Create a new connection if a candidate reuses an address

webrtc/p2p/base/port.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 298 matching lines @@
                   const rtc::SocketAddress& base_address,
                   const rtc::SocketAddress& related_address,
                   const std::string& protocol,
                   const std::string& relay_protocol,
                   const std::string& tcptype,
                   const std::string& type,
                   uint32_t type_preference,
                   uint32_t relay_preference,
                   bool final);
 
-  // Adds the given connection to the list.  (Deleting removes them.)
-  void AddConnection(Connection* conn);
+  // Adds the given connection to the map keyed by the remote candidate address.
+  // If an existing connection has the same address, the existing one will be
+  // replaced and destroyed.
+  void AddOrReplaceConnection(Connection* conn);
 
   // Called when a packet is received from an unknown address that is not
   // currently a connection.  If this is an authenticated STUN binding request,
   // then we will signal the client.
   void OnReadPacket(const char* data, size_t size,
                     const rtc::SocketAddress& addr,
                     ProtocolType proto);
 
   // If the given data comprises a complete and correct STUN message then the
   // return value is true, otherwise false. If the message username corresponds
   // with this port's username fragment, msg will contain the parsed STUN
   // message.  Otherwise, the function may send a STUN response internally.
   // remote_username contains the remote fragment of the STUN username.
   bool GetStunMessage(const char* data,
                       size_t size,
                       const rtc::SocketAddress& addr,
                       std::unique_ptr<IceMessage>* out_msg,
                       std::string* out_username);
 
   // Checks if the address in addr is compatible with the port's ip.
   bool IsCompatibleAddress(const rtc::SocketAddress& addr);
 
   // Returns default DSCP value.
   rtc::DiffServCodePoint DefaultDscpValue() const {
     // No change from what MediaChannel set.
     return rtc::DSCP_NO_CHANGE;
   }
 
+  // Extra work to be done in subclasses when a connection is destroyed.
+  virtual void HandleConnectionDestroyed(Connection* conn) {}
+
  private:
   void Construct();
   // Called when one of our connections deletes itself.
   void OnConnectionDestroyed(Connection* conn);
 
   // Whether this port is dead, and hence, should be destroyed on the controlled
   // side.
   bool dead() const {
     return ice_role_ == ICEROLE_CONTROLLED && connections_.empty();
   }
@@ skipping 307 matching lines @@
            const rtc::PacketOptions& options) override;
   int GetError() override { return error_; }
 
  private:
   int error_ = 0;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_PORT_H_