| /* |
| * libjingle |
| * Copyright 2012, Google Inc. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright notice, |
| * this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright notice, |
| * this list of conditions and the following disclaimer in the documentation |
| * and/or other materials provided with the distribution. |
| * 3. The name of the author may not be used to endorse or promote products |
| * derived from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
| #define TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
| |
| #include <string> |
| |
| #include "talk/app/webrtc/peerconnectioninterface.h" |
| #include "talk/app/webrtc/dtmfsender.h" |
| #include "talk/app/webrtc/mediastreamprovider.h" |
| #include "talk/app/webrtc/datachannel.h" |
| #include "talk/app/webrtc/statstypes.h" |
| #include "talk/base/sigslot.h" |
| #include "talk/base/thread.h" |
| #include "talk/media/base/mediachannel.h" |
| #include "talk/p2p/base/session.h" |
| #include "talk/session/media/mediasession.h" |
| |
| namespace cricket { |
| class BaseChannel; |
| class ChannelManager; |
| class DataChannel; |
| class StatsReport; |
| class Transport; |
| class VideoCapturer; |
| class VideoChannel; |
| class VoiceChannel; |
| } // namespace cricket |
| |
| namespace webrtc { |
| class IceRestartAnswerLatch; |
| class MediaStreamSignaling; |
| class WebRtcSessionDescriptionFactory; |
| |
| extern const char kBundleWithoutRtcpMux[]; |
| extern const char kCreateChannelFailed[]; |
| extern const char kInvalidCandidates[]; |
| extern const char kInvalidSdp[]; |
| extern const char kMlineMismatch[]; |
| extern const char kPushDownTDFailed[]; |
| extern const char kSdpWithoutCrypto[]; |
| extern const char kSdpWithoutIceUfragPwd[]; |
| extern const char kSdpWithoutSdesAndDtlsDisabled[]; |
| extern const char kSessionError[]; |
| extern const char kSessionErrorDesc[]; |
| |
| // ICE state callback interface. |
| class IceObserver { |
| public: |
| IceObserver() {} |
| // Called any time the IceConnectionState changes |
| virtual void OnIceConnectionChange( |
| PeerConnectionInterface::IceConnectionState new_state) {} |
| // Called any time the IceGatheringState changes |
| virtual void OnIceGatheringChange( |
| PeerConnectionInterface::IceGatheringState new_state) {} |
| // New Ice candidate have been found. |
| virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0; |
| // All Ice candidates have been found. |
| // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange. |
| // (via PeerConnectionObserver) |
| virtual void OnIceComplete() {} |
| |
| protected: |
| ~IceObserver() {} |
| |
| private: |
| DISALLOW_COPY_AND_ASSIGN(IceObserver); |
| }; |
| |
| class WebRtcSession : public cricket::BaseSession, |
| public AudioProviderInterface, |
| public DataChannelFactory, |
| public VideoProviderInterface, |
| public DtmfProviderInterface, |
| public DataChannelProviderInterface { |
| public: |
| WebRtcSession(cricket::ChannelManager* channel_manager, |
| talk_base::Thread* signaling_thread, |
| talk_base::Thread* worker_thread, |
| cricket::PortAllocator* port_allocator, |
| MediaStreamSignaling* mediastream_signaling); |
| virtual ~WebRtcSession(); |
| |
| bool Initialize(const PeerConnectionFactoryInterface::Options& options, |
| const MediaConstraintsInterface* constraints, |
| DTLSIdentityServiceInterface* dtls_identity_service); |
| // Deletes the voice, video and data channel and changes the session state |
| // to STATE_RECEIVEDTERMINATE. |
| void Terminate(); |
| |
| void RegisterIceObserver(IceObserver* observer) { |
| ice_observer_ = observer; |
| } |
| |
| virtual cricket::VoiceChannel* voice_channel() { |
| return voice_channel_.get(); |
| } |
| virtual cricket::VideoChannel* video_channel() { |
| return video_channel_.get(); |
| } |
| virtual cricket::DataChannel* data_channel() { |
| return data_channel_.get(); |
| } |
| |
| void SetSecurePolicy(cricket::SecureMediaPolicy secure_policy); |
| cricket::SecureMediaPolicy SecurePolicy() const; |
| |
| // Get current ssl role from transport. |
| bool GetSslRole(talk_base::SSLRole* role); |
| |
| // Generic error message callback from WebRtcSession. |
| // TODO - It may be necessary to supply error code as well. |
| sigslot::signal0<> SignalError; |
| |
| void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const MediaConstraintsInterface* constraints); |
| void CreateAnswer(CreateSessionDescriptionObserver* observer, |
| const MediaConstraintsInterface* constraints); |
| // The ownership of |desc| will be transferred after this call. |
| bool SetLocalDescription(SessionDescriptionInterface* desc, |
| std::string* err_desc); |
| // The ownership of |desc| will be transferred after this call. |
| bool SetRemoteDescription(SessionDescriptionInterface* desc, |
| std::string* err_desc); |
| bool ProcessIceMessage(const IceCandidateInterface* ice_candidate); |
| const SessionDescriptionInterface* local_description() const { |
| return local_desc_.get(); |
| } |
| const SessionDescriptionInterface* remote_description() const { |
| return remote_desc_.get(); |
| } |
| |
| // Get the id used as a media stream track's "id" field from ssrc. |
| virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id); |
| |
| // AudioMediaProviderInterface implementation. |
| virtual void SetAudioPlayout(uint32 ssrc, bool enable, |
| cricket::AudioRenderer* renderer) OVERRIDE; |
| virtual void SetAudioSend(uint32 ssrc, bool enable, |
| const cricket::AudioOptions& options, |
| cricket::AudioRenderer* renderer) OVERRIDE; |
| |
| // Implements VideoMediaProviderInterface. |
| virtual bool SetCaptureDevice(uint32 ssrc, |
| cricket::VideoCapturer* camera) OVERRIDE; |
| virtual void SetVideoPlayout(uint32 ssrc, |
| bool enable, |
| cricket::VideoRenderer* renderer) OVERRIDE; |
| virtual void SetVideoSend(uint32 ssrc, bool enable, |
| const cricket::VideoOptions* options) OVERRIDE; |
| |
| // Implements DtmfProviderInterface. |
| virtual bool CanInsertDtmf(const std::string& track_id); |
| virtual bool InsertDtmf(const std::string& track_id, |
| int code, int duration); |
| virtual sigslot::signal0<>* GetOnDestroyedSignal(); |
| |
| // Implements DataChannelProviderInterface. |
| virtual bool SendData(const cricket::SendDataParams& params, |
| const talk_base::Buffer& payload, |
| cricket::SendDataResult* result) OVERRIDE; |
| virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE; |
| virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE; |
| virtual void AddSctpDataStream(uint32 sid) OVERRIDE; |
| virtual void RemoveSctpDataStream(uint32 sid) OVERRIDE; |
| virtual bool ReadyToSendData() const OVERRIDE; |
| |
| // Implements DataChannelFactory. |
| talk_base::scoped_refptr<DataChannel> CreateDataChannel( |
| const std::string& label, |
| const InternalDataChannelInit* config) OVERRIDE; |
| |
| cricket::DataChannelType data_channel_type() const; |
| |
| bool IceRestartPending() const; |
| |
| void ResetIceRestartLatch(); |
| |
| // Called when an SSLIdentity is generated or retrieved by |
| // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |
| void OnIdentityReady(talk_base::SSLIdentity* identity); |
| |
| // For unit test. |
| bool waiting_for_identity() const; |
| |
| private: |
| // Indicates the type of SessionDescription in a call to SetLocalDescription |
| // and SetRemoteDescription. |
| enum Action { |
| kOffer, |
| kPrAnswer, |
| kAnswer, |
| }; |
| |
| // Invokes ConnectChannels() on transport proxies, which initiates ice |
| // candidates allocation. |
| bool StartCandidatesAllocation(); |
| bool UpdateSessionState(Action action, cricket::ContentSource source, |
| std::string* err_desc); |
| static Action GetAction(const std::string& type); |
| |
| // Transport related callbacks, override from cricket::BaseSession. |
| virtual void OnTransportRequestSignaling(cricket::Transport* transport); |
| virtual void OnTransportConnecting(cricket::Transport* transport); |
| virtual void OnTransportWritable(cricket::Transport* transport); |
| virtual void OnTransportProxyCandidatesReady( |
| cricket::TransportProxy* proxy, |
| const cricket::Candidates& candidates); |
| virtual void OnCandidatesAllocationDone(); |
| |
| // Creates local session description with audio and video contents. |
| bool CreateDefaultLocalDescription(); |
| // Enables media channels to allow sending of media. |
| void EnableChannels(); |
| // Creates a JsepIceCandidate and adds it to the local session description |
| // and notify observers. Called when a new local candidate have been found. |
| void ProcessNewLocalCandidate(const std::string& content_name, |
| const cricket::Candidates& candidates); |
| // Returns the media index for a local ice candidate given the content name. |
| // Returns false if the local session description does not have a media |
| // content called |content_name|. |
| bool GetLocalCandidateMediaIndex(const std::string& content_name, |
| int* sdp_mline_index); |
| // Uses all remote candidates in |remote_desc| in this session. |
| bool UseCandidatesInSessionDescription( |
| const SessionDescriptionInterface* remote_desc); |
| // Uses |candidate| in this session. |
| bool UseCandidate(const IceCandidateInterface* candidate); |
| // Deletes the corresponding channel of contents that don't exist in |desc|. |
| // |desc| can be null. This means that all channels are deleted. |
| void RemoveUnusedChannelsAndTransports( |
| const cricket::SessionDescription* desc); |
| |
| // Allocates media channels based on the |desc|. If |desc| doesn't have |
| // the BUNDLE option, this method will disable BUNDLE in PortAllocator. |
| // This method will also delete any existing media channels before creating. |
| bool CreateChannels(const cricket::SessionDescription* desc); |
| |
| // Helper methods to create media channels. |
| bool CreateVoiceChannel(const cricket::ContentInfo* content); |
| bool CreateVideoChannel(const cricket::ContentInfo* content); |
| bool CreateDataChannel(const cricket::ContentInfo* content); |
| |
| // Copy the candidates from |saved_candidates_| to |dest_desc|. |
| // The |saved_candidates_| will be cleared after this function call. |
| void CopySavedCandidates(SessionDescriptionInterface* dest_desc); |
| |
| // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN |
| // messages. |
| void OnDataChannelMessageReceived(cricket::DataChannel* channel, |
| const cricket::ReceiveDataParams& params, |
| const talk_base::Buffer& payload); |
| |
| bool GetLocalTrackId(uint32 ssrc, std::string* track_id); |
| bool GetRemoteTrackId(uint32 ssrc, std::string* track_id); |
| |
| std::string BadStateErrMsg(State state); |
| void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state); |
| |
| 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); |
| |
| std::string GetSessionErrorMsg(); |
| |
| talk_base::scoped_ptr<cricket::VoiceChannel> voice_channel_; |
| talk_base::scoped_ptr<cricket::VideoChannel> video_channel_; |
| talk_base::scoped_ptr<cricket::DataChannel> data_channel_; |
| cricket::ChannelManager* channel_manager_; |
| MediaStreamSignaling* mediastream_signaling_; |
| IceObserver* ice_observer_; |
| PeerConnectionInterface::IceConnectionState ice_connection_state_; |
| talk_base::scoped_ptr<SessionDescriptionInterface> local_desc_; |
| talk_base::scoped_ptr<SessionDescriptionInterface> remote_desc_; |
| // Candidates that arrived before the remote description was set. |
| std::vector<IceCandidateInterface*> saved_candidates_; |
| // If the remote peer is using a older version of implementation. |
| bool older_version_remote_peer_; |
| bool dtls_enabled_; |
| // Flag will be set based on the constraint value. |
| bool dscp_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). |
| cricket::DataChannelType data_channel_type_; |
| talk_base::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_; |
| |
| talk_base::scoped_ptr<WebRtcSessionDescriptionFactory> |
| webrtc_session_desc_factory_; |
| |
| sigslot::signal0<> SignalVoiceChannelDestroyed; |
| sigslot::signal0<> SignalVideoChannelDestroyed; |
| sigslot::signal0<> SignalDataChannelDestroyed; |
| |
| DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |
| }; |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ |