diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 2f8e401..a9026d9 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -457,11 +457,14 @@
     // Use new combined audio/video bandwidth estimation?
     absl::optional<bool> combined_audio_video_bwe;
 
+#if defined(WEBRTC_FUCHSIA)
+    // TODO(bugs.webrtc.org/11066): Remove entirely once Fuchsia does not use.
     // TODO(bugs.webrtc.org/9891) - Move to crypto_options
     // Can be used to disable DTLS-SRTP. This should never be done, but can be
     // useful for testing purposes, for example in setting up a loopback call
     // with a single PeerConnection.
     absl::optional<bool> enable_dtls_srtp;
+#endif
 
     /////////////////////////////////////////////////
     // The below fields are not part of the standard.
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index e5a83b0..1a17fd4 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -280,8 +280,12 @@
   bool default_enabled =
       (dependencies.cert_generator || !configuration.certificates.empty());
 
+#if defined(WEBRTC_FUCHSIA)
   // The `configuration` can override the default value.
   return configuration.enable_dtls_srtp.value_or(default_enabled);
+#else
+  return default_enabled;
+#endif
 }
 
 }  // namespace
@@ -303,7 +307,9 @@
     bool disable_link_local_networks;
     absl::optional<int> screencast_min_bitrate;
     absl::optional<bool> combined_audio_video_bwe;
+#if defined(WEBRTC_FUCHSIA)
     absl::optional<bool> enable_dtls_srtp;
+#endif
     TcpCandidatePolicy tcp_candidate_policy;
     CandidateNetworkPolicy candidate_network_policy;
     int audio_jitter_buffer_max_packets;
@@ -372,7 +378,9 @@
          disable_link_local_networks == o.disable_link_local_networks &&
          screencast_min_bitrate == o.screencast_min_bitrate &&
          combined_audio_video_bwe == o.combined_audio_video_bwe &&
+#if defined(WEBRTC_FUCHSIA)
          enable_dtls_srtp == o.enable_dtls_srtp &&
+#endif
          ice_candidate_pool_size == o.ice_candidate_pool_size &&
          prune_turn_ports == o.prune_turn_ports &&
          turn_port_prune_policy == o.turn_port_prune_policy &&
diff --git a/pc/peer_connection_crypto_unittest.cc b/pc/peer_connection_crypto_unittest.cc
index 1741b99..9c9e9b6 100644
--- a/pc/peer_connection_crypto_unittest.cc
+++ b/pc/peer_connection_crypto_unittest.cc
@@ -211,7 +211,6 @@
 // no SDES cryptos.
 TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsEnabled) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
 
   auto offer = caller->CreateOffer();
@@ -225,7 +224,6 @@
 }
 TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
   auto callee = CreatePeerConnectionWithAudioVideo(config);
 
@@ -240,6 +238,7 @@
                              answer->description()));
 }
 
+#if defined(WEBRTC_FUCHSIA)
 // When DTLS is disabled, the SDP offer/answer should include SDES cryptos and
 // should not have a DTLS fingerprint.
 TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsDisabled) {
@@ -256,6 +255,7 @@
   EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolSavpf),
                              offer->description()));
 }
+
 TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsDisabled) {
   RTCConfiguration config;
   config.enable_dtls_srtp.emplace(false);
@@ -293,6 +293,7 @@
   EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolAvpf),
                              offer->description()));
 }
+
 TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
   PeerConnectionFactoryInterface::Options options;
   options.disable_encryption = true;
@@ -462,13 +463,13 @@
 
   EXPECT_FALSE(caller->SetRemoteDescription(std::move(answer)));
 }
+#endif
 
 // The following group tests that two PeerConnections can successfully exchange
 // an offer/answer when DTLS is on and that they will refuse any offer/answer
 // applied locally/remotely if it does not include a DTLS fingerprint.
 TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenDtlsOn) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
   auto callee = CreatePeerConnectionWithAudioVideo(config);
 
@@ -483,7 +484,6 @@
 TEST_P(PeerConnectionCryptoTest,
        FailToSetLocalOfferWithNoFingerprintWhenDtlsOn) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
 
   auto offer = caller->CreateOffer();
@@ -494,7 +494,6 @@
 TEST_P(PeerConnectionCryptoTest,
        FailToSetRemoteOfferWithNoFingerprintWhenDtlsOn) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
   auto callee = CreatePeerConnectionWithAudioVideo(config);
 
@@ -506,7 +505,6 @@
 TEST_P(PeerConnectionCryptoTest,
        FailToSetLocalAnswerWithNoFingerprintWhenDtlsOn) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
   auto callee = CreatePeerConnectionWithAudioVideo(config);
 
@@ -517,7 +515,6 @@
 TEST_P(PeerConnectionCryptoTest,
        FailToSetRemoteAnswerWithNoFingerprintWhenDtlsOn) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto caller = CreatePeerConnectionWithAudioVideo(config);
   auto callee = CreatePeerConnectionWithAudioVideo(config);
 
@@ -528,6 +525,7 @@
   EXPECT_FALSE(caller->SetRemoteDescription(std::move(answer)));
 }
 
+#if defined(WEBRTC_FUCHSIA)
 // Test that an offer/answer can be exchanged when encryption is disabled.
 TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenNoEncryption) {
   PeerConnectionFactoryInterface::Options options;
@@ -547,19 +545,18 @@
   ASSERT_TRUE(answer);
   ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
 }
+#endif
 
 // Tests that a DTLS call can be established when the certificate is specified
 // in the PeerConnection config and no certificate generator is specified.
 TEST_P(PeerConnectionCryptoTest,
        ExchangeOfferAnswerWhenDtlsCertificateInConfig) {
   RTCConfiguration caller_config;
-  caller_config.enable_dtls_srtp.emplace(true);
   caller_config.certificates.push_back(
       FakeRTCCertificateGenerator::GenerateCertificate());
   auto caller = CreatePeerConnectionWithAudioVideo(caller_config);
 
   RTCConfiguration callee_config;
-  callee_config.enable_dtls_srtp.emplace(true);
   callee_config.certificates.push_back(
       FakeRTCCertificateGenerator::GenerateCertificate());
   auto callee = CreatePeerConnectionWithAudioVideo(callee_config);
@@ -630,7 +627,6 @@
 
 TEST_P(PeerConnectionCryptoDtlsCertGenTest, TestCertificateGeneration) {
   RTCConfiguration config;
-  config.enable_dtls_srtp.emplace(true);
   auto owned_fake_certificate_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   auto* fake_certificate_generator = owned_fake_certificate_generator.get();
@@ -754,7 +750,6 @@
 
   auto caller = CreatePeerConnectionWithAudioVideo();
   RTCConfiguration callee_config;
-  callee_config.enable_dtls_srtp.emplace(true);
   callee_config.certificates.push_back(callee_certificate);
   auto callee = CreatePeerConnectionWithAudioVideo(callee_config);
 
diff --git a/pc/peer_connection_end_to_end_unittest.cc b/pc/peer_connection_end_to_end_unittest.cc
index 78dcda3..6498be5 100644
--- a/pc/peer_connection_end_to_end_unittest.cc
+++ b/pc/peer_connection_end_to_end_unittest.cc
@@ -396,6 +396,7 @@
   WaitForCallEstablished();
 }
 
+#if defined(IS_FUCHSIA)
 TEST_P(PeerConnectionEndToEndTest, CallWithSdesKeyNegotiation) {
   config_.enable_dtls_srtp = false;
   CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
@@ -404,6 +405,7 @@
   Negotiate();
   WaitForCallEstablished();
 }
+#endif
 
 TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
   class IdLoggingAudioEncoderFactory : public webrtc::AudioEncoderFactory {
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index 1e17fa1..f2e4b1d 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -269,6 +269,7 @@
                                     webrtc::kEnumCounterKeyProtocolSdes));
 }
 
+#if defined(WEBRTC_FUCHSIA)
 // Uses SDES instead of DTLS for key agreement.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
   PeerConnectionInterface::RTCConfiguration sdes_config;
@@ -292,6 +293,7 @@
       0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                     webrtc::kEnumCounterKeyProtocolDtls));
 }
+#endif
 
 // Basic end-to-end test specifying the `enable_encrypted_rtp_header_extensions`
 // option to offer encrypted versions of all header extensions alongside the
diff --git a/rtc_tools/data_channel_benchmark/peer_connection_client.cc b/rtc_tools/data_channel_benchmark/peer_connection_client.cc
index 6d2ee81..cd02e71 100644
--- a/rtc_tools/data_channel_benchmark/peer_connection_client.cc
+++ b/rtc_tools/data_channel_benchmark/peer_connection_client.cc
@@ -159,7 +159,6 @@
   server.urls.push_back(kStunServer);
   config.servers.push_back(server);
   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  config.enable_dtls_srtp = true;
 
   webrtc::PeerConnectionDependencies dependencies(this);
   auto result =