/* 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. */ #ifndef __GST_WEBRTC_ICE_H__ #define __GST_WEBRTC_ICE_H__ #include G_BEGIN_DECLS GST_WEBRTC_API GType gst_webrtc_ice_get_type(void); #define GST_TYPE_WEBRTC_ICE (gst_webrtc_ice_get_type()) #define GST_WEBRTC_ICE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ICE,GstWebRTCICE)) #define GST_IS_WEBRTC_ICE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ICE)) #define GST_WEBRTC_ICE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ICE,GstWebRTCICEClass)) #define GST_IS_WEBRTC_ICE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ICE)) #define GST_WEBRTC_ICE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE,GstWebRTCICEClass)) struct _GstWebRTCICE { GstObject parent; GstWebRTCICEGatheringState ice_gathering_state; GstWebRTCICEConnectionState ice_connection_state; /*< protected >*/ guint min_rtp_port; guint max_rtp_port; gpointer _gst_reserved[GST_PADDING]; }; /** * GstWebRTCICECandidateStats: * @ipaddr: A string containing the address of the candidate. This value may be * an IPv4 address, an IPv6 address, or a fully-qualified domain name * @port: The network port number used by the candidate * @stream_id: A string that uniquely identifies the object that is being * monitored to produce this set of statistics * @type: The candidate type * @proto: A string specifying the protocol (tcp or udp) used to transmit data * on the @port * @replay_proto: A string identifying the protocol used by the endpoint for * communicating with the TURN server; valid values are tcp, udp, and tls * @prio: The candidate's priority, corresponding to RTCIceCandidate.priority * @url: For local candidates, the url property is the URL of the ICE server * from which the candidate was received * @foundation: The ICE foundation as defined in RFC5245 section 15.1 (Since: 1.28) * @related_address: The ICE rel-addr defined in RFC5245 section 15.1 Only * set for serverreflexive, peerreflexive and relay candidates. (Since: 1.28) * @related_port: The ICE rel-addr defined in RFC5245 section 15.1. Only set * for serverreflexive, peerreflexive and relay candidates. (Since: 1.28) * @username_fragment: The ICE username fragment as defined in RFC5245 section 7.1.2.3 (Since: 1.28) * @tcp_type: The ICE candidate TCP type, (Since: 1.28) * * Since: 1.22 */ struct _GstWebRTCICECandidateStats { gchar *ipaddr; guint port; guint stream_id; const gchar *type; const gchar *proto; const gchar *relay_proto; guint prio; gchar *url; /** * GstWebRTCICECandidateStats.ABI: (attributes doc.skip=true) * * ABI compatibility union * * Since: 1.28 */ union { /** * GstWebRTCICECandidateStats.ABI.abi: (attributes doc.skip=true) * * ABI compatibility struct * * Since: 1.28 */ struct { /** * GstWebRTCICECandidateStats.ABI.abi.foundation: * * The foundation of the ICE candidate. * * Since: 1.28 */ gchar *foundation; /** * GstWebRTCICECandidateStats.ABI.abi.related_address: * * The related address (STUN or TURN server) of the candidate. Will be * NULL for host candidates. * * Since: 1.28 */ gchar *related_address; /** * GstWebRTCICECandidateStats.ABI.abi.related_port: * * The related port (STUN or TURN server) of the candidate. Will be * 0 for host candidates. * * Since: 1.28 */ guint related_port; /** * GstWebRTCICECandidateStats.ABI.abi.username_fragment: * * The ICE username for this candidate. * * Since: 1.28 */ gchar *username_fragment; /** * GstWebRTCICECandidateStats.ABI.abi.tcp_type: * * The type of TCP candidate. Will be NULL if the candidate is not a TCP * candidate. * * Since: 1.28 */ GstWebRTCICETcpCandidateType tcp_type; } abi; /*< private >*/ gpointer _gst_reserved[GST_PADDING_LARGE]; } ABI; }; /** * GST_WEBRTC_ICE_CANDIDATE_STATS_ADDRESS: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_ADDRESS(c) ((c)->ipaddr) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_PORT: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_PORT(c) ((c)->port) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_STREAM_ID: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_STREAM_ID(c) ((c)->stream_id) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_TYPE: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_TYPE(c) ((c)->type) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_PROTOCOL: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_PROTOCOL(c) ((c)->proto) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_RELAY_PROTOCOL: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_RELAY_PROTOCOL(c) ((c)->relay_proto) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_PRIORITY: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_PRIORITY(c) ((c)->prio) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_URL: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_URL(c) ((c)->url) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_FOUNDATION: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_FOUNDATION(c) ((c)->ABI.abi.foundation) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_ADDRESS: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_ADDRESS(c) ((c)->ABI.abi.related_address) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_PORT: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_PORT(c) ((c)->ABI.abi.related_port) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_USERNAME_FRAGMENT: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_USERNAME_FRAGMENT(c) ((c)->ABI.abi.username_fragment) /** * GST_WEBRTC_ICE_CANDIDATE_STATS_TCP_TYPE: * * Since: 1.28 */ #define GST_WEBRTC_ICE_CANDIDATE_STATS_TCP_TYPE(c) ((c)->ABI.abi.tcp_type) /** * GstWebRTCICECandidate: * @candidate: String carrying the candidate-attribute as defined in * section 15.1 of RFC5245 * @component: The assigned network component of the candidate (1 for RTP * 2 for RTCP). * @sdp_mid: The media stream "identification-tag" defined in [RFC5888] for the * media component this candidate is associated with. * @sdp_mline_index: The index (starting at zero) of the media description in * the SDP this candidate is associated with. * @stats: The #GstWebRTCICECandidateStats associated to this candidate. * * Since: 1.28 */ struct _GstWebRTCICECandidate { gchar *candidate; gint component; gchar *sdp_mid; gint sdp_mline_index; /* Set to -1 if unknown. */ GstWebRTCICECandidateStats *stats; gpointer _gst_reserved[GST_PADDING_LARGE]; }; struct _GstWebRTCICECandidatePair { GstWebRTCICECandidate *local; GstWebRTCICECandidate *remote; }; /** * GstWebRTCICEOnCandidateFunc: * @ice: The #GstWebRTCICE * @stream_id: The stream id * @candidate: The discovered candidate * @user_data: User data that was set by #gst_webrtc_ice_set_on_ice_candidate * * Callback function to be triggered on discovery of a new candidate * Since: 1.22 */ typedef void (*GstWebRTCICEOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, const gchar * candidate, gpointer user_data); struct _GstWebRTCICEClass { GstObjectClass parent_class; GstWebRTCICEStream * (*add_stream) (GstWebRTCICE * ice, guint session_id); GstWebRTCICETransport * (*find_transport) (GstWebRTCICE * ice, GstWebRTCICEStream * stream, GstWebRTCICEComponent component); gboolean (*gather_candidates) (GstWebRTCICE * ice, GstWebRTCICEStream * stream); void (*add_candidate) (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * candidate, GstPromise * promise); gboolean (*set_local_credentials) (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * ufrag, const gchar * pwd); gboolean (*set_remote_credentials) (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * ufrag, const gchar * pwd); gboolean (*add_turn_server) (GstWebRTCICE * ice, const gchar * uri); void (*set_is_controller) (GstWebRTCICE * ice, gboolean controller); gboolean (*get_is_controller) (GstWebRTCICE * ice); void (*set_force_relay) (GstWebRTCICE * ice, gboolean force_relay); void (*set_stun_server) (GstWebRTCICE * ice, const gchar * uri); gchar * (*get_stun_server) (GstWebRTCICE * ice); void (*set_turn_server) (GstWebRTCICE * ice, const gchar * uri); gchar * (*get_turn_server) (GstWebRTCICE * ice); /** * GstWebRTCICEClass::set_http_proxy: * @ice: a #GstWebRTCICE * @uri: (transfer none): URI of the HTTP proxy of the form * http://[username:password@]hostname[:port] * * Set HTTP Proxy to be used when connecting to TURN server. * * Since: 1.22 */ void (*set_http_proxy) (GstWebRTCICE * ice, const gchar * uri); /** * GstWebRTCICEClass::get_http_proxy: * @ice: a #GstWebRTCICE * * Get HTTP Proxy to be used when connecting to TURN server. * * Returns: (transfer full): URI of the HTTP proxy of the form * http://[username:password@]hostname[:port] * * Since: 1.22 */ gchar * (*get_http_proxy) (GstWebRTCICE * ice); void (*set_tos) (GstWebRTCICE * ice, GstWebRTCICEStream * stream, guint tos); void (*set_on_ice_candidate) (GstWebRTCICE * ice, GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify); GstWebRTCICECandidateStats** (*get_local_candidates)(GstWebRTCICE * ice, GstWebRTCICEStream * stream); GstWebRTCICECandidateStats**(*get_remote_candidates)(GstWebRTCICE * ice, GstWebRTCICEStream * stream); gboolean (*get_selected_pair) (GstWebRTCICE * ice, GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats, GstWebRTCICECandidateStats ** remote_stats); /** * GstWebRTCICEClass::close: * @ice: a #GstWebRTCICE * @promise: (transfer full) (nullable): a #GstPromise to be notified when the task is * complete. * * Invoke the close procedure as specified in * https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close. * * Since: 1.28 */ void (*close) (GstWebRTCICE * ice, GstPromise * promise); gpointer _gst_reserved[GST_PADDING - 1]; }; GST_WEBRTC_API GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice, guint session_id) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API GstWebRTCICETransport * gst_webrtc_ice_find_transport (GstWebRTCICE * ice, GstWebRTCICEStream * stream, GstWebRTCICEComponent component) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API gboolean gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice, GstWebRTCICEStream * stream); /* FIXME: GstStructure-ize the candidate */ GST_WEBRTC_API void gst_webrtc_ice_add_candidate (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * candidate, GstPromise * promise); GST_WEBRTC_API gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * ufrag, const gchar * pwd); GST_WEBRTC_API gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * ufrag, const gchar * pwd); GST_WEBRTC_API gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice, const gchar * uri); GST_WEBRTC_API void gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice, gboolean controller); GST_WEBRTC_API gboolean gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice); GST_WEBRTC_API void gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice, gboolean force_relay); GST_WEBRTC_API void gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice, const gchar * uri); GST_WEBRTC_API gchar * gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API void gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice, const gchar * uri); GST_WEBRTC_API gchar * gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API void gst_webrtc_ice_set_http_proxy (GstWebRTCICE * ice, const gchar * uri); GST_WEBRTC_API gchar * gst_webrtc_ice_get_http_proxy (GstWebRTCICE * ice) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API void gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice, GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify); GST_WEBRTC_API void gst_webrtc_ice_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream, guint tos); GST_WEBRTC_API GstWebRTCICECandidateStats** gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice, GstWebRTCICEStream * stream) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API GstWebRTCICECandidateStats** gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice, GstWebRTCICEStream * stream) G_GNUC_WARN_UNUSED_RESULT; #ifndef GST_DISABLE_DEPRECATED GST_WEBRTC_DEPRECATED_FOR(gst_webrtc_ice_transport_get_selected_pair) gboolean gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice, GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats, GstWebRTCICECandidateStats ** remote_stats); #endif GST_WEBRTC_API void gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats); GST_WEBRTC_API GType gst_webrtc_ice_candidate_stats_get_type (void); G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICE, gst_object_unref) GST_WEBRTC_API GstWebRTCICECandidateStats * gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats *stats) G_GNUC_WARN_UNUSED_RESULT; GST_WEBRTC_API void gst_webrtc_ice_close (GstWebRTCICE * ice, GstPromise * promise); GST_WEBRTC_API void gst_webrtc_ice_candidate_free (GstWebRTCICECandidate * candidate); GST_WEBRTC_API GType gst_webrtc_ice_candidate_get_type (void); GST_WEBRTC_API GstWebRTCICECandidate * gst_webrtc_ice_candidate_copy (GstWebRTCICECandidate * candidate); GST_WEBRTC_API void gst_webrtc_ice_candidate_pair_free (GstWebRTCICECandidatePair * pair); GST_WEBRTC_API GType gst_webrtc_ice_candidate_pair_get_type (void); GST_WEBRTC_API GstWebRTCICECandidatePair * gst_webrtc_ice_candidate_pair_copy (GstWebRTCICECandidatePair * pair); G_END_DECLS #endif /* __GST_WEBRTC_ICE_H__ */