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 "rtsp-server.h"
21 #include "rtsp-client.h"
23 #define DEFAULT_ADDRESS "0.0.0.0"
24 /* #define DEFAULT_ADDRESS "::0" */
25 #define DEFAULT_SERVICE "8554"
26 #define DEFAULT_BACKLOG 5
28 /* Define to use the SO_LINGER option so that the server sockets can be resused
29 * sooner. Disabled for now because it is not very well implemented by various
30 * OSes and it causes clients to fail to read the TEARDOWN response. */
45 G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
47 GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
48 #define GST_CAT_DEFAULT rtsp_server_debug
50 static void gst_rtsp_server_get_property (GObject * object, guint propid,
51 GValue * value, GParamSpec * pspec);
52 static void gst_rtsp_server_set_property (GObject * object, guint propid,
53 const GValue * value, GParamSpec * pspec);
54 static void gst_rtsp_server_finalize (GObject * object);
56 static GstRTSPClient *default_accept_client (GstRTSPServer * server,
57 GIOChannel * channel);
60 gst_rtsp_server_class_init (GstRTSPServerClass * klass)
62 GObjectClass *gobject_class;
64 gobject_class = G_OBJECT_CLASS (klass);
66 gobject_class->get_property = gst_rtsp_server_get_property;
67 gobject_class->set_property = gst_rtsp_server_set_property;
68 gobject_class->finalize = gst_rtsp_server_finalize;
71 * GstRTSPServer::address
73 * The address of the server. This is the address where the server will
76 g_object_class_install_property (gobject_class, PROP_ADDRESS,
77 g_param_spec_string ("address", "Address",
78 "The address the server uses to listen on", DEFAULT_ADDRESS,
79 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
81 * GstRTSPServer::service
83 * The service of the server. This is either a string with the service name or
84 * a port number (as a string) the server will listen on.
86 g_object_class_install_property (gobject_class, PROP_SERVICE,
87 g_param_spec_string ("service", "Service",
88 "The service or port number the server uses to listen on",
89 DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
91 * GstRTSPServer::backlog
93 * The backlog argument defines the maximum length to which the queue of
94 * pending connections for the server may grow. If a connection request arrives
95 * when the queue is full, the client may receive an error with an indication of
96 * ECONNREFUSED or, if the underlying protocol supports retransmission, the
97 * request may be ignored so that a later reattempt at connection succeeds.
99 g_object_class_install_property (gobject_class, PROP_BACKLOG,
100 g_param_spec_int ("backlog", "Backlog",
101 "The maximum length to which the queue "
102 "of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
103 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
105 * GstRTSPServer::session-pool
107 * The session pool of the server. By default each server has a separate
108 * session pool but sessions can be shared between servers by setting the same
109 * session pool on multiple servers.
111 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
112 g_param_spec_object ("session-pool", "Session Pool",
113 "The session pool to use for client session",
114 GST_TYPE_RTSP_SESSION_POOL,
115 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
117 * GstRTSPServer::media-mapping
119 * The media mapping to use for this server. By default the server has no
120 * media mapping and thus cannot map urls to media streams.
122 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
123 g_param_spec_object ("media-mapping", "Media Mapping",
124 "The media mapping to use for client session",
125 GST_TYPE_RTSP_MEDIA_MAPPING,
126 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
128 klass->accept_client = default_accept_client;
130 GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
134 gst_rtsp_server_init (GstRTSPServer * server)
136 server->address = g_strdup (DEFAULT_ADDRESS);
137 server->service = g_strdup (DEFAULT_SERVICE);
138 server->backlog = DEFAULT_BACKLOG;
139 server->session_pool = gst_rtsp_session_pool_new ();
140 server->media_mapping = gst_rtsp_media_mapping_new ();
144 gst_rtsp_server_finalize (GObject * object)
146 GstRTSPServer *server = GST_RTSP_SERVER (object);
148 g_free (server->address);
149 g_free (server->service);
151 g_object_unref (server->session_pool);
152 g_object_unref (server->media_mapping);
156 * gst_rtsp_server_new:
158 * Create a new #GstRTSPServer instance.
161 gst_rtsp_server_new (void)
163 GstRTSPServer *result;
165 result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
171 * gst_rtsp_server_set_address:
172 * @server: a #GstRTSPServer
173 * @address: the address
175 * Configure @server to accept connections on the given address.
177 * This function must be called before the server is bound.
180 gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
182 g_return_if_fail (GST_IS_RTSP_SERVER (server));
183 g_return_if_fail (address != NULL);
185 g_free (server->address);
186 server->address = g_strdup (address);
190 * gst_rtsp_server_get_address:
191 * @server: a #GstRTSPServer
193 * Get the address on which the server will accept connections.
195 * Returns: the server address. g_free() after usage.
198 gst_rtsp_server_get_address (GstRTSPServer * server)
200 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
202 return g_strdup (server->address);
206 * gst_rtsp_server_set_service:
207 * @server: a #GstRTSPServer
208 * @service: the service
210 * Configure @server to accept connections on the given service.
211 * @service should be a string containing the service name (see services(5)) or
212 * a string containing a port number between 1 and 65535.
214 * This function must be called before the server is bound.
217 gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
219 g_return_if_fail (GST_IS_RTSP_SERVER (server));
220 g_return_if_fail (service != NULL);
222 g_free (server->service);
223 server->service = g_strdup (service);
227 * gst_rtsp_server_get_service:
228 * @server: a #GstRTSPServer
230 * Get the service on which the server will accept connections.
232 * Returns: the service. use g_free() after usage.
235 gst_rtsp_server_get_service (GstRTSPServer * server)
237 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
239 return g_strdup (server->service);
243 * gst_rtsp_server_set_backlog:
244 * @server: a #GstRTSPServer
245 * @backlog: the backlog
247 * configure the maximum amount of requests that may be queued for the
250 * This function must be called before the server is bound.
253 gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
255 g_return_if_fail (GST_IS_RTSP_SERVER (server));
257 server->backlog = backlog;
261 * gst_rtsp_server_get_backlog:
262 * @server: a #GstRTSPServer
264 * The maximum amount of queued requests for the server.
266 * Returns: the server backlog.
269 gst_rtsp_server_get_backlog (GstRTSPServer * server)
271 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
273 return server->backlog;
277 * gst_rtsp_server_set_session_pool:
278 * @server: a #GstRTSPServer
279 * @pool: a #GstRTSPSessionPool
281 * configure @pool to be used as the session pool of @server.
284 gst_rtsp_server_set_session_pool (GstRTSPServer * server,
285 GstRTSPSessionPool * pool)
287 GstRTSPSessionPool *old;
289 g_return_if_fail (GST_IS_RTSP_SERVER (server));
291 old = server->session_pool;
296 server->session_pool = pool;
298 g_object_unref (old);
303 * gst_rtsp_server_get_session_pool:
304 * @server: a #GstRTSPServer
306 * Get the #GstRTSPSessionPool used as the session pool of @server.
308 * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
312 gst_rtsp_server_get_session_pool (GstRTSPServer * server)
314 GstRTSPSessionPool *result;
316 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
318 if ((result = server->session_pool))
319 g_object_ref (result);
325 * gst_rtsp_server_set_media_mapping:
326 * @server: a #GstRTSPServer
327 * @mapping: a #GstRTSPMediaMapping
329 * configure @mapping to be used as the media mapping of @server.
332 gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
333 GstRTSPMediaMapping * mapping)
335 GstRTSPMediaMapping *old;
337 g_return_if_fail (GST_IS_RTSP_SERVER (server));
339 old = server->media_mapping;
341 if (old != mapping) {
343 g_object_ref (mapping);
344 server->media_mapping = mapping;
346 g_object_unref (old);
352 * gst_rtsp_server_get_media_mapping:
353 * @server: a #GstRTSPServer
355 * Get the #GstRTSPMediaMapping used as the media mapping of @server.
357 * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
360 GstRTSPMediaMapping *
361 gst_rtsp_server_get_media_mapping (GstRTSPServer * server)
363 GstRTSPMediaMapping *result;
365 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
367 if ((result = server->media_mapping))
368 g_object_ref (result);
374 gst_rtsp_server_get_property (GObject * object, guint propid,
375 GValue * value, GParamSpec * pspec)
377 GstRTSPServer *server = GST_RTSP_SERVER (object);
381 g_value_take_string (value, gst_rtsp_server_get_address (server));
384 g_value_take_string (value, gst_rtsp_server_get_service (server));
387 g_value_set_int (value, gst_rtsp_server_get_backlog (server));
389 case PROP_SESSION_POOL:
390 g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
392 case PROP_MEDIA_MAPPING:
393 g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
396 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
401 gst_rtsp_server_set_property (GObject * object, guint propid,
402 const GValue * value, GParamSpec * pspec)
404 GstRTSPServer *server = GST_RTSP_SERVER (object);
408 gst_rtsp_server_set_address (server, g_value_get_string (value));
411 gst_rtsp_server_set_service (server, g_value_get_string (value));
414 gst_rtsp_server_set_backlog (server, g_value_get_int (value));
416 case PROP_SESSION_POOL:
417 gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
419 case PROP_MEDIA_MAPPING:
420 gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
423 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
427 /* Prepare a server socket for @server and make it listen on the configured port */
429 gst_rtsp_server_sink_init_send (GstRTSPServer * server)
432 struct addrinfo hints;
433 struct addrinfo *result, *rp;
435 struct linger linger;
438 memset (&hints, 0, sizeof (struct addrinfo));
439 hints.ai_family = AF_UNSPEC; /* Allow IPv4 or IPv6 */
440 hints.ai_socktype = SOCK_STREAM; /* stream socket */
441 hints.ai_flags = AI_PASSIVE | AI_CANONNAME; /* For wildcard IP address */
442 hints.ai_protocol = 0; /* Any protocol */
443 hints.ai_canonname = NULL;
444 hints.ai_addr = NULL;
445 hints.ai_next = NULL;
447 GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
450 /* resolve the server IP address */
452 getaddrinfo (server->address, server->service, &hints, &result)) != 0)
455 /* create server socket, we loop through all the addresses until we manage to
456 * create a socket and bind. */
457 for (rp = result; rp; rp = rp->ai_next) {
458 sockfd = socket (rp->ai_family, rp->ai_socktype, rp->ai_protocol);
460 GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
465 /* make address reusable */
467 if (setsockopt (sockfd, SOL_SOCKET, SO_REUSEADDR,
468 (void *) &ret, sizeof (ret)) < 0) {
469 /* warn but try to bind anyway */
470 GST_WARNING_OBJECT (server, "failed to reuse socker (%s)",
474 if (bind (sockfd, rp->ai_addr, rp->ai_addrlen) == 0) {
475 GST_DEBUG_OBJECT (server, "bind on %s", rp->ai_canonname);
479 GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
483 freeaddrinfo (result);
488 server->server_sock.fd = sockfd;
490 GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
491 server->server_sock.fd);
493 /* keep connection alive; avoids SIGPIPE during write */
495 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
496 (void *) &ret, sizeof (ret)) < 0)
497 goto keepalive_failed;
500 /* make sure socket is reset 5 seconds after close. This ensure that we can
501 * reuse the socket quickly while still having a chance to send data to the
505 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_LINGER,
506 (void *) &linger, sizeof (linger)) < 0)
510 /* set the server socket to nonblocking */
511 fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
513 GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
514 server->server_sock.fd, server->backlog);
515 if (listen (server->server_sock.fd, server->backlog) == -1)
518 GST_DEBUG_OBJECT (server,
519 "listened on server socket %d, returning from connection setup",
520 server->server_sock.fd);
522 GST_INFO_OBJECT (server, "listening on service %s", server->service);
529 GST_ERROR_OBJECT (server, "failed to resolve address: %s",
535 GST_ERROR_OBJECT (server, "failed to create socket: %s",
541 GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s",
548 GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
555 GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
561 if (server->server_sock.fd >= 0) {
562 close (server->server_sock.fd);
563 server->server_sock.fd = -1;
569 /* default method for creating a new client object in the server to accept and
570 * handle a client connection on this server */
571 static GstRTSPClient *
572 default_accept_client (GstRTSPServer * server, GIOChannel * channel)
574 GstRTSPClient *client;
576 /* a new client connected, create a session to handle the client. */
577 client = gst_rtsp_client_new ();
579 /* set the session pool that this client should use */
580 gst_rtsp_client_set_session_pool (client, server->session_pool);
581 /* set the media mapping that this client should use */
582 gst_rtsp_client_set_media_mapping (client, server->media_mapping);
584 /* accept connections for that client, this function returns after accepting
585 * the connection and will run the remainder of the communication with the
586 * client asyncronously. */
587 if (!gst_rtsp_client_accept (client, channel))
595 GST_ERROR_OBJECT (server,
596 "Could not accept client on server socket %d: %s (%d)",
597 server->server_sock.fd, g_strerror (errno), errno);
598 gst_object_unref (client);
604 * gst_rtsp_server_io_func:
605 * @channel: a #GIOChannel
606 * @condition: the condition on @source
608 * A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
609 * new connection on @channel or @server.
611 * Returns: TRUE if the source could be connected, FALSE if an error occured.
614 gst_rtsp_server_io_func (GIOChannel * channel, GIOCondition condition,
615 GstRTSPServer * server)
617 GstRTSPClient *client = NULL;
618 GstRTSPServerClass *klass;
620 if (condition & G_IO_IN) {
621 klass = GST_RTSP_SERVER_GET_CLASS (server);
623 /* a new client connected, create a client object to handle the client. */
624 if (klass->accept_client)
625 client = klass->accept_client (server, channel);
629 /* can unref the client now, when the request is finished, it will be
631 gst_object_unref (client);
633 GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
640 GST_ERROR_OBJECT (server, "failed to create a client");
646 * gst_rtsp_server_get_io_channel:
647 * @server: a #GstRTSPServer
649 * Create a #GIOChannel for @server.
651 * Returns: the GIOChannel for @server or NULL when an error occured.
654 gst_rtsp_server_get_io_channel (GstRTSPServer * server)
656 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
658 if (server->io_channel == NULL) {
659 if (!gst_rtsp_server_sink_init_send (server))
662 /* create IO channel for the socket */
663 server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
665 return server->io_channel;
669 GST_ERROR_OBJECT (server, "failed to initialize server");
675 * gst_rtsp_server_create_watch:
676 * @server: a #GstRTSPServer
678 * Create a #GSource for @server. The new source will have a default
679 * #GIOFunc of gst_rtsp_server_io_func().
681 * Returns: the #GSource for @server or NULL when an error occured.
684 gst_rtsp_server_create_watch (GstRTSPServer * server)
686 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
688 if (server->io_watch == NULL) {
691 channel = gst_rtsp_server_get_io_channel (server);
695 /* create a watch for reads (new connections) and possible errors */
696 server->io_watch = g_io_create_watch (channel, G_IO_IN |
697 G_IO_ERR | G_IO_HUP | G_IO_NVAL);
699 /* configure the callback */
700 g_source_set_callback (server->io_watch,
701 (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
703 return server->io_watch;
707 GST_ERROR_OBJECT (server, "failed to create IO channel");
713 * gst_rtsp_server_attach:
714 * @server: a #GstRTSPServer
715 * @context: a #GMainContext
717 * Attaches @server to @context. When the mainloop for @context is run, the
718 * server will be dispatched.
720 * This function should be called when the server properties and urls are fully
721 * configured and the server is ready to start.
723 * Returns: the ID (greater than 0) for the source within the GMainContext.
726 gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
731 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
733 source = gst_rtsp_server_create_watch (server);
737 res = g_source_attach (source, context);
744 GST_ERROR_OBJECT (server, "failed to create watch");