| /* |
| * Copyright 2017 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_JSEP_TRANSPORT_CONTROLLER_H_ |
| #define PC_JSEP_TRANSPORT_CONTROLLER_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/candidate.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/media_transport_interface.h" |
| #include "api/peer_connection_interface.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "media/sctp/sctp_transport_internal.h" |
| #include "p2p/base/dtls_transport.h" |
| #include "p2p/base/p2p_transport_channel.h" |
| #include "p2p/base/transport_factory_interface.h" |
| #include "pc/channel.h" |
| #include "pc/dtls_srtp_transport.h" |
| #include "pc/dtls_transport.h" |
| #include "pc/jsep_transport.h" |
| #include "pc/rtp_transport.h" |
| #include "pc/srtp_transport.h" |
| #include "rtc_base/async_invoker.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| |
| namespace rtc { |
| class Thread; |
| class PacketTransportInternal; |
| } // namespace rtc |
| |
| namespace webrtc { |
| |
| class JsepTransportController : public sigslot::has_slots<> { |
| public: |
| // Used when the RtpTransport/DtlsTransport of the m= section is changed |
| // because the section is rejected or BUNDLE is enabled. |
| class Observer { |
| public: |
| virtual ~Observer() {} |
| |
| // Returns true if media associated with |mid| was successfully set up to be |
| // demultiplexed on |rtp_transport|. Could return false if two bundled m= |
| // sections use the same SSRC, for example. |
| virtual bool OnTransportChanged( |
| const std::string& mid, |
| RtpTransportInternal* rtp_transport, |
| rtc::scoped_refptr<DtlsTransport> dtls_transport, |
| MediaTransportInterface* media_transport) = 0; |
| }; |
| |
| struct Config { |
| // If |redetermine_role_on_ice_restart| is true, ICE role is redetermined |
| // upon setting a local transport description that indicates an ICE |
| // restart. |
| bool redetermine_role_on_ice_restart = true; |
| rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| // |crypto_options| is used to determine if created DTLS transports |
| // negotiate GCM crypto suites or not. |
| webrtc::CryptoOptions crypto_options; |
| PeerConnectionInterface::BundlePolicy bundle_policy = |
| PeerConnectionInterface::kBundlePolicyBalanced; |
| PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy = |
| PeerConnectionInterface::kRtcpMuxPolicyRequire; |
| bool disable_encryption = false; |
| bool enable_external_auth = false; |
| // Used to inject the ICE/DTLS transports created externally. |
| cricket::TransportFactoryInterface* external_transport_factory = nullptr; |
| Observer* transport_observer = nullptr; |
| bool active_reset_srtp_params = false; |
| RtcEventLog* event_log = nullptr; |
| |
| // Whether media transport is used for media. |
| bool use_media_transport_for_media = false; |
| |
| // Whether media transport is used for data channels. |
| bool use_media_transport_for_data_channels = false; |
| |
| // Optional media transport factory (experimental). If provided it will be |
| // used to create media_transport (as long as either |
| // |use_media_transport_for_media| or |
| // |use_media_transport_for_data_channels| is set to true). However, whether |
| // it will be used to send / receive audio and video frames instead of RTP |
| // is determined by |use_media_transport_for_media|. Note that currently |
| // media_transport co-exists with RTP / RTCP transports and may use the same |
| // underlying ICE transport. |
| MediaTransportFactory* media_transport_factory = nullptr; |
| }; |
| |
| // The ICE related events are signaled on the |signaling_thread|. |
| // All the transport related methods are called on the |network_thread|. |
| JsepTransportController(rtc::Thread* signaling_thread, |
| rtc::Thread* network_thread, |
| cricket::PortAllocator* port_allocator, |
| AsyncResolverFactory* async_resolver_factory, |
| Config config); |
| virtual ~JsepTransportController(); |
| |
| // The main method to be called; applies a description at the transport |
| // level, creating/destroying transport objects as needed and updating their |
| // properties. This includes RTP, DTLS, and ICE (but not SCTP). At least not |
| // yet? May make sense to in the future. |
| RTCError SetLocalDescription(SdpType type, |
| const cricket::SessionDescription* description); |
| |
| RTCError SetRemoteDescription(SdpType type, |
| const cricket::SessionDescription* description); |
| |
| // Get transports to be used for the provided |mid|. If bundling is enabled, |
| // calling GetRtpTransport for multiple MIDs may yield the same object. |
| RtpTransportInternal* GetRtpTransport(const std::string& mid) const; |
| cricket::DtlsTransportInternal* GetDtlsTransport(const std::string& mid); |
| const cricket::DtlsTransportInternal* GetRtcpDtlsTransport( |
| const std::string& mid) const; |
| // Gets the externally sharable version of the DtlsTransport. |
| rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid( |
| const std::string& mid); |
| |
| MediaTransportInterface* GetMediaTransport(const std::string& mid) const; |
| MediaTransportState GetMediaTransportState(const std::string& mid) const; |
| |
| /********************* |
| * ICE-related methods |
| ********************/ |
| // This method is public to allow PeerConnection to update it from |
| // SetConfiguration. |
| void SetIceConfig(const cricket::IceConfig& config); |
| // 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& mid) const; |
| // Start gathering candidates for any new transports, or transports doing an |
| // ICE restart. |
| void MaybeStartGathering(); |
| RTCError AddRemoteCandidates( |
| const std::string& mid, |
| const std::vector<cricket::Candidate>& candidates); |
| RTCError RemoveRemoteCandidates( |
| const std::vector<cricket::Candidate>& candidates); |
| |
| /********************** |
| * DTLS-related methods |
| *********************/ |
| // Specifies the identity to use in this session. |
| // Can only be called once. |
| bool SetLocalCertificate( |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
| rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate( |
| const std::string& mid) const; |
| // Caller owns returned certificate chain. This method mainly exists for |
| // stats reporting. |
| std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain( |
| const std::string& mid) const; |
| // Get negotiated role, if one has been negotiated. |
| absl::optional<rtc::SSLRole> GetDtlsRole(const std::string& mid) 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& mid, cricket::TransportStats* stats); |
| |
| bool initial_offerer() const { return initial_offerer_ && *initial_offerer_; } |
| |
| void SetActiveResetSrtpParams(bool active_reset_srtp_params); |
| |
| // Allows to overwrite the settings from config. You may set or reset the |
| // media transport configuration on the jsep transport controller, as long as |
| // you did not call 'GetMediaTransport' or 'MaybeCreateJsepTransport'. Once |
| // Jsep transport is created, you can't change this setting. |
| void SetMediaTransportSettings(bool use_media_transport_for_media, |
| bool use_media_transport_for_data_channels); |
| |
| // All of these signals are fired on the signaling thread. |
| |
| // If any transport failed => failed, |
| // Else if all completed => completed, |
| // Else if all connected => connected, |
| // Else => connecting |
| sigslot::signal1<cricket::IceConnectionState> SignalIceConnectionState; |
| |
| sigslot::signal1<PeerConnectionInterface::PeerConnectionState> |
| SignalConnectionState; |
| sigslot::signal1<PeerConnectionInterface::IceConnectionState> |
| SignalStandardizedIceConnectionState; |
| |
| // If all transports done gathering => complete, |
| // Else if any are gathering => gathering, |
| // Else => new |
| sigslot::signal1<cricket::IceGatheringState> SignalIceGatheringState; |
| |
| // (mid, candidates) |
| sigslot::signal2<const std::string&, const std::vector<cricket::Candidate>&> |
| SignalIceCandidatesGathered; |
| |
| sigslot::signal1<const std::vector<cricket::Candidate>&> |
| SignalIceCandidatesRemoved; |
| |
| sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError; |
| |
| sigslot::signal<> SignalMediaTransportStateChanged; |
| |
| private: |
| RTCError ApplyDescription_n(bool local, |
| SdpType type, |
| const cricket::SessionDescription* description); |
| RTCError ValidateAndMaybeUpdateBundleGroup( |
| bool local, |
| SdpType type, |
| const cricket::SessionDescription* description); |
| RTCError ValidateContent(const cricket::ContentInfo& content_info); |
| |
| void HandleRejectedContent(const cricket::ContentInfo& content_info, |
| const cricket::SessionDescription* description); |
| bool HandleBundledContent(const cricket::ContentInfo& content_info); |
| |
| bool SetTransportForMid(const std::string& mid, |
| cricket::JsepTransport* jsep_transport); |
| void RemoveTransportForMid(const std::string& mid); |
| |
| cricket::JsepTransportDescription CreateJsepTransportDescription( |
| cricket::ContentInfo content_info, |
| cricket::TransportInfo transport_info, |
| const std::vector<int>& encrypted_extension_ids, |
| int rtp_abs_sendtime_extn_id); |
| |
| absl::optional<std::string> bundled_mid() const { |
| absl::optional<std::string> bundled_mid; |
| if (bundle_group_ && bundle_group_->FirstContentName()) { |
| bundled_mid = *(bundle_group_->FirstContentName()); |
| } |
| return bundled_mid; |
| } |
| |
| bool IsBundled(const std::string& mid) const { |
| return bundle_group_ && bundle_group_->HasContentName(mid); |
| } |
| |
| bool ShouldUpdateBundleGroup(SdpType type, |
| const cricket::SessionDescription* description); |
| |
| std::vector<int> MergeEncryptedHeaderExtensionIdsForBundle( |
| const cricket::SessionDescription* description); |
| std::vector<int> GetEncryptedHeaderExtensionIds( |
| const cricket::ContentInfo& content_info); |
| |
| int GetRtpAbsSendTimeHeaderExtensionId( |
| const cricket::ContentInfo& content_info); |
| |
| // This method takes the BUNDLE group into account. If the JsepTransport is |
| // destroyed because of BUNDLE, it would return the transport which other |
| // transports are bundled on (In current implementation, it is the first |
| // content in the BUNDLE group). |
| const cricket::JsepTransport* GetJsepTransportForMid( |
| const std::string& mid) const; |
| cricket::JsepTransport* GetJsepTransportForMid(const std::string& mid); |
| |
| // Get the JsepTransport without considering the BUNDLE group. Return nullptr |
| // if the JsepTransport is destroyed. |
| const cricket::JsepTransport* GetJsepTransportByName( |
| const std::string& transport_name) const; |
| cricket::JsepTransport* GetJsepTransportByName( |
| const std::string& transport_name); |
| |
| // Creates jsep transport. Noop if transport is already created. |
| // Transport is created either during SetLocalDescription (|local| == true) or |
| // during SetRemoteDescription (|local| == false). Passing |local| helps to |
| // differentiate initiator (caller) from answerer (callee). |
| RTCError MaybeCreateJsepTransport( |
| bool local, |
| const cricket::ContentInfo& content_info, |
| const cricket::SessionDescription& description); |
| |
| // Creates media transport if config wants to use it, and pre-shared key is |
| // provided in content info. It modifies the config to disable media transport |
| // if pre-shared key is not provided. |
| std::unique_ptr<webrtc::MediaTransportInterface> MaybeCreateMediaTransport( |
| const cricket::ContentInfo& content_info, |
| const cricket::SessionDescription& description, |
| bool local, |
| cricket::IceTransportInternal* ice_transport); |
| void MaybeDestroyJsepTransport(const std::string& mid); |
| void DestroyAllJsepTransports_n(); |
| |
| void SetIceRole_n(cricket::IceRole ice_role); |
| |
| cricket::IceRole DetermineIceRole( |
| cricket::JsepTransport* jsep_transport, |
| const cricket::TransportInfo& transport_info, |
| SdpType type, |
| bool local); |
| |
| std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport( |
| std::unique_ptr<cricket::IceTransportInternal> ice); |
| std::unique_ptr<cricket::IceTransportInternal> CreateIceTransport( |
| const std::string transport_name, |
| bool rtcp); |
| |
| std::unique_ptr<webrtc::RtpTransport> CreateUnencryptedRtpTransport( |
| const std::string& transport_name, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport); |
| std::unique_ptr<webrtc::SrtpTransport> CreateSdesTransport( |
| const std::string& transport_name, |
| cricket::DtlsTransportInternal* rtp_dtls_transport, |
| cricket::DtlsTransportInternal* rtcp_dtls_transport); |
| std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport( |
| const std::string& transport_name, |
| cricket::DtlsTransportInternal* rtp_dtls_transport, |
| cricket::DtlsTransportInternal* rtcp_dtls_transport); |
| |
| // Collect all the DtlsTransports, including RTP and RTCP, from the |
| // JsepTransports. JsepTransportController can iterate all the DtlsTransports |
| // and update the aggregate states. |
| std::vector<cricket::DtlsTransportInternal*> GetDtlsTransports(); |
| |
| // Handlers for signals from Transport. |
| void OnTransportWritableState_n(rtc::PacketTransportInternal* transport); |
| void OnTransportReceivingState_n(rtc::PacketTransportInternal* transport); |
| void OnTransportGatheringState_n(cricket::IceTransportInternal* transport); |
| void OnTransportCandidateGathered_n(cricket::IceTransportInternal* transport, |
| const cricket::Candidate& candidate); |
| void OnTransportCandidatesRemoved_n(cricket::IceTransportInternal* transport, |
| const cricket::Candidates& candidates); |
| void OnTransportRoleConflict_n(cricket::IceTransportInternal* transport); |
| void OnTransportStateChanged_n(cricket::IceTransportInternal* transport); |
| void OnMediaTransportStateChanged_n(); |
| |
| void UpdateAggregateStates_n(); |
| |
| void OnDtlsHandshakeError(rtc::SSLHandshakeError error); |
| |
| rtc::Thread* const signaling_thread_ = nullptr; |
| rtc::Thread* const network_thread_ = nullptr; |
| cricket::PortAllocator* const port_allocator_ = nullptr; |
| AsyncResolverFactory* const async_resolver_factory_ = nullptr; |
| |
| std::map<std::string, std::unique_ptr<cricket::JsepTransport>> |
| jsep_transports_by_name_; |
| // This keeps track of the mapping between media section |
| // (BaseChannel/SctpTransport) and the JsepTransport underneath. |
| std::map<std::string, cricket::JsepTransport*> mid_to_transport_; |
| |
| // Aggregate states for Transports. |
| // standardized_ice_connection_state_ is intended to replace |
| // ice_connection_state, see bugs.webrtc.org/9308 |
| cricket::IceConnectionState ice_connection_state_ = |
| cricket::kIceConnectionConnecting; |
| PeerConnectionInterface::IceConnectionState |
| standardized_ice_connection_state_ = |
| PeerConnectionInterface::kIceConnectionNew; |
| PeerConnectionInterface::PeerConnectionState combined_connection_state_ = |
| PeerConnectionInterface::PeerConnectionState::kNew; |
| cricket::IceGatheringState ice_gathering_state_ = cricket::kIceGatheringNew; |
| |
| Config config_; |
| // Determines if Config::media_transport_factory should be used |
| // to create a media transport. (when falling back to RTP this may be false). |
| // This is a prerequisite, but is not sufficient to create media transport |
| // (the factory needs to be provided in the config, and config must allow for |
| // media transport). |
| bool is_media_transport_factory_enabled_ = true; |
| const cricket::SessionDescription* local_desc_ = nullptr; |
| const cricket::SessionDescription* remote_desc_ = nullptr; |
| absl::optional<bool> initial_offerer_; |
| |
| absl::optional<cricket::ContentGroup> bundle_group_; |
| |
| cricket::IceConfig ice_config_; |
| cricket::IceRole ice_role_ = cricket::ICEROLE_CONTROLLING; |
| uint64_t ice_tiebreaker_ = rtc::CreateRandomId64(); |
| rtc::scoped_refptr<rtc::RTCCertificate> certificate_; |
| rtc::AsyncInvoker invoker_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransportController); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_JSEP_TRANSPORT_CONTROLLER_H_ |