diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index 9531e42d45..79b79d6eae 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -31,8 +31,6 @@
 #include "webrtcdatachannel.h"
 #include "sctptransport.h"
 
-#include <gst/rtp/rtp.h>
-
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
@@ -249,21 +247,6 @@ gst_webrtc_bin_pad_class_init (GstWebRTCBinPadClass * klass)
           G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 }
 
-static void
-gst_webrtc_bin_pad_update_ssrc_event (GstWebRTCBinPad * wpad)
-{
-  if (wpad->received_caps) {
-    WebRTCTransceiver *trans = (WebRTCTransceiver *) wpad->trans;
-    GstPad *pad = GST_PAD (wpad);
-
-    trans->ssrc_event =
-        gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
-        gst_structure_new ("GstWebRtcBinUpdateTos", "ssrc", G_TYPE_UINT,
-            trans->current_ssrc, NULL));
-    gst_pad_send_event (pad, gst_event_ref (trans->ssrc_event));
-  }
-}
-
 static gboolean
 gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
 {
@@ -289,7 +272,6 @@ gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
 
       s = gst_caps_get_structure (caps, 0);
       gst_structure_get_uint (s, "ssrc", &trans->current_ssrc);
-      gst_webrtc_bin_pad_update_ssrc_event (wpad);
     }
   } else if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
     check_negotiation = TRUE;
@@ -1653,242 +1635,6 @@ _on_dtls_transport_notify_state (GstWebRTCDTLSTransport * transport,
   _update_peer_connection_state (webrtc);
 }
 
-static gboolean
-match_ssrc (GstWebRTCRTPTransceiver * rtp_trans, gconstpointer data)
-{
-  WebRTCTransceiver *trans = (WebRTCTransceiver *) rtp_trans;
-
-  return (trans->current_ssrc == GPOINTER_TO_UINT (data));
-}
-
-static gboolean
-_on_sending_rtcp (GObject * internal_session, GstBuffer * buffer,
-    gboolean early, gpointer user_data)
-{
-  GstWebRTCBin *webrtc = user_data;
-  GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
-  GstRTCPPacket packet;
-
-  if (!gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp))
-    goto done;
-
-  if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
-    if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_SR) {
-      guint32 ssrc;
-      GstWebRTCRTPTransceiver *rtp_trans;
-      WebRTCTransceiver *trans;
-
-      gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
-          NULL);
-
-      rtp_trans = _find_transceiver (webrtc, GUINT_TO_POINTER (ssrc),
-          match_ssrc);
-      trans = (WebRTCTransceiver *) rtp_trans;
-
-      if (rtp_trans && rtp_trans->sender && trans->ssrc_event) {
-        GstPad *pad;
-        gchar *pad_name = NULL;
-
-        pad_name =
-            g_strdup_printf ("send_rtcp_src_%u",
-            rtp_trans->sender->rtcp_transport->session_id);
-        pad = gst_element_get_static_pad (webrtc->rtpbin, pad_name);
-        g_free (pad_name);
-        if (pad) {
-          gst_pad_push_event (pad, gst_event_ref (trans->ssrc_event));
-          gst_object_unref (pad);
-        }
-      }
-    }
-  }
-
-  gst_rtcp_buffer_unmap (&rtcp);
-
-done:
-  /* False means we don't care about suppression */
-  return FALSE;
-}
-
-static void
-gst_webrtc_bin_attach_tos_to_session (GstWebRTCBin * webrtc, guint session_id)
-{
-  GObject *internal_session = NULL;
-
-  g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
-      session_id, &internal_session);
-
-  if (internal_session) {
-    g_signal_connect (internal_session, "on-sending-rtcp",
-        G_CALLBACK (_on_sending_rtcp), webrtc);
-    g_object_unref (internal_session);
-  }
-}
-
-static GstPadProbeReturn
-_nicesink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
-{
-  GstWebRTCBin *webrtc = user_data;
-
-  if (GST_EVENT_TYPE (GST_PAD_PROBE_INFO_EVENT (info))
-      == GST_EVENT_CUSTOM_DOWNSTREAM_STICKY) {
-    const GstStructure *s =
-        gst_event_get_structure (GST_PAD_PROBE_INFO_EVENT (info));
-
-    if (gst_structure_has_name (s, "GstWebRtcBinUpdateTos")) {
-      guint ssrc;
-      gint priority;
-
-      if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
-        GstWebRTCRTPTransceiver *rtp_trans;
-
-        rtp_trans = _find_transceiver (webrtc, GUINT_TO_POINTER (ssrc),
-            match_ssrc);
-        if (rtp_trans) {
-          WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
-          GstWebRTCICEStream *stream = _find_ice_stream_for_session (webrtc,
-              trans->stream->session_id);
-          guint8 dscp = 0;
-
-          /* Set DSCP field based on
-           * https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5
-           */
-          switch (rtp_trans->sender->priority) {
-            case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
-              dscp = 8;         /* CS1 */
-              break;
-            case GST_WEBRTC_PRIORITY_TYPE_LOW:
-              dscp = 0;         /* DF */
-              break;
-            case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
-              switch (rtp_trans->kind) {
-                case GST_WEBRTC_KIND_AUDIO:
-                  dscp = 46;    /* EF */
-                  break;
-                case GST_WEBRTC_KIND_VIDEO:
-                  dscp = 38;    /* AF43 *//* TODO: differentiate non-interactive */
-                  break;
-                case GST_WEBRTC_KIND_UNKNOWN:
-                  dscp = 0;
-                  break;
-              }
-              break;
-            case GST_WEBRTC_PRIORITY_TYPE_HIGH:
-              switch (rtp_trans->kind) {
-                case GST_WEBRTC_KIND_AUDIO:
-                  dscp = 46;    /* EF */
-                  break;
-                case GST_WEBRTC_KIND_VIDEO:
-                  dscp = 36;    /* AF42 *//* TODO: differentiate non-interactive */
-                  break;
-                case GST_WEBRTC_KIND_UNKNOWN:
-                  dscp = 0;
-                  break;
-              }
-              break;
-          }
-
-          gst_webrtc_ice_set_tos (webrtc->priv->ice, stream, dscp << 2);
-        }
-      } else if (gst_structure_get_enum (s, "sctp-priority",
-              GST_TYPE_WEBRTC_PRIORITY_TYPE, &priority)) {
-        guint8 dscp = 0;
-
-        /* Set DSCP field based on
-         * https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5
-         */
-        switch (priority) {
-          case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
-            dscp = 8;           /* CS1 */
-            break;
-          case GST_WEBRTC_PRIORITY_TYPE_LOW:
-            dscp = 0;           /* DF */
-            break;
-          case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
-            dscp = 10;          /* AF11 */
-            break;
-          case GST_WEBRTC_PRIORITY_TYPE_HIGH:
-            dscp = 18;          /* AF21 */
-            break;
-        }
-        if (webrtc->priv->data_channel_transport)
-          gst_webrtc_ice_set_tos (webrtc->priv->ice,
-              webrtc->priv->data_channel_transport->stream, dscp << 2);
-      }
-    }
-  }
-  return GST_PAD_PROBE_OK;
-}
-
-static void gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc);
-
-static void
-gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc)
-{
-  GstWebRTCPriorityType sctp_priority = 0;
-  guint i;
-
-  if (!webrtc->priv->sctp_transport)
-    return;
-
-  for (i = 0; i < webrtc->priv->data_channels->len; i++) {
-    GstWebRTCDataChannel *channel
-        = g_ptr_array_index (webrtc->priv->data_channels, i);
-
-    sctp_priority = MAX (sctp_priority, channel->priority);
-  }
-
-  /* Default priority is low means DSCP field is left as 0 */
-  if (sctp_priority == 0)
-    sctp_priority = GST_WEBRTC_PRIORITY_TYPE_LOW;
-
-  /* Nobody asks for DSCP, leave it as-is */
-  if (sctp_priority == GST_WEBRTC_PRIORITY_TYPE_LOW &&
-      !webrtc->priv->tos_attached)
-    return;
-
-  /* If one stream has a non-default priority, then everyone else does too */
-  gst_webrtc_bin_attach_tos (webrtc);
-
-  gst_webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
-      sctp_priority);
-}
-
-static void
-gst_webrtc_bin_attach_probe_to_ice_sink (GstWebRTCBin * webrtc,
-    GstWebRTCICETransport * transport)
-{
-  GstPad *pad;
-
-  pad = gst_element_get_static_pad (transport->sink, "sink");
-  gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
-      _nicesink_pad_probe, g_object_ref (webrtc),
-      (GDestroyNotify) gst_object_unref);
-  gst_object_unref (pad);
-}
-
-static void
-gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc)
-{
-  guint i;
-
-  if (webrtc->priv->tos_attached)
-    return;
-  webrtc->priv->tos_attached = TRUE;
-
-  for (i = 0; i < webrtc->priv->transports->len; i++) {
-    TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
-
-    gst_webrtc_bin_attach_tos_to_session (webrtc, stream->session_id);
-
-    gst_webrtc_bin_attach_probe_to_ice_sink (webrtc,
-        stream->transport->transport);
-    gst_webrtc_bin_attach_probe_to_ice_sink (webrtc,
-        stream->rtcp_transport->transport);
-  }
-
-  gst_webrtc_bin_update_sctp_priority (webrtc);
-}
-
 static WebRTCTransceiver *
 _create_webrtc_transceiver (GstWebRTCBin * webrtc,
     GstWebRTCRTPTransceiverDirection direction, guint mline)
@@ -1907,9 +1653,6 @@ _create_webrtc_transceiver (GstWebRTCBin * webrtc,
   /* FIXME: We don't support stopping transceiver yet so they're always not stopped */
   rtp_trans->stopped = FALSE;
 
-  g_signal_connect_object (sender, "notify::priority",
-      G_CALLBACK (gst_webrtc_bin_attach_tos), webrtc, G_CONNECT_SWAPPED);
-
   g_ptr_array_add (webrtc->priv->transceivers, trans);
 
   gst_object_unref (sender);
@@ -1938,8 +1681,6 @@ _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
       G_CALLBACK (_on_ice_transport_notify_gathering_state), webrtc);
   g_signal_connect (G_OBJECT (transport), "notify::state",
       G_CALLBACK (_on_dtls_transport_notify_state), webrtc);
-  if (webrtc->priv->tos_attached)
-    gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
 
   if ((transport = ret->rtcp_transport)) {
     g_signal_connect (G_OBJECT (transport->transport),
@@ -1949,8 +1690,6 @@ _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
         G_CALLBACK (_on_ice_transport_notify_gathering_state), webrtc);
     g_signal_connect (G_OBJECT (transport), "notify::state",
         G_CALLBACK (_on_dtls_transport_notify_state), webrtc);
-    if (webrtc->priv->tos_attached)
-      gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
   }
 
   GST_TRACE_OBJECT (webrtc,
@@ -1983,8 +1722,6 @@ _get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id)
     if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
             GST_ELEMENT (ret->send_bin), "rtcp_sink"))
       g_warn_if_reached ();
-    if (webrtc->priv->tos_attached)
-      gst_webrtc_bin_attach_tos_to_session (webrtc, ret->session_id);
     g_free (pad_name);
   }
 
@@ -2024,8 +1761,6 @@ _on_data_channel_ready_state (WebRTCDataChannel * channel,
 
     g_ptr_array_add (webrtc->priv->data_channels, channel);
 
-    gst_webrtc_bin_update_sctp_priority (webrtc);
-
     g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL], 0,
         gst_object_ref (channel));
   }
@@ -2276,7 +2011,6 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
     }
 
     webrtc->priv->sctp_transport = sctp_transport;
-    gst_webrtc_bin_update_sctp_priority (webrtc);
   }
 
   return webrtc->priv->data_channel_transport;
@@ -5565,7 +5299,6 @@ gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
     ret = gst_object_ref (ret);
     ret->webrtcbin = webrtc;
     g_ptr_array_add (webrtc->priv->data_channels, ret);
-    gst_webrtc_bin_update_sctp_priority (webrtc);
     webrtc_data_channel_link_to_sctp (ret, webrtc->priv->sctp_transport);
     if (webrtc->priv->sctp_transport &&
         webrtc->priv->sctp_transport->association_established
diff --git a/ext/webrtc/gstwebrtcbin.h b/ext/webrtc/gstwebrtcbin.h
index 4200b1ad77..e4b462f2e8 100644
--- a/ext/webrtc/gstwebrtcbin.h
+++ b/ext/webrtc/gstwebrtcbin.h
@@ -141,8 +141,6 @@ struct _GstWebRTCBinPrivate
   GstWebRTCSessionDescription *last_generated_answer;
 
   GstStructure *stats;
-
-  gboolean tos_attached;
 };
 
 typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);
diff --git a/ext/webrtc/gstwebrtcice.c b/ext/webrtc/gstwebrtcice.c
index 4a2ae33c57..e917d501a0 100644
--- a/ext/webrtc/gstwebrtcice.c
+++ b/ext/webrtc/gstwebrtcice.c
@@ -867,18 +867,6 @@ gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
   ice->priv->on_candidate_notify = notify;
 }
 
-void
-gst_webrtc_ice_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream,
-    guint tos)
-{
-  struct NiceStreamItem *item;
-
-  item = _find_item (ice, -1, -1, stream);
-  g_return_if_fail (item != NULL);
-
-  nice_agent_set_stream_tos (ice->priv->nice_agent, item->nice_stream_id, tos);
-}
-
 static void
 _clear_ice_stream (struct NiceStreamItem *item)
 {
diff --git a/ext/webrtc/gstwebrtcice.h b/ext/webrtc/gstwebrtcice.h
index 283672fe9e..d7c0965505 100644
--- a/ext/webrtc/gstwebrtcice.h
+++ b/ext/webrtc/gstwebrtcice.h
@@ -101,9 +101,6 @@ void                        gst_webrtc_ice_set_on_ice_candidate     (GstWebRTCIC
                                                                      gpointer user_data,
                                                                      GDestroyNotify notify);
 
-void                        gst_webrtc_ice_set_tos                  (GstWebRTCICE * ice,
-                                                                     GstWebRTCICEStream * stream,
-                                                                     guint tos);
 G_END_DECLS
 
 #endif /* __GST_WEBRTC_ICE_H__ */
diff --git a/ext/webrtc/meson.build b/ext/webrtc/meson.build
index 9c8af7dd44..49e8d4bb23 100644
--- a/ext/webrtc/meson.build
+++ b/ext/webrtc/meson.build
@@ -25,7 +25,7 @@ if libnice_dep.found()
     c_args : gst_plugins_bad_args + ['-DGST_USE_UNSTABLE_API'],
     include_directories : [configinc],
     dependencies : [gio_dep, libnice_dep, gstbase_dep, gstsdp_dep,
-                    gstapp_dep, gstwebrtc_dep, gstsctp_dep, gstrtp_dep],
+                    gstapp_dep, gstwebrtc_dep, gstsctp_dep],
     install : true,
     install_dir : plugins_install_dir,
   )
diff --git a/ext/webrtc/sctptransport.c b/ext/webrtc/sctptransport.c
index 169a49439c..f5a1e9db41 100644
--- a/ext/webrtc/sctptransport.c
+++ b/ext/webrtc/sctptransport.c
@@ -145,20 +145,6 @@ gst_webrtc_sctp_transport_set_property (GObject * object, guint prop_id,
   }
 }
 
-void
-gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * sctp,
-    GstWebRTCPriorityType priority)
-{
-  GstPad *pad;
-
-  pad = gst_element_get_static_pad (sctp->sctpenc, "src");
-  gst_pad_push_event (pad,
-      gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
-          gst_structure_new ("GstWebRtcBinUpdateTos", "sctp-priority",
-              GST_TYPE_WEBRTC_PRIORITY_TYPE, priority, NULL)));
-  gst_object_unref (pad);
-}
-
 static void
 gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
diff --git a/ext/webrtc/sctptransport.h b/ext/webrtc/sctptransport.h
index 8a1466c185..212f15eb9d 100644
--- a/ext/webrtc/sctptransport.h
+++ b/ext/webrtc/sctptransport.h
@@ -61,10 +61,6 @@ struct _GstWebRTCSCTPTransportClass
 
 GstWebRTCSCTPTransport *    gst_webrtc_sctp_transport_new               (void);
 
-void
-gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport *sctp,
-                                        GstWebRTCPriorityType priority);
-
 G_END_DECLS
 
 #endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
diff --git a/ext/webrtc/webrtctransceiver.c b/ext/webrtc/webrtctransceiver.c
index c42fd232ff..f26536741a 100644
--- a/ext/webrtc/webrtctransceiver.c
+++ b/ext/webrtc/webrtctransceiver.c
@@ -173,10 +173,6 @@ webrtc_transceiver_finalize (GObject * object)
 
   gst_caps_replace (&trans->last_configured_caps, NULL);
 
-  if (trans->ssrc_event)
-    gst_event_unref (trans->ssrc_event);
-  trans->ssrc_event = NULL;
-
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
diff --git a/ext/webrtc/webrtctransceiver.h b/ext/webrtc/webrtctransceiver.h
index 6d2d963eff..f834fd329a 100644
--- a/ext/webrtc/webrtctransceiver.h
+++ b/ext/webrtc/webrtctransceiver.h
@@ -40,7 +40,6 @@ struct _WebRTCTransceiver
   TransportStream          *stream;
   GstStructure             *local_rtx_ssrc_map;
   guint                     current_ssrc;
-  GstEvent                 *ssrc_event;
 
   /* Properties */
   GstWebRTCFECType         fec_type;