2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: A client connection state
24 * @see_also: #GstRTSPServer, #GstRTSPThreadPool
26 * The client object handles the connection with a client for as long as a TCP
29 * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
30 * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
31 * #GstRTSPAuth and #GstRTSPThreadPool from the server.
33 * The client connection should be configured with the #GstRTSPConnection using
34 * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
35 * using gst_rtsp_client_attach(). From then on the client will handle requests
38 * Use gst_rtsp_client_session_filter() to iterate or modify all the
39 * #GstRTSPSession objects managed by the client object.
41 * Last reviewed on 2013-07-11 (1.0.0)
47 #include <gst/sdp/gstmikey.h>
48 #include <gst/rtsp/gstrtsp-enumtypes.h>
50 #include "rtsp-client.h"
52 #include "rtsp-params.h"
62 * send_lock, lock, tunnels_lock
65 struct _GstRTSPClientPrivate
67 GMutex lock; /* protects everything else */
70 GstRTSPConnection *connection;
72 GMainContext *watch_context;
76 /* protected by send_lock */
77 GstRTSPClientSendFunc send_func;
79 GDestroyNotify send_notify;
83 GstRTSPSessionPool *session_pool;
84 gulong session_removed_id;
85 GstRTSPMountPoints *mount_points;
87 GstRTSPThreadPool *thread_pool;
89 /* used to cache the media in the last requested DESCRIBE so that
90 * we can pick it up in the next SETUP immediately */
94 GHashTable *transports;
96 guint sessions_cookie;
98 gboolean drop_backlog;
100 guint rtsp_ctrl_timeout_id;
101 guint rtsp_ctrl_timeout_cnt;
103 /* The version currently being used */
104 GstRTSPVersion version;
106 GHashTable *pipelined_requests; /* pipelined_request_id -> session_id */
107 GstRTSPTunnelState tstate;
116 static GMutex tunnels_lock;
117 static GHashTable *tunnels; /* protected by tunnels_lock */
119 #define WATCH_BACKLOG_SIZE 100
121 #define DEFAULT_SESSION_POOL NULL
122 #define DEFAULT_MOUNT_POINTS NULL
123 #define DEFAULT_DROP_BACKLOG TRUE
125 #define RTSP_CTRL_CB_INTERVAL 1
126 #define RTSP_CTRL_TIMEOUT_VALUE 60
141 SIGNAL_PRE_OPTIONS_REQUEST,
142 SIGNAL_OPTIONS_REQUEST,
143 SIGNAL_PRE_DESCRIBE_REQUEST,
144 SIGNAL_DESCRIBE_REQUEST,
145 SIGNAL_PRE_SETUP_REQUEST,
146 SIGNAL_SETUP_REQUEST,
147 SIGNAL_PRE_PLAY_REQUEST,
149 SIGNAL_PRE_PAUSE_REQUEST,
150 SIGNAL_PAUSE_REQUEST,
151 SIGNAL_PRE_TEARDOWN_REQUEST,
152 SIGNAL_TEARDOWN_REQUEST,
153 SIGNAL_PRE_SET_PARAMETER_REQUEST,
154 SIGNAL_SET_PARAMETER_REQUEST,
155 SIGNAL_PRE_GET_PARAMETER_REQUEST,
156 SIGNAL_GET_PARAMETER_REQUEST,
157 SIGNAL_HANDLE_RESPONSE,
159 SIGNAL_PRE_ANNOUNCE_REQUEST,
160 SIGNAL_ANNOUNCE_REQUEST,
161 SIGNAL_PRE_RECORD_REQUEST,
162 SIGNAL_RECORD_REQUEST,
163 SIGNAL_CHECK_REQUIREMENTS,
167 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
168 #define GST_CAT_DEFAULT rtsp_client_debug
170 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
172 static void gst_rtsp_client_get_property (GObject * object, guint propid,
173 GValue * value, GParamSpec * pspec);
174 static void gst_rtsp_client_set_property (GObject * object, guint propid,
175 const GValue * value, GParamSpec * pspec);
176 static void gst_rtsp_client_finalize (GObject * obj);
178 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
179 static gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
180 GstRTSPMedia * media, GstSDPMessage * sdp);
181 static gboolean default_configure_client_media (GstRTSPClient * client,
182 GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
183 static gboolean default_configure_client_transport (GstRTSPClient * client,
184 GstRTSPContext * ctx, GstRTSPTransport * ct);
185 static GstRTSPResult default_params_set (GstRTSPClient * client,
186 GstRTSPContext * ctx);
187 static GstRTSPResult default_params_get (GstRTSPClient * client,
188 GstRTSPContext * ctx);
189 static gchar *default_make_path_from_uri (GstRTSPClient * client,
190 const GstRTSPUrl * uri);
191 static void client_session_removed (GstRTSPSessionPool * pool,
192 GstRTSPSession * session, GstRTSPClient * client);
193 static GstRTSPStatusCode default_pre_signal_handler (GstRTSPClient * client,
194 GstRTSPContext * ctx);
195 static gboolean pre_signal_accumulator (GSignalInvocationHint * ihint,
196 GValue * return_accu, const GValue * handler_return, gpointer data);
198 G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
201 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
203 GObjectClass *gobject_class;
205 gobject_class = G_OBJECT_CLASS (klass);
207 gobject_class->get_property = gst_rtsp_client_get_property;
208 gobject_class->set_property = gst_rtsp_client_set_property;
209 gobject_class->finalize = gst_rtsp_client_finalize;
211 klass->create_sdp = create_sdp;
212 klass->handle_sdp = handle_sdp;
213 klass->configure_client_media = default_configure_client_media;
214 klass->configure_client_transport = default_configure_client_transport;
215 klass->params_set = default_params_set;
216 klass->params_get = default_params_get;
217 klass->make_path_from_uri = default_make_path_from_uri;
219 klass->pre_options_request = default_pre_signal_handler;
220 klass->pre_describe_request = default_pre_signal_handler;
221 klass->pre_setup_request = default_pre_signal_handler;
222 klass->pre_play_request = default_pre_signal_handler;
223 klass->pre_pause_request = default_pre_signal_handler;
224 klass->pre_teardown_request = default_pre_signal_handler;
225 klass->pre_set_parameter_request = default_pre_signal_handler;
226 klass->pre_get_parameter_request = default_pre_signal_handler;
227 klass->pre_announce_request = default_pre_signal_handler;
228 klass->pre_record_request = default_pre_signal_handler;
230 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
231 g_param_spec_object ("session-pool", "Session Pool",
232 "The session pool to use for client session",
233 GST_TYPE_RTSP_SESSION_POOL,
234 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
236 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
237 g_param_spec_object ("mount-points", "Mount Points",
238 "The mount points to use for client session",
239 GST_TYPE_RTSP_MOUNT_POINTS,
240 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
242 g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
243 g_param_spec_boolean ("drop-backlog", "Drop Backlog",
244 "Drop data when the backlog queue is full",
245 DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
247 gst_rtsp_client_signals[SIGNAL_CLOSED] =
248 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
249 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
250 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
252 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
253 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
254 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
255 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
258 * GstRTSPClient::pre-options-request:
259 * @client: a #GstRTSPClient
260 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
262 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
263 * otherwise an appropriate return code
267 gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST] =
268 g_signal_new ("pre-options-request", G_TYPE_FROM_CLASS (klass),
269 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
270 pre_options_request), pre_signal_accumulator, NULL,
271 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
272 GST_TYPE_RTSP_CONTEXT);
275 * GstRTSPClient::options-request:
276 * @client: a #GstRTSPClient
277 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
279 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
280 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
281 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
282 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
283 GST_TYPE_RTSP_CONTEXT);
286 * GstRTSPClient::pre-describe-request:
287 * @client: a #GstRTSPClient
288 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
290 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
291 * otherwise an appropriate return code
295 gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST] =
296 g_signal_new ("pre-describe-request", G_TYPE_FROM_CLASS (klass),
297 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
298 pre_describe_request), pre_signal_accumulator, NULL,
299 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
300 GST_TYPE_RTSP_CONTEXT);
303 * GstRTSPClient::describe-request:
304 * @client: a #GstRTSPClient
305 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
307 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
308 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
309 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
310 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
311 GST_TYPE_RTSP_CONTEXT);
314 * GstRTSPClient::pre-setup-request:
315 * @client: a #GstRTSPClient
316 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
318 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
319 * otherwise an appropriate return code
323 gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST] =
324 g_signal_new ("pre-setup-request", G_TYPE_FROM_CLASS (klass),
325 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
326 pre_setup_request), pre_signal_accumulator, NULL,
327 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
328 GST_TYPE_RTSP_CONTEXT);
331 * GstRTSPClient::setup-request:
332 * @client: a #GstRTSPClient
333 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
335 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
336 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
337 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
338 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
339 GST_TYPE_RTSP_CONTEXT);
342 * GstRTSPClient::pre-play-request:
343 * @client: a #GstRTSPClient
344 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
346 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
347 * otherwise an appropriate return code
351 gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST] =
352 g_signal_new ("pre-play-request", G_TYPE_FROM_CLASS (klass),
353 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
354 pre_play_request), pre_signal_accumulator, NULL,
355 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
356 GST_TYPE_RTSP_CONTEXT);
359 * GstRTSPClient::play-request:
360 * @client: a #GstRTSPClient
361 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
363 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
364 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
365 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
366 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
367 GST_TYPE_RTSP_CONTEXT);
370 * GstRTSPClient::pre-pause-request:
371 * @client: a #GstRTSPClient
372 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
374 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
375 * otherwise an appropriate return code
379 gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST] =
380 g_signal_new ("pre-pause-request", G_TYPE_FROM_CLASS (klass),
381 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
382 pre_pause_request), pre_signal_accumulator, NULL,
383 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
384 GST_TYPE_RTSP_CONTEXT);
387 * GstRTSPClient::pause-request:
388 * @client: a #GstRTSPClient
389 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
391 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
392 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
393 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
394 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
395 GST_TYPE_RTSP_CONTEXT);
398 * GstRTSPClient::pre-teardown-request:
399 * @client: a #GstRTSPClient
400 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
402 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
403 * otherwise an appropriate return code
407 gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST] =
408 g_signal_new ("pre-teardown-request", G_TYPE_FROM_CLASS (klass),
409 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
410 pre_teardown_request), pre_signal_accumulator, NULL,
411 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
412 GST_TYPE_RTSP_CONTEXT);
415 * GstRTSPClient::teardown-request:
416 * @client: a #GstRTSPClient
417 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
419 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
420 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
421 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
422 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
423 GST_TYPE_RTSP_CONTEXT);
426 * GstRTSPClient::pre-set-parameter-request:
427 * @client: a #GstRTSPClient
428 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
430 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
431 * otherwise an appropriate return code
435 gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST] =
436 g_signal_new ("pre-set-parameter-request", G_TYPE_FROM_CLASS (klass),
437 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
438 pre_set_parameter_request), pre_signal_accumulator, NULL,
439 g_cclosure_marshal_generic,
440 GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
443 * GstRTSPClient::set-parameter-request:
444 * @client: a #GstRTSPClient
445 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
447 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
448 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
449 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
450 set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
451 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
454 * GstRTSPClient::pre-get-parameter-request:
455 * @client: a #GstRTSPClient
456 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
458 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
459 * otherwise an appropriate return code
463 gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST] =
464 g_signal_new ("pre-get-parameter-request", G_TYPE_FROM_CLASS (klass),
465 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
466 pre_get_parameter_request), pre_signal_accumulator, NULL,
467 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
468 GST_TYPE_RTSP_CONTEXT);
471 * GstRTSPClient::get-parameter-request:
472 * @client: a #GstRTSPClient
473 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
475 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
476 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
477 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
478 get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
479 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
482 * GstRTSPClient::handle-response:
483 * @client: a #GstRTSPClient
484 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
486 gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
487 g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
488 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
489 handle_response), NULL, NULL, g_cclosure_marshal_generic,
490 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
493 * GstRTSPClient::send-message:
494 * @client: The RTSP client
495 * @session: (type GstRtspServer.RTSPSession): The session
496 * @message: (type GstRtsp.RTSPMessage): The message
498 gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
499 g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
500 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
501 send_message), NULL, NULL, g_cclosure_marshal_generic,
502 G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
505 * GstRTSPClient::pre-announce-request:
506 * @client: a #GstRTSPClient
507 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
509 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
510 * otherwise an appropriate return code
514 gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST] =
515 g_signal_new ("pre-announce-request", G_TYPE_FROM_CLASS (klass),
516 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
517 pre_announce_request), pre_signal_accumulator, NULL,
518 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
519 GST_TYPE_RTSP_CONTEXT);
522 * GstRTSPClient::announce-request:
523 * @client: a #GstRTSPClient
524 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
526 gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST] =
527 g_signal_new ("announce-request", G_TYPE_FROM_CLASS (klass),
528 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, announce_request),
529 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
530 GST_TYPE_RTSP_CONTEXT);
533 * GstRTSPClient::pre-record-request:
534 * @client: a #GstRTSPClient
535 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
537 * Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
538 * otherwise an appropriate return code
542 gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST] =
543 g_signal_new ("pre-record-request", G_TYPE_FROM_CLASS (klass),
544 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
545 pre_record_request), pre_signal_accumulator, NULL,
546 g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
547 GST_TYPE_RTSP_CONTEXT);
550 * GstRTSPClient::record-request:
551 * @client: a #GstRTSPClient
552 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
554 gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST] =
555 g_signal_new ("record-request", G_TYPE_FROM_CLASS (klass),
556 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, record_request),
557 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
558 GST_TYPE_RTSP_CONTEXT);
561 * GstRTSPClient::check-requirements:
562 * @client: a #GstRTSPClient
563 * @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
564 * @arr: a NULL-terminated array of strings
566 * Returns: a newly allocated string with comma-separated list of
567 * unsupported options. An empty string must be returned if
568 * all options are supported.
572 gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS] =
573 g_signal_new ("check-requirements", G_TYPE_FROM_CLASS (klass),
574 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
575 check_requirements), NULL, NULL, g_cclosure_marshal_generic,
576 G_TYPE_STRING, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_STRV);
579 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
580 g_mutex_init (&tunnels_lock);
582 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
586 gst_rtsp_client_init (GstRTSPClient * client)
588 GstRTSPClientPrivate *priv = gst_rtsp_client_get_instance_private (client);
592 g_mutex_init (&priv->lock);
593 g_mutex_init (&priv->send_lock);
594 g_mutex_init (&priv->watch_lock);
596 priv->data_seqs = g_array_new (FALSE, FALSE, sizeof (DataSeq));
597 priv->drop_backlog = DEFAULT_DROP_BACKLOG;
599 g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
601 priv->pipelined_requests = g_hash_table_new_full (g_str_hash,
602 g_str_equal, g_free, g_free);
603 priv->tstate = TUNNEL_STATE_UNKNOWN;
606 static GstRTSPFilterResult
607 filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
610 gboolean *closed = user_data;
613 gboolean is_all_udp = TRUE;
615 media = gst_rtsp_session_media_get_media (sessmedia);
616 n_streams = gst_rtsp_media_n_streams (media);
618 for (i = 0; i < n_streams; i++) {
619 GstRTSPStreamTransport *transport =
620 gst_rtsp_session_media_get_transport (sessmedia, i);
621 const GstRTSPTransport *rtsp_transport;
626 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
628 && rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP
629 && rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP_MCAST) {
635 if (!is_all_udp || gst_rtsp_media_is_stop_on_disconnect (media)) {
636 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
637 return GST_RTSP_FILTER_REMOVE;
640 return GST_RTSP_FILTER_KEEP;
645 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
647 GstRTSPClientPrivate *priv = client->priv;
649 g_mutex_lock (&priv->lock);
650 /* check if we already know about this session */
651 if (g_list_find (priv->sessions, session) == NULL) {
652 GST_INFO ("watching session %p", session);
654 priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
655 priv->sessions_cookie++;
657 /* connect removed session handler, it will be disconnected when the last
658 * session gets removed */
659 if (priv->session_removed_id == 0)
660 priv->session_removed_id = g_signal_connect_data (priv->session_pool,
661 "session-removed", G_CALLBACK (client_session_removed),
662 g_object_ref (client), (GClosureNotify) g_object_unref, 0);
664 g_mutex_unlock (&priv->lock);
669 /* should be called with lock */
671 client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
674 GstRTSPClientPrivate *priv = client->priv;
676 GST_INFO ("client %p: unwatch session %p", client, session);
679 link = g_list_find (priv->sessions, session);
684 priv->sessions = g_list_delete_link (priv->sessions, link);
685 priv->sessions_cookie++;
687 /* if this was the last session, disconnect the handler.
688 * This will also drop the extra client ref */
689 if (!priv->sessions) {
690 g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
691 priv->session_removed_id = 0;
694 if (!priv->drop_backlog) {
695 /* unlink all media managed in this session */
696 gst_rtsp_session_filter (session, filter_session_media, client);
699 /* remove the session */
700 g_object_unref (session);
703 static GstRTSPFilterResult
704 cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
707 gboolean *closed = user_data;
708 GstRTSPClientPrivate *priv = client->priv;
710 if (priv->drop_backlog) {
711 /* unlink all media managed in this session. This needs to happen
712 * without the client lock, so we really want to do it here. */
713 gst_rtsp_session_filter (sess, filter_session_media, user_data);
717 return GST_RTSP_FILTER_REMOVE;
719 return GST_RTSP_FILTER_KEEP;
723 clean_cached_media (GstRTSPClient * client, gboolean unprepare)
725 GstRTSPClientPrivate *priv = client->priv;
733 gst_rtsp_media_unprepare (priv->media);
734 g_object_unref (priv->media);
739 /* A client is finalized when the connection is broken */
741 gst_rtsp_client_finalize (GObject * obj)
743 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
744 GstRTSPClientPrivate *priv = client->priv;
746 GST_INFO ("finalize client %p", client);
749 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
750 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
753 g_source_destroy ((GSource *) priv->watch);
755 if (priv->watch_context)
756 g_main_context_unref (priv->watch_context);
758 /* all sessions should have been removed by now. We keep a ref to
759 * the client object for the session removed handler. The ref is
760 * dropped when the last session is removed from the list. */
761 g_assert (priv->sessions == NULL);
762 g_assert (priv->session_removed_id == 0);
764 g_array_unref (priv->data_seqs);
765 g_hash_table_unref (priv->transports);
766 g_hash_table_unref (priv->pipelined_requests);
768 if (priv->connection)
769 gst_rtsp_connection_free (priv->connection);
770 if (priv->session_pool) {
771 g_object_unref (priv->session_pool);
773 if (priv->mount_points)
774 g_object_unref (priv->mount_points);
776 g_object_unref (priv->auth);
777 if (priv->thread_pool)
778 g_object_unref (priv->thread_pool);
780 clean_cached_media (client, TRUE);
782 g_free (priv->server_ip);
783 g_mutex_clear (&priv->lock);
784 g_mutex_clear (&priv->send_lock);
785 g_mutex_clear (&priv->watch_lock);
787 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
791 gst_rtsp_client_get_property (GObject * object, guint propid,
792 GValue * value, GParamSpec * pspec)
794 GstRTSPClient *client = GST_RTSP_CLIENT (object);
795 GstRTSPClientPrivate *priv = client->priv;
798 case PROP_SESSION_POOL:
799 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
801 case PROP_MOUNT_POINTS:
802 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
804 case PROP_DROP_BACKLOG:
805 g_value_set_boolean (value, priv->drop_backlog);
808 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
813 gst_rtsp_client_set_property (GObject * object, guint propid,
814 const GValue * value, GParamSpec * pspec)
816 GstRTSPClient *client = GST_RTSP_CLIENT (object);
817 GstRTSPClientPrivate *priv = client->priv;
820 case PROP_SESSION_POOL:
821 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
823 case PROP_MOUNT_POINTS:
824 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
826 case PROP_DROP_BACKLOG:
827 g_mutex_lock (&priv->lock);
828 priv->drop_backlog = g_value_get_boolean (value);
829 g_mutex_unlock (&priv->lock);
832 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
837 * gst_rtsp_client_new:
839 * Create a new #GstRTSPClient instance.
841 * Returns: (transfer full): a new #GstRTSPClient
844 gst_rtsp_client_new (void)
846 GstRTSPClient *result;
848 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
854 send_message (GstRTSPClient * client, GstRTSPContext * ctx,
855 GstRTSPMessage * message, gboolean close)
857 GstRTSPClientPrivate *priv = client->priv;
859 gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
860 "GStreamer RTSP server");
862 /* remove any previous header */
863 gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
865 /* add the new session header for new session ids */
867 gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
868 gst_rtsp_session_get_header (ctx->session));
871 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
872 gst_rtsp_message_dump (message);
876 gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
879 message->type_data.response.version =
880 ctx->request->type_data.request.version;
882 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
885 g_mutex_lock (&priv->send_lock);
887 priv->send_func (client, message, close, priv->send_data);
888 g_mutex_unlock (&priv->send_lock);
890 gst_rtsp_message_unset (message);
894 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
895 GstRTSPContext * ctx)
897 gst_rtsp_message_init_response (ctx->response, code,
898 gst_rtsp_status_as_text (code), ctx->request);
902 send_message (client, ctx, ctx->response, FALSE);
906 send_option_not_supported_response (GstRTSPClient * client,
907 GstRTSPContext * ctx, const gchar * unsupported_options)
909 GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
911 gst_rtsp_message_init_response (ctx->response, code,
912 gst_rtsp_status_as_text (code), ctx->request);
914 if (unsupported_options != NULL) {
915 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
916 unsupported_options);
921 send_message (client, ctx, ctx->response, FALSE);
925 paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
927 if (path1 == NULL || path2 == NULL)
930 if (strlen (path1) != len2)
933 if (strncmp (path1, path2, len2))
939 /* this function is called to initially find the media for the DESCRIBE request
940 * but is cached for when the same client (without breaking the connection) is
941 * doing a setup for the exact same url. */
942 static GstRTSPMedia *
943 find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
946 GstRTSPClientPrivate *priv = client->priv;
947 GstRTSPMediaFactory *factory;
951 /* find the longest matching factory for the uri first */
952 if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
956 ctx->factory = factory;
958 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
959 goto no_factory_access;
961 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
967 path_len = strlen (path);
969 if (!paths_are_equal (priv->path, path, path_len)) {
970 /* remove any previously cached values before we try to construct a new
972 clean_cached_media (client, TRUE);
974 /* prepare the media and add it to the pipeline */
975 if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
980 if (!(gst_rtsp_media_get_transport_mode (media) &
981 GST_RTSP_TRANSPORT_MODE_RECORD)) {
982 GstRTSPThread *thread;
984 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
985 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
989 /* prepare the media */
990 if (!gst_rtsp_media_prepare (media, thread))
994 /* now keep track of the uri and the media */
995 priv->path = g_strndup (path, path_len);
998 /* we have seen this path before, used cached media */
1001 GST_INFO ("reusing cached media %p for path %s", media, priv->path);
1004 g_object_unref (factory);
1005 ctx->factory = NULL;
1008 g_object_ref (media);
1015 GST_ERROR ("client %p: no factory for path %s", client, path);
1016 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1021 g_object_unref (factory);
1022 ctx->factory = NULL;
1023 GST_ERROR ("client %p: not authorized to see factory path %s", client,
1025 /* error reply is already sent */
1030 g_object_unref (factory);
1031 ctx->factory = NULL;
1032 GST_ERROR ("client %p: not authorized for factory path %s", client, path);
1033 /* error reply is already sent */
1038 GST_ERROR ("client %p: can't create media", client);
1039 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1040 g_object_unref (factory);
1041 ctx->factory = NULL;
1046 GST_ERROR ("client %p: can't create thread", client);
1047 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1048 g_object_unref (media);
1050 g_object_unref (factory);
1051 ctx->factory = NULL;
1056 GST_ERROR ("client %p: can't prepare media", client);
1057 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1058 g_object_unref (media);
1060 g_object_unref (factory);
1061 ctx->factory = NULL;
1066 static inline DataSeq *
1067 get_data_seq_element (GstRTSPClient * client, guint8 channel)
1069 GstRTSPClientPrivate *priv = client->priv;
1070 GArray *data_seqs = priv->data_seqs;
1073 while (i < data_seqs->len) {
1074 DataSeq *data_seq = &g_array_index (data_seqs, DataSeq, i);
1075 if (data_seq->channel == channel)
1084 add_data_seq (GstRTSPClient * client, guint8 channel)
1086 GstRTSPClientPrivate *priv = client->priv;
1087 DataSeq data_seq = {.channel = channel,.seq = 0 };
1089 if (get_data_seq_element (client, channel) == NULL)
1090 g_array_append_val (priv->data_seqs, data_seq);
1094 set_data_seq (GstRTSPClient * client, guint8 channel, guint seq)
1098 data_seq = get_data_seq_element (client, channel);
1099 g_assert_nonnull (data_seq);
1100 data_seq->seq = seq;
1104 get_data_seq (GstRTSPClient * client, guint8 channel)
1108 data_seq = get_data_seq_element (client, channel);
1109 g_assert_nonnull (data_seq);
1110 return data_seq->seq;
1114 get_data_channel (GstRTSPClient * client, guint seq, guint8 * channel)
1116 GstRTSPClientPrivate *priv = client->priv;
1117 GArray *data_seqs = priv->data_seqs;
1120 while (i < data_seqs->len) {
1121 DataSeq *data_seq = &g_array_index (data_seqs, DataSeq, i);
1122 if (data_seq->seq == seq) {
1123 *channel = data_seq->channel;
1133 do_close (gpointer user_data)
1135 GstRTSPClient *client = user_data;
1137 gst_rtsp_client_close (client);
1139 return G_SOURCE_REMOVE;
1143 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
1145 GstRTSPClientPrivate *priv = client->priv;
1146 GstRTSPMessage message = { 0 };
1147 gboolean ret = TRUE;
1148 GstMapInfo map_info;
1152 gst_rtsp_message_init_data (&message, channel);
1154 /* FIXME, need some sort of iovec RTSPMessage here */
1155 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
1158 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
1160 g_mutex_lock (&priv->send_lock);
1161 if (get_data_seq (client, channel) != 0) {
1162 GST_WARNING ("already a queued data message for channel %d", channel);
1163 g_mutex_unlock (&priv->send_lock);
1166 if (priv->send_func)
1167 ret = priv->send_func (client, &message, FALSE, priv->send_data);
1168 g_mutex_unlock (&priv->send_lock);
1170 gst_rtsp_message_steal_body (&message, &data, &usize);
1171 gst_buffer_unmap (buffer, &map_info);
1173 gst_rtsp_message_unset (&message);
1178 /* close in watch context */
1179 idle_src = g_idle_source_new ();
1180 g_source_set_callback (idle_src, do_close, client, NULL);
1181 g_source_attach (idle_src, priv->watch_context);
1182 g_source_unref (idle_src);
1189 * gst_rtsp_client_close:
1190 * @client: a #GstRTSPClient
1192 * Close the connection of @client and remove all media it was managing.
1197 gst_rtsp_client_close (GstRTSPClient * client)
1199 GstRTSPClientPrivate *priv = client->priv;
1200 const gchar *tunnelid;
1202 GST_DEBUG ("client %p: closing connection", client);
1204 if (priv->connection) {
1205 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
1206 g_mutex_lock (&tunnels_lock);
1207 /* remove from tunnelids */
1208 g_hash_table_remove (tunnels, tunnelid);
1209 g_mutex_unlock (&tunnels_lock);
1211 gst_rtsp_connection_close (priv->connection);
1214 /* connection is now closed, destroy the watch which will also cause the
1215 * closed signal to be emitted */
1217 GST_DEBUG ("client %p: destroying watch", client);
1218 g_source_destroy ((GSource *) priv->watch);
1220 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
1221 g_main_context_unref (priv->watch_context);
1222 priv->watch_context = NULL;
1227 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
1232 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
1234 path = g_strdup (uri->abspath);
1239 /* Default signal handler function for all "pre-command" signals, like
1240 * pre-options-request. It just returns the RTSP return code 200.
1241 * Subclasses can override this to get another default behaviour.
1243 static GstRTSPStatusCode
1244 default_pre_signal_handler (GstRTSPClient * client, GstRTSPContext * ctx)
1246 GST_LOG_OBJECT (client, "returning GST_RTSP_STS_OK");
1247 return GST_RTSP_STS_OK;
1250 /* The pre-signal accumulator function checks the return value of the signal
1251 * handlers. If any of them returns an RTSP status code that does not start
1252 * with 2 it will return FALSE, no more signal handlers will be called, and
1253 * this last RTSP status code will be the result of the signal emission.
1256 pre_signal_accumulator (GSignalInvocationHint * ihint, GValue * return_accu,
1257 const GValue * handler_return, gpointer data)
1259 GstRTSPStatusCode handler_value = g_value_get_enum (handler_return);
1260 GstRTSPStatusCode accumulated_value = g_value_get_enum (return_accu);
1262 if (handler_value < 200 || handler_value > 299) {
1263 GST_DEBUG ("handler_value : %d, returning FALSE", handler_value);
1264 g_value_set_enum (return_accu, handler_value);
1268 /* the accumulated value is initiated to 0 by GLib. if current handler value is
1269 * bigger then use that instead
1271 * FIXME: Should we prioritize the 2xx codes in a smarter way?
1272 * Like, "201 Created" > "250 Low On Storage Space" > "200 OK"?
1274 if (handler_value > accumulated_value)
1275 g_value_set_enum (return_accu, handler_value);
1281 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
1283 GstRTSPClientPrivate *priv = client->priv;
1284 GstRTSPClientClass *klass;
1285 GstRTSPSession *session;
1286 GstRTSPSessionMedia *sessmedia;
1287 GstRTSPStatusCode code;
1290 gboolean keep_session;
1291 GstRTSPStatusCode sig_result;
1296 session = ctx->session;
1301 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1302 path = klass->make_path_from_uri (client, ctx->uri);
1304 /* get a handle to the configuration of the media in the session */
1305 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1309 /* only aggregate control for now.. */
1310 if (path[matched] != '\0')
1315 ctx->sessmedia = sessmedia;
1317 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST],
1318 0, ctx, &sig_result);
1319 if (sig_result != GST_RTSP_STS_OK) {
1323 /* we emit the signal before closing the connection */
1324 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
1327 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
1329 /* unmanage the media in the session, returns false if all media session
1331 keep_session = gst_rtsp_session_release_media (session, sessmedia);
1333 /* construct the response now */
1334 code = GST_RTSP_STS_OK;
1335 gst_rtsp_message_init_response (ctx->response, code,
1336 gst_rtsp_status_as_text (code), ctx->request);
1338 send_message (client, ctx, ctx->response, TRUE);
1340 if (!keep_session) {
1341 /* remove the session */
1342 gst_rtsp_session_pool_remove (priv->session_pool, session);
1350 GST_ERROR ("client %p: no session", client);
1351 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1356 GST_ERROR ("client %p: no uri supplied", client);
1357 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1362 GST_ERROR ("client %p: no media for uri", client);
1363 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1369 GST_ERROR ("client %p: no aggregate path %s", client, path);
1370 send_generic_response (client,
1371 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1377 GST_ERROR ("client %p: pre signal returned error: %s", client,
1378 gst_rtsp_status_as_text (sig_result));
1379 send_generic_response (client, sig_result, ctx);
1384 static GstRTSPResult
1385 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
1389 res = gst_rtsp_params_set (client, ctx);
1394 static GstRTSPResult
1395 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
1399 res = gst_rtsp_params_get (client, ctx);
1405 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
1410 GstRTSPStatusCode sig_result;
1412 g_signal_emit (client,
1413 gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST], 0, ctx,
1415 if (sig_result != GST_RTSP_STS_OK) {
1419 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
1420 if (res != GST_RTSP_OK)
1423 if (size == 0 || !data || strlen ((char *) data) == 0) {
1424 if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
1425 GST_ERROR_OBJECT (client, "Using PLAY request for keep-alive is forbidden"
1430 /* no body (or only '\0'), keep-alive request */
1431 send_generic_response (client, GST_RTSP_STS_OK, ctx);
1433 /* there is a body, handle the params */
1434 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
1435 if (res != GST_RTSP_OK)
1438 send_message (client, ctx, ctx->response, FALSE);
1441 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
1449 GST_ERROR ("client %p: pre signal returned error: %s", client,
1450 gst_rtsp_status_as_text (sig_result));
1451 send_generic_response (client, sig_result, ctx);
1456 GST_ERROR ("client %p: bad request", client);
1457 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1463 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
1468 GstRTSPStatusCode sig_result;
1470 g_signal_emit (client,
1471 gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST], 0, ctx,
1473 if (sig_result != GST_RTSP_STS_OK) {
1477 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
1478 if (res != GST_RTSP_OK)
1481 if (size == 0 || !data || strlen ((char *) data) == 0) {
1482 /* no body (or only '\0'), keep-alive request */
1483 send_generic_response (client, GST_RTSP_STS_OK, ctx);
1485 /* there is a body, handle the params */
1486 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
1487 if (res != GST_RTSP_OK)
1490 send_message (client, ctx, ctx->response, FALSE);
1493 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
1501 GST_ERROR ("client %p: pre signal returned error: %s", client,
1502 gst_rtsp_status_as_text (sig_result));
1503 send_generic_response (client, sig_result, ctx);
1508 GST_ERROR ("client %p: bad request", client);
1509 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1515 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
1517 GstRTSPSession *session;
1518 GstRTSPClientClass *klass;
1519 GstRTSPSessionMedia *sessmedia;
1520 GstRTSPMedia *media;
1521 GstRTSPStatusCode code;
1522 GstRTSPState rtspstate;
1525 GstRTSPStatusCode sig_result;
1528 if (!(session = ctx->session))
1534 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1535 path = klass->make_path_from_uri (client, ctx->uri);
1537 /* get a handle to the configuration of the media in the session */
1538 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1542 if (path[matched] != '\0')
1547 media = gst_rtsp_session_media_get_media (sessmedia);
1548 n = gst_rtsp_media_n_streams (media);
1549 for (i = 0; i < n; i++) {
1550 GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
1552 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
1553 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET)
1557 ctx->sessmedia = sessmedia;
1559 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST], 0,
1561 if (sig_result != GST_RTSP_STS_OK) {
1565 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1566 /* the session state must be playing or recording */
1567 if (rtspstate != GST_RTSP_STATE_PLAYING &&
1568 rtspstate != GST_RTSP_STATE_RECORDING)
1571 /* then pause sending */
1572 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1574 /* construct the response now */
1575 code = GST_RTSP_STS_OK;
1576 gst_rtsp_message_init_response (ctx->response, code,
1577 gst_rtsp_status_as_text (code), ctx->request);
1579 send_message (client, ctx, ctx->response, FALSE);
1581 /* the state is now READY */
1582 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1584 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1591 GST_ERROR ("client %p: no session", client);
1592 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1597 GST_ERROR ("client %p: no uri supplied", client);
1598 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1603 GST_ERROR ("client %p: no media for uri", client);
1604 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1610 GST_ERROR ("client %p: no aggregate path %s", client, path);
1611 send_generic_response (client,
1612 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1618 GST_ERROR ("client %p: pre signal returned error: %s", client,
1619 gst_rtsp_status_as_text (sig_result));
1620 send_generic_response (client, sig_result, ctx);
1625 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1626 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1632 GST_ERROR ("client %p: pausing not supported", client);
1633 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1638 /* convert @url and @path to a URL used as a content base for the factory
1639 * located at @path */
1641 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1647 /* check for trailing '/' and append one */
1648 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1653 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1655 result = gst_rtsp_url_get_request_uri (&tmp);
1656 g_free (tmp.abspath);
1662 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1664 GstRTSPSession *session;
1665 GstRTSPClientClass *klass;
1666 GstRTSPSessionMedia *sessmedia;
1667 GstRTSPMedia *media;
1668 GstRTSPStatusCode code;
1671 GstRTSPTimeRange *range;
1673 GstRTSPState rtspstate;
1674 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1675 gchar *path, *rtpinfo;
1677 gchar *seek_style = NULL;
1678 GstRTSPStatusCode sig_result;
1679 GPtrArray *transports;
1681 if (!(session = ctx->session))
1684 if (!(uri = ctx->uri))
1687 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1688 path = klass->make_path_from_uri (client, uri);
1690 /* get a handle to the configuration of the media in the session */
1691 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1695 if (path[matched] != '\0')
1700 ctx->sessmedia = sessmedia;
1701 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1703 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST], 0,
1705 if (sig_result != GST_RTSP_STS_OK) {
1709 if (!(gst_rtsp_media_get_transport_mode (media) &
1710 GST_RTSP_TRANSPORT_MODE_PLAY))
1711 goto unsupported_mode;
1713 /* the session state must be playing or ready */
1714 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1715 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1718 /* update the pipeline */
1719 transports = gst_rtsp_session_media_get_transports (sessmedia);
1720 if (!gst_rtsp_media_complete_pipeline (media, transports)) {
1721 g_ptr_array_unref (transports);
1722 goto pipeline_error;
1724 g_ptr_array_unref (transports);
1726 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1727 if (!gst_rtsp_media_unsuspend (media))
1728 goto unsuspend_failed;
1730 /* parse the range header if we have one */
1731 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1732 if (res == GST_RTSP_OK) {
1733 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1734 GstRTSPMediaStatus media_status;
1735 GstSeekFlags flags = 0;
1737 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_SEEK_STYLE,
1739 if (g_strcmp0 (seek_style, "RAP") == 0)
1740 flags = GST_SEEK_FLAG_ACCURATE;
1741 else if (g_strcmp0 (seek_style, "CoRAP") == 0)
1742 flags = GST_SEEK_FLAG_KEY_UNIT;
1743 else if (g_strcmp0 (seek_style, "First-Prior") == 0)
1744 flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_BEFORE;
1745 else if (g_strcmp0 (seek_style, "Next") == 0)
1746 flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_AFTER;
1748 GST_FIXME_OBJECT (client, "Add support for seek style %s",
1752 /* we have a range, seek to the position */
1754 gst_rtsp_media_seek_full (media, range, flags);
1755 gst_rtsp_range_free (range);
1757 media_status = gst_rtsp_media_get_status (media);
1758 if (media_status == GST_RTSP_MEDIA_STATUS_ERROR)
1763 /* grab RTPInfo from the media now */
1764 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1766 /* construct the response now */
1767 code = GST_RTSP_STS_OK;
1768 gst_rtsp_message_init_response (ctx->response, code,
1769 gst_rtsp_status_as_text (code), ctx->request);
1771 /* add the RTP-Info header */
1773 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1776 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_SEEK_STYLE,
1780 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1782 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1784 send_message (client, ctx, ctx->response, FALSE);
1786 /* start playing after sending the response */
1787 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1789 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1791 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1798 GST_ERROR ("client %p: no session", client);
1799 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1804 GST_ERROR ("client %p: no uri supplied", client);
1805 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1810 GST_ERROR ("client %p: media not found", client);
1811 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1816 GST_ERROR ("client %p: no aggregate path %s", client, path);
1817 send_generic_response (client,
1818 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1824 GST_ERROR ("client %p: pre signal returned error: %s", client,
1825 gst_rtsp_status_as_text (sig_result));
1826 send_generic_response (client, sig_result, ctx);
1831 GST_ERROR ("client %p: not PLAYING or READY", client);
1832 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1838 GST_ERROR ("client %p: failed to configure the pipeline", client);
1839 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1845 GST_ERROR ("client %p: unsuspend failed", client);
1846 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1851 GST_ERROR ("client %p: seek failed", client);
1852 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1857 GST_ERROR ("client %p: media does not support PLAY", client);
1858 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
1864 do_keepalive (GstRTSPSession * session)
1866 GST_INFO ("keep session %p alive", session);
1867 gst_rtsp_session_touch (session);
1870 /* parse @transport and return a valid transport in @tr. only transports
1871 * supported by @stream are returned. Returns FALSE if no valid transport
1874 parse_transport (const char *transport, GstRTSPStream * stream,
1875 GstRTSPTransport * tr)
1882 gst_rtsp_transport_init (tr);
1884 GST_DEBUG ("parsing transports %s", transport);
1886 transports = g_strsplit (transport, ",", 0);
1888 /* loop through the transports, try to parse */
1889 for (i = 0; transports[i]; i++) {
1890 g_strstrip (transports[i]);
1891 res = gst_rtsp_transport_parse (transports[i], tr);
1892 if (res != GST_RTSP_OK) {
1893 /* no valid transport, search some more */
1894 GST_WARNING ("could not parse transport %s", transports[i]);
1898 /* we have a transport, see if it's supported */
1899 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1900 GST_WARNING ("unsupported transport %s", transports[i]);
1904 /* we have a valid transport */
1905 GST_INFO ("found valid transport %s", transports[i]);
1910 gst_rtsp_transport_init (tr);
1912 g_strfreev (transports);
1918 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1919 GstRTSPStream * stream, GstRTSPContext * ctx)
1921 GstRTSPMessage *request = ctx->request;
1922 gchar *blocksize_str;
1924 if (!gst_rtsp_stream_is_sender (stream))
1927 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1928 &blocksize_str, 0) == GST_RTSP_OK) {
1932 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1933 if (end == blocksize_str)
1936 /* we don't want to change the mtu when this media
1937 * can be shared because it impacts other clients */
1938 if (gst_rtsp_media_is_shared (media))
1941 if (blocksize > G_MAXUINT)
1942 blocksize = G_MAXUINT;
1944 gst_rtsp_stream_set_mtu (stream, blocksize);
1952 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1953 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1959 default_configure_client_transport (GstRTSPClient * client,
1960 GstRTSPContext * ctx, GstRTSPTransport * ct)
1962 GstRTSPClientPrivate *priv = client->priv;
1964 /* we have a valid transport now, set the destination of the client. */
1965 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST ||
1966 ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP) {
1967 /* allocate UDP ports */
1968 GSocketFamily family;
1969 gboolean use_client_settings = FALSE;
1971 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1973 if ((ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) &&
1974 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS) &&
1975 (ct->destination != NULL))
1976 use_client_settings = TRUE;
1978 /* We need to allocate the sockets for both families before starting
1979 * multiudpsink, otherwise multiudpsink won't accept new clients with
1980 * a different family.
1982 /* FIXME: could be more adequately solved by making it possible
1983 * to set a socket on multiudpsink after it has already been started */
1984 if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, G_SOCKET_FAMILY_IPV4, ct,
1985 use_client_settings) && family == G_SOCKET_FAMILY_IPV4)
1986 goto error_allocating_ports;
1988 if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, G_SOCKET_FAMILY_IPV6, ct,
1989 use_client_settings) && family == G_SOCKET_FAMILY_IPV6)
1990 goto error_allocating_ports;
1992 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1993 /* FIXME: the address has been successfully allocated, however, in
1994 * the use_client_settings case we need to verify that the allocated
1995 * address is the one requested by the client and if this address is
1996 * an allowed destination. Verifying this via the address pool in not
1997 * the proper way as the address pool should only be used for choosing
1998 * the server-selected address/port pairs. */
2000 if (!use_client_settings) {
2001 GstRTSPAddress *addr = NULL;
2003 g_free (ct->destination);
2004 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
2007 ct->destination = g_strdup (addr->address);
2008 ct->port.min = addr->port;
2009 ct->port.max = addr->port + addr->n_ports - 1;
2010 ct->ttl = addr->ttl;
2011 gst_rtsp_address_free (addr);
2016 url = gst_rtsp_connection_get_url (priv->connection);
2017 g_free (ct->destination);
2018 ct->destination = g_strdup (url->host);
2023 url = gst_rtsp_connection_get_url (priv->connection);
2024 g_free (ct->destination);
2025 ct->destination = g_strdup (url->host);
2027 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
2029 GSocketAddress *addr;
2031 sock = gst_rtsp_connection_get_read_socket (priv->connection);
2032 if ((addr = g_socket_get_remote_address (sock, NULL))) {
2033 /* our read port is the sender port of client */
2034 ct->client_port.min =
2035 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
2036 g_object_unref (addr);
2038 if ((addr = g_socket_get_local_address (sock, NULL))) {
2039 ct->server_port.max =
2040 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
2041 g_object_unref (addr);
2043 sock = gst_rtsp_connection_get_write_socket (priv->connection);
2044 if ((addr = g_socket_get_remote_address (sock, NULL))) {
2045 /* our write port is the receiver port of client */
2046 ct->client_port.max =
2047 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
2048 g_object_unref (addr);
2050 if ((addr = g_socket_get_local_address (sock, NULL))) {
2051 ct->server_port.min =
2052 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
2053 g_object_unref (addr);
2055 /* check if the client selected channels for TCP */
2056 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
2057 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
2065 error_allocating_ports:
2067 GST_ERROR_OBJECT (client, "Failed to allocate UDP ports");
2072 GST_ERROR_OBJECT (client, "Failed to acquire address for stream");
2077 static GstRTSPTransport *
2078 make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
2079 GstRTSPContext * ctx, GstRTSPTransport * ct)
2081 GstRTSPTransport *st;
2083 GSocketFamily family;
2085 /* prepare the server transport */
2086 gst_rtsp_transport_new (&st);
2088 st->trans = ct->trans;
2089 st->profile = ct->profile;
2090 st->lower_transport = ct->lower_transport;
2091 st->mode_play = ct->mode_play;
2092 st->mode_record = ct->mode_record;
2094 addr = g_inet_address_new_from_string (ct->destination);
2097 GST_ERROR ("failed to get inet addr from client destination");
2098 family = G_SOCKET_FAMILY_IPV4;
2100 family = g_inet_address_get_family (addr);
2101 g_object_unref (addr);
2105 switch (st->lower_transport) {
2106 case GST_RTSP_LOWER_TRANS_UDP:
2107 st->client_port = ct->client_port;
2108 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
2110 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2111 st->port = ct->port;
2112 st->destination = g_strdup (ct->destination);
2115 case GST_RTSP_LOWER_TRANS_TCP:
2116 st->interleaved = ct->interleaved;
2117 st->client_port = ct->client_port;
2118 st->server_port = ct->server_port;
2123 if ((gst_rtsp_media_get_transport_mode (media) &
2124 GST_RTSP_TRANSPORT_MODE_PLAY))
2125 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
2131 rtsp_ctrl_timeout_cb (gpointer user_data)
2133 gboolean res = G_SOURCE_CONTINUE;
2134 GstRTSPClient *client = (GstRTSPClient *) user_data;
2135 GstRTSPClientPrivate *priv = client->priv;
2137 priv->rtsp_ctrl_timeout_cnt += RTSP_CTRL_CB_INTERVAL;
2139 if (priv->rtsp_ctrl_timeout_cnt > RTSP_CTRL_TIMEOUT_VALUE) {
2140 GST_DEBUG ("rtsp control session timeout id=%u expired, closing client.",
2141 priv->rtsp_ctrl_timeout_id);
2142 g_mutex_lock (&priv->lock);
2143 priv->rtsp_ctrl_timeout_id = 0;
2144 priv->rtsp_ctrl_timeout_cnt = 0;
2145 g_mutex_unlock (&priv->lock);
2146 gst_rtsp_client_close (client);
2148 res = G_SOURCE_REMOVE;
2155 rtsp_ctrl_timeout_remove (GstRTSPClientPrivate * priv)
2157 g_mutex_lock (&priv->lock);
2159 if (priv->rtsp_ctrl_timeout_id != 0) {
2160 g_source_destroy (g_main_context_find_source_by_id (priv->watch_context,
2161 priv->rtsp_ctrl_timeout_id));
2162 GST_DEBUG ("rtsp control session removed timeout id=%u.",
2163 priv->rtsp_ctrl_timeout_id);
2164 priv->rtsp_ctrl_timeout_id = 0;
2165 priv->rtsp_ctrl_timeout_cnt = 0;
2168 g_mutex_unlock (&priv->lock);
2172 stream_make_keymgmt (GstRTSPClient * client, const gchar * location,
2173 GstRTSPStream * stream)
2175 gchar *base64, *result = NULL;
2176 GstMIKEYMessage *mikey_msg;
2177 GstCaps *srtcpparams;
2178 GstElement *rtcp_encoder;
2179 gint srtcp_cipher, srtp_cipher;
2180 gint srtcp_auth, srtp_auth;
2182 GType ciphertype, authtype;
2183 GEnumClass *cipher_enum, *auth_enum;
2184 GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value,
2187 rtcp_encoder = gst_rtsp_stream_get_srtp_encoder (stream);
2192 ciphertype = g_type_from_name ("GstSrtpCipherType");
2193 authtype = g_type_from_name ("GstSrtpAuthType");
2195 cipher_enum = g_type_class_ref (ciphertype);
2196 auth_enum = g_type_class_ref (authtype);
2198 /* We need to bring the encoder to READY so that it generates its key */
2199 gst_element_set_state (rtcp_encoder, GST_STATE_READY);
2201 g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth",
2202 &srtcp_auth, "rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key",
2204 g_object_unref (rtcp_encoder);
2206 srtcp_cipher_value = g_enum_get_value (cipher_enum, srtcp_cipher);
2207 srtp_cipher_value = g_enum_get_value (cipher_enum, srtp_cipher);
2208 srtcp_auth_value = g_enum_get_value (auth_enum, srtcp_auth);
2209 srtp_auth_value = g_enum_get_value (auth_enum, srtp_auth);
2211 g_type_class_unref (cipher_enum);
2212 g_type_class_unref (auth_enum);
2214 srtcpparams = gst_caps_new_simple ("application/x-srtcp",
2215 "srtcp-cipher", G_TYPE_STRING, srtcp_cipher_value->value_nick,
2216 "srtcp-auth", G_TYPE_STRING, srtcp_auth_value->value_nick,
2217 "srtp-cipher", G_TYPE_STRING, srtp_cipher_value->value_nick,
2218 "srtp-auth", G_TYPE_STRING, srtp_auth_value->value_nick,
2219 "srtp-key", GST_TYPE_BUFFER, key, NULL);
2221 mikey_msg = gst_mikey_message_new_from_caps (srtcpparams);
2225 gst_rtsp_stream_get_ssrc (stream, &send_ssrc);
2226 gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0);
2228 base64 = gst_mikey_message_base64_encode (mikey_msg);
2229 gst_mikey_message_unref (mikey_msg);
2232 result = gst_sdp_make_keymgmt (location, base64);
2242 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
2244 GstRTSPClientPrivate *priv = client->priv;
2247 gchar *transport, *keymgmt;
2248 GstRTSPTransport *ct, *st;
2249 GstRTSPStatusCode code;
2250 GstRTSPSession *session;
2251 GstRTSPStreamTransport *trans;
2253 GstRTSPSessionMedia *sessmedia;
2254 GstRTSPMedia *media;
2255 GstRTSPStream *stream;
2256 GstRTSPState rtspstate;
2257 GstRTSPClientClass *klass;
2258 gchar *path, *control = NULL;
2260 gboolean new_session = FALSE;
2261 GstRTSPStatusCode sig_result;
2262 gchar *pipelined_request_id = NULL, *accept_range = NULL;
2268 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2269 path = klass->make_path_from_uri (client, uri);
2271 /* parse the transport */
2273 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
2275 if (res != GST_RTSP_OK)
2278 /* Handle Pipelined-requests if using >= 2.0 */
2279 if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0)
2280 gst_rtsp_message_get_header (ctx->request,
2281 GST_RTSP_HDR_PIPELINED_REQUESTS, &pipelined_request_id, 0);
2283 /* we create the session after parsing stuff so that we don't make
2284 * a session for malformed requests */
2285 if (priv->session_pool == NULL)
2288 session = ctx->session;
2291 g_object_ref (session);
2292 /* get a handle to the configuration of the media in the session, this can
2293 * return NULL if this is a new url to manage in this session. */
2294 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
2296 /* we need a new media configuration in this session */
2300 /* we have no session media, find one and manage it */
2301 if (sessmedia == NULL) {
2302 /* get a handle to the configuration of the media in the session */
2303 media = find_media (client, ctx, path, &matched);
2304 /* need to suspend the media, if the protocol has changed */
2306 gst_rtsp_media_suspend (media);
2308 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
2309 g_object_ref (media);
2311 goto media_not_found;
2313 /* no media, not found then */
2315 goto media_not_found_no_reply;
2317 if (path[matched] == '\0') {
2318 if (gst_rtsp_media_n_streams (media) == 1) {
2319 stream = gst_rtsp_media_get_stream (media, 0);
2321 goto control_not_found;
2324 /* path is what matched. */
2325 path[matched] = '\0';
2326 /* control is remainder */
2327 control = &path[matched + 1];
2329 /* find the stream now using the control part */
2330 stream = gst_rtsp_media_find_stream (media, control);
2334 goto stream_not_found;
2336 /* now we have a uri identifying a valid media and stream */
2337 ctx->stream = stream;
2340 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST], 0,
2342 if (sig_result != GST_RTSP_STS_OK) {
2346 if (session == NULL) {
2347 /* create a session if this fails we probably reached our session limit or
2349 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
2350 goto service_unavailable;
2352 /* Pipelined requests should be cleared between sessions */
2353 g_hash_table_remove_all (priv->pipelined_requests);
2355 /* make sure this client is closed when the session is closed */
2356 client_watch_session (client, session);
2359 /* signal new session */
2360 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
2363 ctx->session = session;
2366 if (pipelined_request_id) {
2367 g_hash_table_insert (client->priv->pipelined_requests,
2368 g_strdup (pipelined_request_id),
2369 g_strdup (gst_rtsp_session_get_sessionid (session)));
2371 rtsp_ctrl_timeout_remove (priv);
2373 if (!klass->configure_client_media (client, media, stream, ctx))
2374 goto configure_media_failed_no_reply;
2376 gst_rtsp_transport_new (&ct);
2378 /* parse and find a usable supported transport */
2379 if (!parse_transport (transport, stream, ct))
2380 goto unsupported_transports;
2383 && !(gst_rtsp_media_get_transport_mode (media) &
2384 GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
2385 && !(gst_rtsp_media_get_transport_mode (media) &
2386 GST_RTSP_TRANSPORT_MODE_RECORD)))
2387 goto unsupported_mode;
2389 /* parse the keymgmt */
2390 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
2391 &keymgmt, 0) == GST_RTSP_OK) {
2392 if (!gst_rtsp_stream_handle_keymgmt (ctx->stream, keymgmt))
2396 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
2397 &accept_range, 0) == GST_RTSP_OK) {
2398 GEnumValue *runit = NULL;
2400 gchar **valid_ranges;
2401 GEnumClass *runit_class = g_type_class_ref (GST_TYPE_RTSP_RANGE_UNIT);
2403 gst_rtsp_message_dump (ctx->request);
2404 valid_ranges = g_strsplit (accept_range, ",", -1);
2406 for (i = 0; valid_ranges[i]; i++) {
2407 gchar *range = valid_ranges[i];
2409 while (*range == ' ')
2412 runit = g_enum_get_value_by_nick (runit_class, range);
2416 g_strfreev (valid_ranges);
2417 g_type_class_unref (runit_class);
2420 goto unsupported_range_unit;
2423 if (sessmedia == NULL) {
2424 /* manage the media in our session now, if not done already */
2426 gst_rtsp_session_manage_media (session, path, g_object_ref (media));
2427 /* if we stil have no media, error */
2428 if (sessmedia == NULL)
2429 goto sessmedia_unavailable;
2431 /* don't cache media anymore */
2432 clean_cached_media (client, FALSE);
2435 ctx->sessmedia = sessmedia;
2437 /* update the client transport */
2438 if (!klass->configure_client_transport (client, ctx, ct))
2439 goto unsupported_client_transport;
2441 /* set in the session media transport */
2442 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
2446 /* configure the url used to set this transport, this we will use when
2447 * generating the response for the PLAY request */
2448 gst_rtsp_stream_transport_set_url (trans, uri);
2449 /* configure keepalive for this transport */
2450 gst_rtsp_stream_transport_set_keepalive (trans,
2451 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
2453 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
2454 /* our callbacks to send data on this TCP connection */
2455 gst_rtsp_stream_transport_set_callbacks (trans,
2456 (GstRTSPSendFunc) do_send_data,
2457 (GstRTSPSendFunc) do_send_data, client, NULL);
2459 g_hash_table_insert (priv->transports,
2460 GINT_TO_POINTER (ct->interleaved.min), trans);
2461 g_object_ref (trans);
2462 g_hash_table_insert (priv->transports,
2463 GINT_TO_POINTER (ct->interleaved.max), trans);
2464 g_object_ref (trans);
2465 add_data_seq (client, ct->interleaved.min);
2466 add_data_seq (client, ct->interleaved.max);
2469 /* create and serialize the server transport */
2470 st = make_server_transport (client, media, ctx, ct);
2471 trans_str = gst_rtsp_transport_as_text (st);
2472 gst_rtsp_transport_free (st);
2474 /* construct the response now */
2475 code = GST_RTSP_STS_OK;
2476 gst_rtsp_message_init_response (ctx->response, code,
2477 gst_rtsp_status_as_text (code), ctx->request);
2479 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
2483 if (pipelined_request_id)
2484 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PIPELINED_REQUESTS,
2485 pipelined_request_id);
2487 if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
2488 GstClockTimeDiff seekable = gst_rtsp_media_seekable (media);
2489 GString *media_properties = g_string_new (NULL);
2492 g_string_append (media_properties,
2493 "No-Seeking,Time-Progressing,Time-Duration=0.0");
2494 else if (seekable == 0)
2495 g_string_append (media_properties, "Beginning-Only");
2496 else if (seekable == G_MAXINT64)
2497 g_string_append (media_properties, "Random-Access");
2499 g_string_append_printf (media_properties,
2500 "Random-Access=%f, Unlimited, Immutable",
2501 (gdouble) seekable / GST_SECOND);
2503 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_MEDIA_PROPERTIES,
2504 g_string_free (media_properties, FALSE));
2505 /* TODO Check how Accept-Ranges should be filled */
2506 gst_rtsp_message_add_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
2507 "npt, clock, smpte, clock");
2510 send_message (client, ctx, ctx->response, FALSE);
2512 /* update the state */
2513 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
2514 switch (rtspstate) {
2515 case GST_RTSP_STATE_PLAYING:
2516 case GST_RTSP_STATE_RECORDING:
2517 case GST_RTSP_STATE_READY:
2518 /* no state change */
2521 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
2524 g_object_unref (media);
2525 g_object_unref (session);
2528 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
2535 GST_ERROR ("client %p: no uri", client);
2536 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2541 GST_ERROR ("client %p: no transport", client);
2542 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2547 GST_ERROR ("client %p: no session pool configured", client);
2548 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2551 media_not_found_no_reply:
2553 GST_ERROR ("client %p: media '%s' not found", client, path);
2554 /* error reply is already sent */
2555 goto cleanup_session;
2559 GST_ERROR ("client %p: media '%s' not found", client, path);
2560 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2561 goto cleanup_session;
2565 GST_ERROR ("client %p: no control in path '%s'", client, path);
2566 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2567 g_object_unref (media);
2568 goto cleanup_session;
2572 GST_ERROR ("client %p: stream '%s' not found", client,
2573 GST_STR_NULL (control));
2574 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2575 g_object_unref (media);
2576 goto cleanup_session;
2580 GST_ERROR ("client %p: pre signal returned error: %s", client,
2581 gst_rtsp_status_as_text (sig_result));
2582 send_generic_response (client, sig_result, ctx);
2583 g_object_unref (media);
2586 service_unavailable:
2588 GST_ERROR ("client %p: can't create session", client);
2589 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2590 g_object_unref (media);
2591 goto cleanup_session;
2593 sessmedia_unavailable:
2595 GST_ERROR ("client %p: can't create session media", client);
2596 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2597 goto cleanup_transport;
2599 configure_media_failed_no_reply:
2601 GST_ERROR ("client %p: configure_media failed", client);
2602 g_object_unref (media);
2603 /* error reply is already sent */
2604 goto cleanup_session;
2606 unsupported_transports:
2608 GST_ERROR ("client %p: unsupported transports", client);
2609 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2610 goto cleanup_transport;
2612 unsupported_client_transport:
2614 GST_ERROR ("client %p: unsupported client transport", client);
2615 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2616 goto cleanup_transport;
2620 GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
2621 "mode play: %d, mode record: %d)", client,
2622 ! !(gst_rtsp_media_get_transport_mode (media) &
2623 GST_RTSP_TRANSPORT_MODE_PLAY),
2624 ! !(gst_rtsp_media_get_transport_mode (media) &
2625 GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
2626 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
2627 goto cleanup_transport;
2629 unsupported_range_unit:
2631 GST_ERROR ("Client %p: does not support any range format we support",
2633 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2634 goto cleanup_transport;
2638 GST_ERROR ("client %p: keymgmt error", client);
2639 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
2640 goto cleanup_transport;
2644 gst_rtsp_transport_free (ct);
2646 g_object_unref (media);
2649 gst_rtsp_session_pool_remove (priv->session_pool, session);
2651 g_object_unref (session);
2658 static GstSDPMessage *
2659 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
2661 GstRTSPClientPrivate *priv = client->priv;
2665 guint64 session_id_tmp;
2666 gchar session_id[21];
2668 gst_sdp_message_new (&sdp);
2670 /* some standard things first */
2671 gst_sdp_message_set_version (sdp, "0");
2678 session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
2679 g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
2682 gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
2685 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2686 gst_sdp_message_set_information (sdp, "rtsp-server");
2687 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2688 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2689 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2690 gst_sdp_message_add_attribute (sdp, "control", "*");
2692 info.is_ipv6 = priv->is_ipv6;
2693 info.server_ip = priv->server_ip;
2695 /* create an SDP for the media object */
2696 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2704 GST_ERROR ("client %p: could not create SDP", client);
2705 gst_sdp_message_free (sdp);
2710 /* for the describe we must generate an SDP */
2712 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2714 GstRTSPClientPrivate *priv = client->priv;
2719 GstRTSPMedia *media;
2720 GstRTSPClientClass *klass;
2721 GstRTSPStatusCode sig_result;
2723 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2728 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST],
2729 0, ctx, &sig_result);
2730 if (sig_result != GST_RTSP_STS_OK) {
2734 /* check what kind of format is accepted, we don't really do anything with it
2735 * and always return SDP for now. */
2740 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2742 if (res == GST_RTSP_ENOTIMPL)
2745 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2749 if (!priv->mount_points)
2750 goto no_mount_points;
2752 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2755 /* find the media object for the uri */
2756 if (!(media = find_media (client, ctx, path, NULL)))
2759 if (!(gst_rtsp_media_get_transport_mode (media) &
2760 GST_RTSP_TRANSPORT_MODE_PLAY))
2761 goto unsupported_mode;
2763 /* create an SDP for the media object on this client */
2764 if (!(sdp = klass->create_sdp (client, media)))
2767 /* we suspend after the describe */
2768 gst_rtsp_media_suspend (media);
2769 g_object_unref (media);
2771 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2772 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2774 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2777 /* content base for some clients that might screw up creating the setup uri */
2778 str = make_base_url (client, ctx->uri, path);
2781 GST_INFO ("adding content-base: %s", str);
2782 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2784 /* add SDP to the response body */
2785 str = gst_sdp_message_as_text (sdp);
2786 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2787 gst_sdp_message_free (sdp);
2789 send_message (client, ctx, ctx->response, FALSE);
2791 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2799 GST_ERROR ("client %p: pre signal returned error: %s", client,
2800 gst_rtsp_status_as_text (sig_result));
2801 send_generic_response (client, sig_result, ctx);
2806 GST_ERROR ("client %p: no uri", client);
2807 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2812 GST_ERROR ("client %p: no mount points configured", client);
2813 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2818 GST_ERROR ("client %p: can't find path for url", client);
2819 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2824 GST_ERROR ("client %p: no media", client);
2826 /* error reply is already sent */
2831 GST_ERROR ("client %p: media does not support DESCRIBE", client);
2832 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
2834 g_object_unref (media);
2839 GST_ERROR ("client %p: can't create SDP", client);
2840 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2842 g_object_unref (media);
2848 handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
2849 GstSDPMessage * sdp)
2851 GstRTSPClientPrivate *priv = client->priv;
2852 GstRTSPThread *thread;
2854 /* create an SDP for the media object */
2855 if (!gst_rtsp_media_handle_sdp (media, sdp))
2858 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
2859 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
2863 /* prepare the media */
2864 if (!gst_rtsp_media_prepare (media, thread))
2872 GST_ERROR ("client %p: could not handle SDP", client);
2877 GST_ERROR ("client %p: can't create thread", client);
2882 GST_ERROR ("client %p: can't prepare media", client);
2888 handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
2890 GstRTSPClientPrivate *priv = client->priv;
2891 GstRTSPClientClass *klass;
2894 GstRTSPMedia *media;
2895 gchar *path, *cont = NULL;
2898 GstRTSPStatusCode sig_result;
2901 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2906 if (!priv->mount_points)
2907 goto no_mount_points;
2909 /* check if reply is SDP */
2910 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
2912 /* could not be set but since the request returned OK, we assume it
2913 * was SDP, else check it. */
2915 if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
2916 goto wrong_content_type;
2919 /* get message body and parse as SDP */
2920 gst_rtsp_message_get_body (ctx->request, &data, &size);
2921 if (data == NULL || size == 0)
2924 GST_DEBUG ("client %p: parse SDP...", client);
2925 gst_sdp_message_new (&sdp);
2926 sres = gst_sdp_message_parse_buffer (data, size, sdp);
2927 if (sres != GST_SDP_OK)
2928 goto sdp_parse_failed;
2930 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2933 /* find the media object for the uri */
2934 if (!(media = find_media (client, ctx, path, NULL)))
2939 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST],
2940 0, ctx, &sig_result);
2941 if (sig_result != GST_RTSP_STS_OK) {
2945 if (!(gst_rtsp_media_get_transport_mode (media) &
2946 GST_RTSP_TRANSPORT_MODE_RECORD))
2947 goto unsupported_mode;
2949 /* Tell client subclass about the media */
2950 if (!klass->handle_sdp (client, ctx, media, sdp))
2953 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2954 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2956 n_streams = gst_rtsp_media_n_streams (media);
2957 for (i = 0; i < n_streams; i++) {
2958 GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
2960 g_strdup_printf ("rtsp://%s%s:8554/stream=%d", priv->server_ip, path,
2962 gchar *keymgmt = stream_make_keymgmt (client, location, stream);
2965 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT,
2971 /* we suspend after the announce */
2972 gst_rtsp_media_suspend (media);
2973 g_object_unref (media);
2975 send_message (client, ctx, ctx->response, FALSE);
2977 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
2980 gst_sdp_message_free (sdp);
2986 GST_ERROR ("client %p: no uri", client);
2987 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2992 GST_ERROR ("client %p: no mount points configured", client);
2993 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2998 GST_ERROR ("client %p: can't find path for url", client);
2999 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
3000 gst_sdp_message_free (sdp);
3005 GST_ERROR ("client %p: unknown content type", client);
3006 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
3011 GST_ERROR ("client %p: can't find SDP message", client);
3012 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
3017 GST_ERROR ("client %p: failed to parse SDP message", client);
3018 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
3019 gst_sdp_message_free (sdp);
3024 GST_ERROR ("client %p: no media", client);
3026 /* error reply is already sent */
3027 gst_sdp_message_free (sdp);
3032 GST_ERROR ("client %p: pre signal returned error: %s", client,
3033 gst_rtsp_status_as_text (sig_result));
3034 send_generic_response (client, sig_result, ctx);
3035 gst_sdp_message_free (sdp);
3040 GST_ERROR ("client %p: media does not support ANNOUNCE", client);
3041 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
3043 g_object_unref (media);
3044 gst_sdp_message_free (sdp);
3049 GST_ERROR ("client %p: can't handle SDP", client);
3050 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
3052 g_object_unref (media);
3053 gst_sdp_message_free (sdp);
3059 handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
3061 GstRTSPSession *session;
3062 GstRTSPClientClass *klass;
3063 GstRTSPSessionMedia *sessmedia;
3064 GstRTSPMedia *media;
3066 GstRTSPState rtspstate;
3069 GstRTSPStatusCode sig_result;
3070 GPtrArray *transports;
3072 if (!(session = ctx->session))
3075 if (!(uri = ctx->uri))
3078 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3079 path = klass->make_path_from_uri (client, uri);
3081 /* get a handle to the configuration of the media in the session */
3082 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
3086 if (path[matched] != '\0')
3091 ctx->sessmedia = sessmedia;
3092 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
3094 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST], 0,
3096 if (sig_result != GST_RTSP_STS_OK) {
3100 if (!(gst_rtsp_media_get_transport_mode (media) &
3101 GST_RTSP_TRANSPORT_MODE_RECORD))
3102 goto unsupported_mode;
3104 /* the session state must be playing or ready */
3105 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
3106 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
3109 /* update the pipeline */
3110 transports = gst_rtsp_session_media_get_transports (sessmedia);
3111 if (!gst_rtsp_media_complete_pipeline (media, transports)) {
3112 g_ptr_array_unref (transports);
3113 goto pipeline_error;
3115 g_ptr_array_unref (transports);
3117 /* in record we first unsuspend, media could be suspended from SDP or PAUSED */
3118 if (!gst_rtsp_media_unsuspend (media))
3119 goto unsuspend_failed;
3121 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
3122 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
3124 send_message (client, ctx, ctx->response, FALSE);
3126 /* start playing after sending the response */
3127 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
3129 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
3131 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
3139 GST_ERROR ("client %p: no session", client);
3140 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
3145 GST_ERROR ("client %p: no uri supplied", client);
3146 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
3151 GST_ERROR ("client %p: media not found", client);
3152 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
3157 GST_ERROR ("client %p: no aggregate path %s", client, path);
3158 send_generic_response (client,
3159 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
3165 GST_ERROR ("client %p: pre signal returned error: %s", client,
3166 gst_rtsp_status_as_text (sig_result));
3167 send_generic_response (client, sig_result, ctx);
3172 GST_ERROR ("client %p: media does not support RECORD", client);
3173 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
3178 GST_ERROR ("client %p: not PLAYING or READY", client);
3179 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
3185 GST_ERROR ("client %p: failed to configure the pipeline", client);
3186 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
3192 GST_ERROR ("client %p: unsuspend failed", client);
3193 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
3199 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx,
3200 GstRTSPVersion version)
3202 GstRTSPMethod options;
3204 GstRTSPStatusCode sig_result;
3206 options = GST_RTSP_DESCRIBE |
3211 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
3213 if (version < GST_RTSP_VERSION_2_0) {
3214 options |= GST_RTSP_RECORD;
3215 options |= GST_RTSP_ANNOUNCE;
3218 str = gst_rtsp_options_as_text (options);
3220 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
3221 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
3223 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
3226 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST], 0,
3228 if (sig_result != GST_RTSP_STS_OK) {
3232 send_message (client, ctx, ctx->response, FALSE);
3234 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
3242 GST_ERROR ("client %p: pre signal returned error: %s", client,
3243 gst_rtsp_status_as_text (sig_result));
3244 send_generic_response (client, sig_result, ctx);
3245 gst_rtsp_message_free (ctx->response);
3250 /* remove duplicate and trailing '/' */
3252 sanitize_uri (GstRTSPUrl * uri)
3256 gboolean have_slash, prev_slash;
3258 s = d = uri->abspath;
3259 len = strlen (uri->abspath);
3263 for (i = 0; i < len; i++) {
3264 have_slash = s[i] == '/';
3266 if (!have_slash || !prev_slash)
3268 prev_slash = have_slash;
3270 len = d - uri->abspath;
3271 /* don't remove the first slash if that's the only thing left */
3272 if (len > 1 && *(d - 1) == '/')
3277 /* is called when the session is removed from its session pool. */
3279 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
3280 GstRTSPClient * client)
3282 GstRTSPClientPrivate *priv = client->priv;
3284 GST_INFO ("client %p: session %p removed", client, session);
3286 g_mutex_lock (&priv->lock);
3287 client_unwatch_session (client, session, NULL);
3288 g_mutex_unlock (&priv->lock);
3291 /* Check for Require headers. Returns TRUE if there are no Require headers,
3292 * otherwise lets the application decide which headers are supported.
3293 * By default all headers are unsupported.
3294 * If there are unsupported options, FALSE will be returned together with
3295 * a newly-allocated string of (comma-separated) unsupported options in
3296 * the unsupported_reqs variable.
3298 * There may be multiple Require headers, but we must send one single
3299 * Unsupported header with all the unsupported options as response. If
3300 * an incoming Require header contained a comma-separated list of options
3301 * GstRtspConnection will already have split that list up into multiple
3305 check_request_requirements (GstRTSPContext * ctx, gchar ** unsupported_reqs)
3308 GPtrArray *arr = NULL;
3309 GstRTSPMessage *msg = ctx->request;
3312 gchar *sig_result = NULL;
3313 gboolean result = TRUE;
3317 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
3319 if (res == GST_RTSP_ENOTIMPL)
3323 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
3325 g_ptr_array_add (arr, g_strdup (reqs));
3329 /* if we don't have any Require headers at all, all is fine */
3333 /* otherwise we've now processed at all the Require headers */
3334 g_ptr_array_add (arr, NULL);
3336 g_signal_emit (ctx->client,
3337 gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS], 0, ctx,
3338 (gchar **) arr->pdata, &sig_result);
3340 if (sig_result == NULL) {
3341 /* no supported options, just report all of the required ones as
3343 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
3348 if (strlen (sig_result) == 0)
3349 g_free (sig_result);
3351 *unsupported_reqs = sig_result;
3356 g_ptr_array_unref (arr);
3361 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
3363 GstRTSPClientPrivate *priv = client->priv;
3364 GstRTSPMethod method;
3365 const gchar *uristr;
3366 GstRTSPUrl *uri = NULL;
3367 GstRTSPVersion version;
3369 GstRTSPSession *session = NULL;
3370 GstRTSPContext sctx = { NULL }, *ctx;
3371 GstRTSPMessage response = { 0 };
3372 gchar *unsupported_reqs = NULL;
3373 gchar *sessid = NULL, *pipelined_request_id = NULL;
3375 if (!(ctx = gst_rtsp_context_get_current ())) {
3377 ctx->auth = priv->auth;
3378 gst_rtsp_context_push_current (ctx);
3381 ctx->conn = priv->connection;
3382 ctx->client = client;
3383 ctx->request = request;
3384 ctx->response = &response;
3386 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
3387 gst_rtsp_message_dump (request);
3390 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
3392 GST_INFO ("client %p: received a request %s %s %s", client,
3393 gst_rtsp_method_as_text (method), uristr,
3394 gst_rtsp_version_as_text (version));
3396 /* we can only handle 1.0 requests */
3397 if (version != GST_RTSP_VERSION_1_0 && version != GST_RTSP_VERSION_2_0)
3400 ctx->method = method;
3402 /* we always try to parse the url first */
3403 if (strcmp (uristr, "*") == 0) {
3404 /* special case where we have * as uri, keep uri = NULL */
3405 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
3406 /* check if the uristr is an absolute path <=> scheme and host information
3410 scheme = g_uri_parse_scheme (uristr);
3411 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
3412 gchar *absolute_uristr = NULL;
3414 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
3415 if (priv->server_ip == NULL) {
3416 GST_WARNING_OBJECT (client, "host information missing");
3421 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
3423 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
3424 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
3425 g_free (absolute_uristr);
3428 g_free (absolute_uristr);
3435 /* get the session if there is any */
3436 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_PIPELINED_REQUESTS,
3437 &pipelined_request_id, 0);
3438 if (res == GST_RTSP_OK) {
3439 sessid = g_hash_table_lookup (client->priv->pipelined_requests,
3440 pipelined_request_id);
3443 res = GST_RTSP_ERROR;
3446 if (res != GST_RTSP_OK)
3448 gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
3450 if (res == GST_RTSP_OK) {
3451 if (priv->session_pool == NULL)
3454 /* we had a session in the request, find it again */
3455 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
3456 goto session_not_found;
3458 /* we add the session to the client list of watched sessions. When a session
3459 * disappears because it times out, we will be notified. If all sessions are
3460 * gone, we will close the connection */
3461 client_watch_session (client, session);
3464 /* sanitize the uri */
3468 ctx->session = session;
3470 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
3471 goto not_authorized;
3473 /* handle any 'Require' headers */
3474 if (!check_request_requirements (ctx, &unsupported_reqs))
3475 goto unsupported_requirement;
3477 /* now see what is asked and dispatch to a dedicated handler */
3479 case GST_RTSP_OPTIONS:
3480 priv->version = version;
3481 handle_options_request (client, ctx, version);
3483 case GST_RTSP_DESCRIBE:
3484 handle_describe_request (client, ctx);
3486 case GST_RTSP_SETUP:
3487 handle_setup_request (client, ctx);
3490 handle_play_request (client, ctx);
3492 case GST_RTSP_PAUSE:
3493 handle_pause_request (client, ctx);
3495 case GST_RTSP_TEARDOWN:
3496 handle_teardown_request (client, ctx);
3498 case GST_RTSP_SET_PARAMETER:
3499 handle_set_param_request (client, ctx);
3501 case GST_RTSP_GET_PARAMETER:
3502 handle_get_param_request (client, ctx);
3504 case GST_RTSP_ANNOUNCE:
3505 if (version >= GST_RTSP_VERSION_2_0)
3506 goto invalid_command_for_version;
3507 handle_announce_request (client, ctx);
3509 case GST_RTSP_RECORD:
3510 if (version >= GST_RTSP_VERSION_2_0)
3511 goto invalid_command_for_version;
3512 handle_record_request (client, ctx);
3514 case GST_RTSP_REDIRECT:
3515 goto not_implemented;
3516 case GST_RTSP_INVALID:
3523 gst_rtsp_context_pop_current (ctx);
3525 g_object_unref (session);
3527 gst_rtsp_url_free (uri);
3533 GST_ERROR ("client %p: version %d not supported", client, version);
3534 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
3538 invalid_command_for_version:
3540 GST_ERROR ("client %p: invalid command for version", client);
3541 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
3546 GST_ERROR ("client %p: bad request", client);
3547 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
3552 GST_ERROR ("client %p: no pool configured", client);
3553 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
3558 GST_ERROR ("client %p: session not found", client);
3559 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
3564 GST_ERROR ("client %p: not allowed", client);
3565 /* error reply is already sent */
3568 unsupported_requirement:
3570 GST_ERROR ("client %p: Required option is not supported (%s)", client,
3572 send_option_not_supported_response (client, ctx, unsupported_reqs);
3573 g_free (unsupported_reqs);
3578 GST_ERROR ("client %p: method %d not implemented", client, method);
3579 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
3586 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
3588 GstRTSPClientPrivate *priv = client->priv;
3590 GstRTSPSession *session = NULL;
3591 GstRTSPContext sctx = { NULL }, *ctx;
3594 if (!(ctx = gst_rtsp_context_get_current ())) {
3596 ctx->auth = priv->auth;
3597 gst_rtsp_context_push_current (ctx);
3600 ctx->conn = priv->connection;
3601 ctx->client = client;
3602 ctx->request = NULL;
3604 ctx->method = GST_RTSP_INVALID;
3605 ctx->response = response;
3607 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
3608 gst_rtsp_message_dump (response);
3611 GST_INFO ("client %p: received a response", client);
3613 /* get the session if there is any */
3615 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
3616 if (res == GST_RTSP_OK) {
3617 if (priv->session_pool == NULL)
3620 /* we had a session in the request, find it again */
3621 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
3622 goto session_not_found;
3624 /* we add the session to the client list of watched sessions. When a session
3625 * disappears because it times out, we will be notified. If all sessions are
3626 * gone, we will close the connection */
3627 client_watch_session (client, session);
3630 ctx->session = session;
3632 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
3637 gst_rtsp_context_pop_current (ctx);
3639 g_object_unref (session);
3644 GST_ERROR ("client %p: no pool configured", client);
3649 GST_ERROR ("client %p: session not found", client);
3655 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
3657 GstRTSPClientPrivate *priv = client->priv;
3663 GstRTSPStreamTransport *trans;
3665 /* find the stream for this message */
3666 res = gst_rtsp_message_parse_data (message, &channel);
3667 if (res != GST_RTSP_OK)
3670 gst_rtsp_message_get_body (message, &data, &size);
3672 goto invalid_length;
3674 gst_rtsp_message_steal_body (message, &data, &size);
3676 /* Strip trailing \0 (which GstRTSPConnection adds) */
3679 buffer = gst_buffer_new_wrapped (data, size);
3682 g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
3684 GSocketAddress *addr;
3686 /* Only create the socket address once for the transport, we don't really
3687 * want to do that for every single packet.
3689 * The netaddress meta is later used by the RTP stack to know where
3690 * packets came from and allows us to match it again to a stream transport
3692 * In theory we could use the remote socket address of the RTSP connection
3693 * here, but this would fail with a custom configure_client_transport()
3697 g_object_get_data (G_OBJECT (trans), "rtsp-client.remote-addr"))) {
3698 const GstRTSPTransport *tr;
3699 GInetAddress *iaddr;
3701 tr = gst_rtsp_stream_transport_get_transport (trans);
3702 iaddr = g_inet_address_new_from_string (tr->destination);
3704 addr = g_inet_socket_address_new (iaddr, tr->client_port.min);
3705 g_object_unref (iaddr);
3706 g_object_set_data_full (G_OBJECT (trans), "rtsp-client.remote-addr",
3707 addr, (GDestroyNotify) g_object_unref);
3712 gst_buffer_add_net_address_meta (buffer, addr);
3715 /* dispatch to the stream based on the channel number */
3716 GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
3717 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
3719 GST_DEBUG_OBJECT (client, "received %u bytes of data for "
3720 "unknown channel %u", size, channel);
3721 gst_buffer_unref (buffer);
3729 GST_DEBUG ("client %p: Short message received, ignoring", client);
3735 * gst_rtsp_client_set_session_pool:
3736 * @client: a #GstRTSPClient
3737 * @pool: (transfer none) (nullable): a #GstRTSPSessionPool
3739 * Set @pool as the sessionpool for @client which it will use to find
3740 * or allocate sessions. the sessionpool is usually inherited from the server
3741 * that created the client but can be overridden later.
3744 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
3745 GstRTSPSessionPool * pool)
3747 GstRTSPSessionPool *old;
3748 GstRTSPClientPrivate *priv;
3750 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3752 priv = client->priv;
3755 g_object_ref (pool);
3757 g_mutex_lock (&priv->lock);
3758 old = priv->session_pool;
3759 priv->session_pool = pool;
3761 if (priv->session_removed_id) {
3762 g_signal_handler_disconnect (old, priv->session_removed_id);
3763 priv->session_removed_id = 0;
3765 g_mutex_unlock (&priv->lock);
3767 /* FIXME, should remove all sessions from the old pool for this client */
3769 g_object_unref (old);
3773 * gst_rtsp_client_get_session_pool:
3774 * @client: a #GstRTSPClient
3776 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
3778 * Returns: (transfer full) (nullable): a #GstRTSPSessionPool, unref after usage.
3780 GstRTSPSessionPool *
3781 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
3783 GstRTSPClientPrivate *priv;
3784 GstRTSPSessionPool *result;
3786 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3788 priv = client->priv;
3790 g_mutex_lock (&priv->lock);
3791 if ((result = priv->session_pool))
3792 g_object_ref (result);
3793 g_mutex_unlock (&priv->lock);
3799 * gst_rtsp_client_set_mount_points:
3800 * @client: a #GstRTSPClient
3801 * @mounts: (transfer none) (nullable): a #GstRTSPMountPoints
3803 * Set @mounts as the mount points for @client which it will use to map urls
3804 * to media streams. These mount points are usually inherited from the server that
3805 * created the client but can be overriden later.
3808 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
3809 GstRTSPMountPoints * mounts)
3811 GstRTSPClientPrivate *priv;
3812 GstRTSPMountPoints *old;
3814 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3816 priv = client->priv;
3819 g_object_ref (mounts);
3821 g_mutex_lock (&priv->lock);
3822 old = priv->mount_points;
3823 priv->mount_points = mounts;
3824 g_mutex_unlock (&priv->lock);
3827 g_object_unref (old);
3831 * gst_rtsp_client_get_mount_points:
3832 * @client: a #GstRTSPClient
3834 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
3836 * Returns: (transfer full) (nullable): a #GstRTSPMountPoints, unref after usage.
3838 GstRTSPMountPoints *
3839 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
3841 GstRTSPClientPrivate *priv;
3842 GstRTSPMountPoints *result;
3844 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3846 priv = client->priv;
3848 g_mutex_lock (&priv->lock);
3849 if ((result = priv->mount_points))
3850 g_object_ref (result);
3851 g_mutex_unlock (&priv->lock);
3857 * gst_rtsp_client_set_auth:
3858 * @client: a #GstRTSPClient
3859 * @auth: (transfer none) (nullable): a #GstRTSPAuth
3861 * configure @auth to be used as the authentication manager of @client.
3864 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
3866 GstRTSPClientPrivate *priv;
3869 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3871 priv = client->priv;
3874 g_object_ref (auth);
3876 g_mutex_lock (&priv->lock);
3879 g_mutex_unlock (&priv->lock);
3882 g_object_unref (old);
3887 * gst_rtsp_client_get_auth:
3888 * @client: a #GstRTSPClient
3890 * Get the #GstRTSPAuth used as the authentication manager of @client.
3892 * Returns: (transfer full) (nullable): the #GstRTSPAuth of @client.
3893 * g_object_unref() after usage.
3896 gst_rtsp_client_get_auth (GstRTSPClient * client)
3898 GstRTSPClientPrivate *priv;
3899 GstRTSPAuth *result;
3901 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3903 priv = client->priv;
3905 g_mutex_lock (&priv->lock);
3906 if ((result = priv->auth))
3907 g_object_ref (result);
3908 g_mutex_unlock (&priv->lock);
3914 * gst_rtsp_client_set_thread_pool:
3915 * @client: a #GstRTSPClient
3916 * @pool: (transfer none) (nullable): a #GstRTSPThreadPool
3918 * configure @pool to be used as the thread pool of @client.
3921 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
3922 GstRTSPThreadPool * pool)
3924 GstRTSPClientPrivate *priv;
3925 GstRTSPThreadPool *old;
3927 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
3929 priv = client->priv;
3932 g_object_ref (pool);
3934 g_mutex_lock (&priv->lock);
3935 old = priv->thread_pool;
3936 priv->thread_pool = pool;
3937 g_mutex_unlock (&priv->lock);
3940 g_object_unref (old);
3944 * gst_rtsp_client_get_thread_pool:
3945 * @client: a #GstRTSPClient
3947 * Get the #GstRTSPThreadPool used as the thread pool of @client.
3949 * Returns: (transfer full) (nullable): the #GstRTSPThreadPool of @client. g_object_unref() after
3953 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
3955 GstRTSPClientPrivate *priv;
3956 GstRTSPThreadPool *result;
3958 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3960 priv = client->priv;
3962 g_mutex_lock (&priv->lock);
3963 if ((result = priv->thread_pool))
3964 g_object_ref (result);
3965 g_mutex_unlock (&priv->lock);
3971 * gst_rtsp_client_set_connection:
3972 * @client: a #GstRTSPClient
3973 * @conn: (transfer full): a #GstRTSPConnection
3975 * Set the #GstRTSPConnection of @client. This function takes ownership of
3978 * Returns: %TRUE on success.
3981 gst_rtsp_client_set_connection (GstRTSPClient * client,
3982 GstRTSPConnection * conn)
3984 GstRTSPClientPrivate *priv;
3985 GSocket *read_socket;
3986 GSocketAddress *address;
3988 GError *error = NULL;
3990 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
3991 g_return_val_if_fail (conn != NULL, FALSE);
3993 priv = client->priv;
3995 read_socket = gst_rtsp_connection_get_read_socket (conn);
3997 if (!(address = g_socket_get_local_address (read_socket, &error)))
4000 g_free (priv->server_ip);
4001 /* keep the original ip that the client connected to */
4002 if (G_IS_INET_SOCKET_ADDRESS (address)) {
4003 GInetAddress *iaddr;
4005 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
4007 /* socket might be ipv6 but adress still ipv4 */
4008 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
4009 priv->server_ip = g_inet_address_to_string (iaddr);
4010 g_object_unref (address);
4012 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
4013 priv->server_ip = g_strdup ("unknown");
4016 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
4017 priv->server_ip, priv->is_ipv6);
4019 url = gst_rtsp_connection_get_url (conn);
4020 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
4022 priv->connection = conn;
4029 GST_ERROR ("could not get local address %s", error->message);
4030 g_error_free (error);
4036 * gst_rtsp_client_get_connection:
4037 * @client: a #GstRTSPClient
4039 * Get the #GstRTSPConnection of @client.
4041 * Returns: (transfer none) (nullable): the #GstRTSPConnection of @client.
4042 * The connection object returned remains valid until the client is freed.
4045 gst_rtsp_client_get_connection (GstRTSPClient * client)
4047 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
4049 return client->priv->connection;
4053 * gst_rtsp_client_set_send_func:
4054 * @client: a #GstRTSPClient
4055 * @func: (scope notified): a #GstRTSPClientSendFunc
4056 * @user_data: (closure): user data passed to @func
4057 * @notify: (allow-none): called when @user_data is no longer in use
4059 * Set @func as the callback that will be called when a new message needs to be
4060 * sent to the client. @user_data is passed to @func and @notify is called when
4061 * @user_data is no longer in use.
4063 * By default, the client will send the messages on the #GstRTSPConnection that
4064 * was configured with gst_rtsp_client_attach() was called.
4067 gst_rtsp_client_set_send_func (GstRTSPClient * client,
4068 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
4070 GstRTSPClientPrivate *priv;
4071 GDestroyNotify old_notify;
4074 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
4076 priv = client->priv;
4078 g_mutex_lock (&priv->send_lock);
4079 priv->send_func = func;
4080 old_notify = priv->send_notify;
4081 old_data = priv->send_data;
4082 priv->send_notify = notify;
4083 priv->send_data = user_data;
4084 g_mutex_unlock (&priv->send_lock);
4087 old_notify (old_data);
4091 * gst_rtsp_client_handle_message:
4092 * @client: a #GstRTSPClient
4093 * @message: (transfer none): an #GstRTSPMessage
4095 * Let the client handle @message.
4097 * Returns: a #GstRTSPResult.
4100 gst_rtsp_client_handle_message (GstRTSPClient * client,
4101 GstRTSPMessage * message)
4103 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
4104 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
4106 switch (message->type) {
4107 case GST_RTSP_MESSAGE_REQUEST:
4108 handle_request (client, message);
4110 case GST_RTSP_MESSAGE_RESPONSE:
4111 handle_response (client, message);
4113 case GST_RTSP_MESSAGE_DATA:
4114 handle_data (client, message);
4123 * gst_rtsp_client_send_message:
4124 * @client: a #GstRTSPClient
4125 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
4126 * the message to or %NULL
4127 * @message: (transfer none): The #GstRTSPMessage to send
4129 * Send a message message to the remote end. @message must be a
4130 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
4133 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
4134 GstRTSPMessage * message)
4136 GstRTSPContext sctx = { NULL }
4138 GstRTSPClientPrivate *priv;
4140 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
4141 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
4142 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
4143 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
4145 priv = client->priv;
4147 if (!(ctx = gst_rtsp_context_get_current ())) {
4149 ctx->auth = priv->auth;
4150 gst_rtsp_context_push_current (ctx);
4153 ctx->conn = priv->connection;
4154 ctx->client = client;
4155 ctx->session = session;
4157 send_message (client, ctx, message, FALSE);
4160 gst_rtsp_context_pop_current (ctx);
4166 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
4167 gboolean close, gpointer user_data)
4169 GstRTSPClientPrivate *priv = client->priv;
4173 /* send the message */
4174 ret = gst_rtsp_watch_send_message (priv->watch, message, &id);
4175 if (ret != GST_RTSP_OK)
4178 /* if close flag is set, store the seq number so we can wait until it's
4179 * written to the client to close the connection */
4181 priv->close_seq = id;
4183 if (gst_rtsp_message_get_type (message) == GST_RTSP_MESSAGE_DATA) {
4187 r = gst_rtsp_message_parse_data (message, &channel);
4188 if (r != GST_RTSP_OK) {
4193 /* check if the message has been queued for transmission in watch */
4195 /* store the seq number so we can wait until it has been sent */
4196 GST_DEBUG_OBJECT (client, "wait for message %d, channel %d", id, channel);
4197 set_data_seq (client, channel, id);
4199 GstRTSPStreamTransport *trans;
4202 g_hash_table_lookup (priv->transports,
4203 GINT_TO_POINTER ((gint) channel));
4205 GST_DEBUG_OBJECT (client, "emit 'message-sent' signal");
4206 g_mutex_unlock (&priv->send_lock);
4207 gst_rtsp_stream_transport_message_sent (trans);
4208 g_mutex_lock (&priv->send_lock);
4213 return ret == GST_RTSP_OK;
4218 GST_DEBUG_OBJECT (client, "got error %d", ret);
4223 static GstRTSPResult
4224 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
4227 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
4230 static GstRTSPResult
4231 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
4233 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4234 GstRTSPClientPrivate *priv = client->priv;
4235 GstRTSPStreamTransport *trans = NULL;
4237 gboolean close = FALSE;
4239 g_mutex_lock (&priv->send_lock);
4241 if (get_data_channel (client, cseq, &channel)) {
4242 trans = g_hash_table_lookup (priv->transports, GINT_TO_POINTER (channel));
4243 set_data_seq (client, channel, 0);
4246 if (priv->close_seq && priv->close_seq == cseq) {
4247 GST_INFO ("client %p: send close message", client);
4249 priv->close_seq = 0;
4252 g_mutex_unlock (&priv->send_lock);
4255 GST_DEBUG_OBJECT (client, "emit 'message-sent' signal");
4256 gst_rtsp_stream_transport_message_sent (trans);
4260 gst_rtsp_client_close (client);
4265 static GstRTSPResult
4266 closed (GstRTSPWatch * watch, gpointer user_data)
4268 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4269 GstRTSPClientPrivate *priv = client->priv;
4270 const gchar *tunnelid;
4272 GST_INFO ("client %p: connection closed", client);
4274 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
4275 g_mutex_lock (&tunnels_lock);
4276 /* remove from tunnelids */
4277 g_hash_table_remove (tunnels, tunnelid);
4278 g_mutex_unlock (&tunnels_lock);
4281 gst_rtsp_watch_set_flushing (watch, TRUE);
4282 g_mutex_lock (&priv->watch_lock);
4283 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
4284 g_mutex_unlock (&priv->watch_lock);
4289 static GstRTSPResult
4290 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
4292 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4295 str = gst_rtsp_strresult (result);
4296 GST_INFO ("client %p: received an error %s", client, str);
4302 static GstRTSPResult
4303 error_full (GstRTSPWatch * watch, GstRTSPResult result,
4304 GstRTSPMessage * message, guint id, gpointer user_data)
4306 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4309 str = gst_rtsp_strresult (result);
4311 ("client %p: error when handling message %p with id %d: %s",
4312 client, message, id, str);
4319 remember_tunnel (GstRTSPClient * client)
4321 GstRTSPClientPrivate *priv = client->priv;
4322 const gchar *tunnelid;
4324 /* store client in the pending tunnels */
4325 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
4326 if (tunnelid == NULL)
4329 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
4331 /* we can't have two clients connecting with the same tunnelid */
4332 g_mutex_lock (&tunnels_lock);
4333 if (g_hash_table_lookup (tunnels, tunnelid))
4334 goto tunnel_existed;
4336 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
4337 g_mutex_unlock (&tunnels_lock);
4344 GST_ERROR ("client %p: no tunnelid provided", client);
4349 g_mutex_unlock (&tunnels_lock);
4350 GST_ERROR ("client %p: tunnel session %s already existed", client,
4356 static GstRTSPResult
4357 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
4359 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4360 GstRTSPClientPrivate *priv = client->priv;
4362 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
4365 /* ignore error, it'll only be a problem when the client does a POST again */
4366 remember_tunnel (client);
4371 static GstRTSPStatusCode
4372 handle_tunnel (GstRTSPClient * client)
4374 GstRTSPClientPrivate *priv = client->priv;
4375 GstRTSPClient *oclient;
4376 GstRTSPClientPrivate *opriv;
4377 const gchar *tunnelid;
4379 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
4380 if (tunnelid == NULL)
4383 /* check for previous tunnel */
4384 g_mutex_lock (&tunnels_lock);
4385 oclient = g_hash_table_lookup (tunnels, tunnelid);
4387 if (oclient == NULL) {
4388 /* no previous tunnel, remember tunnel */
4389 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
4390 g_mutex_unlock (&tunnels_lock);
4392 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
4393 client, priv->connection);
4395 /* merge both tunnels into the first client */
4396 /* remove the old client from the table. ref before because removing it will
4397 * remove the ref to it. */
4398 g_object_ref (oclient);
4399 g_hash_table_remove (tunnels, tunnelid);
4400 g_mutex_unlock (&tunnels_lock);
4402 opriv = oclient->priv;
4404 g_mutex_lock (&opriv->watch_lock);
4405 if (opriv->watch == NULL)
4407 if (opriv->tstate == priv->tstate)
4408 goto tunnel_duplicate_id;
4410 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
4411 oclient, opriv->connection, priv->connection);
4413 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
4414 gst_rtsp_watch_reset (priv->watch);
4415 gst_rtsp_watch_reset (opriv->watch);
4416 g_mutex_unlock (&opriv->watch_lock);
4417 g_object_unref (oclient);
4419 /* the old client owns the tunnel now, the new one will be freed */
4420 g_source_destroy ((GSource *) priv->watch);
4422 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
4425 return GST_RTSP_STS_OK;
4430 GST_ERROR ("client %p: no tunnelid provided", client);
4431 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
4435 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
4436 g_mutex_unlock (&opriv->watch_lock);
4437 g_object_unref (oclient);
4438 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
4440 tunnel_duplicate_id:
4442 GST_ERROR ("client %p: tunnel session %s was duplicate", client, tunnelid);
4443 g_mutex_unlock (&opriv->watch_lock);
4444 g_object_unref (oclient);
4445 return GST_RTSP_STS_BAD_REQUEST;
4449 static GstRTSPStatusCode
4450 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
4452 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4454 GST_INFO ("client %p: tunnel get (connection %p)", client,
4455 client->priv->connection);
4457 g_mutex_lock (&client->priv->lock);
4458 client->priv->tstate = TUNNEL_STATE_GET;
4459 g_mutex_unlock (&client->priv->lock);
4461 return handle_tunnel (client);
4464 static GstRTSPResult
4465 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
4467 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4469 GST_INFO ("client %p: tunnel post (connection %p)", client,
4470 client->priv->connection);
4472 g_mutex_lock (&client->priv->lock);
4473 client->priv->tstate = TUNNEL_STATE_POST;
4474 g_mutex_unlock (&client->priv->lock);
4476 if (handle_tunnel (client) != GST_RTSP_STS_OK)
4477 return GST_RTSP_ERROR;
4482 static GstRTSPResult
4483 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
4484 GstRTSPMessage * response, gpointer user_data)
4486 GstRTSPClientClass *klass;
4488 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
4489 klass = GST_RTSP_CLIENT_GET_CLASS (client);
4491 if (klass->tunnel_http_response) {
4492 klass->tunnel_http_response (client, request, response);
4498 static GstRTSPWatchFuncs watch_funcs = {
4507 tunnel_http_response
4511 client_watch_notify (GstRTSPClient * client)
4513 GstRTSPClientPrivate *priv = client->priv;
4514 gboolean closed = TRUE;
4516 GST_INFO ("client %p: watch destroyed", client);
4518 /* remove all sessions if the media says so and so drop the extra client ref */
4519 rtsp_ctrl_timeout_remove (priv);
4520 gst_rtsp_client_session_filter (client, cleanup_session, &closed);
4522 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
4523 g_object_unref (client);
4527 * gst_rtsp_client_attach:
4528 * @client: a #GstRTSPClient
4529 * @context: (allow-none): a #GMainContext
4531 * Attaches @client to @context. When the mainloop for @context is run, the
4532 * client will be dispatched. When @context is %NULL, the default context will be
4535 * This function should be called when the client properties and urls are fully
4536 * configured and the client is ready to start.
4538 * Returns: the ID (greater than 0) for the source within the GMainContext.
4541 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
4543 GstRTSPClientPrivate *priv;
4547 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
4548 priv = client->priv;
4549 g_return_val_if_fail (priv->connection != NULL, 0);
4550 g_return_val_if_fail (priv->watch == NULL, 0);
4552 /* make sure noone will free the context before the watch is destroyed */
4553 priv->watch_context = g_main_context_ref (context);
4555 /* create watch for the connection and attach */
4556 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
4557 g_object_ref (client), (GDestroyNotify) client_watch_notify);
4558 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
4559 (GDestroyNotify) gst_rtsp_watch_unref);
4561 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
4563 GST_INFO ("client %p: attaching to context %p", client, context);
4564 res = gst_rtsp_watch_attach (priv->watch, context);
4566 /* Setting up a timeout for the RTSP control channel until a session
4567 * is up where it is handling timeouts. */
4568 rtsp_ctrl_timeout_remove (priv); /* removing old if any */
4569 g_mutex_lock (&priv->lock);
4571 timer_src = g_timeout_source_new_seconds (RTSP_CTRL_CB_INTERVAL);
4572 g_source_set_callback (timer_src, rtsp_ctrl_timeout_cb, client, NULL);
4573 priv->rtsp_ctrl_timeout_id = g_source_attach (timer_src, priv->watch_context);
4574 g_source_unref (timer_src);
4575 GST_DEBUG ("rtsp control setting up session timeout id=%u.",
4576 priv->rtsp_ctrl_timeout_id);
4578 g_mutex_unlock (&priv->lock);
4584 * gst_rtsp_client_session_filter:
4585 * @client: a #GstRTSPClient
4586 * @func: (scope call) (allow-none): a callback
4587 * @user_data: user data passed to @func
4589 * Call @func for each session managed by @client. The result value of @func
4590 * determines what happens to the session. @func will be called with @client
4591 * locked so no further actions on @client can be performed from @func.
4593 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
4596 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
4598 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
4599 * will also be added with an additional ref to the result #GList of this
4602 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
4604 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
4605 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
4606 * element in the #GList should be unreffed before the list is freed.
4609 gst_rtsp_client_session_filter (GstRTSPClient * client,
4610 GstRTSPClientSessionFilterFunc func, gpointer user_data)
4612 GstRTSPClientPrivate *priv;
4613 GList *result, *walk, *next;
4614 GHashTable *visited;
4617 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
4619 priv = client->priv;
4623 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
4625 g_mutex_lock (&priv->lock);
4627 cookie = priv->sessions_cookie;
4628 for (walk = priv->sessions; walk; walk = next) {
4629 GstRTSPSession *sess = walk->data;
4630 GstRTSPFilterResult res;
4633 next = g_list_next (walk);
4636 /* only visit each session once */
4637 if (g_hash_table_contains (visited, sess))
4640 g_hash_table_add (visited, g_object_ref (sess));
4641 g_mutex_unlock (&priv->lock);
4643 res = func (client, sess, user_data);
4645 g_mutex_lock (&priv->lock);
4647 res = GST_RTSP_FILTER_REF;
4649 changed = (cookie != priv->sessions_cookie);
4652 case GST_RTSP_FILTER_REMOVE:
4653 /* stop watching the session and pretend it went away, if the list was
4654 * changed, we can't use the current list position, try to see if we
4655 * still have the session */
4656 client_unwatch_session (client, sess, changed ? NULL : walk);
4657 cookie = priv->sessions_cookie;
4659 case GST_RTSP_FILTER_REF:
4660 result = g_list_prepend (result, g_object_ref (sess));
4662 case GST_RTSP_FILTER_KEEP:
4669 g_mutex_unlock (&priv->lock);
4672 g_hash_table_unref (visited);