webrtc/talk/p2p/base/rawtransportchannel.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

207 lines
7.1 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_RAWTRANSPORTCHANNEL_H_
#define WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
#include <string>
#include <vector>
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/rawtransport.h"
#include "webrtc/p2p/base/transportchannelimpl.h"
#include "webrtc/base/messagequeue.h"
#if defined(FEATURE_ENABLE_PSTN)
namespace rtc {
class Thread;
}
namespace cricket {
class Connection;
class PortAllocator;
class PortAllocatorSession;
class PortInterface;
class RelayPort;
class StunPort;
// Implements a channel that just sends bare packets once we have received the
// address of the other side. We pick a single address to send them based on
// a simple investigation of NAT type.
class RawTransportChannel : public TransportChannelImpl,
public rtc::MessageHandler {
public:
RawTransportChannel(const std::string& content_name,
int component,
RawTransport* transport,
rtc::Thread *worker_thread,
PortAllocator *allocator);
virtual ~RawTransportChannel();
// Implementation of normal channel packet sending.
virtual int SendPacket(const char *data, size_t len,
const rtc::PacketOptions& options, int flags);
virtual int SetOption(rtc::Socket::Option opt, int value);
virtual int GetError();
// Implements TransportChannelImpl.
virtual Transport* GetTransport() { return raw_transport_; }
virtual void SetIceCredentials(const std::string& ice_ufrag,
const std::string& ice_pwd) {}
virtual void SetRemoteIceCredentials(const std::string& ice_ufrag,
const std::string& ice_pwd) {}
// Creates an allocator session to start figuring out which type of
// port we should send to the other client. This will send
// SignalAvailableCandidate once we have decided.
virtual void Connect();
// Resets state back to unconnected.
virtual void Reset();
// We don't actually worry about signaling since we can't send new candidates.
virtual void OnSignalingReady() {}
// Handles a message setting the remote address. We are writable once we
// have this since we now know where to send.
virtual void OnCandidate(const Candidate& candidate);
void OnRemoteAddress(const rtc::SocketAddress& remote_address);
// Below ICE specific virtual methods not implemented.
virtual IceRole GetIceRole() const { return ICEROLE_UNKNOWN; }
virtual void SetIceRole(IceRole role) {}
virtual void SetIceTiebreaker(uint64 tiebreaker) {}
virtual bool GetIceProtocolType(IceProtocolType* type) const { return false; }
virtual void SetIceProtocolType(IceProtocolType type) {}
virtual void SetIceUfrag(const std::string& ice_ufrag) {}
virtual void SetIcePwd(const std::string& ice_pwd) {}
virtual void SetRemoteIceMode(IceMode mode) {}
virtual size_t GetConnectionCount() const { return 1; }
virtual bool GetStats(ConnectionInfos* infos) {
return false;
}
// DTLS methods.
virtual bool IsDtlsActive() const { return false; }
// Default implementation.
virtual bool GetSslRole(rtc::SSLRole* role) const {
return false;
}
virtual bool SetSslRole(rtc::SSLRole role) {
return false;
}
// Set up the ciphers to use for DTLS-SRTP.
virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
return false;
}
// Find out which DTLS-SRTP cipher was negotiated
virtual bool GetSrtpCipher(std::string* cipher) {
return false;
}
// Returns false because the channel is not DTLS.
virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const {
return false;
}
virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const {
return false;
}
// Allows key material to be extracted for external encryption.
virtual bool ExportKeyingMaterial(
const std::string& label,
const uint8* context,
size_t context_len,
bool use_context,
uint8* result,
size_t result_len) {
return false;
}
virtual bool SetLocalIdentity(rtc::SSLIdentity* identity) {
return false;
}
// Set DTLS Remote fingerprint. Must be after local identity set.
virtual bool SetRemoteFingerprint(
const std::string& digest_alg,
const uint8* digest,
size_t digest_len) {
return false;
}
private:
RawTransport* raw_transport_;
rtc::Thread *worker_thread_;
PortAllocator* allocator_;
PortAllocatorSession* allocator_session_;
StunPort* stun_port_;
RelayPort* relay_port_;
PortInterface* port_;
bool use_relay_;
rtc::SocketAddress remote_address_;
// Called when the allocator creates another port.
void OnPortReady(PortAllocatorSession* session, PortInterface* port);
// Called when one of the ports we are using has determined its address.
void OnCandidatesReady(PortAllocatorSession *session,
const std::vector<Candidate>& candidates);
// Called once we have chosen the port to use for communication with the
// other client. This will send its address and prepare the port for use.
void SetPort(PortInterface* port);
// Called once we have a port and a remote address. This will set mark the
// channel as writable and signal the route to the client.
void SetWritable();
// Called when we receive a packet from the other client.
void OnReadPacket(PortInterface* port, const char* data, size_t size,
const rtc::SocketAddress& addr);
// Handles a message to destroy unused ports.
virtual void OnMessage(rtc::Message *msg);
DISALLOW_EVIL_CONSTRUCTORS(RawTransportChannel);
};
} // namespace cricket
#endif // defined(FEATURE_ENABLE_PSTN)
#endif // WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_