/* 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. */ #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)) struct _GstRTSPClientPrivate { GMutex lock; GMutex send_lock; GstRTSPConnection *connection; GstRTSPWatch *watch; guint close_seq; gchar *server_ip; gboolean is_ipv6; gboolean use_client_settings; GstRTSPClientSendFunc send_func; gpointer send_data; GDestroyNotify send_notify; GstRTSPSessionPool *session_pool; GstRTSPMountPoints *mount_points; GstRTSPAuth *auth; GstRTSPUrl *uri; GstRTSPMedia *media; GList *transports; GList *sessions; }; static GMutex tunnels_lock; static GHashTable *tunnels; #define DEFAULT_SESSION_POOL NULL #define DEFAULT_MOUNT_POINTS NULL #define DEFAULT_USE_CLIENT_SETTINGS FALSE enum { PROP_0, PROP_SESSION_POOL, PROP_MOUNT_POINTS, PROP_USE_CLIENT_SETTINGS, 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_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 * media); 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; 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_USE_CLIENT_SETTINGS, g_param_spec_boolean ("use-client-settings", "Use Client Settings", "Use client settings for ttl and destination in multicast", DEFAULT_USE_CLIENT_SETTINGS, 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); 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->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS; priv->close_seq = 0; } static GstRTSPFilterResult filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * media, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); gst_rtsp_session_media_set_state (media, GST_STATE_NULL); unlink_session_transports (client, sess, media); /* 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_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)) { GstRTSPSession *session = (GstRTSPSession *) sessions->data; g_object_weak_unref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client); client_unlink_session (client, session); } 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); gst_rtsp_client_set_send_func (client, NULL, NULL, NULL); if (priv->watch) g_source_destroy ((GSource *) priv->watch); 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->uri) gst_rtsp_url_free (priv->uri); 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); 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_USE_CLIENT_SETTINGS: g_value_set_boolean (value, gst_rtsp_client_get_use_client_settings (client)); 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); 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_USE_CLIENT_SETTINGS: gst_rtsp_client_set_use_client_settings (client, g_value_get_boolean (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } /** * gst_rtsp_client_new: * * Create a new #GstRTSPClient instance. * * Returns: 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_response (GstRTSPClient * client, GstRTSPSession * session, GstRTSPMessage * response, gboolean close) { GstRTSPClientPrivate *priv = client->priv; gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER, "GStreamer RTSP server"); /* remove any previous header */ gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1); /* add the new session header for new session ids */ if (session) { gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, gst_rtsp_session_get_header (session)); } if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) { gst_rtsp_message_dump (response); } if (close) gst_rtsp_message_add_header (response, GST_RTSP_HDR_CONNECTION, "close"); g_mutex_lock (&priv->send_lock); if (priv->send_func) priv->send_func (client, response, close, priv->send_data); g_mutex_unlock (&priv->send_lock); gst_rtsp_message_unset (response); } static void send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code, GstRTSPClientState * state) { gst_rtsp_message_init_response (state->response, code, gst_rtsp_status_as_text (code), state->request); send_response (client, NULL, state->response, FALSE); } static void handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth, GstRTSPClientState * state) { gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED, gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request); if (auth) { /* and let the authentication manager setup the auth tokens */ gst_rtsp_auth_setup_auth (auth, client, 0, state); } send_response (client, state->session, state->response, FALSE); } static gboolean compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2) { if (uri1 == NULL || uri2 == NULL) return FALSE; if (strcmp (uri1->abspath, uri2->abspath)) 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, GstRTSPClientState * state) { GstRTSPClientPrivate *priv = client->priv; GstRTSPMediaFactory *factory; GstRTSPMedia *media; GstRTSPAuth *auth; if (!compare_uri (priv->uri, state->uri)) { /* remove any previously cached values before we try to construct a new * media for uri */ if (priv->uri) gst_rtsp_url_free (priv->uri); priv->uri = NULL; if (priv->media) { gst_rtsp_media_unprepare (priv->media); g_object_unref (priv->media); } priv->media = NULL; if (!priv->mount_points) goto no_mount_points; /* find the factory for the uri first */ if (!(factory = gst_rtsp_mount_points_find_factory (priv->mount_points, state->uri))) goto no_factory; /* check if we have access to the factory */ if ((auth = gst_rtsp_media_factory_get_auth (factory))) { state->factory = factory; if (!gst_rtsp_auth_check (auth, client, 0, state)) goto not_allowed; state->factory = NULL; g_object_unref (auth); } /* prepare the media and add it to the pipeline */ if (!(media = gst_rtsp_media_factory_construct (factory, state->uri))) goto no_media; g_object_unref (factory); factory = NULL; /* prepare the media */ if (!(gst_rtsp_media_prepare (media))) goto no_prepare; /* now keep track of the uri and the media */ priv->uri = gst_rtsp_url_copy (state->uri); priv->media = media; state->media = media; } else { /* we have seen this uri before, used cached media */ media = priv->media; state->media = media; GST_INFO ("reusing cached media %p", media); } if (media) g_object_ref (media); return media; /* ERRORS */ no_mount_points: { GST_ERROR ("client %p: no mount points configured", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state); return NULL; } no_factory: { GST_ERROR ("client %p: no factory for uri", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state); return NULL; } not_allowed: { GST_ERROR ("client %p: unauthorized request", client); handle_unauthorized_request (client, auth, state); g_object_unref (factory); state->factory = NULL; g_object_unref (auth); return NULL; } no_media: { GST_ERROR ("client %p: can't create media", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state); g_object_unref (factory); return NULL; } no_prepare: { GST_ERROR ("client %p: can't prepare media", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state); g_object_unref (media); 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 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 * media) { guint n_streams, i; n_streams = gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media)); 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 (media, 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); } static gboolean handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPClientPrivate *priv = client->priv; GstRTSPSession *session; GstRTSPSessionMedia *media; GstRTSPStatusCode code; if (!state->session) goto no_session; session = state->session; /* get a handle to the configuration of the media in the session */ media = gst_rtsp_session_get_media (session, state->uri); if (!media) goto not_found; state->sessmedia = media; /* unlink the all TCP callbacks */ unlink_session_transports (client, session, media); /* remove the session from the watched sessions */ g_object_weak_unref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client); priv->sessions = g_list_remove (priv->sessions, session); gst_rtsp_session_media_set_state (media, GST_STATE_NULL); /* unmanage the media in the session, returns false if all media session * are torn down. */ if (!gst_rtsp_session_release_media (session, media)) { /* 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 (state->response, code, gst_rtsp_status_as_text (code), state->request); send_response (client, session, state->response, TRUE); /* we emit the signal before closing the connection */ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST], 0, state); return TRUE; /* ERRORS */ no_session: { GST_ERROR ("client %p: no session", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state); return FALSE; } not_found: { GST_ERROR ("client %p: no media for uri", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state); return FALSE; } } static gboolean handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPResult res; guint8 *data; guint size; res = gst_rtsp_message_get_body (state->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, state); } else { /* there is a body, handle the params */ res = gst_rtsp_params_get (client, state); if (res != GST_RTSP_OK) goto bad_request; send_response (client, state->session, state->response, FALSE); } g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST], 0, state); return TRUE; /* ERRORS */ bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state); return FALSE; } } static gboolean handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPResult res; guint8 *data; guint size; res = gst_rtsp_message_get_body (state->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, state); } else { /* there is a body, handle the params */ res = gst_rtsp_params_set (client, state); if (res != GST_RTSP_OK) goto bad_request; send_response (client, state->session, state->response, FALSE); } g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST], 0, state); return TRUE; /* ERRORS */ bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state); return FALSE; } } static gboolean handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPSession *session; GstRTSPSessionMedia *media; GstRTSPStatusCode code; GstRTSPState rtspstate; if (!(session = state->session)) goto no_session; /* get a handle to the configuration of the media in the session */ media = gst_rtsp_session_get_media (session, state->uri); if (!media) goto not_found; state->sessmedia = media; rtspstate = gst_rtsp_session_media_get_rtsp_state (media); /* 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, media); /* then pause sending */ gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED); /* construct the response now */ code = GST_RTSP_STS_OK; gst_rtsp_message_init_response (state->response, code, gst_rtsp_status_as_text (code), state->request); send_response (client, session, state->response, FALSE); /* the state is now READY */ gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_READY); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, state); return TRUE; /* ERRORS */ no_session: { GST_ERROR ("client %p: no seesion", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state); return FALSE; } not_found: { GST_ERROR ("client %p: no media for uri", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state); 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, state); return FALSE; } } static gboolean handle_play_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPSession *session; GstRTSPSessionMedia *media; GstRTSPStatusCode code; GString *rtpinfo; guint n_streams, i, infocount; gchar *str; GstRTSPTimeRange *range; GstRTSPResult res; GstRTSPState rtspstate; if (!(session = state->session)) goto no_session; /* get a handle to the configuration of the media in the session */ media = gst_rtsp_session_get_media (session, state->uri); if (!media) goto not_found; state->sessmedia = media; /* the session state must be playing or ready */ rtspstate = gst_rtsp_session_media_get_rtsp_state (media); if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY) goto invalid_state; /* parse the range header if we have one */ res = gst_rtsp_message_get_header (state->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 */ gst_rtsp_media_seek (gst_rtsp_session_media_get_media (media), range); gst_rtsp_range_free (range); } } /* grab RTPInfo from the payloaders now */ rtpinfo = g_string_new (""); n_streams = gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media)); for (i = 0, infocount = 0; i < n_streams; i++) { GstRTSPStreamTransport *trans; GstRTSPStream *stream; const GstRTSPTransport *tr; gchar *uristr; guint rtptime, seq; /* get the transport, if there is no transport configured, skip this stream */ trans = gst_rtsp_session_media_get_transport (media, i); if (trans == NULL) { GST_INFO ("stream %d is not configured", i); 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); } stream = gst_rtsp_stream_transport_get_stream (trans); if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) { if (infocount > 0) g_string_append (rtpinfo, ", "); uristr = gst_rtsp_url_get_request_uri (state->uri); g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u", uristr, i, seq, rtptime); g_free (uristr); infocount++; } else { GST_WARNING ("RTP-Info cannot be determined for stream %d", i); } } /* construct the response now */ code = GST_RTSP_STS_OK; gst_rtsp_message_init_response (state->response, code, gst_rtsp_status_as_text (code), state->request); /* add the RTP-Info header */ if (infocount > 0) { str = g_string_free (rtpinfo, FALSE); gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str); } else { g_string_free (rtpinfo, TRUE); } /* add the range */ str = gst_rtsp_media_get_range_string (gst_rtsp_session_media_get_media (media), TRUE); gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str); send_response (client, session, state->response, FALSE); /* start playing after sending the request */ gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING); gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_PLAYING); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, state); return TRUE; /* ERRORS */ no_session: { GST_ERROR ("client %p: no session", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state); return FALSE; } not_found: { GST_ERROR ("client %p: media not found", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state); 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, state); 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 * from @supported are returned. Returns FALSE if no valid transport * was found. */ static gboolean parse_transport (const char *transport, GstRTSPLowerTrans supported, 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 RTP/AVP */ if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) { GST_WARNING ("invalid transport %s", transports[i]); goto next; } if (!(tr->lower_transport & supported)) { 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 handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPMessage * request) { gchar *blocksize_str; gboolean ret = TRUE; 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) { GST_ERROR ("failed to parse blocksize"); ret = FALSE; } else { /* 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)) return TRUE; if (blocksize > G_MAXUINT) blocksize = G_MAXUINT; gst_rtsp_stream_set_mtu (stream, blocksize); } } return ret; } static gboolean configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state, 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) { if (ct->destination && priv->use_client_settings) { GstRTSPAddress *addr; addr = gst_rtsp_stream_reserve_address (state->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; addr = gst_rtsp_stream_get_address (state->stream); 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) { /* check if the client selected channels for TCP */ if (ct->interleaved.min == -1 || ct->interleaved.max == -1) { gst_rtsp_session_media_alloc_channels (state->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, GstRTSPClientState * state, GstRTSPTransport * ct) { GstRTSPTransport *st; /* prepare the server transport */ gst_rtsp_transport_new (&st); st->trans = ct->trans; st->profile = ct->profile; st->lower_transport = ct->lower_transport; switch (st->lower_transport) { case GST_RTSP_LOWER_TRANS_UDP: st->client_port = ct->client_port; gst_rtsp_stream_get_server_port (state->stream, &st->server_port); 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; default: break; } gst_rtsp_stream_get_ssrc (state->stream, &st->ssrc); return st; } static gboolean handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPClientPrivate *priv = client->priv; GstRTSPResult res; GstRTSPUrl *uri; gchar *transport; GstRTSPTransport *ct, *st; GstRTSPLowerTrans supported; GstRTSPStatusCode code; GstRTSPSession *session; GstRTSPStreamTransport *trans; gchar *trans_str, *pos; guint streamid; GstRTSPSessionMedia *sessmedia; GstRTSPMedia *media; GstRTSPStream *stream; GstRTSPState rtspstate; uri = state->uri; /* the uri contains the stream number we added in the SDP config, which is * always /stream=%d so we need to strip that off * parse the stream we need to configure, look for the stream in the abspath * first and then in the query. */ if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) { if (uri->query == NULL || !(pos = strstr (uri->query, "/stream="))) goto bad_request; } /* we can mofify the parsed uri in place */ *pos = '\0'; pos += strlen ("/stream="); if (sscanf (pos, "%u", &streamid) != 1) goto bad_request; /* parse the transport */ res = gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT, &transport, 0); if (res != GST_RTSP_OK) goto no_transport; gst_rtsp_transport_new (&ct); /* our supported transports */ supported = GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP; /* parse and find a usable supported transport */ if (!parse_transport (transport, supported, ct)) goto unsupported_transports; /* 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 = state->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, uri); } else { /* 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; state->session = session; /* we need a new media configuration in this session */ sessmedia = NULL; } /* we have no media, find one and manage it */ if (sessmedia == NULL) { /* get a handle to the configuration of the media in the session */ if ((media = find_media (client, state))) { /* manage the media in our session now */ sessmedia = gst_rtsp_session_manage_media (session, uri, media); } } /* if we stil have no media, error */ if (sessmedia == NULL) goto not_found; state->sessmedia = sessmedia; state->media = media = gst_rtsp_session_media_get_media (sessmedia); /* now get the stream */ stream = gst_rtsp_media_get_stream (media, streamid); if (stream == NULL) goto not_found; state->stream = stream; /* set blocksize on this stream */ if (!handle_blocksize (media, stream, state->request)) goto invalid_blocksize; /* update the client transport */ if (!configure_client_transport (client, state, ct)) goto unsupported_client_transport; /* set in the session media transport */ trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct); /* 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, state, 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 (state->response, code, gst_rtsp_status_as_text (code), state->request); gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT, trans_str); g_free (trans_str); send_response (client, session, state->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_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, state); return TRUE; /* ERRORS */ bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state); return FALSE; } not_found: { GST_ERROR ("client %p: media not found", client); send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state); g_object_unref (session); gst_rtsp_transport_free (ct); return FALSE; } invalid_blocksize: { GST_ERROR ("client %p: invalid blocksize", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state); g_object_unref (session); gst_rtsp_transport_free (ct); return FALSE; } unsupported_client_transport: { GST_ERROR ("client %p: unsupported client transport", client); send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state); g_object_unref (session); gst_rtsp_transport_free (ct); return FALSE; } no_transport: { GST_ERROR ("client %p: no transport", client); send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state); return FALSE; } unsupported_transports: { GST_ERROR ("client %p: unsupported transports", client); send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state); gst_rtsp_transport_free (ct); return FALSE; } no_pool: { GST_ERROR ("client %p: no session pool configured", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state); gst_rtsp_transport_free (ct); return FALSE; } service_unavailable: { GST_ERROR ("client %p: can't create session", client); send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state); gst_rtsp_transport_free (ct); 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.server_proto = proto; info.server_ip = g_strdup (priv->server_ip); /* create an SDP for the media object */ if (!gst_rtsp_sdp_from_media (sdp, &info, media)) goto no_sdp; g_free (info.server_ip); return sdp; /* ERRORS */ no_sdp: { GST_ERROR ("client %p: could not create SDP", client); g_free (info.server_ip); gst_sdp_message_free (sdp); return NULL; } } /* for the describe we must generate an SDP */ static gboolean handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state) { GstRTSPResult res; GstSDPMessage *sdp; guint i, str_len; gchar *str, *content_base; GstRTSPMedia *media; GstRTSPClientClass *klass; klass = GST_RTSP_CLIENT_GET_CLASS (client); /* 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 (state->request, GST_RTSP_HDR_ACCEPT, &accept, i); if (res == GST_RTSP_ENOTIMPL) break; if (g_ascii_strcasecmp (accept, "application/sdp") == 0) break; } /* find the media object for the uri */ if (!(media = find_media (client, state))) goto no_media; /* create an SDP for the media object on this client */ if (!(sdp = klass->create_sdp (client, media))) goto no_sdp; g_object_unref (media); gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK, gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request); gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp"); /* content base for some clients that might screw up creating the setup uri */ str = gst_rtsp_url_get_request_uri (state->uri); str_len = strlen (str); /* check for trailing '/' and append one */ if (str[str_len - 1] != '/') { content_base = g_malloc (str_len + 2); memcpy (content_base, str, str_len); content_base[str_len] = '/'; content_base[str_len + 1] = '\0'; g_free (str); } else { content_base = str; } GST_INFO ("adding content-base: %s", content_base); gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE, content_base); g_free (content_base); /* add SDP to the response body */ str = gst_sdp_message_as_text (sdp); gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str)); gst_sdp_message_free (sdp); send_response (client, state->session, state->response, FALSE); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST], 0, state); return TRUE; /* ERRORS */ no_media: { GST_ERROR ("client %p: no media", client); /* 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, state); g_object_unref (media); return FALSE; } } static gboolean handle_options_request (GstRTSPClient * client, GstRTSPClientState * state) { 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 (state->response, GST_RTSP_STS_OK, gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request); gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str); g_free (str); send_response (client, state->session, state->response, FALSE); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST], 0, state); 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); } } 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); g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0, session); } 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; GstRTSPClientState state = { NULL }; GstRTSPMessage response = { 0 }; gchar *sessid; state.request = request; state.response = &response; if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) { gst_rtsp_message_dump (request); } GST_INFO ("client %p: received a request", client); gst_rtsp_message_parse_request (request, &method, &uristr, &version); /* we can only handle 1.0 requests */ if (version != GST_RTSP_VERSION_1_0) goto not_supported; state.method = method; /* we always try to parse the url first */ if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) 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 */ sanitize_uri (uri); state.uri = uri; state.session = session; if (priv->auth) { if (!gst_rtsp_auth_check (priv->auth, client, 0, &state)) goto not_authorized; } /* now see what is asked and dispatch to a dedicated handler */ switch (method) { case GST_RTSP_OPTIONS: handle_options_request (client, &state); break; case GST_RTSP_DESCRIBE: handle_describe_request (client, &state); break; case GST_RTSP_SETUP: handle_setup_request (client, &state); break; case GST_RTSP_PLAY: handle_play_request (client, &state); break; case GST_RTSP_PAUSE: handle_pause_request (client, &state); break; case GST_RTSP_TEARDOWN: handle_teardown_request (client, &state); break; case GST_RTSP_SET_PARAMETER: handle_set_param_request (client, &state); break; case GST_RTSP_GET_PARAMETER: handle_get_param_request (client, &state); 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 (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, &state); goto done; } bad_request: { GST_ERROR ("client %p: bad request", client); send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state); goto done; } no_pool: { GST_ERROR ("client %p: no pool configured", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state); goto done; } session_not_found: { GST_ERROR ("client %p: session not found", client); send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state); goto done; } not_authorized: { GST_ERROR ("client %p: not allowed", client); handle_unauthorized_request (client, priv->auth, &state); goto done; } not_implemented: { GST_ERROR ("client %p: method %d not implemented", client, method); send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state); 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: 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: 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_use_client_settings: * @client: a #GstRTSPClient * @use_client_settings: whether to use client settings for multicast * * Use client transport settings (destination and ttl) for multicast. * When @use_client_settings is %FALSE, the server settings will be * used. */ void gst_rtsp_client_set_use_client_settings (GstRTSPClient * client, gboolean use_client_settings) { GstRTSPClientPrivate *priv; g_return_if_fail (GST_IS_RTSP_CLIENT (client)); priv = client->priv; g_mutex_lock (&priv->lock); priv->use_client_settings = use_client_settings; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_client_get_use_client_settings: * @client: a #GstRTSPClient * * Check if client transport settings (destination and ttl) for multicast * will be used. */ gboolean gst_rtsp_client_get_use_client_settings (GstRTSPClient * client) { GstRTSPClientPrivate *priv; gboolean res; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE); priv = client->priv; g_mutex_lock (&priv->lock); res = priv->use_client_settings; g_mutex_unlock (&priv->lock); return res; } /** * gst_rtsp_client_set_auth: * @client: a #GstRTSPClient * @auth: 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_send_func: * @client: a #GstRTSPClient * @func: a #GstRTSPClientSendFunc * @user_data: user data passed to @func * @notify: 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. */ 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: 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: break; case GST_RTSP_MESSAGE_DATA: handle_data (client, message); break; default: break; } return GST_RTSP_OK; } static GstRTSPResult do_send_message (GstRTSPClient * client, GstRTSPMessage * message, gboolean close, gpointer user_data) { GstRTSPClientPrivate *priv = client->priv; /* send the response and store the seq number so we can wait until it's * written to the client to close the connection */ return gst_rtsp_watch_send_message (priv->watch, message, close ? &priv->close_seq : NULL); } 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_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: received an error %s when handling message %p with id %d", client, str, message, id); 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 GstRTSPStatusCode tunnel_start (GstRTSPWatch * watch, gpointer user_data) { GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClientPrivate *priv = client->priv; GST_INFO ("client %p: tunnel start (connection %p)", client, priv->connection); if (!remember_tunnel (client)) goto tunnel_error; return GST_RTSP_STS_OK; /* ERRORS */ tunnel_error: { GST_ERROR ("client %p: error starting tunnel", client); return GST_RTSP_STS_SERVICE_UNAVAILABLE; } } 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 GstRTSPResult tunnel_complete (GstRTSPWatch * watch, gpointer user_data) { const gchar *tunnelid; GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClientPrivate *priv = client->priv; GstRTSPClient *oclient; GstRTSPClientPrivate *opriv; GST_INFO ("client %p: tunnel complete", client); /* find previous tunnel */ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection); if (tunnelid == NULL) goto no_tunnelid; g_mutex_lock (&tunnels_lock); if (!(oclient = g_hash_table_lookup (tunnels, tunnelid))) goto no_tunnel; /* 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); opriv = oclient->priv; if (opriv->watch == NULL) goto tunnel_closed; g_mutex_unlock (&tunnels_lock); GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient, opriv->connection, priv->connection); /* merge the tunnels into the first client */ gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection); gst_rtsp_watch_reset (opriv->watch); g_object_unref (oclient); return GST_RTSP_OK; /* ERRORS */ no_tunnelid: { GST_ERROR ("client %p: no tunnelid provided", client); return GST_RTSP_ERROR; } no_tunnel: { g_mutex_unlock (&tunnels_lock); GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid); return GST_RTSP_ERROR; } tunnel_closed: { g_mutex_unlock (&tunnels_lock); GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid); g_object_unref (oclient); return GST_RTSP_ERROR; } } static GstRTSPWatchFuncs watch_funcs = { message_received, message_sent, closed, error, tunnel_start, tunnel_complete, error_full, tunnel_lost }; static void client_watch_notify (GstRTSPClient * client) { GstRTSPClientPrivate *priv = client->priv; GST_INFO ("client %p: watch destroyed", client); priv->watch = NULL; g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL); g_object_unref (client); } static gboolean setup_client (GstRTSPClient * client, GSocket * socket, GstRTSPConnection * conn, GError ** error) { GstRTSPClientPrivate *priv = client->priv; GSocket *read_socket; GSocketAddress *address; GstRTSPUrl *url; read_socket = gst_rtsp_connection_get_read_socket (conn); priv->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6; if (!(address = g_socket_get_remote_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)); priv->server_ip = g_inet_address_to_string (iaddr); g_object_unref (address); } else { 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 remote address %s", (*error)->message); return FALSE; } } /** * gst_rtsp_client_use_socket: * @client: a #GstRTSPClient * @socket: a #GSocket * @ip: the IP address of the remote client * @port: the port used by the other end * @initial_buffer: any zero terminated initial data that was already read from * the socket * @error: a #GError * * Take an existing network socket and use it for an RTSP connection. * * Returns: %TRUE on success. */ gboolean gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket, const gchar * ip, gint port, const gchar * initial_buffer, GError ** error) { GstRTSPConnection *conn; GstRTSPResult res; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE); g_return_val_if_fail (G_IS_SOCKET (socket), FALSE); GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port, initial_buffer, &conn), no_connection); return setup_client (client, socket, conn, error); /* ERRORS */ no_connection: { gchar *str = gst_rtsp_strresult (res); GST_ERROR ("could not create connection from socket %p: %s", socket, str); g_free (str); return FALSE; } } /** * gst_rtsp_client_accept: * @client: a #GstRTSPClient * @socket: a #GSocket * @context: the context to run in * @cancellable: a #GCancellable * @error: a #GError * * Accept a new connection for @client on @socket. * * Returns: %TRUE if the client could be accepted. */ gboolean gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket, GCancellable * cancellable, GError ** error) { GstRTSPConnection *conn; GstRTSPResult res; g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE); g_return_val_if_fail (G_IS_SOCKET (socket), FALSE); /* a new client connected. */ GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable), accept_failed); return setup_client (client, socket, conn, error); /* ERRORS */ accept_failed: { gchar *str = gst_rtsp_strresult (res); GST_ERROR ("Could not accept client on server socket %p: %s", socket, str); g_free (str); return FALSE; } } /** * 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->watch == NULL, 0); /* 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; }