webrtcbin: Allow session level setup attribute in SDP

An SDP answer can declare its setup attribute at the session level or at the
media level. Until this patch we were validating only the latter case and an
assert was raised in the former case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6930>

subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c

@@ -5747,7 +5747,15 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
     GstWebRTCDTLSSetup local_setup, remote_setup;
 
     local_setup = _get_dtls_setup_from_media (local_media);
+    if (local_setup == GST_WEBRTC_DTLS_SETUP_NONE)
+      local_setup =
+          _get_dtls_setup_from_session (webrtc->current_local_description->sdp);
+
     remote_setup = _get_dtls_setup_from_media (remote_media);
+    if (remote_setup == GST_WEBRTC_DTLS_SETUP_NONE)
+      remote_setup =
+          _get_dtls_setup_from_session (webrtc->
+          current_remote_description->sdp);
     new_setup = _get_final_setup (local_setup, remote_setup);
     if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
       g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,

subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c

@@ -236,9 +236,20 @@ _media_get_ice_pwd (const GstSDPMessage * msg, guint media_idx)
 }
 
 static gboolean
-_media_has_setup (const GstSDPMedia * media, guint media_idx, GError ** error)
+_validate_setup_attribute (const gchar * setup, GError ** error)
 {
   static const gchar *valid_setups[] = { "actpass", "active", "passive", NULL };
+  if (!g_strv_contains (valid_setups, setup)) {
+    g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
+        "SDP contains unknown \'setup\' attribute, \'%s\'", setup);
+    return FALSE;
+  }
+  return TRUE;
+}
+
+static gboolean
+_media_has_setup (const GstSDPMedia * media, guint media_idx, GError ** error)
+{
   const gchar *setup = gst_sdp_media_get_attribute_val (media, "setup");
   if (IS_EMPTY_SDP_ATTRIBUTE (setup)) {
     g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
@@ -246,13 +257,7 @@ _media_has_setup (const GstSDPMedia * media, guint media_idx, GError ** error)
         media_idx);
     return FALSE;
   }
-  if (!g_strv_contains (valid_setups, setup)) {
-    g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
-        "media %u contains unknown \'setup\' attribute, \'%s\'", media_idx,
-        setup);
-    return FALSE;
-  }
-  return TRUE;
+  return _validate_setup_attribute (setup, error);
 }
 
 #if 0
@@ -273,9 +278,11 @@ gboolean
 validate_sdp (GstWebRTCSignalingState state, SDPSource source,
     GstWebRTCSessionDescription * sdp, GError ** error)
 {
-  const gchar *group, *bundle_ice_ufrag = NULL, *bundle_ice_pwd = NULL;
+  const gchar *group, *bundle_ice_ufrag = NULL, *bundle_ice_pwd = NULL, *setup =
+      NULL;
   gchar **group_members = NULL;
   gboolean is_bundle = FALSE;
+  gboolean has_session_setup = FALSE;
   int i;
 
   if (!_check_valid_state_for_sdp_change (state, source, sdp->type, error))
@@ -290,6 +297,13 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
   if (is_bundle)
     group_members = g_strsplit (&group[6], " ", -1);
 
+  setup = gst_sdp_message_get_attribute_val (sdp->sdp, "setup");
+  if (setup) {
+    if (!_validate_setup_attribute (setup, error))
+      return FALSE;
+    has_session_setup = TRUE;
+  }
+
   for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
     const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
     const gchar *mid;
@@ -310,7 +324,7 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
           "media %u is missing or contains an empty \'ice-pwd\' attribute", i);
       goto fail;
     }
-    if (!_media_has_setup (media, i, error))
+    if (!has_session_setup && !_media_has_setup (media, i, error))
       goto fail;
     /* check parameters in bundle are the same */
     if (media_in_bundle) {
@@ -527,6 +541,26 @@ _get_dtls_setup_from_media (const GstSDPMedia * media)
   return SETUP (NONE);
 }
 
+GstWebRTCDTLSSetup
+_get_dtls_setup_from_session (const GstSDPMessage * sdp)
+{
+  const gchar *attr = gst_sdp_message_get_attribute_val (sdp, "setup");
+  if (!attr) {
+    GST_LOG ("no setup attribute in session");
+    return SETUP (NONE);
+  }
+  if (g_strcmp0 (attr, "actpass") == 0) {
+    return SETUP (ACTPASS);
+  } else if (g_strcmp0 (attr, "active") == 0) {
+    return SETUP (ACTIVE);
+  } else if (g_strcmp0 (attr, "passive") == 0) {
+    return SETUP (PASSIVE);
+  }
+
+  GST_ERROR ("unknown setup value %s", attr);
+  return SETUP (NONE);
+}
+
 GstWebRTCDTLSSetup
 _intersect_dtls_setup (GstWebRTCDTLSSetup offer)
 {

subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.h

@@ -58,6 +58,8 @@ GstWebRTCRTPTransceiverDirection    _get_final_direction                     (Gs
 G_GNUC_INTERNAL
 GstWebRTCDTLSSetup                  _get_dtls_setup_from_media              (const GstSDPMedia * media);
 G_GNUC_INTERNAL
+GstWebRTCDTLSSetup                  _get_dtls_setup_from_session            (const GstSDPMessage * sdp);
+G_GNUC_INTERNAL
 GstWebRTCDTLSSetup                  _intersect_dtls_setup                   (GstWebRTCDTLSSetup offer);
 G_GNUC_INTERNAL
 void                                _media_replace_setup                    (GstSDPMedia * media,

subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c

@@ -5844,6 +5844,76 @@ GST_START_TEST (test_ice_end_of_candidates)
 
 GST_END_TEST;
 
+static void
+_set_setup_session_attr_on_answer (struct test_webrtc *t, GstElement * element,
+    GstPromise * promise, gpointer user_data)
+{
+  GstSDPMessage *sdp;
+  GstSDPMessage *modified_sdp = NULL;
+  const GstSDPMedia *media;
+  GstSDPMedia *modified_media;
+  const gchar *attr;
+
+  if (TEST_IS_OFFER_ELEMENT (t, element))
+    return;
+
+  sdp = t->answer_desc->sdp;
+  media = gst_sdp_message_get_media (sdp, 0);
+  attr = gst_sdp_media_get_attribute_val (media, "setup");
+
+  /* Remove the setup attribute from first media */
+  gst_sdp_media_copy (media, &modified_media);
+  for (unsigned index = 0;
+      index < gst_sdp_media_attributes_len (modified_media); index++) {
+    const GstSDPAttribute *current =
+        gst_sdp_media_get_attribute (modified_media, index);
+    if (!g_str_equal (current->key, "setup"))
+      continue;
+    gst_sdp_media_remove_attribute (modified_media, index);
+    break;
+  }
+
+  gst_sdp_message_copy (sdp, &modified_sdp);
+
+  /* Add session-level setup attribute to modified answer */
+  gst_sdp_message_add_attribute (modified_sdp, "setup", attr);
+
+  /* Replace first media of answer with a media without session attribute */
+  gst_sdp_message_remove_media (modified_sdp, 0);
+  gst_sdp_message_add_media (modified_sdp, modified_media);
+  gst_sdp_media_free (modified_media);
+
+  gst_sdp_message_free (sdp);
+  t->answer_desc->sdp = modified_sdp;
+}
+
+static void
+_offer_created_do_nothing (struct test_webrtc *t, GstElement * element,
+    GstPromise * promise, gpointer user_data)
+{
+}
+
+GST_START_TEST (test_sdp_session_setup_attribute)
+{
+  struct test_webrtc *t = create_audio_test ();
+
+  t->on_offer_created = _offer_created_do_nothing;
+  t->on_answer_created = _set_setup_session_attr_on_answer;
+
+  fail_if (gst_element_set_state (t->webrtc1, GST_STATE_READY) ==
+      GST_STATE_CHANGE_FAILURE);
+  fail_if (gst_element_set_state (t->webrtc2, GST_STATE_READY) ==
+      GST_STATE_CHANGE_FAILURE);
+  test_webrtc_create_offer (t);
+  test_webrtc_wait_for_state_mask (t, 1 << STATE_ANSWER_SET);
+
+  test_webrtc_wait_for_ice_gathering_complete (t);
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
 static Suite *
 webrtcbin_suite (void)
 {
@@ -5910,6 +5980,7 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_add_turn_server);
     tcase_add_test (tc, test_msid);
     tcase_add_test (tc, test_ice_end_of_candidates);
+    tcase_add_test (tc, test_sdp_session_setup_attribute);
     if (sctpenc && sctpdec) {
       tcase_add_test (tc, test_data_channel_create);
       tcase_add_test (tc, test_data_channel_create_two_channels);