Remove enable_dtls_srtp option

This is part of the removal of support for SDES.

Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}

api/peer_connection_interface.h

@@ -427,12 +427,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // Use new combined audio/video bandwidth estimation?
     absl::optional<bool> combined_audio_video_bwe;
 
-    // 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;
-
     /////////////////////////////////////////////////
     // The below fields are not part of the standard.
     /////////////////////////////////////////////////

examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java

@@ -442,6 +442,10 @@ private void createPeerConnectionFactoryInternal(PeerConnectionFactory.Options o
       decoderFactory = new SoftwareVideoDecoderFactory();
     }
 
+    // Disable encryption for loopback calls.
+    if (peerConnectionParameters.loopback) {
+      options.disableEncryption = true;
+    }
     factory = PeerConnectionFactory.builder()
                   .setOptions(options)
                   .setAudioDeviceModule(adm)
@@ -600,8 +604,6 @@ private void createPeerConnectionInternal() {
     rtcConfig.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY;
     // Use ECDSA encryption.
     rtcConfig.keyType = PeerConnection.KeyType.ECDSA;
-    // Enable DTLS for normal calls and disable for loopback calls.
-    rtcConfig.enableDtlsSrtp = !peerConnectionParameters.loopback;
     rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN;
 
     peerConnection = factory.createPeerConnection(rtcConfig, pcObserver);

examples/androidnativeapi/jni/android_call_client.cc

@@ -177,8 +177,10 @@ void AndroidCallClient::CreatePeerConnection() {
   webrtc::MutexLock lock(&pc_mutex_);
   webrtc::PeerConnectionInterface::RTCConfiguration config;
   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  // DTLS SRTP has to be disabled for loopback to work.
-  config.enable_dtls_srtp = false;
+  // Encryption has to be disabled for loopback to work.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  pcf_->SetOptions(options);
   webrtc::PeerConnectionDependencies deps(pc_observer_.get());
   pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(deps)).MoveValue();
 

examples/objcnativeapi/objc/objc_call_client.mm

@@ -141,8 +141,10 @@
   webrtc::MutexLock lock(&pc_mutex_);
   webrtc::PeerConnectionInterface::RTCConfiguration config;
   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  // DTLS SRTP has to be disabled for loopback to work.
-  config.enable_dtls_srtp = false;
+  // Encryption has to be disabled for loopback to work.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  pcf_->SetOptions(options);
   webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get());
   pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)).MoveValue();
   RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_;

examples/peerconnection/client/conductor.cc

@@ -150,7 +150,7 @@ bool Conductor::InitializePeerConnection() {
     return false;
   }
 
-  if (!CreatePeerConnection(/*dtls=*/true)) {
+  if (!CreatePeerConnection()) {
     main_wnd_->MessageBox("Error", "CreatePeerConnection failed", true);
     DeletePeerConnection();
   }
@@ -165,23 +165,28 @@ bool Conductor::ReinitializePeerConnectionForLoopback() {
   std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> senders =
       peer_connection_->GetSenders();
   peer_connection_ = nullptr;
-  if (CreatePeerConnection(/*dtls=*/false)) {
+  // Loopback is only possible if encryption is disabled.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  peer_connection_factory_->SetOptions(options);
+  if (CreatePeerConnection()) {
     for (const auto& sender : senders) {
       peer_connection_->AddTrack(sender->track(), sender->stream_ids());
     }
     peer_connection_->CreateOffer(
         this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
   }
+  options.disable_encryption = false;
+  peer_connection_factory_->SetOptions(options);
   return peer_connection_ != nullptr;
 }
 
-bool Conductor::CreatePeerConnection(bool dtls) {
+bool Conductor::CreatePeerConnection() {
   RTC_DCHECK(peer_connection_factory_);
   RTC_DCHECK(!peer_connection_);
 
   webrtc::PeerConnectionInterface::RTCConfiguration config;
   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  config.enable_dtls_srtp = dtls;
   webrtc::PeerConnectionInterface::IceServer server;
   server.uri = GetPeerConnectionString();
   config.servers.push_back(server);

examples/peerconnection/client/conductor.h

@@ -54,7 +54,7 @@ class Conductor : public webrtc::PeerConnectionObserver,
   ~Conductor();
   bool InitializePeerConnection();
   bool ReinitializePeerConnectionForLoopback();
-  bool CreatePeerConnection(bool dtls);
+  bool CreatePeerConnection();
   void DeletePeerConnection();
   void EnsureStreamingUI();
   void AddTracks();

examples/unityplugin/simple_peer_connection.cc

@@ -190,7 +190,6 @@ bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
   webrtc::PeerConnectionInterface::IceServer stun_server;
   stun_server.uri = GetPeerConnectionString();
   config_.servers.push_back(stun_server);
-  config_.enable_dtls_srtp = false;
 
   auto result = g_peer_connection_factory->CreatePeerConnectionOrError(
       config_, webrtc::PeerConnectionDependencies(this));

pc/peer_connection.cc

@@ -276,8 +276,8 @@ bool DtlsEnabled(const PeerConnectionInterface::RTCConfiguration& configuration,
   bool default_enabled =
       (dependencies.cert_generator || !configuration.certificates.empty());
 
-  // The `configuration` can override the default value.
-  return configuration.enable_dtls_srtp.value_or(default_enabled);
+  RTC_DCHECK(default_enabled) << "Configuration error: No certs for DTLS";
+  return default_enabled;
 }
 
 }  // namespace
@@ -300,7 +300,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
     bool enable_rtp_data_channel;
     absl::optional<int> screencast_min_bitrate;
     absl::optional<bool> combined_audio_video_bwe;
-    absl::optional<bool> enable_dtls_srtp;
     TcpCandidatePolicy tcp_candidate_policy;
     CandidateNetworkPolicy candidate_network_policy;
     int audio_jitter_buffer_max_packets;
@@ -368,7 +367,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
          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 &&
-         enable_dtls_srtp == o.enable_dtls_srtp &&
          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 &&