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/*
* 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 ChannelManager;
class DataChannel;
class StatsReport;
class Transport;
class VideoCapturer;
class BaseChannel;
class VideoChannel;
class VoiceChannel;
} // namespace cricket
namespace webrtc {
class IceRestartAnswerLatch;
class MediaStreamSignaling;
class WebRtcSessionDescriptionFactory;
extern const char kSetLocalSdpFailed[];
extern const char kSetRemoteSdpFailed[];
extern const char kCreateChannelFailed[];
extern const char kBundleWithoutRtcpMux[];
extern const char kInvalidCandidates[];
extern const char kInvalidSdp[];
extern const char kMlineMismatch[];
extern const char kSdpWithoutCrypto[];
extern const char kSdpWithoutSdesAndDtlsDisabled[];
extern const char kSessionError[];
extern const char kUpdateStateFailed[];
extern const char kPushDownOfferTDFailed[];
extern const char kPushDownPranswerTDFailed[];
extern const char kPushDownAnswerTDFailed[];
// ICE state callback interface.
class IceObserver {
public:
// 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() {}
};
class WebRtcSession : public cricket::BaseSession,
public AudioProviderInterface,
public DataChannelFactory,
public VideoProviderInterface,
public DtmfProviderInterface {
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 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 set_secure_policy(cricket::SecureMediaPolicy secure_policy);
cricket::SecureMediaPolicy secure_policy() 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();
talk_base::scoped_refptr<DataChannel> CreateDataChannel(
const std::string& label,
const DataChannelInit* config);
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,
const cricket::SessionDescription* desc,
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);
void OnDataReceived(
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(const std::string& type, 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* error_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);
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_;
// 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;
};
} // namespace webrtc
#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_