diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index be0d77b3cf..bcea449c01 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -2724,6 +2724,23 @@ typedef struct
   WebRTCTransceiver *trans;
 } RtxSsrcData;
 
+/* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
+static gchar *
+_construct_msid (WebRTCTransceiver * trans, guint ssrc)
+{
+  gchar *ret;
+
+  if (trans->msid_appdata) {
+    ret =
+        g_strdup_printf ("%u msid:%s %s", ssrc, GST_OBJECT_NAME (trans),
+        trans->msid_appdata);
+  } else {
+    ret = g_strdup_printf ("%u msid:%s", ssrc, GST_OBJECT_NAME (trans));
+  }
+
+  return ret;
+}
+
 static gboolean
 _media_add_rtx_ssrc (GQuark field_id, const GValue * value, RtxSsrcData * data)
 {
@@ -2735,10 +2752,8 @@ _media_add_rtx_ssrc (GQuark field_id, const GValue * value, RtxSsrcData * data)
   /* http://www.freesoft.org/CIE/RFC/1889/24.htm */
   cname = gst_structure_get_string (sdes, "cname");
 
-  /* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
-  str =
-      g_strdup_printf ("%u msid:%s %s", g_value_get_uint (value),
-      cname, GST_OBJECT_NAME (data->trans));
+  str = _construct_msid (data->trans, g_value_get_uint (value));
+
   gst_sdp_media_add_attribute (data->media, "ssrc", str);
   g_free (str);
 
@@ -2775,10 +2790,8 @@ _media_add_ssrcs (GstSDPMedia * media, GstCaps * caps, GstWebRTCBin * webrtc,
     if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
       gchar *str;
 
-      /* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
-      str =
-          g_strdup_printf ("%u msid:%s %s", ssrc, cname,
-          GST_OBJECT_NAME (trans));
+      str = _construct_msid (trans, ssrc);
+
       gst_sdp_media_add_attribute (media, "ssrc", str);
       g_free (str);
 
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c b/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c
index ed4f90654c..b754e92f3d 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c
@@ -38,6 +38,7 @@ G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
 #define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
 #define DEFAULT_DO_NACK FALSE
 #define DEFAULT_FEC_PERCENTAGE 100
+#define DEFAULT_MSID_APPDATA NULL
 
 enum
 {
@@ -46,6 +47,7 @@ enum
   PROP_FEC_TYPE,
   PROP_FEC_PERCENTAGE,
   PROP_DO_NACK,
+  PROP_MSID_APPDATA,
 };
 
 void
@@ -112,6 +114,22 @@ webrtc_transceiver_set_property (GObject * object, guint prop_id,
     case PROP_FEC_PERCENTAGE:
       trans->fec_percentage = g_value_get_uint (value);
       break;
+    case PROP_MSID_APPDATA:
+    {
+      gchar *new_msid_appdata = g_value_dup_string (value);
+
+      if (new_msid_appdata && strlen (new_msid_appdata) > 64) {
+        g_warning ("Msid appdata exceeds 64 characters: %s", new_msid_appdata);
+        g_free (new_msid_appdata);
+      } else {
+        if (trans->msid_appdata)
+          g_free (trans->msid_appdata);
+
+        trans->msid_appdata = new_msid_appdata;
+      }
+
+      break;
+    }
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -136,6 +154,9 @@ webrtc_transceiver_get_property (GObject * object, guint prop_id,
     case PROP_FEC_PERCENTAGE:
       g_value_set_uint (value, trans->fec_percentage);
       break;
+    case PROP_MSID_APPDATA:
+      g_value_set_string (value, trans->msid_appdata);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -152,6 +173,7 @@ webrtc_transceiver_finalize (GObject * object)
   gst_clear_object (&trans->ulpfecdec);
   gst_clear_object (&trans->ulpfecenc);
   gst_clear_object (&trans->redenc);
+  g_clear_pointer (&trans->msid_appdata, g_free);
 
   if (trans->local_rtx_ssrc_map)
     gst_structure_free (trans->local_rtx_ssrc_map);
@@ -203,6 +225,23 @@ webrtc_transceiver_class_init (WebRTCTransceiverClass * klass)
           "The amount of Forward Error Correction to apply",
           0, 100, DEFAULT_FEC_PERCENTAGE,
           G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * WebRTCTransceiver:msid-appdata:
+   *
+   * The appdata part of the media stream id, must not exceed 64 characters.
+   *
+   * Consult https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-msid-16#section-2
+   * for more details.
+   *
+   * Since: 1.22
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_MSID_APPDATA,
+      g_param_spec_string ("msid-appdata", "Msid appdata",
+          "The appdata part of the media stream id, must not exceed 64 characters",
+          NULL,
+          G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 }
 
 static void
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.h b/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.h
index a3c80c5117..b0528efec3 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.h
+++ b/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.h
@@ -47,6 +47,7 @@ struct _WebRTCTransceiver
   GstWebRTCFECType         fec_type;
   guint                    fec_percentage;
   gboolean                 do_nack;
+  gchar                   *msid_appdata;
 
   GstCaps                  *last_configured_caps;
 
diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
index 8508bd15e3..f7b44c8a7d 100644
--- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
+++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
@@ -4425,6 +4425,66 @@ GST_START_TEST (test_renego_rtx)
 
 GST_END_TEST;
 
+static void
+_check_msid_appdata (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer expected)
+{
+  guint i = 0;
+
+  for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
+    const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
+
+    if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0
+        || g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
+      int j;
+
+      for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
+        const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
+
+        if (!g_strcmp0 (attr->key, "ssrc")) {
+          gchar **split = g_strsplit (attr->value, " ", 3);
+
+          fail_unless (g_strv_length (split) >= 2);
+
+          if (g_str_has_prefix (split[1], "msid:")) {
+            if (expected) {
+              fail_unless_equals_int (g_strv_length (split), 3);
+              fail_unless_equals_string (split[2], expected);
+            } else {
+              fail_unless_equals_int (g_strv_length (split), 2);
+            }
+          }
+
+          g_strfreev (split);
+        }
+      }
+    }
+  }
+}
+
+GST_START_TEST (test_msid_appdata)
+{
+  struct test_webrtc *t = create_audio_test ();
+  gchar *expected = g_strdup ("foobar");
+  GstWebRTCRTPTransceiver *trans;
+
+  VAL_SDP_INIT (offer, _check_msid_appdata, expected, NULL);
+  VAL_SDP_INIT (answer, _check_msid_appdata, expected, NULL);
+
+  g_signal_emit_by_name (t->webrtc1, "get-transceiver", 0, &trans);
+  g_object_set (trans, "msid-appdata", expected, NULL);
+  g_clear_object (&trans);
+
+  test_validate_sdp (t, &offer, &answer);
+
+  g_free (expected);
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+
 static Suite *
 webrtcbin_suite (void)
 {
@@ -4480,6 +4540,7 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_codec_preferences_incompatible_extmaps);
     tcase_add_test (tc, test_codec_preferences_invalid_extmap);
     tcase_add_test (tc, test_renego_rtx);
+    tcase_add_test (tc, test_msid_appdata);
     if (sctpenc && sctpdec) {
       tcase_add_test (tc, test_data_channel_create);
       tcase_add_test (tc, test_data_channel_remote_notify);