Add config to prune TURN ports

webrtc/p2p/base/p2ptransportchannel.h

 /*
  *  Copyright 2004 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 250 matching lines @@
                                PortInterface* origin_port);
   bool IsPingable(const Connection* conn, int64_t now) const;
   bool IsSelectedConnectionPingable(int64_t now);
   int CalculateActiveWritablePingInterval(const Connection* conn,
                                           int64_t now) const;
   void PingConnection(Connection* conn);
   void AddAllocatorSession(std::unique_ptr<PortAllocatorSession> session);
   void AddConnection(Connection* connection);
 
   void OnPortReady(PortAllocatorSession *session, PortInterface* port);
+  void OnPortPruned(PortAllocatorSession* session, PortInterface* port);
+  // Returns true if the port is found and removed from |ports_|.
+  bool RemovePort(PortInterface* port);
   void OnCandidatesReady(PortAllocatorSession *session,
                          const std::vector<Candidate>& candidates);
   void OnCandidatesAllocationDone(PortAllocatorSession* session);
   void OnUnknownAddress(PortInterface* port,
                         const rtc::SocketAddress& addr,
                         ProtocolType proto,
                         IceMessage* stun_msg,
                         const std::string& remote_username,
                         bool port_muxed);
   void OnPortDestroyed(PortInterface* port);
@@ skipping 93 matching lines @@
   IceConfig config_;
   int last_sent_packet_id_ = -1;  // -1 indicates no packet was sent before.
   bool started_pinging_ = false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_