5 files changed, 147 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/meson.build b/gst-libs/gst/webrtc/meson.build
index efdba6556..c363828b5 100644
--- a/gst-libs/gst/webrtc/meson.build
+++ b/gst-libs/gst/webrtc/meson.build
@@ -6,6 +6,7 @@ webrtc_sources = [
   'rtpsender.c',
   'rtptransceiver.c',
   'datachannel.c',
+  'sctptransport.c',
 ]
 
 webrtc_headers = [
@@ -18,6 +19,7 @@ webrtc_headers = [
   'datachannel.h',
   'webrtc_fwd.h',
   'webrtc.h',
+  'sctptransport.h',
 ]
 
 webrtc_enumtypes_headers = [
diff --git a/gst-libs/gst/webrtc/sctptransport.c b/gst-libs/gst/webrtc/sctptransport.c
new file mode 100644
index 000000000..4d0495a46
--- /dev/null
+++ b/gst-libs/gst/webrtc/sctptransport.c
@@ -0,0 +1,79 @@
+/* GStreamer
+ * Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "sctptransport.h"
+#include "webrtc-priv.h"
+
+G_DEFINE_ABSTRACT_TYPE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
+    GST_TYPE_OBJECT);
+
+static void
+gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  /* all properties should by handled by the plugin class */
+  g_assert_not_reached ();
+}
+
+static void
+gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  guint property_id_dummy = 0;
+
+  gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_object ("transport",
+          "WebRTC DTLS Transport",
+          "DTLS transport used for this SCTP transport",
+          GST_TYPE_WEBRTC_DTLS_TRANSPORT,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_enum ("state",
+          "WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
+          GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
+          GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_uint64 ("max-message-size",
+          "Maximum message size",
+          "Maximum message size as reported by the transport", 0, G_MAXUINT64,
+          0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      ++property_id_dummy,
+      g_param_spec_uint ("max-channels",
+          "Maximum number of channels", "Maximum number of channels",
+          0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
+{
+}
diff --git a/gst-libs/gst/webrtc/sctptransport.h b/gst-libs/gst/webrtc/sctptransport.h
new file mode 100644
index 000000000..99a46eede
--- /dev/null
+++ b/gst-libs/gst/webrtc/sctptransport.h
@@ -0,0 +1,42 @@
+/* GStreamer
+ * Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_WEBRTC_SCTP_TRANSPORT_H__
+#define __GST_WEBRTC_SCTP_TRANSPORT_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc_fwd.h>
+
+G_BEGIN_DECLS
+
+GST_WEBRTC_API
+GType gst_webrtc_sctp_transport_get_type(void);
+
+#define GST_TYPE_WEBRTC_SCTP_TRANSPORT            (gst_webrtc_sctp_transport_get_type())
+#define GST_WEBRTC_SCTP_TRANSPORT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
+#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
+#define GST_WEBRTC_SCTP_TRANSPORT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
+#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
+#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSCTPTransport, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
diff --git a/gst-libs/gst/webrtc/webrtc-priv.h b/gst-libs/gst/webrtc/webrtc-priv.h
index cf8a081c2..7366006b7 100644
--- a/gst-libs/gst/webrtc/webrtc-priv.h
+++ b/gst-libs/gst/webrtc/webrtc-priv.h
@@ -289,6 +289,27 @@ GST_WEBRTC_API
 void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
 
 
+/**
+ * GstWebRTCSCTPTransport:
+ *
+ * Since: 1.20
+ */
+struct _GstWebRTCSCTPTransport
+{
+  GstObject                     parent;
+};
+
+/**
+ * GstWebRTCSCTPTransportClass:
+ *
+ * Since: 1.20
+ */
+struct _GstWebRTCSCTPTransportClass
+{
+  GstObjectClass                parent_class;
+};
+
+
 G_END_DECLS
 
 #endif /* __GST_WEBRTC_PRIV_H__ */
diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h
index f3f9aebb8..189e7b36c 100644
--- a/gst-libs/gst/webrtc/webrtc_fwd.h
+++ b/gst-libs/gst/webrtc/webrtc_fwd.h
@@ -92,6 +92,9 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
 typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
 typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
 
+typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport;
+typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass;
+
 /**
  * GstWebRTCDTLSTransportState:
  * @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new