Do not delete a connection in the turn port with permission error or refresh error.

webrtc/p2p/base/turnport.h

 /*
  *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 20 matching lines @@
 extern const char TURN_PORT_TYPE[];
 class TurnAllocateRequest;
 class TurnEntry;
 
 class TurnPort : public Port {
  public:
   enum PortState {
     STATE_CONNECTING,    // Initial state, cannot send any packets.
     STATE_CONNECTED,     // Socket connected, ready to send stun requests.
     STATE_READY,         // Received allocate success, can send any packets.
-    STATE_DISCONNECTED,  // TCP connection died, cannot send any packets.
+    STATE_RECEIVEONLY,   // Had REFRESH_REQUEST error, cannot send any packets.
+    STATE_DISCONNECTED,  // TCP connection died, cannot send/receive any
+                         // packets.
   };
   static TurnPort* Create(rtc::Thread* thread,
                           rtc::PacketSocketFactory* factory,
                           rtc::Network* network,
                           rtc::AsyncPacketSocket* socket,
                           const std::string& username,  // ice username.
                           const std::string& password,  // ice password.
                           const ProtocolAddress& server_address,
                           const RelayCredentials& credentials,
                           int server_priority,
@@ skipping 143 matching lines @@
   bool CreateTurnClientSocket();
 
   void set_nonce(const std::string& nonce) { nonce_ = nonce; }
   void set_realm(const std::string& realm) {
     if (realm != realm_) {
       realm_ = realm;
       UpdateHash();
     }
   }
 
-  void OnTurnRefreshError();
+  void OnRefreshError();
+  void HandleRefreshError();
   bool SetAlternateServer(const rtc::SocketAddress& address);
   void ResolveTurnAddress(const rtc::SocketAddress& address);
   void OnResolveResult(rtc::AsyncResolverInterface* resolver);
 
   void AddRequestAuthInfo(StunMessage* msg);
   void OnSendStunPacket(const void* data, size_t size, StunRequest* request);
   // Stun address from allocate success response.
   // Currently used only for testing.
   void OnStunAddress(const rtc::SocketAddress& address);
   void OnAllocateSuccess(const rtc::SocketAddress& address,
@@ skipping 22 matching lines @@
   TurnEntry* FindEntry(int channel_id) const;
   bool EntryExists(TurnEntry* e);
   void CreateOrRefreshEntry(const rtc::SocketAddress& address);
   void DestroyEntry(TurnEntry* entry);
   // Destroys the entry only if |timestamp| matches the destruction timestamp
   // in |entry|.
   void DestroyEntryIfNotCancelled(TurnEntry* entry, int64_t timestamp);
   void ScheduleEntryDestruction(TurnEntry* entry);
   void CancelEntryDestruction(TurnEntry* entry);
 
-  // Destroys the connection with remote address |address|. Returns true if
-  // a connection is found and destroyed.
-  bool DestroyConnection(const rtc::SocketAddress& address);
+  // Marks the connection with remote address |address| failed and
+  // pruned (a.k.a. write-timed-out). Returns true if a connection is found.
+  bool FailAndPruneConnection(const rtc::SocketAddress& address);
 
   ProtocolAddress server_address_;
   RelayCredentials credentials_;
   AttemptedServerSet attempted_server_addresses_;
 
   rtc::AsyncPacketSocket* socket_;
   SocketOptionsMap socket_options_;
   rtc::AsyncResolverInterface* resolver_;
   int error_;
 
@@ skipping 18 matching lines @@
   friend class TurnEntry;
   friend class TurnAllocateRequest;
   friend class TurnRefreshRequest;
   friend class TurnCreatePermissionRequest;
   friend class TurnChannelBindRequest;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_TURNPORT_H_