diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index b2d343f20e..5a073e8bba 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -4540,8 +4540,6 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
     }
 
     mid = gst_sdp_media_get_attribute_val (media, "mid");
-    /* XXX: not strictly required but a lot of functionality requires a mid */
-    g_assert (mid);
 
     /* set the a=setup: attribute */
     offer_setup = _get_dtls_setup_from_media (offer_media);
@@ -4603,12 +4601,22 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
 
       _remove_optional_offer_fields (offer_caps);
 
-      rtp_trans = _find_transceiver_for_mid (webrtc, mid);
-      if (!rtp_trans) {
-        g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
-            "Transceiver for media with mid %s not found", mid);
-        gst_caps_unref (offer_caps);
-        goto rejected;
+      if (mid) {
+        rtp_trans = _find_transceiver_for_mid (webrtc, mid);
+        if (!rtp_trans) {
+          g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
+              "Transceiver for media with mid %s not found", mid);
+          gst_caps_unref (offer_caps);
+          goto rejected;
+        }
+      } else {
+        rtp_trans = _find_transceiver_for_mline (webrtc, i);
+        if (!rtp_trans) {
+          g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
+              "Transceiver for media with mline %u not found", i);
+          gst_caps_unref (offer_caps);
+          goto rejected;
+        }
       }
       GstCaps *current_caps =
           _find_codec_preferences (webrtc, rtp_trans, i, error);
@@ -4623,7 +4631,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
         const gchar *last_mid =
             gst_sdp_media_get_attribute_val (last_media, "mid");
         /* FIXME: assumes no shenanigans with recycling transceivers */
-        g_assert (g_strcmp0 (mid, last_mid) == 0);
+        if (mid != last_mid)
+          g_assert (g_strcmp0 (mid, last_mid) == 0);
         if (!current_caps)
           current_caps = _rtp_caps_from_media (last_media);
       }
@@ -6433,11 +6442,10 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
     mid = gst_sdp_media_get_attribute_val (media, "mid");
     direction = _get_direction_from_media (media);
 
-    /* XXX: not strictly required but a lot of functionality requires a mid */
-    if (!mid) {
-      g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
-          "Missing mid attribute in media");
-      goto out;
+    if (mid) {
+      trans = _find_transceiver_for_mid (webrtc, mid);
+    } else {
+      trans = _find_transceiver_for_mline (webrtc, i);
     }
 
     if (bundled)
@@ -6445,7 +6453,6 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
     else
       transport_idx = i;
 
-    trans = _find_transceiver_for_mid (webrtc, mid);
 
     if (sd->source == SDP_LOCAL) {
       /* If the media description was not yet associated with an RTCRtpTransceiver object then run the following steps: */
@@ -6469,8 +6476,10 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
         }
         trans->mline = i;
         /* Set transceiver.[[Mid]] to transceiver.[[JsepMid]] */
-        g_free (trans->mid);
-        trans->mid = g_strdup (mid);
+        g_clear_pointer (&trans->mid, g_free);
+        if (mid) {
+          trans->mid = g_strdup (mid);
+        }
         g_object_notify (G_OBJECT (trans), "mid");
         /* If transceiver.[[Stopped]] is true, abort these sub steps */
         if (trans->stopped)
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c b/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
index 0ece0c6250..4e2636be2c 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/webrtcsdp.c
@@ -193,16 +193,10 @@ _media_has_attribute_key (const GstSDPMedia * media, const gchar * key)
 }
 
 static gboolean
-_media_has_mid (const GstSDPMedia * media, guint media_idx, GError ** error)
+_media_has_mid (const GstSDPMedia * media, guint media_idx)
 {
   const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
-  if (IS_EMPTY_SDP_ATTRIBUTE (mid)) {
-    g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
-        "media %u is missing or contains an empty \'mid\' attribute",
-        media_idx);
-    return FALSE;
-  }
-  return TRUE;
+  return !IS_EMPTY_SDP_ATTRIBUTE (mid);
 }
 
 const gchar *
@@ -308,11 +302,13 @@ validate_sdp (GstWebRTCSignalingState state, SDPSource source,
     const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
     const gchar *mid;
     gboolean media_in_bundle = FALSE;
-    if (!_media_has_mid (media, i, error))
-      goto fail;
-    mid = gst_sdp_media_get_attribute_val (media, "mid");
-    media_in_bundle = is_bundle
-        && g_strv_contains ((const gchar **) group_members, mid);
+
+    if (_media_has_mid (media, i)) {
+      mid = gst_sdp_media_get_attribute_val (media, "mid");
+      media_in_bundle =
+          is_bundle && g_strv_contains ((const gchar **) group_members, mid);
+    }
+
     if (!_media_get_ice_ufrag (sdp->sdp, i)) {
       g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
           "media %u is missing or contains an empty \'ice-ufrag\' attribute",
diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
index c2bb5edf11..bb06c3dc94 100644
--- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
+++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
@@ -5044,6 +5044,166 @@ a=group:BUNDLE \r\n\
 
 GST_END_TEST;
 
+static GstWebRTCSessionDescription *
+remove_sdp_attributes (const GstWebRTCSessionDescription * desc, guint n_attrs,
+    const char *const *attributes)
+{
+  GstSDPMessage *sdp;
+  guint total_medias = gst_sdp_message_medias_len (desc->sdp);
+
+  gst_sdp_message_copy (desc->sdp, &sdp);
+  for (guint i = 0; i < total_medias; i++) {
+    gst_sdp_message_remove_media (sdp, i);
+  }
+
+  for (guint i = 0; i < total_medias; i++) {
+    const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
+    guint total_attributes = gst_sdp_media_attributes_len (media);
+    GstSDPMedia *new_media;
+
+    gst_sdp_media_copy (media, &new_media);
+    for (guint ii = 0; ii < total_attributes; ii++) {
+      const GstSDPAttribute *attribute =
+          gst_sdp_media_get_attribute (new_media, ii);
+      for (guint iii = 0; iii < n_attrs; iii++) {
+        if (!g_strcmp0 (attribute->key, attributes[iii])) {
+          gst_sdp_media_remove_attribute (new_media, ii);
+          ii--;
+          total_attributes--;
+          break;
+        }
+      }
+    }
+    gst_sdp_message_add_media (sdp, new_media);
+    gst_sdp_media_free (new_media);
+  }
+
+  return gst_webrtc_session_description_new (desc->type, sdp);
+}
+
+static void
+do_missing_mid_test (gboolean is_offer)
+{
+  struct test_webrtc *t = test_webrtc_new ();
+  const gchar *attributes_to_remove[] = { "mid", "group" };
+  GstWebRTCSessionDescription *modified_desc = NULL;
+  GstPromise *promise;
+  GstPromiseResult res;
+  const GstStructure *s;
+  GstWebRTCSessionDescription *desc;
+  GstHarness *h1;
+
+  t->on_negotiation_needed = NULL;
+  t->on_ice_candidate = NULL;
+  t->on_pad_added = _pad_added_fakesink;
+
+  h1 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
+  add_audio_test_src_harness (h1, 0xDEADBEEF);
+  t->harnesses = g_list_prepend (t->harnesses, h1);
+
+  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);
+
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc1, "create-offer", NULL, promise);
+  res = gst_promise_wait (promise);
+  fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
+  s = gst_promise_get_reply (promise);
+  fail_unless (s != NULL);
+  fail_if (gst_structure_has_field (s, "error"));
+  gst_structure_get (s, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &desc,
+      NULL);
+  fail_unless (desc != NULL);
+  gst_promise_unref (promise);
+
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc1, "set-local-description", desc, promise);
+  res = gst_promise_wait (promise);
+  fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
+  s = gst_promise_get_reply (promise);
+  fail_if (s && gst_structure_has_field (s, "error"));
+  gst_promise_unref (promise);
+
+  if (is_offer) {
+    modified_desc =
+        remove_sdp_attributes (desc, G_N_ELEMENTS (attributes_to_remove),
+        attributes_to_remove);
+  }
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc2, "set-remote-description",
+      modified_desc ? modified_desc : desc, promise);
+  res = gst_promise_wait (promise);
+  fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
+  s = gst_promise_get_reply (promise);
+  fail_if (s && gst_structure_has_field (s, "error"));
+  gst_promise_unref (promise);
+
+  g_clear_pointer (&modified_desc, gst_webrtc_session_description_free);
+  gst_webrtc_session_description_free (desc);
+
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc2, "create-answer", NULL, promise);
+  res = gst_promise_wait (promise);
+  fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
+  s = gst_promise_get_reply (promise);
+  fail_unless (s != NULL);
+  s = gst_promise_get_reply (promise);
+  fail_unless (s != NULL);
+  fail_if (gst_structure_has_field (s, "error"));
+  gst_structure_get (s, "answer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &desc,
+      NULL);
+  fail_unless (desc != NULL);
+  gst_promise_unref (promise);
+
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc2, "set-local-description", desc, promise);
+  res = gst_promise_wait (promise);
+  fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
+  s = gst_promise_get_reply (promise);
+  fail_if (s && gst_structure_has_field (s, "error"));
+  gst_promise_unref (promise);
+
+  if (!is_offer) {
+    modified_desc =
+        remove_sdp_attributes (desc, G_N_ELEMENTS (attributes_to_remove),
+        attributes_to_remove);
+  }
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc1, "set-remote-description",
+      modified_desc ? modified_desc : desc, promise);
+  res = gst_promise_wait (promise);
+  fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
+  s = gst_promise_get_reply (promise);
+  fail_if (s && gst_structure_has_field (s, "error"));
+  gst_promise_unref (promise);
+
+  g_clear_pointer (&modified_desc, gst_webrtc_session_description_free);
+  gst_webrtc_session_description_free (desc);
+
+  fail_if (gst_element_set_state (t->webrtc1, GST_STATE_PLAYING) ==
+      GST_STATE_CHANGE_FAILURE);
+  fail_if (gst_element_set_state (t->webrtc2, GST_STATE_PLAYING) ==
+      GST_STATE_CHANGE_FAILURE);
+
+  test_webrtc_free (t);
+}
+
+GST_START_TEST (test_missing_mid_in_offer)
+{
+  do_missing_mid_test (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_missing_mid_in_answer)
+{
+  do_missing_mid_test (FALSE);
+}
+
+GST_END_TEST;
+
 static void
 new_jitterbuffer_set_fast_start (GstElement * rtpbin,
     GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
@@ -6624,6 +6784,8 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_sdp_session_setup_attribute);
     tcase_add_test (tc, test_rtp_header_extension_sendonly_recvonly_pair);
     tcase_add_test (tc, test_invalid_bundle_in_pending_remote_description);
+    tcase_add_test (tc, test_missing_mid_in_offer);
+    tcase_add_test (tc, test_missing_mid_in_answer);
     if (sctpenc && sctpdec) {
       tcase_add_test (tc, test_data_channel_create);
       tcase_add_test (tc, test_data_channel_create_two_channels);