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1894293d63
SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
238 lines
7.6 KiB
C
238 lines
7.6 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:gstwebrtc-dtlstransport
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* @short_description: RTCDtlsTransport object
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* @title: GstWebRTCDTLSTransport
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCICETransport
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcdtlstransport">https://www.w3.org/TR/webrtc/#rtcdtlstransport</ulink>
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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 "dtlstransport.h"
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#define GST_CAT_DEFAULT gst_webrtc_dtls_transport_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define gst_webrtc_dtls_transport_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCDTLSTransport, gst_webrtc_dtls_transport,
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GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_dtls_transport_debug,
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"dtlstransport", 0, "dtlstransport");
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);
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enum
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{
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SIGNAL_0,
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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_SESSION_ID,
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PROP_TRANSPORT,
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PROP_STATE,
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PROP_CLIENT,
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PROP_CERTIFICATE,
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PROP_REMOTE_CERTIFICATE,
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PROP_RTCP,
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};
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void
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gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
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GstWebRTCICETransport * ice)
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{
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_ICE_TRANSPORT (ice));
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gst_object_replace ((GstObject **) & transport->transport, GST_OBJECT (ice));
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}
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static void
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gst_webrtc_dtls_transport_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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switch (prop_id) {
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case PROP_SESSION_ID:
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webrtc->session_id = g_value_get_uint (value);
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break;
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case PROP_CLIENT:
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g_object_set_property (G_OBJECT (webrtc->dtlssrtpenc), "is-client",
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value);
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gst_element_set_locked_state (webrtc->dtlssrtpenc, FALSE);
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gst_element_sync_state_with_parent (webrtc->dtlssrtpenc);
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break;
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case PROP_CERTIFICATE:
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g_object_set_property (G_OBJECT (webrtc->dtlssrtpdec), "pem", value);
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break;
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case PROP_RTCP:
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webrtc->is_rtcp = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_dtls_transport_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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switch (prop_id) {
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case PROP_SESSION_ID:
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g_value_set_uint (value, webrtc->session_id);
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break;
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case PROP_TRANSPORT:
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g_value_set_object (value, webrtc->transport);
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break;
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case PROP_STATE:
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g_value_set_enum (value, webrtc->state);
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break;
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case PROP_CLIENT:
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g_object_get_property (G_OBJECT (webrtc->dtlssrtpenc), "is-client",
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value);
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break;
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case PROP_CERTIFICATE:
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g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "pem", value);
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break;
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case PROP_REMOTE_CERTIFICATE:
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g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "peer-pem", value);
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break;
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case PROP_RTCP:
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g_value_set_boolean (value, webrtc->is_rtcp);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_dtls_transport_finalize (GObject * object)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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if (webrtc->transport) {
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gst_object_unref (webrtc->transport);
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}
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webrtc->transport = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_webrtc_dtls_transport_constructed (GObject * object)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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gchar *connection_id;
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/* XXX: this may collide with another connection_id however this is only a
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* problem if multiple dtls element sets are being used within the same
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* process */
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connection_id = g_strdup_printf ("%s_%u_%u", webrtc->is_rtcp ? "rtcp" : "rtp",
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webrtc->session_id, g_random_int ());
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webrtc->dtlssrtpenc = gst_element_factory_make ("dtlssrtpenc", NULL);
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g_object_set (webrtc->dtlssrtpenc, "connection-id", connection_id,
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"is-client", webrtc->client, NULL);
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webrtc->dtlssrtpdec = gst_element_factory_make ("dtlssrtpdec", NULL);
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g_object_set (webrtc->dtlssrtpdec, "connection-id", connection_id, NULL);
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g_free (connection_id);
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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gst_webrtc_dtls_transport_class_init (GstWebRTCDTLSTransportClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->constructed = gst_webrtc_dtls_transport_constructed;
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gobject_class->get_property = gst_webrtc_dtls_transport_get_property;
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gobject_class->set_property = gst_webrtc_dtls_transport_set_property;
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gobject_class->finalize = gst_webrtc_dtls_transport_finalize;
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g_object_class_install_property (gobject_class,
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PROP_SESSION_ID,
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g_param_spec_uint ("session-id", "Session ID",
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"Unique session ID", 0, G_MAXUINT, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_TRANSPORT,
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g_param_spec_object ("transport", "ICE transport",
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"ICE transport used by this dtls transport",
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GST_TYPE_WEBRTC_ICE_TRANSPORT,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/* FIXME: implement */
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g_object_class_install_property (gobject_class,
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PROP_STATE,
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g_param_spec_enum ("state", "DTLS state",
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"State of the DTLS transport",
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GST_TYPE_WEBRTC_DTLS_TRANSPORT_STATE,
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GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_CLIENT,
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g_param_spec_boolean ("client", "DTLS client",
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"Are we the client in the DTLS handshake?", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_CERTIFICATE,
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g_param_spec_string ("certificate", "DTLS certificate",
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"DTLS certificate", NULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_REMOTE_CERTIFICATE,
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g_param_spec_string ("remote-certificate", "Remote DTLS certificate",
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"Remote DTLS certificate", NULL,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_RTCP,
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g_param_spec_boolean ("rtcp", "RTCP",
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"The transport is being used solely for RTCP", FALSE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_dtls_transport_init (GstWebRTCDTLSTransport * webrtc)
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{
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}
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GstWebRTCDTLSTransport *
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gst_webrtc_dtls_transport_new (guint session_id, gboolean is_rtcp)
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{
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return g_object_new (GST_TYPE_WEBRTC_DTLS_TRANSPORT, "session-id", session_id,
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"rtcp", is_rtcp, NULL);
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}
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