| /* |
| * Copyright 2015 The WebRTC Project Authors. All rights reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef PC_TRANSPORTCONTROLLER_H_ |
| #define PC_TRANSPORTCONTROLLER_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/candidate.h" |
| #include "p2p/base/dtlstransport.h" |
| #include "p2p/base/jseptransport.h" |
| #include "p2p/base/p2ptransportchannel.h" |
| #include "pc/dtlssrtptransport.h" |
| #include "pc/rtptransport.h" |
| #include "pc/srtptransport.h" |
| #include "rtc_base/asyncinvoker.h" |
| #include "rtc_base/constructormagic.h" |
| #include "rtc_base/refcountedobject.h" |
| #include "rtc_base/sigslot.h" |
| #include "rtc_base/sslstreamadapter.h" |
| |
| namespace rtc { |
| class Thread; |
| class PacketTransportInternal; |
| } // namespace rtc |
| |
| namespace webrtc { |
| class MetricsObserverInterface; |
| } // namespace webrtc |
| |
| namespace cricket { |
| |
| class TransportController : public sigslot::has_slots<>, |
| public rtc::MessageHandler { |
| public: |
| // If |redetermine_role_on_ice_restart| is true, ICE role is redetermined |
| // upon setting a local transport description that indicates an ICE restart. |
| // For the constructor that doesn't take this parameter, it defaults to true. |
| // |
| // |crypto_options| is used to determine if created DTLS transports negotiate |
| // GCM crypto suites or not. |
| TransportController(rtc::Thread* signaling_thread, |
| rtc::Thread* network_thread, |
| PortAllocator* port_allocator, |
| bool redetermine_role_on_ice_restart, |
| const rtc::CryptoOptions& crypto_options); |
| |
| virtual ~TransportController(); |
| |
| rtc::Thread* signaling_thread() const { return signaling_thread_; } |
| rtc::Thread* network_thread() const { return network_thread_; } |
| |
| PortAllocator* port_allocator() const { return port_allocator_; } |
| |
| // Can only be set before transports are created. |
| // TODO(deadbeef): Make this an argument to the constructor once BaseSession |
| // and WebRtcSession are combined |
| bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version); |
| |
| void SetIceConfig(const IceConfig& config); |
| void SetIceRole(IceRole ice_role); |
| |
| // Set the "needs-ice-restart" flag as described in JSEP. After the flag is |
| // set, offers should generate new ufrags/passwords until an ICE restart |
| // occurs. |
| void SetNeedsIceRestartFlag(); |
| // Returns true if the ICE restart flag above was set, and no ICE restart has |
| // occurred yet for this transport (by applying a local description with |
| // changed ufrag/password). If the transport has been deleted as a result of |
| // bundling, returns false. |
| bool NeedsIceRestart(const std::string& transport_name) const; |
| |
| bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role) const; |
| |
| // Specifies the identity to use in this session. |
| // Can only be called once. |
| bool SetLocalCertificate( |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
| bool GetLocalCertificate( |
| const std::string& transport_name, |
| rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const; |
| // Caller owns returned certificate. This method mainly exists for stats |
| // reporting. |
| std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate( |
| const std::string& transport_name) const; |
| bool SetLocalTransportDescription(const std::string& transport_name, |
| const TransportDescription& tdesc, |
| webrtc::SdpType type, |
| std::string* err); |
| bool SetRemoteTransportDescription(const std::string& transport_name, |
| const TransportDescription& tdesc, |
| webrtc::SdpType type, |
| std::string* err); |
| // Start gathering candidates for any new transports, or transports doing an |
| // ICE restart. |
| void MaybeStartGathering(); |
| bool AddRemoteCandidates(const std::string& transport_name, |
| const Candidates& candidates, |
| std::string* err); |
| bool RemoveRemoteCandidates(const Candidates& candidates, std::string* err); |
| bool ReadyForRemoteCandidates(const std::string& transport_name) const; |
| // TODO(deadbeef): GetStats isn't const because all the way down to |
| // OpenSSLStreamAdapter, |
| // GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not const. Fix this. |
| bool GetStats(const std::string& transport_name, TransportStats* stats); |
| void SetMetricsObserver(webrtc::MetricsObserverInterface* metrics_observer); |
| |
| // Creates a channel if it doesn't exist. Otherwise, increments a reference |
| // count and returns an existing channel. |
| DtlsTransportInternal* CreateDtlsTransport(const std::string& transport_name, |
| int component); |
| virtual DtlsTransportInternal* CreateDtlsTransport_n( |
| const std::string& transport_name, |
| int component); |
| |
| // Decrements a channel's reference count, and destroys the channel if |
| // nothing is referencing it. |
| virtual void DestroyDtlsTransport(const std::string& transport_name, |
| int component); |
| virtual void DestroyDtlsTransport_n(const std::string& transport_name, |
| int component); |
| |
| // Create an SrtpTransport/DtlsSrtpTransport if it doesn't exist. |
| // Otherwise, increments a reference count and returns the existing one. |
| // These methods are not currently used but the plan is to transition |
| // PeerConnection and BaseChannel to use them instead of CreateDtlsTransport. |
| webrtc::SrtpTransport* CreateSdesTransport(const std::string& transport_name, |
| bool rtcp_mux_enabled); |
| webrtc::DtlsSrtpTransport* CreateDtlsSrtpTransport( |
| const std::string& transport_name, |
| bool rtcp_mux_enabled); |
| |
| // Destroy an RTP level transport which can be an RtpTransport, an |
| // SrtpTransport or a DtlsSrtpTransport. |
| void DestroyTransport(const std::string& transport_name); |
| |
| // TODO(deadbeef): Remove all for_testing methods! |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing() |
| const { |
| return certificate_; |
| } |
| std::vector<std::string> transport_names_for_testing(); |
| std::vector<DtlsTransportInternal*> channels_for_testing(); |
| DtlsTransportInternal* get_channel_for_testing( |
| const std::string& transport_name, |
| int component); |
| |
| // All of these signals are fired on the signalling thread. |
| |
| // If any transport failed => failed, |
| // Else if all completed => completed, |
| // Else if all connected => connected, |
| // Else => connecting |
| sigslot::signal1<IceConnectionState> SignalConnectionState; |
| |
| // Receiving if any transport is receiving |
| sigslot::signal1<bool> SignalReceiving; |
| |
| // If all transports done gathering => complete, |
| // Else if any are gathering => gathering, |
| // Else => new |
| sigslot::signal1<IceGatheringState> SignalGatheringState; |
| |
| // (transport_name, candidates) |
| sigslot::signal2<const std::string&, const Candidates&> |
| SignalCandidatesGathered; |
| |
| sigslot::signal1<const Candidates&> SignalCandidatesRemoved; |
| |
| sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError; |
| |
| protected: |
| // TODO(deadbeef): Get rid of these virtual methods. Used by |
| // FakeTransportController currently, but FakeTransportController shouldn't |
| // even be functioning by subclassing TransportController. |
| virtual IceTransportInternal* CreateIceTransportChannel_n( |
| const std::string& transport_name, |
| int component); |
| virtual DtlsTransportInternal* CreateDtlsTransportChannel_n( |
| const std::string& transport_name, |
| int component, |
| IceTransportInternal* ice); |
| |
| private: |
| void OnMessage(rtc::Message* pmsg) override; |
| |
| class ChannelPair; |
| typedef rtc::RefCountedObject<ChannelPair> RefCountedChannel; |
| |
| // Wrapper for RtpTransport that keeps a reference count. |
| // When using SDES, |srtp_transport| is non-null, |dtls_srtp_transport| is |
| // null and |rtp_transport.get()| == |srtp_transport|, |
| // When using DTLS-SRTP, |dtls_srtp_transport| is non-null, |srtp_transport| |
| // is null and |rtp_transport.get()| == |dtls_srtp_transport|, |
| // When using unencrypted RTP, only |rtp_transport| is non-null. |
| struct RtpTransportWrapper { |
| // |rtp_transport| is always non-null. |
| std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport; |
| webrtc::SrtpTransport* srtp_transport = nullptr; |
| webrtc::DtlsSrtpTransport* dtls_srtp_transport = nullptr; |
| }; |
| |
| typedef rtc::RefCountedObject<RtpTransportWrapper> RefCountedRtpTransport; |
| |
| const RefCountedRtpTransport* FindRtpTransport( |
| const std::string& transport_name); |
| |
| // Helper functions to get a channel or transport, or iterator to it (in case |
| // it needs to be erased). |
| std::vector<RefCountedChannel*>::iterator GetChannelIterator_n( |
| const std::string& transport_name, |
| int component); |
| std::vector<RefCountedChannel*>::const_iterator GetChannelIterator_n( |
| const std::string& transport_name, |
| int component) const; |
| const JsepTransport* GetJsepTransport( |
| const std::string& transport_name) const; |
| JsepTransport* GetJsepTransport(const std::string& transport_name); |
| const RefCountedChannel* GetChannel_n(const std::string& transport_name, |
| int component) const; |
| RefCountedChannel* GetChannel_n(const std::string& transport_name, |
| int component); |
| |
| JsepTransport* GetOrCreateJsepTransport(const std::string& transport_name); |
| void DestroyAllChannels_n(); |
| |
| bool SetSslMaxProtocolVersion_n(rtc::SSLProtocolVersion version); |
| void SetIceConfig_n(const IceConfig& config); |
| void SetIceRole_n(IceRole ice_role); |
| bool GetSslRole_n(const std::string& transport_name, |
| rtc::SSLRole* role) const; |
| bool SetLocalCertificate_n( |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
| bool GetLocalCertificate_n( |
| const std::string& transport_name, |
| rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const; |
| std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate_n( |
| const std::string& transport_name) const; |
| bool SetLocalTransportDescription_n(const std::string& transport_name, |
| const TransportDescription& tdesc, |
| webrtc::SdpType type, |
| std::string* err); |
| bool SetRemoteTransportDescription_n(const std::string& transport_name, |
| const TransportDescription& tdesc, |
| webrtc::SdpType type, |
| std::string* err); |
| void MaybeStartGathering_n(); |
| bool AddRemoteCandidates_n(const std::string& transport_name, |
| const Candidates& candidates, |
| std::string* err); |
| bool RemoveRemoteCandidates_n(const Candidates& candidates, std::string* err); |
| bool ReadyForRemoteCandidates_n(const std::string& transport_name) const; |
| bool GetStats_n(const std::string& transport_name, TransportStats* stats); |
| void SetMetricsObserver_n(webrtc::MetricsObserverInterface* metrics_observer); |
| |
| // Handlers for signals from Transport. |
| void OnChannelWritableState_n(rtc::PacketTransportInternal* transport); |
| void OnChannelReceivingState_n(rtc::PacketTransportInternal* transport); |
| void OnChannelGatheringState_n(IceTransportInternal* channel); |
| void OnChannelCandidateGathered_n(IceTransportInternal* channel, |
| const Candidate& candidate); |
| void OnChannelCandidatesRemoved(const Candidates& candidates); |
| void OnChannelCandidatesRemoved_n(IceTransportInternal* channel, |
| const Candidates& candidates); |
| void OnChannelRoleConflict_n(IceTransportInternal* channel); |
| void OnChannelStateChanged_n(IceTransportInternal* channel); |
| |
| void UpdateAggregateStates_n(); |
| |
| void OnDtlsHandshakeError(rtc::SSLHandshakeError error); |
| |
| rtc::Thread* const signaling_thread_ = nullptr; |
| rtc::Thread* const network_thread_ = nullptr; |
| PortAllocator* const port_allocator_ = nullptr; |
| |
| std::map<std::string, std::unique_ptr<JsepTransport>> transports_; |
| std::vector<RefCountedChannel*> channels_; |
| |
| std::map<std::string, RefCountedRtpTransport*> rtp_transports_; |
| |
| // Aggregate state for TransportChannelImpls. |
| IceConnectionState connection_state_ = kIceConnectionConnecting; |
| bool receiving_ = false; |
| IceGatheringState gathering_state_ = kIceGatheringNew; |
| |
| IceConfig ice_config_; |
| IceRole ice_role_ = ICEROLE_CONTROLLING; |
| bool redetermine_role_on_ice_restart_; |
| uint64_t ice_tiebreaker_ = rtc::CreateRandomId64(); |
| rtc::CryptoOptions crypto_options_; |
| rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12; |
| rtc::scoped_refptr<rtc::RTCCertificate> certificate_; |
| rtc::AsyncInvoker invoker_; |
| |
| webrtc::MetricsObserverInterface* metrics_observer_ = nullptr; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(TransportController); |
| }; |
| |
| } // namespace cricket |
| |
| #endif // PC_TRANSPORTCONTROLLER_H_ |