webrtcbin: an element that handles the transport aspects of webrtc connections

SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523

1 files changed, 70 insertions, 0 deletions

diff --git a/gst-libs/gst/webrtc/dtlstransport.h b/gst-libs/gst/webrtc/dtlstransport.h
new file mode 100644
index 000000000..366a602a2
--- /dev/null
+++ b/gst-libs/gst/webrtc/dtlstransport.h
@@ -0,0 +1,70 @@
+/* GStreamer
+ * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_WEBRTC_DTLS_TRANSPORT_H__
+#define __GST_WEBRTC_DTLS_TRANSPORT_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc_fwd.h>
+#include <gst/webrtc/icetransport.h>
+
+G_BEGIN_DECLS
+
+GST_EXPORT
+GType gst_webrtc_dtls_transport_get_type(void);
+#define GST_TYPE_WEBRTC_DTLS_TRANSPORT            (gst_webrtc_dtls_transport_get_type())
+#define GST_WEBRTC_DTLS_TRANSPORT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DTLS_TRANSPORT,GstWebRTCDTLSTransport))
+#define GST_IS_WEBRTC_DTLS_TRANSPORT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DTLS_TRANSPORT))
+#define GST_WEBRTC_DTLS_TRANSPORT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT,GstWebRTCDTLSTransportClass))
+#define GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT))
+#define GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT,GstWebRTCDTLSTransportClass))
+
+struct _GstWebRTCDTLSTransport
+{
+  GstObject                          parent;
+
+  GstWebRTCICETransport             *transport;
+  GstWebRTCDTLSTransportState        state;
+
+  gboolean                           is_rtcp;
+  gboolean                           client;
+  guint                              session_id;
+  GstElement                        *dtlssrtpenc;
+  GstElement                        *dtlssrtpdec;
+
+  gpointer                          _padding[GST_PADDING];
+};
+
+struct _GstWebRTCDTLSTransportClass
+{
+  GstBinClass               parent_class;
+
+  gpointer                  _padding[GST_PADDING];
+};
+
+GST_EXPORT
+GstWebRTCDTLSTransport *    gst_webrtc_dtls_transport_new               (guint session_id, gboolean rtcp);
+
+GST_EXPORT
+void                        gst_webrtc_dtls_transport_set_transport     (GstWebRTCDTLSTransport * transport,
+                                                                         GstWebRTCICETransport * ice);
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_DTLS_TRANSPORT_H__ */