Adding TLS support to the TCP Client sockets.

content/renderer/media/media_stream_dependency_factory.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.
 
 #include "content/renderer/media/media_stream_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 99 matching lines @@
       config.stun_server_port = stun_servers[0].server.port();
     }
     config.legacy_relay = false;
     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 first turn server as the stun server.
     if (turn_configurations.size() > 0) {
       config.stun_server = config.relays[0].server_address;
       config.stun_server_port = config.relays[0].port;
     }
 
     return new P2PPortAllocator(
@@ skipping 686 matching lines @@
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer.
   if (is_new_capturer)
     GetWebRtcAudioDevice()->AddAudioCapturer(capturer);
 
   return capturer;
 }
 
 }  // namespace content