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

.gitignore

@@ -46,6 +46,8 @@ Makefile
 tmp-orc.c
 gst*orc.h
 
+/gst-libs/gst/*/*-enumtypes.[ch]
+
 /tests/check/orc
 /tests/check/media/
 /tests/check/libs/player
@@ -71,6 +73,9 @@ gst*orc.h
 /tests/examples/opencv/gstfacedetect_test
 /tests/examples/playout
 /tests/examples/waylandsink/gtkwaylandsink
+/tests/examples/webrtc/webrtc
+/tests/examples/webrtc/webrtcbidirectional
+/tests/examples/webrtc/webrtcswap
 
 Build
 *.user

configure.ac

@@ -2425,6 +2425,16 @@ AG_GST_CHECK_FEATURE(WEBRTCDSP, [WebRTC Audio Processing], webrtcdsp, [
   AC_LANG_POP([C++])
 ])
 
+dnl *** WebRTC ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_WEBRTC, true)
+AG_GST_CHECK_FEATURE(WEBRTC, [WebRTC], webrtc, [
+  AG_GST_PKG_CHECK_MODULES(GST_SDP, gstreamer-sdp-1.0)
+  PKG_CHECK_MODULES(NICE, nice >= 0.1, [
+    HAVE_WEBRTC="yes" ], [
+    HAVE_WEBRTC="no"
+  ])
+])
+
 else
 
 dnl not building plugins with external dependencies,
@@ -2501,6 +2511,7 @@ AM_CONDITIONAL(USE_RTMP, false)
 AM_CONDITIONAL(USE_TELETEXTDEC, false)
 AM_CONDITIONAL(USE_UVCH264, false)
 AM_CONDITIONAL(USE_WEBP, false)
+AM_CONDITIONAL(USE_WEBRTC, false)
 AM_CONDITIONAL(USE_WEBRTCDSP, false)
 AM_CONDITIONAL(USE_OPENH264, false)
 AM_CONDITIONAL(USE_X265, false)
@@ -2676,6 +2687,7 @@ gst-libs/gst/codecparsers/Makefile
 gst-libs/gst/mpegts/Makefile
 gst-libs/gst/uridownloader/Makefile
 gst-libs/gst/wayland/Makefile
+gst-libs/gst/webrtc/Makefile
 gst-libs/gst/player/Makefile
 gst-libs/gst/video/Makefile
 gst-libs/gst/audio/Makefile
@@ -2724,6 +2736,7 @@ tests/examples/mxf/Makefile
 tests/examples/opencv/Makefile
 tests/examples/uvch264/Makefile
 tests/examples/waylandsink/Makefile
+tests/examples/webrtc/Makefile
 tests/icles/Makefile
 ext/voamrwbenc/Makefile
 ext/voaacenc/Makefile
@@ -2793,6 +2806,7 @@ ext/webp/Makefile
 ext/x265/Makefile
 ext/zbar/Makefile
 ext/dtls/Makefile
+ext/webrtc/Makefile
 ext/webrtcdsp/Makefile
 ext/ttml/Makefile
 po/Makefile.in
@@ -2813,6 +2827,8 @@ pkgconfig/gstreamer-player.pc
 pkgconfig/gstreamer-player-uninstalled.pc
 pkgconfig/gstreamer-wayland.pc
 pkgconfig/gstreamer-wayland-uninstalled.pc
+pkgconfig/gstreamer-webrtc.pc
+pkgconfig/gstreamer-webrtc-uninstalled.pc
 pkgconfig/gstreamer-bad-video.pc
 pkgconfig/gstreamer-bad-video-uninstalled.pc
 pkgconfig/gstreamer-bad-audio.pc

docs/libs/Makefile.am

@@ -61,6 +61,7 @@ GTKDOC_LIBS = \
 	$(top_builddir)/gst-libs/gst/insertbin/libgstinsertbin-@GST_API_VERSION@.la \
 	$(top_builddir)/gst-libs/gst/mpegts/libgstmpegts-@GST_API_VERSION@.la \
 	$(top_builddir)/gst-libs/gst/player/libgstplayer-@GST_API_VERSION@.la \
+	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la \
   $(GST_BASE_LIBS)
 
 # If you need to override some of the declarations, place them in this file

docs/libs/gst-plugins-bad-libs-docs.sgml

@@ -73,6 +73,16 @@
       <xi:include href="xml/gstplayer-visualization.xml"/>
     </chapter>
 
+    <chapter id="webrtc">
+      <title>WebRTC Library</title>
+      <xi:include href="xml/gstwebrtc-dtlstransport.xml"/>
+      <xi:include href="xml/gstwebrtc-icetransport.xml"/>
+      <xi:include href="xml/gstwebrtc-receiver.xml"/>
+      <xi:include href="xml/gstwebrtc-sender.xml"/>
+      <xi:include href="xml/gstwebrtc-sessiondescription.xml"/>
+      <xi:include href="xml/gstwebrtc-transceiver.xml"/>
+    </chapter>
+
     <chapter>
       <title>Interfaces</title>
       <xi:include href="xml/gstphotography.xml" />

docs/libs/gst-plugins-bad-libs-sections.txt

@@ -1065,3 +1065,104 @@ GstPlayerSubtitleInfoClass
 gst_player_subtitle_info_get_type
 </SECTION>
 
+
+<SECTION>
+<FILE>gstwebrtc-dtlstransport</FILE>
+GstWebRTCDTLSTransportState
+
+gst_webrtc_dtls_transport_new
+
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_DTLS_TRANSPORT
+gst_webrtc_dtls_transport_get_type
+GstWebRTCDTLSTransport
+GST_WEBRTC_DTLS_TRANSPORT
+GST_IS_WEBRTC_DTLS_TRANSPORT
+GstWebRTCDTLSTransportClass
+GST_WEBRTC_DTLS_TRANSPORT_CLASS
+GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS
+GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-icetransport</FILE>
+GstWebRTCIceRole
+GstWebRTCICEConnectionState
+GstWebRTCICEGatheringState
+
+
+
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_ICE_TRANSPORT
+gst_webrtc_ice_transport_get_type
+GstWebRTCICETransport
+GST_WEBRTC_ICE_TRANSPORT
+GST_IS_WEBRTC_ICE_TRANSPORT
+GstWebRTCICETransportClass
+GST_WEBRTC_ICE_TRANSPORT_CLASS
+GST_WEBRTC_ICE_TRANSPORT_GET_CLASS
+GST_IS_WEBRTC_ICE_TRANSPORT_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-receiver</FILE>
+gst_webrtc_rtp_receiver_new
+gst_webrtc_rtp_receiver_get_parameters
+gst_webrtc_rtp_receiver_set_parameters
+gst_webrtc_rtp_receiver_set_rtcp_transport
+gst_webrtc_rtp_receiver_set_transport
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_RTP_RECEIVER
+gst_webrtc_rtp_receiver_get_type
+GstWebRTCRTPReceiver
+GST_WEBRTC_RTP_RECEIVER
+GST_IS_WEBRTC_RTP_RECEIVER
+GstWebRTCRTPReceiverClass
+GST_WEBRTC_RTP_RECEIVER_CLASS
+GST_WEBRTC_RTP_RECEIVER_GET_CLASS
+GST_IS_WEBRTC_RTP_RECEIVER_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-sender</FILE>
+gst_webrtc_rtp_sender_new
+gst_webrtc_rtp_sender_get_parameters
+gst_webrtc_rtp_sender_set_parameters
+gst_webrtc_rtp_sender_set_rtcp_transport
+gst_webrtc_rtp_sender_set_transport
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_RTP_SENDER
+gst_webrtc_rtp_sender_get_type
+GstWebRTCRTPSender
+GST_WEBRTC_RTP_SENDER
+GST_IS_WEBRTC_RTP_SENDER
+GstWebRTCRTPSenderClass
+GST_WEBRTC_RTP_SENDER_CLASS
+GST_WEBRTC_RTP_SENDER_GET_CLASS
+GST_IS_WEBRTC_RTP_SENDER_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-sessiondescription</FILE>
+GstWebRTCSessionDescription
+gst_webrtc_session_description_new
+gst_webrtc_session_description_copy
+gst_webrtc_session_description_free
+<SUBSECTION Standard>
+gst_webrtc_session_description_get_type
+GST_TYPE_WEBRTC_SESSION_DESCRIPTION
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-transceiver</FILE>
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_RTP_TRANSCEIVER
+gst_webrtc_rtp_transceiver_get_type
+GstWebRTCRTPTransceiver
+GST_WEBRTC_RTP_TRANSCEIVER
+GST_IS_WEBRTC_RTP_TRANSCEIVER
+GstWebRTCRTPTransceiverClass
+GST_WEBRTC_RTP_TRANSCEIVER_CLASS
+GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS
+GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS
+</SECTION>

docs/libs/gst-plugins-bad-libs.types

@@ -7,6 +7,7 @@
 #include <gst/insertbin/gstinsertbin.h>
 #include <gst/mpegts/mpegts.h>
 #include <gst/player/player.h>
+#include <gst/webrtc/webrtc.h>
 
 gst_audio_aggregator_get_type
 gst_audio_aggregator_pad_get_type
@@ -49,3 +50,22 @@ gst_player_video_overlay_video_renderer_get_type
 gst_player_video_renderer_get_type
 gst_player_visualization_get_type
 
+
+gst_webrtc_dtls_setup_get_type
+gst_webrtc_dtls_transport_get_type
+gst_webrtc_dtls_transport_state_get_type
+gst_webrtc_ice_component_get_type
+gst_webrtc_ice_connection_state_get_type
+gst_webrtc_ice_gathering_state_get_type
+gst_webrtc_ice_role_get_type
+gst_webrtc_sdp_type_get_type
+gst_webrtc_ice_transport_get_type
+gst_webrtc_peer_connection_state_get_type
+gst_webrtc_rtp_receiver_get_type
+gst_webrtc_rtp_sender_get_type
+gst_webrtc_session_description_get_type
+gst_webrtc_signaling_state_get_type
+gst_webrtc_rtp_transceiver_direction_get_type
+gst_webrtc_rtp_transceiver_get_type
+gst_webrtc_stats_type_get_type
+

ext/Makefile.am

@@ -406,6 +406,12 @@ else
 WEBRTCDSP_DIR=
 endif
 
+if USE_WEBRTC
+WEBRTC_DIR=webrtc
+else
+WEBRTC_DIR=
+endif
+
 if USE_TTML
 TTML_DIR=ttml
 else
@@ -482,7 +488,8 @@ SUBDIRS=\
 	$(DTLS_DIR) \
 	$(VULKAN_DIR) \
 	$(WEBRTCDSP_DIR) \
-	$(TTML_DIR)
+	$(TTML_DIR) \
+	$(WEBRTC_DIR)
 
 DIST_SUBDIRS = \
 	assrender \
@@ -551,6 +558,7 @@ DIST_SUBDIRS = \
 	dtls \
 	vulkan \
 	webrtcdsp \
-	ttml
+	ttml \
+	webrtc
 
 include $(top_srcdir)/common/parallel-subdirs.mak

ext/meson.build

@@ -65,6 +65,7 @@ subdir('voaacenc')
 subdir('vulkan')
 subdir('wayland')
 subdir('webrtcdsp')
+subdir('webrtc')
 subdir('webp')
 subdir('x265')
 subdir('zbar')

ext/webrtc/Makefile.am

+plugin_LTLIBRARIES = libgstwebrtc.la
+
+noinst_HEADERS = \
+	fwd.h \
+	gstwebrtcbin.h \
+	gstwebrtcice.h \
+	gstwebrtcstats.h \
+	icestream.h \
+	nicetransport.h \
+	transportstream.h \
+	transportsendbin.h \
+	transportreceivebin.h \
+	utils.h \
+	webrtcsdp.h \
+	webrtctransceiver.h
+
+libgstwebrtc_la_SOURCES = \
+	gstwebrtc.c \
+	gstwebrtcbin.c \
+	gstwebrtcice.c \
+	gstwebrtcstats.c \
+	icestream.c \
+	nicetransport.c \
+	transportstream.c \
+	transportsendbin.c \
+	transportreceivebin.c \
+	utils.c \
+	webrtcsdp.c \
+	webrtctransceiver.c
+
+libgstwebrtc_la_SOURCES += $(BUILT_SOURCES)
+noinst_HEADERS += $(built_headers)
+
+libgstwebrtc_la_CFLAGS = \
+	-I$(top_builddir)/gst-libs \
+	-I$(top_srcdir)/gst-libs \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_BASE_CFLAGS) \
+	$(GST_CFLAGS) \
+	$(GST_SDP_CFLAGS) \
+	$(NICE_CFLAGS)
+libgstwebrtc_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) \
+	$(GST_BASE_LIBS) \
+	$(GST_LIBS) \
+	$(GST_SDP_LIBS) \
+	$(NICE_LIBS) \
+	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la
+
+libgstwebrtc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstwebrtc_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+include $(top_srcdir)/common/gst-glib-gen.mak

ext/webrtc/fwd.h

+/* 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 __WEBRTC_FWD_H__
+#define __WEBRTC_FWD_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstWebRTCBin GstWebRTCBin;
+typedef struct _GstWebRTCBinClass GstWebRTCBinClass;
+typedef struct _GstWebRTCBinPrivate GstWebRTCBinPrivate;
+
+typedef struct _GstWebRTCICE GstWebRTCICE;
+typedef struct _GstWebRTCICEClass GstWebRTCICEClass;
+typedef struct _GstWebRTCICEPrivate GstWebRTCICEPrivate;
+
+typedef struct _GstWebRTCICEStream GstWebRTCICEStream;
+typedef struct _GstWebRTCICEStreamClass GstWebRTCICEStreamClass;
+typedef struct _GstWebRTCICEStreamPrivate GstWebRTCICEStreamPrivate;
+
+typedef struct _GstWebRTCNiceTransport GstWebRTCNiceTransport;
+typedef struct _GstWebRTCNiceTransportClass GstWebRTCNiceTransportClass;
+typedef struct _GstWebRTCNiceTransportPrivate GstWebRTCNiceTransportPrivate;
+
+typedef struct _TransportStream TransportStream;
+typedef struct _TransportStreamClass TransportStreamClass;
+
+typedef struct _TransportSendBin TransportSendBin;
+typedef struct _TransportSendBinClass TransportSendBinClass;
+
+typedef struct _TransportReceiveBin TransportReceiveBin;
+typedef struct _TransportReceiveBinClass TransportReceiveBinClass;
+
+typedef struct _WebRTCTransceiver WebRTCTransceiver;
+typedef struct _WebRTCTransceiverClass WebRTCTransceiverClass;
+
+G_END_DECLS
+
+#endif /* __WEBRTC_FWD_H__ */

ext/webrtc/gstwebrtc.c

+/* 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "gstwebrtcbin.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "webrtcbin", GST_RANK_PRIMARY,
+          GST_TYPE_WEBRTC_BIN))
+    return FALSE;
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    webrtc,
+    "WebRTC plugins",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

ext/webrtc/gstwebrtcbin.h

+/* 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_BIN_H__
+#define __GST_WEBRTC_BIN_H__
+
+#include <gst/sdp/sdp.h>
+#include "fwd.h"
+#include "gstwebrtcice.h"
+
+G_BEGIN_DECLS
+
+#define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark ()
+GQuark gst_webrtc_bin_error_quark (void);
+
+typedef enum
+{
+  GST_WEBRTC_BIN_ERROR_FAILED,
+  GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX,
+  GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION,
+  GST_WEBRTC_BIN_ERROR_INVALID_STATE,
+  GST_WEBRTC_BIN_ERROR_BAD_SDP,
+  GST_WEBRTC_BIN_ERROR_FINGERPRINT,
+} GstWebRTCJSEPSDPError;
+
+GType gst_webrtc_bin_pad_get_type(void);
+#define GST_TYPE_WEBRTC_BIN_PAD            (gst_webrtc_bin_pad_get_type())
+#define GST_WEBRTC_BIN_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPad))
+#define GST_IS_WEBRTC_BIN_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN_PAD))
+#define GST_WEBRTC_BIN_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
+#define GST_IS_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN_PAD))
+#define GST_WEBRTC_BIN_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
+
+typedef struct _GstWebRTCBinPad GstWebRTCBinPad;
+typedef struct _GstWebRTCBinPadClass GstWebRTCBinPadClass;
+
+struct _GstWebRTCBinPad
+{
+  GstGhostPad           parent;
+
+  guint                 mlineindex;
+
+  GstWebRTCRTPTransceiver *trans;
+};
+
+struct _GstWebRTCBinPadClass
+{
+  GstGhostPadClass      parent_class;
+};
+
+GType gst_webrtc_bin_get_type(void);
+#define GST_TYPE_WEBRTC_BIN            (gst_webrtc_bin_get_type())
+#define GST_WEBRTC_BIN(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN,GstWebRTCBin))
+#define GST_IS_WEBRTC_BIN(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN))
+#define GST_WEBRTC_BIN_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
+#define GST_IS_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN))
+#define GST_WEBRTC_BIN_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
+
+struct _GstWebRTCBin
+{
+  GstBin                            parent;
+
+  GstElement                       *rtpbin;
+
+  GstWebRTCSignalingState           signaling_state;
+  GstWebRTCICEGatheringState        ice_gathering_state;
+  GstWebRTCICEConnectionState       ice_connection_state;
+  GstWebRTCPeerConnectionState      peer_connection_state;
+
+  GstWebRTCSessionDescription      *current_local_description;
+  GstWebRTCSessionDescription      *pending_local_description;
+  GstWebRTCSessionDescription      *current_remote_description;
+  GstWebRTCSessionDescription      *pending_remote_description;
+
+  GstWebRTCBinPrivate              *priv;
+};
+
+struct _GstWebRTCBinClass
+{
+  GstBinClass           parent_class;
+};
+
+struct _GstWebRTCBinPrivate
+{
+  guint max_sink_pad_serial;
+
+  gboolean bundle;
+  GArray *transceivers;
+  GArray *session_mid_map;
+  GArray *transports;
+
+  GstWebRTCICE *ice;
+  GArray *ice_stream_map;
+  GArray *pending_ice_candidates;
+
+  /* peerconnection variables */
+  gboolean is_closed;
+  gboolean need_negotiation;
+  gpointer sctp_transport;      /* FIXME */
+
+  /* peerconnection helper thread for promises */
+  GMainContext *main_context;
+  GMainLoop *loop;
+  GThread *thread;
+  GMutex pc_lock;
+  GCond pc_cond;
+
+  gboolean running;
+  gboolean async_pending;
+
+  GList *pending_pads;
+
+  /* count of the number of media streams we've offered for uniqueness */
+  /* FIXME: overflow? */
+  guint media_counter;
+
+  GstStructure *stats;
+};
+
+typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);
+
+typedef struct
+{
+  GstWebRTCBin *webrtc;
+  GstWebRTCBinFunc op;
+  gpointer data;
+  GDestroyNotify notify;
+//  GstPromise *promise;      /* FIXME */
+} GstWebRTCBinTask;
+
+void            gst_webrtc_bin_enqueue_task             (GstWebRTCBin * pc,
+                                                         GstWebRTCBinFunc func,
+                                                         gpointer data,
+                                                         GDestroyNotify notify);
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_BIN_H__ */