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

#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/media_transport_config.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;

    // Whether an RtpMediaTransport should be created as default, when no
    // MediaTransportFactory is provided.
    bool use_rtp_media_transport = false;

    // Use encrypted datagram transport to send packets.
    bool use_datagram_transport = 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);

  MediaTransportConfig GetMediaTransportConfig(const std::string& mid) const;

  MediaTransportInterface* GetMediaTransportForDataChannel(
      const std::string& mid) const;

  // TODO(sukhanov): Deprecate, return only config.
  MediaTransportInterface* GetMediaTransport(const std::string& mid) const {
    return GetMediaTransportConfig(mid).media_transport;
  }

  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,
                                 bool use_datagram_transport);

  // If media transport is present enabled and supported,
  // when this method is called, it creates a media transport and generates its
  // offer. The new offer is then returned, and the created media transport will
  // subsequently be used.
  absl::optional<cricket::SessionDescription::MediaTransportSetting>
  GenerateOrGetLastMediaTransportOffer();

  // Gets the transport parameters for the transport identified by |mid|.
  // If |mid| is bundled, returns the parameters for the bundled transport.
  // If the transport for |mid| has not been created yet, it may be allocated in
  // order to generate transport parameters.
  absl::optional<cricket::OpaqueTransportParameters> GetTransportParameters(
      const std::string& mid);

  // 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 cricket::IceCandidateErrorEvent&>
      SignalIceCandidateError;

  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(
      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);

  // Creates media transport if config wants to use it, and a=x-mt line is
  // present for the current media transport. Returned MediaTransportInterface
  // is not connected, and must be connected to ICE. You must call
  // |GenerateOrGetLastMediaTransportOffer| on the caller before calling
  // MaybeCreateMediaTransport.
  std::unique_ptr<webrtc::MediaTransportInterface> MaybeCreateMediaTransport(
      const cricket::ContentInfo& content_info,
      const cricket::SessionDescription& description,
      bool local);

  // Creates datagram transport if config wants to use it, and a=x-mt line is
  // present for the current media transport. Returned
  // DatagramTransportInterface is not connected, and must be connected to ICE.
  // You must call |GenerateOrGetLastMediaTransportOffer| on the caller before
  // calling MaybeCreateDatagramTransport.
  std::unique_ptr<webrtc::DatagramTransportInterface>
  MaybeCreateDatagramTransport(const cricket::ContentInfo& content_info,
                               const cricket::SessionDescription& description,
                               bool local);

  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(
      cricket::IceTransportInternal* ice,
      DatagramTransportInterface* datagram_transport);
  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 OnTransportCandidateError_n(
      cricket::IceTransportInternal* transport,
      const cricket::IceCandidateErrorEvent& event);
  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_;

  // Early on in the call we don't know if media transport is going to be used,
  // but we need to get the server-supported parameters to add to an SDP.
  // This server media transport will be promoted to the used media transport
  // after the local description is set, and the ownership will be transferred
  // to the actual JsepTransport.
  // This "offer" media transport is not created if it's done on the party that
  // provides answer. This offer media transport is only created once at the
  // beginning of the connection, and never again.
  std::unique_ptr<MediaTransportInterface> offer_media_transport_ = nullptr;

  // Contains the offer of the |offer_media_transport_|, in case if it needs to
  // be repeated.
  absl::optional<cricket::SessionDescription::MediaTransportSetting>
      media_transport_offer_settings_;

  // Early on in the call we don't know if datagram transport is going to be
  // used, but we need to get the server-supported parameters to add to an SDP.
  // This server datagram transport will be promoted to the used datagram
  // transport after the local description is set, and the ownership will be
  // transferred to the actual JsepTransport. This "offer" datagram transport is
  // not created if it's done on the party that provides answer. This offer
  // datagram transport is only created once at the beginning of the connection,
  // and never again.
  std::unique_ptr<DatagramTransportInterface> offer_datagram_transport_ =
      nullptr;

  // Contains the offer of the |offer_datagram_transport_|, in case if it needs
  // to be repeated.
  absl::optional<cricket::SessionDescription::MediaTransportSetting>
      datagram_transport_offer_settings_;

  // When the new offer is regenerated (due to upgrade), we don't want to
  // re-create media transport. New streams might be created; but media
  // transport stays the same. This flag prevents re-creation of the transport
  // on the offerer.
  // The first media transport is created in jsep transport controller as the
  // |offer_media_transport_|, and then the ownership is moved to the
  // appropriate JsepTransport, at which point |offer_media_transport_| is
  // zeroed out. On the callee (answerer), the first media transport is not even
  // assigned to |offer_media_transport_|. Both offerer and answerer can
  // recreate the Offer (e.g. after adding streams in Plan B), and so we want to
  // prevent recreation of the media transport when that happens.
  bool media_transport_created_once_ = false;

  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_