Generate bindings for the new GstWebRTC library

32 files changed, 3589 insertions, 6 deletions

diff --git a/girs/GstWebRTC-1.0.gir b/girs/GstWebRTC-1.0.gir
new file mode 100644
index 0000000000..951089f479
--- /dev/null
+++ b/girs/GstWebRTC-1.0.gir
@@ -0,0 +1,1003 @@
+<?xml version="1.0"?>
+<!-- This file was automatically generated from C sources - DO NOT EDIT!
+To affect the contents of this file, edit the original C definitions,
+and/or use gtk-doc annotations.  -->
+<repository version="1.2"
+            xmlns="http://www.gtk.org/introspection/core/1.0"
+            xmlns:c="http://www.gtk.org/introspection/c/1.0"
+            xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
+  <include name="Gst" version="1.0"/>
+  <include name="GstSdp" version="1.0"/>
+  <package name="gstreamer-webrtc-1.0"/>
+  <c:include name="gst/webrtc/webrtc.h"/>
+  <namespace name="GstWebRTC"
+             version="1.0"
+             shared-library="libgstwebrtc-1.0.so.0"
+             c:identifier-prefixes="Gst"
+             c:symbol-prefixes="gst">
+    <enumeration name="WebRTCDTLSSetup"
+                 glib:type-name="GstWebRTCDTLSSetup"
+                 glib:get-type="gst_webrtc_dtls_setup_get_type"
+                 c:type="GstWebRTCDTLSSetup">
+      <doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
+GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
+GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
+GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
+      <member name="none"
+              value="0"
+              c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
+              glib:nick="none">
+      </member>
+      <member name="actpass"
+              value="1"
+              c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
+              glib:nick="actpass">
+      </member>
+      <member name="active"
+              value="2"
+              c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
+              glib:nick="active">
+      </member>
+      <member name="passive"
+              value="3"
+              c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
+              glib:nick="passive">
+      </member>
+    </enumeration>
+    <class name="WebRTCDTLSTransport"
+           c:symbol-prefix="webrtc_dtls_transport"
+           c:type="GstWebRTCDTLSTransport"
+           parent="Gst.Object"
+           glib:type-name="GstWebRTCDTLSTransport"
+           glib:get-type="gst_webrtc_dtls_transport_get_type"
+           glib:type-struct="WebRTCDTLSTransportClass">
+      <constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
+        <return-value transfer-ownership="none">
+          <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+        </return-value>
+        <parameters>
+          <parameter name="session_id" transfer-ownership="none">
+            <type name="guint" c:type="guint"/>
+          </parameter>
+          <parameter name="rtcp" transfer-ownership="none">
+            <type name="gboolean" c:type="gboolean"/>
+          </parameter>
+        </parameters>
+      </constructor>
+      <method name="set_transport"
+              c:identifier="gst_webrtc_dtls_transport_set_transport">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="transport" transfer-ownership="none">
+            <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+          </instance-parameter>
+          <parameter name="ice" transfer-ownership="none">
+            <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <property name="certificate" writable="1" transfer-ownership="none">
+        <type name="utf8" c:type="gchar*"/>
+      </property>
+      <property name="client" writable="1" transfer-ownership="none">
+        <type name="gboolean" c:type="gboolean"/>
+      </property>
+      <property name="remote-certificate" transfer-ownership="none">
+        <type name="utf8" c:type="gchar*"/>
+      </property>
+      <property name="rtcp"
+                writable="1"
+                construct-only="1"
+                transfer-ownership="none">
+        <type name="gboolean" c:type="gboolean"/>
+      </property>
+      <property name="session-id"
+                writable="1"
+                construct-only="1"
+                transfer-ownership="none">
+        <type name="guint" c:type="guint"/>
+      </property>
+      <property name="state" transfer-ownership="none">
+        <type name="WebRTCDTLSTransportState"/>
+      </property>
+      <property name="transport" transfer-ownership="none">
+        <type name="WebRTCICETransport"/>
+      </property>
+      <field name="parent">
+        <type name="Gst.Object" c:type="GstObject"/>
+      </field>
+      <field name="transport">
+        <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+      </field>
+      <field name="state">
+        <type name="WebRTCDTLSTransportState"
+              c:type="GstWebRTCDTLSTransportState"/>
+      </field>
+      <field name="is_rtcp">
+        <type name="gboolean" c:type="gboolean"/>
+      </field>
+      <field name="client">
+        <type name="gboolean" c:type="gboolean"/>
+      </field>
+      <field name="session_id">
+        <type name="guint" c:type="guint"/>
+      </field>
+      <field name="dtlssrtpenc">
+        <type name="Gst.Element" c:type="GstElement*"/>
+      </field>
+      <field name="dtlssrtpdec">
+        <type name="Gst.Element" c:type="GstElement*"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </class>
+    <record name="WebRTCDTLSTransportClass"
+            c:type="GstWebRTCDTLSTransportClass"
+            glib:is-gtype-struct-for="WebRTCDTLSTransport">
+      <field name="parent_class">
+        <type name="Gst.BinClass" c:type="GstBinClass"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </record>
+    <enumeration name="WebRTCDTLSTransportState"
+                 glib:type-name="GstWebRTCDTLSTransportState"
+                 glib:get-type="gst_webrtc_dtls_transport_state_get_type"
+                 c:type="GstWebRTCDTLSTransportState">
+      <doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
+GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
+GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
+GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
+GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
+      <member name="new"
+              value="0"
+              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
+              glib:nick="new">
+      </member>
+      <member name="closed"
+              value="1"
+              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
+              glib:nick="closed">
+      </member>
+      <member name="failed"
+              value="2"
+              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
+              glib:nick="failed">
+      </member>
+      <member name="connecting"
+              value="3"
+              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
+              glib:nick="connecting">
+      </member>
+      <member name="connected"
+              value="4"
+              c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
+              glib:nick="connected">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCFECType"
+                 glib:type-name="GstWebRTCFECType"
+                 glib:get-type="gst_webrtc_fec_type_get_type"
+                 c:type="GstWebRTCFECType">
+      <doc xml:space="preserve">GST_WEBRTC_FEC_TYPE_NONE: none
+GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red</doc>
+      <member name="none"
+              value="0"
+              c:identifier="GST_WEBRTC_FEC_TYPE_NONE"
+              glib:nick="none">
+      </member>
+      <member name="ulp_red"
+              value="1"
+              c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED"
+              glib:nick="ulp-red">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCICEComponent"
+                 glib:type-name="GstWebRTCICEComponent"
+                 glib:get-type="gst_webrtc_ice_component_get_type"
+                 c:type="GstWebRTCICEComponent">
+      <doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
+GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
+      <member name="rtp"
+              value="0"
+              c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP"
+              glib:nick="rtp">
+      </member>
+      <member name="rtcp"
+              value="1"
+              c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
+              glib:nick="rtcp">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCICEConnectionState"
+                 glib:type-name="GstWebRTCICEConnectionState"
+                 glib:get-type="gst_webrtc_ice_connection_state_get_type"
+                 c:type="GstWebRTCICEConnectionState">
+      <doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
+GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
+GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
+GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
+GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
+GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
+GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
+See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate&lt;/ulink&gt;</doc>
+      <member name="new"
+              value="0"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
+              glib:nick="new">
+      </member>
+      <member name="checking"
+              value="1"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING"
+              glib:nick="checking">
+      </member>
+      <member name="connected"
+              value="2"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED"
+              glib:nick="connected">
+      </member>
+      <member name="completed"
+              value="3"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED"
+              glib:nick="completed">
+      </member>
+      <member name="failed"
+              value="4"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED"
+              glib:nick="failed">
+      </member>
+      <member name="disconnected"
+              value="5"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED"
+              glib:nick="disconnected">
+      </member>
+      <member name="closed"
+              value="6"
+              c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
+              glib:nick="closed">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCICEGatheringState"
+                 glib:type-name="GstWebRTCICEGatheringState"
+                 glib:get-type="gst_webrtc_ice_gathering_state_get_type"
+                 c:type="GstWebRTCICEGatheringState">
+      <doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
+GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
+GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
+See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&lt;/ulink&gt;</doc>
+      <member name="new"
+              value="0"
+              c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW"
+              glib:nick="new">
+      </member>
+      <member name="gathering"
+              value="1"
+              c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING"
+              glib:nick="gathering">
+      </member>
+      <member name="complete"
+              value="2"
+              c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
+              glib:nick="complete">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCICERole"
+                 glib:type-name="GstWebRTCICERole"
+                 glib:get-type="gst_webrtc_ice_role_get_type"
+                 c:type="GstWebRTCICERole">
+      <doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
+GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
+      <member name="controlled"
+              value="0"
+              c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
+              glib:nick="controlled">
+      </member>
+      <member name="controlling"
+              value="1"
+              c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
+              glib:nick="controlling">
+      </member>
+    </enumeration>
+    <class name="WebRTCICETransport"
+           c:symbol-prefix="webrtc_ice_transport"
+           c:type="GstWebRTCICETransport"
+           parent="Gst.Object"
+           abstract="1"
+           glib:type-name="GstWebRTCICETransport"
+           glib:get-type="gst_webrtc_ice_transport_get_type"
+           glib:type-struct="WebRTCICETransportClass">
+      <virtual-method name="gather_candidates">
+        <return-value transfer-ownership="none">
+          <type name="gboolean" c:type="gboolean"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="transport" transfer-ownership="none">
+            <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+          </instance-parameter>
+        </parameters>
+      </virtual-method>
+      <method name="connection_state_change"
+              c:identifier="gst_webrtc_ice_transport_connection_state_change">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="ice" transfer-ownership="none">
+            <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+          </instance-parameter>
+          <parameter name="new_state" transfer-ownership="none">
+            <type name="WebRTCICEConnectionState"
+                  c:type="GstWebRTCICEConnectionState"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="gathering_state_change"
+              c:identifier="gst_webrtc_ice_transport_gathering_state_change">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="ice" transfer-ownership="none">
+            <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+          </instance-parameter>
+          <parameter name="new_state" transfer-ownership="none">
+            <type name="WebRTCICEGatheringState"
+                  c:type="GstWebRTCICEGatheringState"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="new_candidate"
+              c:identifier="gst_webrtc_ice_transport_new_candidate">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="ice" transfer-ownership="none">
+            <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+          </instance-parameter>
+          <parameter name="stream_id" transfer-ownership="none">
+            <type name="guint" c:type="guint"/>
+          </parameter>
+          <parameter name="component" transfer-ownership="none">
+            <type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
+          </parameter>
+          <parameter name="attr" transfer-ownership="none">
+            <type name="utf8" c:type="gchar*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="selected_pair_change"
+              c:identifier="gst_webrtc_ice_transport_selected_pair_change">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="ice" transfer-ownership="none">
+            <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
+      <property name="component"
+                writable="1"
+                construct-only="1"
+                transfer-ownership="none">
+        <type name="WebRTCICEComponent"/>
+      </property>
+      <property name="gathering-state" transfer-ownership="none">
+        <type name="WebRTCICEGatheringState"/>
+      </property>
+      <property name="state" transfer-ownership="none">
+        <type name="WebRTCICEConnectionState"/>
+      </property>
+      <field name="parent">
+        <type name="Gst.Object" c:type="GstObject"/>
+      </field>
+      <field name="role">
+        <type name="WebRTCICERole" c:type="GstWebRTCICERole"/>
+      </field>
+      <field name="component">
+        <type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
+      </field>
+      <field name="state">
+        <type name="WebRTCICEConnectionState"
+              c:type="GstWebRTCICEConnectionState"/>
+      </field>
+      <field name="gathering_state">
+        <type name="WebRTCICEGatheringState"
+              c:type="GstWebRTCICEGatheringState"/>
+      </field>
+      <field name="src">
+        <type name="Gst.Element" c:type="GstElement*"/>
+      </field>
+      <field name="sink">
+        <type name="Gst.Element" c:type="GstElement*"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+      <glib:signal name="on-new-candidate" when="last">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <parameter name="object" transfer-ownership="none">
+            <type name="utf8" c:type="gchar*"/>
+          </parameter>
+        </parameters>
+      </glib:signal>
+      <glib:signal name="on-selected-candidate-pair-change" when="last">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+      </glib:signal>
+    </class>
+    <record name="WebRTCICETransportClass"
+            c:type="GstWebRTCICETransportClass"
+            glib:is-gtype-struct-for="WebRTCICETransport">
+      <field name="parent_class">
+        <type name="Gst.BinClass" c:type="GstBinClass"/>
+      </field>
+      <field name="gather_candidates">
+        <callback name="gather_candidates">
+          <return-value transfer-ownership="none">
+            <type name="gboolean" c:type="gboolean"/>
+          </return-value>
+          <parameters>
+            <parameter name="transport" transfer-ownership="none">
+              <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
+            </parameter>
+          </parameters>
+        </callback>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </record>
+    <enumeration name="WebRTCPeerConnectionState"
+                 glib:type-name="GstWebRTCPeerConnectionState"
+                 glib:get-type="gst_webrtc_peer_connection_state_get_type"
+                 c:type="GstWebRTCPeerConnectionState">
+      <doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
+GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
+GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
+GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
+GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
+GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
+See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate&lt;/ulink&gt;</doc>
+      <member name="new"
+              value="0"
+              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW"
+              glib:nick="new">
+      </member>
+      <member name="connecting"
+              value="1"
+              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING"
+              glib:nick="connecting">
+      </member>
+      <member name="connected"
+              value="2"
+              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED"
+              glib:nick="connected">
+      </member>
+      <member name="disconnected"
+              value="3"
+              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED"
+              glib:nick="disconnected">
+      </member>
+      <member name="failed"
+              value="4"
+              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED"
+              glib:nick="failed">
+      </member>
+      <member name="closed"
+              value="5"
+              c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
+              glib:nick="closed">
+      </member>
+    </enumeration>
+    <class name="WebRTCRTPReceiver"
+           c:symbol-prefix="webrtc_rtp_receiver"
+           c:type="GstWebRTCRTPReceiver"
+           parent="Gst.Object"
+           glib:type-name="GstWebRTCRTPReceiver"
+           glib:get-type="gst_webrtc_rtp_receiver_get_type"
+           glib:type-struct="WebRTCRTPReceiverClass">
+      <constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
+        <return-value transfer-ownership="none">
+          <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
+        </return-value>
+      </constructor>
+      <method name="set_rtcp_transport"
+              c:identifier="gst_webrtc_rtp_receiver_set_rtcp_transport">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="receiver" transfer-ownership="none">
+            <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
+          </instance-parameter>
+          <parameter name="transport" transfer-ownership="none">
+            <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="set_transport"
+              c:identifier="gst_webrtc_rtp_receiver_set_transport">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="receiver" transfer-ownership="none">
+            <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
+          </instance-parameter>
+          <parameter name="transport" transfer-ownership="none">
+            <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <field name="parent">
+        <type name="Gst.Object" c:type="GstObject"/>
+      </field>
+      <field name="transport">
+        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+      </field>
+      <field name="rtcp_transport">
+        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </class>
+    <record name="WebRTCRTPReceiverClass"
+            c:type="GstWebRTCRTPReceiverClass"
+            glib:is-gtype-struct-for="WebRTCRTPReceiver">
+      <field name="parent_class">
+        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </record>
+    <class name="WebRTCRTPSender"
+           c:symbol-prefix="webrtc_rtp_sender"
+           c:type="GstWebRTCRTPSender"
+           parent="Gst.Object"
+           glib:type-name="GstWebRTCRTPSender"
+           glib:get-type="gst_webrtc_rtp_sender_get_type"
+           glib:type-struct="WebRTCRTPSenderClass">
+      <constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
+        <return-value transfer-ownership="none">
+          <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
+        </return-value>
+      </constructor>
+      <method name="set_rtcp_transport"
+              c:identifier="gst_webrtc_rtp_sender_set_rtcp_transport">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="sender" transfer-ownership="none">
+            <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
+          </instance-parameter>
+          <parameter name="transport" transfer-ownership="none">
+            <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="set_transport"
+              c:identifier="gst_webrtc_rtp_sender_set_transport">
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="sender" transfer-ownership="none">
+            <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
+          </instance-parameter>
+          <parameter name="transport" transfer-ownership="none">
+            <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <field name="parent">
+        <type name="Gst.Object" c:type="GstObject"/>
+      </field>
+      <field name="transport">
+        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+      </field>
+      <field name="rtcp_transport">
+        <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
+      </field>
+      <field name="send_encodings">
+        <array name="GLib.Array" c:type="GArray*">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </class>
+    <record name="WebRTCRTPSenderClass"
+            c:type="GstWebRTCRTPSenderClass"
+            glib:is-gtype-struct-for="WebRTCRTPSender">
+      <field name="parent_class">
+        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </record>
+    <class name="WebRTCRTPTransceiver"
+           c:symbol-prefix="webrtc_rtp_transceiver"
+           c:type="GstWebRTCRTPTransceiver"
+           parent="Gst.Object"
+           abstract="1"
+           glib:type-name="GstWebRTCRTPTransceiver"
+           glib:get-type="gst_webrtc_rtp_transceiver_get_type"
+           glib:type-struct="WebRTCRTPTransceiverClass">
+      <property name="mlineindex"
+                writable="1"
+                construct-only="1"
+                transfer-ownership="none">
+        <type name="guint" c:type="guint"/>
+      </property>
+      <property name="receiver"
+                writable="1"
+                construct-only="1"
+                transfer-ownership="none">
+        <type name="WebRTCRTPReceiver"/>
+      </property>
+      <property name="sender"
+                writable="1"
+                construct-only="1"
+                transfer-ownership="none">
+        <type name="WebRTCRTPSender"/>
+      </property>
+      <field name="parent">
+        <type name="Gst.Object" c:type="GstObject"/>
+      </field>
+      <field name="mline">
+        <type name="guint" c:type="guint"/>
+      </field>
+      <field name="mid">
+        <type name="utf8" c:type="gchar*"/>
+      </field>
+      <field name="stopped">
+        <type name="gboolean" c:type="gboolean"/>
+      </field>
+      <field name="sender">
+        <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
+      </field>
+      <field name="receiver">
+        <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
+      </field>
+      <field name="direction">
+        <type name="WebRTCRTPTransceiverDirection"
+              c:type="GstWebRTCRTPTransceiverDirection"/>
+      </field>
+      <field name="current_direction">
+        <type name="WebRTCRTPTransceiverDirection"
+              c:type="GstWebRTCRTPTransceiverDirection"/>
+      </field>
+      <field name="codec_preferences">
+        <type name="Gst.Caps" c:type="GstCaps*"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </class>
+    <record name="WebRTCRTPTransceiverClass"
+            c:type="GstWebRTCRTPTransceiverClass"
+            glib:is-gtype-struct-for="WebRTCRTPTransceiver">
+      <field name="parent_class">
+        <type name="Gst.ObjectClass" c:type="GstObjectClass"/>
+      </field>
+      <field name="_padding">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </record>
+    <enumeration name="WebRTCRTPTransceiverDirection"
+                 glib:type-name="GstWebRTCRTPTransceiverDirection"
+                 glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type"
+                 c:type="GstWebRTCRTPTransceiverDirection">
+      <member name="none"
+              value="0"
+              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE"
+              glib:nick="none">
+      </member>
+      <member name="inactive"
+              value="1"
+              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE"
+              glib:nick="inactive">
+      </member>
+      <member name="sendonly"
+              value="2"
+              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY"
+              glib:nick="sendonly">
+      </member>
+      <member name="recvonly"
+              value="3"
+              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY"
+              glib:nick="recvonly">
+      </member>
+      <member name="sendrecv"
+              value="4"
+              c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
+              glib:nick="sendrecv">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCSDPType"
+                 glib:type-name="GstWebRTCSDPType"
+                 glib:get-type="gst_webrtc_sdp_type_get_type"
+                 c:type="GstWebRTCSDPType">
+      <doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
+GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
+GST_WEBRTC_SDP_TYPE_ANSWER: answer
+GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
+See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.github.io/webrtc-pc/#rtcsdptype&lt;/ulink&gt;</doc>
+      <member name="offer"
+              value="1"
+              c:identifier="GST_WEBRTC_SDP_TYPE_OFFER"
+              glib:nick="offer">
+      </member>
+      <member name="pranswer"
+              value="2"
+              c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
+              glib:nick="pranswer">
+      </member>
+      <member name="answer"
+              value="3"
+              c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
+              glib:nick="answer">
+      </member>
+      <member name="rollback"
+              value="4"
+              c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
+              glib:nick="rollback">
+      </member>
+      <function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
+     recognized.</doc>
+          <type name="utf8" c:type="const gchar*"/>
+        </return-value>
+        <parameters>
+          <parameter name="type" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstWebRTCSDPType</doc>
+            <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
+          </parameter>
+        </parameters>
+      </function>
+    </enumeration>
+    <record name="WebRTCSessionDescription"
+            c:type="GstWebRTCSessionDescription"
+            glib:type-name="GstWebRTCSessionDescription"
+            glib:get-type="gst_webrtc_session_description_get_type"
+            c:symbol-prefix="webrtc_session_description">
+      <doc xml:space="preserve">See &lt;ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class"&gt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&lt;/ulink&gt;</doc>
+      <field name="type" writable="1">
+        <doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
+        <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
+      </field>
+      <field name="sdp" writable="1">
+        <doc xml:space="preserve">the #GstSDPMessage of the description</doc>
+        <type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
+      </field>
+      <constructor name="new"
+                   c:identifier="gst_webrtc_session_description_new">
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a new #GstWebRTCSessionDescription from @type
+     and @sdp</doc>
+          <type name="WebRTCSessionDescription"
+                c:type="GstWebRTCSessionDescription*"/>
+        </return-value>
+        <parameters>
+          <parameter name="type" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstWebRTCSDPType</doc>
+            <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
+          </parameter>
+          <parameter name="sdp" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstSDPMessage</doc>
+            <type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
+          </parameter>
+        </parameters>
+      </constructor>
+      <method name="copy" c:identifier="gst_webrtc_session_description_copy">
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a new copy of @src</doc>
+          <type name="WebRTCSessionDescription"
+                c:type="GstWebRTCSessionDescription*"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="src" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstWebRTCSessionDescription</doc>
+            <type name="WebRTCSessionDescription"
+                  c:type="const GstWebRTCSessionDescription*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
+      <method name="free" c:identifier="gst_webrtc_session_description_free">
+        <doc xml:space="preserve">Free @desc and all associated resources</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="desc" transfer-ownership="full">
+            <doc xml:space="preserve">a #GstWebRTCSessionDescription</doc>
+            <type name="WebRTCSessionDescription"
+                  c:type="GstWebRTCSessionDescription*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
+    </record>
+    <enumeration name="WebRTCSignalingState"
+                 glib:type-name="GstWebRTCSignalingState"
+                 glib:get-type="gst_webrtc_signaling_state_get_type"
+                 c:type="GstWebRTCSignalingState">
+      <doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
+GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
+GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
+GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
+GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
+GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
+See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate&lt;/ulink&gt;</doc>
+      <member name="stable"
+              value="0"
+              c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE"
+              glib:nick="stable">
+      </member>
+      <member name="closed"
+              value="1"
+              c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED"
+              glib:nick="closed">
+      </member>
+      <member name="have_local_offer"
+              value="2"
+              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER"
+              glib:nick="have-local-offer">
+      </member>
+      <member name="have_remote_offer"
+              value="3"
+              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER"
+              glib:nick="have-remote-offer">
+      </member>
+      <member name="have_local_pranswer"
+              value="4"
+              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER"
+              glib:nick="have-local-pranswer">
+      </member>
+      <member name="have_remote_pranswer"
+              value="5"
+              c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
+              glib:nick="have-remote-pranswer">
+      </member>
+    </enumeration>
+    <enumeration name="WebRTCStatsType"
+                 glib:type-name="GstWebRTCStatsType"
+                 glib:get-type="gst_webrtc_stats_type_get_type"
+                 c:type="GstWebRTCStatsType">
+      <doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
+GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
+GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
+GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
+GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
+GST_WEBRTC_STATS_CSRC: csrc
+GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
+GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
+GST_WEBRTC_STATS_STREAM: stream
+GST_WEBRTC_STATS_TRANSPORT: transport
+GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
+GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
+GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
+GST_WEBRTC_STATS_CERTIFICATE: certificate</doc>
+      <member name="codec"
+              value="1"
+              c:identifier="GST_WEBRTC_STATS_CODEC"
+              glib:nick="codec">
+      </member>
+      <member name="inbound_rtp"
+              value="2"
+              c:identifier="GST_WEBRTC_STATS_INBOUND_RTP"
+              glib:nick="inbound-rtp">
+      </member>
+      <member name="outbound_rtp"
+              value="3"
+              c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP"
+              glib:nick="outbound-rtp">
+      </member>
+      <member name="remote_inbound_rtp"
+              value="4"
+              c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
+              glib:nick="remote-inbound-rtp">
+      </member>
+      <member name="remote_outbound_rtp"
+              value="5"
+              c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
+              glib:nick="remote-outbound-rtp">
+      </member>
+      <member name="csrc"
+              value="6"
+              c:identifier="GST_WEBRTC_STATS_CSRC"
+              glib:nick="csrc">
+      </member>
+      <member name="peer_connection"
+              value="7"
+              c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION"
+              glib:nick="peer-connection">
+      </member>
+      <member name="data_channel"
+              value="8"
+              c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
+              glib:nick="data-channel">
+      </member>
+      <member name="stream"
+              value="9"
+              c:identifier="GST_WEBRTC_STATS_STREAM"
+              glib:nick="stream">
+      </member>
+      <member name="transport"
+              value="10"
+              c:identifier="GST_WEBRTC_STATS_TRANSPORT"
+              glib:nick="transport">
+      </member>
+      <member name="candidate_pair"
+              value="11"
+              c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR"
+              glib:nick="candidate-pair">
+      </member>
+      <member name="local_candidate"
+              value="12"
+              c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE"
+              glib:nick="local-candidate">
+      </member>
+      <member name="remote_candidate"
+              value="13"
+              c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
+              glib:nick="remote-candidate">
+      </member>
+      <member name="certificate"
+              value="14"
+              c:identifier="GST_WEBRTC_STATS_CERTIFICATE"
+              glib:nick="certificate">
+      </member>
+    </enumeration>
+    <function name="webrtc_sdp_type_to_string"
+              c:identifier="gst_webrtc_sdp_type_to_string"
+              moved-to="WebRTCSDPType.to_string">
+      <return-value transfer-ownership="none">
+        <doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
+     recognized.</doc>
+        <type name="utf8" c:type="const gchar*"/>
+      </return-value>
+      <parameters>
+        <parameter name="type" transfer-ownership="none">
+          <doc xml:space="preserve">a #GstWebRTCSDPType</doc>
+          <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
+        </parameter>
+      </parameters>
+    </function>
+  </namespace>
+</repository>
diff --git a/meson.build b/meson.build
index 0b92311654..978aff217d 100644
--- a/meson.build
+++ b/meson.build
@@ -79,7 +79,9 @@ gst_deps_defs = [
   ['gstreamer-rtsp', ['gst-plugins-base', 'rtsp_dep'], 'gst_rtsp'],
   ['gstreamer-sdp', ['gst-plugins-base', 'sdp_dep'], 'gstsdp'],
   ['gstreamer-tag', ['gst-plugins-base', 'tag_dep'], 'gsttag'],
-  ['gstreamer-video', ['gst-plugins-base', 'video_dep'], 'gstvideo'],]
+  ['gstreamer-video', ['gst-plugins-base', 'video_dep'], 'gstvideo'],
+  ['gstreamer-webrtc', ['gst-plugins-bad', 'gstwebrtc_dep'], 'gstwebrtc'],
+]
 
 foreach dep: gst_deps_defs
   gst_deps += [dependency(dep.get(0) + '-' + apiversion, version: gst_required_version,
@@ -165,7 +167,7 @@ if bindinator.get_variable('found')
     run_target('bindinate_gstreamer',
         command: [bindinate,
             '--name=gstreamer', '--regenerate=true',
-            '--merge-with=GstApp-1.0,GstAudio-1.0,GstBase-1.0,GstController-1.0,GstNet-1.0,GstPbutils-1.0,GstRtp-1.0,GstRtsp-1.0,GstSdp-1.0,GstTag-1.0,GstVideo-1.0',
+            '--merge-with=GstApp-1.0,GstAudio-1.0,GstBase-1.0,GstController-1.0,GstNet-1.0,GstPbutils-1.0,GstRtp-1.0,GstRtsp-1.0,GstSdp-1.0,GstTag-1.0,GstVideo-1.0,GstWebRTC-1.0',
             '--gir=Gst-1.0',
             '--copy-girs=@0@'.format(join_paths(meson.current_source_dir(), 'girs'))],
         depends: []
@@ -183,4 +185,4 @@ if bindinator.get_variable('found')
     run_target('update-all', command: [find_program('update_sources.py'), 'bindinate'])
 else
     warning('Bindinator not usable as some required dependencies are not avalaible.')
-endif
\ No newline at end of file
+endif
diff --git a/sources/custom/Application.cs b/sources/custom/Application.cs
index 4ac9abb47c..6a5f022622 100644
--- a/sources/custom/Application.cs
+++ b/sources/custom/Application.cs
@@ -32,7 +32,8 @@ namespace Gst {
 			GLib.GType.Register (FractionRange.GType, typeof(FractionRange));
 			GLib.GType.Register (DateTime.GType, typeof(DateTime));
 			GLib.GType.Register (Gst.Array.GType, typeof(Gst.Array));
-
+			GLib.GType.Register(Promise.GType, typeof(Promise));
+			GLib.GType.Register(Gst.WebRTC.WebRTCSessionDescription.GType, typeof(Gst.WebRTC.WebRTCSessionDescription));
 		}
 
 		[DllImport("libgstreamer-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
diff --git a/sources/generated/Gst.WebRTC/Constants.cs b/sources/generated/Gst.WebRTC/Constants.cs
new file mode 100644
index 0000000000..640f6820c2
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/Constants.cs
@@ -0,0 +1,16 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class Constants {
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/Global.cs b/sources/generated/Gst.WebRTC/Global.cs
new file mode 100644
index 0000000000..460a1cc078
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/Global.cs
@@ -0,0 +1,25 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class Global {
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_sdp_type_to_string(int type);
+
+		public static string WebrtcSdpTypeToString(Gst.WebRTC.WebRTCSDPType type) {
+			IntPtr raw_ret = gst_webrtc_sdp_type_to_string((int) type);
+			string ret = GLib.Marshaller.Utf8PtrToString (raw_ret);
+			return ret;
+		}
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/OnNewCandidateHandler.cs b/sources/generated/Gst.WebRTC/OnNewCandidateHandler.cs
new file mode 100644
index 0000000000..42d9524004
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/OnNewCandidateHandler.cs
@@ -0,0 +1,18 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+
+	public delegate void OnNewCandidateHandler(object o, OnNewCandidateArgs args);
+
+	public class OnNewCandidateArgs : GLib.SignalArgs {
+		public string Object{
+			get {
+				return (string) Args [0];
+			}
+		}
+
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCDTLSSetup.cs b/sources/generated/Gst.WebRTC/WebRTCDTLSSetup.cs
new file mode 100644
index 0000000000..208f6bd472
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCDTLSSetup.cs
@@ -0,0 +1,30 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCDTLSSetupGType))]
+	public enum WebRTCDTLSSetup {
+
+		None = 0,
+		Actpass = 1,
+		Active = 2,
+		Passive = 3,
+	}
+
+	internal class WebRTCDTLSSetupGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_dtls_setup_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_dtls_setup_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs b/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs
new file mode 100644
index 0000000000..bcbdde6af4
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs
@@ -0,0 +1,333 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class WebRTCDTLSTransport : Gst.Object {
+
+		public WebRTCDTLSTransport (IntPtr raw) : base(raw) {}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_dtls_transport_new(uint session_id, bool rtcp);
+
+		public WebRTCDTLSTransport (uint session_id, bool rtcp) : base (IntPtr.Zero)
+		{
+			if (GetType () != typeof (WebRTCDTLSTransport)) {
+				var vals = new List<GLib.Value> ();
+				var names = new List<string> ();
+				names.Add ("session_id");
+				vals.Add (new GLib.Value (session_id));
+				names.Add ("rtcp");
+				vals.Add (new GLib.Value (rtcp));
+				CreateNativeObject (names.ToArray (), vals.ToArray ());
+				return;
+			}
+			Raw = gst_webrtc_dtls_transport_new(session_id, rtcp);
+		}
+
+		[GLib.Property ("certificate")]
+		public string Certificate {
+			get {
+				GLib.Value val = GetProperty ("certificate");
+				string ret = (string) val;
+				val.Dispose ();
+				return ret;
+			}
+			set {
+				GLib.Value val = new GLib.Value(value);
+				SetProperty("certificate", val);
+				val.Dispose ();
+			}
+		}
+
+		[GLib.Property ("client")]
+		public bool Client {
+			get {
+				GLib.Value val = GetProperty ("client");
+				bool ret = (bool) val;
+				val.Dispose ();
+				return ret;
+			}
+			set {
+				GLib.Value val = new GLib.Value(value);
+				SetProperty("client", val);
+				val.Dispose ();
+			}
+		}
+
+		[GLib.Property ("remote-certificate")]
+		public string RemoteCertificate {
+			get {
+				GLib.Value val = GetProperty ("remote-certificate");
+				string ret = (string) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("rtcp")]
+		public bool Rtcp {
+			get {
+				GLib.Value val = GetProperty ("rtcp");
+				bool ret = (bool) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("session-id")]
+		public uint SessionId {
+			get {
+				GLib.Value val = GetProperty ("session-id");
+				uint ret = (uint) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("state")]
+		public Gst.WebRTC.WebRTCDTLSTransportState State {
+			get {
+				GLib.Value val = GetProperty ("state");
+				Gst.WebRTC.WebRTCDTLSTransportState ret = (Gst.WebRTC.WebRTCDTLSTransportState) (Enum) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_dtls_transport_set_transport(IntPtr raw, IntPtr ice);
+
+		[GLib.Property ("transport")]
+		public Gst.WebRTC.WebRTCICETransport Transport {
+			get {
+				GLib.Value val = GetProperty ("transport");
+				Gst.WebRTC.WebRTCICETransport ret = (Gst.WebRTC.WebRTCICETransport) val;
+				val.Dispose ();
+				return ret;
+			}
+			set  {
+				gst_webrtc_dtls_transport_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+			}
+		}
+
+		public Gst.WebRTC.WebRTCICETransport TransportField {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCICETransport;
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCDTLSTransportState StateField {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("state"));
+					return (Gst.WebRTC.WebRTCDTLSTransportState) (*raw_ptr);
+				}
+			}
+		}
+
+		public bool IsRtcp {
+			get {
+				unsafe {
+					bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("is_rtcp"));
+					return (*raw_ptr);
+				}
+			}
+		}
+
+		public bool ClientField {
+			get {
+				unsafe {
+					bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("client"));
+					return (*raw_ptr);
+				}
+			}
+		}
+
+		public uint SessionIdField {
+			get {
+				unsafe {
+					uint* raw_ptr = (uint*)(((byte*)Handle) + abi_info.GetFieldOffset("session_id"));
+					return (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.Element Dtlssrtpenc {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("dtlssrtpenc"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+				}
+			}
+		}
+
+		public Gst.Element Dtlssrtpdec {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("dtlssrtpdec"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+				}
+			}
+		}
+
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _class_abi = null;
+		static public new GLib.AbiStruct class_abi {
+			get {
+				if (_class_abi == null)
+					_class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("_padding"
+							, Gst.Object.class_abi.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, null
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _class_abi;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_dtls_transport_get_type();
+
+		public static new GLib.GType GType { 
+			get {
+				IntPtr raw_ret = gst_webrtc_dtls_transport_get_type();
+				GLib.GType ret = new GLib.GType(raw_ret);
+				return ret;
+			}
+		}
+
+
+		static WebRTCDTLSTransport ()
+		{
+			GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+		}
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _abi_info = null;
+		static public new GLib.AbiStruct abi_info {
+			get {
+				if (_abi_info == null)
+					_abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("transport"
+							, Gst.Object.abi_info.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
+							, null
+							, "state"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("state"
+							, -1
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCDTLSTransportState))) // state
+							, "transport"
+							, "is_rtcp"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_stateAlign), "state")
+							, 0
+							),
+						new GLib.AbiField("is_rtcp"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(bool)) // is_rtcp
+							, "state"
+							, "client"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_is_rtcpAlign), "is_rtcp")
+							, 0
+							),
+						new GLib.AbiField("client"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(bool)) // client
+							, "is_rtcp"
+							, "session_id"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_clientAlign), "client")
+							, 0
+							),
+						new GLib.AbiField("session_id"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(uint)) // session_id
+							, "client"
+							, "dtlssrtpenc"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_session_idAlign), "session_id")
+							, 0
+							),
+						new GLib.AbiField("dtlssrtpenc"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // dtlssrtpenc
+							, "session_id"
+							, "dtlssrtpdec"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("dtlssrtpdec"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // dtlssrtpdec
+							, "dtlssrtpenc"
+							, "_padding"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("_padding"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, "dtlssrtpdec"
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _abi_info;
+			}
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCDTLSTransport_stateAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCDTLSTransportState state;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCDTLSTransport_is_rtcpAlign
+		{
+			sbyte f1;
+			private bool is_rtcp;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCDTLSTransport_clientAlign
+		{
+			sbyte f1;
+			private bool client;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCDTLSTransport_session_idAlign
+		{
+			sbyte f1;
+			private uint session_id;
+		}
+
+
+		// End of the ABI representation.
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCDTLSTransportState.cs b/sources/generated/Gst.WebRTC/WebRTCDTLSTransportState.cs
new file mode 100644
index 0000000000..dae707ccc6
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCDTLSTransportState.cs
@@ -0,0 +1,31 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCDTLSTransportStateGType))]
+	public enum WebRTCDTLSTransportState {
+
+		New = 0,
+		Closed = 1,
+		Failed = 2,
+		Connecting = 3,
+		Connected = 4,
+	}
+
+	internal class WebRTCDTLSTransportStateGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_dtls_transport_state_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_dtls_transport_state_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCICEComponent.cs b/sources/generated/Gst.WebRTC/WebRTCICEComponent.cs
new file mode 100644
index 0000000000..925bda9626
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCICEComponent.cs
@@ -0,0 +1,28 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCICEComponentGType))]
+	public enum WebRTCICEComponent {
+
+		Rtp = 0,
+		Rtcp = 1,
+	}
+
+	internal class WebRTCICEComponentGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_ice_component_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_ice_component_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCICEConnectionState.cs b/sources/generated/Gst.WebRTC/WebRTCICEConnectionState.cs
new file mode 100644
index 0000000000..f894208eb5
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCICEConnectionState.cs
@@ -0,0 +1,33 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCICEConnectionStateGType))]
+	public enum WebRTCICEConnectionState {
+
+		New = 0,
+		Checking = 1,
+		Connected = 2,
+		Completed = 3,
+		Failed = 4,
+		Disconnected = 5,
+		Closed = 6,
+	}
+
+	internal class WebRTCICEConnectionStateGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_ice_connection_state_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_ice_connection_state_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCICEGatheringState.cs b/sources/generated/Gst.WebRTC/WebRTCICEGatheringState.cs
new file mode 100644
index 0000000000..73f3975fcb
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCICEGatheringState.cs
@@ -0,0 +1,29 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCICEGatheringStateGType))]
+	public enum WebRTCICEGatheringState {
+
+		New = 0,
+		Gathering = 1,
+		Complete = 2,
+	}
+
+	internal class WebRTCICEGatheringStateGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_ice_gathering_state_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_ice_gathering_state_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCICERole.cs b/sources/generated/Gst.WebRTC/WebRTCICERole.cs
new file mode 100644
index 0000000000..921d6377ce
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCICERole.cs
@@ -0,0 +1,28 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCICERoleGType))]
+	public enum WebRTCICERole {
+
+		Controlled = 0,
+		Controlling = 1,
+	}
+
+	internal class WebRTCICERoleGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_ice_role_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_ice_role_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCICETransport.cs b/sources/generated/Gst.WebRTC/WebRTCICETransport.cs
new file mode 100644
index 0000000000..886fd40f00
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCICETransport.cs
@@ -0,0 +1,463 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class WebRTCICETransport : Gst.Object {
+
+		protected WebRTCICETransport (IntPtr raw) : base(raw) {}
+
+		protected WebRTCICETransport() : base(IntPtr.Zero)
+		{
+			CreateNativeObject (new string [0], new GLib.Value [0]);
+		}
+
+		[GLib.Property ("component")]
+		public Gst.WebRTC.WebRTCICEComponent Component {
+			get {
+				GLib.Value val = GetProperty ("component");
+				Gst.WebRTC.WebRTCICEComponent ret = (Gst.WebRTC.WebRTCICEComponent) (Enum) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("gathering-state")]
+		public Gst.WebRTC.WebRTCICEGatheringState GatheringState {
+			get {
+				GLib.Value val = GetProperty ("gathering-state");
+				Gst.WebRTC.WebRTCICEGatheringState ret = (Gst.WebRTC.WebRTCICEGatheringState) (Enum) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("state")]
+		public Gst.WebRTC.WebRTCICEConnectionState State {
+			get {
+				GLib.Value val = GetProperty ("state");
+				Gst.WebRTC.WebRTCICEConnectionState ret = (Gst.WebRTC.WebRTCICEConnectionState) (Enum) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		public Gst.WebRTC.WebRTCICERole Role {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("role"));
+					return (Gst.WebRTC.WebRTCICERole) (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCICEComponent ComponentField {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("component"));
+					return (Gst.WebRTC.WebRTCICEComponent) (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCICEConnectionState StateField {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("state"));
+					return (Gst.WebRTC.WebRTCICEConnectionState) (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCICEGatheringState GatheringStateField {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("gathering_state"));
+					return (Gst.WebRTC.WebRTCICEGatheringState) (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.Element Src {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("src"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+				}
+			}
+		}
+
+		public Gst.Element Sink {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("sink"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
+				}
+			}
+		}
+
+		[GLib.Signal("on-selected-candidate-pair-change")]
+		public event System.EventHandler OnSelectedCandidatePairChange {
+			add {
+				this.AddSignalHandler ("on-selected-candidate-pair-change", value);
+			}
+			remove {
+				this.RemoveSignalHandler ("on-selected-candidate-pair-change", value);
+			}
+		}
+
+		[GLib.Signal("on-new-candidate")]
+		public event Gst.WebRTC.OnNewCandidateHandler OnNewCandidate {
+			add {
+				this.AddSignalHandler ("on-new-candidate", value, typeof (Gst.WebRTC.OnNewCandidateArgs));
+			}
+			remove {
+				this.RemoveSignalHandler ("on-new-candidate", value);
+			}
+		}
+
+		static OnNewCandidateNativeDelegate OnNewCandidate_cb_delegate;
+		static OnNewCandidateNativeDelegate OnNewCandidateVMCallback {
+			get {
+				if (OnNewCandidate_cb_delegate == null)
+					OnNewCandidate_cb_delegate = new OnNewCandidateNativeDelegate (OnNewCandidate_cb);
+				return OnNewCandidate_cb_delegate;
+			}
+		}
+
+		static void OverrideOnNewCandidate (GLib.GType gtype)
+		{
+			OverrideOnNewCandidate (gtype, OnNewCandidateVMCallback);
+		}
+
+		static void OverrideOnNewCandidate (GLib.GType gtype, OnNewCandidateNativeDelegate callback)
+		{
+			OverrideVirtualMethod (gtype, "on-new-candidate", callback);
+		}
+		[UnmanagedFunctionPointer (CallingConvention.Cdecl)]
+		delegate void OnNewCandidateNativeDelegate (IntPtr inst, IntPtr _object);
+
+		static void OnNewCandidate_cb (IntPtr inst, IntPtr _object)
+		{
+			try {
+				WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
+				__obj.OnOnNewCandidate (GLib.Marshaller.Utf8PtrToString (_object));
+			} catch (Exception e) {
+				GLib.ExceptionManager.RaiseUnhandledException (e, false);
+			}
+		}
+
+		[GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideOnNewCandidate")]
+		protected virtual void OnOnNewCandidate (string _object)
+		{
+			InternalOnNewCandidate (_object);
+		}
+
+		private void InternalOnNewCandidate (string _object)
+		{
+			GLib.Value ret = GLib.Value.Empty;
+			GLib.ValueArray inst_and_params = new GLib.ValueArray (2);
+			GLib.Value[] vals = new GLib.Value [2];
+			vals [0] = new GLib.Value (this);
+			inst_and_params.Append (vals [0]);
+			vals [1] = new GLib.Value (_object);
+			inst_and_params.Append (vals [1]);
+			g_signal_chain_from_overridden (inst_and_params.ArrayPtr, ref ret);
+			foreach (GLib.Value v in vals)
+				v.Dispose ();
+		}
+
+		static OnSelectedCandidatePairChangeNativeDelegate OnSelectedCandidatePairChange_cb_delegate;
+		static OnSelectedCandidatePairChangeNativeDelegate OnSelectedCandidatePairChangeVMCallback {
+			get {
+				if (OnSelectedCandidatePairChange_cb_delegate == null)
+					OnSelectedCandidatePairChange_cb_delegate = new OnSelectedCandidatePairChangeNativeDelegate (OnSelectedCandidatePairChange_cb);
+				return OnSelectedCandidatePairChange_cb_delegate;
+			}
+		}
+
+		static void OverrideOnSelectedCandidatePairChange (GLib.GType gtype)
+		{
+			OverrideOnSelectedCandidatePairChange (gtype, OnSelectedCandidatePairChangeVMCallback);
+		}
+
+		static void OverrideOnSelectedCandidatePairChange (GLib.GType gtype, OnSelectedCandidatePairChangeNativeDelegate callback)
+		{
+			OverrideVirtualMethod (gtype, "on-selected-candidate-pair-change", callback);
+		}
+		[UnmanagedFunctionPointer (CallingConvention.Cdecl)]
+		delegate void OnSelectedCandidatePairChangeNativeDelegate (IntPtr inst);
+
+		static void OnSelectedCandidatePairChange_cb (IntPtr inst)
+		{
+			try {
+				WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
+				__obj.OnOnSelectedCandidatePairChange ();
+			} catch (Exception e) {
+				GLib.ExceptionManager.RaiseUnhandledException (e, false);
+			}
+		}
+
+		[GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideOnSelectedCandidatePairChange")]
+		protected virtual void OnOnSelectedCandidatePairChange ()
+		{
+			InternalOnSelectedCandidatePairChange ();
+		}
+
+		private void InternalOnSelectedCandidatePairChange ()
+		{
+			GLib.Value ret = GLib.Value.Empty;
+			GLib.ValueArray inst_and_params = new GLib.ValueArray (1);
+			GLib.Value[] vals = new GLib.Value [1];
+			vals [0] = new GLib.Value (this);
+			inst_and_params.Append (vals [0]);
+			g_signal_chain_from_overridden (inst_and_params.ArrayPtr, ref ret);
+			foreach (GLib.Value v in vals)
+				v.Dispose ();
+		}
+
+		static GatherCandidatesNativeDelegate GatherCandidates_cb_delegate;
+		static GatherCandidatesNativeDelegate GatherCandidatesVMCallback {
+			get {
+				if (GatherCandidates_cb_delegate == null)
+					GatherCandidates_cb_delegate = new GatherCandidatesNativeDelegate (GatherCandidates_cb);
+				return GatherCandidates_cb_delegate;
+			}
+		}
+
+		static void OverrideGatherCandidates (GLib.GType gtype)
+		{
+			OverrideGatherCandidates (gtype, GatherCandidatesVMCallback);
+		}
+
+		static void OverrideGatherCandidates (GLib.GType gtype, GatherCandidatesNativeDelegate callback)
+		{
+			unsafe {
+				IntPtr* raw_ptr = (IntPtr*)(((long) gtype.GetClassPtr()) + (long) class_abi.GetFieldOffset("gather_candidates"));
+				*raw_ptr = Marshal.GetFunctionPointerForDelegate((Delegate) callback);
+			}
+		}
+
+		[UnmanagedFunctionPointer (CallingConvention.Cdecl)]
+		delegate bool GatherCandidatesNativeDelegate (IntPtr inst);
+
+		static bool GatherCandidates_cb (IntPtr inst)
+		{
+			try {
+				WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
+				bool __result;
+				__result = __obj.OnGatherCandidates ();
+				return __result;
+			} catch (Exception e) {
+				GLib.ExceptionManager.RaiseUnhandledException (e, true);
+				// NOTREACHED: above call does not return.
+				throw e;
+			}
+		}
+
+		[GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideGatherCandidates")]
+		protected virtual bool OnGatherCandidates ()
+		{
+			return InternalGatherCandidates ();
+		}
+
+		private bool InternalGatherCandidates ()
+		{
+			GatherCandidatesNativeDelegate unmanaged = null;
+			unsafe {
+				IntPtr* raw_ptr = (IntPtr*)(((long) this.LookupGType().GetThresholdType().GetClassPtr()) + (long) class_abi.GetFieldOffset("gather_candidates"));
+				unmanaged = (GatherCandidatesNativeDelegate) Marshal.GetDelegateForFunctionPointer(*raw_ptr, typeof(GatherCandidatesNativeDelegate));
+			}
+			if (unmanaged == null) return false;
+
+			bool __result = unmanaged (this.Handle);
+			return __result;
+		}
+
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _class_abi = null;
+		static public new GLib.AbiStruct class_abi {
+			get {
+				if (_class_abi == null)
+					_class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("gather_candidates"
+							, Gst.Object.class_abi.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // gather_candidates
+							, null
+							, "_padding"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("_padding"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, "gather_candidates"
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _class_abi;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_ice_transport_get_type();
+
+		public static new GLib.GType GType { 
+			get {
+				IntPtr raw_ret = gst_webrtc_ice_transport_get_type();
+				GLib.GType ret = new GLib.GType(raw_ret);
+				return ret;
+			}
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_ice_transport_connection_state_change(IntPtr raw, int new_state);
+
+		public void ConnectionStateChange(Gst.WebRTC.WebRTCICEConnectionState new_state) {
+			gst_webrtc_ice_transport_connection_state_change(Handle, (int) new_state);
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_ice_transport_gathering_state_change(IntPtr raw, int new_state);
+
+		public void GatheringStateChange(Gst.WebRTC.WebRTCICEGatheringState new_state) {
+			gst_webrtc_ice_transport_gathering_state_change(Handle, (int) new_state);
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_ice_transport_new_candidate(IntPtr raw, uint stream_id, int component, IntPtr attr);
+
+		public void NewCandidate(uint stream_id, Gst.WebRTC.WebRTCICEComponent component, string attr) {
+			IntPtr native_attr = GLib.Marshaller.StringToPtrGStrdup (attr);
+			gst_webrtc_ice_transport_new_candidate(Handle, stream_id, (int) component, native_attr);
+			GLib.Marshaller.Free (native_attr);
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_ice_transport_selected_pair_change(IntPtr raw);
+
+		public void SelectedPairChange() {
+			gst_webrtc_ice_transport_selected_pair_change(Handle);
+		}
+
+
+		static WebRTCICETransport ()
+		{
+			GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+		}
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _abi_info = null;
+		static public new GLib.AbiStruct abi_info {
+			get {
+				if (_abi_info == null)
+					_abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("role"
+							, Gst.Object.abi_info.Fields
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICERole))) // role
+							, null
+							, "component"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_roleAlign), "role")
+							, 0
+							),
+						new GLib.AbiField("component"
+							, -1
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEComponent))) // component
+							, "role"
+							, "state"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_componentAlign), "component")
+							, 0
+							),
+						new GLib.AbiField("state"
+							, -1
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEConnectionState))) // state
+							, "component"
+							, "gathering_state"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_stateAlign), "state")
+							, 0
+							),
+						new GLib.AbiField("gathering_state"
+							, -1
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEGatheringState))) // gathering_state
+							, "state"
+							, "src"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_gathering_stateAlign), "gathering_state")
+							, 0
+							),
+						new GLib.AbiField("src"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // src
+							, "gathering_state"
+							, "sink"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("sink"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // sink
+							, "src"
+							, "_padding"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("_padding"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, "sink"
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _abi_info;
+			}
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCICETransport_roleAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCICERole role;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCICETransport_componentAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCICEComponent component;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCICETransport_stateAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCICEConnectionState state;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCICETransport_gathering_stateAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCICEGatheringState gathering_state;
+		}
+
+
+		// End of the ABI representation.
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCPeerConnectionState.cs b/sources/generated/Gst.WebRTC/WebRTCPeerConnectionState.cs
new file mode 100644
index 0000000000..3b3524f994
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCPeerConnectionState.cs
@@ -0,0 +1,32 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCPeerConnectionStateGType))]
+	public enum WebRTCPeerConnectionState {
+
+		New = 0,
+		Connecting = 1,
+		Connected = 2,
+		Disconnected = 3,
+		Failed = 4,
+		Closed = 5,
+	}
+
+	internal class WebRTCPeerConnectionStateGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_peer_connection_state_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_peer_connection_state_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs b/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs
new file mode 100644
index 0000000000..4ed2981c8c
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs
@@ -0,0 +1,140 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class WebRTCRTPReceiver : Gst.Object {
+
+		public WebRTCRTPReceiver (IntPtr raw) : base(raw) {}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_rtp_receiver_new();
+
+		public WebRTCRTPReceiver () : base (IntPtr.Zero)
+		{
+			if (GetType () != typeof (WebRTCRTPReceiver)) {
+				CreateNativeObject (new string [0], new GLib.Value[0]);
+				return;
+			}
+			Raw = gst_webrtc_rtp_receiver_new();
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_rtp_receiver_set_transport(IntPtr raw, IntPtr transport);
+
+		public Gst.WebRTC.WebRTCDTLSTransport Transport {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+				}
+			}
+			set  {
+				gst_webrtc_rtp_receiver_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+			}
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_rtp_receiver_set_rtcp_transport(IntPtr raw, IntPtr transport);
+
+		public Gst.WebRTC.WebRTCDTLSTransport RtcpTransport {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("rtcp_transport"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+				}
+			}
+			set  {
+				gst_webrtc_rtp_receiver_set_rtcp_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+			}
+		}
+
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _class_abi = null;
+		static public new GLib.AbiStruct class_abi {
+			get {
+				if (_class_abi == null)
+					_class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("_padding"
+							, Gst.Object.class_abi.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, null
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _class_abi;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_rtp_receiver_get_type();
+
+		public static new GLib.GType GType { 
+			get {
+				IntPtr raw_ret = gst_webrtc_rtp_receiver_get_type();
+				GLib.GType ret = new GLib.GType(raw_ret);
+				return ret;
+			}
+		}
+
+
+		static WebRTCRTPReceiver ()
+		{
+			GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+		}
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _abi_info = null;
+		static public new GLib.AbiStruct abi_info {
+			get {
+				if (_abi_info == null)
+					_abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("transport"
+							, Gst.Object.abi_info.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
+							, null
+							, "rtcp_transport"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("rtcp_transport"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // rtcp_transport
+							, "transport"
+							, "_padding"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("_padding"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, "rtcp_transport"
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _abi_info;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs b/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
new file mode 100644
index 0000000000..d6b4924c72
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
@@ -0,0 +1,148 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class WebRTCRTPSender : Gst.Object {
+
+		public WebRTCRTPSender (IntPtr raw) : base(raw) {}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_rtp_sender_new();
+
+		public WebRTCRTPSender () : base (IntPtr.Zero)
+		{
+			if (GetType () != typeof (WebRTCRTPSender)) {
+				CreateNativeObject (new string [0], new GLib.Value[0]);
+				return;
+			}
+			Raw = gst_webrtc_rtp_sender_new();
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_rtp_sender_set_transport(IntPtr raw, IntPtr transport);
+
+		public Gst.WebRTC.WebRTCDTLSTransport Transport {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+				}
+			}
+			set  {
+				gst_webrtc_rtp_sender_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+			}
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern void gst_webrtc_rtp_sender_set_rtcp_transport(IntPtr raw, IntPtr transport);
+
+		public Gst.WebRTC.WebRTCDTLSTransport RtcpTransport {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("rtcp_transport"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
+				}
+			}
+			set  {
+				gst_webrtc_rtp_sender_set_rtcp_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
+			}
+		}
+
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _class_abi = null;
+		static public new GLib.AbiStruct class_abi {
+			get {
+				if (_class_abi == null)
+					_class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("_padding"
+							, Gst.Object.class_abi.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, null
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _class_abi;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_rtp_sender_get_type();
+
+		public static new GLib.GType GType { 
+			get {
+				IntPtr raw_ret = gst_webrtc_rtp_sender_get_type();
+				GLib.GType ret = new GLib.GType(raw_ret);
+				return ret;
+			}
+		}
+
+
+		static WebRTCRTPSender ()
+		{
+			GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+		}
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _abi_info = null;
+		static public new GLib.AbiStruct abi_info {
+			get {
+				if (_abi_info == null)
+					_abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("transport"
+							, Gst.Object.abi_info.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
+							, null
+							, "rtcp_transport"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("rtcp_transport"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // rtcp_transport
+							, "transport"
+							, "send_encodings"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("send_encodings"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // send_encodings
+							, "rtcp_transport"
+							, "_padding"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("_padding"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, "send_encodings"
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _abi_info;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs b/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs
new file mode 100644
index 0000000000..af4436cfda
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs
@@ -0,0 +1,281 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	public partial class WebRTCRTPTransceiver : Gst.Object {
+
+		protected WebRTCRTPTransceiver (IntPtr raw) : base(raw) {}
+
+		protected WebRTCRTPTransceiver() : base(IntPtr.Zero)
+		{
+			CreateNativeObject (new string [0], new GLib.Value [0]);
+		}
+
+		[GLib.Property ("mlineindex")]
+		public uint Mlineindex {
+			get {
+				GLib.Value val = GetProperty ("mlineindex");
+				uint ret = (uint) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("receiver")]
+		public Gst.WebRTC.WebRTCRTPReceiver Receiver {
+			get {
+				GLib.Value val = GetProperty ("receiver");
+				Gst.WebRTC.WebRTCRTPReceiver ret = (Gst.WebRTC.WebRTCRTPReceiver) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		[GLib.Property ("sender")]
+		public Gst.WebRTC.WebRTCRTPSender Sender {
+			get {
+				GLib.Value val = GetProperty ("sender");
+				Gst.WebRTC.WebRTCRTPSender ret = (Gst.WebRTC.WebRTCRTPSender) val;
+				val.Dispose ();
+				return ret;
+			}
+		}
+
+		public uint Mline {
+			get {
+				unsafe {
+					uint* raw_ptr = (uint*)(((byte*)Handle) + abi_info.GetFieldOffset("mline"));
+					return (*raw_ptr);
+				}
+			}
+		}
+
+		public string Mid {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("mid"));
+					return GLib.Marshaller.Utf8PtrToString ((*raw_ptr));
+				}
+			}
+		}
+
+		public bool Stopped {
+			get {
+				unsafe {
+					bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("stopped"));
+					return (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCRTPSender SenderField {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("sender"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCRTPSender;
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCRTPReceiver ReceiverField {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("receiver"));
+					return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCRTPReceiver;
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCRTPTransceiverDirection Direction {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("direction"));
+					return (Gst.WebRTC.WebRTCRTPTransceiverDirection) (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.WebRTC.WebRTCRTPTransceiverDirection CurrentDirection {
+			get {
+				unsafe {
+					int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("current_direction"));
+					return (Gst.WebRTC.WebRTCRTPTransceiverDirection) (*raw_ptr);
+				}
+			}
+		}
+
+		public Gst.Caps CodecPreferences {
+			get {
+				unsafe {
+					IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("codec_preferences"));
+					return (*raw_ptr) == IntPtr.Zero ? null : (Gst.Caps) GLib.Opaque.GetOpaque ((*raw_ptr), typeof (Gst.Caps), false);
+				}
+			}
+		}
+
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _class_abi = null;
+		static public new GLib.AbiStruct class_abi {
+			get {
+				if (_class_abi == null)
+					_class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("_padding"
+							, Gst.Object.class_abi.Fields
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, null
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _class_abi;
+			}
+		}
+
+
+		// End of the ABI representation.
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_rtp_transceiver_get_type();
+
+		public static new GLib.GType GType { 
+			get {
+				IntPtr raw_ret = gst_webrtc_rtp_transceiver_get_type();
+				GLib.GType ret = new GLib.GType(raw_ret);
+				return ret;
+			}
+		}
+
+
+		static WebRTCRTPTransceiver ()
+		{
+			GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
+		}
+
+		// Internal representation of the wrapped structure ABI.
+		static GLib.AbiStruct _abi_info = null;
+		static public new GLib.AbiStruct abi_info {
+			get {
+				if (_abi_info == null)
+					_abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{ 
+						new GLib.AbiField("mline"
+							, Gst.Object.abi_info.Fields
+							, (uint) Marshal.SizeOf(typeof(uint)) // mline
+							, null
+							, "mid"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_mlineAlign), "mline")
+							, 0
+							),
+						new GLib.AbiField("mid"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // mid
+							, "mline"
+							, "stopped"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("stopped"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(bool)) // stopped
+							, "mid"
+							, "sender"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_stoppedAlign), "stopped")
+							, 0
+							),
+						new GLib.AbiField("sender"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // sender
+							, "stopped"
+							, "receiver"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("receiver"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // receiver
+							, "sender"
+							, "direction"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("direction"
+							, -1
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCRTPTransceiverDirection))) // direction
+							, "receiver"
+							, "current_direction"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_directionAlign), "direction")
+							, 0
+							),
+						new GLib.AbiField("current_direction"
+							, -1
+							, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCRTPTransceiverDirection))) // current_direction
+							, "direction"
+							, "codec_preferences"
+							, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_current_directionAlign), "current_direction")
+							, 0
+							),
+						new GLib.AbiField("codec_preferences"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) // codec_preferences
+							, "current_direction"
+							, "_padding"
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+						new GLib.AbiField("_padding"
+							, -1
+							, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
+							, "codec_preferences"
+							, null
+							, (uint) Marshal.SizeOf(typeof(IntPtr))
+							, 0
+							),
+					});
+
+				return _abi_info;
+			}
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCRTPTransceiver_mlineAlign
+		{
+			sbyte f1;
+			private uint mline;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCRTPTransceiver_stoppedAlign
+		{
+			sbyte f1;
+			private bool stopped;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCRTPTransceiver_directionAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCRTPTransceiverDirection direction;
+		}
+
+		[StructLayout(LayoutKind.Sequential)]
+		public struct GstWebRTCRTPTransceiver_current_directionAlign
+		{
+			sbyte f1;
+			private Gst.WebRTC.WebRTCRTPTransceiverDirection current_direction;
+		}
+
+
+		// End of the ABI representation.
+
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPTransceiverDirection.cs b/sources/generated/Gst.WebRTC/WebRTCRTPTransceiverDirection.cs
new file mode 100644
index 0000000000..9c1e68fd2d
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPTransceiverDirection.cs
@@ -0,0 +1,31 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCRTPTransceiverDirectionGType))]
+	public enum WebRTCRTPTransceiverDirection {
+
+		None = 0,
+		Inactive = 1,
+		Sendonly = 2,
+		Recvonly = 3,
+		Sendrecv = 4,
+	}
+
+	internal class WebRTCRTPTransceiverDirectionGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_rtp_transceiver_direction_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_rtp_transceiver_direction_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCSDPType.cs b/sources/generated/Gst.WebRTC/WebRTCSDPType.cs
new file mode 100644
index 0000000000..b39f5673bd
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCSDPType.cs
@@ -0,0 +1,30 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCSDPTypeGType))]
+	public enum WebRTCSDPType {
+
+		Offer = 1,
+		Pranswer = 2,
+		Answer = 3,
+		Rollback = 4,
+	}
+
+	internal class WebRTCSDPTypeGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_sdp_type_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_sdp_type_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCSessionDescription.cs b/sources/generated/Gst.WebRTC/WebRTCSessionDescription.cs
new file mode 100644
index 0000000000..c34ed23358
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCSessionDescription.cs
@@ -0,0 +1,83 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Collections;
+	using System.Collections.Generic;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[StructLayout(LayoutKind.Sequential)]
+	public partial struct WebRTCSessionDescription : IEquatable<WebRTCSessionDescription> {
+
+		public Gst.WebRTC.WebRTCSDPType Type;
+		private IntPtr _sdp;
+		public Gst.Sdp.SDPMessage Sdp {
+			get {
+				return _sdp == IntPtr.Zero ? null : (Gst.Sdp.SDPMessage) GLib.Opaque.GetOpaque (_sdp, typeof (Gst.Sdp.SDPMessage), false);
+			}
+			set {
+				_sdp = value == null ? IntPtr.Zero : value.Handle;
+			}
+		}
+
+		public static Gst.WebRTC.WebRTCSessionDescription Zero = new Gst.WebRTC.WebRTCSessionDescription ();
+
+		public static Gst.WebRTC.WebRTCSessionDescription New(IntPtr raw) {
+			if (raw == IntPtr.Zero)
+				return Gst.WebRTC.WebRTCSessionDescription.Zero;
+			return (Gst.WebRTC.WebRTCSessionDescription) Marshal.PtrToStructure (raw, typeof (Gst.WebRTC.WebRTCSessionDescription));
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_session_description_new(int type, IntPtr sdp);
+
+		public static WebRTCSessionDescription New(Gst.WebRTC.WebRTCSDPType type, Gst.Sdp.SDPMessage sdp)
+		{
+			WebRTCSessionDescription result = WebRTCSessionDescription.New (gst_webrtc_session_description_new((int) type, sdp == null ? IntPtr.Zero : sdp.Handle));
+			return result;
+		}
+
+		[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_session_description_get_type();
+
+		public static GLib.GType GType { 
+			get {
+				IntPtr raw_ret = gst_webrtc_session_description_get_type();
+				GLib.GType ret = new GLib.GType(raw_ret);
+				return ret;
+			}
+		}
+
+		public bool Equals (WebRTCSessionDescription other)
+		{
+			return true && Type.Equals (other.Type) && Sdp.Equals (other.Sdp);
+		}
+
+		public override bool Equals (object other)
+		{
+			return other is WebRTCSessionDescription && Equals ((WebRTCSessionDescription) other);
+		}
+
+		public override int GetHashCode ()
+		{
+			return this.GetType ().FullName.GetHashCode () ^ Type.GetHashCode () ^ Sdp.GetHashCode ();
+		}
+
+		public static explicit operator GLib.Value (Gst.WebRTC.WebRTCSessionDescription boxed)
+		{
+			GLib.Value val = GLib.Value.Empty;
+			val.Init (Gst.WebRTC.WebRTCSessionDescription.GType);
+			val.Val = boxed;
+			return val;
+		}
+
+		public static explicit operator Gst.WebRTC.WebRTCSessionDescription (GLib.Value val)
+		{
+			return (Gst.WebRTC.WebRTCSessionDescription) val.Val;
+		}
+#endregion
+	}
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCSignalingState.cs b/sources/generated/Gst.WebRTC/WebRTCSignalingState.cs
new file mode 100644
index 0000000000..ccad44b223
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCSignalingState.cs
@@ -0,0 +1,32 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCSignalingStateGType))]
+	public enum WebRTCSignalingState {
+
+		Stable = 0,
+		Closed = 1,
+		HaveLocalOffer = 2,
+		HaveRemoteOffer = 3,
+		HaveLocalPranswer = 4,
+		HaveRemotePranswer = 5,
+	}
+
+	internal class WebRTCSignalingStateGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_signaling_state_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_signaling_state_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCStatsType.cs b/sources/generated/Gst.WebRTC/WebRTCStatsType.cs
new file mode 100644
index 0000000000..b8916f4415
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCStatsType.cs
@@ -0,0 +1,40 @@
+// This file was generated by the Gtk# code generator.
+// Any changes made will be lost if regenerated.
+
+namespace Gst.WebRTC {
+
+	using System;
+	using System.Runtime.InteropServices;
+
+#region Autogenerated code
+	[GLib.GType (typeof (Gst.WebRTC.WebRTCStatsTypeGType))]
+	public enum WebRTCStatsType {
+
+		Codec = 1,
+		InboundRtp = 2,
+		OutboundRtp = 3,
+		RemoteInboundRtp = 4,
+		RemoteOutboundRtp = 5,
+		Csrc = 6,
+		PeerConnection = 7,
+		DataChannel = 8,
+		Stream = 9,
+		Transport = 10,
+		CandidatePair = 11,
+		LocalCandidate = 12,
+		RemoteCandidate = 13,
+		Certificate = 14,
+	}
+
+	internal class WebRTCStatsTypeGType {
+		[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+		static extern IntPtr gst_webrtc_stats_type_get_type ();
+
+		public static GLib.GType GType {
+			get {
+				return new GLib.GType (gst_webrtc_stats_type_get_type ());
+			}
+		}
+	}
+#endregion
+}
diff --git a/sources/generated/GtkSharp/ObjectManager.cs b/sources/generated/GtkSharp/ObjectManager.cs
index ba47db25c3..6b410a6f96 100644
--- a/sources/generated/GtkSharp/ObjectManager.cs
+++ b/sources/generated/GtkSharp/ObjectManager.cs
@@ -69,6 +69,11 @@ namespace GtkSharp.GstreamerSharp {
 			GLib.GType.Register (Gst.Video.VideoEncoder.GType, typeof (Gst.Video.VideoEncoder));
 			GLib.GType.Register (Gst.Video.VideoFilter.GType, typeof (Gst.Video.VideoFilter));
 			GLib.GType.Register (Gst.Video.VideoSink.GType, typeof (Gst.Video.VideoSink));
+			GLib.GType.Register (Gst.WebRTC.WebRTCDTLSTransport.GType, typeof (Gst.WebRTC.WebRTCDTLSTransport));
+			GLib.GType.Register (Gst.WebRTC.WebRTCICETransport.GType, typeof (Gst.WebRTC.WebRTCICETransport));
+			GLib.GType.Register (Gst.WebRTC.WebRTCRTPReceiver.GType, typeof (Gst.WebRTC.WebRTCRTPReceiver));
+			GLib.GType.Register (Gst.WebRTC.WebRTCRTPSender.GType, typeof (Gst.WebRTC.WebRTCRTPSender));
+			GLib.GType.Register (Gst.WebRTC.WebRTCRTPTransceiver.GType, typeof (Gst.WebRTC.WebRTCRTPTransceiver));
 		}
 	}
 }
diff --git a/sources/generated/gstreamer-sharp-abi.c b/sources/generated/gstreamer-sharp-abi.c
index 1e42011056..5b7f57283e 100644
--- a/sources/generated/gstreamer-sharp-abi.c
+++ b/sources/generated/gstreamer-sharp-abi.c
@@ -21,6 +21,7 @@
 #include <gst/video/video.h>
 #include <gst/video/gstvideoaffinetransformationmeta.h>
 #include <gst/net/gstnetcontrolmessagemeta.h>
+#include <gst/webrtc/webrtc.h>
 
 int main (int argc, char *argv[]) {
 	g_print("\"sizeof(GstAllocatorClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstAllocatorClass));
@@ -944,5 +945,52 @@ int main (int argc, char *argv[]) {
 	g_print("\"GstVideoInfo.fps_d\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, fps_d));
 	g_print("\"GstVideoInfo.offset\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, offset));
 	g_print("\"GstVideoInfo.stride\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, stride));
+	g_print("\"sizeof(GstWebRTCDTLSTransportClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCDTLSTransportClass));
+	g_print("\"GstWebRTCDTLSTransportClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransportClass, _padding));
+	g_print("\"sizeof(GstWebRTCDTLSTransport)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCDTLSTransport));
+	g_print("\"GstWebRTCDTLSTransport.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, transport));
+	g_print("\"GstWebRTCDTLSTransport.state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, state));
+	g_print("\"GstWebRTCDTLSTransport.is_rtcp\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, is_rtcp));
+	g_print("\"GstWebRTCDTLSTransport.client\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, client));
+	g_print("\"GstWebRTCDTLSTransport.session_id\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, session_id));
+	g_print("\"GstWebRTCDTLSTransport.dtlssrtpenc\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, dtlssrtpenc));
+	g_print("\"GstWebRTCDTLSTransport.dtlssrtpdec\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, dtlssrtpdec));
+	g_print("\"GstWebRTCDTLSTransport._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, _padding));
+	g_print("\"sizeof(GstWebRTCICETransportClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCICETransportClass));
+	g_print("\"GstWebRTCICETransportClass.gather_candidates\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransportClass, gather_candidates));
+	g_print("\"GstWebRTCICETransportClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransportClass, _padding));
+	g_print("\"sizeof(GstWebRTCICETransport)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCICETransport));
+	g_print("\"GstWebRTCICETransport.role\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, role));
+	g_print("\"GstWebRTCICETransport.component\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, component));
+	g_print("\"GstWebRTCICETransport.state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, state));
+	g_print("\"GstWebRTCICETransport.gathering_state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, gathering_state));
+	g_print("\"GstWebRTCICETransport.src\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, src));
+	g_print("\"GstWebRTCICETransport.sink\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, sink));
+	g_print("\"GstWebRTCICETransport._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, _padding));
+	g_print("\"sizeof(GstWebRTCRTPReceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPReceiverClass));
+	g_print("\"GstWebRTCRTPReceiverClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiverClass, _padding));
+	g_print("\"sizeof(GstWebRTCRTPReceiver)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPReceiver));
+	g_print("\"GstWebRTCRTPReceiver.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, transport));
+	g_print("\"GstWebRTCRTPReceiver.rtcp_transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, rtcp_transport));
+	g_print("\"GstWebRTCRTPReceiver._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, _padding));
+	g_print("\"sizeof(GstWebRTCRTPSenderClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPSenderClass));
+	g_print("\"GstWebRTCRTPSenderClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSenderClass, _padding));
+	g_print("\"sizeof(GstWebRTCRTPSender)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPSender));
+	g_print("\"GstWebRTCRTPSender.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, transport));
+	g_print("\"GstWebRTCRTPSender.rtcp_transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, rtcp_transport));
+	g_print("\"GstWebRTCRTPSender.send_encodings\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, send_encodings));
+	g_print("\"GstWebRTCRTPSender._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, _padding));
+	g_print("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiverClass));
+	g_print("\"GstWebRTCRTPTransceiverClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiverClass, _padding));
+	g_print("\"sizeof(GstWebRTCRTPTransceiver)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiver));
+	g_print("\"GstWebRTCRTPTransceiver.mline\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, mline));
+	g_print("\"GstWebRTCRTPTransceiver.mid\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, mid));
+	g_print("\"GstWebRTCRTPTransceiver.stopped\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, stopped));
+	g_print("\"GstWebRTCRTPTransceiver.sender\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, sender));
+	g_print("\"GstWebRTCRTPTransceiver.receiver\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, receiver));
+	g_print("\"GstWebRTCRTPTransceiver.direction\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, direction));
+	g_print("\"GstWebRTCRTPTransceiver.current_direction\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, current_direction));
+	g_print("\"GstWebRTCRTPTransceiver.codec_preferences\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, codec_preferences));
+	g_print("\"GstWebRTCRTPTransceiver._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, _padding));
 	return 0;
 }
diff --git a/sources/generated/gstreamer-sharp-abi.cs b/sources/generated/gstreamer-sharp-abi.cs
index 27332da914..df93275720 100644
--- a/sources/generated/gstreamer-sharp-abi.cs
+++ b/sources/generated/gstreamer-sharp-abi.cs
@@ -939,6 +939,53 @@ namespace AbiTester {
 			Console.WriteLine("\"GstVideoInfo.fps_d\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("fps_d") + "\"");
 			Console.WriteLine("\"GstVideoInfo.offset\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("offset") + "\"");
 			Console.WriteLine("\"GstVideoInfo.stride\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("stride") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCDTLSTransportClass)\": \"" + Gst.WebRTC.WebRTCDTLSTransport.class_abi.Size + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransportClass._padding\": \"" + Gst.WebRTC.WebRTCDTLSTransport.class_abi.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCDTLSTransport)\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.Size + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.transport\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("transport") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.state\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("state") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.is_rtcp\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("is_rtcp") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.client\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("client") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.session_id\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("session_id") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.dtlssrtpenc\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("dtlssrtpenc") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport.dtlssrtpdec\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("dtlssrtpdec") + "\"");
+			Console.WriteLine("\"GstWebRTCDTLSTransport._padding\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCICETransportClass)\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.Size + "\"");
+			Console.WriteLine("\"GstWebRTCICETransportClass.gather_candidates\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.GetFieldOffset("gather_candidates") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransportClass._padding\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCICETransport)\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.Size + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport.role\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("role") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport.component\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("component") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport.state\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("state") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport.gathering_state\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("gathering_state") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport.src\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("src") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport.sink\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("sink") + "\"");
+			Console.WriteLine("\"GstWebRTCICETransport._padding\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCRTPReceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPReceiver.class_abi.Size + "\"");
+			Console.WriteLine("\"GstWebRTCRTPReceiverClass._padding\": \"" + Gst.WebRTC.WebRTCRTPReceiver.class_abi.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCRTPReceiver)\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.Size + "\"");
+			Console.WriteLine("\"GstWebRTCRTPReceiver.transport\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("transport") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPReceiver.rtcp_transport\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("rtcp_transport") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPReceiver._padding\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCRTPSenderClass)\": \"" + Gst.WebRTC.WebRTCRTPSender.class_abi.Size + "\"");
+			Console.WriteLine("\"GstWebRTCRTPSenderClass._padding\": \"" + Gst.WebRTC.WebRTCRTPSender.class_abi.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCRTPSender)\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.Size + "\"");
+			Console.WriteLine("\"GstWebRTCRTPSender.transport\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("transport") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPSender.rtcp_transport\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("rtcp_transport") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPSender.send_encodings\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("send_encodings") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPSender._padding\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.Size + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiverClass._padding\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.GetFieldOffset("_padding") + "\"");
+			Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiver)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.Size + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.mline\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mline") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.mid\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mid") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.stopped\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("stopped") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.sender\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("sender") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.receiver\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("receiver") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.direction\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("direction") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.current_direction\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("current_direction") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver.codec_preferences\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("codec_preferences") + "\"");
+			Console.WriteLine("\"GstWebRTCRTPTransceiver._padding\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("_padding") + "\"");
 		}
 	}
 }
diff --git a/sources/generated/gstreamer-sharp-api.xml b/sources/generated/gstreamer-sharp-api.xml
index d7bf61f8aa..53e047e00b 100644
--- a/sources/generated/gstreamer-sharp-api.xml
+++ b/sources/generated/gstreamer-sharp-api.xml
@@ -28814,4 +28814,305 @@
       <constant value="16" ctype="gint" gtype="gint" name="VIDEO_TILE_Y_TILES_SHIFT" />
     </object>
   </namespace>
+  <namespace name="Gst.WebRTC" library="libgstwebrtc-1.0-0.dll">
+    <enum name="WebRTCDTLSSetup" cname="GstWebRTCDTLSSetup" type="enum" gtype="gst_webrtc_dtls_setup_get_type">
+      <member cname="GST_WEBRTC_DTLS_SETUP_NONE" name="None" value="0" />
+      <member cname="GST_WEBRTC_DTLS_SETUP_ACTPASS" name="Actpass" value="1" />
+      <member cname="GST_WEBRTC_DTLS_SETUP_ACTIVE" name="Active" value="2" />
+      <member cname="GST_WEBRTC_DTLS_SETUP_PASSIVE" name="Passive" value="3" />
+    </enum>
+    <enum name="WebRTCDTLSTransportState" cname="GstWebRTCDTLSTransportState" type="enum" gtype="gst_webrtc_dtls_transport_state_get_type">
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" name="New" value="0" />
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" name="Closed" value="1" />
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" name="Failed" value="2" />
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" name="Connecting" value="3" />
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" name="Connected" value="4" />
+    </enum>
+    <enum name="WebRTCICEComponent" cname="GstWebRTCICEComponent" type="enum" gtype="gst_webrtc_ice_component_get_type">
+      <member cname="GST_WEBRTC_ICE_COMPONENT_RTP" name="Rtp" value="0" />
+      <member cname="GST_WEBRTC_ICE_COMPONENT_RTCP" name="Rtcp" value="1" />
+    </enum>
+    <enum name="WebRTCICEConnectionState" cname="GstWebRTCICEConnectionState" type="enum" gtype="gst_webrtc_ice_connection_state_get_type">
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" name="New" value="0" />
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" name="Checking" value="1" />
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" name="Connected" value="2" />
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" name="Completed" value="3" />
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" name="Failed" value="4" />
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="5" />
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" name="Closed" value="6" />
+    </enum>
+    <enum name="WebRTCICEGatheringState" cname="GstWebRTCICEGatheringState" type="enum" gtype="gst_webrtc_ice_gathering_state_get_type">
+      <member cname="GST_WEBRTC_ICE_GATHERING_STATE_NEW" name="New" value="0" />
+      <member cname="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" name="Gathering" value="1" />
+      <member cname="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" name="Complete" value="2" />
+    </enum>
+    <enum name="WebRTCICERole" cname="GstWebRTCICERole" type="enum" gtype="gst_webrtc_ice_role_get_type">
+      <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLED" name="Controlled" value="0" />
+      <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLING" name="Controlling" value="1" />
+    </enum>
+    <enum name="WebRTCPeerConnectionState" cname="GstWebRTCPeerConnectionState" type="enum" gtype="gst_webrtc_peer_connection_state_get_type">
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" name="New" value="0" />
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" name="Connecting" value="1" />
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" name="Connected" value="2" />
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="3" />
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" name="Failed" value="4" />
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" name="Closed" value="5" />
+    </enum>
+    <enum name="WebRTCRTPTransceiverDirection" cname="GstWebRTCRTPTransceiverDirection" type="enum" gtype="gst_webrtc_rtp_transceiver_direction_get_type">
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" name="None" value="0" />
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" name="Inactive" value="1" />
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" name="Sendonly" value="2" />
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" name="Recvonly" value="3" />
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" name="Sendrecv" value="4" />
+    </enum>
+    <enum name="WebRTCSDPType" cname="GstWebRTCSDPType" type="enum" gtype="gst_webrtc_sdp_type_get_type">
+      <member cname="GST_WEBRTC_SDP_TYPE_OFFER" name="Offer" value="1" />
+      <member cname="GST_WEBRTC_SDP_TYPE_PRANSWER" name="Pranswer" value="2" />
+      <member cname="GST_WEBRTC_SDP_TYPE_ANSWER" name="Answer" value="3" />
+      <member cname="GST_WEBRTC_SDP_TYPE_ROLLBACK" name="Rollback" value="4" />
+    </enum>
+    <enum name="WebRTCSignalingState" cname="GstWebRTCSignalingState" type="enum" gtype="gst_webrtc_signaling_state_get_type">
+      <member cname="GST_WEBRTC_SIGNALING_STATE_STABLE" name="Stable" value="0" />
+      <member cname="GST_WEBRTC_SIGNALING_STATE_CLOSED" name="Closed" value="1" />
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" name="HaveLocalOffer" value="2" />
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" name="HaveRemoteOffer" value="3" />
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" name="HaveLocalPranswer" value="4" />
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" name="HaveRemotePranswer" value="5" />
+    </enum>
+    <enum name="WebRTCStatsType" cname="GstWebRTCStatsType" type="enum" gtype="gst_webrtc_stats_type_get_type">
+      <member cname="GST_WEBRTC_STATS_CODEC" name="Codec" value="1" />
+      <member cname="GST_WEBRTC_STATS_INBOUND_RTP" name="InboundRtp" value="2" />
+      <member cname="GST_WEBRTC_STATS_OUTBOUND_RTP" name="OutboundRtp" value="3" />
+      <member cname="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" name="RemoteInboundRtp" value="4" />
+      <member cname="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" name="RemoteOutboundRtp" value="5" />
+      <member cname="GST_WEBRTC_STATS_CSRC" name="Csrc" value="6" />
+      <member cname="GST_WEBRTC_STATS_PEER_CONNECTION" name="PeerConnection" value="7" />
+      <member cname="GST_WEBRTC_STATS_DATA_CHANNEL" name="DataChannel" value="8" />
+      <member cname="GST_WEBRTC_STATS_STREAM" name="Stream" value="9" />
+      <member cname="GST_WEBRTC_STATS_TRANSPORT" name="Transport" value="10" />
+      <member cname="GST_WEBRTC_STATS_CANDIDATE_PAIR" name="CandidatePair" value="11" />
+      <member cname="GST_WEBRTC_STATS_LOCAL_CANDIDATE" name="LocalCandidate" value="12" />
+      <member cname="GST_WEBRTC_STATS_REMOTE_CANDIDATE" name="RemoteCandidate" value="13" />
+      <member cname="GST_WEBRTC_STATS_CERTIFICATE" name="Certificate" value="14" />
+    </enum>
+    <object name="WebRTCDTLSTransport" cname="GstWebRTCDTLSTransport" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCDTLSTransportClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_dtls_transport_get_type" shared="true">
+        <return-type type="GType" />
+      </method>
+      <constructor cname="gst_webrtc_dtls_transport_new">
+        <parameters>
+          <parameter name="session_id" type="guint" />
+          <parameter name="rtcp" type="gboolean" />
+        </parameters>
+      </constructor>
+      <method name="SetTransport" cname="gst_webrtc_dtls_transport_set_transport">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="ice" type="GstWebRTCICETransport*" />
+        </parameters>
+      </method>
+      <property name="Certificate" cname="certificate" type="gchar*" readable="true" writeable="true" construct="false" construct-only="false" />
+      <property name="Client" cname="client" type="gboolean" readable="true" writeable="true" construct="false" construct-only="false" />
+      <property name="RemoteCertificate" cname="remote-certificate" type="gchar*" readable="true" writeable="false" construct="false" construct-only="false" />
+      <property name="Rtcp" cname="rtcp" type="gboolean" readable="true" writeable="true" construct="false" construct-only="true" />
+      <property name="SessionId" cname="session-id" type="guint" readable="true" writeable="true" construct="false" construct-only="true" />
+      <property name="State" cname="state" type="GstWebRTCDTLSTransportState" readable="true" writeable="false" construct="false" construct-only="false" />
+      <property name="Transport" cname="transport" type="GstWebRTCICETransport*" readable="true" writeable="false" construct="false" construct-only="false" />
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+      <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="TransportField" type="GstWebRTCICETransport*" />
+      <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCDTLSTransportState" />
+      <field cname="is_rtcp" access="public" writeable="false" readable="true" is_callback="false" name="IsRtcp" type="gboolean" />
+      <field cname="client" access="public" writeable="false" readable="true" is_callback="false" name="ClientField" type="gboolean" />
+      <field cname="session_id" access="public" writeable="false" readable="true" is_callback="false" name="SessionIdField" type="guint" />
+      <field cname="dtlssrtpenc" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpenc" type="GstElement*" />
+      <field cname="dtlssrtpdec" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpdec" type="GstElement*" />
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+    </object>
+    <object name="WebRTCICETransport" cname="GstWebRTCICETransport" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCICETransportClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <method vm="gather_candidates" />
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_ice_transport_get_type" shared="true">
+        <return-type type="GType" />
+      </method>
+      <virtual_method name="GatherCandidates" cname="gather_candidates">
+        <return-type type="gboolean" />
+        <parameters />
+      </virtual_method>
+      <method name="ConnectionStateChange" cname="gst_webrtc_ice_transport_connection_state_change">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="new_state" type="GstWebRTCICEConnectionState" />
+        </parameters>
+      </method>
+      <method name="GatheringStateChange" cname="gst_webrtc_ice_transport_gathering_state_change">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="new_state" type="GstWebRTCICEGatheringState" />
+        </parameters>
+      </method>
+      <method name="NewCandidate" cname="gst_webrtc_ice_transport_new_candidate">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="stream_id" type="guint" />
+          <parameter name="component" type="GstWebRTCICEComponent" />
+          <parameter name="attr" type="const-gchar*" />
+        </parameters>
+      </method>
+      <method name="SelectedPairChange" cname="gst_webrtc_ice_transport_selected_pair_change">
+        <return-type type="void" />
+        <parameters />
+      </method>
+      <property name="Component" cname="component" type="GstWebRTCICEComponent" readable="true" writeable="true" construct="false" construct-only="true" />
+      <property name="GatheringState" cname="gathering-state" type="GstWebRTCICEGatheringState" readable="true" writeable="false" construct="false" construct-only="false" />
+      <property name="State" cname="state" type="GstWebRTCICEConnectionState" readable="true" writeable="false" construct="false" construct-only="false" />
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+      <field cname="role" access="public" writeable="false" readable="true" is_callback="false" name="Role" type="GstWebRTCICERole" />
+      <field cname="component" access="public" writeable="false" readable="true" is_callback="false" name="ComponentField" type="GstWebRTCICEComponent" />
+      <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCICEConnectionState" />
+      <field cname="gathering_state" access="public" writeable="false" readable="true" is_callback="false" name="GatheringStateField" type="GstWebRTCICEGatheringState" />
+      <field cname="src" access="public" writeable="false" readable="true" is_callback="false" name="Src" type="GstElement*" />
+      <field cname="sink" access="public" writeable="false" readable="true" is_callback="false" name="Sink" type="GstElement*" />
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+      <signal name="OnNewCandidate" cname="on-new-candidate" when="last">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="_object" type="const-gchar*" />
+        </parameters>
+      </signal>
+      <signal name="OnSelectedCandidatePairChange" cname="on-selected-candidate-pair-change" when="last">
+        <return-type type="void" />
+        <parameters />
+      </signal>
+    </object>
+    <object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCRTPReceiverClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_rtp_receiver_get_type" shared="true">
+        <return-type type="GType" />
+      </method>
+      <constructor cname="gst_webrtc_rtp_receiver_new" disable_void_ctor="" />
+      <method name="SetRtcpTransport" cname="gst_webrtc_rtp_receiver_set_rtcp_transport">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+        </parameters>
+      </method>
+      <method name="SetTransport" cname="gst_webrtc_rtp_receiver_set_transport">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+        </parameters>
+      </method>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+      <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*" />
+      <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*" />
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+    </object>
+    <object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCRTPSenderClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_rtp_sender_get_type" shared="true">
+        <return-type type="GType" />
+      </method>
+      <constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor="" />
+      <method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+        </parameters>
+      </method>
+      <method name="SetTransport" cname="gst_webrtc_rtp_sender_set_transport">
+        <return-type type="void" />
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*" />
+        </parameters>
+      </method>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+      <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*" />
+      <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*" />
+      <field cname="send_encodings" access="public" writeable="false" readable="true" is_callback="false" name="SendEncodings" type="GArray*" array="true" null_term_array="true" />
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+    </object>
+    <object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCRTPTransceiverClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_rtp_transceiver_get_type" shared="true">
+        <return-type type="GType" />
+      </method>
+      <property name="Mlineindex" cname="mlineindex" type="guint" readable="true" writeable="true" construct="false" construct-only="true" />
+      <property name="Receiver" cname="receiver" type="GstWebRTCRTPReceiver*" readable="true" writeable="true" construct="false" construct-only="true" />
+      <property name="Sender" cname="sender" type="GstWebRTCRTPSender*" readable="true" writeable="true" construct="false" construct-only="true" />
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
+      <field cname="mline" access="public" writeable="false" readable="true" is_callback="false" name="Mline" type="guint" />
+      <field cname="mid" access="public" writeable="false" readable="true" is_callback="false" name="Mid" type="gchar*" />
+      <field cname="stopped" access="public" writeable="false" readable="true" is_callback="false" name="Stopped" type="gboolean" />
+      <field cname="sender" access="public" writeable="false" readable="true" is_callback="false" name="SenderField" type="GstWebRTCRTPSender*" />
+      <field cname="receiver" access="public" writeable="false" readable="true" is_callback="false" name="ReceiverField" type="GstWebRTCRTPReceiver*" />
+      <field cname="direction" access="public" writeable="false" readable="true" is_callback="false" name="Direction" type="GstWebRTCRTPTransceiverDirection" />
+      <field cname="current_direction" access="public" writeable="false" readable="true" is_callback="false" name="CurrentDirection" type="GstWebRTCRTPTransceiverDirection" />
+      <field cname="codec_preferences" access="public" writeable="false" readable="true" is_callback="false" name="CodecPreferences" type="GstCaps*">
+        <warning>missing glib:type-name</warning>
+      </field>
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
+    </object>
+    <boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
+      <method name="GetType" cname="gst_webrtc_session_description_get_type" shared="true">
+        <return-type type="GType" />
+      </method>
+      <field cname="type" access="public" writeable="true" readable="true" is_callback="false" name="Type" type="GstWebRTCSDPType" />
+      <field cname="sdp" access="public" writeable="true" readable="true" is_callback="false" name="Sdp" type="GstSDPMessage*">
+        <warning>missing glib:type-name</warning>
+      </field>
+      <constructor cname="gst_webrtc_session_description_new">
+        <parameters>
+          <parameter name="type" type="GstWebRTCSDPType" />
+          <parameter name="sdp" type="GstSDPMessage*">
+            <warning>missing glib:type-name</warning>
+          </parameter>
+        </parameters>
+      </constructor>
+      <method name="Copy" cname="gst_webrtc_session_description_copy">
+        <return-type type="GstWebRTCSessionDescription*" owned="true">
+          <warning>missing glib:type-name</warning>
+        </return-type>
+        <parameters />
+      </method>
+      <method name="Free" cname="gst_webrtc_session_description_free">
+        <return-type type="void" />
+        <parameters />
+      </method>
+    </boxed>
+    <object name="Global" cname="GstWebRTCGlobal" opaque="true">
+      <method name="WebrtcSdpTypeToString" cname="gst_webrtc_sdp_type_to_string" shared="true">
+        <return-type type="const-gchar*" />
+        <parameters>
+          <parameter name="type" type="GstWebRTCSDPType" />
+        </parameters>
+      </method>
+    </object>
+    <object name="Constants" cname="GstWebRTCConstants" opaque="true" />
+  </namespace>
 </api>
\ No newline at end of file
diff --git a/sources/generated/meson.build b/sources/generated/meson.build
index 5a82c3c068..942ef08eb5 100644
--- a/sources/generated/meson.build
+++ b/sources/generated/meson.build
@@ -722,6 +722,26 @@ generated_sources = [
     'Gst.Rtsp/Gst.RtspSharp.RTSPConnectionAcceptCertificateFuncNative.cs',
     'Gst.Audio/AudioStreamAlign.cs',
     'Gst.Video/VideoOverlayProperties.cs',
+    'Gst.WebRTC/WebRTCPeerConnectionState.cs',
+    'Gst.WebRTC/WebRTCSessionDescription.cs',
+    'Gst.WebRTC/WebRTCICEGatheringState.cs',
+    'Gst.WebRTC/WebRTCRTPTransceiverDirection.cs',
+    'Gst.WebRTC/WebRTCRTPTransceiver.cs',
+    'Gst.WebRTC/OnNewCandidateHandler.cs',
+    'Gst.WebRTC/WebRTCICERole.cs',
+    'Gst.WebRTC/Global.cs',
+    'Gst.WebRTC/WebRTCICEComponent.cs',
+    'Gst.WebRTC/WebRTCICEConnectionState.cs',
+    'Gst.WebRTC/WebRTCDTLSTransport.cs',
+    'Gst.WebRTC/WebRTCICETransport.cs',
+    'Gst.WebRTC/WebRTCSDPType.cs',
+    'Gst.WebRTC/WebRTCRTPSender.cs',
+    'Gst.WebRTC/WebRTCSignalingState.cs',
+    'Gst.WebRTC/WebRTCDTLSTransportState.cs',
+    'Gst.WebRTC/WebRTCDTLSSetup.cs',
+    'Gst.WebRTC/WebRTCRTPReceiver.cs',
+    'Gst.WebRTC/WebRTCStatsType.cs',
+    'Gst.WebRTC/Constants.cs',
 ]
 
 run_target('update_gstreamer_code',
diff --git a/sources/gstreamer-sharp-api.raw b/sources/gstreamer-sharp-api.raw
index 23c852e1fb..d132b09d6e 100644
--- a/sources/gstreamer-sharp-api.raw
+++ b/sources/gstreamer-sharp-api.raw
@@ -29184,4 +29184,305 @@
       <constant value="16" ctype="gint" gtype="gint" name="VIDEO_TILE_Y_TILES_SHIFT"/>
     </object>
   </namespace>
+  <namespace name="GstWebRTC" library="gstwebrtc-1.0">
+    <enum name="WebRTCDTLSSetup" cname="GstWebRTCDTLSSetup" type="enum" gtype="gst_webrtc_dtls_setup_get_type">
+      <member cname="GST_WEBRTC_DTLS_SETUP_NONE" name="None" value="0"/>
+      <member cname="GST_WEBRTC_DTLS_SETUP_ACTPASS" name="Actpass" value="1"/>
+      <member cname="GST_WEBRTC_DTLS_SETUP_ACTIVE" name="Active" value="2"/>
+      <member cname="GST_WEBRTC_DTLS_SETUP_PASSIVE" name="Passive" value="3"/>
+    </enum>
+    <enum name="WebRTCDTLSTransportState" cname="GstWebRTCDTLSTransportState" type="enum" gtype="gst_webrtc_dtls_transport_state_get_type">
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" name="New" value="0"/>
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" name="Closed" value="1"/>
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" name="Failed" value="2"/>
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" name="Connecting" value="3"/>
+      <member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" name="Connected" value="4"/>
+    </enum>
+    <enum name="WebRTCICEComponent" cname="GstWebRTCICEComponent" type="enum" gtype="gst_webrtc_ice_component_get_type">
+      <member cname="GST_WEBRTC_ICE_COMPONENT_RTP" name="Rtp" value="0"/>
+      <member cname="GST_WEBRTC_ICE_COMPONENT_RTCP" name="Rtcp" value="1"/>
+    </enum>
+    <enum name="WebRTCICEConnectionState" cname="GstWebRTCICEConnectionState" type="enum" gtype="gst_webrtc_ice_connection_state_get_type">
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" name="New" value="0"/>
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" name="Checking" value="1"/>
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" name="Connected" value="2"/>
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" name="Completed" value="3"/>
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" name="Failed" value="4"/>
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="5"/>
+      <member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" name="Closed" value="6"/>
+    </enum>
+    <enum name="WebRTCICEGatheringState" cname="GstWebRTCICEGatheringState" type="enum" gtype="gst_webrtc_ice_gathering_state_get_type">
+      <member cname="GST_WEBRTC_ICE_GATHERING_STATE_NEW" name="New" value="0"/>
+      <member cname="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" name="Gathering" value="1"/>
+      <member cname="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" name="Complete" value="2"/>
+    </enum>
+    <enum name="WebRTCICERole" cname="GstWebRTCICERole" type="enum" gtype="gst_webrtc_ice_role_get_type">
+      <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLED" name="Controlled" value="0"/>
+      <member cname="GST_WEBRTC_ICE_ROLE_CONTROLLING" name="Controlling" value="1"/>
+    </enum>
+    <enum name="WebRTCPeerConnectionState" cname="GstWebRTCPeerConnectionState" type="enum" gtype="gst_webrtc_peer_connection_state_get_type">
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" name="New" value="0"/>
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" name="Connecting" value="1"/>
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" name="Connected" value="2"/>
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="3"/>
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" name="Failed" value="4"/>
+      <member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" name="Closed" value="5"/>
+    </enum>
+    <enum name="WebRTCRTPTransceiverDirection" cname="GstWebRTCRTPTransceiverDirection" type="enum" gtype="gst_webrtc_rtp_transceiver_direction_get_type">
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" name="None" value="0"/>
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" name="Inactive" value="1"/>
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" name="Sendonly" value="2"/>
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" name="Recvonly" value="3"/>
+      <member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" name="Sendrecv" value="4"/>
+    </enum>
+    <enum name="WebRTCSDPType" cname="GstWebRTCSDPType" type="enum" gtype="gst_webrtc_sdp_type_get_type">
+      <member cname="GST_WEBRTC_SDP_TYPE_OFFER" name="Offer" value="1"/>
+      <member cname="GST_WEBRTC_SDP_TYPE_PRANSWER" name="Pranswer" value="2"/>
+      <member cname="GST_WEBRTC_SDP_TYPE_ANSWER" name="Answer" value="3"/>
+      <member cname="GST_WEBRTC_SDP_TYPE_ROLLBACK" name="Rollback" value="4"/>
+    </enum>
+    <enum name="WebRTCSignalingState" cname="GstWebRTCSignalingState" type="enum" gtype="gst_webrtc_signaling_state_get_type">
+      <member cname="GST_WEBRTC_SIGNALING_STATE_STABLE" name="Stable" value="0"/>
+      <member cname="GST_WEBRTC_SIGNALING_STATE_CLOSED" name="Closed" value="1"/>
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" name="HaveLocalOffer" value="2"/>
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" name="HaveRemoteOffer" value="3"/>
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" name="HaveLocalPranswer" value="4"/>
+      <member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" name="HaveRemotePranswer" value="5"/>
+    </enum>
+    <enum name="WebRTCStatsType" cname="GstWebRTCStatsType" type="enum" gtype="gst_webrtc_stats_type_get_type">
+      <member cname="GST_WEBRTC_STATS_CODEC" name="Codec" value="1"/>
+      <member cname="GST_WEBRTC_STATS_INBOUND_RTP" name="InboundRtp" value="2"/>
+      <member cname="GST_WEBRTC_STATS_OUTBOUND_RTP" name="OutboundRtp" value="3"/>
+      <member cname="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" name="RemoteInboundRtp" value="4"/>
+      <member cname="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" name="RemoteOutboundRtp" value="5"/>
+      <member cname="GST_WEBRTC_STATS_CSRC" name="Csrc" value="6"/>
+      <member cname="GST_WEBRTC_STATS_PEER_CONNECTION" name="PeerConnection" value="7"/>
+      <member cname="GST_WEBRTC_STATS_DATA_CHANNEL" name="DataChannel" value="8"/>
+      <member cname="GST_WEBRTC_STATS_STREAM" name="Stream" value="9"/>
+      <member cname="GST_WEBRTC_STATS_TRANSPORT" name="Transport" value="10"/>
+      <member cname="GST_WEBRTC_STATS_CANDIDATE_PAIR" name="CandidatePair" value="11"/>
+      <member cname="GST_WEBRTC_STATS_LOCAL_CANDIDATE" name="LocalCandidate" value="12"/>
+      <member cname="GST_WEBRTC_STATS_REMOTE_CANDIDATE" name="RemoteCandidate" value="13"/>
+      <member cname="GST_WEBRTC_STATS_CERTIFICATE" name="Certificate" value="14"/>
+    </enum>
+    <object name="WebRTCDTLSTransport" cname="GstWebRTCDTLSTransport" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCDTLSTransportClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_dtls_transport_get_type" shared="true">
+        <return-type type="GType"/>
+      </method>
+      <constructor cname="gst_webrtc_dtls_transport_new">
+        <parameters>
+          <parameter name="session_id" type="guint"/>
+          <parameter name="rtcp" type="gboolean"/>
+        </parameters>
+      </constructor>
+      <method name="SetTransport" cname="gst_webrtc_dtls_transport_set_transport">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="ice" type="GstWebRTCICETransport*"/>
+        </parameters>
+      </method>
+      <property name="Certificate" cname="certificate" type="gchar*" readable="true" writeable="true" construct="false" construct-only="false"/>
+      <property name="Client" cname="client" type="gboolean" readable="true" writeable="true" construct="false" construct-only="false"/>
+      <property name="RemoteCertificate" cname="remote-certificate" type="gchar*" readable="true" writeable="false" construct="false" construct-only="false"/>
+      <property name="Rtcp" cname="rtcp" type="gboolean" readable="true" writeable="true" construct="false" construct-only="true"/>
+      <property name="SessionId" cname="session-id" type="guint" readable="true" writeable="true" construct="false" construct-only="true"/>
+      <property name="State" cname="state" type="GstWebRTCDTLSTransportState" readable="true" writeable="false" construct="false" construct-only="false"/>
+      <property name="Transport" cname="transport" type="GstWebRTCICETransport*" readable="true" writeable="false" construct="false" construct-only="false"/>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+      <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="TransportField" type="GstWebRTCICETransport*"/>
+      <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCDTLSTransportState"/>
+      <field cname="is_rtcp" access="public" writeable="false" readable="true" is_callback="false" name="IsRtcp" type="gboolean"/>
+      <field cname="client" access="public" writeable="false" readable="true" is_callback="false" name="ClientField" type="gboolean"/>
+      <field cname="session_id" access="public" writeable="false" readable="true" is_callback="false" name="SessionIdField" type="guint"/>
+      <field cname="dtlssrtpenc" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpenc" type="GstElement*"/>
+      <field cname="dtlssrtpdec" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpdec" type="GstElement*"/>
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+    </object>
+    <object name="WebRTCICETransport" cname="GstWebRTCICETransport" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCICETransportClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <method vm="gather_candidates"/>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_ice_transport_get_type" shared="true">
+        <return-type type="GType"/>
+      </method>
+      <virtual_method name="GatherCandidates" cname="gather_candidates">
+        <return-type type="gboolean"/>
+        <parameters/>
+      </virtual_method>
+      <method name="ConnectionStateChange" cname="gst_webrtc_ice_transport_connection_state_change">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="new_state" type="GstWebRTCICEConnectionState"/>
+        </parameters>
+      </method>
+      <method name="GatheringStateChange" cname="gst_webrtc_ice_transport_gathering_state_change">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="new_state" type="GstWebRTCICEGatheringState"/>
+        </parameters>
+      </method>
+      <method name="NewCandidate" cname="gst_webrtc_ice_transport_new_candidate">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="stream_id" type="guint"/>
+          <parameter name="component" type="GstWebRTCICEComponent"/>
+          <parameter name="attr" type="const-gchar*"/>
+        </parameters>
+      </method>
+      <method name="SelectedPairChange" cname="gst_webrtc_ice_transport_selected_pair_change">
+        <return-type type="void"/>
+        <parameters/>
+      </method>
+      <property name="Component" cname="component" type="GstWebRTCICEComponent" readable="true" writeable="true" construct="false" construct-only="true"/>
+      <property name="GatheringState" cname="gathering-state" type="GstWebRTCICEGatheringState" readable="true" writeable="false" construct="false" construct-only="false"/>
+      <property name="State" cname="state" type="GstWebRTCICEConnectionState" readable="true" writeable="false" construct="false" construct-only="false"/>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+      <field cname="role" access="public" writeable="false" readable="true" is_callback="false" name="Role" type="GstWebRTCICERole"/>
+      <field cname="component" access="public" writeable="false" readable="true" is_callback="false" name="ComponentField" type="GstWebRTCICEComponent"/>
+      <field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCICEConnectionState"/>
+      <field cname="gathering_state" access="public" writeable="false" readable="true" is_callback="false" name="GatheringStateField" type="GstWebRTCICEGatheringState"/>
+      <field cname="src" access="public" writeable="false" readable="true" is_callback="false" name="Src" type="GstElement*"/>
+      <field cname="sink" access="public" writeable="false" readable="true" is_callback="false" name="Sink" type="GstElement*"/>
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+      <signal name="OnNewCandidate" cname="on-new-candidate" when="last">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="_object" type="const-gchar*"/>
+        </parameters>
+      </signal>
+      <signal name="OnSelectedCandidatePairChange" cname="on-selected-candidate-pair-change" when="last">
+        <return-type type="void"/>
+        <parameters/>
+      </signal>
+    </object>
+    <object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCRTPReceiverClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_rtp_receiver_get_type" shared="true">
+        <return-type type="GType"/>
+      </method>
+      <constructor cname="gst_webrtc_rtp_receiver_new" disable_void_ctor=""/>
+      <method name="SetRtcpTransport" cname="gst_webrtc_rtp_receiver_set_rtcp_transport">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+        </parameters>
+      </method>
+      <method name="SetTransport" cname="gst_webrtc_rtp_receiver_set_transport">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+        </parameters>
+      </method>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+      <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*"/>
+      <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*"/>
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+    </object>
+    <object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCRTPSenderClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_rtp_sender_get_type" shared="true">
+        <return-type type="GType"/>
+      </method>
+      <constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor=""/>
+      <method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+        </parameters>
+      </method>
+      <method name="SetTransport" cname="gst_webrtc_rtp_sender_set_transport">
+        <return-type type="void"/>
+        <parameters>
+          <parameter name="transport" type="GstWebRTCDTLSTransport*"/>
+        </parameters>
+      </method>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+      <field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*"/>
+      <field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*"/>
+      <field cname="send_encodings" access="public" writeable="false" readable="true" is_callback="false" name="SendEncodings" type="GArray*" array="true" null_term_array="true"/>
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+    </object>
+    <object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
+      <class_struct cname="GstWebRTCRTPTransceiverClass">
+        <field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
+          <warning>missing glib:type-name</warning>
+        </field>
+        <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+      </class_struct>
+      <method name="GetType" cname="gst_webrtc_rtp_transceiver_get_type" shared="true">
+        <return-type type="GType"/>
+      </method>
+      <property name="Mlineindex" cname="mlineindex" type="guint" readable="true" writeable="true" construct="false" construct-only="true"/>
+      <property name="Receiver" cname="receiver" type="GstWebRTCRTPReceiver*" readable="true" writeable="true" construct="false" construct-only="true"/>
+      <property name="Sender" cname="sender" type="GstWebRTCRTPSender*" readable="true" writeable="true" construct="false" construct-only="true"/>
+      <field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
+      <field cname="mline" access="public" writeable="false" readable="true" is_callback="false" name="Mline" type="guint"/>
+      <field cname="mid" access="public" writeable="false" readable="true" is_callback="false" name="Mid" type="gchar*"/>
+      <field cname="stopped" access="public" writeable="false" readable="true" is_callback="false" name="Stopped" type="gboolean"/>
+      <field cname="sender" access="public" writeable="false" readable="true" is_callback="false" name="SenderField" type="GstWebRTCRTPSender*"/>
+      <field cname="receiver" access="public" writeable="false" readable="true" is_callback="false" name="ReceiverField" type="GstWebRTCRTPReceiver*"/>
+      <field cname="direction" access="public" writeable="false" readable="true" is_callback="false" name="Direction" type="GstWebRTCRTPTransceiverDirection"/>
+      <field cname="current_direction" access="public" writeable="false" readable="true" is_callback="false" name="CurrentDirection" type="GstWebRTCRTPTransceiverDirection"/>
+      <field cname="codec_preferences" access="public" writeable="false" readable="true" is_callback="false" name="CodecPreferences" type="GstCaps*">
+        <warning>missing glib:type-name</warning>
+      </field>
+      <field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
+    </object>
+    <boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
+      <method name="GetType" cname="gst_webrtc_session_description_get_type" shared="true">
+        <return-type type="GType"/>
+      </method>
+      <field cname="type" access="public" writeable="true" readable="true" is_callback="false" name="Type" type="GstWebRTCSDPType"/>
+      <field cname="sdp" access="public" writeable="true" readable="true" is_callback="false" name="Sdp" type="GstSDPMessage*">
+        <warning>missing glib:type-name</warning>
+      </field>
+      <constructor cname="gst_webrtc_session_description_new">
+        <parameters>
+          <parameter name="type" type="GstWebRTCSDPType"/>
+          <parameter name="sdp" type="GstSDPMessage*">
+            <warning>missing glib:type-name</warning>
+          </parameter>
+        </parameters>
+      </constructor>
+      <method name="Copy" cname="gst_webrtc_session_description_copy">
+        <return-type type="GstWebRTCSessionDescription*" owned="true">
+          <warning>missing glib:type-name</warning>
+        </return-type>
+        <parameters/>
+      </method>
+      <method name="Free" cname="gst_webrtc_session_description_free">
+        <return-type type="void"/>
+        <parameters/>
+      </method>
+    </boxed>
+    <object name="Global" cname="GstWebRTCGlobal" opaque="true">
+      <method name="WebrtcSdpTypeToString" cname="gst_webrtc_sdp_type_to_string" shared="true">
+        <return-type type="const-gchar*"/>
+        <parameters>
+          <parameter name="type" type="GstWebRTCSDPType"/>
+        </parameters>
+      </method>
+    </object>
+    <object name="Constants" cname="GstWebRTCConstants" opaque="true"/>
+  </namespace>
 </api>
diff --git a/sources/gstreamer-sharp.dll.config b/sources/gstreamer-sharp.dll.config
index cb98e23dc8..63059b373f 100644
--- a/sources/gstreamer-sharp.dll.config
+++ b/sources/gstreamer-sharp.dll.config
@@ -12,6 +12,7 @@
   <dllmap dll="libgstrtp-1.0-0.dll" target="libgstrtp-1.0.so.0" os="linux"/>
   <dllmap dll="libgstrtsp-1.0-0.dll" target="libgstrtsp-1.0.so.0" os="linux"/>
   <dllmap dll="libgstsdp-1.0-0.dll" target="libgstsdp-1.0.so.0" os="linux"/>
+  <dllmap dll="libgstwebrtc-1.0-0.dll" target="libgstwebrtc-1.0.so.0" os="linux"/>
   <dllmap dll="libgstcontroller-1.0-0.dll" target="libgstcontroller-1.0.so.0" os="linux"/>
   <dllmap dll="libglib-2.0-0.dll" target="libglib-2.0.so.0" os="linux"/>
   <dllmap dll="libgobject-2.0-0.dll" target="libgobject-2.0.so.0" os="linux"/>
@@ -29,6 +30,7 @@
   <dllmap dll="libgstrtp-1.0-0.dll" target="libgstrtp-1.0.dylib" os="osx"/>
   <dllmap dll="libgstrtsp-1.0-0.dll" target="libgstrtsp-1.0.dylib" os="osx"/>
   <dllmap dll="libgstsdp-1.0-0.dll" target="libgstsdp-1.0.dylib" os="osx"/>
+  <dllmap dll="libgstwebrtc-1.0-0.dll" target="libgstwebrtc-1.0.dylib" os="osx"/>
   <dllmap dll="libgstcontroller-1.0-0.dll" target="libgstcontroller-1.0.dylib" os="osx"/>
   <dllmap dll="libglib-2.0-0.dll" target="libglib-2.0.dylib" os="osx"/>
   <dllmap dll="libgobject-2.0-0.dll" target="libgobject-2.0.dylib" os="osx"/>
diff --git a/sources/gstreamer-sharp.metadata b/sources/gstreamer-sharp.metadata
index d710b4aebb..fb726b2880 100644
--- a/sources/gstreamer-sharp.metadata
+++ b/sources/gstreamer-sharp.metadata
@@ -243,6 +243,7 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
 	<attr path="/api/namespace[@name='GstRtp']" name="name">Gst.Rtp</attr>
 	<attr path="/api/namespace[@name='GstRtsp']" name="name">Gst.Rtsp</attr>
 	<attr path="/api/namespace[@name='GstSdp']" name="name">Gst.Sdp</attr>
+	<attr path="/api/namespace[@name='GstWebRTC']" name="name">Gst.WebRTC</attr>
 
 	<attr path="/api/namespace" name="library">libgstreamer-1.0-0.dll</attr>
 	<attr path="/api/namespace[@name='Gst.Base']" name="library">libgstbase-1.0-0.dll</attr>
@@ -258,6 +259,7 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
 	<attr path="/api/namespace[@name='Gst.Rtp']" name="library">libgstrtp-1.0-0.dll</attr>
 	<attr path="/api/namespace[@name='Gst.Rtsp']" name="library">libgstrtsp-1.0-0.dll</attr>
 	<attr path="/api/namespace[@name='Gst.Sdp']" name="library">libgstsdp-1.0-0.dll</attr>
+	<attr path="/api/namespace[@name='Gst.WebRTC']" name="library">libgstwebrtc-1.0-0.dll</attr>
 
 	<!-- DoubleRange and Fraction are in Value.cs -->
 	<attr path="//struct[@name='DoubleRange' or @name='Fraction' or @name='IntRange' or @name='FractionRange']" name="hidden">true</attr>
diff --git a/sources/meson.build b/sources/meson.build
index 2eea17268a..6b44a7d7ea 100644
--- a/sources/meson.build
+++ b/sources/meson.build
@@ -1,7 +1,7 @@
 raw_api_fname = join_paths(meson.current_source_dir(), meson.project_name() + '-api.raw')
 metadata = files(meson.project_name() + '.metadata')
 
-abi_includes = 'glib.h,gst/gst.h,gst/video/video.h,gst/audio/audio.h,gst/rtsp/rtsp.h,gst/app/app.h,gst/audio/audio.h,gst/base/base.h,gst/controller/controller.h,gst/fft/fft.h,gst/net/net.h,gst/pbutils/gstaudiovisualizer.h,gst/pbutils/pbutils.h,gst/rtp/rtp.h,gst/rtsp/rtsp.h,gst/sdp/sdp.h,gst/tag/tag.h,gst/video/video.h,gst/video/gstvideoaffinetransformationmeta.h,gst/net/gstnetcontrolmessagemeta.h'
+abi_includes = 'glib.h,gst/gst.h,gst/video/video.h,gst/audio/audio.h,gst/rtsp/rtsp.h,gst/app/app.h,gst/audio/audio.h,gst/base/base.h,gst/controller/controller.h,gst/fft/fft.h,gst/net/net.h,gst/pbutils/gstaudiovisualizer.h,gst/pbutils/pbutils.h,gst/rtp/rtp.h,gst/rtsp/rtsp.h,gst/sdp/sdp.h,gst/tag/tag.h,gst/video/video.h,gst/video/gstvideoaffinetransformationmeta.h,gst/net/gstnetcontrolmessagemeta.h,gst/webrtc/webrtc.h'
 
 sources = [
     'custom/Adapter.cs',
@@ -43,7 +43,7 @@ gst_sharp_dep = declare_dependency(dependencies: [glib_sharp_dep, gio_sharp_dep]
 
 if add_languages('c', required: false) and csc.get_id() == 'mono'
     c_abi_exe = executable('gst_sharp_c_abi', c_abi,
-            cs_args: ['-nowarn:169', '-nowarn:108', '-nowarn:114', '-unsafe'],
+            c_args: ['-DGST_USE_UNSTABLE_API'],
             dependencies: [gst_deps])
 
     cs_abi_exe = executable('gst_sharp_cs_abi', cs_abi,