webrtc/talk/p2p/base/stunport.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

256 lines
8.8 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_STUNPORT_H_
#define WEBRTC_P2P_BASE_STUNPORT_H_
#include <string>
#include "webrtc/p2p/base/port.h"
#include "webrtc/p2p/base/stunrequest.h"
#include "webrtc/base/asyncpacketsocket.h"
// TODO(mallinath) - Rename stunport.cc|h to udpport.cc|h.
namespace rtc {
class AsyncResolver;
class SignalThread;
}
namespace cricket {
// Communicates using the address on the outside of a NAT.
class UDPPort : public Port {
public:
static UDPPort* Create(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
rtc::AsyncPacketSocket* socket,
const std::string& username,
const std::string& password) {
UDPPort* port = new UDPPort(thread, factory, network, socket,
username, password);
if (!port->Init()) {
delete port;
port = NULL;
}
return port;
}
static UDPPort* 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) {
UDPPort* port = new UDPPort(thread, factory, network,
ip, min_port, max_port,
username, password);
if (!port->Init()) {
delete port;
port = NULL;
}
return port;
}
virtual ~UDPPort();
rtc::SocketAddress GetLocalAddress() const {
return socket_->GetLocalAddress();
}
const ServerAddresses& server_addresses() const {
return server_addresses_;
}
void
set_server_addresses(const ServerAddresses& addresses) {
server_addresses_ = addresses;
}
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();
virtual bool HandleIncomingPacket(
rtc::AsyncPacketSocket* socket, const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time) {
// All packets given to UDP port will be consumed.
OnReadPacket(socket, data, size, remote_addr, packet_time);
return true;
}
void set_stun_keepalive_delay(int delay) {
stun_keepalive_delay_ = delay;
}
int stun_keepalive_delay() const {
return stun_keepalive_delay_;
}
protected:
UDPPort(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);
UDPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
rtc::Network* network, rtc::AsyncPacketSocket* socket,
const std::string& username, const std::string& password);
bool Init();
virtual int SendTo(const void* data, size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload);
void OnLocalAddressReady(rtc::AsyncPacketSocket* socket,
const rtc::SocketAddress& address);
void OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
// This method will send STUN binding request if STUN server address is set.
void MaybePrepareStunCandidate();
void SendStunBindingRequests();
private:
// A helper class which can be called repeatedly to resolve multiple
// addresses, as opposed to rtc::AsyncResolverInterface, which can only
// resolve one address per instance.
class AddressResolver : public sigslot::has_slots<> {
public:
explicit AddressResolver(rtc::PacketSocketFactory* factory);
~AddressResolver();
void Resolve(const rtc::SocketAddress& address);
bool GetResolvedAddress(const rtc::SocketAddress& input,
int family,
rtc::SocketAddress* output) const;
// The signal is sent when resolving the specified address is finished. The
// first argument is the input address, the second argument is the error
// or 0 if it succeeded.
sigslot::signal2<const rtc::SocketAddress&, int> SignalDone;
private:
typedef std::map<rtc::SocketAddress,
rtc::AsyncResolverInterface*> ResolverMap;
void OnResolveResult(rtc::AsyncResolverInterface* resolver);
rtc::PacketSocketFactory* socket_factory_;
ResolverMap resolvers_;
};
// DNS resolution of the STUN server.
void ResolveStunAddress(const rtc::SocketAddress& stun_addr);
void OnResolveResult(const rtc::SocketAddress& input, int error);
void SendStunBindingRequest(const rtc::SocketAddress& stun_addr);
// Below methods handles binding request responses.
void OnStunBindingRequestSucceeded(
const rtc::SocketAddress& stun_server_addr,
const rtc::SocketAddress& stun_reflected_addr);
void OnStunBindingOrResolveRequestFailed(
const rtc::SocketAddress& stun_server_addr);
// Sends STUN requests to the server.
void OnSendPacket(const void* data, size_t size, StunRequest* req);
// TODO(mallinaht) - Move this up to cricket::Port when SignalAddressReady is
// changed to SignalPortReady.
void MaybeSetPortCompleteOrError();
bool HasCandidateWithAddress(const rtc::SocketAddress& addr) const;
ServerAddresses server_addresses_;
ServerAddresses bind_request_succeeded_servers_;
ServerAddresses bind_request_failed_servers_;
StunRequestManager requests_;
rtc::AsyncPacketSocket* socket_;
int error_;
rtc::scoped_ptr<AddressResolver> resolver_;
bool ready_;
int stun_keepalive_delay_;
friend class StunBindingRequest;
};
class StunPort : public UDPPort {
public:
static StunPort* 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,
const ServerAddresses& servers) {
StunPort* port = new StunPort(thread, factory, network,
ip, min_port, max_port,
username, password, servers);
if (!port->Init()) {
delete port;
port = NULL;
}
return port;
}
virtual ~StunPort() {}
virtual void PrepareAddress() {
SendStunBindingRequests();
}
protected:
StunPort(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,
const ServerAddresses& servers)
: UDPPort(thread, factory, network, ip, min_port, max_port, username,
password) {
// UDPPort will set these to local udp, updating these to STUN.
set_type(STUN_PORT_TYPE);
set_server_addresses(servers);
}
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_STUNPORT_H_