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
105 lines
3.8 KiB
C++
105 lines
3.8 KiB
C++
/*
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* libjingle
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* Copyright 2004--2011, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef WEBRTC_P2P_BASE_PORTPROXY_H_
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#define WEBRTC_P2P_BASE_PORTPROXY_H_
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#include "webrtc/p2p/base/portinterface.h"
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#include "webrtc/base/sigslot.h"
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namespace rtc {
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class Network;
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}
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namespace cricket {
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class PortProxy : public PortInterface, public sigslot::has_slots<> {
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public:
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PortProxy() {}
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virtual ~PortProxy() {}
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PortInterface* impl() { return impl_; }
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void set_impl(PortInterface* port);
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virtual const std::string& Type() const;
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virtual rtc::Network* Network() const;
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virtual void SetIceProtocolType(IceProtocolType protocol);
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virtual IceProtocolType IceProtocol() const;
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// Methods to set/get ICE role and tiebreaker values.
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virtual void SetIceRole(IceRole role);
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virtual IceRole GetIceRole() const;
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virtual void SetIceTiebreaker(uint64 tiebreaker);
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virtual uint64 IceTiebreaker() const;
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virtual bool SharedSocket() const;
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// Forwards call to the actual Port.
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virtual void PrepareAddress();
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virtual Connection* CreateConnection(const Candidate& remote_candidate,
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CandidateOrigin origin);
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virtual Connection* GetConnection(
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const rtc::SocketAddress& remote_addr);
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virtual int SendTo(const void* data, size_t size,
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const rtc::SocketAddress& addr,
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const rtc::PacketOptions& options,
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bool payload);
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virtual int SetOption(rtc::Socket::Option opt, int value);
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virtual int GetOption(rtc::Socket::Option opt, int* value);
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virtual int GetError();
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virtual const std::vector<Candidate>& Candidates() const;
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virtual void SendBindingResponse(StunMessage* request,
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const rtc::SocketAddress& addr);
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virtual void SendBindingErrorResponse(
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StunMessage* request, const rtc::SocketAddress& addr,
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int error_code, const std::string& reason);
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virtual void EnablePortPackets();
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virtual std::string ToString() const;
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private:
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void OnUnknownAddress(PortInterface *port,
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const rtc::SocketAddress &addr,
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ProtocolType proto,
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IceMessage *stun_msg,
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const std::string &remote_username,
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bool port_muxed);
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void OnRoleConflict(PortInterface* port);
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void OnPortDestroyed(PortInterface* port);
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PortInterface* impl_;
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};
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} // namespace cricket
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#endif // WEBRTC_P2P_BASE_PORTPROXY_H_
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