Stop using TCP TURN servers as STUN servers.

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 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.
 
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/location.h"
@@ skipping 129 matching lines @@
     }
     for (size_t i = 0; i < turn_configurations.size(); ++i) {
       P2PPortAllocator::Config::RelayServerConfig relay_config;
       relay_config.server_address = turn_configurations[i].server.hostname();
       relay_config.port = turn_configurations[i].server.port();
       relay_config.username = turn_configurations[i].username;
       relay_config.password = turn_configurations[i].password;
       relay_config.transport_type = turn_configurations[i].transport_type;
       relay_config.secure = turn_configurations[i].secure;
       config.relays.push_back(relay_config);
-
-      // Use turn servers as stun servers.
-      config.stun_servers.insert(rtc::SocketAddress(
-          turn_configurations[i].server.hostname(),
-          turn_configurations[i].server.port()));
     }
     config.enable_multiple_routes = enable_multiple_routes_;
 
     return new P2PPortAllocator(
         socket_dispatcher_.get(), network_manager_,
         socket_factory_, config, origin_);
   }
 
  protected:
   ~P2PPortAllocatorFactory() override {}
@@ skipping 527 matching lines @@
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content