webrtcbin: Expose SCTP Transport

Being able to access the SCTP Transport object from the application
means the application can access the associated DTLS Transport object
and its ICE Transport object. This means we can observe the ICE state
also for a data-channel-only session. The collated
ice-connection-state on webrtcbin only includes the ICE Transport
objects that resides on the RTP transceivers (which is exactly how it
is specified in
https://w3c.github.io/webrtc-pc/#rtciceconnectionstate-enum).

For the consent freshness functionality (RFC 7675) to work the ICE
state must be accessible and consequently the SCTP transport must be
accessible for enabling consent freshness checking for a
data-channel-only session.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>

diff --git a/docs/plugins/gst_plugins_cache.json b/docs/plugins/gst_plugins_cache.json
index dd166e4..86dd1c5 100644 (file)
--- a/docs/plugins/gst_plugins_cache.json
+++ b/docs/plugins/gst_plugins_cache.json
@@ -227981,6 +227981,17 @@
                         "type": "GstWebRTCSessionDescription",
                         "writable": false
                     },
+                    "sctp-transport": {
+                        "blurb": "The WebRTC SCTP Transport",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "mutable": "null",
+                        "readable": true,
+                        "type": "GstObject",
+                        "writable": false
+                    },
                     "signaling-state": {
                         "blurb": "The signaling state of this element",
                         "conditionally-available": false,

diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index fdc53f5..c6855fc 100644 (file)
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -467,7 +467,8 @@ enum
   PROP_BUNDLE_POLICY,
   PROP_ICE_TRANSPORT_POLICY,
   PROP_ICE_AGENT,
-  PROP_LATENCY
+  PROP_LATENCY,
+  PROP_SCTP_TRANSPORT,
 };
 
 static guint gst_webrtc_bin_signals[LAST_SIGNAL] = { 0 };
@@ -6914,6 +6915,9 @@ gst_webrtc_bin_get_property (GObject * object, guint prop_id,
     case PROP_LATENCY:
       g_value_set_uint (value, webrtc->priv->jb_latency);
       break;
+    case PROP_SCTP_TRANSPORT:
+      g_value_set_object (value, webrtc->priv->sctp_transport);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -7192,6 +7196,19 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
   /**
+   * GstWebRTCBin:sctp-transport:
+   *
+   * The WebRTC SCTP Transport
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_SCTP_TRANSPORT,
+      g_param_spec_object ("sctp-transport", "WebRTC SCTP Transport",
+          "The WebRTC SCTP Transport",
+          GST_TYPE_OBJECT, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  /**
    * GstWebRTCBin::create-offer:
    * @object: the #webrtcbin
    * @options: (nullable): create-offer options