webrtc/talk/p2p/base/tcpport.h
henrike@webrtc.org 269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00

154 lines
5.5 KiB
C++

/*
* libjingle
* Copyright 2004--2005, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef WEBRTC_P2P_BASE_TCPPORT_H_
#define WEBRTC_P2P_BASE_TCPPORT_H_
#include <list>
#include <string>
#include "webrtc/p2p/base/port.h"
#include "webrtc/base/asyncpacketsocket.h"
namespace cricket {
class TCPConnection;
// Communicates using a local TCP port.
//
// This class is designed to allow subclasses to take advantage of the
// connection management provided by this class. A subclass should take of all
// packet sending and preparation, but when a packet is received, it should
// call this TCPPort::OnReadPacket (3 arg) to dispatch to a connection.
class TCPPort : public Port {
public:
static TCPPort* Create(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
int min_port, int max_port,
const std::string& username,
const std::string& password,
bool allow_listen) {
TCPPort* port = new TCPPort(thread, factory, network,
ip, min_port, max_port,
username, password, allow_listen);
if (!port->Init()) {
delete port;
port = NULL;
}
return port;
}
virtual ~TCPPort();
virtual Connection* CreateConnection(const Candidate& address,
CandidateOrigin origin);
virtual void PrepareAddress();
virtual int GetOption(rtc::Socket::Option opt, int* value);
virtual int SetOption(rtc::Socket::Option opt, int value);
virtual int GetError();
protected:
TCPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
rtc::Network* network, const rtc::IPAddress& ip,
int min_port, int max_port, const std::string& username,
const std::string& password, bool allow_listen);
bool Init();
// Handles sending using the local TCP socket.
virtual int SendTo(const void* data, size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload);
// Accepts incoming TCP connection.
void OnNewConnection(rtc::AsyncPacketSocket* socket,
rtc::AsyncPacketSocket* new_socket);
private:
struct Incoming {
rtc::SocketAddress addr;
rtc::AsyncPacketSocket* socket;
};
rtc::AsyncPacketSocket* GetIncoming(
const rtc::SocketAddress& addr, bool remove = false);
// Receives packet signal from the local TCP Socket.
void OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
void OnAddressReady(rtc::AsyncPacketSocket* socket,
const rtc::SocketAddress& address);
// TODO: Is this still needed?
bool incoming_only_;
bool allow_listen_;
rtc::AsyncPacketSocket* socket_;
int error_;
std::list<Incoming> incoming_;
friend class TCPConnection;
};
class TCPConnection : public Connection {
public:
// Connection is outgoing unless socket is specified
TCPConnection(TCPPort* port, const Candidate& candidate,
rtc::AsyncPacketSocket* socket = 0);
virtual ~TCPConnection();
virtual int Send(const void* data, size_t size,
const rtc::PacketOptions& options);
virtual int GetError();
rtc::AsyncPacketSocket* socket() { return socket_; }
private:
void OnConnect(rtc::AsyncPacketSocket* socket);
void OnClose(rtc::AsyncPacketSocket* socket, int error);
void OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
rtc::AsyncPacketSocket* socket_;
int error_;
friend class TCPPort;
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_TCPPORT_H_