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/*
* Copyright 2018 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_H_
#define PC_JSEP_TRANSPORT_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/transport/datagram_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/transport_info.h"
#include "pc/composite_data_channel_transport.h"
#include "pc/composite_rtp_transport.h"
#include "pc/dtls_srtp_transport.h"
#include "pc/dtls_transport.h"
#include "pc/rtcp_mux_filter.h"
#include "pc/rtp_transport.h"
#include "pc/sctp_transport.h"
#include "pc/session_description.h"
#include "pc/srtp_filter.h"
#include "pc/srtp_transport.h"
#include "pc/transport_stats.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread_checker.h"
namespace cricket {
class DtlsTransportInternal;
struct JsepTransportDescription {
public:
JsepTransportDescription();
JsepTransportDescription(
bool rtcp_mux_enabled,
const std::vector<CryptoParams>& cryptos,
const std::vector<int>& encrypted_header_extension_ids,
int rtp_abs_sendtime_extn_id,
const TransportDescription& transport_description,
absl::optional<std::string> media_alt_protocol,
absl::optional<std::string> data_alt_protocol);
JsepTransportDescription(const JsepTransportDescription& from);
~JsepTransportDescription();
JsepTransportDescription& operator=(const JsepTransportDescription& from);
bool rtcp_mux_enabled = true;
std::vector<CryptoParams> cryptos;
std::vector<int> encrypted_header_extension_ids;
int rtp_abs_sendtime_extn_id = -1;
// TODO(zhihuang): Add the ICE and DTLS related variables and methods from
// TransportDescription and remove this extra layer of abstraction.
TransportDescription transport_desc;
// Alt-protocols that apply to this JsepTransport. Presence indicates a
// request to use an alternative protocol for media and/or data. The
// alt-protocol is handled by a datagram transport. If one or both of these
// values are present, JsepTransport will attempt to negotiate use of the
// datagram transport for media and/or data.
absl::optional<std::string> media_alt_protocol;
absl::optional<std::string> data_alt_protocol;
};
// Helper class used by JsepTransportController that processes
// TransportDescriptions. A TransportDescription represents the
// transport-specific properties of an SDP m= section, processed according to
// JSEP. Each transport consists of DTLS and ICE transport channels for RTP
// (and possibly RTCP, if rtcp-mux isn't used).
//
// On Threading: JsepTransport performs work solely on the network thread, and
// so its methods should only be called on the network thread.
class JsepTransport : public sigslot::has_slots<> {
public:
// |mid| is just used for log statements in order to identify the Transport.
// Note that |local_certificate| is allowed to be null since a remote
// description may be set before a local certificate is generated.
JsepTransport(
const std::string& mid,
const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport,
rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport,
std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport,
std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
std::unique_ptr<webrtc::RtpTransportInternal> datagram_rtp_transport,
std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
std::unique_ptr<SctpTransportInternal> sctp_transport,
std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport,
webrtc::DataChannelTransportInterface* data_channel_transport);
~JsepTransport() override;
// Returns the MID of this transport. This is only used for logging.
const std::string& mid() const { return mid_; }
// Must be called before applying local session description.
// Needed in order to verify the local fingerprint.
void SetLocalCertificate(
const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate) {
RTC_DCHECK_RUN_ON(network_thread_);
local_certificate_ = local_certificate;
}
// Return the local certificate provided by SetLocalCertificate.
rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const {
RTC_DCHECK_RUN_ON(network_thread_);
return local_certificate_;
}
webrtc::RTCError SetLocalJsepTransportDescription(
const JsepTransportDescription& jsep_description,
webrtc::SdpType type);
// Set the remote TransportDescription to be used by DTLS and ICE channels
// that are part of this Transport.
webrtc::RTCError SetRemoteJsepTransportDescription(
const JsepTransportDescription& jsep_description,
webrtc::SdpType type);
webrtc::RTCError AddRemoteCandidates(const Candidates& candidates);
// 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.
//
// This and the below method can be called safely from any thread as long as
// SetXTransportDescription is not in progress.
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).
bool needs_ice_restart() const {
rtc::CritScope scope(&accessor_lock_);
return needs_ice_restart_;
}
// Returns role if negotiated, or empty absl::optional if it hasn't been
// negotiated yet.
absl::optional<rtc::SSLRole> GetDtlsRole() const;
absl::optional<OpaqueTransportParameters> GetTransportParameters() const;
// TODO(deadbeef): Make this const. See comment in transportcontroller.h.
bool GetStats(TransportStats* stats);
const JsepTransportDescription* local_description() const {
RTC_DCHECK_RUN_ON(network_thread_);
return local_description_.get();
}
const JsepTransportDescription* remote_description() const {
RTC_DCHECK_RUN_ON(network_thread_);
return remote_description_.get();
}
webrtc::RtpTransportInternal* rtp_transport() const {
rtc::CritScope scope(&accessor_lock_);
if (composite_rtp_transport_) {
return composite_rtp_transport_.get();
} else if (datagram_rtp_transport_) {
return datagram_rtp_transport_.get();
} else {
return default_rtp_transport();
}
}
const DtlsTransportInternal* rtp_dtls_transport() const {
rtc::CritScope scope(&accessor_lock_);
if (rtp_dtls_transport_) {
return rtp_dtls_transport_->internal();
} else {
return nullptr;
}
}
DtlsTransportInternal* rtp_dtls_transport() {
rtc::CritScope scope(&accessor_lock_);
if (rtp_dtls_transport_) {
return rtp_dtls_transport_->internal();
} else {
return nullptr;
}
}
const DtlsTransportInternal* rtcp_dtls_transport() const {
rtc::CritScope scope(&accessor_lock_);
if (rtcp_dtls_transport_) {
return rtcp_dtls_transport_->internal();
} else {
return nullptr;
}
}
DtlsTransportInternal* rtcp_dtls_transport() {
rtc::CritScope scope(&accessor_lock_);
if (rtcp_dtls_transport_) {
return rtcp_dtls_transport_->internal();
} else {
return nullptr;
}
}
rtc::scoped_refptr<webrtc::DtlsTransport> RtpDtlsTransport() {
rtc::CritScope scope(&accessor_lock_);
return rtp_dtls_transport_;
}
rtc::scoped_refptr<webrtc::SctpTransport> SctpTransport() const {
rtc::CritScope scope(&accessor_lock_);
return sctp_transport_;
}
webrtc::DataChannelTransportInterface* data_channel_transport() const {
rtc::CritScope scope(&accessor_lock_);
if (composite_data_channel_transport_) {
return composite_data_channel_transport_.get();
} else if (sctp_data_channel_transport_) {
return sctp_data_channel_transport_.get();
}
return data_channel_transport_;
}
// Returns datagram transport, if available.
webrtc::DatagramTransportInterface* datagram_transport() const {
rtc::CritScope scope(&accessor_lock_);
return datagram_transport_.get();
}
// This is signaled when RTCP-mux becomes active and
// |rtcp_dtls_transport_| is destroyed. The JsepTransportController will
// handle the signal and update the aggregate transport states.
sigslot::signal<> SignalRtcpMuxActive;
// Signals that a data channel transport was negotiated and may be used to
// send data. The first parameter is |this|. The second parameter is the
// transport that was negotiated, or null if negotiation rejected the data
// channel transport. The third parameter (bool) indicates whether the
// negotiation was provisional or final. If true, it is provisional, if
// false, it is final.
sigslot::signal2<JsepTransport*, webrtc::DataChannelTransportInterface*>
SignalDataChannelTransportNegotiated;
// TODO(deadbeef): The methods below are only public for testing. Should make
// them utility functions or objects so they can be tested independently from
// this class.
// Returns an error if the certificate's identity does not match the
// fingerprint, or either is NULL.
webrtc::RTCError VerifyCertificateFingerprint(
const rtc::RTCCertificate* certificate,
const rtc::SSLFingerprint* fingerprint) const;
void SetActiveResetSrtpParams(bool active_reset_srtp_params);
private:
bool SetRtcpMux(bool enable, webrtc::SdpType type, ContentSource source);
void ActivateRtcpMux();
bool SetSdes(const std::vector<CryptoParams>& cryptos,
const std::vector<int>& encrypted_extension_ids,
webrtc::SdpType type,
ContentSource source);
// Negotiates and sets the DTLS parameters based on the current local and
// remote transport description, such as the DTLS role to use, and whether
// DTLS should be activated.
//
// Called when an answer TransportDescription is applied.
webrtc::RTCError NegotiateAndSetDtlsParameters(
webrtc::SdpType local_description_type);
// Negotiates the DTLS role based off the offer and answer as specified by
// RFC 4145, section-4.1. Returns an RTCError if role cannot be determined
// from the local description and remote description.
webrtc::RTCError NegotiateDtlsRole(
webrtc::SdpType local_description_type,
ConnectionRole local_connection_role,
ConnectionRole remote_connection_role,
absl::optional<rtc::SSLRole>* negotiated_dtls_role);
// Pushes down the ICE parameters from the local description, such
// as the ICE ufrag and pwd.
void SetLocalIceParameters(IceTransportInternal* ice);
// Pushes down the ICE parameters from the remote description.
void SetRemoteIceParameters(IceTransportInternal* ice);
// Pushes down the DTLS parameters obtained via negotiation.
webrtc::RTCError SetNegotiatedDtlsParameters(
DtlsTransportInternal* dtls_transport,
absl::optional<rtc::SSLRole> dtls_role,
rtc::SSLFingerprint* remote_fingerprint);
bool GetTransportStats(DtlsTransportInternal* dtls_transport,
TransportStats* stats);
// Deactivates, signals removal, and deletes |composite_rtp_transport_| if the
// current state of negotiation is sufficient to determine which rtp_transport
// and data channel transport to use.
void NegotiateDatagramTransport(webrtc::SdpType type)
RTC_RUN_ON(network_thread_);
// Returns the default (non-datagram) rtp transport, if any.
webrtc::RtpTransportInternal* default_rtp_transport() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(accessor_lock_) {
if (dtls_srtp_transport_) {
return dtls_srtp_transport_.get();
} else if (sdes_transport_) {
return sdes_transport_.get();
} else if (unencrypted_rtp_transport_) {
return unencrypted_rtp_transport_.get();
} else {
return nullptr;
}
}
// Owning thread, for safety checks
const rtc::Thread* const network_thread_;
// Critical scope for fields accessed off-thread
// TODO(https://bugs.webrtc.org/10300): Stop doing this.
rtc::CriticalSection accessor_lock_;
const std::string mid_;
// needs-ice-restart bit as described in JSEP.
bool needs_ice_restart_ RTC_GUARDED_BY(accessor_lock_) = false;
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_
RTC_GUARDED_BY(network_thread_);
std::unique_ptr<JsepTransportDescription> local_description_
RTC_GUARDED_BY(network_thread_);
std::unique_ptr<JsepTransportDescription> remote_description_
RTC_GUARDED_BY(network_thread_);
// Ice transport which may be used by any of upper-layer transports (below).
// Owned by JsepTransport and guaranteed to outlive the transports below.
const rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport_;
const rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport_;
// To avoid downcasting and make it type safe, keep three unique pointers for
// different SRTP mode and only one of these is non-nullptr.
std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport_
RTC_GUARDED_BY(accessor_lock_);
std::unique_ptr<webrtc::SrtpTransport> sdes_transport_
RTC_GUARDED_BY(accessor_lock_);
std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport_
RTC_GUARDED_BY(accessor_lock_);
// If multiple RTP transports are in use, |composite_rtp_transport_| will be
// passed to callers. This is only valid for offer-only, receive-only
// scenarios, as it is not possible for the composite to correctly choose
// which transport to use for sending.
std::unique_ptr<webrtc::CompositeRtpTransport> composite_rtp_transport_
RTC_GUARDED_BY(accessor_lock_);
rtc::scoped_refptr<webrtc::DtlsTransport> rtp_dtls_transport_
RTC_GUARDED_BY(accessor_lock_);
rtc::scoped_refptr<webrtc::DtlsTransport> rtcp_dtls_transport_
RTC_GUARDED_BY(accessor_lock_);
rtc::scoped_refptr<webrtc::DtlsTransport> datagram_dtls_transport_
RTC_GUARDED_BY(accessor_lock_);
std::unique_ptr<webrtc::DataChannelTransportInterface>
sctp_data_channel_transport_ RTC_GUARDED_BY(accessor_lock_);
rtc::scoped_refptr<webrtc::SctpTransport> sctp_transport_
RTC_GUARDED_BY(accessor_lock_);
SrtpFilter sdes_negotiator_ RTC_GUARDED_BY(network_thread_);
RtcpMuxFilter rtcp_mux_negotiator_ RTC_GUARDED_BY(network_thread_);
// Cache the encrypted header extension IDs for SDES negoitation.
absl::optional<std::vector<int>> send_extension_ids_
RTC_GUARDED_BY(network_thread_);
absl::optional<std::vector<int>> recv_extension_ids_
RTC_GUARDED_BY(network_thread_);
// Optional datagram transport (experimental).
std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport_
RTC_GUARDED_BY(accessor_lock_);
std::unique_ptr<webrtc::RtpTransportInternal> datagram_rtp_transport_
RTC_GUARDED_BY(accessor_lock_);
// Non-SCTP data channel transport. Set to |datagram_transport_| if that
// transport should be used for data chanels. Unset otherwise.
webrtc::DataChannelTransportInterface* data_channel_transport_
RTC_GUARDED_BY(accessor_lock_) = nullptr;
// Composite data channel transport, used during negotiation.
std::unique_ptr<webrtc::CompositeDataChannelTransport>
composite_data_channel_transport_ RTC_GUARDED_BY(accessor_lock_);
RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
};
} // namespace cricket
#endif // PC_JSEP_TRANSPORT_H_