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
134 lines
4.4 KiB
C++
134 lines
4.4 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_STUNREQUEST_H_
|
|
#define WEBRTC_P2P_BASE_STUNREQUEST_H_
|
|
|
|
#include <map>
|
|
#include <string>
|
|
#include "webrtc/p2p/base/stun.h"
|
|
#include "webrtc/base/sigslot.h"
|
|
#include "webrtc/base/thread.h"
|
|
|
|
namespace cricket {
|
|
|
|
class StunRequest;
|
|
|
|
// Manages a set of STUN requests, sending and resending until we receive a
|
|
// response or determine that the request has timed out.
|
|
class StunRequestManager {
|
|
public:
|
|
StunRequestManager(rtc::Thread* thread);
|
|
~StunRequestManager();
|
|
|
|
// Starts sending the given request (perhaps after a delay).
|
|
void Send(StunRequest* request);
|
|
void SendDelayed(StunRequest* request, int delay);
|
|
|
|
// Removes a stun request that was added previously. This will happen
|
|
// automatically when a request succeeds, fails, or times out.
|
|
void Remove(StunRequest* request);
|
|
|
|
// Removes all stun requests that were added previously.
|
|
void Clear();
|
|
|
|
// Determines whether the given message is a response to one of the
|
|
// outstanding requests, and if so, processes it appropriately.
|
|
bool CheckResponse(StunMessage* msg);
|
|
bool CheckResponse(const char* data, size_t size);
|
|
|
|
bool empty() { return requests_.empty(); }
|
|
|
|
// Raised when there are bytes to be sent.
|
|
sigslot::signal3<const void*, size_t, StunRequest*> SignalSendPacket;
|
|
|
|
private:
|
|
typedef std::map<std::string, StunRequest*> RequestMap;
|
|
|
|
rtc::Thread* thread_;
|
|
RequestMap requests_;
|
|
|
|
friend class StunRequest;
|
|
};
|
|
|
|
// Represents an individual request to be sent. The STUN message can either be
|
|
// constructed beforehand or built on demand.
|
|
class StunRequest : public rtc::MessageHandler {
|
|
public:
|
|
StunRequest();
|
|
StunRequest(StunMessage* request);
|
|
virtual ~StunRequest();
|
|
|
|
// Causes our wrapped StunMessage to be Prepared
|
|
void Construct();
|
|
|
|
// The manager handling this request (if it has been scheduled for sending).
|
|
StunRequestManager* manager() { return manager_; }
|
|
|
|
// Returns the transaction ID of this request.
|
|
const std::string& id() { return msg_->transaction_id(); }
|
|
|
|
// Returns the STUN type of the request message.
|
|
int type();
|
|
|
|
// Returns a const pointer to |msg_|.
|
|
const StunMessage* msg() const;
|
|
|
|
// Time elapsed since last send (in ms)
|
|
uint32 Elapsed() const;
|
|
|
|
protected:
|
|
int count_;
|
|
bool timeout_;
|
|
|
|
// Fills in a request object to be sent. Note that request's transaction ID
|
|
// will already be set and cannot be changed.
|
|
virtual void Prepare(StunMessage* request) {}
|
|
|
|
// Called when the message receives a response or times out.
|
|
virtual void OnResponse(StunMessage* response) {}
|
|
virtual void OnErrorResponse(StunMessage* response) {}
|
|
virtual void OnTimeout() {}
|
|
virtual int GetNextDelay();
|
|
|
|
private:
|
|
void set_manager(StunRequestManager* manager);
|
|
|
|
// Handles messages for sending and timeout.
|
|
void OnMessage(rtc::Message* pmsg);
|
|
|
|
StunRequestManager* manager_;
|
|
StunMessage* msg_;
|
|
uint32 tstamp_;
|
|
|
|
friend class StunRequestManager;
|
|
};
|
|
|
|
} // namespace cricket
|
|
|
|
#endif // WEBRTC_P2P_BASE_STUNREQUEST_H_
|