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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, 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 (g_list_length (session->medias) > 0) {
184 GstRTSPSessionMedia *media = g_list_first (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 */
315 if (session->timeout != 60)
317 g_strdup_printf ("%s; timeout=%d", session->sessionid,
320 str = g_strdup (session->sessionid);
322 gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
325 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
326 gst_rtsp_message_dump (response);
329 gst_rtsp_watch_send_message (client->watch, response, NULL);
330 gst_rtsp_message_unset (response);
334 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
335 GstRTSPClientState * state)
337 gst_rtsp_message_init_response (state->response, code,
338 gst_rtsp_status_as_text (code), state->request);
340 send_response (client, NULL, state->response);
344 handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
345 GstRTSPClientState * state)
347 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
348 gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
351 /* and let the authentication manager setup the auth tokens */
352 gst_rtsp_auth_setup_auth (auth, client, 0, state);
355 send_response (client, state->session, state->response);
360 compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
362 if (uri1 == NULL || uri2 == NULL)
365 if (strcmp (uri1->abspath, uri2->abspath))
371 /* this function is called to initially find the media for the DESCRIBE request
372 * but is cached for when the same client (without breaking the connection) is
373 * doing a setup for the exact same url. */
374 static GstRTSPMedia *
375 find_media (GstRTSPClient * client, GstRTSPClientState * state)
377 GstRTSPMediaFactory *factory;
381 if (!compare_uri (client->uri, state->uri)) {
382 /* remove any previously cached values before we try to construct a new
385 gst_rtsp_url_free (client->uri);
388 g_object_unref (client->media);
389 client->media = NULL;
391 if (!client->media_mapping)
394 /* find the factory for the uri first */
396 gst_rtsp_media_mapping_find_factory (client->media_mapping,
400 state->factory = factory;
402 /* check if we have access to the factory */
403 if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
404 if (!gst_rtsp_auth_check (auth, client, 0, state))
407 g_object_unref (auth);
410 /* prepare the media and add it to the pipeline */
411 if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
414 g_object_unref (factory);
416 state->factory = NULL;
418 /* set ipv6 on the media before preparing */
419 media->is_ipv6 = client->is_ipv6;
420 state->media = media;
422 /* prepare the media */
423 if (!(gst_rtsp_media_prepare (media)))
426 /* now keep track of the uri and the media */
427 client->uri = gst_rtsp_url_copy (state->uri);
428 client->media = media;
430 /* we have seen this uri before, used cached media */
431 media = client->media;
432 state->media = media;
433 GST_INFO ("reusing cached media %p", media);
437 g_object_ref (media);
444 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
449 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
454 handle_unauthorized_request (client, auth, state);
455 g_object_unref (factory);
456 g_object_unref (auth);
461 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
462 g_object_unref (factory);
467 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
468 g_object_unref (media);
474 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
476 GstRTSPMessage message = { 0 };
481 gst_rtsp_message_init_data (&message, channel);
483 /* FIXME, need some sort of iovec RTSPMessage here */
484 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
487 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
489 /* FIXME, client->watch could have been finalized here, we need to keep an
490 * extra refcount to the watch. */
491 gst_rtsp_watch_send_message (client->watch, &message, NULL);
493 gst_rtsp_message_steal_body (&message, &data, &usize);
494 gst_buffer_unmap (buffer, &map_info);
496 gst_rtsp_message_unset (&message);
502 link_transport (GstRTSPClient * client, GstRTSPSession * session,
503 GstRTSPStreamTransport * trans)
505 GST_DEBUG ("client %p: linking transport %p", client, trans);
506 gst_rtsp_stream_transport_set_callbacks (trans,
507 (GstRTSPSendFunc) do_send_data,
508 (GstRTSPSendFunc) do_send_data, client, NULL);
509 client->transports = g_list_prepend (client->transports, trans);
510 /* make sure our session can't expire */
511 gst_rtsp_session_prevent_expire (session);
515 unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
516 GstRTSPStreamTransport * trans)
518 GST_DEBUG ("client %p: unlinking transport %p", client, trans);
519 gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
520 client->transports = g_list_remove (client->transports, trans);
521 /* our session can now expire */
522 gst_rtsp_session_allow_expire (session);
526 unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
527 GstRTSPSessionMedia * media)
531 n_streams = gst_rtsp_media_n_streams (media->media);
532 for (i = 0; i < n_streams; i++) {
533 GstRTSPStreamTransport *trans;
534 GstRTSPTransport *tr;
536 /* get the stream as configured in the session */
537 trans = gst_rtsp_session_media_get_transport (media, i);
538 /* get the transport, if there is no transport configured, skip this stream */
539 if (!(tr = trans->transport))
542 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
543 /* for TCP, unlink the stream from the TCP connection of the client */
544 unlink_transport (client, session, trans);
550 close_connection (GstRTSPClient * client)
552 const gchar *tunnelid;
554 GST_DEBUG ("client %p: closing connection", client);
556 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
557 g_mutex_lock (&tunnels_lock);
558 /* remove from tunnelids */
559 g_hash_table_remove (tunnels, tunnelid);
560 g_mutex_unlock (&tunnels_lock);
563 gst_rtsp_connection_close (client->connection);
567 handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
569 GstRTSPSession *session;
570 GstRTSPSessionMedia *media;
571 GstRTSPStatusCode code;
576 session = state->session;
578 /* get a handle to the configuration of the media in the session */
579 media = gst_rtsp_session_get_media (session, state->uri);
583 state->sessmedia = media;
585 /* unlink the all TCP callbacks */
586 unlink_session_transports (client, session, media);
588 /* remove the session from the watched sessions */
589 g_object_weak_unref (G_OBJECT (session),
590 (GWeakNotify) client_session_finalized, client);
591 client->sessions = g_list_remove (client->sessions, session);
593 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
595 /* unmanage the media in the session, returns false if all media session
597 if (!gst_rtsp_session_release_media (session, media)) {
598 /* remove the session */
599 gst_rtsp_session_pool_remove (client->session_pool, session);
601 /* construct the response now */
602 code = GST_RTSP_STS_OK;
603 gst_rtsp_message_init_response (state->response, code,
604 gst_rtsp_status_as_text (code), state->request);
606 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
609 send_response (client, session, state->response);
611 /* we emit the signal before closing the connection */
612 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
615 close_connection (client);
622 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
627 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
633 handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
639 res = gst_rtsp_message_get_body (state->request, &data, &size);
640 if (res != GST_RTSP_OK)
644 /* no body, keep-alive request */
645 send_generic_response (client, GST_RTSP_STS_OK, state);
647 /* there is a body, handle the params */
648 res = gst_rtsp_params_get (client, state);
649 if (res != GST_RTSP_OK)
652 send_response (client, state->session, state->response);
655 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
663 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
669 handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
675 res = gst_rtsp_message_get_body (state->request, &data, &size);
676 if (res != GST_RTSP_OK)
680 /* no body, keep-alive request */
681 send_generic_response (client, GST_RTSP_STS_OK, state);
683 /* there is a body, handle the params */
684 res = gst_rtsp_params_set (client, state);
685 if (res != GST_RTSP_OK)
688 send_response (client, state->session, state->response);
691 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
699 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
705 handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
707 GstRTSPSession *session;
708 GstRTSPSessionMedia *media;
709 GstRTSPStatusCode code;
711 if (!(session = state->session))
714 /* get a handle to the configuration of the media in the session */
715 media = gst_rtsp_session_get_media (session, state->uri);
719 state->sessmedia = media;
721 /* the session state must be playing or recording */
722 if (media->state != GST_RTSP_STATE_PLAYING &&
723 media->state != GST_RTSP_STATE_RECORDING)
726 /* unlink the all TCP callbacks */
727 unlink_session_transports (client, session, media);
729 /* then pause sending */
730 gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
732 /* construct the response now */
733 code = GST_RTSP_STS_OK;
734 gst_rtsp_message_init_response (state->response, code,
735 gst_rtsp_status_as_text (code), state->request);
737 send_response (client, session, state->response);
739 /* the state is now READY */
740 media->state = GST_RTSP_STATE_READY;
742 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
750 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
755 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
760 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
767 handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
769 GstRTSPSession *session;
770 GstRTSPSessionMedia *media;
771 GstRTSPStatusCode code;
773 guint n_streams, i, infocount;
775 GstRTSPTimeRange *range;
778 if (!(session = state->session))
781 /* get a handle to the configuration of the media in the session */
782 media = gst_rtsp_session_get_media (session, state->uri);
786 state->sessmedia = media;
788 /* the session state must be playing or ready */
789 if (media->state != GST_RTSP_STATE_PLAYING &&
790 media->state != GST_RTSP_STATE_READY)
793 /* parse the range header if we have one */
795 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
796 if (res == GST_RTSP_OK) {
797 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
798 /* we have a range, seek to the position */
799 gst_rtsp_media_seek (media->media, range);
800 gst_rtsp_range_free (range);
804 /* grab RTPInfo from the payloaders now */
805 rtpinfo = g_string_new ("");
807 n_streams = gst_rtsp_media_n_streams (media->media);
808 for (i = 0, infocount = 0; i < n_streams; i++) {
809 GstRTSPStreamTransport *trans;
810 GstRTSPTransport *tr;
814 /* get the stream as configured in the session */
815 trans = gst_rtsp_session_media_get_transport (media, i);
816 /* get the transport, if there is no transport configured, skip this stream */
817 if (!(tr = trans->transport)) {
818 GST_INFO ("stream %d is not configured", i);
822 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
823 /* for TCP, link the stream to the TCP connection of the client */
824 link_transport (client, session, trans);
827 if (gst_rtsp_stream_get_rtpinfo (trans->stream, &rtptime, &seq)) {
829 g_string_append (rtpinfo, ", ");
831 uristr = gst_rtsp_url_get_request_uri (state->uri);
832 g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
833 uristr, i, seq, rtptime);
838 GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
842 /* construct the response now */
843 code = GST_RTSP_STS_OK;
844 gst_rtsp_message_init_response (state->response, code,
845 gst_rtsp_status_as_text (code), state->request);
847 /* add the RTP-Info header */
849 str = g_string_free (rtpinfo, FALSE);
850 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
852 g_string_free (rtpinfo, TRUE);
856 str = gst_rtsp_media_get_range_string (media->media, TRUE);
857 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
859 send_response (client, session, state->response);
861 /* start playing after sending the request */
862 gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
864 media->state = GST_RTSP_STATE_PLAYING;
866 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
874 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
879 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
884 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
891 do_keepalive (GstRTSPSession * session)
893 GST_INFO ("keep session %p alive", session);
894 gst_rtsp_session_touch (session);
897 /* parse @transport and return a valid transport in @tr. only transports
898 * from @supported are returned. Returns FALSE if no valid transport
901 parse_transport (const char *transport, GstRTSPLowerTrans supported,
902 GstRTSPTransport * tr)
909 gst_rtsp_transport_init (tr);
911 GST_WARNING ("parsing transports %s", transport);
913 transports = g_strsplit (transport, ",", 0);
915 /* loop through the transports, try to parse */
916 for (i = 0; transports[i]; i++) {
917 res = gst_rtsp_transport_parse (transports[i], tr);
918 if (res != GST_RTSP_OK) {
919 /* no valid transport, search some more */
920 GST_WARNING ("could not parse transport %s", transports[i]);
924 /* we have a transport, see if it's RTP/AVP */
925 if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
926 GST_WARNING ("invalid transport %s", transports[i]);
930 if (!(tr->lower_transport & supported)) {
931 GST_WARNING ("unsupported transport %s", transports[i]);
935 /* we have a valid transport */
936 GST_INFO ("found valid transport %s", transports[i]);
941 gst_rtsp_transport_init (tr);
943 g_strfreev (transports);
949 handle_blocksize (GstRTSPMedia * media, GstRTSPMessage * request)
951 gchar *blocksize_str;
954 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
955 &blocksize_str, 0) == GST_RTSP_OK) {
959 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
960 if (end == blocksize_str) {
961 GST_ERROR ("failed to parse blocksize");
964 /* we don't want to change the mtu when this media
965 * can be shared because it impacts other clients */
966 if (gst_rtsp_media_is_shared (media))
969 if (blocksize > G_MAXUINT)
970 blocksize = G_MAXUINT;
971 gst_rtsp_media_set_mtu (media, blocksize);
979 handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
984 GstRTSPTransport *ct, *st;
985 GstRTSPLowerTrans supported;
986 GstRTSPStatusCode code;
987 GstRTSPSession *session;
988 GstRTSPStreamTransport *trans;
989 gchar *trans_str, *pos;
991 GstRTSPSessionMedia *media;
995 /* the uri contains the stream number we added in the SDP config, which is
996 * always /stream=%d so we need to strip that off
997 * parse the stream we need to configure, look for the stream in the abspath
998 * first and then in the query. */
999 if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
1000 if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
1004 /* we can mofify the parsed uri in place */
1007 pos += strlen ("/stream=");
1008 if (sscanf (pos, "%u", &streamid) != 1)
1011 /* parse the transport */
1013 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
1015 if (res != GST_RTSP_OK)
1018 gst_rtsp_transport_new (&ct);
1020 /* our supported transports */
1021 supported = GST_RTSP_LOWER_TRANS_UDP |
1022 GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
1024 /* parse and find a usable supported transport */
1025 if (!parse_transport (transport, supported, ct))
1026 goto unsupported_transports;
1028 if (client->session_pool == NULL)
1031 session = state->session;
1034 g_object_ref (session);
1035 /* get a handle to the configuration of the media in the session, this can
1036 * return NULL if this is a new url to manage in this session. */
1037 media = gst_rtsp_session_get_media (session, uri);
1039 /* create a session if this fails we probably reached our session limit or
1041 if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
1042 goto service_unavailable;
1044 state->session = session;
1046 /* we need a new media configuration in this session */
1050 /* we have no media, find one and manage it */
1051 if (media == NULL) {
1054 /* get a handle to the configuration of the media in the session */
1055 if ((m = find_media (client, state))) {
1056 /* manage the media in our session now */
1057 media = gst_rtsp_session_manage_media (session, uri, m);
1061 /* if we stil have no media, error */
1065 state->sessmedia = media;
1067 if (!handle_blocksize (media->media, state->request))
1068 goto invalid_blocksize;
1070 /* we have a valid transport now, set the destination of the client. */
1071 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1072 if (ct->destination == NULL || !client->use_client_settings) {
1073 g_free (ct->destination);
1074 ct->destination = gst_rtsp_media_get_multicast_group (media->media);
1076 /* reset ttl if client settings are not allowed */
1077 if (!client->use_client_settings) {
1083 url = gst_rtsp_connection_get_url (client->connection);
1084 g_free (ct->destination);
1085 ct->destination = g_strdup (url->host);
1087 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1088 /* check if the client selected channels for TCP */
1089 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1090 gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
1095 /* get a handle to the transport of the media in this session */
1096 if (!(trans = gst_rtsp_session_media_get_transport (media, streamid)))
1097 goto no_stream_transport;
1099 st = gst_rtsp_stream_transport_set_transport (trans, ct);
1101 /* configure keepalive for this transport */
1102 gst_rtsp_stream_transport_set_keepalive (trans,
1103 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1105 /* serialize the server transport */
1106 trans_str = gst_rtsp_transport_as_text (st);
1107 gst_rtsp_transport_free (st);
1109 /* construct the response now */
1110 code = GST_RTSP_STS_OK;
1111 gst_rtsp_message_init_response (state->response, code,
1112 gst_rtsp_status_as_text (code), state->request);
1114 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
1118 send_response (client, session, state->response);
1120 /* update the state */
1121 switch (media->state) {
1122 case GST_RTSP_STATE_PLAYING:
1123 case GST_RTSP_STATE_RECORDING:
1124 case GST_RTSP_STATE_READY:
1125 /* no state change */
1128 media->state = GST_RTSP_STATE_READY;
1131 g_object_unref (session);
1133 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
1141 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1146 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1147 g_object_unref (session);
1148 gst_rtsp_transport_free (ct);
1153 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1154 g_object_unref (session);
1155 gst_rtsp_transport_free (ct);
1158 no_stream_transport:
1160 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1161 g_object_unref (session);
1162 gst_rtsp_transport_free (ct);
1167 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1170 unsupported_transports:
1172 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1173 gst_rtsp_transport_free (ct);
1178 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1179 gst_rtsp_transport_free (ct);
1182 service_unavailable:
1184 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1185 gst_rtsp_transport_free (ct);
1190 static GstSDPMessage *
1191 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1196 GstRTSPLowerTrans protocols;
1198 gst_sdp_message_new (&sdp);
1200 /* some standard things first */
1201 gst_sdp_message_set_version (sdp, "0");
1203 if (client->is_ipv6)
1208 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
1211 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
1212 gst_sdp_message_set_information (sdp, "rtsp-server");
1213 gst_sdp_message_add_time (sdp, "0", "0", NULL);
1214 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
1215 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
1216 gst_sdp_message_add_attribute (sdp, "control", "*");
1218 info.server_proto = proto;
1219 protocols = gst_rtsp_media_get_protocols (media);
1220 if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
1221 info.server_ip = gst_rtsp_media_get_multicast_group (media);
1223 info.server_ip = g_strdup (client->server_ip);
1225 /* create an SDP for the media object */
1226 if (!gst_rtsp_sdp_from_media (sdp, &info, media))
1229 g_free (info.server_ip);
1236 g_free (info.server_ip);
1237 gst_sdp_message_free (sdp);
1242 /* for the describe we must generate an SDP */
1244 handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
1249 gchar *str, *content_base;
1250 GstRTSPMedia *media;
1251 GstRTSPClientClass *klass;
1253 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1255 /* check what kind of format is accepted, we don't really do anything with it
1256 * and always return SDP for now. */
1261 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
1263 if (res == GST_RTSP_ENOTIMPL)
1266 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
1270 /* find the media object for the uri */
1271 if (!(media = find_media (client, state)))
1274 /* create an SDP for the media object on this client */
1275 if (!(sdp = klass->create_sdp (client, media)))
1278 g_object_unref (media);
1280 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1281 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1283 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
1286 /* content base for some clients that might screw up creating the setup uri */
1287 str = gst_rtsp_url_get_request_uri (state->uri);
1288 str_len = strlen (str);
1290 /* check for trailing '/' and append one */
1291 if (str[str_len - 1] != '/') {
1292 content_base = g_malloc (str_len + 2);
1293 memcpy (content_base, str, str_len);
1294 content_base[str_len] = '/';
1295 content_base[str_len + 1] = '\0';
1301 GST_INFO ("adding content-base: %s", content_base);
1303 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
1305 g_free (content_base);
1307 /* add SDP to the response body */
1308 str = gst_sdp_message_as_text (sdp);
1309 gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
1310 gst_sdp_message_free (sdp);
1312 send_response (client, state->session, state->response);
1314 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
1322 /* error reply is already sent */
1327 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1328 g_object_unref (media);
1334 handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
1336 GstRTSPMethod options;
1339 options = GST_RTSP_DESCRIBE |
1344 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
1346 str = gst_rtsp_options_as_text (options);
1348 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1349 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1351 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
1354 send_response (client, state->session, state->response);
1356 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
1362 /* remove duplicate and trailing '/' */
1364 sanitize_uri (GstRTSPUrl * uri)
1368 gboolean have_slash, prev_slash;
1370 s = d = uri->abspath;
1371 len = strlen (uri->abspath);
1375 for (i = 0; i < len; i++) {
1376 have_slash = s[i] == '/';
1378 if (!have_slash || !prev_slash)
1380 prev_slash = have_slash;
1382 len = d - uri->abspath;
1383 /* don't remove the first slash if that's the only thing left */
1384 if (len > 1 && *(d - 1) == '/')
1390 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
1392 GST_INFO ("client %p: session %p finished", client, session);
1394 /* unlink all media managed in this session */
1395 client_unlink_session (client, session);
1397 /* remove the session */
1398 if (!(client->sessions = g_list_remove (client->sessions, session))) {
1399 GST_INFO ("client %p: all sessions finalized, close the connection",
1401 close_connection (client);
1406 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
1410 for (walk = client->sessions; walk; walk = g_list_next (walk)) {
1411 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
1413 /* we already know about this session */
1414 if (msession == session)
1418 GST_INFO ("watching session %p", session);
1420 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
1422 client->sessions = g_list_prepend (client->sessions, session);
1424 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1429 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
1431 GstRTSPMethod method;
1432 const gchar *uristr;
1434 GstRTSPVersion version;
1436 GstRTSPSession *session;
1437 GstRTSPClientState state = { NULL };
1438 GstRTSPMessage response = { 0 };
1441 state.request = request;
1442 state.response = &response;
1444 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
1445 gst_rtsp_message_dump (request);
1448 GST_INFO ("client %p: received a request", client);
1450 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
1452 if (version != GST_RTSP_VERSION_1_0) {
1453 /* we can only handle 1.0 requests */
1454 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
1458 state.method = method;
1460 /* we always try to parse the url first */
1461 if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
1462 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1466 /* sanitize the uri */
1470 /* get the session if there is any */
1471 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
1472 if (res == GST_RTSP_OK) {
1473 if (client->session_pool == NULL)
1476 /* we had a session in the request, find it again */
1477 if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
1478 goto session_not_found;
1480 /* we add the session to the client list of watched sessions. When a session
1481 * disappears because it times out, we will be notified. If all sessions are
1482 * gone, we will close the connection */
1483 client_watch_session (client, session);
1487 state.session = session;
1490 if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
1491 goto not_authorized;
1494 /* now see what is asked and dispatch to a dedicated handler */
1496 case GST_RTSP_OPTIONS:
1497 handle_options_request (client, &state);
1499 case GST_RTSP_DESCRIBE:
1500 handle_describe_request (client, &state);
1502 case GST_RTSP_SETUP:
1503 handle_setup_request (client, &state);
1506 handle_play_request (client, &state);
1508 case GST_RTSP_PAUSE:
1509 handle_pause_request (client, &state);
1511 case GST_RTSP_TEARDOWN:
1512 handle_teardown_request (client, &state);
1514 case GST_RTSP_SET_PARAMETER:
1515 handle_set_param_request (client, &state);
1517 case GST_RTSP_GET_PARAMETER:
1518 handle_get_param_request (client, &state);
1520 case GST_RTSP_ANNOUNCE:
1521 case GST_RTSP_RECORD:
1522 case GST_RTSP_REDIRECT:
1523 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
1525 case GST_RTSP_INVALID:
1527 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1531 g_object_unref (session);
1533 gst_rtsp_url_free (uri);
1539 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
1544 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
1549 handle_unauthorized_request (client, client->auth, &state);
1555 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
1565 /* find the stream for this message */
1566 res = gst_rtsp_message_parse_data (message, &channel);
1567 if (res != GST_RTSP_OK)
1570 gst_rtsp_message_steal_body (message, &data, &size);
1572 buffer = gst_buffer_new_wrapped (data, size);
1575 for (walk = client->transports; walk; walk = g_list_next (walk)) {
1576 GstRTSPStreamTransport *trans = (GstRTSPStreamTransport *) walk->data;
1577 GstRTSPStream *stream;
1578 GstRTSPTransport *tr;
1580 /* get the transport, if there is no transport configured, skip this stream */
1581 if (!(tr = trans->transport))
1584 /* we also need a media stream */
1585 if (!(stream = trans->stream))
1588 /* check for TCP transport */
1589 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1590 /* dispatch to the stream based on the channel number */
1591 if (tr->interleaved.min == channel) {
1592 gst_rtsp_stream_recv_rtp (stream, buffer);
1595 } else if (tr->interleaved.max == channel) {
1596 gst_rtsp_stream_recv_rtcp (stream, buffer);
1603 gst_buffer_unref (buffer);
1607 * gst_rtsp_client_set_session_pool:
1608 * @client: a #GstRTSPClient
1609 * @pool: a #GstRTSPSessionPool
1611 * Set @pool as the sessionpool for @client which it will use to find
1612 * or allocate sessions. the sessionpool is usually inherited from the server
1613 * that created the client but can be overridden later.
1616 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
1617 GstRTSPSessionPool * pool)
1619 GstRTSPSessionPool *old;
1621 old = client->session_pool;
1624 g_object_ref (pool);
1625 client->session_pool = pool;
1627 g_object_unref (old);
1632 * gst_rtsp_client_get_session_pool:
1633 * @client: a #GstRTSPClient
1635 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
1637 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
1639 GstRTSPSessionPool *
1640 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
1642 GstRTSPSessionPool *result;
1644 if ((result = client->session_pool))
1645 g_object_ref (result);
1651 * gst_rtsp_client_set_server:
1652 * @client: a #GstRTSPClient
1653 * @server: a #GstRTSPServer
1655 * Set @server as the server that created @client.
1658 gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
1662 old = client->server;
1663 if (old != server) {
1665 g_object_ref (server);
1666 client->server = server;
1668 g_object_unref (old);
1673 * gst_rtsp_client_get_server:
1674 * @client: a #GstRTSPClient
1676 * Get the #GstRTSPServer object that @client was created from.
1678 * Returns: (transfer full): a #GstRTSPServer, unref after usage.
1681 gst_rtsp_client_get_server (GstRTSPClient * client)
1683 GstRTSPServer *result;
1685 if ((result = client->server))
1686 g_object_ref (result);
1692 * gst_rtsp_client_set_media_mapping:
1693 * @client: a #GstRTSPClient
1694 * @mapping: a #GstRTSPMediaMapping
1696 * Set @mapping as the media mapping for @client which it will use to map urls
1697 * to media streams. These mapping is usually inherited from the server that
1698 * created the client but can be overriden later.
1701 gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
1702 GstRTSPMediaMapping * mapping)
1704 GstRTSPMediaMapping *old;
1706 old = client->media_mapping;
1708 if (old != mapping) {
1710 g_object_ref (mapping);
1711 client->media_mapping = mapping;
1713 g_object_unref (old);
1718 * gst_rtsp_client_get_media_mapping:
1719 * @client: a #GstRTSPClient
1721 * Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
1723 * Returns: (transfer full): a #GstRTSPMediaMapping, unref after usage.
1725 GstRTSPMediaMapping *
1726 gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
1728 GstRTSPMediaMapping *result;
1730 if ((result = client->media_mapping))
1731 g_object_ref (result);
1737 * gst_rtsp_client_set_use_client_settings:
1738 * @client: a #GstRTSPClient
1739 * @use_client_settings: whether to use client settings for multicast
1741 * Use client transport settings (destination and ttl) for multicast.
1742 * When @use_client_settings is %FALSE, the server settings will be
1746 gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
1747 gboolean use_client_settings)
1749 client->use_client_settings = use_client_settings;
1753 * gst_rtsp_client_get_use_client_settings:
1754 * @client: a #GstRTSPClient
1756 * Check if client transport settings (destination and ttl) for multicast
1760 gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
1762 return client->use_client_settings;
1766 * gst_rtsp_client_set_auth:
1767 * @client: a #GstRTSPClient
1768 * @auth: a #GstRTSPAuth
1770 * configure @auth to be used as the authentication manager of @client.
1773 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
1777 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1783 g_object_ref (auth);
1784 client->auth = auth;
1786 g_object_unref (old);
1792 * gst_rtsp_client_get_auth:
1793 * @client: a #GstRTSPClient
1795 * Get the #GstRTSPAuth used as the authentication manager of @client.
1797 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
1801 gst_rtsp_client_get_auth (GstRTSPClient * client)
1803 GstRTSPAuth *result;
1805 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
1807 if ((result = client->auth))
1808 g_object_ref (result);
1813 static GstRTSPResult
1814 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
1817 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1819 switch (message->type) {
1820 case GST_RTSP_MESSAGE_REQUEST:
1821 handle_request (client, message);
1823 case GST_RTSP_MESSAGE_RESPONSE:
1825 case GST_RTSP_MESSAGE_DATA:
1826 handle_data (client, message);
1834 static GstRTSPResult
1835 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
1837 /* GstRTSPClient *client; */
1839 /* client = GST_RTSP_CLIENT (user_data); */
1841 /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
1846 static GstRTSPResult
1847 closed (GstRTSPWatch * watch, gpointer user_data)
1849 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1850 const gchar *tunnelid;
1852 GST_INFO ("client %p: connection closed", client);
1854 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
1855 g_mutex_lock (&tunnels_lock);
1856 /* remove from tunnelids */
1857 g_hash_table_remove (tunnels, tunnelid);
1858 g_mutex_unlock (&tunnels_lock);
1864 static GstRTSPResult
1865 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
1867 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1870 str = gst_rtsp_strresult (result);
1871 GST_INFO ("client %p: received an error %s", client, str);
1877 static GstRTSPResult
1878 error_full (GstRTSPWatch * watch, GstRTSPResult result,
1879 GstRTSPMessage * message, guint id, gpointer user_data)
1881 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1884 str = gst_rtsp_strresult (result);
1886 ("client %p: received an error %s when handling message %p with id %d",
1887 client, str, message, id);
1894 remember_tunnel (GstRTSPClient * client)
1896 const gchar *tunnelid;
1898 /* store client in the pending tunnels */
1899 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1900 if (tunnelid == NULL)
1903 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
1905 /* we can't have two clients connecting with the same tunnelid */
1906 g_mutex_lock (&tunnels_lock);
1907 if (g_hash_table_lookup (tunnels, tunnelid))
1908 goto tunnel_existed;
1910 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
1911 g_mutex_unlock (&tunnels_lock);
1918 GST_ERROR ("client %p: no tunnelid provided", client);
1923 g_mutex_unlock (&tunnels_lock);
1924 GST_ERROR ("client %p: tunnel session %s already existed", client,
1930 static GstRTSPStatusCode
1931 tunnel_start (GstRTSPWatch * watch, gpointer user_data)
1933 GstRTSPClient *client;
1935 client = GST_RTSP_CLIENT (user_data);
1937 GST_INFO ("client %p: tunnel start (connection %p)", client,
1938 client->connection);
1940 if (!remember_tunnel (client))
1943 return GST_RTSP_STS_OK;
1948 GST_ERROR ("client %p: error starting tunnel", client);
1949 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1953 static GstRTSPResult
1954 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
1956 GstRTSPClient *client;
1958 client = GST_RTSP_CLIENT (user_data);
1960 GST_INFO ("client %p: tunnel lost (connection %p)", client,
1961 client->connection);
1963 /* ignore error, it'll only be a problem when the client does a POST again */
1964 remember_tunnel (client);
1969 static GstRTSPResult
1970 tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
1972 const gchar *tunnelid;
1973 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1974 GstRTSPClient *oclient;
1976 GST_INFO ("client %p: tunnel complete", client);
1978 /* find previous tunnel */
1979 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1980 if (tunnelid == NULL)
1983 g_mutex_lock (&tunnels_lock);
1984 if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
1987 /* remove the old client from the table. ref before because removing it will
1988 * remove the ref to it. */
1989 g_object_ref (oclient);
1990 g_hash_table_remove (tunnels, tunnelid);
1992 if (oclient->watch == NULL)
1994 g_mutex_unlock (&tunnels_lock);
1996 GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
1997 oclient->connection, client->connection);
1999 /* merge the tunnels into the first client */
2000 gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
2001 gst_rtsp_watch_reset (oclient->watch);
2002 g_object_unref (oclient);
2009 GST_INFO ("client %p: no tunnelid provided", client);
2010 return GST_RTSP_ERROR;
2014 g_mutex_unlock (&tunnels_lock);
2015 GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
2016 return GST_RTSP_ERROR;
2020 g_mutex_unlock (&tunnels_lock);
2021 GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
2022 g_object_unref (oclient);
2023 return GST_RTSP_ERROR;
2027 static GstRTSPWatchFuncs watch_funcs = {
2039 client_watch_notify (GstRTSPClient * client)
2041 GST_INFO ("client %p: watch destroyed", client);
2042 client->watchid = 0;
2043 client->watch = NULL;
2044 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
2045 g_object_unref (client);
2049 attach_client (GstRTSPClient * client, GSocket * socket,
2050 GstRTSPConnection * conn, GError ** error)
2052 GSocket *read_socket;
2053 GSocketAddress *address;
2055 GMainContext *context;
2058 read_socket = gst_rtsp_connection_get_read_socket (conn);
2059 client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
2061 if (!(address = g_socket_get_remote_address (read_socket, error)))
2064 g_free (client->server_ip);
2065 /* keep the original ip that the client connected to */
2066 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2067 GInetAddress *iaddr;
2069 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2071 client->server_ip = g_inet_address_to_string (iaddr);
2072 g_object_unref (address);
2074 client->server_ip = g_strdup ("unknown");
2077 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2078 client->server_ip, client->is_ipv6);
2080 url = gst_rtsp_connection_get_url (conn);
2081 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2083 client->connection = conn;
2085 /* create watch for the connection and attach */
2086 client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
2087 g_object_ref (client), (GDestroyNotify) client_watch_notify);
2089 /* find the context to add the watch */
2090 if ((source = g_main_current_source ()))
2091 context = g_source_get_context (source);
2095 GST_INFO ("attaching to context %p", context);
2097 client->watchid = gst_rtsp_watch_attach (client->watch, context);
2098 gst_rtsp_watch_unref (client->watch);
2105 GST_ERROR ("could not get remote address %s", (*error)->message);
2111 * gst_rtsp_client_create_from_socket:
2112 * @client: a #GstRTSPClient
2113 * @socket: a #GSocket
2114 * @ip: the IP address of the remote client
2115 * @port: the port used by the other end
2116 * @initial_buffer: any initial data that was already read from the socket
2119 * Take an existing network socket and use it for an RTSP connection.
2121 * Returns: %TRUE on success.
2124 gst_rtsp_client_create_from_socket (GstRTSPClient * client, GSocket * socket,
2125 const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
2127 GstRTSPConnection *conn;
2130 GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
2131 initial_buffer, &conn), no_connection);
2133 return attach_client (client, socket, conn, error);
2138 gchar *str = gst_rtsp_strresult (res);
2140 GST_ERROR ("could not create connection from socket %p: %s", socket, str);
2147 * gst_rtsp_client_accept:
2148 * @client: a #GstRTSPClient
2149 * @socket: a #GSocket
2150 * @cancellable: a #GCancellable
2153 * Accept a new connection for @client on @socket.
2155 * This function should be called when the client properties and urls are fully
2156 * configured and the client is ready to start.
2158 * Returns: %TRUE if the client could be accepted.
2161 gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
2162 GCancellable * cancellable, GError ** error)
2164 GstRTSPConnection *conn;
2167 /* a new client connected. */
2168 GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
2171 return attach_client (client, socket, conn, error);
2176 gchar *str = gst_rtsp_strresult (res);
2178 GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);