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This is a reland of 9185aca9ce
> Original change's description:
> > > Clean up libjingle API dependencies.
> > >
> > > This CL moves candidate.h into the public API, since it has
> > > been implicitly included before.
> > >
> > > This is a straightforward way of solving the circular
> > > dependencies involving that file. For instance,
> > > libjingle_peerconnection_api includes candidate.h from
> > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> > > depends on _api. In fact, _api can't depend on much at all
> > > since it's a very high level abstraction; instead, things
> > > should depend on it.
> > >
> > > Furthermore, we have the case where deprecated headers
> > > include headers in internal modules. I just have to turn
> > > off include checking for those, but that's not a big deal.
> > >
> > > This CL punts the problem of callfactoryinterface.h being
> > > implicitly included, and pulling in most of the call
> > > module with it. This should be addressed in a follow-up
> > > CL.
> Bug: webrtc:7504
> Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6
> Reviewed-on: https://webrtc-review.googlesource.com/6460
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20156}
TBR=deadbeef@webrtc.org
Bug: webrtc:7504
Change-Id: Ic6598ac2af9355b60bbd289c86dc75e0ae9fed2e
Reviewed-on: https://webrtc-review.googlesource.com/6801
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20167}
680 lines
28 KiB
C++
680 lines
28 KiB
C++
/*
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* Copyright 2012 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_WEBRTCSESSION_H_
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#define PC_WEBRTCSESSION_H_
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#include <memory>
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#include <set>
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#include <string>
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#include <vector>
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#include "api/candidate.h"
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#include "api/optional.h"
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#include "api/peerconnectioninterface.h"
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#include "api/statstypes.h"
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#include "call/call.h"
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#include "pc/datachannel.h"
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#include "pc/mediasession.h"
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#include "pc/transportcontroller.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/sigslot.h"
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#include "rtc_base/sslidentity.h"
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#include "rtc_base/thread.h"
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#ifdef HAVE_QUIC
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#include "pc/quicdatatransport.h"
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#endif // HAVE_QUIC
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namespace cricket {
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class ChannelManager;
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class RtpDataChannel;
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class SctpTransportInternal;
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class SctpTransportInternalFactory;
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class StatsReport;
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class VideoChannel;
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class VoiceChannel;
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#ifdef HAVE_QUIC
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class QuicTransportChannel;
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#endif // HAVE_QUIC
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} // namespace cricket
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namespace webrtc {
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class IceRestartAnswerLatch;
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class JsepIceCandidate;
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class MediaStreamSignaling;
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class RtcEventLog;
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class WebRtcSessionDescriptionFactory;
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extern const char kBundleWithoutRtcpMux[];
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extern const char kCreateChannelFailed[];
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extern const char kInvalidCandidates[];
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extern const char kInvalidSdp[];
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extern const char kMlineMismatchInAnswer[];
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extern const char kMlineMismatchInSubsequentOffer[];
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extern const char kPushDownTDFailed[];
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extern const char kSdpWithoutDtlsFingerprint[];
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extern const char kSdpWithoutSdesCrypto[];
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extern const char kSdpWithoutIceUfragPwd[];
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extern const char kSdpWithoutSdesAndDtlsDisabled[];
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extern const char kSessionError[];
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extern const char kSessionErrorDesc[];
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extern const char kDtlsSrtpSetupFailureRtp[];
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extern const char kDtlsSrtpSetupFailureRtcp[];
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extern const char kEnableBundleFailed[];
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// Maximum number of received video streams that will be processed by webrtc
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// even if they are not signalled beforehand.
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extern const int kMaxUnsignalledRecvStreams;
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// ICE state callback interface.
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class IceObserver {
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public:
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IceObserver() {}
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// Called any time the IceConnectionState changes
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virtual void OnIceConnectionStateChange(
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PeerConnectionInterface::IceConnectionState new_state) {}
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// Called any time the IceGatheringState changes
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virtual void OnIceGatheringChange(
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PeerConnectionInterface::IceGatheringState new_state) {}
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// New Ice candidate have been found.
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virtual void OnIceCandidate(
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std::unique_ptr<IceCandidateInterface> candidate) = 0;
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// Some local ICE candidates have been removed.
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virtual void OnIceCandidatesRemoved(
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const std::vector<cricket::Candidate>& candidates) = 0;
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// Called whenever the state changes between receiving and not receiving.
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virtual void OnIceConnectionReceivingChange(bool receiving) {}
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protected:
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~IceObserver() {}
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
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};
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// Statistics for all the transports of the session.
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typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
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typedef std::map<std::string, std::string> ProxyTransportMap;
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// TODO(pthatcher): Think of a better name for this. We already have
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// a TransportStats in transport.h. Perhaps TransportsStats?
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struct SessionStats {
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ProxyTransportMap proxy_to_transport;
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TransportStatsMap transport_stats;
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};
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struct ChannelNamePair {
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ChannelNamePair(
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const std::string& content_name, const std::string& transport_name)
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: content_name(content_name), transport_name(transport_name) {}
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std::string content_name;
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std::string transport_name;
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};
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struct ChannelNamePairs {
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rtc::Optional<ChannelNamePair> voice;
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rtc::Optional<ChannelNamePair> video;
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rtc::Optional<ChannelNamePair> data;
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};
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// A WebRtcSession manages general session state. This includes negotiation
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// of both the application-level and network-level protocols: the former
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// defines what will be sent and the latter defines how it will be sent. Each
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// network-level protocol is represented by a Transport object. Each Transport
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// participates in the network-level negotiation. The individual streams of
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// packets are represented by TransportChannels. The application-level protocol
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// is represented by SessionDecription objects.
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class WebRtcSession :
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public DataChannelProviderInterface,
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public sigslot::has_slots<> {
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public:
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enum State {
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STATE_INIT = 0,
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STATE_SENTOFFER, // Sent offer, waiting for answer.
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STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
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STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
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STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
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STATE_INPROGRESS, // Offer/answer exchange completed.
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STATE_CLOSED, // Close() was called.
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};
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enum Error {
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ERROR_NONE = 0, // no error
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ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
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ERROR_TRANSPORT = 2, // transport error of some kind
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};
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// |sctp_factory| may be null, in which case SCTP is treated as unsupported.
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WebRtcSession(
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Call* call,
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cricket::ChannelManager* channel_manager,
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const cricket::MediaConfig& media_config,
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RtcEventLog* event_log,
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rtc::Thread* network_thread,
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rtc::Thread* worker_thread,
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rtc::Thread* signaling_thread,
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cricket::PortAllocator* port_allocator,
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std::unique_ptr<cricket::TransportController> transport_controller,
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std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory);
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virtual ~WebRtcSession();
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// These are const to allow them to be called from const methods.
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rtc::Thread* network_thread() const { return network_thread_; }
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rtc::Thread* worker_thread() const { return worker_thread_; }
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rtc::Thread* signaling_thread() const { return signaling_thread_; }
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// The ID of this session.
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const std::string& id() const { return sid_; }
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bool Initialize(
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const PeerConnectionFactoryInterface::Options& options,
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
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const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
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// Deletes the voice, video and data channel and changes the session state
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// to STATE_CLOSED.
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void Close();
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// Returns true if we were the initial offerer.
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bool initial_offerer() const { return initial_offerer_; }
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// Returns the current state of the session. See the enum above for details.
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// Each time the state changes, we will fire this signal.
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State state() const { return state_; }
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sigslot::signal2<WebRtcSession*, State> SignalState;
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// Returns the last error in the session. See the enum above for details.
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Error error() const { return error_; }
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const std::string& error_desc() const { return error_desc_; }
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void RegisterIceObserver(IceObserver* observer) {
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ice_observer_ = observer;
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}
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// Exposed for stats collecting.
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// TODO(steveanton): Switch callers to use the plural form and remove these.
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virtual cricket::VoiceChannel* voice_channel() {
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if (voice_channels_.empty()) {
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return nullptr;
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} else {
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return voice_channels_[0];
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}
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}
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virtual cricket::VideoChannel* video_channel() {
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if (video_channels_.empty()) {
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return nullptr;
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} else {
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return video_channels_[0];
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}
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}
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virtual std::vector<cricket::VoiceChannel*> voice_channels() const {
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return voice_channels_;
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}
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virtual std::vector<cricket::VideoChannel*> video_channels() const {
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return video_channels_;
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}
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// Only valid when using deprecated RTP data channels.
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virtual cricket::RtpDataChannel* rtp_data_channel() {
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return rtp_data_channel_.get();
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}
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virtual rtc::Optional<std::string> sctp_content_name() const {
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return sctp_content_name_;
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}
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virtual rtc::Optional<std::string> sctp_transport_name() const {
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return sctp_transport_name_;
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}
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cricket::BaseChannel* GetChannel(const std::string& content_name);
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cricket::SecurePolicy SdesPolicy() const;
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// Get current SSL role used by SCTP's underlying transport.
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bool GetSctpSslRole(rtc::SSLRole* role);
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// Get SSL role for an arbitrary m= section (handles bundling correctly).
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// TODO(deadbeef): This is only used internally by the session description
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// factory, it shouldn't really be public).
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bool GetSslRole(const std::string& content_name, rtc::SSLRole* role);
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void CreateOffer(
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CreateSessionDescriptionObserver* observer,
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const PeerConnectionInterface::RTCOfferAnswerOptions& options,
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const cricket::MediaSessionOptions& session_options);
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void CreateAnswer(CreateSessionDescriptionObserver* observer,
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const cricket::MediaSessionOptions& session_options);
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// The ownership of |desc| will be transferred after this call.
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bool SetLocalDescription(SessionDescriptionInterface* desc,
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std::string* err_desc);
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// The ownership of |desc| will be transferred after this call.
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bool SetRemoteDescription(SessionDescriptionInterface* desc,
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std::string* err_desc);
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bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
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bool RemoveRemoteIceCandidates(
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const std::vector<cricket::Candidate>& candidates);
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cricket::IceConfig ParseIceConfig(
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const PeerConnectionInterface::RTCConfiguration& config) const;
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void SetIceConfig(const cricket::IceConfig& ice_config);
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// Start gathering candidates for any new transports, or transports doing an
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// ICE restart.
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void MaybeStartGathering();
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const SessionDescriptionInterface* local_description() const {
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return pending_local_description_ ? pending_local_description_.get()
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: current_local_description_.get();
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}
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const SessionDescriptionInterface* remote_description() const {
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return pending_remote_description_ ? pending_remote_description_.get()
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: current_remote_description_.get();
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}
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const SessionDescriptionInterface* current_local_description() const {
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return current_local_description_.get();
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}
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const SessionDescriptionInterface* current_remote_description() const {
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return current_remote_description_.get();
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}
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const SessionDescriptionInterface* pending_local_description() const {
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return pending_local_description_.get();
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}
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const SessionDescriptionInterface* pending_remote_description() const {
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return pending_remote_description_.get();
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}
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// Get the id used as a media stream track's "id" field from ssrc.
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virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
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virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
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// Implements DataChannelProviderInterface.
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bool SendData(const cricket::SendDataParams& params,
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const rtc::CopyOnWriteBuffer& payload,
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cricket::SendDataResult* result) override;
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bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
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void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
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void AddSctpDataStream(int sid) override;
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void RemoveSctpDataStream(int sid) override;
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bool ReadyToSendData() const override;
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virtual Call::Stats GetCallStats();
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// Returns stats for all channels of all transports.
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// This avoids exposing the internal structures used to track them.
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// The parameterless version creates |ChannelNamePairs| from |voice_channel|,
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// |video_channel| and |voice_channel| if available - this requires it to be
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// called on the signaling thread - and invokes the other |GetStats|. The
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// other |GetStats| can be invoked on any thread; if not invoked on the
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// network thread a thread hop will happen.
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std::unique_ptr<SessionStats> GetStats_s();
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virtual std::unique_ptr<SessionStats> GetStats(
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const ChannelNamePairs& channel_name_pairs);
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// virtual so it can be mocked in unit tests
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virtual bool GetLocalCertificate(
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const std::string& transport_name,
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rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
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// Caller owns returned certificate
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virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
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const std::string& transport_name);
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cricket::DataChannelType data_channel_type() const;
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// Returns true if there was an ICE restart initiated by the remote offer.
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bool IceRestartPending(const std::string& content_name) const;
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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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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). If the transport has been deleted as a result of
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// bundling, returns false.
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bool NeedsIceRestart(const std::string& content_name) const;
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// Called when an RTCCertificate is generated or retrieved by
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// WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
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void OnCertificateReady(
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const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
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void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp);
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// For unit test.
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bool waiting_for_certificate_for_testing() const;
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const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
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void set_metrics_observer(
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webrtc::MetricsObserverInterface* metrics_observer) {
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metrics_observer_ = metrics_observer;
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transport_controller_->SetMetricsObserver(metrics_observer);
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}
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// Called when voice_channel_, video_channel_ and
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// rtp_data_channel_/sctp_transport_ are created and destroyed. As a result
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// of, for example, setting a new description.
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sigslot::signal0<> SignalVoiceChannelCreated;
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sigslot::signal0<> SignalVoiceChannelDestroyed;
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sigslot::signal0<> SignalVideoChannelCreated;
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sigslot::signal0<> SignalVideoChannelDestroyed;
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sigslot::signal0<> SignalDataChannelCreated;
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sigslot::signal0<> SignalDataChannelDestroyed;
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// Called when a valid data channel OPEN message is received.
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// std::string represents the data channel label.
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sigslot::signal2<const std::string&, const InternalDataChannelInit&>
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SignalDataChannelOpenMessage;
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#ifdef HAVE_QUIC
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QuicDataTransport* quic_data_transport() {
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return quic_data_transport_.get();
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}
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#endif // HAVE_QUIC
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private:
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// Indicates the type of SessionDescription in a call to SetLocalDescription
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// and SetRemoteDescription.
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enum Action {
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kOffer,
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kPrAnswer,
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kAnswer,
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};
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// Return all managed, non-null channels.
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std::vector<cricket::BaseChannel*> Channels() const;
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// Non-const versions of local_description()/remote_description(), for use
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// internally.
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SessionDescriptionInterface* mutable_local_description() {
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return pending_local_description_ ? pending_local_description_.get()
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: current_local_description_.get();
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}
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SessionDescriptionInterface* mutable_remote_description() {
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return pending_remote_description_ ? pending_remote_description_.get()
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: current_remote_description_.get();
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}
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// Log session state.
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void LogState(State old_state, State new_state);
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// Updates the state, signaling if necessary.
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virtual void SetState(State state);
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// Updates the error state, signaling if necessary.
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// TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
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virtual void SetError(Error error, const std::string& error_desc);
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bool UpdateSessionState(Action action, cricket::ContentSource source,
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std::string* err_desc);
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static Action GetAction(const std::string& type);
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// Push the media parts of the local or remote session description
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// down to all of the channels.
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bool PushdownMediaDescription(cricket::ContentAction action,
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cricket::ContentSource source,
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std::string* error_desc);
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bool PushdownSctpParameters_n(cricket::ContentSource source);
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bool PushdownTransportDescription(cricket::ContentSource source,
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cricket::ContentAction action,
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std::string* error_desc);
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// Helper methods to push local and remote transport descriptions.
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bool PushdownLocalTransportDescription(
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const cricket::SessionDescription* sdesc,
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cricket::ContentAction action,
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std::string* error_desc);
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bool PushdownRemoteTransportDescription(
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const cricket::SessionDescription* sdesc,
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cricket::ContentAction action,
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std::string* error_desc);
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// Returns true and the TransportInfo of the given |content_name|
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// from |description|. Returns false if it's not available.
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static bool GetTransportDescription(
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const cricket::SessionDescription* description,
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const std::string& content_name,
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cricket::TransportDescription* info);
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// Returns the name of the transport channel when BUNDLE is enabled, or
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// nullptr if the channel is not part of any bundle.
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const std::string* GetBundleTransportName(
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const cricket::ContentInfo* content,
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const cricket::ContentGroup* bundle);
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// Cause all the BaseChannels in the bundle group to have the same
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// transport channel.
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bool EnableBundle(const cricket::ContentGroup& bundle);
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// Enables media channels to allow sending of media.
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void EnableChannels();
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// Returns the media index for a local ice candidate given the content name.
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// Returns false if the local session description does not have a media
|
|
// content called |content_name|.
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bool GetLocalCandidateMediaIndex(const std::string& content_name,
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int* sdp_mline_index);
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// Uses all remote candidates in |remote_desc| in this session.
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bool UseCandidatesInSessionDescription(
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const SessionDescriptionInterface* remote_desc);
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// Uses |candidate| in this session.
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bool UseCandidate(const IceCandidateInterface* candidate);
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// Deletes the corresponding channel of contents that don't exist in |desc|.
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// |desc| can be null. This means that all channels are deleted.
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void RemoveUnusedChannels(const cricket::SessionDescription* desc);
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// Allocates media channels based on the |desc|. If |desc| doesn't have
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// the BUNDLE option, this method will disable BUNDLE in PortAllocator.
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// This method will also delete any existing media channels before creating.
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bool CreateChannels(const cricket::SessionDescription* desc);
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|
|
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// Helper methods to create media channels.
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bool CreateVoiceChannel(const cricket::ContentInfo* content,
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const std::string* bundle_transport);
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bool CreateVideoChannel(const cricket::ContentInfo* content,
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const std::string* bundle_transport);
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bool CreateDataChannel(const cricket::ContentInfo* content,
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const std::string* bundle_transport);
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|
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std::unique_ptr<SessionStats> GetStats_n(
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const ChannelNamePairs& channel_name_pairs);
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|
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bool CreateSctpTransport_n(const std::string& content_name,
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const std::string& transport_name);
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// For bundling.
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void ChangeSctpTransport_n(const std::string& transport_name);
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void DestroySctpTransport_n();
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// SctpTransport signal handlers. Needed to marshal signals from the network
|
|
// to signaling thread.
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void OnSctpTransportReadyToSendData_n();
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// This may be called with "false" if the direction of the m= section causes
|
|
// us to tear down the SCTP connection.
|
|
void OnSctpTransportReadyToSendData_s(bool ready);
|
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void OnSctpTransportDataReceived_n(const cricket::ReceiveDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload);
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|
// Beyond just firing the signal to the signaling thread, listens to SCTP
|
|
// CONTROL messages on unused SIDs and processes them as OPEN messages.
|
|
void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload);
|
|
void OnSctpStreamClosedRemotely_n(int sid);
|
|
|
|
std::string BadStateErrMsg(State state);
|
|
void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
|
|
void SetIceConnectionReceiving(bool receiving);
|
|
|
|
bool ValidateBundleSettings(const cricket::SessionDescription* desc);
|
|
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
|
|
// Below methods are helper methods which verifies SDP.
|
|
bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
|
|
cricket::ContentSource source,
|
|
std::string* err_desc);
|
|
|
|
// Check if a call to SetLocalDescription is acceptable with |action|.
|
|
bool ExpectSetLocalDescription(Action action);
|
|
// Check if a call to SetRemoteDescription is acceptable with |action|.
|
|
bool ExpectSetRemoteDescription(Action action);
|
|
// Verifies a=setup attribute as per RFC 5763.
|
|
bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
|
|
Action action);
|
|
|
|
// Returns true if we are ready to push down the remote candidate.
|
|
// |remote_desc| is the new remote description, or NULL if the current remote
|
|
// description should be used. Output |valid| is true if the candidate media
|
|
// index is valid.
|
|
bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
|
|
const SessionDescriptionInterface* remote_desc,
|
|
bool* valid);
|
|
|
|
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
|
|
// this session.
|
|
bool SrtpRequired() const;
|
|
|
|
// TransportController signal handlers.
|
|
void OnTransportControllerConnectionState(cricket::IceConnectionState state);
|
|
void OnTransportControllerReceiving(bool receiving);
|
|
void OnTransportControllerGatheringState(cricket::IceGatheringState state);
|
|
void OnTransportControllerCandidatesGathered(
|
|
const std::string& transport_name,
|
|
const std::vector<cricket::Candidate>& candidates);
|
|
void OnTransportControllerCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates);
|
|
void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
|
|
|
|
std::string GetSessionErrorMsg();
|
|
|
|
// Invoked when TransportController connection completion is signaled.
|
|
// Reports stats for all transports in use.
|
|
void ReportTransportStats();
|
|
|
|
// Gather the usage of IPv4/IPv6 as best connection.
|
|
void ReportBestConnectionState(const cricket::TransportStats& stats);
|
|
|
|
void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
|
|
|
|
void OnSentPacket_w(const rtc::SentPacket& sent_packet);
|
|
|
|
const std::string GetTransportName(const std::string& content_name);
|
|
|
|
void DestroyRtcpTransport_n(const std::string& transport_name);
|
|
void RemoveAndDestroyVideoChannel(cricket::VideoChannel* video_channel);
|
|
void DestroyVideoChannel(cricket::VideoChannel* video_channel);
|
|
void RemoveAndDestroyVoiceChannel(cricket::VoiceChannel* voice_channel);
|
|
void DestroyVoiceChannel(cricket::VoiceChannel* voice_channel);
|
|
void DestroyDataChannel();
|
|
|
|
rtc::Thread* const network_thread_;
|
|
rtc::Thread* const worker_thread_;
|
|
rtc::Thread* const signaling_thread_;
|
|
|
|
State state_ = STATE_INIT;
|
|
Error error_ = ERROR_NONE;
|
|
std::string error_desc_;
|
|
|
|
const std::string sid_;
|
|
bool initial_offerer_ = false;
|
|
|
|
const std::unique_ptr<cricket::TransportController> transport_controller_;
|
|
const std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_;
|
|
const cricket::MediaConfig media_config_;
|
|
RtcEventLog* event_log_;
|
|
Call* call_;
|
|
// TODO(steveanton): voice_channels_ and video_channels_ used to be a single
|
|
// VoiceChannel/VideoChannel respectively but are being changed to support
|
|
// multiple m= lines in unified plan. But until more work is done, these can
|
|
// only have 0 or 1 channel each.
|
|
// These channels are owned by ChannelManager.
|
|
std::vector<cricket::VoiceChannel*> voice_channels_;
|
|
std::vector<cricket::VideoChannel*> video_channels_;
|
|
// |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_|
|
|
// when using SCTP.
|
|
// TODO(steveanton): This should be changed to a bare pointer because
|
|
// WebRtcSession doesn't actually own the RtpDataChannel
|
|
// (ChannelManager does).
|
|
std::unique_ptr<cricket::RtpDataChannel> rtp_data_channel_;
|
|
|
|
std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
|
|
// |sctp_transport_name_| keeps track of what DTLS transport the SCTP
|
|
// transport is using (which can change due to bundling).
|
|
rtc::Optional<std::string> sctp_transport_name_;
|
|
// |sctp_content_name_| is the content name (MID) in SDP.
|
|
rtc::Optional<std::string> sctp_content_name_;
|
|
// Value cached on signaling thread. Only updated when SctpReadyToSendData
|
|
// fires on the signaling thread.
|
|
bool sctp_ready_to_send_data_ = false;
|
|
// Same as signals provided by SctpTransport, but these are guaranteed to
|
|
// fire on the signaling thread, whereas SctpTransport fires on the networking
|
|
// thread.
|
|
// |sctp_invoker_| is used so that any signals queued on the signaling thread
|
|
// from the network thread are immediately discarded if the SctpTransport is
|
|
// destroyed (due to m= section being rejected).
|
|
// TODO(deadbeef): Use a proxy object to ensure that method calls/signals
|
|
// are marshalled to the right thread. Could almost use proxy.h for this,
|
|
// but it doesn't have a mechanism for marshalling sigslot::signals
|
|
std::unique_ptr<rtc::AsyncInvoker> sctp_invoker_;
|
|
sigslot::signal1<bool> SignalSctpReadyToSendData;
|
|
sigslot::signal2<const cricket::ReceiveDataParams&,
|
|
const rtc::CopyOnWriteBuffer&>
|
|
SignalSctpDataReceived;
|
|
sigslot::signal1<int> SignalSctpStreamClosedRemotely;
|
|
|
|
cricket::ChannelManager* channel_manager_;
|
|
IceObserver* ice_observer_;
|
|
PeerConnectionInterface::IceConnectionState ice_connection_state_;
|
|
bool ice_connection_receiving_;
|
|
std::unique_ptr<SessionDescriptionInterface> current_local_description_;
|
|
std::unique_ptr<SessionDescriptionInterface> pending_local_description_;
|
|
std::unique_ptr<SessionDescriptionInterface> current_remote_description_;
|
|
std::unique_ptr<SessionDescriptionInterface> pending_remote_description_;
|
|
// If the remote peer is using a older version of implementation.
|
|
bool older_version_remote_peer_;
|
|
bool dtls_enabled_;
|
|
// Specifies which kind of data channel is allowed. This is controlled
|
|
// by the chrome command-line flag and constraints:
|
|
// 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
|
|
// constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
|
|
// not set or false, SCTP is allowed (DCT_SCTP);
|
|
// 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
|
|
// 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
|
|
// The data channel type could be DCT_QUIC if the QUIC data channel is
|
|
// enabled.
|
|
cricket::DataChannelType data_channel_type_;
|
|
// List of content names for which the remote side triggered an ICE restart.
|
|
std::set<std::string> pending_ice_restarts_;
|
|
|
|
std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
|
|
|
|
// Member variables for caching global options.
|
|
cricket::AudioOptions audio_options_;
|
|
cricket::VideoOptions video_options_;
|
|
MetricsObserverInterface* metrics_observer_;
|
|
|
|
// Declares the bundle policy for the WebRTCSession.
|
|
PeerConnectionInterface::BundlePolicy bundle_policy_;
|
|
|
|
// Declares the RTCP mux policy for the WebRTCSession.
|
|
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
|
|
|
|
bool received_first_video_packet_ = false;
|
|
bool received_first_audio_packet_ = false;
|
|
|
|
#ifdef HAVE_QUIC
|
|
std::unique_ptr<QuicDataTransport> quic_data_transport_;
|
|
#endif // HAVE_QUIC
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // PC_WEBRTCSESSION_H_
|