/* GStreamer * Copyright (C) 2008 Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:rtsp-client * @short_description: A client connection state * @see_also: #GstRTSPServer, #GstRTSPThreadPool * * The client object handles the connection with a client for as long as a TCP * connection is open. * * A #GstRTSPClient is created by #GstRTSPServer when a new connection is * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool, * #GstRTSPAuth and #GstRTSPThreadPool from the server. * * The client connection should be configured with the #GstRTSPConnection using * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext * using gst_rtsp_client_attach(). From then on the client will handle requests * on the connection. * * Use gst_rtsp_client_session_filter() to iterate or modify all the * #GstRTSPSession objects managed by the client object. * * Last reviewed on 2013-07-11 (1.0.0) */ #include #include #include #include "rtsp-client.h" #include "rtsp-sdp.h" #include "rtsp-params.h" #define GST_RTSP_CLIENT_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate)) /* locking order: * send_lock, lock, tunnels_lock */ struct _GstRTSPClientPrivate { GMutex lock; /* protects everything else */ GMutex send_lock; GstRTSPConnection *connection; GstRTSPWatch *watch; GMainContext *watch_context; guint close_seq; gchar *server_ip; gboolean is_ipv6; GstRTSPClientSendFunc send_func; /* protected by send_lock */ gpointer send_data; /* protected by send_lock */ GDestroyNotify send_notify; /* protected by send_lock */ GstRTSPSessionPool *session_pool; GstRTSPMountPoints *mount_points; GstRTSPAuth *auth; GstRTSPThreadPool *thread_pool; /* used to cache the media in the last requested DESCRIBE so that * we can pick it up in the next SETUP immediately */ gchar *path; GstRTSPMedia *media; GList *transports; GList *sessions; gboolean drop_backlog; }; static GMutex tunnels_lock; static GHashTable *tunnels; /* protected by tunnels_lock */ #define DEFAULT_SESSION_POOL NULL #define DEFAULT_MOUNT_POINTS NULL #define DEFAULT_DROP_BACKLOG TRUE enum { PROP_0, PROP_SESSION_POOL, PROP_MOUNT_POINTS, PROP_DROP_BACKLOG, PROP_LAST }; enum { SIGNAL_CLOSED, SIGNAL_NEW_SESSION, SIGNAL_OPTIONS_REQUEST, SIGNAL_DESCRIBE_REQUEST, SIGNAL_SETUP_REQUEST, SIGNAL_PLAY_REQUEST, SIGNAL_PAUSE_REQUEST, SIGNAL_TEARDOWN_REQUEST, SIGNAL_SET_PARAMETER_REQUEST, SIGNAL_GET_PARAMETER_REQUEST, SIGNAL_HANDLE_RESPONSE, SIGNAL_SEND_MESSAGE, SIGNAL_LAST }; GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug); #define GST_CAT_DEFAULT rtsp_client_debug static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 }; static void gst_rtsp_client_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec); static void gst_rtsp_client_set_property (GObject * object, guint propid, const GValue * value, GParamSpec * pspec); static void gst_rtsp_client_finalize (GObject * obj); static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media); static void client_session_finalized (GstRTSPClient * client, GstRTSPSession * session); static void unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session, GstRTSPSessionMedia * sessmedia); static gboolean default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx); static gboolean default_configure_client_transport (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPTransport * ct); static GstRTSPResult default_params_set (GstRTSPClient * client, GstRTSPContext * ctx); static GstRTSPResult default_params_get (GstRTSPClient * client, GstRTSPContext * ctx); static gchar *default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri); G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT); static void gst_rtsp_client_class_init (GstRTSPClientClass * klass) { GObjectClass *gobject_class; g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate)); gobject_class = G_OBJECT_CLASS (klass); gobject_class->get_property = gst_rtsp_client_get_property; gobject_class->set_property = gst_rtsp_client_set_property; gobject_class->finalize = gst_rtsp_client_finalize; klass->create_sdp = create_sdp; klass->configure_client_media = default_configure_client_media; klass->configure_client_transport = default_configure_client_transport; klass->params_set = default_params_set; klass->params_get = default_params_get; klass->make_path_from_uri = default_make_path_from_uri; g_object_class_install_property (gobject_class, PROP_SESSION_POOL, g_param_spec_object ("session-pool", "Session Pool", "The session pool to use for client session", GST_TYPE_RTSP_SESSION_POOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS, g_param_spec_object ("mount-points", "Mount Points", "The mount points to use for client session", GST_TYPE_RTSP_MOUNT_POINTS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG, g_param_spec_boolean ("drop-backlog", "Drop Backlog", "Drop data when the backlog queue is full", DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_rtsp_client_signals[SIGNAL_CLOSED] = g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); gst_rtsp_client_signals[SIGNAL_NEW_SESSION] = g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION); gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] = g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] = g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] = g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] = g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] = g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] = g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] = g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] = g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] = g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, handle_response), NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1, G_TYPE_POINTER); gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] = g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_POINTER, G_TYPE_POINTER); tunnels = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref); g_mutex_init (&tunnels_lock); GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient"); } static void gst_rtsp_client_init (GstRTSPClient * client) { GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client); client->priv = priv; g_mutex_init (&priv->lock); g_mutex_init (&priv->send_lock); priv->close_seq = 0; priv->drop_backlog = DEFAULT_DROP_BACKLOG; } static GstRTSPFilterResult filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL); unlink_session_transports (client, sess, sessmedia); /* unmanage the media in the session */ return GST_RTSP_FILTER_REMOVE; } static void client_unlink_session (GstRTSPClient * client, GstRTSPSession * session) { /* unlink all media managed in this session */ gst_rtsp_session_filter (session, filter_session, client); } static void client_watch_session (GstRTSPClient * client, GstRTSPSession * session) { GstRTSPClientPrivate *priv = client->priv; GList *walk; for (walk = priv->sessions; walk; walk = g_list_next (walk)) { GstRTSPSession *msession = (GstRTSPSession *) walk->data; /* we already know about this session */ if (msession == session) return; } GST_INFO ("watching session %p", session); g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client); priv->sessions = g_list_prepend (priv->sessions, session); } static void client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session) { GstRTSPClientPrivate *priv = client->priv; GST_INFO ("unwatching session %p", session); g_object_weak_unref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client); priv->sessions = g_list_remove (priv->sessions, session); } static void client_cleanup_session (GstRTSPClient * client, GstRTSPSession * session) { g_object_weak_unref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client); client_unlink_session (client, session); } static void client_cleanup_sessions (GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; GList *sessions; /* remove weak-ref from sessions */ for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) { client_cleanup_session (client, (GstRTSPSession *) sessions->data); } g_list_free (priv->sessions); priv->sessions = NULL; } /* A client is finalized when the connection is broken */ static void gst_rtsp_client_finalize (GObject * obj) { GstRTSPClient *client = GST_RTSP_CLIENT (obj); GstRTSPClientPrivate *priv = client->priv; GST_INFO ("finalize client %p", client); if (priv->watch) gst_rtsp_watch_set_flushing (priv->watch, TRUE); gst_rtsp_client_set_send_func (client, NULL, NULL, NULL); if (priv->watch) g_source_destroy ((GSource *) priv->watch); if (priv->watch_context) g_main_context_unref (priv->watch_context); client_cleanup_sessions (client); if (priv->connection) gst_rtsp_connection_free (priv->connection); if (priv->session_pool) g_object_unref (priv->session_pool); if (priv->mount_points) g_object_unref (priv->mount_points); if (priv->auth) g_object_unref (priv->auth); if (priv->thread_pool) g_object_unref (priv->thread_pool); if (priv->path) g_free (priv->path); if (priv->media) { gst_rtsp_media_unprepare (priv->media); g_object_unref (priv->media); } g_free (priv->server_ip); g_mutex_clear (&priv->lock); g_mutex_clear (&priv->send_lock); G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj); } static void gst_rtsp_client_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec) { GstRTSPClient *client = GST_RTSP_CLIENT (object); GstRTSPClientPrivate *priv = client->priv; switch (propid) { case PROP_SESSION_POOL: g_value_take_object (value, gst_rtsp_client_get_session_pool (client)); break; case PROP_MOUNT_POINTS: g_value_take_object (value, gst_rtsp_client_get_mount_points (client)); break; case PROP_DROP_BACKLOG: g_value_set_boolean (value, priv->drop_backlog); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } static void gst_rtsp_client_set_property (GObject * object, guint propid, const GValue * value, GParamSpec * pspec) { GstRTSPClient *client = GST_RTSP_CLIENT (object); GstRTSPClientPrivate *priv = client->priv; switch (propid) { case PROP_SESSION_POOL: gst_rtsp_client_set_session_pool (client, g_value_get_object (value)); break; case PROP_MOUNT_POINTS: gst_rtsp_client_set_mount_points (client, g_value_get_object (value)); break; case PROP_DROP_BACKLOG: g_mutex_lock (&priv->lock); priv->drop_backlog = g_value_get_boolean (value); g_mutex_unlock (&priv->lock); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } /** * gst_rtsp_client_new: * * Create a new #GstRTSPClient instance. * * Returns: (transfer full): a new #GstRTSPClient */ GstRTSPClient * gst_rtsp_client_new (void) { GstRTSPClient *result; result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL); return result; } static void send_message (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMessage * message, gboolean close) { GstRTSPClientPrivate *priv = client->priv; gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER, "GStreamer RTSP server"); /* remove any previous header */ gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1); /* add the new session header for new session ids */ if (ctx->session) { gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION, gst_rtsp_session_get_header (ctx->session)); } if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) { gst_rtsp_message_dump (message); } if (close) gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close"); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE], 0, ctx, message); g_mutex_lock (&priv->send_lock); if (priv->send_func) priv->send_func (client, message, close, priv->send_data); g_mutex_unlock (&priv->send_lock); gst_rtsp_message_unset (message); } static void send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code, GstRTSPContext * ctx) { gst_rtsp_message_init_response (ctx->response, code, gst_rtsp_status_as_text (code), ctx->request); ctx->session = NULL; send_message (client, ctx, ctx->response, FALSE); } static void send_option_not_supported_response (GstRTSPClient * client, GstRTSPContext * ctx, const gchar * unsupported_options) { GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED; gst_rtsp_message_init_response (ctx->response, code, gst_rtsp_status_as_text (code), ctx->request); if (unsupported_options != NULL) { gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED, unsupported_options); } ctx->session = NULL; send_message (client, ctx, ctx->response, FALSE); } static gboolean paths_are_equal (const gchar * path1, const gchar * path2, gint len2) { if (path1 == NULL || path2 == NULL) return FALSE; if (strlen (path1) != len2) return FALSE; if (strncmp (path1, path2, len2)) return FALSE; return TRUE; } /* this function is called to initially find the media for the DESCRIBE request * but is cached for when the same client (without breaking the connection) is * doing a setup for the exact same url. */ static GstRTSPMedia * find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path, gint * matched) { GstRTSPClientPrivate *priv = client->priv; GstRTSPMediaFactory *factory; GstRTSPMedia *media; gint path_len; /* find the longest matching factory for the uri first */ if (!(factory = gst_rtsp_mount_points_match (priv->mount_points, path, matched))) goto no_factory; ctx->factory = factory; if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS)) goto no_factory_access; if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT)) goto not_authorized; if (matched) path_len = *matched; else path_len = strlen (path); if (!paths_are_equal (priv->path, path, path_len)) { GstRTSPThread *thread; /* remove any previously cached values before we try to construct a new * media for uri */ if (priv->path) g_free (priv->path); priv->path = NULL; if (priv->media) { gst_rtsp_media_unprepare (priv->media); g_object_unref (priv->media); } priv->media = NULL; /* prepare the media and add it to the pipeline */ if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri))) goto no_media; ctx->media = media; thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool, GST_RTSP_THREAD_TYPE_MEDIA, ctx); if (thread == NULL) goto no_thread; /* prepare the media */ if (!(gst_rtsp_media_prepare (media, thread))) goto no_prepare; /* now keep track of the uri and the media */ priv->path = g_strndup (path, path_len); priv->media = media; } else { /* we have seen this path before, used cached media */ media = priv->media; ctx->media = media; GST_INFO ("reusing cached media %p for path %s", media, priv->path); } g_object_unref (factory); ctx->factory = NULL; if (media) g_object_ref (media); return media; /* ERRORS */ no_factory: { GST_ERROR ("client %p: no factory for path %s", client, path); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); return NULL; } no_factory_access: { GST_ERROR ("client %p: not authorized to see factory path %s", client, path); /* error reply is already sent */ return NULL; } not_authorized: { GST_ERROR ("client %p: not authorized for factory path %s", client, path); /* error reply is already sent */ return NULL; } no_media: { GST_ERROR ("client %p: can't create media", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); g_object_unref (factory); ctx->factory = NULL; return NULL; } no_thread: { GST_ERROR ("client %p: can't create thread", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx); g_object_unref (media); ctx->media = NULL; g_object_unref (factory); ctx->factory = NULL; return NULL; } no_prepare: { GST_ERROR ("client %p: can't prepare media", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx); g_object_unref (media); ctx->media = NULL; g_object_unref (factory); ctx->factory = NULL; return NULL; } } static gboolean do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; GstRTSPMessage message = { 0 }; GstMapInfo map_info; guint8 *data; guint usize; gst_rtsp_message_init_data (&message, channel); /* FIXME, need some sort of iovec RTSPMessage here */ if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ)) return FALSE; gst_rtsp_message_take_body (&message, map_info.data, map_info.size); g_mutex_lock (&priv->send_lock); if (priv->send_func) priv->send_func (client, &message, FALSE, priv->send_data); g_mutex_unlock (&priv->send_lock); gst_rtsp_message_steal_body (&message, &data, &usize); gst_buffer_unmap (buffer, &map_info); gst_rtsp_message_unset (&message); return TRUE; } static void link_transport (GstRTSPClient * client, GstRTSPSession * session, GstRTSPStreamTransport * trans) { GstRTSPClientPrivate *priv = client->priv; GST_DEBUG ("client %p: linking transport %p", client, trans); gst_rtsp_stream_transport_set_callbacks (trans, (GstRTSPSendFunc) do_send_data, (GstRTSPSendFunc) do_send_data, client, NULL); priv->transports = g_list_prepend (priv->transports, trans); /* make sure our session can't expire */ gst_rtsp_session_prevent_expire (session); } static void link_session_transports (GstRTSPClient * client, GstRTSPSession * session, GstRTSPSessionMedia * sessmedia) { guint n_streams, i; n_streams = gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia)); for (i = 0; i < n_streams; i++) { GstRTSPStreamTransport *trans; const GstRTSPTransport *tr; /* get the transport, if there is no transport configured, skip this stream */ trans = gst_rtsp_session_media_get_transport (sessmedia, i); if (trans == NULL) continue; tr = gst_rtsp_stream_transport_get_transport (trans); if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) { /* for TCP, link the stream to the TCP connection of the client */ link_transport (client, session, trans); } } } static void unlink_transport (GstRTSPClient * client, GstRTSPSession * session, GstRTSPStreamTransport * trans) { GstRTSPClientPrivate *priv = client->priv; GST_DEBUG ("client %p: unlinking transport %p", client, trans); gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL); priv->transports = g_list_remove (priv->transports, trans); /* our session can now expire */ gst_rtsp_session_allow_expire (session); } static void unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session, GstRTSPSessionMedia * sessmedia) { guint n_streams, i; n_streams = gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia)); for (i = 0; i < n_streams; i++) { GstRTSPStreamTransport *trans; const GstRTSPTransport *tr; /* get the transport, if there is no transport configured, skip this stream */ trans = gst_rtsp_session_media_get_transport (sessmedia, i); if (trans == NULL) continue; tr = gst_rtsp_stream_transport_get_transport (trans); if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) { /* for TCP, unlink the stream from the TCP connection of the client */ unlink_transport (client, session, trans); } } } static void close_connection (GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; const gchar *tunnelid; GST_DEBUG ("client %p: closing connection", client); if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) { g_mutex_lock (&tunnels_lock); /* remove from tunnelids */ g_hash_table_remove (tunnels, tunnelid); g_mutex_unlock (&tunnels_lock); } gst_rtsp_connection_close (priv->connection); /* connection is now closed, destroy the watch which will also cause the * closed signal to be emitted */ if (priv->watch) { GST_DEBUG ("client %p: destroying watch", client); g_source_destroy ((GSource *) priv->watch); priv->watch = NULL; gst_rtsp_client_set_send_func (client, NULL, NULL, NULL); } } static gchar * default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri) { gchar *path; if (uri->query) path = g_strconcat (uri->abspath, "?", uri->query, NULL); else path = g_strdup (uri->abspath); return path; } static gboolean handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPClientPrivate *priv = client->priv; GstRTSPClientClass *klass; GstRTSPSession *session; GstRTSPSessionMedia *sessmedia; GstRTSPStatusCode code; gchar *path; gint matched; if (!ctx->session) goto no_session; session = ctx->session; if (!ctx->uri) goto no_uri; klass = GST_RTSP_CLIENT_GET_CLASS (client); path = klass->make_path_from_uri (client, ctx->uri); /* get a handle to the configuration of the media in the session */ sessmedia = gst_rtsp_session_get_media (session, path, &matched); if (!sessmedia) goto not_found; /* only aggregate control for now.. */ if (path[matched] != '\0') goto no_aggregate; g_free (path); ctx->sessmedia = sessmedia; /* we emit the signal before closing the connection */ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST], 0, ctx); /* make sure we unblock the backlog and don't accept new messages * on the watch */ gst_rtsp_watch_set_flushing (priv->watch, TRUE); /* unlink the all TCP callbacks */ unlink_session_transports (client, session, sessmedia); /* remove the session from the watched sessions */ client_unwatch_session (client, session); gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL); /* allow messages again so that we can send the reply */ gst_rtsp_watch_set_flushing (priv->watch, FALSE); /* unmanage the media in the session, returns false if all media session * are torn down. */ if (!gst_rtsp_session_release_media (session, sessmedia)) { /* remove the session */ gst_rtsp_session_pool_remove (priv->session_pool, session); } /* construct the response now */ code = GST_RTSP_STS_OK; gst_rtsp_message_init_response (ctx->response, code, gst_rtsp_status_as_text (code), ctx->request); send_message (client, ctx, ctx->response, TRUE); return TRUE; /* ERRORS */ no_session: { GST_ERROR ("client %p: no session", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx); return FALSE; } no_uri: { GST_ERROR ("client %p: no uri supplied", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } not_found: { GST_ERROR ("client %p: no media for uri", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); g_free (path); return FALSE; } no_aggregate: { GST_ERROR ("client %p: no aggregate path %s", client, path); send_generic_response (client, GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx); g_free (path); return FALSE; } } static GstRTSPResult default_params_set (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPResult res; res = gst_rtsp_params_set (client, ctx); return res; } static GstRTSPResult default_params_get (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPResult res; res = gst_rtsp_params_get (client, ctx); return res; } static gboolean handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPResult res; guint8 *data; guint size; res = gst_rtsp_message_get_body (ctx->request, &data, &size); if (res != GST_RTSP_OK) goto bad_request; if (size == 0) { /* no body, keep-alive request */ send_generic_response (client, GST_RTSP_STS_OK, ctx); } else { /* there is a body, handle the params */ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx); if (res != GST_RTSP_OK) goto bad_request; send_message (client, ctx, ctx->response, FALSE); } g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST], 0, ctx); return TRUE; /* ERRORS */ bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } } static gboolean handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPResult res; guint8 *data; guint size; res = gst_rtsp_message_get_body (ctx->request, &data, &size); if (res != GST_RTSP_OK) goto bad_request; if (size == 0) { /* no body, keep-alive request */ send_generic_response (client, GST_RTSP_STS_OK, ctx); } else { /* there is a body, handle the params */ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx); if (res != GST_RTSP_OK) goto bad_request; send_message (client, ctx, ctx->response, FALSE); } g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST], 0, ctx); return TRUE; /* ERRORS */ bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } } static gboolean handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPSession *session; GstRTSPClientClass *klass; GstRTSPSessionMedia *sessmedia; GstRTSPStatusCode code; GstRTSPState rtspstate; gchar *path; gint matched; if (!(session = ctx->session)) goto no_session; if (!ctx->uri) goto no_uri; klass = GST_RTSP_CLIENT_GET_CLASS (client); path = klass->make_path_from_uri (client, ctx->uri); /* get a handle to the configuration of the media in the session */ sessmedia = gst_rtsp_session_get_media (session, path, &matched); if (!sessmedia) goto not_found; if (path[matched] != '\0') goto no_aggregate; g_free (path); ctx->sessmedia = sessmedia; rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia); /* the session state must be playing or recording */ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_RECORDING) goto invalid_state; /* unlink the all TCP callbacks */ unlink_session_transports (client, session, sessmedia); /* then pause sending */ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED); /* construct the response now */ code = GST_RTSP_STS_OK; gst_rtsp_message_init_response (ctx->response, code, gst_rtsp_status_as_text (code), ctx->request); send_message (client, ctx, ctx->response, FALSE); /* the state is now READY */ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx); return TRUE; /* ERRORS */ no_session: { GST_ERROR ("client %p: no seesion", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx); return FALSE; } no_uri: { GST_ERROR ("client %p: no uri supplied", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } not_found: { GST_ERROR ("client %p: no media for uri", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); g_free (path); return FALSE; } no_aggregate: { GST_ERROR ("client %p: no aggregate path %s", client, path); send_generic_response (client, GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx); g_free (path); return FALSE; } invalid_state: { GST_ERROR ("client %p: not PLAYING or RECORDING", client); send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, ctx); return FALSE; } } /* convert @url and @path to a URL used as a content base for the factory * located at @path */ static gchar * make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path) { GstRTSPUrl tmp; gchar *result; const gchar *trail; /* check for trailing '/' and append one */ trail = (path[strlen (path) - 1] != '/' ? "/" : ""); tmp = *url; tmp.user = NULL; tmp.passwd = NULL; tmp.abspath = g_strdup_printf ("%s%s", path, trail); tmp.query = NULL; result = gst_rtsp_url_get_request_uri (&tmp); g_free (tmp.abspath); return result; } static gboolean handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPSession *session; GstRTSPClientClass *klass; GstRTSPSessionMedia *sessmedia; GstRTSPMedia *media; GstRTSPStatusCode code; GstRTSPUrl *uri; gchar *str; GstRTSPTimeRange *range; GstRTSPResult res; GstRTSPState rtspstate; GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT; gchar *path, *rtpinfo; gint matched; if (!(session = ctx->session)) goto no_session; if (!(uri = ctx->uri)) goto no_uri; klass = GST_RTSP_CLIENT_GET_CLASS (client); path = klass->make_path_from_uri (client, uri); /* get a handle to the configuration of the media in the session */ sessmedia = gst_rtsp_session_get_media (session, path, &matched); if (!sessmedia) goto not_found; if (path[matched] != '\0') goto no_aggregate; g_free (path); ctx->sessmedia = sessmedia; ctx->media = media = gst_rtsp_session_media_get_media (sessmedia); /* the session state must be playing or ready */ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia); if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY) goto invalid_state; /* in play we first unsuspend, media could be suspended from SDP or PAUSED */ if (!gst_rtsp_media_unsuspend (media)) goto unsuspend_failed; /* parse the range header if we have one */ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0); if (res == GST_RTSP_OK) { if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) { /* we have a range, seek to the position */ unit = range->unit; gst_rtsp_media_seek (media, range); gst_rtsp_range_free (range); } } /* link the all TCP callbacks */ link_session_transports (client, session, sessmedia); /* grab RTPInfo from the media now */ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia); /* construct the response now */ code = GST_RTSP_STS_OK; gst_rtsp_message_init_response (ctx->response, code, gst_rtsp_status_as_text (code), ctx->request); /* add the RTP-Info header */ if (rtpinfo) gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, rtpinfo); /* add the range */ str = gst_rtsp_media_get_range_string (media, TRUE, unit); if (str) gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str); send_message (client, ctx, ctx->response, FALSE); /* start playing after sending the response */ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING); gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx); return TRUE; /* ERRORS */ no_session: { GST_ERROR ("client %p: no session", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx); return FALSE; } no_uri: { GST_ERROR ("client %p: no uri supplied", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } not_found: { GST_ERROR ("client %p: media not found", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); return FALSE; } no_aggregate: { GST_ERROR ("client %p: no aggregate path %s", client, path); send_generic_response (client, GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx); g_free (path); return FALSE; } invalid_state: { GST_ERROR ("client %p: not PLAYING or READY", client); send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, ctx); return FALSE; } unsuspend_failed: { GST_ERROR ("client %p: unsuspend failed", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx); return FALSE; } } static void do_keepalive (GstRTSPSession * session) { GST_INFO ("keep session %p alive", session); gst_rtsp_session_touch (session); } /* parse @transport and return a valid transport in @tr. only transports * supported by @stream are returned. Returns FALSE if no valid transport * was found. */ static gboolean parse_transport (const char *transport, GstRTSPStream * stream, GstRTSPTransport * tr) { gint i; gboolean res; gchar **transports; res = FALSE; gst_rtsp_transport_init (tr); GST_DEBUG ("parsing transports %s", transport); transports = g_strsplit (transport, ",", 0); /* loop through the transports, try to parse */ for (i = 0; transports[i]; i++) { res = gst_rtsp_transport_parse (transports[i], tr); if (res != GST_RTSP_OK) { /* no valid transport, search some more */ GST_WARNING ("could not parse transport %s", transports[i]); goto next; } /* we have a transport, see if it's supported */ if (!gst_rtsp_stream_is_transport_supported (stream, tr)) { GST_WARNING ("unsupported transport %s", transports[i]); goto next; } /* we have a valid transport */ GST_INFO ("found valid transport %s", transports[i]); res = TRUE; break; next: gst_rtsp_transport_init (tr); } g_strfreev (transports); return res; } static gboolean default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx) { GstRTSPMessage *request = ctx->request; gchar *blocksize_str; if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE, &blocksize_str, 0) == GST_RTSP_OK) { guint64 blocksize; gchar *end; blocksize = g_ascii_strtoull (blocksize_str, &end, 10); if (end == blocksize_str) goto parse_failed; /* we don't want to change the mtu when this media * can be shared because it impacts other clients */ if (gst_rtsp_media_is_shared (media)) goto done; if (blocksize > G_MAXUINT) blocksize = G_MAXUINT; gst_rtsp_stream_set_mtu (stream, blocksize); } done: return TRUE; /* ERRORS */ parse_failed: { GST_ERROR_OBJECT (client, "failed to parse blocksize"); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } } static gboolean default_configure_client_transport (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPTransport * ct) { GstRTSPClientPrivate *priv = client->priv; /* we have a valid transport now, set the destination of the client. */ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) { gboolean use_client_settings; use_client_settings = gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS); if (ct->destination && use_client_settings) { GstRTSPAddress *addr; addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination, ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl); if (addr == NULL) goto no_address; gst_rtsp_address_free (addr); } else { GstRTSPAddress *addr; GSocketFamily family; family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4; addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family); if (addr == NULL) goto no_address; g_free (ct->destination); ct->destination = g_strdup (addr->address); ct->port.min = addr->port; ct->port.max = addr->port + addr->n_ports - 1; ct->ttl = addr->ttl; gst_rtsp_address_free (addr); } } else { GstRTSPUrl *url; url = gst_rtsp_connection_get_url (priv->connection); g_free (ct->destination); ct->destination = g_strdup (url->host); if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) { GSocket *sock; GSocketAddress *addr; sock = gst_rtsp_connection_get_read_socket (priv->connection); if ((addr = g_socket_get_remote_address (sock, NULL))) { /* our read port is the sender port of client */ ct->client_port.min = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr)); g_object_unref (addr); } if ((addr = g_socket_get_local_address (sock, NULL))) { ct->server_port.max = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr)); g_object_unref (addr); } sock = gst_rtsp_connection_get_write_socket (priv->connection); if ((addr = g_socket_get_remote_address (sock, NULL))) { /* our write port is the receiver port of client */ ct->client_port.max = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr)); g_object_unref (addr); } if ((addr = g_socket_get_local_address (sock, NULL))) { ct->server_port.min = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr)); g_object_unref (addr); } /* check if the client selected channels for TCP */ if (ct->interleaved.min == -1 || ct->interleaved.max == -1) { gst_rtsp_session_media_alloc_channels (ctx->sessmedia, &ct->interleaved); } } } return TRUE; /* ERRORS */ no_address: { GST_ERROR_OBJECT (client, "failed to acquire address for stream"); return FALSE; } } static GstRTSPTransport * make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPTransport * ct) { GstRTSPTransport *st; GInetAddress *addr; GSocketFamily family; /* prepare the server transport */ gst_rtsp_transport_new (&st); st->trans = ct->trans; st->profile = ct->profile; st->lower_transport = ct->lower_transport; addr = g_inet_address_new_from_string (ct->destination); if (!addr) { GST_ERROR ("failed to get inet addr from client destination"); family = G_SOCKET_FAMILY_IPV4; } else { family = g_inet_address_get_family (addr); g_object_unref (addr); addr = NULL; } switch (st->lower_transport) { case GST_RTSP_LOWER_TRANS_UDP: st->client_port = ct->client_port; gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family); break; case GST_RTSP_LOWER_TRANS_UDP_MCAST: st->port = ct->port; st->destination = g_strdup (ct->destination); st->ttl = ct->ttl; break; case GST_RTSP_LOWER_TRANS_TCP: st->interleaved = ct->interleaved; st->client_port = ct->client_port; st->server_port = ct->server_port; default: break; } gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc); return st; } #define AES_128_KEY_LEN 16 #define AES_256_KEY_LEN 32 #define HMAC_32_KEY_LEN 4 #define HMAC_80_KEY_LEN 10 static gboolean mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy) { const gchar *srtp_cipher; const gchar *srtp_auth; const GstMIKEYPayload *sp; guint i; /* loop over Security policy until we find one containing policy */ for (i = 0;; i++) { if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL) break; if (((GstMIKEYPayloadSP *) sp)->policy == policy) break; } /* the default ciphers */ srtp_cipher = "aes-128-icm"; srtp_auth = "hmac-sha1-80"; /* now override the defaults with what is in the Security Policy */ if (sp != NULL) { guint len; /* collect all the params and go over them */ len = gst_mikey_payload_sp_get_n_params (sp); for (i = 0; i < len; i++) { const GstMIKEYPayloadSPParam *param = gst_mikey_payload_sp_get_param (sp, i); switch (param->type) { case GST_MIKEY_SP_SRTP_ENC_ALG: switch (param->val[0]) { case 0: srtp_cipher = "null"; break; case 2: case 1: srtp_cipher = "aes-128-icm"; break; default: break; } break; case GST_MIKEY_SP_SRTP_ENC_KEY_LEN: switch (param->val[0]) { case AES_128_KEY_LEN: srtp_cipher = "aes-128-icm"; break; case AES_256_KEY_LEN: srtp_cipher = "aes-256-icm"; break; default: break; } break; case GST_MIKEY_SP_SRTP_AUTH_ALG: switch (param->val[0]) { case 0: srtp_auth = "null"; break; case 2: case 1: srtp_auth = "hmac-sha1-80"; break; default: break; } break; case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN: switch (param->val[0]) { case HMAC_32_KEY_LEN: srtp_auth = "hmac-sha1-32"; break; case HMAC_80_KEY_LEN: srtp_auth = "hmac-sha1-80"; break; default: break; } break; case GST_MIKEY_SP_SRTP_SRTP_ENC: break; case GST_MIKEY_SP_SRTP_SRTCP_ENC: break; default: break; } } } /* now configure the SRTP parameters */ gst_caps_set_simple (caps, "srtp-cipher", G_TYPE_STRING, srtp_cipher, "srtp-auth", G_TYPE_STRING, srtp_auth, "srtcp-cipher", G_TYPE_STRING, srtp_cipher, "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL); return TRUE; } static gboolean handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx, guint8 * data, gsize size) { GstMIKEYMessage *msg; guint i, n_cs; GstCaps *caps = NULL; GstMIKEYPayloadKEMAC *kemac; const GstMIKEYPayloadKeyData *pkd; GstBuffer *key; /* the MIKEY message contains a CSB or crypto session bundle. It is a * set of Crypto Sessions protected with the same master key. * In the context of SRTP, an RTP and its RTCP stream is part of a * crypto session */ if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL) goto parse_failed; /* we can only handle SRTP crypto sessions for now */ if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP) goto invalid_map_type; /* get the number of crypto sessions. This maps SSRC to its * security parameters */ n_cs = gst_mikey_message_get_n_cs (msg); if (n_cs == 0) goto no_crypto_sessions; /* we also need keys */ if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0))) goto no_keys; /* we don't support encrypted keys */ if (kemac->enc_alg != GST_MIKEY_ENC_NULL || kemac->mac_alg != GST_MIKEY_MAC_NULL) goto unsupported_encryption; /* get Key data sub-payload */ pkd = (const GstMIKEYPayloadKeyData *) gst_mikey_payload_kemac_get_sub (&kemac->pt, 0); key = gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len), pkd->key_len); /* go over all crypto sessions and create the security policy for each * SSRC */ for (i = 0; i < n_cs; i++) { const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i); caps = gst_caps_new_simple ("application/x-srtp", "ssrc", G_TYPE_UINT, map->ssrc, "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL); mikey_apply_policy (caps, msg, map->policy); gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps); gst_caps_unref (caps); } gst_mikey_message_free (msg); return TRUE; /* ERRORS */ parse_failed: { GST_DEBUG_OBJECT (client, "failed to parse MIKEY message"); return FALSE; } invalid_map_type: { GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type); goto cleanup_message; } no_crypto_sessions: { GST_DEBUG_OBJECT (client, "no crypto sessions"); goto cleanup_message; } no_keys: { GST_DEBUG_OBJECT (client, "no keys found"); goto cleanup_message; } unsupported_encryption: { GST_DEBUG_OBJECT (client, "unsupported key encryption"); goto cleanup_message; } cleanup_message: { gst_mikey_message_free (msg); return FALSE; } } #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"')) static void strip_chars (gchar * str) { gchar *s; gsize len; len = strlen (str); while (len--) { if (!IS_STRIP_CHAR (str[len])) break; str[len] = '\0'; } for (s = str; *s && IS_STRIP_CHAR (*s); s++); memmove (str, s, len + 1); } /** * KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec) * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"] */ static gboolean handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt) { gchar **specs; gint i, j; specs = g_strsplit (keymgmt, ",", 0); for (i = 0; specs[i]; i++) { gchar **split; split = g_strsplit (specs[i], ";", 0); for (j = 0; split[j]; j++) { g_strstrip (split[j]); if (g_str_has_prefix (split[j], "prot=")) { g_strstrip (split[j] + 5); if (!g_str_equal (split[j] + 5, "mikey")) break; GST_DEBUG ("found mikey"); } else if (g_str_has_prefix (split[j], "uri=")) { strip_chars (split[j] + 4); GST_DEBUG ("found uri '%s'", split[j] + 4); } else if (g_str_has_prefix (split[j], "data=")) { guchar *data; gsize size; strip_chars (split[j] + 5); GST_DEBUG ("found data '%s'", split[j] + 5); data = g_base64_decode_inplace (split[j] + 5, &size); handle_mikey_data (client, ctx, data, size); } } } return TRUE; } static gboolean handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPClientPrivate *priv = client->priv; GstRTSPResult res; GstRTSPUrl *uri; gchar *transport, *keymgmt; GstRTSPTransport *ct, *st; GstRTSPStatusCode code; GstRTSPSession *session; GstRTSPStreamTransport *trans; gchar *trans_str; GstRTSPSessionMedia *sessmedia; GstRTSPMedia *media; GstRTSPStream *stream; GstRTSPState rtspstate; GstRTSPClientClass *klass; gchar *path, *control; gint matched; if (!ctx->uri) goto no_uri; uri = ctx->uri; klass = GST_RTSP_CLIENT_GET_CLASS (client); path = klass->make_path_from_uri (client, uri); /* parse the transport */ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT, &transport, 0); if (res != GST_RTSP_OK) goto no_transport; /* we create the session after parsing stuff so that we don't make * a session for malformed requests */ if (priv->session_pool == NULL) goto no_pool; session = ctx->session; if (session) { g_object_ref (session); /* get a handle to the configuration of the media in the session, this can * return NULL if this is a new url to manage in this session. */ sessmedia = gst_rtsp_session_get_media (session, path, &matched); } else { /* we need a new media configuration in this session */ sessmedia = NULL; } /* we have no session media, find one and manage it */ if (sessmedia == NULL) { /* get a handle to the configuration of the media in the session */ media = find_media (client, ctx, path, &matched); } else { if ((media = gst_rtsp_session_media_get_media (sessmedia))) g_object_ref (media); else goto media_not_found; } /* no media, not found then */ if (media == NULL) goto media_not_found_no_reply; if (path[matched] == '\0') goto control_not_found; /* path is what matched. */ path[matched] = '\0'; /* control is remainder */ control = &path[matched + 1]; /* find the stream now using the control part */ stream = gst_rtsp_media_find_stream (media, control); if (stream == NULL) goto stream_not_found; /* now we have a uri identifying a valid media and stream */ ctx->stream = stream; ctx->media = media; if (session == NULL) { /* create a session if this fails we probably reached our session limit or * something. */ if (!(session = gst_rtsp_session_pool_create (priv->session_pool))) goto service_unavailable; /* make sure this client is closed when the session is closed */ client_watch_session (client, session); /* signal new session */ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0, session); ctx->session = session; } if (sessmedia == NULL) { /* manage the media in our session now, if not done already */ sessmedia = gst_rtsp_session_manage_media (session, path, media); /* if we stil have no media, error */ if (sessmedia == NULL) goto sessmedia_unavailable; } else { g_object_unref (media); } ctx->sessmedia = sessmedia; if (!klass->configure_client_media (client, media, stream, ctx)) goto configure_media_failed_no_reply; gst_rtsp_transport_new (&ct); /* parse and find a usable supported transport */ if (!parse_transport (transport, stream, ct)) goto unsupported_transports; /* update the client transport */ if (!klass->configure_client_transport (client, ctx, ct)) goto unsupported_client_transport; /* parse the keymgmt */ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT, &keymgmt, 0) == GST_RTSP_OK) { if (!handle_keymgmt (client, ctx, keymgmt)) goto keymgmt_error; } /* set in the session media transport */ trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct); /* configure the url used to set this transport, this we will use when * generating the response for the PLAY request */ gst_rtsp_stream_transport_set_url (trans, uri); /* configure keepalive for this transport */ gst_rtsp_stream_transport_set_keepalive (trans, (GstRTSPKeepAliveFunc) do_keepalive, session, NULL); /* create and serialize the server transport */ st = make_server_transport (client, ctx, ct); trans_str = gst_rtsp_transport_as_text (st); gst_rtsp_transport_free (st); /* construct the response now */ code = GST_RTSP_STS_OK; gst_rtsp_message_init_response (ctx->response, code, gst_rtsp_status_as_text (code), ctx->request); gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT, trans_str); g_free (trans_str); send_message (client, ctx, ctx->response, FALSE); /* update the state */ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia); switch (rtspstate) { case GST_RTSP_STATE_PLAYING: case GST_RTSP_STATE_RECORDING: case GST_RTSP_STATE_READY: /* no state change */ break; default: gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY); break; } g_object_unref (session); g_free (path); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx); return TRUE; /* ERRORS */ no_uri: { GST_ERROR ("client %p: no uri", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } no_transport: { GST_ERROR ("client %p: no transport", client); send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx); goto cleanup_path; } no_pool: { GST_ERROR ("client %p: no session pool configured", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx); goto cleanup_path; } media_not_found_no_reply: { GST_ERROR ("client %p: media '%s' not found", client, path); /* error reply is already sent */ goto cleanup_path; } media_not_found: { GST_ERROR ("client %p: media '%s' not found", client, path); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); goto cleanup_path; } control_not_found: { GST_ERROR ("client %p: no control in path '%s'", client, path); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); g_object_unref (media); goto cleanup_path; } stream_not_found: { GST_ERROR ("client %p: stream '%s' not found", client, control); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); g_object_unref (media); goto cleanup_path; } service_unavailable: { GST_ERROR ("client %p: can't create session", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx); g_object_unref (media); goto cleanup_path; } sessmedia_unavailable: { GST_ERROR ("client %p: can't create session media", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx); g_object_unref (media); goto cleanup_session; } configure_media_failed_no_reply: { GST_ERROR ("client %p: configure_media failed", client); /* error reply is already sent */ goto cleanup_session; } unsupported_transports: { GST_ERROR ("client %p: unsupported transports", client); send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx); goto cleanup_transport; } unsupported_client_transport: { GST_ERROR ("client %p: unsupported client transport", client); send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx); goto cleanup_transport; } keymgmt_error: { GST_ERROR ("client %p: keymgmt error", client); send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx); goto cleanup_transport; } { cleanup_transport: gst_rtsp_transport_free (ct); cleanup_session: g_object_unref (session); cleanup_path: g_free (path); return FALSE; } } static GstSDPMessage * create_sdp (GstRTSPClient * client, GstRTSPMedia * media) { GstRTSPClientPrivate *priv = client->priv; GstSDPMessage *sdp; GstSDPInfo info; const gchar *proto; gst_sdp_message_new (&sdp); /* some standard things first */ gst_sdp_message_set_version (sdp, "0"); if (priv->is_ipv6) proto = "IP6"; else proto = "IP4"; gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto, priv->server_ip); gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer"); gst_sdp_message_set_information (sdp, "rtsp-server"); gst_sdp_message_add_time (sdp, "0", "0", NULL); gst_sdp_message_add_attribute (sdp, "tool", "GStreamer"); gst_sdp_message_add_attribute (sdp, "type", "broadcast"); gst_sdp_message_add_attribute (sdp, "control", "*"); info.is_ipv6 = priv->is_ipv6; info.server_ip = priv->server_ip; /* create an SDP for the media object */ if (!gst_rtsp_media_setup_sdp (media, sdp, &info)) goto no_sdp; return sdp; /* ERRORS */ no_sdp: { GST_ERROR ("client %p: could not create SDP", client); gst_sdp_message_free (sdp); return NULL; } } /* for the describe we must generate an SDP */ static gboolean handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPClientPrivate *priv = client->priv; GstRTSPResult res; GstSDPMessage *sdp; guint i; gchar *path, *str; GstRTSPMedia *media; GstRTSPClientClass *klass; klass = GST_RTSP_CLIENT_GET_CLASS (client); if (!ctx->uri) goto no_uri; /* check what kind of format is accepted, we don't really do anything with it * and always return SDP for now. */ for (i = 0;; i++) { gchar *accept; res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT, &accept, i); if (res == GST_RTSP_ENOTIMPL) break; if (g_ascii_strcasecmp (accept, "application/sdp") == 0) break; } if (!priv->mount_points) goto no_mount_points; if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri))) goto no_path; /* find the media object for the uri */ if (!(media = find_media (client, ctx, path, NULL))) goto no_media; /* create an SDP for the media object on this client */ if (!(sdp = klass->create_sdp (client, media))) goto no_sdp; /* we suspend after the describe */ gst_rtsp_media_suspend (media); g_object_unref (media); gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK, gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request); gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp"); /* content base for some clients that might screw up creating the setup uri */ str = make_base_url (client, ctx->uri, path); g_free (path); GST_INFO ("adding content-base: %s", str); gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str); /* add SDP to the response body */ str = gst_sdp_message_as_text (sdp); gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str)); gst_sdp_message_free (sdp); send_message (client, ctx, ctx->response, FALSE); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST], 0, ctx); return TRUE; /* ERRORS */ no_uri: { GST_ERROR ("client %p: no uri", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); return FALSE; } no_mount_points: { GST_ERROR ("client %p: no mount points configured", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); return FALSE; } no_path: { GST_ERROR ("client %p: can't find path for url", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx); return FALSE; } no_media: { GST_ERROR ("client %p: no media", client); g_free (path); /* error reply is already sent */ return FALSE; } no_sdp: { GST_ERROR ("client %p: can't create SDP", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx); g_free (path); g_object_unref (media); return FALSE; } } static gboolean handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx) { GstRTSPMethod options; gchar *str; options = GST_RTSP_DESCRIBE | GST_RTSP_OPTIONS | GST_RTSP_PAUSE | GST_RTSP_PLAY | GST_RTSP_SETUP | GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN; str = gst_rtsp_options_as_text (options); gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK, gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request); gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str); g_free (str); send_message (client, ctx, ctx->response, FALSE); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST], 0, ctx); return TRUE; } /* remove duplicate and trailing '/' */ static void sanitize_uri (GstRTSPUrl * uri) { gint i, len; gchar *s, *d; gboolean have_slash, prev_slash; s = d = uri->abspath; len = strlen (uri->abspath); prev_slash = FALSE; for (i = 0; i < len; i++) { have_slash = s[i] == '/'; *d = s[i]; if (!have_slash || !prev_slash) d++; prev_slash = have_slash; } len = d - uri->abspath; /* don't remove the first slash if that's the only thing left */ if (len > 1 && *(d - 1) == '/') d--; *d = '\0'; } static void client_session_finalized (GstRTSPClient * client, GstRTSPSession * session) { GstRTSPClientPrivate *priv = client->priv; GST_INFO ("client %p: session %p finished", client, session); /* unlink all media managed in this session */ client_unlink_session (client, session); /* remove the session */ if (!(priv->sessions = g_list_remove (priv->sessions, session))) { GST_INFO ("client %p: all sessions finalized, close the connection", client); close_connection (client); } } /* Returns TRUE if there are no Require headers, otherwise returns FALSE * and also returns a newly-allocated string of (comma-separated) unsupported * options in the unsupported_reqs variable . * * There may be multiple Require headers, but we must send one single * Unsupported header with all the unsupported options as response. If * an incoming Require header contained a comma-separated list of options * GstRtspConnection will already have split that list up into multiple * headers. * * TODO: allow the application to decide what features are supported */ static gboolean check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs) { GstRTSPResult res; GPtrArray *arr = NULL; gchar *reqs = NULL; gint i; i = 0; do { res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++); if (res == GST_RTSP_ENOTIMPL) break; if (arr == NULL) arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free); g_ptr_array_add (arr, g_strdup (reqs)); } while (TRUE); /* if we don't have any Require headers at all, all is fine */ if (i == 1) return TRUE; /* otherwise we've now processed at all the Require headers */ g_ptr_array_add (arr, NULL); /* for now we don't commit to supporting anything, so will just report * all of the required options as unsupported */ *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata); g_ptr_array_unref (arr); return FALSE; } static void handle_request (GstRTSPClient * client, GstRTSPMessage * request) { GstRTSPClientPrivate *priv = client->priv; GstRTSPMethod method; const gchar *uristr; GstRTSPUrl *uri = NULL; GstRTSPVersion version; GstRTSPResult res; GstRTSPSession *session = NULL; GstRTSPContext sctx = { NULL }, *ctx; GstRTSPMessage response = { 0 }; gchar *unsupported_reqs = NULL; gchar *sessid; if (!(ctx = gst_rtsp_context_get_current ())) { ctx = &sctx; ctx->auth = priv->auth; gst_rtsp_context_push_current (ctx); } ctx->conn = priv->connection; ctx->client = client; ctx->request = request; ctx->response = &response; if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) { gst_rtsp_message_dump (request); } gst_rtsp_message_parse_request (request, &method, &uristr, &version); GST_INFO ("client %p: received a request %s %s %s", client, gst_rtsp_method_as_text (method), uristr, gst_rtsp_version_as_text (version)); /* we can only handle 1.0 requests */ if (version != GST_RTSP_VERSION_1_0) goto not_supported; ctx->method = method; /* we always try to parse the url first */ if (strcmp (uristr, "*") == 0) { /* special case where we have * as uri, keep uri = NULL */ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) { /* check if the uristr is an absolute path <=> scheme and host information * is missing */ gchar *scheme; scheme = g_uri_parse_scheme (uristr); if (scheme == NULL && g_str_has_prefix (uristr, "/")) { gchar *absolute_uristr = NULL; GST_WARNING_OBJECT (client, "request doesn't contain absolute url"); if (priv->server_ip == NULL) { GST_WARNING_OBJECT (client, "host information missing"); goto bad_request; } absolute_uristr = g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr); GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr); if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) { g_free (absolute_uristr); goto bad_request; } g_free (absolute_uristr); } else { g_free (scheme); goto bad_request; } } /* get the session if there is any */ res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0); if (res == GST_RTSP_OK) { if (priv->session_pool == NULL) goto no_pool; /* we had a session in the request, find it again */ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid))) goto session_not_found; /* we add the session to the client list of watched sessions. When a session * disappears because it times out, we will be notified. If all sessions are * gone, we will close the connection */ client_watch_session (client, session); } /* sanitize the uri */ if (uri) sanitize_uri (uri); ctx->uri = uri; ctx->session = session; if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL)) goto not_authorized; /* handle any 'Require' headers */ if (!check_request_requirements (ctx->request, &unsupported_reqs)) goto unsupported_requirement; /* now see what is asked and dispatch to a dedicated handler */ switch (method) { case GST_RTSP_OPTIONS: handle_options_request (client, ctx); break; case GST_RTSP_DESCRIBE: handle_describe_request (client, ctx); break; case GST_RTSP_SETUP: handle_setup_request (client, ctx); break; case GST_RTSP_PLAY: handle_play_request (client, ctx); break; case GST_RTSP_PAUSE: handle_pause_request (client, ctx); break; case GST_RTSP_TEARDOWN: handle_teardown_request (client, ctx); break; case GST_RTSP_SET_PARAMETER: handle_set_param_request (client, ctx); break; case GST_RTSP_GET_PARAMETER: handle_get_param_request (client, ctx); break; case GST_RTSP_ANNOUNCE: case GST_RTSP_RECORD: case GST_RTSP_REDIRECT: goto not_implemented; case GST_RTSP_INVALID: default: goto bad_request; } done: if (ctx == &sctx) gst_rtsp_context_pop_current (ctx); if (session) g_object_unref (session); if (uri) gst_rtsp_url_free (uri); return; /* ERRORS */ not_supported: { GST_ERROR ("client %p: version %d not supported", client, version); send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, ctx); goto done; } bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx); goto done; } no_pool: { GST_ERROR ("client %p: no pool configured", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx); goto done; } session_not_found: { GST_ERROR ("client %p: session not found", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx); goto done; } not_authorized: { GST_ERROR ("client %p: not allowed", client); /* error reply is already sent */ goto done; } unsupported_requirement: { GST_ERROR ("client %p: Required option is not supported (%s)", client, unsupported_reqs); send_option_not_supported_response (client, ctx, unsupported_reqs); g_free (unsupported_reqs); goto done; } not_implemented: { GST_ERROR ("client %p: method %d not implemented", client, method); send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx); goto done; } } static void handle_response (GstRTSPClient * client, GstRTSPMessage * response) { GstRTSPClientPrivate *priv = client->priv; GstRTSPResult res; GstRTSPSession *session = NULL; GstRTSPContext sctx = { NULL }, *ctx; gchar *sessid; if (!(ctx = gst_rtsp_context_get_current ())) { ctx = &sctx; ctx->auth = priv->auth; gst_rtsp_context_push_current (ctx); } ctx->conn = priv->connection; ctx->client = client; ctx->request = NULL; ctx->uri = NULL; ctx->method = GST_RTSP_INVALID; ctx->response = response; if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) { gst_rtsp_message_dump (response); } GST_INFO ("client %p: received a response", client); /* get the session if there is any */ res = gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0); if (res == GST_RTSP_OK) { if (priv->session_pool == NULL) goto no_pool; /* we had a session in the request, find it again */ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid))) goto session_not_found; /* we add the session to the client list of watched sessions. When a session * disappears because it times out, we will be notified. If all sessions are * gone, we will close the connection */ client_watch_session (client, session); } ctx->session = session; g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE], 0, ctx); done: if (ctx == &sctx) gst_rtsp_context_pop_current (ctx); if (session) g_object_unref (session); return; no_pool: { GST_ERROR ("client %p: no pool configured", client); goto done; } session_not_found: { GST_ERROR ("client %p: session not found", client); goto done; } } static void handle_data (GstRTSPClient * client, GstRTSPMessage * message) { GstRTSPClientPrivate *priv = client->priv; GstRTSPResult res; guint8 channel; GList *walk; guint8 *data; guint size; GstBuffer *buffer; gboolean handled; /* find the stream for this message */ res = gst_rtsp_message_parse_data (message, &channel); if (res != GST_RTSP_OK) return; gst_rtsp_message_steal_body (message, &data, &size); buffer = gst_buffer_new_wrapped (data, size); handled = FALSE; for (walk = priv->transports; walk; walk = g_list_next (walk)) { GstRTSPStreamTransport *trans; GstRTSPStream *stream; const GstRTSPTransport *tr; trans = walk->data; tr = gst_rtsp_stream_transport_get_transport (trans); stream = gst_rtsp_stream_transport_get_stream (trans); /* check for TCP transport */ if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) { /* dispatch to the stream based on the channel number */ if (tr->interleaved.min == channel) { gst_rtsp_stream_recv_rtp (stream, buffer); handled = TRUE; break; } else if (tr->interleaved.max == channel) { gst_rtsp_stream_recv_rtcp (stream, buffer); handled = TRUE; break; } } } if (!handled) gst_buffer_unref (buffer); } /** * gst_rtsp_client_set_session_pool: * @client: a #GstRTSPClient * @pool: (transfer none): a #GstRTSPSessionPool * * Set @pool as the sessionpool for @client which it will use to find * or allocate sessions. the sessionpool is usually inherited from the server * that created the client but can be overridden later. */ void gst_rtsp_client_set_session_pool (GstRTSPClient * client, GstRTSPSessionPool * pool) { GstRTSPSessionPool *old; GstRTSPClientPrivate *priv; g_return_if_fail (GST_IS_RTSP_CLIENT (client)); priv = client->priv; if (pool) g_object_ref (pool); g_mutex_lock (&priv->lock); old = priv->session_pool; priv->session_pool = pool; g_mutex_unlock (&priv->lock); if (old) g_object_unref (old); } /** * gst_rtsp_client_get_session_pool: * @client: a #GstRTSPClient * * Get the #GstRTSPSessionPool object that @client uses to manage its sessions. * * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage. */ GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient * client) { GstRTSPClientPrivate *priv; GstRTSPSessionPool *result; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL); priv = client->priv; g_mutex_lock (&priv->lock); if ((result = priv->session_pool)) g_object_ref (result); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_client_set_mount_points: * @client: a #GstRTSPClient * @mounts: (transfer none): a #GstRTSPMountPoints * * Set @mounts as the mount points for @client which it will use to map urls * to media streams. These mount points are usually inherited from the server that * created the client but can be overriden later. */ void gst_rtsp_client_set_mount_points (GstRTSPClient * client, GstRTSPMountPoints * mounts) { GstRTSPClientPrivate *priv; GstRTSPMountPoints *old; g_return_if_fail (GST_IS_RTSP_CLIENT (client)); priv = client->priv; if (mounts) g_object_ref (mounts); g_mutex_lock (&priv->lock); old = priv->mount_points; priv->mount_points = mounts; g_mutex_unlock (&priv->lock); if (old) g_object_unref (old); } /** * gst_rtsp_client_get_mount_points: * @client: a #GstRTSPClient * * Get the #GstRTSPMountPoints object that @client uses to manage its sessions. * * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage. */ GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient * client) { GstRTSPClientPrivate *priv; GstRTSPMountPoints *result; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL); priv = client->priv; g_mutex_lock (&priv->lock); if ((result = priv->mount_points)) g_object_ref (result); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_client_set_auth: * @client: a #GstRTSPClient * @auth: (transfer none): a #GstRTSPAuth * * configure @auth to be used as the authentication manager of @client. */ void gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth) { GstRTSPClientPrivate *priv; GstRTSPAuth *old; g_return_if_fail (GST_IS_RTSP_CLIENT (client)); priv = client->priv; if (auth) g_object_ref (auth); g_mutex_lock (&priv->lock); old = priv->auth; priv->auth = auth; g_mutex_unlock (&priv->lock); if (old) g_object_unref (old); } /** * gst_rtsp_client_get_auth: * @client: a #GstRTSPClient * * Get the #GstRTSPAuth used as the authentication manager of @client. * * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after * usage. */ GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient * client) { GstRTSPClientPrivate *priv; GstRTSPAuth *result; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL); priv = client->priv; g_mutex_lock (&priv->lock); if ((result = priv->auth)) g_object_ref (result); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_client_set_thread_pool: * @client: a #GstRTSPClient * @pool: (transfer none): a #GstRTSPThreadPool * * configure @pool to be used as the thread pool of @client. */ void gst_rtsp_client_set_thread_pool (GstRTSPClient * client, GstRTSPThreadPool * pool) { GstRTSPClientPrivate *priv; GstRTSPThreadPool *old; g_return_if_fail (GST_IS_RTSP_CLIENT (client)); priv = client->priv; if (pool) g_object_ref (pool); g_mutex_lock (&priv->lock); old = priv->thread_pool; priv->thread_pool = pool; g_mutex_unlock (&priv->lock); if (old) g_object_unref (old); } /** * gst_rtsp_client_get_thread_pool: * @client: a #GstRTSPClient * * Get the #GstRTSPThreadPool used as the thread pool of @client. * * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after * usage. */ GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient * client) { GstRTSPClientPrivate *priv; GstRTSPThreadPool *result; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL); priv = client->priv; g_mutex_lock (&priv->lock); if ((result = priv->thread_pool)) g_object_ref (result); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_client_set_connection: * @client: a #GstRTSPClient * @conn: (transfer full): a #GstRTSPConnection * * Set the #GstRTSPConnection of @client. This function takes ownership of * @conn. * * Returns: %TRUE on success. */ gboolean gst_rtsp_client_set_connection (GstRTSPClient * client, GstRTSPConnection * conn) { GstRTSPClientPrivate *priv; GSocket *read_socket; GSocketAddress *address; GstRTSPUrl *url; GError *error = NULL; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE); g_return_val_if_fail (conn != NULL, FALSE); priv = client->priv; read_socket = gst_rtsp_connection_get_read_socket (conn); if (!(address = g_socket_get_local_address (read_socket, &error))) goto no_address; g_free (priv->server_ip); /* keep the original ip that the client connected to */ if (G_IS_INET_SOCKET_ADDRESS (address)) { GInetAddress *iaddr; iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address)); /* socket might be ipv6 but adress still ipv4 */ priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6; priv->server_ip = g_inet_address_to_string (iaddr); g_object_unref (address); } else { priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6; priv->server_ip = g_strdup ("unknown"); } GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client, priv->server_ip, priv->is_ipv6); url = gst_rtsp_connection_get_url (conn); GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port); priv->connection = conn; return TRUE; /* ERRORS */ no_address: { GST_ERROR ("could not get local address %s", error->message); g_error_free (error); return FALSE; } } /** * gst_rtsp_client_get_connection: * @client: a #GstRTSPClient * * Get the #GstRTSPConnection of @client. * * Returns: (transfer none): the #GstRTSPConnection of @client. * The connection object returned remains valid until the client is freed. */ GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient * client) { g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL); return client->priv->connection; } /** * gst_rtsp_client_set_send_func: * @client: a #GstRTSPClient * @func: (scope notified): a #GstRTSPClientSendFunc * @user_data: (closure): user data passed to @func * @notify: (allow-none): called when @user_data is no longer in use * * Set @func as the callback that will be called when a new message needs to be * sent to the client. @user_data is passed to @func and @notify is called when * @user_data is no longer in use. * * By default, the client will send the messages on the #GstRTSPConnection that * was configured with gst_rtsp_client_attach() was called. */ void gst_rtsp_client_set_send_func (GstRTSPClient * client, GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify) { GstRTSPClientPrivate *priv; GDestroyNotify old_notify; gpointer old_data; g_return_if_fail (GST_IS_RTSP_CLIENT (client)); priv = client->priv; g_mutex_lock (&priv->send_lock); priv->send_func = func; old_notify = priv->send_notify; old_data = priv->send_data; priv->send_notify = notify; priv->send_data = user_data; g_mutex_unlock (&priv->send_lock); if (old_notify) old_notify (old_data); } /** * gst_rtsp_client_handle_message: * @client: a #GstRTSPClient * @message: (transfer none): an #GstRTSPMessage * * Let the client handle @message. * * Returns: a #GstRTSPResult. */ GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient * client, GstRTSPMessage * message) { g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL); g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL); switch (message->type) { case GST_RTSP_MESSAGE_REQUEST: handle_request (client, message); break; case GST_RTSP_MESSAGE_RESPONSE: handle_response (client, message); break; case GST_RTSP_MESSAGE_DATA: handle_data (client, message); break; default: break; } return GST_RTSP_OK; } /** * gst_rtsp_client_send_message: * @client: a #GstRTSPClient * @session: (transfer none): a #GstRTSPSession to send the message to or %NULL * @message: (transfer none): The #GstRTSPMessage to send * * Send a message message to the remote end. @message must be a * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE. */ GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session, GstRTSPMessage * message) { GstRTSPContext sctx = { NULL } , *ctx; GstRTSPClientPrivate *priv; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL); g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL); g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST || message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL); priv = client->priv; if (!(ctx = gst_rtsp_context_get_current ())) { ctx = &sctx; ctx->auth = priv->auth; gst_rtsp_context_push_current (ctx); } ctx->conn = priv->connection; ctx->client = client; ctx->session = session; send_message (client, ctx, message, FALSE); if (ctx == &sctx) gst_rtsp_context_pop_current (ctx); return GST_RTSP_OK; } static GstRTSPResult do_send_message (GstRTSPClient * client, GstRTSPMessage * message, gboolean close, gpointer user_data) { GstRTSPClientPrivate *priv = client->priv; GstRTSPResult ret; GTimeVal time; time.tv_sec = 1; time.tv_usec = 0; do { /* send the response and store the seq number so we can wait until it's * written to the client to close the connection */ ret = gst_rtsp_watch_send_message (priv->watch, message, close ? &priv->close_seq : NULL); if (ret == GST_RTSP_OK) break; if (ret != GST_RTSP_ENOMEM) goto error; /* drop backlog */ if (priv->drop_backlog) break; /* queue was full, wait for more space */ GST_DEBUG_OBJECT (client, "waiting for backlog"); ret = gst_rtsp_watch_wait_backlog (priv->watch, &time); GST_DEBUG_OBJECT (client, "Resend due to backlog full"); } while (ret != GST_RTSP_EINTR); return ret; /* ERRORS */ error: { GST_DEBUG_OBJECT (client, "got error %d", ret); return ret; } } static GstRTSPResult message_received (GstRTSPWatch * watch, GstRTSPMessage * message, gpointer user_data) { return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message); } static GstRTSPResult message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClientPrivate *priv = client->priv; if (priv->close_seq && priv->close_seq == cseq) { priv->close_seq = 0; close_connection (client); } return GST_RTSP_OK; } static GstRTSPResult closed (GstRTSPWatch * watch, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClientPrivate *priv = client->priv; const gchar *tunnelid; GST_INFO ("client %p: connection closed", client); if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) { g_mutex_lock (&tunnels_lock); /* remove from tunnelids */ g_hash_table_remove (tunnels, tunnelid); g_mutex_unlock (&tunnels_lock); } gst_rtsp_watch_set_flushing (watch, TRUE); gst_rtsp_client_set_send_func (client, NULL, NULL, NULL); return GST_RTSP_OK; } static GstRTSPResult error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); gchar *str; str = gst_rtsp_strresult (result); GST_INFO ("client %p: received an error %s", client, str); g_free (str); return GST_RTSP_OK; } static GstRTSPResult error_full (GstRTSPWatch * watch, GstRTSPResult result, GstRTSPMessage * message, guint id, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); gchar *str; str = gst_rtsp_strresult (result); GST_INFO ("client %p: error when handling message %p with id %d: %s", client, message, id, str); g_free (str); return GST_RTSP_OK; } static gboolean remember_tunnel (GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; const gchar *tunnelid; /* store client in the pending tunnels */ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection); if (tunnelid == NULL) goto no_tunnelid; GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid); /* we can't have two clients connecting with the same tunnelid */ g_mutex_lock (&tunnels_lock); if (g_hash_table_lookup (tunnels, tunnelid)) goto tunnel_existed; g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client)); g_mutex_unlock (&tunnels_lock); return TRUE; /* ERRORS */ no_tunnelid: { GST_ERROR ("client %p: no tunnelid provided", client); return FALSE; } tunnel_existed: { g_mutex_unlock (&tunnels_lock); GST_ERROR ("client %p: tunnel session %s already existed", client, tunnelid); return FALSE; } } static GstRTSPResult tunnel_lost (GstRTSPWatch * watch, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClientPrivate *priv = client->priv; GST_WARNING ("client %p: tunnel lost (connection %p)", client, priv->connection); /* ignore error, it'll only be a problem when the client does a POST again */ remember_tunnel (client); return GST_RTSP_OK; } static gboolean handle_tunnel (GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; GstRTSPClient *oclient; GstRTSPClientPrivate *opriv; const gchar *tunnelid; tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection); if (tunnelid == NULL) goto no_tunnelid; /* check for previous tunnel */ g_mutex_lock (&tunnels_lock); oclient = g_hash_table_lookup (tunnels, tunnelid); if (oclient == NULL) { /* no previous tunnel, remember tunnel */ g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client)); g_mutex_unlock (&tunnels_lock); GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)", client, priv->connection); } else { /* merge both tunnels into the first client */ /* remove the old client from the table. ref before because removing it will * remove the ref to it. */ g_object_ref (oclient); g_hash_table_remove (tunnels, tunnelid); g_mutex_unlock (&tunnels_lock); opriv = oclient->priv; if (opriv->watch == NULL) goto tunnel_closed; GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client, oclient, opriv->connection, priv->connection); gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection); gst_rtsp_watch_reset (priv->watch); gst_rtsp_watch_reset (opriv->watch); g_object_unref (oclient); /* the old client owns the tunnel now, the new one will be freed */ g_source_destroy ((GSource *) priv->watch); priv->watch = NULL; g_main_context_unref (priv->watch_context); priv->watch_context = NULL; gst_rtsp_client_set_send_func (client, NULL, NULL, NULL); } return TRUE; /* ERRORS */ no_tunnelid: { GST_ERROR ("client %p: no tunnelid provided", client); return FALSE; } tunnel_closed: { GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid); g_object_unref (oclient); return FALSE; } } static GstRTSPStatusCode tunnel_get (GstRTSPWatch * watch, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GST_INFO ("client %p: tunnel get (connection %p)", client, client->priv->connection); if (!handle_tunnel (client)) { return GST_RTSP_STS_SERVICE_UNAVAILABLE; } return GST_RTSP_STS_OK; } static GstRTSPResult tunnel_post (GstRTSPWatch * watch, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GST_INFO ("client %p: tunnel post (connection %p)", client, client->priv->connection); if (!handle_tunnel (client)) { return GST_RTSP_ERROR; } return GST_RTSP_OK; } static GstRTSPResult tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request, GstRTSPMessage * response, gpointer user_data) { GstRTSPClientClass *klass; GstRTSPClient *client = GST_RTSP_CLIENT (user_data); klass = GST_RTSP_CLIENT_GET_CLASS (client); if (klass->tunnel_http_response) { klass->tunnel_http_response (client, request, response); } return GST_RTSP_OK; } static GstRTSPWatchFuncs watch_funcs = { message_received, message_sent, closed, error, tunnel_get, tunnel_post, error_full, tunnel_lost, tunnel_http_response }; static void client_watch_notify (GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; GST_INFO ("client %p: watch destroyed", client); priv->watch = NULL; g_main_context_unref (priv->watch_context); priv->watch_context = NULL; g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL); g_object_unref (client); } /** * gst_rtsp_client_attach: * @client: a #GstRTSPClient * @context: (allow-none): a #GMainContext * * Attaches @client to @context. When the mainloop for @context is run, the * client will be dispatched. When @context is %NULL, the default context will be * used). * * This function should be called when the client properties and urls are fully * configured and the client is ready to start. * * Returns: the ID (greater than 0) for the source within the GMainContext. */ guint gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context) { GstRTSPClientPrivate *priv; guint res; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0); priv = client->priv; g_return_val_if_fail (priv->connection != NULL, 0); g_return_val_if_fail (priv->watch == NULL, 0); /* make sure noone will free the context before the watch is destroyed */ priv->watch_context = g_main_context_ref (context); /* create watch for the connection and attach */ priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs, g_object_ref (client), (GDestroyNotify) client_watch_notify); gst_rtsp_client_set_send_func (client, do_send_message, priv->watch, (GDestroyNotify) gst_rtsp_watch_unref); /* FIXME make this configurable. We don't want to do this yet because it will * be superceeded by a cache object later */ gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100); GST_INFO ("attaching to context %p", context); res = gst_rtsp_watch_attach (priv->watch, context); return res; } /** * gst_rtsp_client_session_filter: * @client: a #GstRTSPClient * @func: (scope call) (allow-none): a callback * @user_data: user data passed to @func * * Call @func for each session managed by @client. The result value of @func * determines what happens to the session. @func will be called with @client * locked so no further actions on @client can be performed from @func. * * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from * @client. * * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client. * * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but * will also be added with an additional ref to the result #GList of this * function.. * * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session. * * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each * element in the #GList should be unreffed before the list is freed. */ GList * gst_rtsp_client_session_filter (GstRTSPClient * client, GstRTSPClientSessionFilterFunc func, gpointer user_data) { GstRTSPClientPrivate *priv; GList *result, *walk, *next; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL); priv = client->priv; result = NULL; g_mutex_lock (&priv->lock); for (walk = priv->sessions; walk; walk = next) { GstRTSPSession *sess = walk->data; GstRTSPFilterResult res; next = g_list_next (walk); if (func) res = func (client, sess, user_data); else res = GST_RTSP_FILTER_REF; switch (res) { case GST_RTSP_FILTER_REMOVE: /* stop watching the session and pretent it went away */ client_cleanup_session (client, sess); break; case GST_RTSP_FILTER_REF: result = g_list_prepend (result, g_object_ref (sess)); break; case GST_RTSP_FILTER_KEEP: default: break; } } g_mutex_unlock (&priv->lock); return result; }