269fb4bc90
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
154 lines
5.5 KiB
C++
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_
|