/* GStreamer * Copyright (C) 2017 Matthew Waters * * 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. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "ice.h" #include "icestream.h" #include "webrtc-priv.h" #define GST_CAT_DEFAULT gst_webrtc_ice_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); enum { SIGNAL_0, ADD_LOCAL_IP_ADDRESS_SIGNAL, LAST_SIGNAL, }; enum { PROP_0, PROP_MIN_RTP_PORT, PROP_MAX_RTP_PORT, }; static guint gst_webrtc_ice_signals[LAST_SIGNAL] = { 0 }; #define gst_webrtc_ice_parent_class parent_class G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICE, gst_webrtc_ice, GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_debug, "webrtcice", 0, "webrtcice");); /** * gst_webrtc_ice_add_stream: * @ice: The #GstWebRTCICE * @session_id: The session id * * Returns: (transfer full) (nullable): The #GstWebRTCICEStream, or %NULL * Since: 1.22 */ GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice, guint session_id) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream); return GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream (ice, session_id); } /** * gst_webrtc_ice_find_transport: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * @component: The #GstWebRTCICEComponent * * Returns: (transfer full) (nullable): The #GstWebRTCICETransport, or %NULL * Since: 1.22 */ GstWebRTCICETransport * gst_webrtc_ice_find_transport (GstWebRTCICE * ice, GstWebRTCICEStream * stream, GstWebRTCICEComponent component) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport); return GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport (ice, stream, component); } /** * gst_webrtc_ice_add_candidate: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * @candidate: The ICE candidate * Since: 1.22 */ void gst_webrtc_ice_add_candidate (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * candidate) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate); GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate (ice, stream, candidate); } /** * gst_webrtc_ice_set_remote_credentials: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * @ufrag: ICE username * @pwd: ICE password * Returns: FALSE on error, TRUE otherwise * Since: 1.22 */ gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice, GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials); return GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials (ice, stream, ufrag, pwd); } /** * gst_webrtc_ice_add_turn_server: * @ice: The #GstWebRTCICE * @uri: URI of the TURN server * Returns: FALSE on error, TRUE otherwise * Since: 1.22 */ gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice, const gchar * uri) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server); return GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server (ice, uri); } /** * gst_webrtc_ice_set_local_credentials: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * @ufrag: ICE username * @pwd: ICE password * Returns: FALSE on error, TRUE otherwise * Since: 1.22 */ gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice, GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials); return GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials (ice, stream, ufrag, pwd); } /** * gst_webrtc_ice_gather_candidates: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * Returns: FALSE on error, TRUE otherwise * Since: 1.22 */ gboolean gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice, GstWebRTCICEStream * stream) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates); return GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates (ice, stream); } /** * gst_webrtc_ice_set_is_controller: * @ice: The #GstWebRTCICE * @controller: TRUE to set as controller * Since: 1.22 */ void gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice, gboolean controller) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller); GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller (ice, controller); } /** * gst_webrtc_ice_get_is_controller: * @ice: The #GstWebRTCICE * Returns: TRUE if set as controller, FALSE otherwise * Since: 1.22 */ gboolean gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller); return GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller (ice); } /** * gst_webrtc_ice_set_force_relay: * @ice: The #GstWebRTCICE * @force_relay: TRUE to enable force relay * Since: 1.22 */ void gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice, gboolean force_relay) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay); GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay (ice, force_relay); } /** * gst_webrtc_ice_set_tos: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * @tos: ToS to be set * Since: 1.22 */ void gst_webrtc_ice_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream, guint tos) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos); GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos (ice, stream, tos); } /** * gst_webrtc_ice_get_local_candidates: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * Returns: (transfer full) (element-type GstWebRTCICECandidateStats): List of local candidates * Since: 1.22 */ GArray * gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice, GstWebRTCICEStream * stream) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates); return GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates (ice, stream); } /** * gst_webrtc_ice_get_remote_candidates: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * Returns: (transfer full) (element-type GstWebRTCICECandidateStats): List of remote candidates * Since: 1.22 */ GArray * gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice, GstWebRTCICEStream * stream) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates); return GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates (ice, stream); } /** * gst_webrtc_ice_get_selected_pair: * @ice: The #GstWebRTCICE * @stream: The #GstWebRTCICEStream * @local_stats: A pointer to #GstWebRTCICECandidateStats for local candidate * @remote_stats: A pointer to #GstWebRTCICECandidateStats for remote candidate * * Returns: FALSE on failure, otherwise @local_stats @remote_stats will be set * Since: 1.22 */ gboolean gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice, GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats, GstWebRTCICECandidateStats ** remote_stats) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair); return GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair (ice, stream, local_stats, remote_stats); } /** * gst_webrtc_ice_candidate_stats_free: * @stats: The #GstWebRTCICECandidateStats to be free'd * * Helper function to free #GstWebRTCICECandidateStats * Since: 1.22 */ void gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats) { if (stats) { g_free (stats->ipaddr); g_free (stats->url); } g_free (stats); } /** * gst_webrtc_ice_set_on_ice_candidate: * @ice: The #GstWebRTCICE * @func: The #GstWebRTCICEOnCandidateFunc callback function * @user_data: User data passed to the callback function * @notify: a #GDestroyNotify when the candidate is no longer needed * Since: 1.22 */ void gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice, GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate); GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate (ice, func, user_data, notify); } /** * gst_webrtc_ice_set_stun_server: * @ice: The #GstWebRTCICE * @uri: URI of the STUN server * Since: 1.22 */ void gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice, const gchar * uri_s) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server); GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server (ice, uri_s); } /** * gst_webrtc_ice_get_stun_server: * @ice: The #GstWebRTCICE * Returns: URI of the STUN sever * Since: 1.22 */ gchar * gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server); return GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server (ice); } /** * gst_webrtc_ice_set_turn_server: * @ice: The #GstWebRTCICE * @uri: URI of the TURN sever * Since: 1.22 */ void gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice, const gchar * uri_s) { g_return_if_fail (GST_IS_WEBRTC_ICE (ice)); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server); GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server (ice, uri_s); } /** * gst_webrtc_ice_get_turn_server: * @ice: The #GstWebRTCICE * Returns: URI of the TURN sever * Since: 1.22 */ gchar * gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice) { g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL); g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server); return GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server (ice); } static void gst_webrtc_ice_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWebRTCICE *ice = GST_WEBRTC_ICE (object); switch (prop_id) { case PROP_MIN_RTP_PORT: ice->min_rtp_port = g_value_get_uint (value); if (ice->min_rtp_port > ice->max_rtp_port) g_warning ("Set min-rtp-port to %u which is larger than" " max-rtp-port %u", ice->min_rtp_port, ice->max_rtp_port); break; case PROP_MAX_RTP_PORT: ice->max_rtp_port = g_value_get_uint (value); if (ice->min_rtp_port > ice->max_rtp_port) g_warning ("Set max-rtp-port to %u which is smaller than" " min-rtp-port %u", ice->max_rtp_port, ice->min_rtp_port); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_ice_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWebRTCICE *ice = GST_WEBRTC_ICE (object); switch (prop_id) { case PROP_MIN_RTP_PORT: g_value_set_uint (value, ice->min_rtp_port); break; case PROP_MAX_RTP_PORT: g_value_set_uint (value, ice->max_rtp_port); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_ice_class_init (GstWebRTCICEClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; klass->add_stream = NULL; klass->find_transport = NULL; klass->gather_candidates = NULL; klass->add_candidate = NULL; klass->set_local_credentials = NULL; klass->set_remote_credentials = NULL; klass->add_turn_server = NULL; klass->set_is_controller = NULL; klass->get_is_controller = NULL; klass->set_force_relay = NULL; klass->set_stun_server = NULL; klass->get_stun_server = NULL; klass->set_turn_server = NULL; klass->get_turn_server = NULL; klass->set_tos = NULL; klass->set_on_ice_candidate = NULL; klass->get_local_candidates = NULL; klass->get_remote_candidates = NULL; klass->get_selected_pair = NULL; gobject_class->get_property = gst_webrtc_ice_get_property; gobject_class->set_property = gst_webrtc_ice_set_property; /** * GstWebRTCICE:min-rtp-port: * * Minimum port for local rtp port range. * min-rtp-port must be <= max-rtp-port * * Since: 1.20 */ g_object_class_install_property (gobject_class, PROP_MIN_RTP_PORT, g_param_spec_uint ("min-rtp-port", "ICE RTP candidate min port", "Minimum port for local rtp port range. " "min-rtp-port must be <= max-rtp-port", 0, 65535, 0, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); /** * GstWebRTCICE:max-rtp-port: * * Maximum port for local rtp port range. * min-rtp-port must be <= max-rtp-port * * Since: 1.20 */ g_object_class_install_property (gobject_class, PROP_MAX_RTP_PORT, g_param_spec_uint ("max-rtp-port", "ICE RTP candidate max port", "Maximum port for local rtp port range. " "max-rtp-port must be >= min-rtp-port", 0, 65535, 65535, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); /** * GstWebRTCICE::add-local-ip-address: * @object: the #GstWebRTCICE * @address: The local IP address * * Add a local IP address to use for ICE candidate gathering. If none * are supplied, they will be discovered automatically. Calling this signal * stops automatic ICE gathering. * * Returns: whether the address could be added. */ gst_webrtc_ice_signals[ADD_LOCAL_IP_ADDRESS_SIGNAL] = g_signal_new_class_handler ("add-local-ip-address", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, NULL, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_STRING); } static void gst_webrtc_ice_init (GstWebRTCICE * ice) { }