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
207 lines
7.1 KiB
C++
207 lines
7.1 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_RAWTRANSPORTCHANNEL_H_
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#define WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_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/rawtransport.h"
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#include "webrtc/p2p/base/transportchannelimpl.h"
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#include "webrtc/base/messagequeue.h"
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#if defined(FEATURE_ENABLE_PSTN)
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namespace rtc {
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class Thread;
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}
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namespace cricket {
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class Connection;
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class PortAllocator;
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class PortAllocatorSession;
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class PortInterface;
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class RelayPort;
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class StunPort;
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// Implements a channel that just sends bare packets once we have received the
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// address of the other side. We pick a single address to send them based on
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// a simple investigation of NAT type.
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class RawTransportChannel : public TransportChannelImpl,
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public rtc::MessageHandler {
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public:
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RawTransportChannel(const std::string& content_name,
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int component,
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RawTransport* transport,
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rtc::Thread *worker_thread,
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PortAllocator *allocator);
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virtual ~RawTransportChannel();
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// Implementation of normal channel packet sending.
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virtual int SendPacket(const char *data, size_t len,
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const rtc::PacketOptions& options, int flags);
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virtual int SetOption(rtc::Socket::Option opt, int value);
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virtual int GetError();
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// Implements TransportChannelImpl.
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virtual Transport* GetTransport() { return raw_transport_; }
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virtual void SetIceCredentials(const std::string& ice_ufrag,
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const std::string& ice_pwd) {}
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virtual void SetRemoteIceCredentials(const std::string& ice_ufrag,
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const std::string& ice_pwd) {}
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// Creates an allocator session to start figuring out which type of
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// port we should send to the other client. This will send
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// SignalAvailableCandidate once we have decided.
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virtual void Connect();
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// Resets state back to unconnected.
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virtual void Reset();
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// We don't actually worry about signaling since we can't send new candidates.
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virtual void OnSignalingReady() {}
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// Handles a message setting the remote address. We are writable once we
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// have this since we now know where to send.
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virtual void OnCandidate(const Candidate& candidate);
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void OnRemoteAddress(const rtc::SocketAddress& remote_address);
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// Below ICE specific virtual methods not implemented.
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virtual IceRole GetIceRole() const { return ICEROLE_UNKNOWN; }
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virtual void SetIceRole(IceRole role) {}
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virtual void SetIceTiebreaker(uint64 tiebreaker) {}
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virtual bool GetIceProtocolType(IceProtocolType* type) const { return false; }
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virtual void SetIceProtocolType(IceProtocolType type) {}
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virtual void SetIceUfrag(const std::string& ice_ufrag) {}
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virtual void SetIcePwd(const std::string& ice_pwd) {}
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virtual void SetRemoteIceMode(IceMode mode) {}
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virtual size_t GetConnectionCount() const { return 1; }
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virtual bool GetStats(ConnectionInfos* infos) {
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return false;
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}
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// DTLS methods.
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virtual bool IsDtlsActive() const { return false; }
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// Default implementation.
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virtual bool GetSslRole(rtc::SSLRole* role) const {
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return false;
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}
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virtual bool SetSslRole(rtc::SSLRole role) {
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return false;
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}
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// Set up the ciphers to use for DTLS-SRTP.
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virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
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return false;
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}
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// Find out which DTLS-SRTP cipher was negotiated
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virtual bool GetSrtpCipher(std::string* cipher) {
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return false;
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}
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// Returns false because the channel is not DTLS.
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virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const {
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return false;
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}
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virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const {
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return false;
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}
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// Allows key material to be extracted for external encryption.
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virtual bool ExportKeyingMaterial(
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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) {
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return false;
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}
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virtual bool SetLocalIdentity(rtc::SSLIdentity* identity) {
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return false;
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}
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// Set DTLS Remote fingerprint. Must be after local identity set.
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virtual bool SetRemoteFingerprint(
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const std::string& digest_alg,
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const uint8* digest,
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size_t digest_len) {
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return false;
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}
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private:
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RawTransport* raw_transport_;
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rtc::Thread *worker_thread_;
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PortAllocator* allocator_;
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PortAllocatorSession* allocator_session_;
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StunPort* stun_port_;
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RelayPort* relay_port_;
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PortInterface* port_;
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bool use_relay_;
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rtc::SocketAddress remote_address_;
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// Called when the allocator creates another port.
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void OnPortReady(PortAllocatorSession* session, PortInterface* port);
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// Called when one of the ports we are using has determined its address.
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void OnCandidatesReady(PortAllocatorSession *session,
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const std::vector<Candidate>& candidates);
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// Called once we have chosen the port to use for communication with the
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// other client. This will send its address and prepare the port for use.
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void SetPort(PortInterface* port);
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// Called once we have a port and a remote address. This will set mark the
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// channel as writable and signal the route to the client.
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void SetWritable();
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// Called when we receive a packet from the other client.
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void OnReadPacket(PortInterface* port, const char* data, size_t size,
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const rtc::SocketAddress& addr);
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// Handles a message to destroy unused ports.
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virtual void OnMessage(rtc::Message *msg);
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DISALLOW_EVIL_CONSTRUCTORS(RawTransportChannel);
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};
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} // namespace cricket
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#endif // defined(FEATURE_ENABLE_PSTN)
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#endif // WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
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