webrtc/talk/p2p/base/portproxy.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

105 lines
3.8 KiB
C++

/*
* libjingle
* Copyright 2004--2011, 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_PORTPROXY_H_
#define WEBRTC_P2P_BASE_PORTPROXY_H_
#include "webrtc/p2p/base/portinterface.h"
#include "webrtc/base/sigslot.h"
namespace rtc {
class Network;
}
namespace cricket {
class PortProxy : public PortInterface, public sigslot::has_slots<> {
public:
PortProxy() {}
virtual ~PortProxy() {}
PortInterface* impl() { return impl_; }
void set_impl(PortInterface* port);
virtual const std::string& Type() const;
virtual rtc::Network* Network() const;
virtual void SetIceProtocolType(IceProtocolType protocol);
virtual IceProtocolType IceProtocol() const;
// Methods to set/get ICE role and tiebreaker values.
virtual void SetIceRole(IceRole role);
virtual IceRole GetIceRole() const;
virtual void SetIceTiebreaker(uint64 tiebreaker);
virtual uint64 IceTiebreaker() const;
virtual bool SharedSocket() const;
// Forwards call to the actual Port.
virtual void PrepareAddress();
virtual Connection* CreateConnection(const Candidate& remote_candidate,
CandidateOrigin origin);
virtual Connection* GetConnection(
const rtc::SocketAddress& remote_addr);
virtual int SendTo(const void* data, size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload);
virtual int SetOption(rtc::Socket::Option opt, int value);
virtual int GetOption(rtc::Socket::Option opt, int* value);
virtual int GetError();
virtual const std::vector<Candidate>& Candidates() const;
virtual void SendBindingResponse(StunMessage* request,
const rtc::SocketAddress& addr);
virtual void SendBindingErrorResponse(
StunMessage* request, const rtc::SocketAddress& addr,
int error_code, const std::string& reason);
virtual void EnablePortPackets();
virtual std::string ToString() const;
private:
void OnUnknownAddress(PortInterface *port,
const rtc::SocketAddress &addr,
ProtocolType proto,
IceMessage *stun_msg,
const std::string &remote_username,
bool port_muxed);
void OnRoleConflict(PortInterface* port);
void OnPortDestroyed(PortInterface* port);
PortInterface* impl_;
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_PORTPROXY_H_