diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index 0b10519389..9b84fa317d 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/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,
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c b/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
index 221ac4eb9a..67c8143d99 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
+++ b/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)
 {
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.h b/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.h
index c55709b504..80d21203c2 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.h
+++ b/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,
diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
index cb720bd37b..8fa8eeaf76 100644
--- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
+++ b/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);