gstreamer/ext/webrtc/gstwebrtcice.h
Ilya Kreymer 92626535c7 webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
2021-03-01 14:42:17 +00:00

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5.9 KiB
C

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* 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 <gst/gst.h>
#include <gst/sdp/sdp.h>
#include <gst/webrtc/webrtc.h>
#include "fwd.h"
G_BEGIN_DECLS
#define GST_WEBRTC_ICE_ERROR gst_webrtc_ice_error_quark ()
GQuark gst_webrtc_ice_error_quark (void);
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;
GstUri *stun_server;
GstUri *turn_server;
GHashTable *turn_servers;
GstWebRTCICEPrivate *priv;
guint min_rtp_port;
guint max_rtp_port;
};
struct _GstWebRTCICEClass
{
GstObjectClass parent_class;
};
GstWebRTCICE * gst_webrtc_ice_new (const gchar * name);
GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice,
guint session_id);
GstWebRTCICETransport * gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
GstWebRTCICEComponent component);
gboolean gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
/* FIXME: GstStructure-ize the candidate */
void gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
const gchar * candidate);
gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice,
const gchar * uri);
void gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice,
gboolean controller);
gboolean gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice);
void gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice,
gboolean force_relay);
void gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice,
const gchar * uri);
gchar * gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice);
void gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice,
const gchar * uri);
gchar * gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice);
typedef void (*GstWebRTCIceOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, gchar * candidate, gpointer user_data);
void gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
GstWebRTCIceOnCandidateFunc func,
gpointer user_data,
GDestroyNotify notify);
void gst_webrtc_ice_set_tos (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
guint tos);
G_END_DECLS
#endif /* __GST_WEBRTC_ICE_H__ */