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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.
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/jsep.h"
#include "api/media_transport_interface.h"
#include "p2p/base/dtlstransport.h"
#include "p2p/base/p2pconstants.h"
#include "p2p/base/transportinfo.h"
#include "pc/dtlssrtptransport.h"
#include "pc/rtcpmuxfilter.h"
#include "pc/rtptransport.h"
#include "pc/sessiondescription.h"
#include "pc/srtpfilter.h"
#include "pc/srtptransport.h"
#include "pc/transportstats.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/messagequeue.h"
#include "rtc_base/rtccertificate.h"
#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace cricket {
class DtlsTransportInternal;
struct 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);
JsepTransportDescription(const JsepTransportDescription& from);
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;
// 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 webrtc::MediaTransportStateCallback {
// |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.
// |media_trasport| is optional (experimental). If available it will be used
// to send / receive encoded audio and video frames instead of RTP.
// Currently |media_transport| can co-exist with RTP / RTCP transports.
const std::string& mid,
const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
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<DtlsTransportInternal> rtp_dtls_transport,
std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
std::unique_ptr<webrtc::MediaTransportInterface> media_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) {
local_certificate_ = local_certificate;
// Return the local certificate provided by SetLocalCertificate.
rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const {
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 { 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;
// TODO(deadbeef): Make this const. See comment in transportcontroller.h.
bool GetStats(TransportStats* stats);
const JsepTransportDescription* local_description() const {
return local_description_.get();
const JsepTransportDescription* remote_description() const {
return remote_description_.get();
webrtc::RtpTransportInternal* rtp_transport() const {
if (dtls_srtp_transport_) {
return dtls_srtp_transport_.get();
} else if (sdes_transport_) {
return sdes_transport_.get();
} else {
return unencrypted_rtp_transport_.get();
DtlsTransportInternal* rtp_dtls_transport() const {
return rtp_dtls_transport_.get();
DtlsTransportInternal* rtcp_dtls_transport() const {
return rtcp_dtls_transport_.get();
// Returns media transport, if available.
// Note that media transport is owned by jseptransport and the pointer
// to media transport will becomes invalid after destruction of jseptransport.
webrtc::MediaTransportInterface* media_transport() const {
return media_transport_.get();
// Returns the latest media transport state.
webrtc::MediaTransportState media_transport_state() const {
return media_transport_state_;
// 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;
// This is signaled for changes in |media_transport_| state.
sigslot::signal<> SignalMediaTransportStateChanged;
// 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);
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);
// Invoked whenever the state of the media transport changes.
void OnStateChanged(webrtc::MediaTransportState state) override;
const std::string mid_;
// needs-ice-restart bit as described in JSEP.
bool needs_ice_restart_ = false;
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
std::unique_ptr<JsepTransportDescription> local_description_;
std::unique_ptr<JsepTransportDescription> remote_description_;
// 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_;
std::unique_ptr<webrtc::SrtpTransport> sdes_transport_;
std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport_;
std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport_;
std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport_;
SrtpFilter sdes_negotiator_;
RtcpMuxFilter rtcp_mux_negotiator_;
// Cache the encrypted header extension IDs for SDES negoitation.
absl::optional<std::vector<int>> send_extension_ids_;
absl::optional<std::vector<int>> recv_extension_ids_;
// Optional media transport (experimental).
std::unique_ptr<webrtc::MediaTransportInterface> media_transport_;
// If |media_transport_| is provided, this variable represents the state of
// media transport.
webrtc::MediaTransportState media_transport_state_ =
} // namespace cricket