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
84 lines
3.2 KiB
C++
84 lines
3.2 KiB
C++
/*
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* libjingle
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* Copyright 2004--2005, 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_STUNSERVER_H_
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#define WEBRTC_P2P_BASE_STUNSERVER_H_
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#include "webrtc/p2p/base/stun.h"
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#include "webrtc/base/asyncudpsocket.h"
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#include "webrtc/base/scoped_ptr.h"
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namespace cricket {
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const int STUN_SERVER_PORT = 3478;
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class StunServer : public sigslot::has_slots<> {
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public:
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// Creates a STUN server, which will listen on the given socket.
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explicit StunServer(rtc::AsyncUDPSocket* socket);
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// Removes the STUN server from the socket and deletes the socket.
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~StunServer();
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protected:
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// Slot for AsyncSocket.PacketRead:
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void OnPacket(
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rtc::AsyncPacketSocket* socket, const char* buf, size_t size,
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const rtc::SocketAddress& remote_addr,
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const rtc::PacketTime& packet_time);
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// Handlers for the different types of STUN/TURN requests:
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virtual void OnBindingRequest(StunMessage* msg,
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const rtc::SocketAddress& addr);
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void OnAllocateRequest(StunMessage* msg,
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const rtc::SocketAddress& addr);
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void OnSharedSecretRequest(StunMessage* msg,
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const rtc::SocketAddress& addr);
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void OnSendRequest(StunMessage* msg,
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const rtc::SocketAddress& addr);
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// Sends an error response to the given message back to the user.
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void SendErrorResponse(
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const StunMessage& msg, const rtc::SocketAddress& addr,
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int error_code, const char* error_desc);
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// Sends the given message to the appropriate destination.
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void SendResponse(const StunMessage& msg,
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const rtc::SocketAddress& addr);
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// A helper method to compose a STUN binding response.
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void GetStunBindReqponse(StunMessage* request,
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const rtc::SocketAddress& remote_addr,
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StunMessage* response) const;
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private:
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rtc::scoped_ptr<rtc::AsyncUDPSocket> socket_;
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};
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} // namespace cricket
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#endif // WEBRTC_P2P_BASE_STUNSERVER_H_
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