webrtc/talk/p2p/base/relayport.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

119 lines
4.5 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_RELAYPORT_H_
#define WEBRTC_P2P_BASE_RELAYPORT_H_
#include <deque>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/p2p/base/port.h"
#include "webrtc/p2p/base/stunrequest.h"
namespace cricket {
class RelayEntry;
class RelayConnection;
// Communicates using an allocated port on the relay server. For each
// remote candidate that we try to send data to a RelayEntry instance
// is created. The RelayEntry will try to reach the remote destination
// by connecting to all available server addresses in a pre defined
// order with a small delay in between. When a connection is
// successful all other connection attemts are aborted.
class RelayPort : public Port {
public:
typedef std::pair<rtc::Socket::Option, int> OptionValue;
// RelayPort doesn't yet do anything fancy in the ctor.
static RelayPort* 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) {
return new RelayPort(thread, factory, network, ip, min_port, max_port,
username, password);
}
virtual ~RelayPort();
void AddServerAddress(const ProtocolAddress& addr);
void AddExternalAddress(const ProtocolAddress& addr);
const std::vector<OptionValue>& options() const { return options_; }
bool HasMagicCookie(const char* data, size_t size);
virtual void PrepareAddress();
virtual Connection* CreateConnection(const Candidate& address,
CandidateOrigin origin);
virtual int SetOption(rtc::Socket::Option opt, int value);
virtual int GetOption(rtc::Socket::Option opt, int* value);
virtual int GetError();
const ProtocolAddress * ServerAddress(size_t index) const;
bool IsReady() { return ready_; }
// Used for testing.
sigslot::signal1<const ProtocolAddress*> SignalConnectFailure;
sigslot::signal1<const ProtocolAddress*> SignalSoftTimeout;
protected:
RelayPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
rtc::Network*, const rtc::IPAddress& ip,
int min_port, int max_port, const std::string& username,
const std::string& password);
bool Init();
void SetReady();
virtual int SendTo(const void* data, size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload);
// Dispatches the given packet to the port or connection as appropriate.
void OnReadPacket(const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
ProtocolType proto,
const rtc::PacketTime& packet_time);
private:
friend class RelayEntry;
std::deque<ProtocolAddress> server_addr_;
std::vector<ProtocolAddress> external_addr_;
bool ready_;
std::vector<RelayEntry*> entries_;
std::vector<OptionValue> options_;
int error_;
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_RELAYPORT_H_