Do not delete the turn port entry right away when the respective connection is deleted.

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.
  */
 
 #ifndef WEBRTC_P2P_BASE_TURNPORT_H_
 #define WEBRTC_P2P_BASE_TURNPORT_H_
 
 #include <stdio.h>
 #include <list>
 #include <set>
 #include <string>
 
+#include "webrtc/base/asyncinvoker.h"
+#include "webrtc/base/asyncpacketsocket.h"
 #include "webrtc/p2p/base/port.h"
 #include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/base/asyncpacketsocket.h"
 
 namespace rtc {
 class AsyncResolver;
 class SignalThread;
 }
 
 namespace cricket {
 
 extern const char TURN_PORT_TYPE[];
 class TurnAllocateRequest;
@@ skipping 83 matching lines @@
   const std::string& nonce() const { return nonce_; }
 
   int error() const { return error_; }
 
   void OnAllocateMismatch();
 
   rtc::AsyncPacketSocket* socket() const {
     return socket_;
   }
 
+  // For testing only.
+  rtc::AsyncInvoker* invoker() { return &invoker_; }
+
   // Signal with resolved server address.
   // Parameters are port, server address and resolved server address.
   // This signal will be sent only if server address is resolved successfully.
   sigslot::signal3<TurnPort*,
                    const rtc::SocketAddress&,
                    const rtc::SocketAddress&> SignalResolvedServerAddress;
 
   // This signal is only for testing purpose.
   sigslot::signal3<TurnPort*, const rtc::SocketAddress&, int>
       SignalCreatePermissionResult;
@@ skipping 73 matching lines @@
   bool ScheduleRefresh(int lifetime);
   void SendRequest(StunRequest* request, int delay);
   int Send(const void* data, size_t size,
            const rtc::PacketOptions& options);
   void UpdateHash();
   bool UpdateNonce(StunMessage* response);
 
   bool HasPermission(const rtc::IPAddress& ipaddr) const;
   TurnEntry* FindEntry(const rtc::SocketAddress& address) const;
   TurnEntry* FindEntry(int channel_id) const;
-  TurnEntry* CreateEntry(const rtc::SocketAddress& address);
-  void DestroyEntry(const rtc::SocketAddress& address);
+  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, uint32_t timestamp);
+  void ScheduleEntryDestruction(TurnEntry* entry);
+  void CancelEntryDestruction(TurnEntry* entry);
   void OnConnectionDestroyed(Connection* conn);
 
   ProtocolAddress server_address_;
   RelayCredentials credentials_;
   AttemptedServerSet attempted_server_addresses_;
 
   rtc::AsyncPacketSocket* socket_;
   SocketOptionsMap socket_options_;
   rtc::AsyncResolverInterface* resolver_;
   int error_;
 
   StunRequestManager request_manager_;
   std::string realm_;       // From 401/438 response message.
   std::string nonce_;       // From 401/438 response message.
   std::string hash_;        // Digest of username:realm:password
 
   int next_channel_number_;
   EntryList entries_;
 
   PortState state_;
   // By default the value will be set to 0. This value will be used in
   // calculating the candidate priority.
   int server_priority_;
 
   // The number of retries made due to allocate mismatch error.
   size_t allocate_mismatch_retries_;
 
+  rtc::AsyncInvoker invoker_;
+
   friend class TurnEntry;
   friend class TurnAllocateRequest;
   friend class TurnRefreshRequest;
   friend class TurnCreatePermissionRequest;
   friend class TurnChannelBindRequest;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_TURNPORT_H_