Files
platform-external-webrtc/pc/jsep_transport_controller.h
Lahiru Ginnaliya Gamathige e99c68dd21 Replace one use of sigslot with RoboCaller
The eventual goal is to replace sigslot entirely, but we need to
  start small, tread carefully, and evaluate how it works out.
  Also add a few more RoboCaller unit tests to cover the types we
  now use with RoboCaller.

Change-Id: I9a5814d1668a37546ea484ca88ec9c2be1913d25
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184660
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32266}
2020-09-30 22:55:44 +00:00

402 lines
17 KiB
C++

/*
* 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/ice_transport_factory.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/dtls_transport_factory.h"
#include "p2p/base/p2p_transport_channel.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/robo_caller.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.
//
// 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;
bool active_reset_srtp_params = false;
RtcEventLog* event_log = nullptr;
// Factory for SCTP transports.
SctpTransportFactoryInterface* sctp_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);
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();
// 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
RoboCaller<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<const cricket::CandidatePairChangeEvent&>
SignalIceCandidatePairChanged;
sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
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);
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(
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);
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 OnTransportCandidatePairChanged_n(
const cricket::CandidatePairChangeEvent& event);
void UpdateAggregateStates_n();
void OnRtcpPacketReceived_n(rtc::CopyOnWriteBuffer* packet,
int64_t packet_time_us);
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;
Config config_;
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_