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.
20 #include <sys/ioctl.h>
22 #include "rtsp-server.h"
23 #include "rtsp-client.h"
25 #define DEFAULT_ADDRESS "0.0.0.0"
26 /* #define DEFAULT_ADDRESS "::0" */
27 #define DEFAULT_SERVICE "8554"
28 #define DEFAULT_BACKLOG 5
30 /* Define to use the SO_LINGER option so that the server sockets can be resused
31 * sooner. Disabled for now because it is not very well implemented by various
32 * OSes and it causes clients to fail to read the TEARDOWN response. */
47 G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
49 GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
50 #define GST_CAT_DEFAULT rtsp_server_debug
52 static void gst_rtsp_server_get_property (GObject * object, guint propid,
53 GValue * value, GParamSpec * pspec);
54 static void gst_rtsp_server_set_property (GObject * object, guint propid,
55 const GValue * value, GParamSpec * pspec);
56 static void gst_rtsp_server_finalize (GObject * object);
58 static GstRTSPClient *default_accept_client (GstRTSPServer * server,
59 GIOChannel * channel);
62 gst_rtsp_server_class_init (GstRTSPServerClass * klass)
64 GObjectClass *gobject_class;
66 gobject_class = G_OBJECT_CLASS (klass);
68 gobject_class->get_property = gst_rtsp_server_get_property;
69 gobject_class->set_property = gst_rtsp_server_set_property;
70 gobject_class->finalize = gst_rtsp_server_finalize;
73 * GstRTSPServer::address
75 * The address of the server. This is the address where the server will
78 g_object_class_install_property (gobject_class, PROP_ADDRESS,
79 g_param_spec_string ("address", "Address",
80 "The address the server uses to listen on", DEFAULT_ADDRESS,
81 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
83 * GstRTSPServer::service
85 * The service of the server. This is either a string with the service name or
86 * a port number (as a string) the server will listen on.
88 g_object_class_install_property (gobject_class, PROP_SERVICE,
89 g_param_spec_string ("service", "Service",
90 "The service or port number the server uses to listen on",
91 DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
93 * GstRTSPServer::backlog
95 * The backlog argument defines the maximum length to which the queue of
96 * pending connections for the server may grow. If a connection request arrives
97 * when the queue is full, the client may receive an error with an indication of
98 * ECONNREFUSED or, if the underlying protocol supports retransmission, the
99 * request may be ignored so that a later reattempt at connection succeeds.
101 g_object_class_install_property (gobject_class, PROP_BACKLOG,
102 g_param_spec_int ("backlog", "Backlog",
103 "The maximum length to which the queue "
104 "of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
105 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
107 * GstRTSPServer::session-pool
109 * The session pool of the server. By default each server has a separate
110 * session pool but sessions can be shared between servers by setting the same
111 * session pool on multiple servers.
113 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
114 g_param_spec_object ("session-pool", "Session Pool",
115 "The session pool to use for client session",
116 GST_TYPE_RTSP_SESSION_POOL,
117 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
119 * GstRTSPServer::media-mapping
121 * The media mapping to use for this server. By default the server has no
122 * media mapping and thus cannot map urls to media streams.
124 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
125 g_param_spec_object ("media-mapping", "Media Mapping",
126 "The media mapping to use for client session",
127 GST_TYPE_RTSP_MEDIA_MAPPING,
128 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
130 klass->accept_client = default_accept_client;
132 GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
136 gst_rtsp_server_init (GstRTSPServer * server)
138 server->address = g_strdup (DEFAULT_ADDRESS);
139 server->service = g_strdup (DEFAULT_SERVICE);
140 server->backlog = DEFAULT_BACKLOG;
141 server->session_pool = gst_rtsp_session_pool_new ();
142 server->media_mapping = gst_rtsp_media_mapping_new ();
146 gst_rtsp_server_finalize (GObject * object)
148 GstRTSPServer *server = GST_RTSP_SERVER (object);
150 g_free (server->address);
151 g_free (server->service);
153 g_object_unref (server->session_pool);
154 g_object_unref (server->media_mapping);
158 * gst_rtsp_server_new:
160 * Create a new #GstRTSPServer instance.
163 gst_rtsp_server_new (void)
165 GstRTSPServer *result;
167 result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
173 * gst_rtsp_server_set_address:
174 * @server: a #GstRTSPServer
175 * @address: the address
177 * Configure @server to accept connections on the given address.
179 * This function must be called before the server is bound.
182 gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
184 g_return_if_fail (GST_IS_RTSP_SERVER (server));
185 g_return_if_fail (address != NULL);
187 g_free (server->address);
188 server->address = g_strdup (address);
192 * gst_rtsp_server_get_address:
193 * @server: a #GstRTSPServer
195 * Get the address on which the server will accept connections.
197 * Returns: the server address. g_free() after usage.
200 gst_rtsp_server_get_address (GstRTSPServer * server)
202 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
204 return g_strdup (server->address);
208 * gst_rtsp_server_set_service:
209 * @server: a #GstRTSPServer
210 * @service: the service
212 * Configure @server to accept connections on the given service.
213 * @service should be a string containing the service name (see services(5)) or
214 * a string containing a port number between 1 and 65535.
216 * This function must be called before the server is bound.
219 gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
221 g_return_if_fail (GST_IS_RTSP_SERVER (server));
222 g_return_if_fail (service != NULL);
224 g_free (server->service);
225 server->service = g_strdup (service);
229 * gst_rtsp_server_get_service:
230 * @server: a #GstRTSPServer
232 * Get the service on which the server will accept connections.
234 * Returns: the service. use g_free() after usage.
237 gst_rtsp_server_get_service (GstRTSPServer * server)
239 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
241 return g_strdup (server->service);
245 * gst_rtsp_server_set_backlog:
246 * @server: a #GstRTSPServer
247 * @backlog: the backlog
249 * configure the maximum amount of requests that may be queued for the
252 * This function must be called before the server is bound.
255 gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
257 g_return_if_fail (GST_IS_RTSP_SERVER (server));
259 server->backlog = backlog;
263 * gst_rtsp_server_get_backlog:
264 * @server: a #GstRTSPServer
266 * The maximum amount of queued requests for the server.
268 * Returns: the server backlog.
271 gst_rtsp_server_get_backlog (GstRTSPServer * server)
273 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
275 return server->backlog;
279 * gst_rtsp_server_set_session_pool:
280 * @server: a #GstRTSPServer
281 * @pool: a #GstRTSPSessionPool
283 * configure @pool to be used as the session pool of @server.
286 gst_rtsp_server_set_session_pool (GstRTSPServer * server,
287 GstRTSPSessionPool * pool)
289 GstRTSPSessionPool *old;
291 g_return_if_fail (GST_IS_RTSP_SERVER (server));
293 old = server->session_pool;
298 server->session_pool = pool;
300 g_object_unref (old);
305 * gst_rtsp_server_get_session_pool:
306 * @server: a #GstRTSPServer
308 * Get the #GstRTSPSessionPool used as the session pool of @server.
310 * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
314 gst_rtsp_server_get_session_pool (GstRTSPServer * server)
316 GstRTSPSessionPool *result;
318 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
320 if ((result = server->session_pool))
321 g_object_ref (result);
327 * gst_rtsp_server_set_media_mapping:
328 * @server: a #GstRTSPServer
329 * @mapping: a #GstRTSPMediaMapping
331 * configure @mapping to be used as the media mapping of @server.
334 gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
335 GstRTSPMediaMapping * mapping)
337 GstRTSPMediaMapping *old;
339 g_return_if_fail (GST_IS_RTSP_SERVER (server));
341 old = server->media_mapping;
343 if (old != mapping) {
345 g_object_ref (mapping);
346 server->media_mapping = mapping;
348 g_object_unref (old);
354 * gst_rtsp_server_get_media_mapping:
355 * @server: a #GstRTSPServer
357 * Get the #GstRTSPMediaMapping used as the media mapping of @server.
359 * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
362 GstRTSPMediaMapping *
363 gst_rtsp_server_get_media_mapping (GstRTSPServer * server)
365 GstRTSPMediaMapping *result;
367 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
369 if ((result = server->media_mapping))
370 g_object_ref (result);
376 * gst_rtsp_server_set_auth:
377 * @server: a #GstRTSPServer
378 * @auth: a #GstRTSPAuth
380 * configure @auth to be used as the authentication manager of @server.
383 gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
387 g_return_if_fail (GST_IS_RTSP_SERVER (server));
396 g_object_unref (old);
402 * gst_rtsp_server_get_auth:
403 * @server: a #GstRTSPServer
405 * Get the #GstRTSPAuth used as the authentication manager of @server.
407 * Returns: the #GstRTSPAuth of @server. g_object_unref() after
411 gst_rtsp_server_get_auth (GstRTSPServer * server)
415 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
417 if ((result = server->auth))
418 g_object_ref (result);
424 gst_rtsp_server_get_property (GObject * object, guint propid,
425 GValue * value, GParamSpec * pspec)
427 GstRTSPServer *server = GST_RTSP_SERVER (object);
431 g_value_take_string (value, gst_rtsp_server_get_address (server));
434 g_value_take_string (value, gst_rtsp_server_get_service (server));
437 g_value_set_int (value, gst_rtsp_server_get_backlog (server));
439 case PROP_SESSION_POOL:
440 g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
442 case PROP_MEDIA_MAPPING:
443 g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
446 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
451 gst_rtsp_server_set_property (GObject * object, guint propid,
452 const GValue * value, GParamSpec * pspec)
454 GstRTSPServer *server = GST_RTSP_SERVER (object);
458 gst_rtsp_server_set_address (server, g_value_get_string (value));
461 gst_rtsp_server_set_service (server, g_value_get_string (value));
464 gst_rtsp_server_set_backlog (server, g_value_get_int (value));
466 case PROP_SESSION_POOL:
467 gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
469 case PROP_MEDIA_MAPPING:
470 gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
473 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
477 /* Prepare a server socket for @server and make it listen on the configured port */
479 gst_rtsp_server_sink_init_send (GstRTSPServer * server)
482 struct addrinfo hints;
483 struct addrinfo *result, *rp;
485 struct linger linger;
488 memset (&hints, 0, sizeof (struct addrinfo));
489 hints.ai_family = AF_UNSPEC; /* Allow IPv4 or IPv6 */
490 hints.ai_socktype = SOCK_STREAM; /* stream socket */
491 hints.ai_flags = AI_PASSIVE | AI_CANONNAME; /* For wildcard IP address */
492 hints.ai_protocol = 0; /* Any protocol */
493 hints.ai_canonname = NULL;
494 hints.ai_addr = NULL;
495 hints.ai_next = NULL;
497 GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
500 /* resolve the server IP address */
502 getaddrinfo (server->address, server->service, &hints, &result)) != 0)
505 /* create server socket, we loop through all the addresses until we manage to
506 * create a socket and bind. */
507 for (rp = result; rp; rp = rp->ai_next) {
508 sockfd = socket (rp->ai_family, rp->ai_socktype, rp->ai_protocol);
510 GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
515 /* make address reusable */
517 if (setsockopt (sockfd, SOL_SOCKET, SO_REUSEADDR,
518 (void *) &ret, sizeof (ret)) < 0) {
519 /* warn but try to bind anyway */
520 GST_WARNING_OBJECT (server, "failed to reuse socker (%s)",
524 if (bind (sockfd, rp->ai_addr, rp->ai_addrlen) == 0) {
525 GST_DEBUG_OBJECT (server, "bind on %s", rp->ai_canonname);
529 GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
533 freeaddrinfo (result);
538 server->server_sock.fd = sockfd;
540 GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
541 server->server_sock.fd);
543 /* keep connection alive; avoids SIGPIPE during write */
545 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
546 (void *) &ret, sizeof (ret)) < 0)
547 goto keepalive_failed;
550 /* make sure socket is reset 5 seconds after close. This ensure that we can
551 * reuse the socket quickly while still having a chance to send data to the
555 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_LINGER,
556 (void *) &linger, sizeof (linger)) < 0)
560 /* set the server socket to nonblocking */
561 fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
563 GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
564 server->server_sock.fd, server->backlog);
565 if (listen (server->server_sock.fd, server->backlog) == -1)
568 GST_DEBUG_OBJECT (server,
569 "listened on server socket %d, returning from connection setup",
570 server->server_sock.fd);
572 GST_INFO_OBJECT (server, "listening on service %s", server->service);
579 GST_ERROR_OBJECT (server, "failed to resolve address: %s",
585 GST_ERROR_OBJECT (server, "failed to create socket: %s",
591 GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s",
598 GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
605 GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
611 if (server->server_sock.fd >= 0) {
612 close (server->server_sock.fd);
613 server->server_sock.fd = -1;
619 /* default method for creating a new client object in the server to accept and
620 * handle a client connection on this server */
621 static GstRTSPClient *
622 default_accept_client (GstRTSPServer * server, GIOChannel * channel)
624 GstRTSPClient *client;
626 /* a new client connected, create a session to handle the client. */
627 client = gst_rtsp_client_new ();
629 /* set the session pool that this client should use */
630 gst_rtsp_client_set_session_pool (client, server->session_pool);
631 /* set the media mapping that this client should use */
632 gst_rtsp_client_set_media_mapping (client, server->media_mapping);
633 /* set authentication manager */
634 gst_rtsp_client_set_auth (client, server->auth);
636 /* accept connections for that client, this function returns after accepting
637 * the connection and will run the remainder of the communication with the
638 * client asyncronously. */
639 if (!gst_rtsp_client_accept (client, channel))
647 GST_ERROR_OBJECT (server,
648 "Could not accept client on server socket %d: %s (%d)",
649 server->server_sock.fd, g_strerror (errno), errno);
650 gst_object_unref (client);
656 * gst_rtsp_server_io_func:
657 * @channel: a #GIOChannel
658 * @condition: the condition on @source
660 * A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
661 * new connection on @channel or @server.
663 * Returns: TRUE if the source could be connected, FALSE if an error occured.
666 gst_rtsp_server_io_func (GIOChannel * channel, GIOCondition condition,
667 GstRTSPServer * server)
669 GstRTSPClient *client = NULL;
670 GstRTSPServerClass *klass;
672 if (condition & G_IO_IN) {
673 klass = GST_RTSP_SERVER_GET_CLASS (server);
675 /* a new client connected, create a client object to handle the client. */
676 if (klass->accept_client)
677 client = klass->accept_client (server, channel);
681 /* can unref the client now, when the request is finished, it will be
683 gst_object_unref (client);
685 GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
692 GST_ERROR_OBJECT (server, "failed to create a client");
698 * gst_rtsp_server_get_io_channel:
699 * @server: a #GstRTSPServer
701 * Create a #GIOChannel for @server.
703 * Returns: the GIOChannel for @server or NULL when an error occured.
706 gst_rtsp_server_get_io_channel (GstRTSPServer * server)
708 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
710 if (server->io_channel == NULL) {
711 if (!gst_rtsp_server_sink_init_send (server))
714 /* create IO channel for the socket */
715 server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
717 return server->io_channel;
721 GST_ERROR_OBJECT (server, "failed to initialize server");
727 * gst_rtsp_server_create_watch:
728 * @server: a #GstRTSPServer
730 * Create a #GSource for @server. The new source will have a default
731 * #GIOFunc of gst_rtsp_server_io_func().
733 * Returns: the #GSource for @server or NULL when an error occured.
736 gst_rtsp_server_create_watch (GstRTSPServer * server)
738 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
740 if (server->io_watch == NULL) {
743 channel = gst_rtsp_server_get_io_channel (server);
747 /* create a watch for reads (new connections) and possible errors */
748 server->io_watch = g_io_create_watch (channel, G_IO_IN |
749 G_IO_ERR | G_IO_HUP | G_IO_NVAL);
751 /* configure the callback */
752 g_source_set_callback (server->io_watch,
753 (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
755 return server->io_watch;
759 GST_ERROR_OBJECT (server, "failed to create IO channel");
765 * gst_rtsp_server_attach:
766 * @server: a #GstRTSPServer
767 * @context: a #GMainContext
769 * Attaches @server to @context. When the mainloop for @context is run, the
770 * server will be dispatched.
772 * This function should be called when the server properties and urls are fully
773 * configured and the server is ready to start.
775 * Returns: the ID (greater than 0) for the source within the GMainContext.
778 gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
783 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
785 source = gst_rtsp_server_create_watch (server);
789 res = g_source_attach (source, context);
796 GST_ERROR_OBJECT (server, "failed to create watch");