const GValue * value, GParamSpec * pspec);
100 static void gst_rtsp_client_finalize (GObject * obj);
102 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
103 static void client_session_finalized (GstRTSPClient * client,
104 GstRTSPSession * session);
105 static void unlink_session_transports (GstRTSPClient * client,
106 GstRTSPSession * session, GstRTSPSessionMedia * media);
108 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
111 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
113 GObjectClass *gobject_class;
115 g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
117 gobject_class = G_OBJECT_CLASS (klass);
119 gobject_class->get_property = gst_rtsp_client_get_property;
120 gobject_class->set_property = gst_rtsp_client_set_property;
121 gobject_class->finalize = gst_rtsp_client_finalize;
123 klass->create_sdp = create_sdp;
125 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
126 g_param_spec_object ("session-pool", "Session Pool",
127 "The session pool to use for client session",
128 GST_TYPE_RTSP_SESSION_POOL,
129 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
131 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
132 g_param_spec_object ("mount-points", "Mount Points",
133 "The mount points to use for client session",
134 GST_TYPE_RTSP_MOUNT_POINTS,
135 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
137 g_object_class_install_property (gobject_class, PROP_USE_CLIENT_SETTINGS,
138 g_param_spec_boolean ("use-client-settings", "Use Client Settings",
139 "Use client settings for ttl and destination in multicast",
140 DEFAULT_USE_CLIENT_SETTINGS,
141 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
143 gst_rtsp_client_signals[SIGNAL_CLOSED] =
144 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
145 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
146 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
148 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
149 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
150 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
151 g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
153 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
154 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
155 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
156 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
159 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
160 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
161 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
162 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
165 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
166 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
167 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
168 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
171 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
172 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
173 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
174 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
177 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
178 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
179 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
180 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
183 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
184 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
185 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
186 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
189 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
190 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
191 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
192 set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
193 G_TYPE_NONE, 1, G_TYPE_POINTER);
195 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
196 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
197 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
198 get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
199 G_TYPE_NONE, 1, G_TYPE_POINTER);
202 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
203 g_mutex_init (&tunnels_lock);
205 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
209 gst_rtsp_client_init (GstRTSPClient * client)
211 GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
215 g_mutex_init (&priv->lock);
216 g_mutex_init (&priv->send_lock);
217 priv->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
221 static GstRTSPFilterResult
222 filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * media,
225 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
227 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
228 unlink_session_transports (client, sess, media);
230 /* unmanage the media in the session */
231 return GST_RTSP_FILTER_REMOVE;
235 client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
237 /* unlink all media managed in this session */
238 gst_rtsp_session_filter (session, filter_session, client);
242 client_cleanup_sessions (GstRTSPClient * client)
244 GstRTSPClientPrivate *priv = client->priv;
247 /* remove weak-ref from sessions */
248 for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
249 GstRTSPSession *session = (GstRTSPSession *) sessions->data;
250 g_object_weak_unref (G_OBJECT (session),
251 (GWeakNotify) client_session_finalized, client);
252 client_unlink_session (client, session);
254 g_list_free (priv->sessions);
255 priv->sessions = NULL;
258 /* A client is finalized when the connection is broken */
260 gst_rtsp_client_finalize (GObject * obj)
262 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
263 GstRTSPClientPrivate *priv = client->priv;
265 GST_INFO ("finalize client %p", client);
267 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
270 g_source_destroy ((GSource *) priv->watch);
272 client_cleanup_sessions (client);
274 if (priv->connection)
275 gst_rtsp_connection_free (priv->connection);
276 if (priv->session_pool)
277 g_object_unref (priv->session_pool);
278 if (priv->mount_points)
279 g_object_unref (priv->mount_points);
281 g_object_unref (priv->auth);
284 gst_rtsp_url_free (priv->uri);
286 gst_rtsp_media_unprepare (priv->media);
287 g_object_unref (priv->media);
290 g_free (priv->server_ip);
291 g_mutex_clear (&priv->lock);
292 g_mutex_clear (&priv->send_lock);
294 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
298 gst_rtsp_client_get_property (GObject * object, guint propid,
299 GValue * value, GParamSpec * pspec)
301 GstRTSPClient *client = GST_RTSP_CLIENT (object);
304 case PROP_SESSION_POOL:
305 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
307 case PROP_MOUNT_POINTS:
308 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
310 case PROP_USE_CLIENT_SETTINGS:
311 g_value_set_boolean (value,
312 gst_rtsp_client_get_use_client_settings (client));
315 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
320 gst_rtsp_client_set_property (GObject * object, guint propid,
321 const GValue * value, GParamSpec * pspec)
323 GstRTSPClient *client = GST_RTSP_CLIENT (object);
326 case PROP_SESSION_POOL:
327 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
329 case PROP_MOUNT_POINTS:
330 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
332 case PROP_USE_CLIENT_SETTINGS:
333 gst_rtsp_client_set_use_client_settings (client,
334 g_value_get_boolean (value));
337 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
342 * gst_rtsp_client_new:
344 * Create a new #GstRTSPClient instance.
346 * Returns: a new #GstRTSPClient
349 gst_rtsp_client_new (void)
351 GstRTSPClient *result;
353 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
359 send_response (GstRTSPClient * client, GstRTSPSession * session,
360 GstRTSPMessage * response, gboolean close)
362 GstRTSPClientPrivate *priv = client->priv;
364 gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
365 "GStreamer RTSP server");
367 /* remove any previous header */
368 gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
370 /* add the new session header for new session ids */
372 gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION,
373 gst_rtsp_session_get_header (session));
376 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
377 gst_rtsp_message_dump (response);
381 gst_rtsp_message_add_header (response, GST_RTSP_HDR_CONNECTION, "close");
383 g_mutex_lock (&priv->send_lock);
385 priv->send_func (client, response, close, priv->send_data);
386 g_mutex_unlock (&priv->send_lock);
388 gst_rtsp_message_unset (response);
392 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
393 GstRTSPClientState * state)
395 gst_rtsp_message_init_response (state->response, code,
396 gst_rtsp_status_as_text (code), state->request);
398 send_response (client, NULL, state->response, FALSE);
402 handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
403 GstRTSPClientState * state)
405 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
406 gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
409 /* and let the authentication manager setup the auth tokens */
410 gst_rtsp_auth_setup_auth (auth, client, 0, state);
413 send_response (client, state->session, state->response, FALSE);
418 compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
420 if (uri1 == NULL || uri2 == NULL)
423 if (strcmp (uri1->abspath, uri2->abspath))
429 /* this function is called to initially find the media for the DESCRIBE request
430 * but is cached for when the same client (without breaking the connection) is
431 * doing a setup for the exact same url. */
432 static GstRTSPMedia *
433 find_media (GstRTSPClient * client, GstRTSPClientState * state)
435 GstRTSPClientPrivate *priv = client->priv;
436 GstRTSPMediaFactory *factory;
440 if (!compare_uri (priv->uri, state->uri)) {
441 /* remove any previously cached values before we try to construct a new
444 gst_rtsp_url_free (priv->uri);
447 gst_rtsp_media_unprepare (priv->media);
448 g_object_unref (priv->media);
452 if (!priv->mount_points)
453 goto no_mount_points;
455 /* find the factory for the uri first */
457 gst_rtsp_mount_points_find_factory (priv->mount_points,
461 /* check if we have access to the factory */
462 if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
463 state->factory = factory;
465 if (!gst_rtsp_auth_check (auth, client, 0, state))
468 state->factory = NULL;
469 g_object_unref (auth);
472 /* prepare the media and add it to the pipeline */
473 if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
476 g_object_unref (factory);
479 /* prepare the media */
480 if (!(gst_rtsp_media_prepare (media)))
483 /* now keep track of the uri and the media */
484 priv->uri = gst_rtsp_url_copy (state->uri);
486 state->media = media;
488 /* we have seen this uri before, used cached media */
490 state->media = media;
491 GST_INFO ("reusing cached media %p", media);
495 g_object_ref (media);
502 GST_ERROR ("client %p: no mount points configured", client);
503 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
508 GST_ERROR ("client %p: no factory for uri", client);
509 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
514 GST_ERROR ("client %p: unauthorized request", client);
515 handle_unauthorized_request (client, auth, state);
516 g_object_unref (factory);
517 state->factory = NULL;
518 g_object_unref (auth);
523 GST_ERROR ("client %p: can't create media", client);
524 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
525 g_object_unref (factory);
530 GST_ERROR ("client %p: can't prepare media", client);
531 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
532 g_object_unref (media);
538 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
540 GstRTSPClientPrivate *priv = client->priv;
541 GstRTSPMessage message = { 0 };
546 gst_rtsp_message_init_data (&message, channel);
548 /* FIXME, need some sort of iovec RTSPMessage here */
549 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
552 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
554 g_mutex_lock (&priv->send_lock);
556 priv->send_func (client, &message, FALSE, priv->send_data);
557 g_mutex_unlock (&priv->send_lock);
559 gst_rtsp_message_steal_body (&message, &data, &usize);
560 gst_buffer_unmap (buffer, &map_info);
562 gst_rtsp_message_unset (&message);
568 link_transport (GstRTSPClient * client, GstRTSPSession * session,
569 GstRTSPStreamTransport * trans)
571 GstRTSPClientPrivate *priv = client->priv;
573 GST_DEBUG ("client %p: linking transport %p", client, trans);
575 gst_rtsp_stream_transport_set_callbacks (trans,
576 (GstRTSPSendFunc) do_send_data,
577 (GstRTSPSendFunc) do_send_data, client, NULL);
579 priv->transports = g_list_prepend (priv->transports, trans);
581 /* make sure our session can't expire */
582 gst_rtsp_session_prevent_expire (session);
586 unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
587 GstRTSPStreamTransport * trans)
589 GstRTSPClientPrivate *priv = client->priv;
591 GST_DEBUG ("client %p: unlinking transport %p", client, trans);
593 gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
595 priv->transports = g_list_remove (priv->transports, trans);
597 /* our session can now expire */
598 gst_rtsp_session_allow_expire (session);
602 unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
603 GstRTSPSessionMedia * media)
608 gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
609 for (i = 0; i < n_streams; i++) {
610 GstRTSPStreamTransport *trans;
611 const GstRTSPTransport *tr;
613 /* get the transport, if there is no transport configured, skip this stream */
614 trans = gst_rtsp_session_media_get_transport (media, i);
618 tr = gst_rtsp_stream_transport_get_transport (trans);
620 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
621 /* for TCP, unlink the stream from the TCP connection of the client */
622 unlink_transport (client, session, trans);
628 close_connection (GstRTSPClient * client)
630 GstRTSPClientPrivate *priv = client->priv;
631 const gchar *tunnelid;
633 GST_DEBUG ("client %p: closing connection", client);
635 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
636 g_mutex_lock (&tunnels_lock);
637 /* remove from tunnelids */
638 g_hash_table_remove (tunnels, tunnelid);
639 g_mutex_unlock (&tunnels_lock);
642 gst_rtsp_connection_close (priv->connection);
646 handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
648 GstRTSPClientPrivate *priv = client->priv;
649 GstRTSPSession *session;
650 GstRTSPSessionMedia *media;
651 GstRTSPStatusCode code;
656 session = state->session;
658 /* get a handle to the configuration of the media in the session */
659 media = gst_rtsp_session_get_media (session, state->uri);
663 state->sessmedia = media;
665 /* unlink the all TCP callbacks */
666 unlink_session_transports (client, session, media);
668 /* remove the session from the watched sessions */
669 g_object_weak_unref (G_OBJECT (session),
670 (GWeakNotify) client_session_finalized, client);
671 priv->sessions = g_list_remove (priv->sessions, session);
673 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
675 /* unmanage the media in the session, returns false if all media session
677 if (!gst_rtsp_session_release_media (session, media)) {
678 /* remove the session */
679 gst_rtsp_session_pool_remove (priv->session_pool, session);
681 /* construct the response now */
682 code = GST_RTSP_STS_OK;
683 gst_rtsp_message_init_response (state->response, code,
684 gst_rtsp_status_as_text (code), state->request);
686 send_response (client, session, state->response, TRUE);
688 /* we emit the signal before closing the connection */
689 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
697 GST_ERROR ("client %p: no session", client);
698 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
703 GST_ERROR ("client %p: no media for uri", client);
704 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
710 handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
716 res = gst_rtsp_message_get_body (state->request, &data, &size);
717 if (res != GST_RTSP_OK)
721 /* no body, keep-alive request */
722 send_generic_response (client, GST_RTSP_STS_OK, state);
724 /* there is a body, handle the params */
725 res = gst_rtsp_params_get (client, state);
726 if (res != GST_RTSP_OK)
729 send_response (client, state->session, state->response, FALSE);
732 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
740 GST_ERROR ("client %p: bad request", client);
741 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
747 handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
753 res = gst_rtsp_message_get_body (state->request, &data, &size);
754 if (res != GST_RTSP_OK)
758 /* no body, keep-alive request */
759 send_generic_response (client, GST_RTSP_STS_OK, state);
761 /* there is a body, handle the params */
762 res = gst_rtsp_params_set (client, state);
763 if (res != GST_RTSP_OK)
766 send_response (client, state->session, state->response, FALSE);
769 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
777 GST_ERROR ("client %p: bad request", client);
778 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
784 handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
786 GstRTSPSession *session;
787 GstRTSPSessionMedia *media;
788 GstRTSPStatusCode code;
789 GstRTSPState rtspstate;
791 if (!(session = state->session))
794 /* get a handle to the configuration of the media in the session */
795 media = gst_rtsp_session_get_media (session, state->uri);
799 state->sessmedia = media;
801 rtspstate = gst_rtsp_session_media_get_rtsp_state (media);
802 /* the session state must be playing or recording */
803 if (rtspstate != GST_RTSP_STATE_PLAYING &&
804 rtspstate != GST_RTSP_STATE_RECORDING)
807 /* unlink the all TCP callbacks */
808 unlink_session_transports (client, session, media);
810 /* then pause sending */
811 gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
813 /* construct the response now */
814 code = GST_RTSP_STS_OK;
815 gst_rtsp_message_init_response (state->response, code,
816 gst_rtsp_status_as_text (code), state->request);
818 send_response (client, session, state->response, FALSE);
820 /* the state is now READY */
821 gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_READY);
823 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
831 GST_ERROR ("client %p: no seesion", client);
832 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
837 GST_ERROR ("client %p: no media for uri", client);
838 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
843 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
844 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
851 handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
853 GstRTSPSession *session;
854 GstRTSPSessionMedia *media;
855 GstRTSPStatusCode code;
857 guint n_streams, i, infocount;
859 GstRTSPTimeRange *range;
861 GstRTSPState rtspstate;
863 if (!(session = state->session))
866 /* get a handle to the configuration of the media in the session */
867 media = gst_rtsp_session_get_media (session, state->uri);
871 state->sessmedia = media;
873 /* the session state must be playing or ready */
874 rtspstate = gst_rtsp_session_media_get_rtsp_state (media);
875 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
878 /* parse the range header if we have one */
880 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
881 if (res == GST_RTSP_OK) {
882 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
883 /* we have a range, seek to the position */
884 gst_rtsp_media_seek (gst_rtsp_session_media_get_media (media), range);
885 gst_rtsp_range_free (range);
889 /* grab RTPInfo from the payloaders now */
890 rtpinfo = g_string_new ("");
893 gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
894 for (i = 0, infocount = 0; i < n_streams; i++) {
895 GstRTSPStreamTransport *trans;
896 GstRTSPStream *stream;
897 const GstRTSPTransport *tr;
901 /* get the transport, if there is no transport configured, skip this stream */
902 trans = gst_rtsp_session_media_get_transport (media, i);
904 GST_INFO ("stream %d is not configured", i);
907 tr = gst_rtsp_stream_transport_get_transport (trans);
909 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
910 /* for TCP, link the stream to the TCP connection of the client */
911 link_transport (client, session, trans);
914 stream = gst_rtsp_stream_transport_get_stream (trans);
915 if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
917 g_string_append (rtpinfo, ", ");
919 uristr = gst_rtsp_url_get_request_uri (state->uri);
920 g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
921 uristr, i, seq, rtptime);
926 GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
930 /* construct the response now */
931 code = GST_RTSP_STS_OK;
932 gst_rtsp_message_init_response (state->response, code,
933 gst_rtsp_status_as_text (code), state->request);
935 /* add the RTP-Info header */
937 str = g_string_free (rtpinfo, FALSE);
938 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
940 g_string_free (rtpinfo, TRUE);
945 gst_rtsp_media_get_range_string (gst_rtsp_session_media_get_media (media),
947 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
949 send_response (client, session, state->response, FALSE);
951 /* start playing after sending the request */
952 gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
954 gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_PLAYING);
956 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
964 GST_ERROR ("client %p: no session", client);
965 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
970 GST_ERROR ("client %p: media not found", client);
971 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
976 GST_ERROR ("client %p: not PLAYING or READY", client);
977 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
984 do_keepalive (GstRTSPSession * session)
986 GST_INFO ("keep session %p alive", session);
987 gst_rtsp_session_touch (session);
990 /* parse @transport and return a valid transport in @tr. only transports
991 * from @supported are returned. Returns FALSE if no valid transport
994 parse_transport (const char *transport, GstRTSPLowerTrans supported,
995 GstRTSPTransport * tr)
1002 gst_rtsp_transport_init (tr);
1004 GST_DEBUG ("parsing transports %s", transport);
1006 transports = g_strsplit (transport, ",", 0);
1008 /* loop through the transports, try to parse */
1009 for (i = 0; transports[i]; i++) {
1010 res = gst_rtsp_transport_parse (transports[i], tr);
1011 if (res != GST_RTSP_OK) {
1012 /* no valid transport, search some more */
1013 GST_WARNING ("could not parse transport %s", transports[i]);
1017 /* we have a transport, see if it's RTP/AVP */
1018 if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
1019 GST_WARNING ("invalid transport %s", transports[i]);
1023 if (!(tr->lower_transport & supported)) {
1024 GST_WARNING ("unsupported transport %s", transports[i]);
1028 /* we have a valid transport */
1029 GST_INFO ("found valid transport %s", transports[i]);
1034 gst_rtsp_transport_init (tr);
1036 g_strfreev (transports);
1042 handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
1043 GstRTSPMessage * request)
1045 gchar *blocksize_str;
1046 gboolean ret = TRUE;
1048 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1049 &blocksize_str, 0) == GST_RTSP_OK) {
1053 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1054 if (end == blocksize_str) {
1055 GST_ERROR ("failed to parse blocksize");
1058 /* we don't want to change the mtu when this media
1059 * can be shared because it impacts other clients */
1060 if (gst_rtsp_media_is_shared (media))
1063 if (blocksize > G_MAXUINT)
1064 blocksize = G_MAXUINT;
1065 gst_rtsp_stream_set_mtu (stream, blocksize);
1072 configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state,
1073 GstRTSPTransport * ct)
1075 GstRTSPClientPrivate *priv = client->priv;
1077 /* we have a valid transport now, set the destination of the client. */
1078 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1079 if (ct->destination && priv->use_client_settings) {
1080 GstRTSPAddress *addr;
1082 addr = gst_rtsp_stream_reserve_address (state->stream, ct->destination,
1083 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1088 gst_rtsp_address_free (addr);
1090 GstRTSPAddress *addr;
1092 addr = gst_rtsp_stream_get_address (state->stream);
1096 g_free (ct->destination);
1097 ct->destination = g_strdup (addr->address);
1098 ct->port.min = addr->port;
1099 ct->port.max = addr->port + addr->n_ports - 1;
1100 ct->ttl = addr->ttl;
1102 gst_rtsp_address_free (addr);
1107 url = gst_rtsp_connection_get_url (priv->connection);
1108 g_free (ct->destination);
1109 ct->destination = g_strdup (url->host);
1111 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1112 /* check if the client selected channels for TCP */
1113 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1114 gst_rtsp_session_media_alloc_channels (state->sessmedia,
1124 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1129 static GstRTSPTransport *
1130 make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
1131 GstRTSPTransport * ct)
1133 GstRTSPTransport *st;
1135 /* prepare the server transport */
1136 gst_rtsp_transport_new (&st);
1138 st->trans = ct->trans;
1139 st->profile = ct->profile;
1140 st->lower_transport = ct->lower_transport;
1142 switch (st->lower_transport) {
1143 case GST_RTSP_LOWER_TRANS_UDP:
1144 st->client_port = ct->client_port;
1145 gst_rtsp_stream_get_server_port (state->stream, &st->server_port);
1147 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1148 st->port = ct->port;
1149 st->destination = g_strdup (ct->destination);
1152 case GST_RTSP_LOWER_TRANS_TCP:
1153 st->interleaved = ct->interleaved;
1158 gst_rtsp_stream_get_ssrc (state->stream, &st->ssrc);
1164 handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
1166 GstRTSPClientPrivate *priv = client->priv;
1170 GstRTSPTransport *ct, *st;
1171 GstRTSPLowerTrans supported;
1172 GstRTSPStatusCode code;
1173 GstRTSPSession *session;
1174 GstRTSPStreamTransport *trans;
1175 gchar *trans_str, *pos;
1177 GstRTSPSessionMedia *sessmedia;
1178 GstRTSPMedia *media;
1179 GstRTSPStream *stream;
1180 GstRTSPState rtspstate;
1184 /* the uri contains the stream number we added in the SDP config, which is
1185 * always /stream=%d so we need to strip that off
1186 * parse the stream we need to configure, look for the stream in the abspath
1187 * first and then in the query. */
1188 if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
1189 if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
1193 /* we can mofify the parsed uri in place */
1196 pos += strlen ("/stream=");
1197 if (sscanf (pos, "%u", &streamid) != 1)
1200 /* parse the transport */
1202 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
1204 if (res != GST_RTSP_OK)
1207 gst_rtsp_transport_new (&ct);
1209 /* our supported transports */
1210 supported = GST_RTSP_LOWER_TRANS_UDP |
1211 GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
1213 /* parse and find a usable supported transport */
1214 if (!parse_transport (transport, supported, ct))
1215 goto unsupported_transports;
1217 /* we create the session after parsing stuff so that we don't make
1218 * a session for malformed requests */
1219 if (priv->session_pool == NULL)
1222 session = state->session;
1225 g_object_ref (session);
1226 /* get a handle to the configuration of the media in the session, this can
1227 * return NULL if this is a new url to manage in this session. */
1228 sessmedia = gst_rtsp_session_get_media (session, uri);
1230 /* create a session if this fails we probably reached our session limit or
1232 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1233 goto service_unavailable;
1235 state->session = session;
1237 /* we need a new media configuration in this session */
1241 /* we have no media, find one and manage it */
1242 if (sessmedia == NULL) {
1243 /* get a handle to the configuration of the media in the session */
1244 if ((media = find_media (client, state))) {
1245 /* manage the media in our session now */
1246 sessmedia = gst_rtsp_session_manage_media (session, uri, media);
1250 /* if we stil have no media, error */
1251 if (sessmedia == NULL)
1254 state->sessmedia = sessmedia;
1255 state->media = media = gst_rtsp_session_media_get_media (sessmedia);
1257 /* now get the stream */
1258 stream = gst_rtsp_media_get_stream (media, streamid);
1262 state->stream = stream;
1264 /* set blocksize on this stream */
1265 if (!handle_blocksize (media, stream, state->request))
1266 goto invalid_blocksize;
1268 /* update the client transport */
1269 if (!configure_client_transport (client, state, ct))
1270 goto unsupported_client_transport;
1272 /* set in the session media transport */
1273 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1275 /* configure keepalive for this transport */
1276 gst_rtsp_stream_transport_set_keepalive (trans,
1277 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1279 /* create and serialize the server transport */
1280 st = make_server_transport (client, state, ct);
1281 trans_str = gst_rtsp_transport_as_text (st);
1282 gst_rtsp_transport_free (st);
1284 /* construct the response now */
1285 code = GST_RTSP_STS_OK;
1286 gst_rtsp_message_init_response (state->response, code,
1287 gst_rtsp_status_as_text (code), state->request);
1289 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
1293 send_response (client, session, state->response, FALSE);
1295 /* update the state */
1296 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1297 switch (rtspstate) {
1298 case GST_RTSP_STATE_PLAYING:
1299 case GST_RTSP_STATE_RECORDING:
1300 case GST_RTSP_STATE_READY:
1301 /* no state change */
1304 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1307 g_object_unref (session);
1309 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
1317 GST_ERROR ("client %p: bad request", client);
1318 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1323 GST_ERROR ("client %p: media not found", client);
1324 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1325 g_object_unref (session);
1326 gst_rtsp_transport_free (ct);
1331 GST_ERROR ("client %p: invalid blocksize", client);
1332 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1333 g_object_unref (session);
1334 gst_rtsp_transport_free (ct);
1337 unsupported_client_transport:
1339 GST_ERROR ("client %p: unsupported client transport", client);
1340 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1341 g_object_unref (session);
1342 gst_rtsp_transport_free (ct);
1347 GST_ERROR ("client %p: no transport", client);
1348 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1351 unsupported_transports:
1353 GST_ERROR ("client %p: unsupported transports", client);
1354 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1355 gst_rtsp_transport_free (ct);
1360 GST_ERROR ("client %p: no session pool configured", client);
1361 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
1362 gst_rtsp_transport_free (ct);
1365 service_unavailable:
1367 GST_ERROR ("client %p: can't create session", client);
1368 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1369 gst_rtsp_transport_free (ct);
1374 static GstSDPMessage *
1375 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1377 GstRTSPClientPrivate *priv = client->priv;
1382 gst_sdp_message_new (&sdp);
1384 /* some standard things first */
1385 gst_sdp_message_set_version (sdp, "0");
1392 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
1395 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
1396 gst_sdp_message_set_information (sdp, "rtsp-server");
1397 gst_sdp_message_add_time (sdp, "0", "0", NULL);
1398 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
1399 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
1400 gst_sdp_message_add_attribute (sdp, "control", "*");
1402 info.server_proto = proto;
1403 info.server_ip = g_strdup (priv->server_ip);
1405 /* create an SDP for the media object */
1406 if (!gst_rtsp_sdp_from_media (sdp, &info, media))
1409 g_free (info.server_ip);
1416 GST_ERROR ("client %p: could not create SDP", client);
1417 g_free (info.server_ip);
1418 gst_sdp_message_free (sdp);
1423 /* for the describe we must generate an SDP */
1425 handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
1430 gchar *str, *content_base;
1431 GstRTSPMedia *media;
1432 GstRTSPClientClass *klass;
1434 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1436 /* check what kind of format is accepted, we don't really do anything with it
1437 * and always return SDP for now. */
1442 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
1444 if (res == GST_RTSP_ENOTIMPL)
1447 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
1451 /* find the media object for the uri */
1452 if (!(media = find_media (client, state)))
1455 /* create an SDP for the media object on this client */
1456 if (!(sdp = klass->create_sdp (client, media)))
1459 g_object_unref (media);
1461 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1462 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1464 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
1467 /* content base for some clients that might screw up creating the setup uri */
1468 str = gst_rtsp_url_get_request_uri (state->uri);
1469 str_len = strlen (str);
1471 /* check for trailing '/' and append one */
1472 if (str[str_len - 1] != '/') {
1473 content_base = g_malloc (str_len + 2);
1474 memcpy (content_base, str, str_len);
1475 content_base[str_len] = '/';
1476 content_base[str_len + 1] = '\0';
1482 GST_INFO ("adding content-base: %s", content_base);
1484 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
1486 g_free (content_base);
1488 /* add SDP to the response body */
1489 str = gst_sdp_message_as_text (sdp);
1490 gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
1491 gst_sdp_message_free (sdp);
1493 send_response (client, state->session, state->response, FALSE);
1495 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
1503 GST_ERROR ("client %p: no media", client);
1504 /* error reply is already sent */
1509 GST_ERROR ("client %p: can't create SDP", client);
1510 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1511 g_object_unref (media);
1517 handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
1519 GstRTSPMethod options;
1522 options = GST_RTSP_DESCRIBE |
1527 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
1529 str = gst_rtsp_options_as_text (options);
1531 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1532 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1534 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
1537 send_response (client, state->session, state->response, FALSE);
1539 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
1545 /* remove duplicate and trailing '/' */
1547 sanitize_uri (GstRTSPUrl * uri)
1551 gboolean have_slash, prev_slash;
1553 s = d = uri->abspath;
1554 len = strlen (uri->abspath);
1558 for (i = 0; i < len; i++) {
1559 have_slash = s[i] == '/';
1561 if (!have_slash || !prev_slash)
1563 prev_slash = have_slash;
1565 len = d - uri->abspath;
1566 /* don't remove the first slash if that's the only thing left */
1567 if (len > 1 && *(d - 1) == '/')
1573 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
1575 GstRTSPClientPrivate *priv = client->priv;
1577 GST_INFO ("client %p: session %p finished", client, session);
1579 /* unlink all media managed in this session */
1580 client_unlink_session (client, session);
1582 /* remove the session */
1583 if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
1584 GST_INFO ("client %p: all sessions finalized, close the connection",
1586 close_connection (client);
1591 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
1593 GstRTSPClientPrivate *priv = client->priv;
1596 for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
1597 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
1599 /* we already know about this session */
1600 if (msession == session)
1604 GST_INFO ("watching session %p", session);
1606 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
1608 priv->sessions = g_list_prepend (priv->sessions, session);
1610 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1615 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
1617 GstRTSPClientPrivate *priv = client->priv;
1618 GstRTSPMethod method;
1619 const gchar *uristr;
1620 GstRTSPUrl *uri = NULL;
1621 GstRTSPVersion version;
1623 GstRTSPSession *session = NULL;
1624 GstRTSPClientState state = { NULL };
1625 GstRTSPMessage response = { 0 };
1628 state.request = request;
1629 state.response = &response;
1631 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
1632 gst_rtsp_message_dump (request);
1635 GST_INFO ("client %p: received a request", client);
1637 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
1639 /* we can only handle 1.0 requests */
1640 if (version != GST_RTSP_VERSION_1_0)
1643 state.method = method;
1645 /* we always try to parse the url first */
1646 if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK)
1649 /* get the session if there is any */
1650 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
1651 if (res == GST_RTSP_OK) {
1652 if (priv->session_pool == NULL)
1655 /* we had a session in the request, find it again */
1656 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
1657 goto session_not_found;
1659 /* we add the session to the client list of watched sessions. When a session
1660 * disappears because it times out, we will be notified. If all sessions are
1661 * gone, we will close the connection */
1662 client_watch_session (client, session);
1665 /* sanitize the uri */
1668 state.session = session;
1671 if (!gst_rtsp_auth_check (priv->auth, client, 0, &state))
1672 goto not_authorized;
1675 /* now see what is asked and dispatch to a dedicated handler */
1677 case GST_RTSP_OPTIONS:
1678 handle_options_request (client, &state);
1680 case GST_RTSP_DESCRIBE:
1681 handle_describe_request (client, &state);
1683 case GST_RTSP_SETUP:
1684 handle_setup_request (client, &state);
1687 handle_play_request (client, &state);
1689 case GST_RTSP_PAUSE:
1690 handle_pause_request (client, &state);
1692 case GST_RTSP_TEARDOWN:
1693 handle_teardown_request (client, &state);
1695 case GST_RTSP_SET_PARAMETER:
1696 handle_set_param_request (client, &state);
1698 case GST_RTSP_GET_PARAMETER:
1699 handle_get_param_request (client, &state);
1701 case GST_RTSP_ANNOUNCE:
1702 case GST_RTSP_RECORD:
1703 case GST_RTSP_REDIRECT:
1704 goto not_implemented;
1705 case GST_RTSP_INVALID:
1712 g_object_unref (session);
1714 gst_rtsp_url_free (uri);
1720 GST_ERROR ("client %p: version %d not supported", client, version);
1721 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
1727 GST_ERROR ("client %p: bad request", client);
1728 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1733 GST_ERROR ("client %p: no pool configured", client);
1734 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
1739 GST_ERROR ("client %p: session not found", client);
1740 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
1745 GST_ERROR ("client %p: not allowed", client);
1746 handle_unauthorized_request (client, priv->auth, &state);
1751 GST_ERROR ("client %p: method %d not implemented", client, method);
1752 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
1758 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
1760 GstRTSPClientPrivate *priv = client->priv;
1769 /* find the stream for this message */
1770 res = gst_rtsp_message_parse_data (message, &channel);
1771 if (res != GST_RTSP_OK)
1774 gst_rtsp_message_steal_body (message, &data, &size);
1776 buffer = gst_buffer_new_wrapped (data, size);
1779 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1780 GstRTSPStreamTransport *trans;
1781 GstRTSPStream *stream;
1782 const GstRTSPTransport *tr;
1786 tr = gst_rtsp_stream_transport_get_transport (trans);
1787 stream = gst_rtsp_stream_transport_get_stream (trans);
1789 /* check for TCP transport */
1790 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1791 /* dispatch to the stream based on the channel number */
1792 if (tr->interleaved.min == channel) {
1793 gst_rtsp_stream_recv_rtp (stream, buffer);
1796 } else if (tr->interleaved.max == channel) {
1797 gst_rtsp_stream_recv_rtcp (stream, buffer);
1804 gst_buffer_unref (buffer);
1808 * gst_rtsp_client_set_session_pool:
1809 * @client: a #GstRTSPClient
1810 * @pool: a #GstRTSPSessionPool
1812 * Set @pool as the sessionpool for @client which it will use to find
1813 * or allocate sessions. the sessionpool is usually inherited from the server
1814 * that created the client but can be overridden later.
1817 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
1818 GstRTSPSessionPool * pool)
1820 GstRTSPSessionPool *old;
1821 GstRTSPClientPrivate *priv;
1823 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1825 priv = client->priv;
1828 g_object_ref (pool);
1830 g_mutex_lock (&priv->lock);
1831 old = priv->session_pool;
1832 priv->session_pool = pool;
1833 g_mutex_unlock (&priv->lock);
1836 g_object_unref (old);
1840 * gst_rtsp_client_get_session_pool:
1841 * @client: a #GstRTSPClient
1843 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
1845 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
1847 GstRTSPSessionPool *
1848 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
1850 GstRTSPClientPrivate *priv;
1851 GstRTSPSessionPool *result;
1853 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
1855 priv = client->priv;
1857 g_mutex_lock (&priv->lock);
1858 if ((result = priv->session_pool))
1859 g_object_ref (result);
1860 g_mutex_unlock (&priv->lock);
1866 * gst_rtsp_client_set_mount_points:
1867 * @client: a #GstRTSPClient
1868 * @mounts: a #GstRTSPMountPoints
1870 * Set @mounts as the mount points for @client which it will use to map urls
1871 * to media streams. These mount points are usually inherited from the server that
1872 * created the client but can be overriden later.
1875 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
1876 GstRTSPMountPoints * mounts)
1878 GstRTSPClientPrivate *priv;
1879 GstRTSPMountPoints *old;
1881 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1883 priv = client->priv;
1886 g_object_ref (mounts);
1888 g_mutex_lock (&priv->lock);
1889 old = priv->mount_points;
1890 priv->mount_points = mounts;
1891 g_mutex_unlock (&priv->lock);
1894 g_object_unref (old);
1898 * gst_rtsp_client_get_mount_points:
1899 * @client: a #GstRTSPClient
1901 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
1903 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
1905 GstRTSPMountPoints *
1906 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
1908 GstRTSPClientPrivate *priv;
1909 GstRTSPMountPoints *result;
1911 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
1913 priv = client->priv;
1915 g_mutex_lock (&priv->lock);
1916 if ((result = priv->mount_points))
1917 g_object_ref (result);
1918 g_mutex_unlock (&priv->lock);
1924 * gst_rtsp_client_set_use_client_settings:
1925 * @client: a #GstRTSPClient
1926 * @use_client_settings: whether to use client settings for multicast
1928 * Use client transport settings (destination and ttl) for multicast.
1929 * When @use_client_settings is %FALSE, the server settings will be
1933 gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
1934 gboolean use_client_settings)
1936 GstRTSPClientPrivate *priv;
1938 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1940 priv = client->priv;
1942 g_mutex_lock (&priv->lock);
1943 priv->use_client_settings = use_client_settings;
1944 g_mutex_unlock (&priv->lock);
1948 * gst_rtsp_client_get_use_client_settings:
1949 * @client: a #GstRTSPClient
1951 * Check if client transport settings (destination and ttl) for multicast
1955 gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
1957 GstRTSPClientPrivate *priv;
1960 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
1962 priv = client->priv;
1964 g_mutex_lock (&priv->lock);
1965 res = priv->use_client_settings;
1966 g_mutex_unlock (&priv->lock);
1972 * gst_rtsp_client_set_auth:
1973 * @client: a #GstRTSPClient
1974 * @auth: a #GstRTSPAuth
1976 * configure @auth to be used as the authentication manager of @client.
1979 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
1981 GstRTSPClientPrivate *priv;
1984 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1986 priv = client->priv;
1989 g_object_ref (auth);
1991 g_mutex_lock (&priv->lock);
1994 g_mutex_unlock (&priv->lock);
1997 g_object_unref (old);
2002 * gst_rtsp_client_get_auth:
2003 * @client: a #GstRTSPClient
2005 * Get the #GstRTSPAuth used as the authentication manager of @client.
2007 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2011 gst_rtsp_client_get_auth (GstRTSPClient * client)
2013 GstRTSPClientPrivate *priv;
2014 GstRTSPAuth *result;
2016 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2018 priv = client->priv;
2020 g_mutex_lock (&priv->lock);
2021 if ((result = priv->auth))
2022 g_object_ref (result);
2023 g_mutex_unlock (&priv->lock);
2029 * gst_rtsp_client_set_send_func:
2030 * @client: a #GstRTSPClient
2031 * @func: a #GstRTSPClientSendFunc
2032 * @user_data: user data passed to @func
2033 * @notify: called when @user_data is no longer in use
2035 * Set @func as the callback that will be called when a new message needs to be
2036 * sent to the client. @user_data is passed to @func and @notify is called when
2037 * @user_data is no longer in use.
2040 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2041 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2043 GstRTSPClientPrivate *priv;
2044 GDestroyNotify old_notify;
2047 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2049 priv = client->priv;
2051 g_mutex_lock (&priv->send_lock);
2052 priv->send_func = func;
2053 old_notify = priv->send_notify;
2054 old_data = priv->send_data;
2055 priv->send_notify = notify;
2056 priv->send_data = user_data;
2057 g_mutex_unlock (&priv->send_lock);
2060 old_notify (old_data);
2064 * gst_rtsp_client_handle_message:
2065 * @client: a #GstRTSPClient
2066 * @message: an #GstRTSPMessage
2068 * Let the client handle @message.
2070 * Returns: a #GstRTSPResult.
2073 gst_rtsp_client_handle_message (GstRTSPClient * client,
2074 GstRTSPMessage * message)
2076 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2077 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2079 switch (message->type) {
2080 case GST_RTSP_MESSAGE_REQUEST:
2081 handle_request (client, message);
2083 case GST_RTSP_MESSAGE_RESPONSE:
2085 case GST_RTSP_MESSAGE_DATA:
2086 handle_data (client, message);
2094 static GstRTSPResult
2095 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
2096 gboolean close, gpointer user_data)
2098 GstRTSPClientPrivate *priv = client->priv;
2100 /* send the response and store the seq number so we can wait until it's
2101 * written to the client to close the connection */
2102 return gst_rtsp_watch_send_message (priv->watch, message, close ?
2103 &priv->close_seq : NULL);
2106 static GstRTSPResult
2107 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
2110 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
2113 static GstRTSPResult
2114 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
2116 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2117 GstRTSPClientPrivate *priv = client->priv;
2119 if (priv->close_seq && priv->close_seq == cseq) {
2120 priv->close_seq = 0;
2121 close_connection (client);
2127 static GstRTSPResult
2128 closed (GstRTSPWatch * watch, gpointer user_data)
2130 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2131 GstRTSPClientPrivate *priv = client->priv;
2132 const gchar *tunnelid;
2134 GST_INFO ("client %p: connection closed", client);
2136 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
2137 g_mutex_lock (&tunnels_lock);
2138 /* remove from tunnelids */
2139 g_hash_table_remove (tunnels, tunnelid);
2140 g_mutex_unlock (&tunnels_lock);
2143 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
2148 static GstRTSPResult
2149 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
2151 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2154 str = gst_rtsp_strresult (result);
2155 GST_INFO ("client %p: received an error %s", client, str);
2161 static GstRTSPResult
2162 error_full (GstRTSPWatch * watch, GstRTSPResult result,
2163 GstRTSPMessage * message, guint id, gpointer user_data)
2165 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2168 str = gst_rtsp_strresult (result);
2170 ("client %p: received an error %s when handling message %p with id %d",
2171 client, str, message, id);
2178 remember_tunnel (GstRTSPClient * client)
2180 GstRTSPClientPrivate *priv = client->priv;
2181 const gchar *tunnelid;
2183 /* store client in the pending tunnels */
2184 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
2185 if (tunnelid == NULL)
2188 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
2190 /* we can't have two clients connecting with the same tunnelid */
2191 g_mutex_lock (&tunnels_lock);
2192 if (g_hash_table_lookup (tunnels, tunnelid))
2193 goto tunnel_existed;
2195 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
2196 g_mutex_unlock (&tunnels_lock);
2203 GST_ERROR ("client %p: no tunnelid provided", client);
2208 g_mutex_unlock (&tunnels_lock);
2209 GST_ERROR ("client %p: tunnel session %s already existed", client,
2215 static GstRTSPStatusCode
2216 tunnel_start (GstRTSPWatch * watch, gpointer user_data)
2218 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2219 GstRTSPClientPrivate *priv = client->priv;
2221 GST_INFO ("client %p: tunnel start (connection %p)", client,
2224 if (!remember_tunnel (client))
2227 return GST_RTSP_STS_OK;
2232 GST_ERROR ("client %p: error starting tunnel", client);
2233 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
2237 static GstRTSPResult
2238 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
2240 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2241 GstRTSPClientPrivate *priv = client->priv;
2243 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
2246 /* ignore error, it'll only be a problem when the client does a POST again */
2247 remember_tunnel (client);
2252 static GstRTSPResult
2253 tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
2255 const gchar *tunnelid;
2256 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2257 GstRTSPClientPrivate *priv = client->priv;
2258 GstRTSPClient *oclient;
2259 GstRTSPClientPrivate *opriv;
2261 GST_INFO ("client %p: tunnel complete", client);
2263 /* find previous tunnel */
2264 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
2265 if (tunnelid == NULL)
2268 g_mutex_lock (&tunnels_lock);
2269 if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
2272 /* remove the old client from the table. ref before because removing it will
2273 * remove the ref to it. */
2274 g_object_ref (oclient);
2275 g_hash_table_remove (tunnels, tunnelid);
2277 opriv = oclient->priv;
2279 if (opriv->watch == NULL)
2281 g_mutex_unlock (&tunnels_lock);
2283 GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
2284 opriv->connection, priv->connection);
2286 /* merge the tunnels into the first client */
2287 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
2288 gst_rtsp_watch_reset (opriv->watch);
2289 g_object_unref (oclient);
2296 GST_ERROR ("client %p: no tunnelid provided", client);
2297 return GST_RTSP_ERROR;
2301 g_mutex_unlock (&tunnels_lock);
2302 GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid);
2303 return GST_RTSP_ERROR;
2307 g_mutex_unlock (&tunnels_lock);
2308 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
2309 g_object_unref (oclient);
2310 return GST_RTSP_ERROR;
2314 static GstRTSPWatchFuncs watch_funcs = {
2326 client_watch_notify (GstRTSPClient * client)
2328 GstRTSPClientPrivate *priv = client->priv;
2330 GST_INFO ("client %p: watch destroyed", client);
2332 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
2333 g_object_unref (client);
2337 setup_client (GstRTSPClient * client, GSocket * socket,
2338 GstRTSPConnection * conn, GError ** error)
2340 GstRTSPClientPrivate *priv = client->priv;
2341 GSocket *read_socket;
2342 GSocketAddress *address;
2345 read_socket = gst_rtsp_connection_get_read_socket (conn);
2346 priv->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
2348 if (!(address = g_socket_get_remote_address (read_socket, error)))
2351 g_free (priv->server_ip);
2352 /* keep the original ip that the client connected to */
2353 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2354 GInetAddress *iaddr;
2356 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2358 priv->server_ip = g_inet_address_to_string (iaddr);
2359 g_object_unref (address);
2361 priv->server_ip = g_strdup ("unknown");
2364 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2365 priv->server_ip, priv->is_ipv6);
2367 url = gst_rtsp_connection_get_url (conn);
2368 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2370 priv->connection = conn;
2377 GST_ERROR ("could not get remote address %s", (*error)->message);
2383 * gst_rtsp_client_use_socket:
2384 * @client: a #GstRTSPClient
2385 * @socket: a #GSocket
2386 * @ip: the IP address of the remote client
2387 * @port: the port used by the other end
2388 * @initial_buffer: any zero terminated initial data that was already read from
2392 * Take an existing network socket and use it for an RTSP connection.
2394 * Returns: %TRUE on success.
2397 gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket,
2398 const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
2400 GstRTSPConnection *conn;
2403 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2404 g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
2406 GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
2407 initial_buffer, &conn), no_connection);
2409 return setup_client (client, socket, conn, error);
2414 gchar *str = gst_rtsp_strresult (res);
2416 GST_ERROR ("could not create connection from socket %p: %s", socket, str);
2423 * gst_rtsp_client_accept:
2424 * @client: a #GstRTSPClient
2425 * @socket: a #GSocket
2426 * @context: the context to run in
2427 * @cancellable: a #GCancellable
2430 * Accept a new connection for @client on @socket.
2432 * Returns: %TRUE if the client could be accepted.
2435 gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
2436 GCancellable * cancellable, GError ** error)
2438 GstRTSPConnection *conn;
2441 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2442 g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
2444 /* a new client connected. */
2445 GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
2448 return setup_client (client, socket, conn, error);
2453 gchar *str = gst_rtsp_strresult (res);
2455 GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
2462 * gst_rtsp_client_attach:
2463 * @client: a #GstRTSPClient
2464 * @context: (allow-none): a #GMainContext
2466 * Attaches @client to @context. When the mainloop for @context is run, the
2467 * client will be dispatched. When @context is NULL, the default context will be
2470 * This function should be called when the client properties and urls are fully
2471 * configured and the client is ready to start.
2473 * Returns: the ID (greater than 0) for the source within the GMainContext.
2476 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
2478 GstRTSPClientPrivate *priv;
2481 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
2482 priv = client->priv;
2483 g_return_val_if_fail (priv->watch == NULL, 0);
2485 /* create watch for the connection and attach */
2486 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
2487 g_object_ref (client), (GDestroyNotify) client_watch_notify);
2488 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
2489 (GDestroyNotify) gst_rtsp_watch_unref);
2491 /* FIXME make this configurable. We don't want to do this yet because it will
2492 * be superceeded by a cache object later */
2493 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
2495 GST_INFO ("attaching to context %p", context);
2496 res = gst_rtsp_watch_attach (priv->watch, context);