pc/jsep_transport_controller.h

/*
 *  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 <stdint.h>

#include <functional>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>

#include "absl/types/optional.h"
#include "api/async_resolver_factory.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/ice_transport_factory.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/scoped_refptr.h"
#include "api/transport/data_channel_transport_interface.h"
#include "api/transport/sctp_transport_factory_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/dtls_transport_factory.h"
#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_transport_channel.h"
#include "p2p/base/packet_transport_internal.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.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/rtp_transport_internal.h"
#include "pc/sctp_transport.h"
#include "pc/session_description.h"
#include "pc/srtp_transport.h"
#include "pc/transport_stats.h"
#include "rtc_base/async_invoker.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/helpers.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.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.
    //
    // If a data channel transport must be negotiated, |data_channel_transport|
    // and |negotiation_state| indicate negotiation status.  If
    // |data_channel_transport| is null, the data channel transport should not
    // be used.  Otherwise, the value is a pointer to the transport to be used
    // for data channels on |mid|, if any.
    //
    // The observer should not send data on |data_channel_transport| until
    // |negotiation_state| is provisional or final.  It should not delete
    // |data_channel_transport| or any fallback transport until
    // |negotiation_state| is final.
    virtual bool OnTransportChanged(
        const std::string& mid,
        RtpTransportInternal* rtp_transport,
        rtc::scoped_refptr<DtlsTransport> dtls_transport,
        DataChannelTransportInterface* data_channel_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.
    webrtc::IceTransportFactory* ice_transport_factory = nullptr;
    cricket::DtlsTransportFactory* dtls_transport_factory = nullptr;
    Observer* transport_observer = nullptr;
    // Must be provided and valid for the lifetime of the
    // JsepTransportController instance.
    std::function<void(const rtc::CopyOnWriteBuffer& packet,
                       int64_t packet_time_us)>
        rtcp_handler;
    // Initial value for whether DtlsTransport reset causes a reset
    // of SRTP parameters.
    bool active_reset_srtp_params = false;
    RtcEventLog* event_log = nullptr;

    // Factory for SCTP transports.
    SctpTransportFactoryInterface* sctp_factory = nullptr;
    std::function<void(const rtc::SSLHandshakeError)> on_dtls_handshake_error_;
  };

  // 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);
  rtc::scoped_refptr<SctpTransport> GetSctpTransport(
      const std::string& mid) const;

  DataChannelTransportInterface* GetDataChannelTransport(
      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);

  // For now the rollback only removes mid to transport mappings
  // and deletes unused transports, but doesn't consider anything more complex.
  void RollbackTransports();

  // F: void(const std::string&, const std::vector<cricket::Candidate>&)
  template <typename F>
  void SubscribeIceCandidateGathered(F&& callback) {
    signal_ice_candidates_gathered_.AddReceiver(std::forward<F>(callback));
  }

  // F: void(cricket::IceConnectionState)
  template <typename F>
  void SubscribeIceConnectionState(F&& callback) {
    signal_ice_connection_state_.AddReceiver(std::forward<F>(callback));
  }

  // F: void(PeerConnectionInterface::PeerConnectionState)
  template <typename F>
  void SubscribeConnectionState(F&& callback) {
    signal_connection_state_.AddReceiver(std::forward<F>(callback));
  }

  // F: void(PeerConnectionInterface::IceConnectionState)
  template <typename F>
  void SubscribeStandardizedIceConnectionState(F&& callback) {
    signal_standardized_ice_connection_state_.AddReceiver(
        std::forward<F>(callback));
  }

  // F: void(cricket::IceGatheringState)
  template <typename F>
  void SubscribeIceGatheringState(F&& callback) {
    signal_ice_gathering_state_.AddReceiver(std::forward<F>(callback));
  }

  // F: void(const cricket::IceCandidateErrorEvent&)
  template <typename F>
  void SubscribeIceCandidateError(F&& callback) {
    signal_ice_candidate_error_.AddReceiver(std::forward<F>(callback));
  }

  // F: void(const std::vector<cricket::Candidate>&)
  template <typename F>
  void SubscribeIceCandidatesRemoved(F&& callback) {
    signal_ice_candidates_removed_.AddReceiver(std::forward<F>(callback));
  }

  // F: void(const cricket::CandidatePairChangeEvent&)
  template <typename F>
  void SubscribeIceCandidatePairChanged(F&& callback) {
    signal_ice_candidate_pair_changed_.AddReceiver(std::forward<F>(callback));
  }

 private:
  // All of these callbacks are fired on the signaling thread.

  // If any transport failed => failed,
  // Else if all completed => completed,
  // Else if all connected => connected,
  // Else => connecting
  CallbackList<cricket::IceConnectionState> signal_ice_connection_state_;

  CallbackList<PeerConnectionInterface::PeerConnectionState>
      signal_connection_state_;

  CallbackList<PeerConnectionInterface::IceConnectionState>
      signal_standardized_ice_connection_state_;

  // If all transports done gathering => complete,
  // Else if any are gathering => gathering,
  // Else => new
  CallbackList<cricket::IceGatheringState> signal_ice_gathering_state_;

  // [mid, candidates]
  CallbackList<const std::string&, const std::vector<cricket::Candidate>&>
      signal_ice_candidates_gathered_;

  CallbackList<const cricket::IceCandidateErrorEvent&>
      signal_ice_candidate_error_;

  CallbackList<const std::vector<cricket::Candidate>&>
      signal_ice_candidates_removed_;

  CallbackList<const cricket::CandidatePairChangeEvent&>
      signal_ice_candidate_pair_changed_;

  RTCError ApplyDescription_n(bool local,
                              SdpType type,
                              const cricket::SessionDescription* description)
      RTC_RUN_ON(network_thread_);
  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)
      RTC_RUN_ON(network_thread_);
  bool HandleBundledContent(const cricket::ContentInfo& content_info)
      RTC_RUN_ON(network_thread_);

  bool SetTransportForMid(const std::string& mid,
                          cricket::JsepTransport* jsep_transport);
  void RemoveTransportForMid(const std::string& mid);

  cricket::JsepTransportDescription CreateJsepTransportDescription(
      const cricket::ContentInfo& content_info,
      const 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)
      RTC_RUN_ON(network_thread_);

  void MaybeDestroyJsepTransport(const std::string& mid)
      RTC_RUN_ON(network_thread_);
  void DestroyAllJsepTransports_n() RTC_RUN_ON(network_thread_);

  void SetIceRole_n(cricket::IceRole ice_role) RTC_RUN_ON(network_thread_);

  cricket::IceRole DetermineIceRole(
      cricket::JsepTransport* jsep_transport,
      const cricket::TransportInfo& transport_info,
      SdpType type,
      bool local);

  std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
      const cricket::ContentInfo& content_info,
      cricket::IceTransportInternal* ice);
  rtc::scoped_refptr<webrtc::IceTransportInterface> 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)
      RTC_RUN_ON(network_thread_);
  void OnTransportReceivingState_n(rtc::PacketTransportInternal* transport)
      RTC_RUN_ON(network_thread_);
  void OnTransportGatheringState_n(cricket::IceTransportInternal* transport)
      RTC_RUN_ON(network_thread_);
  void OnTransportCandidateGathered_n(cricket::IceTransportInternal* transport,
                                      const cricket::Candidate& candidate)
      RTC_RUN_ON(network_thread_);
  void OnTransportCandidateError_n(cricket::IceTransportInternal* transport,
                                   const cricket::IceCandidateErrorEvent& event)
      RTC_RUN_ON(network_thread_);
  void OnTransportCandidatesRemoved_n(cricket::IceTransportInternal* transport,
                                      const cricket::Candidates& candidates)
      RTC_RUN_ON(network_thread_);
  void OnTransportRoleConflict_n(cricket::IceTransportInternal* transport)
      RTC_RUN_ON(network_thread_);
  void OnTransportStateChanged_n(cricket::IceTransportInternal* transport)
      RTC_RUN_ON(network_thread_);
  void OnTransportCandidatePairChanged_n(
      const cricket::CandidatePairChangeEvent& event)
      RTC_RUN_ON(network_thread_);
  void UpdateAggregateStates_n() RTC_RUN_ON(network_thread_);

  void OnRtcpPacketReceived_n(rtc::CopyOnWriteBuffer* packet,
                              int64_t packet_time_us)
      RTC_RUN_ON(network_thread_);

  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_;
  // Keep track of mids that have been mapped to transports. Used for rollback.
  std::vector<std::string> pending_mids_ RTC_GUARDED_BY(network_thread_);
  // 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;

  const Config config_;
  bool active_reset_srtp_params_ RTC_GUARDED_BY(network_thread_);

  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_