diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index da67c093ec..3fb9cfea72 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -118,6 +118,10 @@ config("common_config") {
     defines += [ "RTC_DISABLE_VP9" ]
   }
 
+  if (rtc_enable_sctp) {
+    defines += [ "HAVE_SCTP" ]
+  }
+
   if (build_with_chromium) {
     defines += [
       # NOTICE: Since common_inherited_config is used in public_configs for our
@@ -126,7 +130,6 @@ config("common_config") {
       # source when webrtc:4256 is completed.
       "ENABLE_EXTERNAL_AUTH",
       "HAVE_OPENSSL_SSL_H",
-      "HAVE_SCTP",
       "HAVE_SRTP",
       "HAVE_WEBRTC_VIDEO",
       "HAVE_WEBRTC_VOICE",
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 8eeccb3a05..45fcd91508 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -262,7 +262,9 @@ if (rtc_include_tests) {
       "webrtcsession_unittest.cc",
     ]
 
-    defines = [ "HAVE_SCTP" ]
+    if (rtc_enable_sctp) {
+      defines = [ "HAVE_SCTP" ]
+    }
 
     configs += [ ":peerconnection_unittests_config" ]
 
diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc
index b213ecb00f..4f34cfa83e 100644
--- a/webrtc/api/peerconnection_unittest.cc
+++ b/webrtc/api/peerconnection_unittest.cc
@@ -1789,6 +1789,7 @@ TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
   VerifyRenderedAspectRatio(640, 480);
 }
 
+#ifdef HAVE_SCTP
 // This test verifies that the negotiation will succeed with data channel only
 // in max-bundle mode.
 TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
@@ -1799,6 +1800,7 @@ TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
   initializing_client()->CreateDataChannel();
   initializing_client()->Negotiate();
 }
+#endif
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive video.
@@ -2088,6 +2090,7 @@ TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) {
   EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
 }
 
+#ifdef HAVE_SCTP
 // This test sets up a call between two parties with audio, video and an SCTP
 // data channel.
 TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
@@ -2175,6 +2178,7 @@ TEST_F(P2PTestConductor, UnorderedSctpDataChannel) {
                    kMaxWaitMs);
   EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
 }
+#endif  // HAVE_SCTP
 
 // This test sets up a call between two parties and creates a data channel.
 // The test tests that received data is buffered unless an observer has been
diff --git a/webrtc/api/peerconnectionendtoend_unittest.cc b/webrtc/api/peerconnectionendtoend_unittest.cc
index fedadd9e12..4110db07c9 100644
--- a/webrtc/api/peerconnectionendtoend_unittest.cc
+++ b/webrtc/api/peerconnectionendtoend_unittest.cc
@@ -194,6 +194,7 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
 }
 #endif  // !defined(ADDRESS_SANITIZER)
 
+#ifdef HAVE_SCTP
 // Verifies that a DataChannel created before the negotiation can transition to
 // "OPEN" and transfer data.
 TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
@@ -319,6 +320,7 @@ TEST_F(PeerConnectionEndToEndTest,
   EXPECT_EQ(1U, dc_1_observer->received_message_count());
   EXPECT_EQ(1U, dc_2_observer->received_message_count());
 }
+#endif  // HAVE_SCTP
 
 #ifdef HAVE_QUIC
 // Test that QUIC data channels can be used and that messages go to the correct
@@ -396,6 +398,7 @@ TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
 }
 #endif  // HAVE_QUIC
 
+#ifdef HAVE_SCTP
 // Verifies that a DataChannel added from an OPEN message functions after
 // a channel has been previously closed (webrtc issue 3778).
 // This previously failed because the new channel re-uses the ID of the closed
@@ -455,3 +458,4 @@ TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
   // close message and be destroyed.
   rtc::Thread::Current()->ProcessMessages(100);
 }
+#endif  // HAVE_SCTP
diff --git a/webrtc/api/rtcstats_integrationtest.cc b/webrtc/api/rtcstats_integrationtest.cc
index c28130fe8a..d78ba3529f 100644
--- a/webrtc/api/rtcstats_integrationtest.cc
+++ b/webrtc/api/rtcstats_integrationtest.cc
@@ -523,6 +523,7 @@ class RTCStatsReportVerifier {
   rtc::scoped_refptr<const RTCStatsReport> report_;
 };
 
+#ifdef HAVE_SCTP
 TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) {
   StartCall();
 
@@ -549,6 +550,7 @@ TEST_F(RTCStatsIntegrationTest, GetsStatsWhileDestroyingPeerConnections) {
   // the peer connection.
   EXPECT_TRUE(stats_obtainer->report());
 }
+#endif  // HAVE_SCTP
 
 }  // namespace
 
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 17405c3cb3..1f97724380 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -47,6 +47,9 @@ declare_args() {
   # Set this to true to enable BWE test logging.
   rtc_enable_bwe_test_logging = false
 
+  # Set this to disable building with support for SCTP data channels.
+  rtc_enable_sctp = true
+
   # Disable these to not build components which can be externally provided.
   rtc_build_expat = true
   rtc_build_json = true
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index 4ed14138b6..04936de177 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -143,10 +143,15 @@ rtc_static_library("rtc_media") {
     "engine/webrtcvoe.h",
     "engine/webrtcvoiceengine.cc",
     "engine/webrtcvoiceengine.h",
-    "sctp/sctpdataengine.cc",
-    "sctp/sctpdataengine.h",
   ]
 
+  if (rtc_enable_sctp) {
+    sources += [
+      "sctp/sctpdataengine.cc",
+      "sctp/sctpdataengine.h",
+    ]
+  }
+
   configs += [ ":rtc_media_warnings_config" ]
 
   if (!build_with_chromium && is_clang) {
@@ -179,7 +184,7 @@ rtc_static_library("rtc_media") {
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
-  if (rtc_build_usrsctp) {
+  if (rtc_enable_sctp && rtc_build_usrsctp) {
     include_dirs += [
       # TODO(jiayl): move this into the public_configs of
       # //third_party/usrsctp/BUILD.gn.
@@ -338,9 +343,12 @@ if (rtc_include_tests) {
       "engine/webrtcvideocapturer_unittest.cc",
       "engine/webrtcvideoengine2_unittest.cc",
       "engine/webrtcvoiceengine_unittest.cc",
-      "sctp/sctpdataengine_unittest.cc",
     ]
 
+    if (rtc_enable_sctp) {
+      sources += [ "sctp/sctpdataengine_unittest.cc" ]
+    }
+
     configs += [ ":rtc_media_unittests_config" ]
 
     if (rtc_use_h264) {
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index fe8711fa26..f2bdbc3568 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -15,10 +15,10 @@ group("pc") {
 }
 
 config("rtc_pc_config") {
-  defines = [
-    "HAVE_SCTP",
-    "HAVE_SRTP",
-  ]
+  defines = [ "HAVE_SRTP" ]
+  if (rtc_enable_sctp) {
+    defines += [ "HAVE_SCTP" ]
+  }
 }
 
 rtc_static_library("rtc_pc") {