diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index f554ec2c15..0e315ac466 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -4476,6 +4476,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
       gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
       offer_caps = _rtp_caps_from_media (offer_media);
 
+      _remove_optional_offer_fields (offer_caps);
+
       if (last_answer && i < gst_sdp_message_medias_len (last_answer)
           && (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) {
         const GstSDPMedia *last_media =
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/utils.c b/subprojects/gst-plugins-bad/ext/webrtc/utils.c
index f0741d1e55..e3552b3814 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/utils.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/utils.c
@@ -170,6 +170,31 @@ _g_checksum_to_webrtc_string (GChecksumType type)
   }
 }
 
+void
+_remove_optional_offer_fields (GstCaps * offer_caps)
+{
+  int i;
+
+  for (i = 0; i < gst_caps_get_size (offer_caps); i++) {
+    GstStructure *s = gst_caps_get_structure (offer_caps, i);
+    const gchar *mtype = gst_structure_get_string (s, "media");
+    const gchar *encoding_name = gst_structure_get_string (s, "encoding-name");
+
+    if (mtype == NULL || encoding_name == NULL) {
+      continue;
+    }
+
+    /* Special cases for different codecs - sender-only fields
+     * that we don't need to care about for SDP intersection */
+    if (g_str_equal (mtype, "audio")) {
+      if (g_str_equal (encoding_name, "OPUS")) {
+        gst_structure_remove_fields (s, "sprop-stereo", "sprop-maxcapturerate",
+            NULL);
+      }
+    }
+  }
+}
+
 GstCaps *
 _rtp_caps_from_media (const GstSDPMedia * media)
 {
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/utils.h b/subprojects/gst-plugins-bad/ext/webrtc/utils.h
index e5d3d124ab..bcbe906a73 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/utils.h
+++ b/subprojects/gst-plugins-bad/ext/webrtc/utils.h
@@ -63,6 +63,8 @@ const gchar *                 _enum_value_to_string       (GType type, guint val
 G_GNUC_INTERNAL
 const gchar *           _g_checksum_to_webrtc_string (GChecksumType type);
 G_GNUC_INTERNAL
+void                    _remove_optional_offer_fields (GstCaps *offer_caps);
+G_GNUC_INTERNAL
 GstCaps *               _rtp_caps_from_media        (const GstSDPMedia * media);
 G_GNUC_INTERNAL
 GstWebRTCKind           webrtc_kind_from_caps       (const GstCaps * caps);