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
161 lines
6.1 KiB
C++
161 lines
6.1 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_TRANSPORTCHANNEL_H_
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#define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
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#include <string>
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#include <vector>
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#include "webrtc/p2p/base/candidate.h"
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#include "webrtc/p2p/base/transport.h"
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#include "webrtc/p2p/base/transportdescription.h"
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#include "webrtc/base/asyncpacketsocket.h"
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/dscp.h"
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#include "webrtc/base/sigslot.h"
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#include "webrtc/base/socket.h"
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#include "webrtc/base/sslidentity.h"
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#include "webrtc/base/sslstreamadapter.h"
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namespace cricket {
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class Candidate;
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// Flags for SendPacket/SignalReadPacket.
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enum PacketFlags {
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PF_NORMAL = 0x00, // A normal packet.
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PF_SRTP_BYPASS = 0x01, // An encrypted SRTP packet; bypass any additional
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// crypto provided by the transport (e.g. DTLS)
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};
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// A TransportChannel represents one logical stream of packets that are sent
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// between the two sides of a session.
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class TransportChannel : public sigslot::has_slots<> {
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public:
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explicit TransportChannel(const std::string& content_name, int component)
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: content_name_(content_name),
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component_(component),
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readable_(false), writable_(false) {}
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virtual ~TransportChannel() {}
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// TODO(mallinath) - Remove this API, as it's no longer useful.
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// Returns the session id of this channel.
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virtual const std::string SessionId() const { return std::string(); }
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const std::string& content_name() const { return content_name_; }
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int component() const { return component_; }
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// Returns the readable and states of this channel. Each time one of these
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// states changes, a signal is raised. These states are aggregated by the
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// TransportManager.
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bool readable() const { return readable_; }
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bool writable() const { return writable_; }
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sigslot::signal1<TransportChannel*> SignalReadableState;
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sigslot::signal1<TransportChannel*> SignalWritableState;
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// Emitted when the TransportChannel's ability to send has changed.
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sigslot::signal1<TransportChannel*> SignalReadyToSend;
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// Attempts to send the given packet. The return value is < 0 on failure.
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// TODO: Remove the default argument once channel code is updated.
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virtual int SendPacket(const char* data, size_t len,
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const rtc::PacketOptions& options,
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int flags = 0) = 0;
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// Sets a socket option on this channel. Note that not all options are
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// supported by all transport types.
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virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
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// Returns the most recent error that occurred on this channel.
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virtual int GetError() = 0;
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// Returns the current stats for this connection.
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virtual bool GetStats(ConnectionInfos* infos) = 0;
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// Is DTLS active?
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virtual bool IsDtlsActive() const = 0;
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// Default implementation.
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virtual bool GetSslRole(rtc::SSLRole* role) const = 0;
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// Sets up the ciphers to use for DTLS-SRTP.
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virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) = 0;
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// Finds out which DTLS-SRTP cipher was negotiated
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virtual bool GetSrtpCipher(std::string* cipher) = 0;
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// Gets a copy of the local SSL identity, owned by the caller.
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virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const = 0;
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// Gets a copy of the remote side's SSL certificate, owned by the caller.
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virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const = 0;
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// Allows key material to be extracted for external encryption.
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virtual bool ExportKeyingMaterial(const std::string& label,
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const uint8* context,
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size_t context_len,
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bool use_context,
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uint8* result,
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size_t result_len) = 0;
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// Signalled each time a packet is received on this channel.
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sigslot::signal5<TransportChannel*, const char*,
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size_t, const rtc::PacketTime&, int> SignalReadPacket;
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// This signal occurs when there is a change in the way that packets are
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// being routed, i.e. to a different remote location. The candidate
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// indicates where and how we are currently sending media.
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sigslot::signal2<TransportChannel*, const Candidate&> SignalRouteChange;
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// Invoked when the channel is being destroyed.
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sigslot::signal1<TransportChannel*> SignalDestroyed;
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// Debugging description of this transport channel.
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std::string ToString() const;
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protected:
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// Sets the readable state, signaling if necessary.
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void set_readable(bool readable);
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// Sets the writable state, signaling if necessary.
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void set_writable(bool writable);
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private:
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// Used mostly for debugging.
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std::string content_name_;
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int component_;
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bool readable_;
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bool writable_;
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DISALLOW_EVIL_CONSTRUCTORS(TransportChannel);
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};
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} // namespace cricket
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#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
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