diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 3fb9cfea72..ba58565dc8 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -122,13 +122,16 @@ config("common_config") {
     defines += [ "HAVE_SCTP" ]
   }
 
+  if (rtc_enable_external_auth) {
+    defines += [ "ENABLE_EXTERNAL_AUTH" ]
+  }
+
   if (build_with_chromium) {
     defines += [
       # NOTICE: Since common_inherited_config is used in public_configs for our
       # targets, there's no point including the defines in that config here.
       # TODO(kjellander): Cleanup unused ones and move defines closer to the
       # source when webrtc:4256 is completed.
-      "ENABLE_EXTERNAL_AUTH",
       "HAVE_OPENSSL_SSL_H",
       "HAVE_SRTP",
       "HAVE_WEBRTC_VIDEO",
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 1f97724380..ce59871576 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -40,6 +40,10 @@ declare_args() {
   # Disable the code for the intelligibility enhancer by default.
   rtc_enable_intelligibility_enhancer = false
 
+  # Enable when an external authentication mechanism is used for performing
+  # packet authentication for RTP packets instead of libsrtp.
+  rtc_enable_external_auth = build_with_chromium
+
   # Selects whether debug dumps for the audio processing module
   # should be generated.
   apm_debug_dump = false