Enable TURN on the host when using WebRTC.

remoting/host/remoting_me2me_host.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 //
 // This file implements a standalone host process for Me2Me.
 
 #include <stddef.h>
 #include <stdint.h>
 
 #include <string>
@@ skipping 1492 matching lines @@
   if (!udp_port_range_.is_null()) {
     network_settings.port_range = udp_port_range_;
   } else if (!allow_nat_traversal_) {
     // For legacy reasons we have to restrict the port range to a set of default
     // values when nat traversal is disabled, even if the port range was not
     // set in policy.
     network_settings.port_range.min_port = NetworkSettings::kDefaultMinPort;
     network_settings.port_range.max_port = NetworkSettings::kDefaultMaxPort;
   }
 
-  scoped_refptr<protocol::TransportContext> transport_context =
+  scoped_refptr<protocol::TransportContext> ice_transport_context =
       new protocol::TransportContext(
           signal_strategy_.get(),
           make_scoped_ptr(new protocol::ChromiumPortAllocatorFactory()),
           make_scoped_ptr(new ChromiumUrlRequestFactory(
               context_->url_request_context_getter())),
           network_settings, protocol::TransportRole::SERVER);
 
+  scoped_refptr<protocol::TransportContext> webrtc_transport_context;
+
   scoped_ptr<protocol::SessionManager> session_manager(
       new protocol::JingleSessionManager(signal_strategy_.get()));
 
   scoped_ptr<protocol::CandidateSessionConfig> protocol_config =
       protocol::CandidateSessionConfig::CreateDefault();
   if (!desktop_environment_factory_->SupportsAudioCapture())
     protocol_config->DisableAudioChannel();
   if (enable_vp9_)
     protocol_config->set_vp9_experiment_enabled(true);
   if (base::CommandLine::ForCurrentProcess()->HasSwitch(
           kEnableWebrtcSwitchName)) {
     protocol_config->set_webrtc_supported(true);
+
+    webrtc_transport_context = new protocol::TransportContext(
+        signal_strategy_.get(),
+        make_scoped_ptr(new protocol::ChromiumPortAllocatorFactory()),
+        make_scoped_ptr(new ChromiumUrlRequestFactory(
+            context_->url_request_context_getter())),
+        network_settings, protocol::TransportRole::SERVER);
+    webrtc_transport_context->UseTurn(
+        ServiceUrls::GetInstance()->ice_config_url());
   }
   session_manager->set_protocol_config(std::move(protocol_config));
 
-  host_.reset(new ChromotingHost(desktop_environment_factory_.get(),
-                                 std::move(session_manager), transport_context,
-                                 context_->audio_task_runner(),
-                                 context_->video_encode_task_runner()));
+  host_.reset(new ChromotingHost(
+      desktop_environment_factory_.get(), std::move(session_manager),
+      ice_transport_context, webrtc_transport_context,
+      context_->audio_task_runner(), context_->video_encode_task_runner()));
 
   if (gnubby_auth_policy_enabled_ && gnubby_extension_supported_) {
     host_->AddExtension(make_scoped_ptr(new GnubbyExtension()));
   }
 
   if (frame_recorder_buffer_size_ > 0) {
     scoped_ptr<VideoFrameRecorderHostExtension> frame_recorder_extension(
         new VideoFrameRecorderHostExtension());
     frame_recorder_extension->SetMaxContentBytes(frame_recorder_buffer_size_);
     host_->AddExtension(std::move(frame_recorder_extension));
@@ skipping 193 matching lines @@
       base::TimeDelta::FromSeconds(kShutdownTimeoutSeconds));
   new HostProcess(std::move(context), &exit_code, &shutdown_watchdog);
 
   // Run the main (also UI) message loop until the host no longer needs it.
   message_loop.Run();
 
   return exit_code;
 }
 
 }  // namespace remoting