webrtc/talk/p2p/base/transportchannel.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

161 lines
6.1 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_TRANSPORTCHANNEL_H_
#define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
#include <string>
#include <vector>
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/transport.h"
#include "webrtc/p2p/base/transportdescription.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/socket.h"
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/sslstreamadapter.h"
namespace cricket {
class Candidate;
// Flags for SendPacket/SignalReadPacket.
enum PacketFlags {
PF_NORMAL = 0x00, // A normal packet.
PF_SRTP_BYPASS = 0x01, // An encrypted SRTP packet; bypass any additional
// crypto provided by the transport (e.g. DTLS)
};
// A TransportChannel represents one logical stream of packets that are sent
// between the two sides of a session.
class TransportChannel : public sigslot::has_slots<> {
public:
explicit TransportChannel(const std::string& content_name, int component)
: content_name_(content_name),
component_(component),
readable_(false), writable_(false) {}
virtual ~TransportChannel() {}
// TODO(mallinath) - Remove this API, as it's no longer useful.
// Returns the session id of this channel.
virtual const std::string SessionId() const { return std::string(); }
const std::string& content_name() const { return content_name_; }
int component() const { return component_; }
// Returns the readable and states of this channel. Each time one of these
// states changes, a signal is raised. These states are aggregated by the
// TransportManager.
bool readable() const { return readable_; }
bool writable() const { return writable_; }
sigslot::signal1<TransportChannel*> SignalReadableState;
sigslot::signal1<TransportChannel*> SignalWritableState;
// Emitted when the TransportChannel's ability to send has changed.
sigslot::signal1<TransportChannel*> SignalReadyToSend;
// Attempts to send the given packet. The return value is < 0 on failure.
// TODO: Remove the default argument once channel code is updated.
virtual int SendPacket(const char* data, size_t len,
const rtc::PacketOptions& options,
int flags = 0) = 0;
// Sets a socket option on this channel. Note that not all options are
// supported by all transport types.
virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
// Returns the most recent error that occurred on this channel.
virtual int GetError() = 0;
// Returns the current stats for this connection.
virtual bool GetStats(ConnectionInfos* infos) = 0;
// Is DTLS active?
virtual bool IsDtlsActive() const = 0;
// Default implementation.
virtual bool GetSslRole(rtc::SSLRole* role) const = 0;
// Sets up the ciphers to use for DTLS-SRTP.
virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) = 0;
// Finds out which DTLS-SRTP cipher was negotiated
virtual bool GetSrtpCipher(std::string* cipher) = 0;
// Gets a copy of the local SSL identity, owned by the caller.
virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const = 0;
// Gets a copy of the remote side's SSL certificate, owned by the caller.
virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const = 0;
// 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) = 0;
// Signalled each time a packet is received on this channel.
sigslot::signal5<TransportChannel*, const char*,
size_t, const rtc::PacketTime&, int> SignalReadPacket;
// This signal occurs when there is a change in the way that packets are
// being routed, i.e. to a different remote location. The candidate
// indicates where and how we are currently sending media.
sigslot::signal2<TransportChannel*, const Candidate&> SignalRouteChange;
// Invoked when the channel is being destroyed.
sigslot::signal1<TransportChannel*> SignalDestroyed;
// Debugging description of this transport channel.
std::string ToString() const;
protected:
// Sets the readable state, signaling if necessary.
void set_readable(bool readable);
// Sets the writable state, signaling if necessary.
void set_writable(bool writable);
private:
// Used mostly for debugging.
std::string content_name_;
int component_;
bool readable_;
bool writable_;
DISALLOW_EVIL_CONSTRUCTORS(TransportChannel);
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_