Add TransportContext class.

remoting/host/cast_extension_session.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 "remoting/host/cast_extension_session.h"
 
 #include "base/bind.h"
 #include "base/json/json_reader.h"
 #include "base/json/json_writer.h"
 #include "base/logging.h"
@@ skipping 31 matching lines @@
 const char kWebRtcCandidate[] = "candidate";
 const char kWebRtcSessionDescType[] = "type";
 const char kWebRtcSessionDescSDP[] = "sdp";
 const char kWebRtcSDPMid[] = "sdpMid";
 const char kWebRtcSDPMLineIndex[] = "sdpMLineIndex";
 
 // Media labels used over the PeerConnection.
 const char kVideoLabel[] = "cast_video_label";
 const char kStreamLabel[] = "stream_label";
 
-// Default STUN server used to construct
-// ChromiumPortAllocator for the PeerConnection.
-const char kDefaultStunHost[] = "stun.l.google.com";
-const int kDefaultStunPort = 19302;
-
 const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread";
 
 // Interval between each call to PollPeerConnectionStats().
 const int kStatsLogIntervalSec = 10;
 
 // Minimum frame rate for video streaming over the PeerConnection in frames per
 // second, added as a media constraint when constructing the video source for
 // the Peer Connection.
 const int kMinFramesPerSecond = 5;
 
@@ skipping 417 matching lines @@
 
   // DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the
   // peer connection uses SDES. Disabling SDES as well will cause the peer
   // connection to fail to connect.
   // Note: For protection and unprotection of SRTP packets, the libjingle
   // ENABLE_EXTERNAL_AUTH flag must not be set.
   webrtc::FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            webrtc::MediaConstraintsInterface::kValueTrue);
 
-  rtc::scoped_ptr<protocol::ChromiumPortAllocator> port_allocator(
-      protocol::ChromiumPortAllocator::Create(url_request_context_getter_,
-                                              network_settings_)
-          .release());
-  std::vector<rtc::SocketAddress> stun_hosts;
-  stun_hosts.push_back(rtc::SocketAddress(kDefaultStunHost, kDefaultStunPort));
-  port_allocator->SetStunHosts(stun_hosts);
+  scoped_ptr<cricket::PortAllocator> port_allocator =
+      protocol::ChromiumPortAllocator::Create(url_request_context_getter_);
 
   webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
   peer_connection_ = peer_conn_factory_->CreatePeerConnection(
-      rtc_config, &constraints, port_allocator.Pass(), nullptr, this);
+      rtc_config, &constraints,
+      rtc::scoped_ptr<cricket::PortAllocator>(port_allocator.release()),
+      nullptr, this);
 
   if (!peer_connection_.get()) {
     CleanupPeerConnection();
     return false;
   }
 
   VLOG(1) << "Created PeerConnection successfully.";
 
   create_session_desc_observer_ =
       CastCreateSessionDescriptionObserver::Create(this);
@@ skipping 137 matching lines @@
   json.SetString(kWebRtcCandidate, candidate_str);
   std::string json_str;
   if (!base::JSONWriter::Write(json, &json_str)) {
     LOG(ERROR) << "Failed to serialize candidate message.";
     return;
   }
   SendMessageToClient(kSubjectNewCandidate, json_str);
 }
 
 }  // namespace remoting