2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
26 #include <sys/types.h>
27 #include <netinet/in.h>
29 #include <sys/socket.h>
32 #include <arpa/inet.h>
33 #include <sys/ioctl.h>
35 #include "rtsp-server.h"
36 #include "rtsp-client.h"
38 #define DEFAULT_ADDRESS "0.0.0.0"
39 /* #define DEFAULT_ADDRESS "::0" */
40 #define DEFAULT_SERVICE "8554"
41 #define DEFAULT_BACKLOG 5
43 /* Define to use the SO_LINGER option so that the server sockets can be resused
44 * sooner. Disabled for now because it is not very well implemented by various
45 * OSes and it causes clients to fail to read the TEARDOWN response. */
60 G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
62 GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
63 #define GST_CAT_DEFAULT rtsp_server_debug
65 static void gst_rtsp_server_get_property (GObject * object, guint propid,
66 GValue * value, GParamSpec * pspec);
67 static void gst_rtsp_server_set_property (GObject * object, guint propid,
68 const GValue * value, GParamSpec * pspec);
69 static void gst_rtsp_server_finalize (GObject * object);
71 static GstRTSPClient *default_create_client (GstRTSPServer * server);
72 static gboolean default_accept_client (GstRTSPServer * server,
73 GstRTSPClient * client, GIOChannel * channel);
76 gst_rtsp_server_class_init (GstRTSPServerClass * klass)
78 GObjectClass *gobject_class;
80 gobject_class = G_OBJECT_CLASS (klass);
82 gobject_class->get_property = gst_rtsp_server_get_property;
83 gobject_class->set_property = gst_rtsp_server_set_property;
84 gobject_class->finalize = gst_rtsp_server_finalize;
87 * GstRTSPServer::address
89 * The address of the server. This is the address where the server will
92 g_object_class_install_property (gobject_class, PROP_ADDRESS,
93 g_param_spec_string ("address", "Address",
94 "The address the server uses to listen on", DEFAULT_ADDRESS,
95 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
97 * GstRTSPServer::service
99 * The service of the server. This is either a string with the service name or
100 * a port number (as a string) the server will listen on.
102 g_object_class_install_property (gobject_class, PROP_SERVICE,
103 g_param_spec_string ("service", "Service",
104 "The service or port number the server uses to listen on",
105 DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
107 * GstRTSPServer::backlog
109 * The backlog argument defines the maximum length to which the queue of
110 * pending connections for the server may grow. If a connection request arrives
111 * when the queue is full, the client may receive an error with an indication of
112 * ECONNREFUSED or, if the underlying protocol supports retransmission, the
113 * request may be ignored so that a later reattempt at connection succeeds.
115 g_object_class_install_property (gobject_class, PROP_BACKLOG,
116 g_param_spec_int ("backlog", "Backlog",
117 "The maximum length to which the queue "
118 "of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
119 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
121 * GstRTSPServer::session-pool
123 * The session pool of the server. By default each server has a separate
124 * session pool but sessions can be shared between servers by setting the same
125 * session pool on multiple servers.
127 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
128 g_param_spec_object ("session-pool", "Session Pool",
129 "The session pool to use for client session",
130 GST_TYPE_RTSP_SESSION_POOL,
131 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
133 * GstRTSPServer::media-mapping
135 * The media mapping to use for this server. By default the server has no
136 * media mapping and thus cannot map urls to media streams.
138 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
139 g_param_spec_object ("media-mapping", "Media Mapping",
140 "The media mapping to use for client session",
141 GST_TYPE_RTSP_MEDIA_MAPPING,
142 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
144 klass->create_client = default_create_client;
145 klass->accept_client = default_accept_client;
147 GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
151 gst_rtsp_server_init (GstRTSPServer * server)
153 server->lock = g_mutex_new ();
154 server->address = g_strdup (DEFAULT_ADDRESS);
155 server->service = g_strdup (DEFAULT_SERVICE);
156 server->backlog = DEFAULT_BACKLOG;
157 server->session_pool = gst_rtsp_session_pool_new ();
158 server->media_mapping = gst_rtsp_media_mapping_new ();
162 gst_rtsp_server_finalize (GObject * object)
164 GstRTSPServer *server = GST_RTSP_SERVER (object);
166 GST_DEBUG_OBJECT (server, "finalize server");
168 g_mutex_free (server->lock);
169 g_free (server->address);
170 g_free (server->service);
172 g_object_unref (server->session_pool);
173 g_object_unref (server->media_mapping);
176 g_object_unref (server->auth);
178 G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
182 * gst_rtsp_server_new:
184 * Create a new #GstRTSPServer instance.
187 gst_rtsp_server_new (void)
189 GstRTSPServer *result;
191 result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
197 * gst_rtsp_server_set_address:
198 * @server: a #GstRTSPServer
199 * @address: the address
201 * Configure @server to accept connections on the given address.
203 * This function must be called before the server is bound.
206 gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
208 g_return_if_fail (GST_IS_RTSP_SERVER (server));
209 g_return_if_fail (address != NULL);
211 g_free (server->address);
212 server->address = g_strdup (address);
216 * gst_rtsp_server_get_address:
217 * @server: a #GstRTSPServer
219 * Get the address on which the server will accept connections.
221 * Returns: the server address. g_free() after usage.
224 gst_rtsp_server_get_address (GstRTSPServer * server)
226 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
228 return g_strdup (server->address);
232 * gst_rtsp_server_set_service:
233 * @server: a #GstRTSPServer
234 * @service: the service
236 * Configure @server to accept connections on the given service.
237 * @service should be a string containing the service name (see services(5)) or
238 * a string containing a port number between 1 and 65535.
240 * This function must be called before the server is bound.
243 gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
245 g_return_if_fail (GST_IS_RTSP_SERVER (server));
246 g_return_if_fail (service != NULL);
248 g_free (server->service);
249 server->service = g_strdup (service);
253 * gst_rtsp_server_get_service:
254 * @server: a #GstRTSPServer
256 * Get the service on which the server will accept connections.
258 * Returns: the service. use g_free() after usage.
261 gst_rtsp_server_get_service (GstRTSPServer * server)
263 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
265 return g_strdup (server->service);
269 * gst_rtsp_server_set_backlog:
270 * @server: a #GstRTSPServer
271 * @backlog: the backlog
273 * configure the maximum amount of requests that may be queued for the
276 * This function must be called before the server is bound.
279 gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
281 g_return_if_fail (GST_IS_RTSP_SERVER (server));
283 server->backlog = backlog;
287 * gst_rtsp_server_get_backlog:
288 * @server: a #GstRTSPServer
290 * The maximum amount of queued requests for the server.
292 * Returns: the server backlog.
295 gst_rtsp_server_get_backlog (GstRTSPServer * server)
297 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
299 return server->backlog;
303 * gst_rtsp_server_set_session_pool:
304 * @server: a #GstRTSPServer
305 * @pool: a #GstRTSPSessionPool
307 * configure @pool to be used as the session pool of @server.
310 gst_rtsp_server_set_session_pool (GstRTSPServer * server,
311 GstRTSPSessionPool * pool)
313 GstRTSPSessionPool *old;
315 g_return_if_fail (GST_IS_RTSP_SERVER (server));
317 old = server->session_pool;
322 server->session_pool = pool;
324 g_object_unref (old);
329 * gst_rtsp_server_get_session_pool:
330 * @server: a #GstRTSPServer
332 * Get the #GstRTSPSessionPool used as the session pool of @server.
334 * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
338 gst_rtsp_server_get_session_pool (GstRTSPServer * server)
340 GstRTSPSessionPool *result;
342 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
344 if ((result = server->session_pool))
345 g_object_ref (result);
351 * gst_rtsp_server_set_media_mapping:
352 * @server: a #GstRTSPServer
353 * @mapping: a #GstRTSPMediaMapping
355 * configure @mapping to be used as the media mapping of @server.
358 gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
359 GstRTSPMediaMapping * mapping)
361 GstRTSPMediaMapping *old;
363 g_return_if_fail (GST_IS_RTSP_SERVER (server));
365 old = server->media_mapping;
367 if (old != mapping) {
369 g_object_ref (mapping);
370 server->media_mapping = mapping;
372 g_object_unref (old);
378 * gst_rtsp_server_get_media_mapping:
379 * @server: a #GstRTSPServer
381 * Get the #GstRTSPMediaMapping used as the media mapping of @server.
383 * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
386 GstRTSPMediaMapping *
387 gst_rtsp_server_get_media_mapping (GstRTSPServer * server)
389 GstRTSPMediaMapping *result;
391 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
393 if ((result = server->media_mapping))
394 g_object_ref (result);
400 * gst_rtsp_server_set_auth:
401 * @server: a #GstRTSPServer
402 * @auth: a #GstRTSPAuth
404 * configure @auth to be used as the authentication manager of @server.
407 gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
411 g_return_if_fail (GST_IS_RTSP_SERVER (server));
420 g_object_unref (old);
426 * gst_rtsp_server_get_auth:
427 * @server: a #GstRTSPServer
429 * Get the #GstRTSPAuth used as the authentication manager of @server.
431 * Returns: the #GstRTSPAuth of @server. g_object_unref() after
435 gst_rtsp_server_get_auth (GstRTSPServer * server)
439 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
441 if ((result = server->auth))
442 g_object_ref (result);
448 gst_rtsp_server_get_property (GObject * object, guint propid,
449 GValue * value, GParamSpec * pspec)
451 GstRTSPServer *server = GST_RTSP_SERVER (object);
455 g_value_take_string (value, gst_rtsp_server_get_address (server));
458 g_value_take_string (value, gst_rtsp_server_get_service (server));
461 g_value_set_int (value, gst_rtsp_server_get_backlog (server));
463 case PROP_SESSION_POOL:
464 g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
466 case PROP_MEDIA_MAPPING:
467 g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
470 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
475 gst_rtsp_server_set_property (GObject * object, guint propid,
476 const GValue * value, GParamSpec * pspec)
478 GstRTSPServer *server = GST_RTSP_SERVER (object);
482 gst_rtsp_server_set_address (server, g_value_get_string (value));
485 gst_rtsp_server_set_service (server, g_value_get_string (value));
488 gst_rtsp_server_set_backlog (server, g_value_get_int (value));
490 case PROP_SESSION_POOL:
491 gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
493 case PROP_MEDIA_MAPPING:
494 gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
497 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
501 /* Prepare a server socket for @server and make it listen on the configured port */
503 gst_rtsp_server_sink_init_send (GstRTSPServer * server)
507 struct addrinfo hints;
508 struct addrinfo *result, *rp;
510 struct linger linger;
513 memset (&hints, 0, sizeof (struct addrinfo));
514 hints.ai_family = AF_UNSPEC; /* Allow IPv4 or IPv6 */
515 hints.ai_socktype = SOCK_STREAM; /* stream socket */
516 hints.ai_flags = AI_PASSIVE | AI_CANONNAME; /* For wildcard IP address */
517 hints.ai_protocol = 0; /* Any protocol */
518 hints.ai_canonname = NULL;
519 hints.ai_addr = NULL;
520 hints.ai_next = NULL;
522 GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
525 /* resolve the server IP address */
527 getaddrinfo (server->address, server->service, &hints, &result)) != 0)
530 /* create server socket, we loop through all the addresses until we manage to
531 * create a socket and bind. */
532 for (rp = result; rp; rp = rp->ai_next) {
533 sockfd = socket (rp->ai_family, rp->ai_socktype, rp->ai_protocol);
535 GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
540 /* make address reusable */
542 if (setsockopt (sockfd, SOL_SOCKET, SO_REUSEADDR,
543 (void *) &ret, sizeof (ret)) < 0) {
544 /* warn but try to bind anyway */
545 GST_WARNING_OBJECT (server, "failed to reuse socker (%s)",
549 if (bind (sockfd, rp->ai_addr, rp->ai_addrlen) == 0) {
550 GST_DEBUG_OBJECT (server, "bind on %s", rp->ai_canonname);
554 GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
558 freeaddrinfo (result);
563 GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d", sockfd);
565 /* keep connection alive; avoids SIGPIPE during write */
567 if (setsockopt (sockfd, SOL_SOCKET, SO_KEEPALIVE,
568 (void *) &ret, sizeof (ret)) < 0)
569 goto keepalive_failed;
572 /* make sure socket is reset 5 seconds after close. This ensure that we can
573 * reuse the socket quickly while still having a chance to send data to the
577 if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
578 (void *) &linger, sizeof (linger)) < 0)
582 /* set the server socket to nonblocking */
583 fcntl (sockfd, F_SETFL, O_NONBLOCK);
585 GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
586 sockfd, server->backlog);
587 if (listen (sockfd, server->backlog) == -1)
590 GST_DEBUG_OBJECT (server,
591 "listened on server socket %d, returning from connection setup", sockfd);
593 /* create IO channel for the socket */
594 channel = g_io_channel_unix_new (sockfd);
595 g_io_channel_set_close_on_unref (channel, TRUE);
597 GST_INFO_OBJECT (server, "listening on service %s", server->service);
604 GST_ERROR_OBJECT (server, "failed to resolve address: %s",
610 GST_ERROR_OBJECT (server, "failed to create socket: %s",
616 GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s",
623 GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
630 GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
644 unmanage_client (GstRTSPClient * client, GstRTSPServer * server)
646 GST_DEBUG_OBJECT (server, "unmanage client %p", client);
648 g_mutex_lock (server->lock);
649 server->clients = g_list_remove (server->clients, client);
650 g_mutex_unlock (server->lock);
652 g_object_unref (client);
655 /* add the client to the active list of clients, takes ownership of
658 manage_client (GstRTSPServer * server, GstRTSPClient * client)
660 GST_DEBUG_OBJECT (server, "manage client %p", client);
661 gst_rtsp_client_set_server (client, server);
663 g_mutex_lock (server->lock);
664 g_signal_connect (client, "closed", (GCallback) unmanage_client, server);
665 server->clients = g_list_prepend (server->clients, client);
666 g_mutex_unlock (server->lock);
669 static GstRTSPClient *
670 default_create_client (GstRTSPServer * server)
672 GstRTSPClient *client;
674 /* a new client connected, create a session to handle the client. */
675 client = gst_rtsp_client_new ();
677 /* set the session pool that this client should use */
678 gst_rtsp_client_set_session_pool (client, server->session_pool);
679 /* set the media mapping that this client should use */
680 gst_rtsp_client_set_media_mapping (client, server->media_mapping);
681 /* set authentication manager */
682 gst_rtsp_client_set_auth (client, server->auth);
687 /* default method for creating a new client object in the server to accept and
688 * handle a client connection on this server */
690 default_accept_client (GstRTSPServer * server, GstRTSPClient * client,
691 GIOChannel * channel)
693 /* accept connections for that client, this function returns after accepting
694 * the connection and will run the remainder of the communication with the
695 * client asyncronously. */
696 if (!gst_rtsp_client_accept (client, channel))
704 GST_ERROR_OBJECT (server,
705 "Could not accept client on server : %s (%d)", g_strerror (errno),
712 * gst_rtsp_server_io_func:
713 * @channel: a #GIOChannel
714 * @condition: the condition on @source
716 * A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
717 * new connection on @channel or @server.
719 * Returns: TRUE if the source could be connected, FALSE if an error occured.
722 gst_rtsp_server_io_func (GIOChannel * channel, GIOCondition condition,
723 GstRTSPServer * server)
726 GstRTSPClient *client = NULL;
727 GstRTSPServerClass *klass;
729 if (condition & G_IO_IN) {
730 klass = GST_RTSP_SERVER_GET_CLASS (server);
732 if (klass->create_client)
733 client = klass->create_client (server);
737 /* a new client connected, create a client object to handle the client. */
738 if (klass->accept_client)
739 result = klass->accept_client (server, client, channel);
743 /* manage the client connection */
744 manage_client (server, client);
746 GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
753 GST_ERROR_OBJECT (server, "failed to create a client");
758 GST_ERROR_OBJECT (server, "failed to accept client");
759 gst_object_unref (client);
765 * gst_rtsp_server_get_io_channel:
766 * @server: a #GstRTSPServer
768 * Create a #GIOChannel for @server.
770 * Returns: the GIOChannel for @server or NULL when an error occured.
773 gst_rtsp_server_get_io_channel (GstRTSPServer * server)
777 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
779 if (!(channel = gst_rtsp_server_sink_init_send (server)))
786 GST_ERROR_OBJECT (server, "failed to initialize server");
792 watch_destroyed (GstRTSPServer * server)
794 GST_DEBUG_OBJECT (server, "source destroyed");
798 * gst_rtsp_server_create_watch:
799 * @server: a #GstRTSPServer
801 * Create a #GSource for @server. The new source will have a default
802 * #GIOFunc of gst_rtsp_server_io_func().
804 * Returns: the #GSource for @server or NULL when an error occured.
807 gst_rtsp_server_create_watch (GstRTSPServer * server)
812 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
814 channel = gst_rtsp_server_get_io_channel (server);
818 /* create a watch for reads (new connections) and possible errors */
819 source = g_io_create_watch (channel, G_IO_IN |
820 G_IO_ERR | G_IO_HUP | G_IO_NVAL);
821 g_io_channel_unref (channel);
823 /* configure the callback */
824 g_source_set_callback (source,
825 (GSourceFunc) gst_rtsp_server_io_func, server,
826 (GDestroyNotify) watch_destroyed);
832 GST_ERROR_OBJECT (server, "failed to create IO channel");
838 * gst_rtsp_server_attach:
839 * @server: a #GstRTSPServer
840 * @context: a #GMainContext
842 * Attaches @server to @context. When the mainloop for @context is run, the
843 * server will be dispatched.
845 * This function should be called when the server properties and urls are fully
846 * configured and the server is ready to start.
848 * Returns: the ID (greater than 0) for the source within the GMainContext.
851 gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
856 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
858 source = gst_rtsp_server_create_watch (server);
862 res = g_source_attach (source, context);
863 g_source_unref (source);
870 GST_ERROR_OBJECT (server, "failed to create watch");