mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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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
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstwebrtcice
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* @title: GstWebRTCICE
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* @short_description: Base class WebRTC ICE handling
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* @symbols:
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* - GstWebRTCICE
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "ice.h"
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#include "icestream.h"
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#include "webrtc-priv.h"
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#define GST_CAT_DEFAULT gst_webrtc_ice_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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enum
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{
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SIGNAL_0,
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ADD_LOCAL_IP_ADDRESS_SIGNAL,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_MIN_RTP_PORT,
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PROP_MAX_RTP_PORT,
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};
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static guint gst_webrtc_ice_signals[LAST_SIGNAL] = { 0 };
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#define gst_webrtc_ice_parent_class parent_class
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G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICE, gst_webrtc_ice,
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GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_debug,
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"webrtcice", 0, "webrtcice"););
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/**
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* gst_webrtc_ice_add_stream:
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* @ice: The #GstWebRTCICE
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* @session_id: The session id
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*
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* Returns: (transfer full) (nullable): The #GstWebRTCICEStream, or %NULL
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*
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* Since: 1.22
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*/
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GstWebRTCICEStream *
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gst_webrtc_ice_add_stream (GstWebRTCICE * ice, guint session_id)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream (ice, session_id);
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}
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/**
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* gst_webrtc_ice_find_transport:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* @component: The #GstWebRTCICEComponent
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*
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* Returns: (transfer full) (nullable): The #GstWebRTCICETransport, or %NULL
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*
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* Since: 1.22
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*/
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GstWebRTCICETransport *
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gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream, GstWebRTCICEComponent component)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport (ice, stream,
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component);
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}
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/**
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* gst_webrtc_ice_add_candidate:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* @candidate: The ICE candidate
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* @promise: (nullable): A #GstPromise for task notifications (Since: 1.24)
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream, const gchar * candidate, GstPromise * promise)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate);
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GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate (ice, stream, candidate,
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promise);
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}
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/**
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* gst_webrtc_ice_set_remote_credentials:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* @ufrag: ICE username
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* @pwd: ICE password
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*
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* Returns: FALSE on error, TRUE otherwise
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*
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* Since: 1.22
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*/
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gboolean
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gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream, const gchar * ufrag, const gchar * pwd)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials (ice, stream,
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ufrag, pwd);
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}
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/**
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* gst_webrtc_ice_add_turn_server:
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* @ice: The #GstWebRTCICE
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* @uri: URI of the TURN server
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*
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* Returns: FALSE on error, TRUE otherwise
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*
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* Since: 1.22
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*/
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gboolean
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gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice, const gchar * uri)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server (ice, uri);
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}
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/**
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* gst_webrtc_ice_set_local_credentials:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* @ufrag: ICE username
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* @pwd: ICE password
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*
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* Returns: FALSE on error, TRUE otherwise
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*
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* Since: 1.22
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*/
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gboolean
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gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream, const gchar * ufrag, const gchar * pwd)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials (ice, stream,
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ufrag, pwd);
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}
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/**
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* gst_webrtc_ice_gather_candidates:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* Returns: FALSE on error, TRUE otherwise
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*
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* Since: 1.22
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*/
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gboolean
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gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates (ice, stream);
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}
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/**
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* gst_webrtc_ice_set_is_controller:
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* @ice: The #GstWebRTCICE
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* @controller: TRUE to set as controller
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice, gboolean controller)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller);
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GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller (ice, controller);
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}
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/**
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* gst_webrtc_ice_get_is_controller:
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* @ice: The #GstWebRTCICE
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* Returns: TRUE if set as controller, FALSE otherwise
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*
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* Since: 1.22
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*/
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gboolean
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gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller (ice);
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}
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/**
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* gst_webrtc_ice_set_force_relay:
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* @ice: The #GstWebRTCICE
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* @force_relay: TRUE to enable force relay
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice, gboolean force_relay)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay);
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GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay (ice, force_relay);
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}
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/**
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* gst_webrtc_ice_set_tos:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* @tos: ToS to be set
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream,
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guint tos)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos);
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GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos (ice, stream, tos);
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}
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/**
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* gst_webrtc_ice_get_local_candidates:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* Returns: (transfer full)(array zero-terminated=1): List of local candidates
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*
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* Since: 1.22
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*/
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GstWebRTCICECandidateStats **
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gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates (ice, stream);
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}
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/**
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* gst_webrtc_ice_get_remote_candidates:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* Returns: (transfer full) (array zero-terminated=1): List of remote candidates
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*
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* Since: 1.22
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*/
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GstWebRTCICECandidateStats **
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gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates (ice, stream);
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}
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/**
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* gst_webrtc_ice_get_selected_pair:
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* @ice: The #GstWebRTCICE
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* @stream: The #GstWebRTCICEStream
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* @local_stats: (out) (transfer full): A pointer to #GstWebRTCICECandidateStats for local candidate
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* @remote_stats: (out) (transfer full): pointer to #GstWebRTCICECandidateStats for remote candidate
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*
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* Returns: FALSE on failure, otherwise @local_stats @remote_stats will be set
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*
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* Since: 1.22
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*/
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gboolean
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gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats,
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GstWebRTCICECandidateStats ** remote_stats)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair (ice, stream,
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local_stats, remote_stats);
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}
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/**
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* gst_webrtc_ice_candidate_stats_free:
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* @stats: The #GstWebRTCICECandidateStats to be free'd
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*
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* Helper function to free #GstWebRTCICECandidateStats
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats)
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{
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if (stats) {
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g_free (stats->ipaddr);
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g_free (stats->url);
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}
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g_free (stats);
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}
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/**
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* gst_webrtc_ice_candidate_stats_copy:
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* @stats: The #GstWebRTCICE
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*
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* Returns: (transfer full): A copy of @stats
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*
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* Since: 1.22
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*/
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GstWebRTCICECandidateStats *
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gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats * stats)
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{
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GstWebRTCICECandidateStats *copy =
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g_malloc (sizeof (GstWebRTCICECandidateStats));
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*copy = *stats;
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copy->ipaddr = g_strdup (stats->ipaddr);
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copy->url = g_strdup (stats->url);
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return copy;
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}
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G_DEFINE_BOXED_TYPE (GstWebRTCICECandidateStats, gst_webrtc_ice_candidate_stats,
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(GBoxedCopyFunc) gst_webrtc_ice_candidate_stats_copy,
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(GBoxedFreeFunc) gst_webrtc_ice_candidate_stats_free);
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/**
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* gst_webrtc_ice_set_on_ice_candidate:
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* @ice: The #GstWebRTCICE
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* @func: The #GstWebRTCICEOnCandidateFunc callback function
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* @user_data: User data passed to the callback function
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* @notify: a #GDestroyNotify when the candidate is no longer needed
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
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GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate);
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GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate (ice, func, user_data,
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notify);
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}
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/**
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* gst_webrtc_ice_set_stun_server:
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* @ice: The #GstWebRTCICE
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* @uri: (nullable): URI of the STUN server
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice, const gchar * uri_s)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server);
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GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server (ice, uri_s);
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}
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/**
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* gst_webrtc_ice_get_stun_server:
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* @ice: The #GstWebRTCICE
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*
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* Returns: (nullable): URI of the STUN sever
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*
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* Since: 1.22
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*/
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gchar *
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gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server (ice);
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}
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/**
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* gst_webrtc_ice_set_turn_server:
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* @ice: The #GstWebRTCICE
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* @uri: (nullable): URI of the TURN sever
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*
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* Since: 1.22
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*/
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void
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gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice, const gchar * uri_s)
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{
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g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server);
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GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server (ice, uri_s);
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}
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/**
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* gst_webrtc_ice_get_turn_server:
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* @ice: The #GstWebRTCICE
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*
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* Returns: (nullable): URI of the TURN sever
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*
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* Since: 1.22
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*/
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gchar *
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gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice)
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{
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g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
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g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server);
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return GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server (ice);
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}
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/**
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* gst_webrtc_ice_set_http_proxy:
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* @ice: The #GstWebRTCICE
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* @uri: (transfer none): URI of the HTTP proxy of the form
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* http://[username:password@]hostname[:port][?alpn=<alpn>]
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*
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* Set HTTP Proxy to be used when connecting to TURN server.
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*
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* Since: 1.22
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|
*/
|
|
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)
|
|
{
|
|
}
|