c08de0f4b7
Add methods to TransportController so that it can create RTP level transports (RtpTransport, SrtpTransport and DtlsSrtpTransport.). The RTP level transports are ref-counted since they could be shared by multiple BaseChannels and TransportController manages the life time of the transports. Bug: webrtc:7013 Change-Id: Ifd31062426e513d95473e257c9c9ff84a8c537fd Reviewed-on: https://webrtc-review.googlesource.com/5080 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21196}
313 lines
13 KiB
C++
313 lines
13 KiB
C++
/*
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* Copyright 2015 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_TRANSPORTCONTROLLER_H_
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#define PC_TRANSPORTCONTROLLER_H_
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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 "api/candidate.h"
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#include "p2p/base/dtlstransport.h"
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#include "p2p/base/jseptransport.h"
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#include "p2p/base/p2ptransportchannel.h"
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#include "pc/dtlssrtptransport.h"
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#include "pc/rtptransport.h"
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#include "pc/srtptransport.h"
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#include "rtc_base/asyncinvoker.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/refcountedobject.h"
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#include "rtc_base/sigslot.h"
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#include "rtc_base/sslstreamadapter.h"
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namespace rtc {
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class Thread;
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class PacketTransportInternal;
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} // namespace rtc
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namespace webrtc {
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class MetricsObserverInterface;
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} // namespace webrtc
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namespace cricket {
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class TransportController : public sigslot::has_slots<>,
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public rtc::MessageHandler {
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public:
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// If |redetermine_role_on_ice_restart| is true, ICE role is redetermined
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// upon setting a local transport description that indicates an ICE restart.
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// For the constructor that doesn't take this parameter, it defaults to true.
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//
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// |crypto_options| is used to determine if created DTLS transports negotiate
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// GCM crypto suites or not.
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TransportController(rtc::Thread* signaling_thread,
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rtc::Thread* network_thread,
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PortAllocator* port_allocator,
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bool redetermine_role_on_ice_restart,
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const rtc::CryptoOptions& crypto_options);
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virtual ~TransportController();
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rtc::Thread* signaling_thread() const { return signaling_thread_; }
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rtc::Thread* network_thread() const { return network_thread_; }
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PortAllocator* port_allocator() const { return port_allocator_; }
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// Can only be set before transports are created.
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// TODO(deadbeef): Make this an argument to the constructor once BaseSession
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// and WebRtcSession are combined
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bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version);
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void SetIceConfig(const IceConfig& config);
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void SetIceRole(IceRole ice_role);
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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& transport_name) const;
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bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role) const;
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// Specifies the identity to use in this session.
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// Can only be called once.
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bool SetLocalCertificate(
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const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
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bool GetLocalCertificate(
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const std::string& transport_name,
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rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const;
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// Caller owns returned certificate. This method mainly exists for stats
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// reporting.
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std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
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const std::string& transport_name) const;
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bool SetLocalTransportDescription(const std::string& transport_name,
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const TransportDescription& tdesc,
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webrtc::SdpType type,
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std::string* err);
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bool SetRemoteTransportDescription(const std::string& transport_name,
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const TransportDescription& tdesc,
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webrtc::SdpType type,
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std::string* err);
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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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bool AddRemoteCandidates(const std::string& transport_name,
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const Candidates& candidates,
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std::string* err);
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bool RemoveRemoteCandidates(const Candidates& candidates, std::string* err);
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bool ReadyForRemoteCandidates(const std::string& transport_name) const;
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// TODO(deadbeef): GetStats isn't const because all the way down to
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// OpenSSLStreamAdapter,
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// GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not const. Fix this.
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bool GetStats(const std::string& transport_name, TransportStats* stats);
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void SetMetricsObserver(webrtc::MetricsObserverInterface* metrics_observer);
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// Creates a channel if it doesn't exist. Otherwise, increments a reference
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// count and returns an existing channel.
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DtlsTransportInternal* CreateDtlsTransport(const std::string& transport_name,
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int component);
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virtual DtlsTransportInternal* CreateDtlsTransport_n(
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const std::string& transport_name,
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int component);
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// Decrements a channel's reference count, and destroys the channel if
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// nothing is referencing it.
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virtual void DestroyDtlsTransport(const std::string& transport_name,
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int component);
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virtual void DestroyDtlsTransport_n(const std::string& transport_name,
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int component);
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// Create an SrtpTransport/DtlsSrtpTransport if it doesn't exist.
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// Otherwise, increments a reference count and returns the existing one.
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// These methods are not currently used but the plan is to transition
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// PeerConnection and BaseChannel to use them instead of CreateDtlsTransport.
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webrtc::SrtpTransport* CreateSdesTransport(const std::string& transport_name,
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bool rtcp_mux_enabled);
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webrtc::DtlsSrtpTransport* CreateDtlsSrtpTransport(
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const std::string& transport_name,
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bool rtcp_mux_enabled);
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// Destroy an RTP level transport which can be an RtpTransport, an
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// SrtpTransport or a DtlsSrtpTransport.
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void DestroyTransport(const std::string& transport_name);
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// TODO(deadbeef): Remove all for_testing methods!
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const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing()
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const {
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return certificate_;
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}
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std::vector<std::string> transport_names_for_testing();
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std::vector<DtlsTransportInternal*> channels_for_testing();
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DtlsTransportInternal* get_channel_for_testing(
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const std::string& transport_name,
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int component);
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// All of these signals are fired on the signalling thread.
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// If any transport failed => failed,
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// Else if all completed => completed,
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// Else if all connected => connected,
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// Else => connecting
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sigslot::signal1<IceConnectionState> SignalConnectionState;
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// Receiving if any transport is receiving
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sigslot::signal1<bool> SignalReceiving;
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// If all transports done gathering => complete,
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// Else if any are gathering => gathering,
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// Else => new
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sigslot::signal1<IceGatheringState> SignalGatheringState;
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// (transport_name, candidates)
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sigslot::signal2<const std::string&, const Candidates&>
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SignalCandidatesGathered;
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sigslot::signal1<const Candidates&> SignalCandidatesRemoved;
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sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
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protected:
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// TODO(deadbeef): Get rid of these virtual methods. Used by
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// FakeTransportController currently, but FakeTransportController shouldn't
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// even be functioning by subclassing TransportController.
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virtual IceTransportInternal* CreateIceTransportChannel_n(
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const std::string& transport_name,
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int component);
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virtual DtlsTransportInternal* CreateDtlsTransportChannel_n(
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const std::string& transport_name,
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int component,
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IceTransportInternal* ice);
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private:
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void OnMessage(rtc::Message* pmsg) override;
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class ChannelPair;
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typedef rtc::RefCountedObject<ChannelPair> RefCountedChannel;
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// Wrapper for RtpTransport that keeps a reference count.
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// When using SDES, |srtp_transport| is non-null, |dtls_srtp_transport| is
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// null and |rtp_transport.get()| == |srtp_transport|,
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// When using DTLS-SRTP, |dtls_srtp_transport| is non-null, |srtp_transport|
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// is null and |rtp_transport.get()| == |dtls_srtp_transport|,
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// When using unencrypted RTP, only |rtp_transport| is non-null.
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struct RtpTransportWrapper {
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// |rtp_transport| is always non-null.
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std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport;
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webrtc::SrtpTransport* srtp_transport = nullptr;
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webrtc::DtlsSrtpTransport* dtls_srtp_transport = nullptr;
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};
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typedef rtc::RefCountedObject<RtpTransportWrapper> RefCountedRtpTransport;
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const RefCountedRtpTransport* FindRtpTransport(
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const std::string& transport_name);
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// Helper functions to get a channel or transport, or iterator to it (in case
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// it needs to be erased).
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std::vector<RefCountedChannel*>::iterator GetChannelIterator_n(
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const std::string& transport_name,
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int component);
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std::vector<RefCountedChannel*>::const_iterator GetChannelIterator_n(
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const std::string& transport_name,
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int component) const;
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const JsepTransport* GetJsepTransport(
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const std::string& transport_name) const;
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JsepTransport* GetJsepTransport(const std::string& transport_name);
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const RefCountedChannel* GetChannel_n(const std::string& transport_name,
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int component) const;
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RefCountedChannel* GetChannel_n(const std::string& transport_name,
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int component);
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JsepTransport* GetOrCreateJsepTransport(const std::string& transport_name);
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void DestroyAllChannels_n();
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bool SetSslMaxProtocolVersion_n(rtc::SSLProtocolVersion version);
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void SetIceConfig_n(const IceConfig& config);
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void SetIceRole_n(IceRole ice_role);
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bool GetSslRole_n(const std::string& transport_name,
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rtc::SSLRole* role) const;
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bool SetLocalCertificate_n(
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const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
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bool GetLocalCertificate_n(
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const std::string& transport_name,
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rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const;
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std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate_n(
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const std::string& transport_name) const;
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bool SetLocalTransportDescription_n(const std::string& transport_name,
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const TransportDescription& tdesc,
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webrtc::SdpType type,
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std::string* err);
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bool SetRemoteTransportDescription_n(const std::string& transport_name,
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const TransportDescription& tdesc,
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webrtc::SdpType type,
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std::string* err);
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void MaybeStartGathering_n();
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bool AddRemoteCandidates_n(const std::string& transport_name,
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const Candidates& candidates,
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std::string* err);
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bool RemoveRemoteCandidates_n(const Candidates& candidates, std::string* err);
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bool ReadyForRemoteCandidates_n(const std::string& transport_name) const;
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bool GetStats_n(const std::string& transport_name, TransportStats* stats);
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void SetMetricsObserver_n(webrtc::MetricsObserverInterface* metrics_observer);
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// Handlers for signals from Transport.
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void OnChannelWritableState_n(rtc::PacketTransportInternal* transport);
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void OnChannelReceivingState_n(rtc::PacketTransportInternal* transport);
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void OnChannelGatheringState_n(IceTransportInternal* channel);
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void OnChannelCandidateGathered_n(IceTransportInternal* channel,
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const Candidate& candidate);
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void OnChannelCandidatesRemoved(const Candidates& candidates);
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void OnChannelCandidatesRemoved_n(IceTransportInternal* channel,
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const Candidates& candidates);
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void OnChannelRoleConflict_n(IceTransportInternal* channel);
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void OnChannelStateChanged_n(IceTransportInternal* channel);
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void UpdateAggregateStates_n();
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void OnDtlsHandshakeError(rtc::SSLHandshakeError error);
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rtc::Thread* const signaling_thread_ = nullptr;
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rtc::Thread* const network_thread_ = nullptr;
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PortAllocator* const port_allocator_ = nullptr;
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std::map<std::string, std::unique_ptr<JsepTransport>> transports_;
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std::vector<RefCountedChannel*> channels_;
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std::map<std::string, RefCountedRtpTransport*> rtp_transports_;
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// Aggregate state for TransportChannelImpls.
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IceConnectionState connection_state_ = kIceConnectionConnecting;
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bool receiving_ = false;
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IceGatheringState gathering_state_ = kIceGatheringNew;
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IceConfig ice_config_;
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IceRole ice_role_ = ICEROLE_CONTROLLING;
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bool redetermine_role_on_ice_restart_;
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uint64_t ice_tiebreaker_ = rtc::CreateRandomId64();
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rtc::CryptoOptions crypto_options_;
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rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
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rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
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rtc::AsyncInvoker invoker_;
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webrtc::MetricsObserverInterface* metrics_observer_ = nullptr;
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RTC_DISALLOW_COPY_AND_ASSIGN(TransportController);
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};
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} // namespace cricket
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#endif // PC_TRANSPORTCONTROLLER_H_
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