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* 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.
#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<> {
// Used when the RtpTransport/DtlsTransport of the m= section is changed
// because the section is rejected or BUNDLE is enabled.
class Observer {
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::RtcpMuxPolicy rtcp_mux_policy =
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;
// 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);
// 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.
// 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;
// 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>&>
sigslot::signal1<const std::vector<cricket::Candidate>&>
sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
sigslot::signal<> SignalMediaTransportStateChanged;
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 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);
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>>
// 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
cricket::IceConnectionState ice_connection_state_ =
standardized_ice_connection_state_ =
PeerConnectionInterface::PeerConnectionState combined_connection_state_ =
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.
// 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_;
} // namespace webrtc