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 static GMutex tunnels_lock;
28 static GHashTable *tunnels;
30 #define DEFAULT_SESSION_POOL NULL
31 #define DEFAULT_MEDIA_MAPPING NULL
32 #define DEFAULT_USE_CLIENT_SETTINGS FALSE
39 PROP_USE_CLIENT_SETTINGS,
47 SIGNAL_OPTIONS_REQUEST,
48 SIGNAL_DESCRIBE_REQUEST,
52 SIGNAL_TEARDOWN_REQUEST,
53 SIGNAL_SET_PARAMETER_REQUEST,
54 SIGNAL_GET_PARAMETER_REQUEST,
58 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
59 #define GST_CAT_DEFAULT rtsp_client_debug
61 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
63 static void gst_rtsp_client_get_property (GObject * object, guint propid,
64 GValue * value, GParamSpec * pspec);
65 static void gst_rtsp_client_set_property (GObject * object, guint propid,
66 const GValue * value, GParamSpec * pspec);
67 static void gst_rtsp_client_finalize (GObject * obj);
69 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
70 static void client_session_finalized (GstRTSPClient * client,
71 GstRTSPSession * session);
72 static void unlink_session_transports (GstRTSPClient * client,
73 GstRTSPSession * session, GstRTSPSessionMedia * media);
75 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
78 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
80 GObjectClass *gobject_class;
82 gobject_class = G_OBJECT_CLASS (klass);
84 gobject_class->get_property = gst_rtsp_client_get_property;
85 gobject_class->set_property = gst_rtsp_client_set_property;
86 gobject_class->finalize = gst_rtsp_client_finalize;
88 klass->create_sdp = create_sdp;
90 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
91 g_param_spec_object ("session-pool", "Session Pool",
92 "The session pool to use for client session",
93 GST_TYPE_RTSP_SESSION_POOL,
94 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
96 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
97 g_param_spec_object ("media-mapping", "Media Mapping",
98 "The media mapping to use for client session",
99 GST_TYPE_RTSP_MEDIA_MAPPING,
100 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
102 g_object_class_install_property (gobject_class, PROP_USE_CLIENT_SETTINGS,
103 g_param_spec_boolean ("use-client-settings", "Use Client Settings",
104 "Use client settings for ttl and destination in multicast",
105 DEFAULT_USE_CLIENT_SETTINGS,
106 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
108 gst_rtsp_client_signals[SIGNAL_CLOSED] =
109 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
110 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
111 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
113 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
114 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
115 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
116 g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
118 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
119 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
120 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
121 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
124 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
125 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
126 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
127 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
130 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
131 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
132 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
133 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
136 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
137 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
138 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
139 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
142 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
143 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
144 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
145 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
148 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
149 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
150 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
151 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
154 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
155 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
156 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
157 set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
158 G_TYPE_NONE, 1, G_TYPE_POINTER);
160 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
161 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
162 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
163 get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
164 G_TYPE_NONE, 1, G_TYPE_POINTER);
167 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
168 g_mutex_init (&tunnels_lock);
170 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
174 gst_rtsp_client_init (GstRTSPClient * client)
176 client->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
180 client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
182 /* unlink all media managed in this session */
183 while (session->medias) {
184 GstRTSPSessionMedia *media = session->medias->data;
186 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
187 unlink_session_transports (client, session, media);
188 /* unmanage the media in the session. this will modify session->medias */
189 gst_rtsp_session_release_media (session, media);
194 client_cleanup_sessions (GstRTSPClient * client)
198 /* remove weak-ref from sessions */
199 for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
200 GstRTSPSession *session = (GstRTSPSession *) sessions->data;
201 g_object_weak_unref (G_OBJECT (session),
202 (GWeakNotify) client_session_finalized, client);
203 client_unlink_session (client, session);
205 g_list_free (client->sessions);
206 client->sessions = NULL;
209 /* A client is finalized when the connection is broken */
211 gst_rtsp_client_finalize (GObject * obj)
213 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
215 GST_INFO ("finalize client %p", client);
218 g_source_destroy ((GSource *) client->watch);
220 client_cleanup_sessions (client);
222 gst_rtsp_connection_free (client->connection);
223 if (client->session_pool)
224 g_object_unref (client->session_pool);
225 if (client->media_mapping)
226 g_object_unref (client->media_mapping);
228 g_object_unref (client->auth);
231 gst_rtsp_url_free (client->uri);
233 g_object_unref (client->media);
235 g_free (client->server_ip);
237 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
241 gst_rtsp_client_get_property (GObject * object, guint propid,
242 GValue * value, GParamSpec * pspec)
244 GstRTSPClient *client = GST_RTSP_CLIENT (object);
247 case PROP_SESSION_POOL:
248 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
250 case PROP_MEDIA_MAPPING:
251 g_value_take_object (value, gst_rtsp_client_get_media_mapping (client));
253 case PROP_USE_CLIENT_SETTINGS:
254 g_value_set_boolean (value,
255 gst_rtsp_client_get_use_client_settings (client));
258 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
263 gst_rtsp_client_set_property (GObject * object, guint propid,
264 const GValue * value, GParamSpec * pspec)
266 GstRTSPClient *client = GST_RTSP_CLIENT (object);
269 case PROP_SESSION_POOL:
270 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
272 case PROP_MEDIA_MAPPING:
273 gst_rtsp_client_set_media_mapping (client, g_value_get_object (value));
275 case PROP_USE_CLIENT_SETTINGS:
276 gst_rtsp_client_set_use_client_settings (client,
277 g_value_get_boolean (value));
280 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
285 * gst_rtsp_client_new:
287 * Create a new #GstRTSPClient instance.
289 * Returns: a new #GstRTSPClient
292 gst_rtsp_client_new (void)
294 GstRTSPClient *result;
296 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
302 send_response (GstRTSPClient * client, GstRTSPSession * session,
303 GstRTSPMessage * response)
305 gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
306 "GStreamer RTSP server");
308 /* remove any previous header */
309 gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
311 /* add the new session header for new session ids */
313 gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION,
314 gst_rtsp_session_get_header (session));
317 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
318 gst_rtsp_message_dump (response);
321 gst_rtsp_watch_send_message (client->watch, response, NULL);
322 gst_rtsp_message_unset (response);
326 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
327 GstRTSPClientState * state)
329 gst_rtsp_message_init_response (state->response, code,
330 gst_rtsp_status_as_text (code), state->request);
332 send_response (client, NULL, state->response);
336 handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
337 GstRTSPClientState * state)
339 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
340 gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
343 /* and let the authentication manager setup the auth tokens */
344 gst_rtsp_auth_setup_auth (auth, client, 0, state);
347 send_response (client, state->session, state->response);
352 compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
354 if (uri1 == NULL || uri2 == NULL)
357 if (strcmp (uri1->abspath, uri2->abspath))
363 /* this function is called to initially find the media for the DESCRIBE request
364 * but is cached for when the same client (without breaking the connection) is
365 * doing a setup for the exact same url. */
366 static GstRTSPMedia *
367 find_media (GstRTSPClient * client, GstRTSPClientState * state)
369 GstRTSPMediaFactory *factory;
373 if (!compare_uri (client->uri, state->uri)) {
374 /* remove any previously cached values before we try to construct a new
377 gst_rtsp_url_free (client->uri);
380 g_object_unref (client->media);
381 client->media = NULL;
383 if (!client->media_mapping)
386 /* find the factory for the uri first */
388 gst_rtsp_media_mapping_find_factory (client->media_mapping,
392 state->factory = factory;
394 /* check if we have access to the factory */
395 if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
396 if (!gst_rtsp_auth_check (auth, client, 0, state))
399 g_object_unref (auth);
402 /* prepare the media and add it to the pipeline */
403 if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
406 g_object_unref (factory);
408 state->factory = NULL;
410 /* set ipv6 on the media before preparing */
411 media->is_ipv6 = client->is_ipv6;
412 state->media = media;
414 /* prepare the media */
415 if (!(gst_rtsp_media_prepare (media)))
418 /* now keep track of the uri and the media */
419 client->uri = gst_rtsp_url_copy (state->uri);
420 client->media = media;
422 /* we have seen this uri before, used cached media */
423 media = client->media;
424 state->media = media;
425 GST_INFO ("reusing cached media %p", media);
429 g_object_ref (media);
436 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
441 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
446 handle_unauthorized_request (client, auth, state);
447 g_object_unref (factory);
448 g_object_unref (auth);
453 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
454 g_object_unref (factory);
459 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
460 g_object_unref (media);
466 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
468 GstRTSPMessage message = { 0 };
473 gst_rtsp_message_init_data (&message, channel);
475 /* FIXME, need some sort of iovec RTSPMessage here */
476 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
479 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
481 /* FIXME, client->watch could have been finalized here, we need to keep an
482 * extra refcount to the watch. */
483 gst_rtsp_watch_send_message (client->watch, &message, NULL);
485 gst_rtsp_message_steal_body (&message, &data, &usize);
486 gst_buffer_unmap (buffer, &map_info);
488 gst_rtsp_message_unset (&message);
494 link_transport (GstRTSPClient * client, GstRTSPSession * session,
495 GstRTSPStreamTransport * trans)
497 GST_DEBUG ("client %p: linking transport %p", client, trans);
498 gst_rtsp_stream_transport_set_callbacks (trans,
499 (GstRTSPSendFunc) do_send_data,
500 (GstRTSPSendFunc) do_send_data, client, NULL);
502 client->transports = g_list_prepend (client->transports, trans);
504 /* make sure our session can't expire */
505 gst_rtsp_session_prevent_expire (session);
509 unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
510 GstRTSPStreamTransport * trans)
512 GST_DEBUG ("client %p: unlinking transport %p", client, trans);
513 gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
515 client->transports = g_list_remove (client->transports, trans);
517 /* our session can now expire */
518 gst_rtsp_session_allow_expire (session);
522 unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
523 GstRTSPSessionMedia * media)
527 n_streams = gst_rtsp_media_n_streams (media->media);
528 for (i = 0; i < n_streams; i++) {
529 GstRTSPStreamTransport *trans;
530 GstRTSPTransport *tr;
532 /* get the transport, if there is no transport configured, skip this stream */
533 trans = gst_rtsp_session_media_get_transport (media, i);
537 tr = trans->transport;
539 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
540 /* for TCP, unlink the stream from the TCP connection of the client */
541 unlink_transport (client, session, trans);
547 close_connection (GstRTSPClient * client)
549 const gchar *tunnelid;
551 GST_DEBUG ("client %p: closing connection", client);
553 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
554 g_mutex_lock (&tunnels_lock);
555 /* remove from tunnelids */
556 g_hash_table_remove (tunnels, tunnelid);
557 g_mutex_unlock (&tunnels_lock);
560 gst_rtsp_connection_close (client->connection);
564 handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
566 GstRTSPSession *session;
567 GstRTSPSessionMedia *media;
568 GstRTSPStatusCode code;
573 session = state->session;
575 /* get a handle to the configuration of the media in the session */
576 media = gst_rtsp_session_get_media (session, state->uri);
580 state->sessmedia = media;
582 /* unlink the all TCP callbacks */
583 unlink_session_transports (client, session, media);
585 /* remove the session from the watched sessions */
586 g_object_weak_unref (G_OBJECT (session),
587 (GWeakNotify) client_session_finalized, client);
588 client->sessions = g_list_remove (client->sessions, session);
590 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
592 /* unmanage the media in the session, returns false if all media session
594 if (!gst_rtsp_session_release_media (session, media)) {
595 /* remove the session */
596 gst_rtsp_session_pool_remove (client->session_pool, session);
598 /* construct the response now */
599 code = GST_RTSP_STS_OK;
600 gst_rtsp_message_init_response (state->response, code,
601 gst_rtsp_status_as_text (code), state->request);
603 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
606 send_response (client, session, state->response);
608 /* we emit the signal before closing the connection */
609 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
612 close_connection (client);
619 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
624 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
630 handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
636 res = gst_rtsp_message_get_body (state->request, &data, &size);
637 if (res != GST_RTSP_OK)
641 /* no body, keep-alive request */
642 send_generic_response (client, GST_RTSP_STS_OK, state);
644 /* there is a body, handle the params */
645 res = gst_rtsp_params_get (client, state);
646 if (res != GST_RTSP_OK)
649 send_response (client, state->session, state->response);
652 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
660 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
666 handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
672 res = gst_rtsp_message_get_body (state->request, &data, &size);
673 if (res != GST_RTSP_OK)
677 /* no body, keep-alive request */
678 send_generic_response (client, GST_RTSP_STS_OK, state);
680 /* there is a body, handle the params */
681 res = gst_rtsp_params_set (client, state);
682 if (res != GST_RTSP_OK)
685 send_response (client, state->session, state->response);
688 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
696 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
702 handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
704 GstRTSPSession *session;
705 GstRTSPSessionMedia *media;
706 GstRTSPStatusCode code;
708 if (!(session = state->session))
711 /* get a handle to the configuration of the media in the session */
712 media = gst_rtsp_session_get_media (session, state->uri);
716 state->sessmedia = media;
718 /* the session state must be playing or recording */
719 if (media->state != GST_RTSP_STATE_PLAYING &&
720 media->state != GST_RTSP_STATE_RECORDING)
723 /* unlink the all TCP callbacks */
724 unlink_session_transports (client, session, media);
726 /* then pause sending */
727 gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
729 /* construct the response now */
730 code = GST_RTSP_STS_OK;
731 gst_rtsp_message_init_response (state->response, code,
732 gst_rtsp_status_as_text (code), state->request);
734 send_response (client, session, state->response);
736 /* the state is now READY */
737 media->state = GST_RTSP_STATE_READY;
739 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
747 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
752 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
757 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
764 handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
766 GstRTSPSession *session;
767 GstRTSPSessionMedia *media;
768 GstRTSPStatusCode code;
770 guint n_streams, i, infocount;
772 GstRTSPTimeRange *range;
775 if (!(session = state->session))
778 /* get a handle to the configuration of the media in the session */
779 media = gst_rtsp_session_get_media (session, state->uri);
783 state->sessmedia = media;
785 /* the session state must be playing or ready */
786 if (media->state != GST_RTSP_STATE_PLAYING &&
787 media->state != GST_RTSP_STATE_READY)
790 /* parse the range header if we have one */
792 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
793 if (res == GST_RTSP_OK) {
794 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
795 /* we have a range, seek to the position */
796 gst_rtsp_media_seek (media->media, range);
797 gst_rtsp_range_free (range);
801 /* grab RTPInfo from the payloaders now */
802 rtpinfo = g_string_new ("");
804 n_streams = gst_rtsp_media_n_streams (media->media);
805 for (i = 0, infocount = 0; i < n_streams; i++) {
806 GstRTSPStreamTransport *trans;
807 GstRTSPTransport *tr;
811 /* get the transport, if there is no transport configured, skip this stream */
812 trans = gst_rtsp_session_media_get_transport (media, i);
814 GST_INFO ("stream %d is not configured", i);
817 tr = trans->transport;
819 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
820 /* for TCP, link the stream to the TCP connection of the client */
821 link_transport (client, session, trans);
824 if (gst_rtsp_stream_get_rtpinfo (trans->stream, &rtptime, &seq)) {
826 g_string_append (rtpinfo, ", ");
828 uristr = gst_rtsp_url_get_request_uri (state->uri);
829 g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
830 uristr, i, seq, rtptime);
835 GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
839 /* construct the response now */
840 code = GST_RTSP_STS_OK;
841 gst_rtsp_message_init_response (state->response, code,
842 gst_rtsp_status_as_text (code), state->request);
844 /* add the RTP-Info header */
846 str = g_string_free (rtpinfo, FALSE);
847 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
849 g_string_free (rtpinfo, TRUE);
853 str = gst_rtsp_media_get_range_string (media->media, TRUE);
854 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
856 send_response (client, session, state->response);
858 /* start playing after sending the request */
859 gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
861 media->state = GST_RTSP_STATE_PLAYING;
863 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
871 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
876 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
881 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
888 do_keepalive (GstRTSPSession * session)
890 GST_INFO ("keep session %p alive", session);
891 gst_rtsp_session_touch (session);
894 /* parse @transport and return a valid transport in @tr. only transports
895 * from @supported are returned. Returns FALSE if no valid transport
898 parse_transport (const char *transport, GstRTSPLowerTrans supported,
899 GstRTSPTransport * tr)
906 gst_rtsp_transport_init (tr);
908 GST_WARNING ("parsing transports %s", transport);
910 transports = g_strsplit (transport, ",", 0);
912 /* loop through the transports, try to parse */
913 for (i = 0; transports[i]; i++) {
914 res = gst_rtsp_transport_parse (transports[i], tr);
915 if (res != GST_RTSP_OK) {
916 /* no valid transport, search some more */
917 GST_WARNING ("could not parse transport %s", transports[i]);
921 /* we have a transport, see if it's RTP/AVP */
922 if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
923 GST_WARNING ("invalid transport %s", transports[i]);
927 if (!(tr->lower_transport & supported)) {
928 GST_WARNING ("unsupported transport %s", transports[i]);
932 /* we have a valid transport */
933 GST_INFO ("found valid transport %s", transports[i]);
938 gst_rtsp_transport_init (tr);
940 g_strfreev (transports);
946 handle_blocksize (GstRTSPMedia * media, GstRTSPMessage * request)
948 gchar *blocksize_str;
951 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
952 &blocksize_str, 0) == GST_RTSP_OK) {
956 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
957 if (end == blocksize_str) {
958 GST_ERROR ("failed to parse blocksize");
961 /* we don't want to change the mtu when this media
962 * can be shared because it impacts other clients */
963 if (gst_rtsp_media_is_shared (media))
966 if (blocksize > G_MAXUINT)
967 blocksize = G_MAXUINT;
968 gst_rtsp_media_set_mtu (media, blocksize);
975 configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state,
976 GstRTSPTransport * ct)
978 /* we have a valid transport now, set the destination of the client. */
979 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
980 if (ct->destination == NULL || !client->use_client_settings) {
981 GstRTSPAddressPool *pool;
987 pool = gst_rtsp_media_get_address_pool (state->media);
991 id = gst_rtsp_address_pool_acquire_address (pool,
992 GST_RTSP_ADDRESS_FLAG_EVEN_PORT, 2, &address, &port, &ttl);
996 g_free (ct->destination);
997 ct->destination = address;
999 ct->port.max = port + 1;
1005 url = gst_rtsp_connection_get_url (client->connection);
1006 g_free (ct->destination);
1007 ct->destination = g_strdup (url->host);
1009 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1010 /* check if the client selected channels for TCP */
1011 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1012 gst_rtsp_session_media_alloc_channels (state->sessmedia,
1030 static GstRTSPTransport *
1031 make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
1032 GstRTSPTransport * ct)
1034 GstRTSPTransport *st;
1036 /* prepare the server transport */
1037 gst_rtsp_transport_new (&st);
1039 st->trans = ct->trans;
1040 st->profile = ct->profile;
1041 st->lower_transport = ct->lower_transport;
1043 switch (st->lower_transport) {
1044 case GST_RTSP_LOWER_TRANS_UDP:
1045 st->client_port = ct->client_port;
1046 st->server_port = state->stream->server_port;
1048 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1049 st->port = ct->port;
1050 st->destination = g_strdup (ct->destination);
1053 case GST_RTSP_LOWER_TRANS_TCP:
1054 st->interleaved = ct->interleaved;
1059 if (state->stream->session)
1060 g_object_get (state->stream->session, "internal-ssrc", &st->ssrc, NULL);
1066 handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
1071 GstRTSPTransport *ct, *st;
1072 GstRTSPLowerTrans supported;
1073 GstRTSPStatusCode code;
1074 GstRTSPSession *session;
1075 GstRTSPStreamTransport *trans;
1076 gchar *trans_str, *pos;
1078 GstRTSPSessionMedia *sessmedia;
1079 GstRTSPMedia *media;
1080 GstRTSPStream *stream;
1084 /* the uri contains the stream number we added in the SDP config, which is
1085 * always /stream=%d so we need to strip that off
1086 * parse the stream we need to configure, look for the stream in the abspath
1087 * first and then in the query. */
1088 if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
1089 if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
1093 /* we can mofify the parsed uri in place */
1096 pos += strlen ("/stream=");
1097 if (sscanf (pos, "%u", &streamid) != 1)
1100 /* parse the transport */
1102 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
1104 if (res != GST_RTSP_OK)
1107 gst_rtsp_transport_new (&ct);
1109 /* our supported transports */
1110 supported = GST_RTSP_LOWER_TRANS_UDP |
1111 GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
1113 /* parse and find a usable supported transport */
1114 if (!parse_transport (transport, supported, ct))
1115 goto unsupported_transports;
1117 /* we create the session after parsing stuff so that we don't make
1118 * a session for malformed requests */
1119 if (client->session_pool == NULL)
1122 session = state->session;
1125 g_object_ref (session);
1126 /* get a handle to the configuration of the media in the session, this can
1127 * return NULL if this is a new url to manage in this session. */
1128 sessmedia = gst_rtsp_session_get_media (session, uri);
1130 /* create a session if this fails we probably reached our session limit or
1132 if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
1133 goto service_unavailable;
1135 state->session = session;
1137 /* we need a new media configuration in this session */
1141 /* we have no media, find one and manage it */
1142 if (sessmedia == NULL) {
1143 /* get a handle to the configuration of the media in the session */
1144 if ((media = find_media (client, state))) {
1145 /* manage the media in our session now */
1146 sessmedia = gst_rtsp_session_manage_media (session, uri, media);
1150 /* if we stil have no media, error */
1151 if (sessmedia == NULL)
1154 state->sessmedia = sessmedia;
1155 state->media = media = sessmedia->media;
1157 /* now get the stream */
1158 stream = gst_rtsp_media_get_stream (media, streamid);
1162 state->stream = stream;
1164 /* FIXME set only on this stream */
1165 if (!handle_blocksize (media, state->request))
1166 goto invalid_blocksize;
1168 /* update the client transport */
1169 if (!configure_client_transport (client, state, ct))
1170 goto unsupported_client_transport;
1172 /* set in the session media transport */
1173 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1175 /* configure keepalive for this transport */
1176 gst_rtsp_stream_transport_set_keepalive (trans,
1177 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1179 /* create and serialize the server transport */
1180 st = make_server_transport (client, state, ct);
1181 trans_str = gst_rtsp_transport_as_text (st);
1182 gst_rtsp_transport_free (st);
1184 /* construct the response now */
1185 code = GST_RTSP_STS_OK;
1186 gst_rtsp_message_init_response (state->response, code,
1187 gst_rtsp_status_as_text (code), state->request);
1189 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
1193 send_response (client, session, state->response);
1195 /* update the state */
1196 switch (sessmedia->state) {
1197 case GST_RTSP_STATE_PLAYING:
1198 case GST_RTSP_STATE_RECORDING:
1199 case GST_RTSP_STATE_READY:
1200 /* no state change */
1203 sessmedia->state = GST_RTSP_STATE_READY;
1206 g_object_unref (session);
1208 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
1216 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1221 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1222 g_object_unref (session);
1223 gst_rtsp_transport_free (ct);
1228 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1229 g_object_unref (session);
1230 gst_rtsp_transport_free (ct);
1233 unsupported_client_transport:
1235 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1236 g_object_unref (session);
1237 gst_rtsp_transport_free (ct);
1242 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1245 unsupported_transports:
1247 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1248 gst_rtsp_transport_free (ct);
1253 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1254 gst_rtsp_transport_free (ct);
1257 service_unavailable:
1259 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1260 gst_rtsp_transport_free (ct);
1265 static GstSDPMessage *
1266 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1271 GstRTSPLowerTrans protocols;
1273 gst_sdp_message_new (&sdp);
1275 /* some standard things first */
1276 gst_sdp_message_set_version (sdp, "0");
1278 if (client->is_ipv6)
1283 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
1286 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
1287 gst_sdp_message_set_information (sdp, "rtsp-server");
1288 gst_sdp_message_add_time (sdp, "0", "0", NULL);
1289 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
1290 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
1291 gst_sdp_message_add_attribute (sdp, "control", "*");
1293 info.server_proto = proto;
1294 protocols = gst_rtsp_media_get_protocols (media);
1295 if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
1297 info.server_ip = gst_rtsp_media_get_multicast_group (media);
1299 info.server_ip = g_strdup (client->server_ip);
1302 info.server_ip = g_strdup (client->server_ip);
1304 /* create an SDP for the media object */
1305 if (!gst_rtsp_sdp_from_media (sdp, &info, media))
1308 g_free (info.server_ip);
1315 g_free (info.server_ip);
1316 gst_sdp_message_free (sdp);
1321 /* for the describe we must generate an SDP */
1323 handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
1328 gchar *str, *content_base;
1329 GstRTSPMedia *media;
1330 GstRTSPClientClass *klass;
1332 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1334 /* check what kind of format is accepted, we don't really do anything with it
1335 * and always return SDP for now. */
1340 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
1342 if (res == GST_RTSP_ENOTIMPL)
1345 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
1349 /* find the media object for the uri */
1350 if (!(media = find_media (client, state)))
1353 /* create an SDP for the media object on this client */
1354 if (!(sdp = klass->create_sdp (client, media)))
1357 g_object_unref (media);
1359 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1360 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1362 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
1365 /* content base for some clients that might screw up creating the setup uri */
1366 str = gst_rtsp_url_get_request_uri (state->uri);
1367 str_len = strlen (str);
1369 /* check for trailing '/' and append one */
1370 if (str[str_len - 1] != '/') {
1371 content_base = g_malloc (str_len + 2);
1372 memcpy (content_base, str, str_len);
1373 content_base[str_len] = '/';
1374 content_base[str_len + 1] = '\0';
1380 GST_INFO ("adding content-base: %s", content_base);
1382 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
1384 g_free (content_base);
1386 /* add SDP to the response body */
1387 str = gst_sdp_message_as_text (sdp);
1388 gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
1389 gst_sdp_message_free (sdp);
1391 send_response (client, state->session, state->response);
1393 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
1401 /* error reply is already sent */
1406 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1407 g_object_unref (media);
1413 handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
1415 GstRTSPMethod options;
1418 options = GST_RTSP_DESCRIBE |
1423 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
1425 str = gst_rtsp_options_as_text (options);
1427 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1428 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1430 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
1433 send_response (client, state->session, state->response);
1435 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
1441 /* remove duplicate and trailing '/' */
1443 sanitize_uri (GstRTSPUrl * uri)
1447 gboolean have_slash, prev_slash;
1449 s = d = uri->abspath;
1450 len = strlen (uri->abspath);
1454 for (i = 0; i < len; i++) {
1455 have_slash = s[i] == '/';
1457 if (!have_slash || !prev_slash)
1459 prev_slash = have_slash;
1461 len = d - uri->abspath;
1462 /* don't remove the first slash if that's the only thing left */
1463 if (len > 1 && *(d - 1) == '/')
1469 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
1471 GST_INFO ("client %p: session %p finished", client, session);
1473 /* unlink all media managed in this session */
1474 client_unlink_session (client, session);
1476 /* remove the session */
1477 if (!(client->sessions = g_list_remove (client->sessions, session))) {
1478 GST_INFO ("client %p: all sessions finalized, close the connection",
1480 close_connection (client);
1485 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
1489 for (walk = client->sessions; walk; walk = g_list_next (walk)) {
1490 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
1492 /* we already know about this session */
1493 if (msession == session)
1497 GST_INFO ("watching session %p", session);
1499 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
1501 client->sessions = g_list_prepend (client->sessions, session);
1503 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1508 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
1510 GstRTSPMethod method;
1511 const gchar *uristr;
1513 GstRTSPVersion version;
1515 GstRTSPSession *session;
1516 GstRTSPClientState state = { NULL };
1517 GstRTSPMessage response = { 0 };
1520 state.request = request;
1521 state.response = &response;
1523 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
1524 gst_rtsp_message_dump (request);
1527 GST_INFO ("client %p: received a request", client);
1529 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
1531 if (version != GST_RTSP_VERSION_1_0) {
1532 /* we can only handle 1.0 requests */
1533 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
1537 state.method = method;
1539 /* we always try to parse the url first */
1540 if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
1541 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1545 /* sanitize the uri */
1549 /* get the session if there is any */
1550 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
1551 if (res == GST_RTSP_OK) {
1552 if (client->session_pool == NULL)
1555 /* we had a session in the request, find it again */
1556 if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
1557 goto session_not_found;
1559 /* we add the session to the client list of watched sessions. When a session
1560 * disappears because it times out, we will be notified. If all sessions are
1561 * gone, we will close the connection */
1562 client_watch_session (client, session);
1566 state.session = session;
1569 if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
1570 goto not_authorized;
1573 /* now see what is asked and dispatch to a dedicated handler */
1575 case GST_RTSP_OPTIONS:
1576 handle_options_request (client, &state);
1578 case GST_RTSP_DESCRIBE:
1579 handle_describe_request (client, &state);
1581 case GST_RTSP_SETUP:
1582 handle_setup_request (client, &state);
1585 handle_play_request (client, &state);
1587 case GST_RTSP_PAUSE:
1588 handle_pause_request (client, &state);
1590 case GST_RTSP_TEARDOWN:
1591 handle_teardown_request (client, &state);
1593 case GST_RTSP_SET_PARAMETER:
1594 handle_set_param_request (client, &state);
1596 case GST_RTSP_GET_PARAMETER:
1597 handle_get_param_request (client, &state);
1599 case GST_RTSP_ANNOUNCE:
1600 case GST_RTSP_RECORD:
1601 case GST_RTSP_REDIRECT:
1602 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
1604 case GST_RTSP_INVALID:
1606 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1610 g_object_unref (session);
1612 gst_rtsp_url_free (uri);
1618 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
1623 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
1628 handle_unauthorized_request (client, client->auth, &state);
1634 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
1644 /* find the stream for this message */
1645 res = gst_rtsp_message_parse_data (message, &channel);
1646 if (res != GST_RTSP_OK)
1649 gst_rtsp_message_steal_body (message, &data, &size);
1651 buffer = gst_buffer_new_wrapped (data, size);
1654 for (walk = client->transports; walk; walk = g_list_next (walk)) {
1655 GstRTSPStreamTransport *trans;
1656 GstRTSPStream *stream;
1657 GstRTSPTransport *tr;
1661 /* we only add clients with a transport to the list */
1662 tr = trans->transport;
1663 stream = trans->stream;
1665 /* check for TCP transport */
1666 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1667 /* dispatch to the stream based on the channel number */
1668 if (tr->interleaved.min == channel) {
1669 gst_rtsp_stream_recv_rtp (stream, buffer);
1672 } else if (tr->interleaved.max == channel) {
1673 gst_rtsp_stream_recv_rtcp (stream, buffer);
1680 gst_buffer_unref (buffer);
1684 * gst_rtsp_client_set_session_pool:
1685 * @client: a #GstRTSPClient
1686 * @pool: a #GstRTSPSessionPool
1688 * Set @pool as the sessionpool for @client which it will use to find
1689 * or allocate sessions. the sessionpool is usually inherited from the server
1690 * that created the client but can be overridden later.
1693 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
1694 GstRTSPSessionPool * pool)
1696 GstRTSPSessionPool *old;
1698 old = client->session_pool;
1701 g_object_ref (pool);
1702 client->session_pool = pool;
1704 g_object_unref (old);
1709 * gst_rtsp_client_get_session_pool:
1710 * @client: a #GstRTSPClient
1712 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
1714 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
1716 GstRTSPSessionPool *
1717 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
1719 GstRTSPSessionPool *result;
1721 if ((result = client->session_pool))
1722 g_object_ref (result);
1728 * gst_rtsp_client_set_server:
1729 * @client: a #GstRTSPClient
1730 * @server: a #GstRTSPServer
1732 * Set @server as the server that created @client.
1735 gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
1739 old = client->server;
1740 if (old != server) {
1742 g_object_ref (server);
1743 client->server = server;
1745 g_object_unref (old);
1750 * gst_rtsp_client_get_server:
1751 * @client: a #GstRTSPClient
1753 * Get the #GstRTSPServer object that @client was created from.
1755 * Returns: (transfer full): a #GstRTSPServer, unref after usage.
1758 gst_rtsp_client_get_server (GstRTSPClient * client)
1760 GstRTSPServer *result;
1762 if ((result = client->server))
1763 g_object_ref (result);
1769 * gst_rtsp_client_set_media_mapping:
1770 * @client: a #GstRTSPClient
1771 * @mapping: a #GstRTSPMediaMapping
1773 * Set @mapping as the media mapping for @client which it will use to map urls
1774 * to media streams. These mapping is usually inherited from the server that
1775 * created the client but can be overriden later.
1778 gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
1779 GstRTSPMediaMapping * mapping)
1781 GstRTSPMediaMapping *old;
1783 old = client->media_mapping;
1785 if (old != mapping) {
1787 g_object_ref (mapping);
1788 client->media_mapping = mapping;
1790 g_object_unref (old);
1795 * gst_rtsp_client_get_media_mapping:
1796 * @client: a #GstRTSPClient
1798 * Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
1800 * Returns: (transfer full): a #GstRTSPMediaMapping, unref after usage.
1802 GstRTSPMediaMapping *
1803 gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
1805 GstRTSPMediaMapping *result;
1807 if ((result = client->media_mapping))
1808 g_object_ref (result);
1814 * gst_rtsp_client_set_use_client_settings:
1815 * @client: a #GstRTSPClient
1816 * @use_client_settings: whether to use client settings for multicast
1818 * Use client transport settings (destination and ttl) for multicast.
1819 * When @use_client_settings is %FALSE, the server settings will be
1823 gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
1824 gboolean use_client_settings)
1826 client->use_client_settings = use_client_settings;
1830 * gst_rtsp_client_get_use_client_settings:
1831 * @client: a #GstRTSPClient
1833 * Check if client transport settings (destination and ttl) for multicast
1837 gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
1839 return client->use_client_settings;
1843 * gst_rtsp_client_set_auth:
1844 * @client: a #GstRTSPClient
1845 * @auth: a #GstRTSPAuth
1847 * configure @auth to be used as the authentication manager of @client.
1850 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
1854 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1860 g_object_ref (auth);
1861 client->auth = auth;
1863 g_object_unref (old);
1869 * gst_rtsp_client_get_auth:
1870 * @client: a #GstRTSPClient
1872 * Get the #GstRTSPAuth used as the authentication manager of @client.
1874 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
1878 gst_rtsp_client_get_auth (GstRTSPClient * client)
1880 GstRTSPAuth *result;
1882 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
1884 if ((result = client->auth))
1885 g_object_ref (result);
1890 static GstRTSPResult
1891 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
1894 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1896 switch (message->type) {
1897 case GST_RTSP_MESSAGE_REQUEST:
1898 handle_request (client, message);
1900 case GST_RTSP_MESSAGE_RESPONSE:
1902 case GST_RTSP_MESSAGE_DATA:
1903 handle_data (client, message);
1911 static GstRTSPResult
1912 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
1914 /* GstRTSPClient *client; */
1916 /* client = GST_RTSP_CLIENT (user_data); */
1918 /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
1923 static GstRTSPResult
1924 closed (GstRTSPWatch * watch, gpointer user_data)
1926 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1927 const gchar *tunnelid;
1929 GST_INFO ("client %p: connection closed", client);
1931 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
1932 g_mutex_lock (&tunnels_lock);
1933 /* remove from tunnelids */
1934 g_hash_table_remove (tunnels, tunnelid);
1935 g_mutex_unlock (&tunnels_lock);
1941 static GstRTSPResult
1942 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
1944 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1947 str = gst_rtsp_strresult (result);
1948 GST_INFO ("client %p: received an error %s", client, str);
1954 static GstRTSPResult
1955 error_full (GstRTSPWatch * watch, GstRTSPResult result,
1956 GstRTSPMessage * message, guint id, gpointer user_data)
1958 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1961 str = gst_rtsp_strresult (result);
1963 ("client %p: received an error %s when handling message %p with id %d",
1964 client, str, message, id);
1971 remember_tunnel (GstRTSPClient * client)
1973 const gchar *tunnelid;
1975 /* store client in the pending tunnels */
1976 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1977 if (tunnelid == NULL)
1980 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
1982 /* we can't have two clients connecting with the same tunnelid */
1983 g_mutex_lock (&tunnels_lock);
1984 if (g_hash_table_lookup (tunnels, tunnelid))
1985 goto tunnel_existed;
1987 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
1988 g_mutex_unlock (&tunnels_lock);
1995 GST_ERROR ("client %p: no tunnelid provided", client);
2000 g_mutex_unlock (&tunnels_lock);
2001 GST_ERROR ("client %p: tunnel session %s already existed", client,
2007 static GstRTSPStatusCode
2008 tunnel_start (GstRTSPWatch * watch, gpointer user_data)
2010 GstRTSPClient *client;
2012 client = GST_RTSP_CLIENT (user_data);
2014 GST_INFO ("client %p: tunnel start (connection %p)", client,
2015 client->connection);
2017 if (!remember_tunnel (client))
2020 return GST_RTSP_STS_OK;
2025 GST_ERROR ("client %p: error starting tunnel", client);
2026 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
2030 static GstRTSPResult
2031 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
2033 GstRTSPClient *client;
2035 client = GST_RTSP_CLIENT (user_data);
2037 GST_INFO ("client %p: tunnel lost (connection %p)", client,
2038 client->connection);
2040 /* ignore error, it'll only be a problem when the client does a POST again */
2041 remember_tunnel (client);
2046 static GstRTSPResult
2047 tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
2049 const gchar *tunnelid;
2050 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2051 GstRTSPClient *oclient;
2053 GST_INFO ("client %p: tunnel complete", client);
2055 /* find previous tunnel */
2056 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
2057 if (tunnelid == NULL)
2060 g_mutex_lock (&tunnels_lock);
2061 if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
2064 /* remove the old client from the table. ref before because removing it will
2065 * remove the ref to it. */
2066 g_object_ref (oclient);
2067 g_hash_table_remove (tunnels, tunnelid);
2069 if (oclient->watch == NULL)
2071 g_mutex_unlock (&tunnels_lock);
2073 GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
2074 oclient->connection, client->connection);
2076 /* merge the tunnels into the first client */
2077 gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
2078 gst_rtsp_watch_reset (oclient->watch);
2079 g_object_unref (oclient);
2086 GST_INFO ("client %p: no tunnelid provided", client);
2087 return GST_RTSP_ERROR;
2091 g_mutex_unlock (&tunnels_lock);
2092 GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
2093 return GST_RTSP_ERROR;
2097 g_mutex_unlock (&tunnels_lock);
2098 GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
2099 g_object_unref (oclient);
2100 return GST_RTSP_ERROR;
2104 static GstRTSPWatchFuncs watch_funcs = {
2116 client_watch_notify (GstRTSPClient * client)
2118 GST_INFO ("client %p: watch destroyed", client);
2119 client->watch = NULL;
2120 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
2121 g_object_unref (client);
2125 setup_client (GstRTSPClient * client, GSocket * socket,
2126 GstRTSPConnection * conn, GError ** error)
2128 GSocket *read_socket;
2129 GSocketAddress *address;
2132 read_socket = gst_rtsp_connection_get_read_socket (conn);
2133 client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
2135 if (!(address = g_socket_get_remote_address (read_socket, error)))
2138 g_free (client->server_ip);
2139 /* keep the original ip that the client connected to */
2140 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2141 GInetAddress *iaddr;
2143 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2145 client->server_ip = g_inet_address_to_string (iaddr);
2146 g_object_unref (address);
2148 client->server_ip = g_strdup ("unknown");
2151 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2152 client->server_ip, client->is_ipv6);
2154 url = gst_rtsp_connection_get_url (conn);
2155 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2157 client->connection = conn;
2164 GST_ERROR ("could not get remote address %s", (*error)->message);
2170 * gst_rtsp_client_use_socket:
2171 * @client: a #GstRTSPClient
2172 * @socket: a #GSocket
2173 * @ip: the IP address of the remote client
2174 * @port: the port used by the other end
2175 * @initial_buffer: any zero terminated initial data that was already read from
2179 * Take an existing network socket and use it for an RTSP connection.
2181 * Returns: %TRUE on success.
2184 gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket,
2185 const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
2187 GstRTSPConnection *conn;
2190 GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
2191 initial_buffer, &conn), no_connection);
2193 return setup_client (client, socket, conn, error);
2198 gchar *str = gst_rtsp_strresult (res);
2200 GST_ERROR ("could not create connection from socket %p: %s", socket, str);
2207 * gst_rtsp_client_accept:
2208 * @client: a #GstRTSPClient
2209 * @socket: a #GSocket
2210 * @context: the context to run in
2211 * @cancellable: a #GCancellable
2214 * Accept a new connection for @client on @socket.
2216 * Returns: %TRUE if the client could be accepted.
2219 gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
2220 GCancellable * cancellable, GError ** error)
2222 GstRTSPConnection *conn;
2225 /* a new client connected. */
2226 GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
2229 return setup_client (client, socket, conn, error);
2234 gchar *str = gst_rtsp_strresult (res);
2236 GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
2243 * gst_rtsp_client_attach:
2244 * @client: a #GstRTSPClient
2245 * @context: (allow-none): a #GMainContext
2247 * Attaches @client to @context. When the mainloop for @context is run, the
2248 * client will be dispatched. When @context is NULL, the default context will be
2251 * This function should be called when the client properties and urls are fully
2252 * configured and the client is ready to start.
2254 * Returns: the ID (greater than 0) for the source within the GMainContext.
2257 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
2261 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
2262 g_return_val_if_fail (client->watch == NULL, 0);
2264 /* create watch for the connection and attach */
2265 client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
2266 g_object_ref (client), (GDestroyNotify) client_watch_notify);
2268 GST_INFO ("attaching to context %p", context);
2269 res = gst_rtsp_watch_attach (client->watch, context);
2270 gst_rtsp_watch_unref (client->watch);