diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index 56afd5c27cc..74f0e00dbc8 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -353,6 +353,14 @@ rtc_server {
     # @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
     # default: off
     merge_nalus off;
+    # For WebRTC over TCP directly, not TURN, see https://github.com/ossrs/srs/issues/2852
+    # Some network does not support UDP, or not very well, so we use TCP like HTTP/80 port for firewall traversing.
+    tcp {
+        # The TCP listen port for WebRTC. Highly recommend is some normally used ports, such as TCP/80, TCP/443,
+        # TCP/8000, TCP/8080 etc. However SRS default to TCP/8000 corresponding to UDP/8000.
+        # Default: 8000
+        listen 8000;
+    }
     # The black-hole to copy packet to, for debugging.
     # For example, when debugging Chrome publish stream, the received packets are encrypted cipher,
     # we can set the publisher black-hole, SRS will copy the plaintext packets to black-hole, and