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320
TMessagesProj/jni/voip/webrtc/pc/jsep_transport.h
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320
TMessagesProj/jni/voip/webrtc/pc/jsep_transport.h
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/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_JSEP_TRANSPORT_H_
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#define PC_JSEP_TRANSPORT_H_
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#include <functional>
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/candidate.h"
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#include "api/ice_transport_interface.h"
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#include "api/jsep.h"
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#include "api/rtc_error.h"
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#include "api/scoped_refptr.h"
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#include "api/sequence_checker.h"
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#include "api/transport/data_channel_transport_interface.h"
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#include "media/sctp/sctp_transport_internal.h"
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#include "p2p/base/dtls_transport.h"
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#include "p2p/base/dtls_transport_internal.h"
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#include "p2p/base/ice_transport_internal.h"
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#include "p2p/base/p2p_constants.h"
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#include "p2p/base/transport_description.h"
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#include "p2p/base/transport_info.h"
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#include "pc/dtls_srtp_transport.h"
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#include "pc/dtls_transport.h"
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#include "pc/rtcp_mux_filter.h"
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#include "pc/rtp_transport.h"
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#include "pc/rtp_transport_internal.h"
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#include "pc/sctp_transport.h"
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#include "pc/session_description.h"
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#include "pc/srtp_transport.h"
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#include "pc/transport_stats.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/rtc_certificate.h"
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#include "rtc_base/ssl_fingerprint.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_annotations.h"
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namespace cricket {
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class DtlsTransportInternal;
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struct JsepTransportDescription {
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public:
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JsepTransportDescription();
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JsepTransportDescription(
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bool rtcp_mux_enabled,
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const std::vector<int>& encrypted_header_extension_ids,
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int rtp_abs_sendtime_extn_id,
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const TransportDescription& transport_description);
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JsepTransportDescription(const JsepTransportDescription& from);
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~JsepTransportDescription();
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JsepTransportDescription& operator=(const JsepTransportDescription& from);
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bool rtcp_mux_enabled = true;
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std::vector<int> encrypted_header_extension_ids;
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int rtp_abs_sendtime_extn_id = -1;
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// TODO(zhihuang): Add the ICE and DTLS related variables and methods from
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// TransportDescription and remove this extra layer of abstraction.
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TransportDescription transport_desc;
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};
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// Helper class used by JsepTransportController that processes
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// TransportDescriptions. A TransportDescription represents the
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// transport-specific properties of an SDP m= section, processed according to
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// JSEP. Each transport consists of DTLS and ICE transport channels for RTP
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// (and possibly RTCP, if rtcp-mux isn't used).
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//
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// On Threading: JsepTransport performs work solely on the network thread, and
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// so its methods should only be called on the network thread.
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class JsepTransport {
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public:
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// `mid` is just used for log statements in order to identify the Transport.
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// Note that `local_certificate` is allowed to be null since a remote
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// description may be set before a local certificate is generated.
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JsepTransport(
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const std::string& mid,
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const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
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rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport,
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rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport,
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std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport,
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std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
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std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
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std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
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std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
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std::unique_ptr<SctpTransportInternal> sctp_transport,
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std::function<void()> rtcp_mux_active_callback);
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~JsepTransport();
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JsepTransport(const JsepTransport&) = delete;
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JsepTransport& operator=(const JsepTransport&) = delete;
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// Returns the MID of this transport. This is only used for logging.
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const std::string& mid() const { return mid_; }
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// Must be called before applying local session description.
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// Needed in order to verify the local fingerprint.
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void SetLocalCertificate(
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const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate) {
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RTC_DCHECK_RUN_ON(network_thread_);
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local_certificate_ = local_certificate;
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}
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// Return the local certificate provided by SetLocalCertificate.
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rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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return local_certificate_;
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}
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webrtc::RTCError SetLocalJsepTransportDescription(
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const JsepTransportDescription& jsep_description,
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webrtc::SdpType type);
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// Set the remote TransportDescription to be used by DTLS and ICE channels
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// that are part of this Transport.
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webrtc::RTCError SetRemoteJsepTransportDescription(
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const JsepTransportDescription& jsep_description,
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webrtc::SdpType type);
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webrtc::RTCError AddRemoteCandidates(const Candidates& candidates);
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// Set the "needs-ice-restart" flag as described in JSEP. After the flag is
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// set, offers should generate new ufrags/passwords until an ICE restart
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// occurs.
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//
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// This and `needs_ice_restart()` must be called on the network thread.
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void SetNeedsIceRestartFlag();
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// Returns true if the ICE restart flag above was set, and no ICE restart has
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// occurred yet for this transport (by applying a local description with
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// changed ufrag/password).
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bool needs_ice_restart() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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return needs_ice_restart_;
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}
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// Returns role if negotiated, or empty absl::optional if it hasn't been
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// negotiated yet.
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absl::optional<rtc::SSLRole> GetDtlsRole() const;
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// TODO(deadbeef): Make this const. See comment in transportcontroller.h.
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bool GetStats(TransportStats* stats);
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const JsepTransportDescription* local_description() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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return local_description_.get();
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}
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const JsepTransportDescription* remote_description() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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return remote_description_.get();
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}
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// Returns the rtp transport, if any.
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webrtc::RtpTransportInternal* rtp_transport() const {
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if (dtls_srtp_transport_) {
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return dtls_srtp_transport_.get();
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}
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if (sdes_transport_) {
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return sdes_transport_.get();
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}
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if (unencrypted_rtp_transport_) {
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return unencrypted_rtp_transport_.get();
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}
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return nullptr;
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}
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const DtlsTransportInternal* rtp_dtls_transport() const {
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if (rtp_dtls_transport_) {
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return rtp_dtls_transport_->internal();
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}
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return nullptr;
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}
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DtlsTransportInternal* rtp_dtls_transport() {
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if (rtp_dtls_transport_) {
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return rtp_dtls_transport_->internal();
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}
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return nullptr;
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}
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const DtlsTransportInternal* rtcp_dtls_transport() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtcp_dtls_transport_) {
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return rtcp_dtls_transport_->internal();
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}
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return nullptr;
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}
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DtlsTransportInternal* rtcp_dtls_transport() {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtcp_dtls_transport_) {
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return rtcp_dtls_transport_->internal();
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}
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return nullptr;
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}
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rtc::scoped_refptr<webrtc::DtlsTransport> RtpDtlsTransport() {
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return rtp_dtls_transport_;
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}
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rtc::scoped_refptr<webrtc::SctpTransport> SctpTransport() const {
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return sctp_transport_;
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}
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// TODO(bugs.webrtc.org/9719): Delete method, update callers to use
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// SctpTransport() instead.
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webrtc::DataChannelTransportInterface* data_channel_transport() const {
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return sctp_transport_.get();
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}
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// TODO(deadbeef): The methods below are only public for testing. Should make
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// them utility functions or objects so they can be tested independently from
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// this class.
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// Returns an error if the certificate's identity does not match the
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// fingerprint, or either is NULL.
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webrtc::RTCError VerifyCertificateFingerprint(
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const rtc::RTCCertificate* certificate,
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const rtc::SSLFingerprint* fingerprint) const;
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void SetActiveResetSrtpParams(bool active_reset_srtp_params);
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private:
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bool SetRtcpMux(bool enable, webrtc::SdpType type, ContentSource source);
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void ActivateRtcpMux() RTC_RUN_ON(network_thread_);
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// Negotiates and sets the DTLS parameters based on the current local and
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// remote transport description, such as the DTLS role to use, and whether
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// DTLS should be activated.
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//
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// Called when an answer TransportDescription is applied.
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webrtc::RTCError NegotiateAndSetDtlsParameters(
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webrtc::SdpType local_description_type);
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// Negotiates the DTLS role based off the offer and answer as specified by
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// RFC 4145, section-4.1. Returns an RTCError if role cannot be determined
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// from the local description and remote description.
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webrtc::RTCError NegotiateDtlsRole(
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webrtc::SdpType local_description_type,
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ConnectionRole local_connection_role,
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ConnectionRole remote_connection_role,
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absl::optional<rtc::SSLRole>* negotiated_dtls_role);
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// Pushes down the ICE parameters from the remote description.
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void SetRemoteIceParameters(const IceParameters& ice_parameters,
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IceTransportInternal* ice);
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// Pushes down the DTLS parameters obtained via negotiation.
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static webrtc::RTCError SetNegotiatedDtlsParameters(
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DtlsTransportInternal* dtls_transport,
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absl::optional<rtc::SSLRole> dtls_role,
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rtc::SSLFingerprint* remote_fingerprint);
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bool GetTransportStats(DtlsTransportInternal* dtls_transport,
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int component,
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TransportStats* stats);
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// Owning thread, for safety checks
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const rtc::Thread* const network_thread_;
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const std::string mid_;
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// needs-ice-restart bit as described in JSEP.
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bool needs_ice_restart_ RTC_GUARDED_BY(network_thread_) = false;
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rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_
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RTC_GUARDED_BY(network_thread_);
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std::unique_ptr<JsepTransportDescription> local_description_
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RTC_GUARDED_BY(network_thread_);
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std::unique_ptr<JsepTransportDescription> remote_description_
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RTC_GUARDED_BY(network_thread_);
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// Ice transport which may be used by any of upper-layer transports (below).
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// Owned by JsepTransport and guaranteed to outlive the transports below.
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const rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport_;
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const rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport_;
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// To avoid downcasting and make it type safe, keep three unique pointers for
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// different SRTP mode and only one of these is non-nullptr.
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const std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport_;
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const std::unique_ptr<webrtc::SrtpTransport> sdes_transport_;
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const std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport_;
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const rtc::scoped_refptr<webrtc::DtlsTransport> rtp_dtls_transport_;
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// The RTCP transport is const for all usages, except that it is cleared
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// when RTCP multiplexing is turned on; this happens on the network thread.
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rtc::scoped_refptr<webrtc::DtlsTransport> rtcp_dtls_transport_
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RTC_GUARDED_BY(network_thread_);
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const rtc::scoped_refptr<webrtc::SctpTransport> sctp_transport_;
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RtcpMuxFilter rtcp_mux_negotiator_ RTC_GUARDED_BY(network_thread_);
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// Cache the encrypted header extension IDs for SDES negoitation.
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absl::optional<std::vector<int>> send_extension_ids_
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RTC_GUARDED_BY(network_thread_);
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absl::optional<std::vector<int>> recv_extension_ids_
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RTC_GUARDED_BY(network_thread_);
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// This is invoked when RTCP-mux becomes active and
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// `rtcp_dtls_transport_` is destroyed. The JsepTransportController will
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// receive the callback and update the aggregate transport states.
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std::function<void()> rtcp_mux_active_callback_;
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};
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} // namespace cricket
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#endif // PC_JSEP_TRANSPORT_H_
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