mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-26 11:41:09 +00:00
5b64cfaca3
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP proxy to access the Internet it MUST include the "ALPN" header. This commit adds this header. By default the ALPN used when connecting to the TURN/TCP server via a proxy is set to "webrtc". It can be changed by adding an alpn url option for the http-proxy. For example: http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT request. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
624 lines
17 KiB
C
624 lines
17 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.
|
|
*/
|
|
/**
|
|
* SECTION:gstwebrtcice
|
|
* @title: GstWebRTCICE
|
|
* @short_description: Base class WebRTC ICE handling
|
|
* @symbols:
|
|
* - GstWebRTCICE
|
|
*/
|
|
|
|
#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
|
|
* @promise: (nullable): A #GstPromise for task notifications (Since: 1.24)
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
void
|
|
gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
|
|
GstWebRTCICEStream * stream, const gchar * candidate, GstPromise * promise)
|
|
{
|
|
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,
|
|
promise);
|
|
}
|
|
|
|
/**
|
|
* 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, const gchar * ufrag, const 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, const gchar * ufrag, const 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)(array zero-terminated=1): List of local candidates
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
GstWebRTCICECandidateStats **
|
|
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) (array zero-terminated=1): List of remote candidates
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
GstWebRTCICECandidateStats **
|
|
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: (out) (transfer full): A pointer to #GstWebRTCICECandidateStats for local candidate
|
|
* @remote_stats: (out) (transfer full): 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_candidate_stats_copy:
|
|
* @stats: The #GstWebRTCICE
|
|
*
|
|
* Returns: (transfer full): A copy of @stats
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
GstWebRTCICECandidateStats *
|
|
gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats * stats)
|
|
{
|
|
GstWebRTCICECandidateStats *copy =
|
|
g_malloc (sizeof (GstWebRTCICECandidateStats));
|
|
|
|
*copy = *stats;
|
|
|
|
copy->ipaddr = g_strdup (stats->ipaddr);
|
|
copy->url = g_strdup (stats->url);
|
|
|
|
return copy;
|
|
}
|
|
|
|
G_DEFINE_BOXED_TYPE (GstWebRTCICECandidateStats, gst_webrtc_ice_candidate_stats,
|
|
(GBoxedCopyFunc) gst_webrtc_ice_candidate_stats_copy,
|
|
(GBoxedFreeFunc) gst_webrtc_ice_candidate_stats_free);
|
|
|
|
/**
|
|
* 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: (nullable): 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: (nullable): 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: (nullable): 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: (nullable): 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);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_ice_set_http_proxy:
|
|
* @ice: The #GstWebRTCICE
|
|
* @uri: (transfer none): URI of the HTTP proxy of the form
|
|
* http://[username:password@]hostname[:port][?alpn=<alpn>]
|
|
*
|
|
* Set HTTP Proxy to be used when connecting to TURN server.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
void
|
|
gst_webrtc_ice_set_http_proxy (GstWebRTCICE * ice, const gchar * uri_s)
|
|
{
|
|
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
|
|
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_http_proxy);
|
|
|
|
GST_WEBRTC_ICE_GET_CLASS (ice)->set_http_proxy (ice, uri_s);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_ice_get_http_proxy:
|
|
* @ice: The #GstWebRTCICE
|
|
*
|
|
* Returns: (transfer full): URI of the HTTP proxy of the form
|
|
* http://[username:password@]hostname[:port][?alpn=<alpn>]
|
|
*
|
|
* Get HTTP Proxy to be used when connecting to TURN server.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
gchar *
|
|
gst_webrtc_ice_get_http_proxy (GstWebRTCICE * ice)
|
|
{
|
|
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
|
|
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_http_proxy);
|
|
|
|
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_http_proxy (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->get_http_proxy = NULL;
|
|
klass->set_http_proxy = 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)
|
|
{
|
|
}
|