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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2012, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_
29#define TALK_APP_WEBRTC_WEBRTCSESSION_H_
30
31#include <string>
32
33#include "talk/app/webrtc/peerconnectioninterface.h"
34#include "talk/app/webrtc/dtmfsender.h"
35#include "talk/app/webrtc/mediastreamprovider.h"
36#include "talk/app/webrtc/datachannel.h"
37#include "talk/app/webrtc/statstypes.h"
38#include "talk/base/sigslot.h"
39#include "talk/base/thread.h"
40#include "talk/media/base/mediachannel.h"
41#include "talk/p2p/base/session.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042#include "talk/session/media/mediasession.h"
43
44namespace cricket {
wu@webrtc.org364f2042013-11-20 21:49:41 +000045class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046class ChannelManager;
47class DataChannel;
48class StatsReport;
49class Transport;
50class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000051class VideoChannel;
52class VoiceChannel;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053} // namespace cricket
54
55namespace webrtc {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056class IceRestartAnswerLatch;
57class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000058class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059
60extern const char kSetLocalSdpFailed[];
61extern const char kSetRemoteSdpFailed[];
62extern const char kCreateChannelFailed[];
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000063extern const char kBundleWithoutRtcpMux[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000064extern const char kInvalidCandidates[];
65extern const char kInvalidSdp[];
66extern const char kMlineMismatch[];
67extern const char kSdpWithoutCrypto[];
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +000068extern const char kSdpWithoutSdesAndDtlsDisabled[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000069extern const char kSdpWithoutIceUfragPwd[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000070extern const char kSessionError[];
71extern const char kUpdateStateFailed[];
72extern const char kPushDownOfferTDFailed[];
73extern const char kPushDownPranswerTDFailed[];
74extern const char kPushDownAnswerTDFailed[];
75
76// ICE state callback interface.
77class IceObserver {
78 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000079 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080 // Called any time the IceConnectionState changes
81 virtual void OnIceConnectionChange(
82 PeerConnectionInterface::IceConnectionState new_state) {}
83 // Called any time the IceGatheringState changes
84 virtual void OnIceGatheringChange(
85 PeerConnectionInterface::IceGatheringState new_state) {}
86 // New Ice candidate have been found.
87 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
88 // All Ice candidates have been found.
89 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
90 // (via PeerConnectionObserver)
91 virtual void OnIceComplete() {}
92
93 protected:
94 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000095
96 private:
97 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000098};
99
100class WebRtcSession : public cricket::BaseSession,
101 public AudioProviderInterface,
102 public DataChannelFactory,
103 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000104 public DtmfProviderInterface,
105 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106 public:
107 WebRtcSession(cricket::ChannelManager* channel_manager,
108 talk_base::Thread* signaling_thread,
109 talk_base::Thread* worker_thread,
110 cricket::PortAllocator* port_allocator,
111 MediaStreamSignaling* mediastream_signaling);
112 virtual ~WebRtcSession();
113
wu@webrtc.org97077a32013-10-25 21:18:33 +0000114 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
115 const MediaConstraintsInterface* constraints,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000116 DTLSIdentityServiceInterface* dtls_identity_service);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117 // Deletes the voice, video and data channel and changes the session state
118 // to STATE_RECEIVEDTERMINATE.
119 void Terminate();
120
121 void RegisterIceObserver(IceObserver* observer) {
122 ice_observer_ = observer;
123 }
124
125 virtual cricket::VoiceChannel* voice_channel() {
126 return voice_channel_.get();
127 }
128 virtual cricket::VideoChannel* video_channel() {
129 return video_channel_.get();
130 }
131 virtual cricket::DataChannel* data_channel() {
132 return data_channel_.get();
133 }
134
wu@webrtc.org364f2042013-11-20 21:49:41 +0000135 void SetSecurePolicy(cricket::SecureMediaPolicy secure_policy);
136 cricket::SecureMediaPolicy SecurePolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000137
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000138 // Get current ssl role from transport.
139 bool GetSslRole(talk_base::SSLRole* role);
140
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000141 // Generic error message callback from WebRtcSession.
142 // TODO - It may be necessary to supply error code as well.
143 sigslot::signal0<> SignalError;
144
wu@webrtc.org91053e72013-08-10 07:18:04 +0000145 void CreateOffer(CreateSessionDescriptionObserver* observer,
146 const MediaConstraintsInterface* constraints);
147 void CreateAnswer(CreateSessionDescriptionObserver* observer,
148 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000149 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000150 bool SetLocalDescription(SessionDescriptionInterface* desc,
151 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000152 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000153 bool SetRemoteDescription(SessionDescriptionInterface* desc,
154 std::string* err_desc);
155 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
156 const SessionDescriptionInterface* local_description() const {
157 return local_desc_.get();
158 }
159 const SessionDescriptionInterface* remote_description() const {
160 return remote_desc_.get();
161 }
162
163 // Get the id used as a media stream track's "id" field from ssrc.
164 virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id);
165
166 // AudioMediaProviderInterface implementation.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000167 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
168 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000169 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000170 const cricket::AudioOptions& options,
171 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172
173 // Implements VideoMediaProviderInterface.
174 virtual bool SetCaptureDevice(uint32 ssrc,
175 cricket::VideoCapturer* camera) OVERRIDE;
176 virtual void SetVideoPlayout(uint32 ssrc,
177 bool enable,
178 cricket::VideoRenderer* renderer) OVERRIDE;
179 virtual void SetVideoSend(uint32 ssrc, bool enable,
180 const cricket::VideoOptions* options) OVERRIDE;
181
182 // Implements DtmfProviderInterface.
183 virtual bool CanInsertDtmf(const std::string& track_id);
184 virtual bool InsertDtmf(const std::string& track_id,
185 int code, int duration);
186 virtual sigslot::signal0<>* GetOnDestroyedSignal();
187
wu@webrtc.org78187522013-10-07 23:32:02 +0000188 // Implements DataChannelProviderInterface.
189 virtual bool SendData(const cricket::SendDataParams& params,
190 const talk_base::Buffer& payload,
191 cricket::SendDataResult* result) OVERRIDE;
192 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
193 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000194 virtual void AddSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000195 virtual void RemoveSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000196 virtual bool ReadyToSendData() const OVERRIDE;
wu@webrtc.org78187522013-10-07 23:32:02 +0000197
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000198 talk_base::scoped_refptr<DataChannel> CreateDataChannel(
199 const std::string& label,
henrika@webrtc.org44461fa2014-01-13 09:35:02 +0000200 const DataChannelInit* config);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000201
202 cricket::DataChannelType data_channel_type() const;
203
wu@webrtc.org91053e72013-08-10 07:18:04 +0000204 bool IceRestartPending() const;
205
206 void ResetIceRestartLatch();
207
208 // Called when an SSLIdentity is generated or retrieved by
209 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
210 void OnIdentityReady(talk_base::SSLIdentity* identity);
211
212 // For unit test.
213 bool waiting_for_identity() const;
214
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000215 private:
216 // Indicates the type of SessionDescription in a call to SetLocalDescription
217 // and SetRemoteDescription.
218 enum Action {
219 kOffer,
220 kPrAnswer,
221 kAnswer,
222 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000223
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000224 // Invokes ConnectChannels() on transport proxies, which initiates ice
225 // candidates allocation.
226 bool StartCandidatesAllocation();
227 bool UpdateSessionState(Action action, cricket::ContentSource source,
228 const cricket::SessionDescription* desc,
229 std::string* err_desc);
230 static Action GetAction(const std::string& type);
231
232 // Transport related callbacks, override from cricket::BaseSession.
233 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
234 virtual void OnTransportConnecting(cricket::Transport* transport);
235 virtual void OnTransportWritable(cricket::Transport* transport);
236 virtual void OnTransportProxyCandidatesReady(
237 cricket::TransportProxy* proxy,
238 const cricket::Candidates& candidates);
239 virtual void OnCandidatesAllocationDone();
240
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000241 // Creates local session description with audio and video contents.
242 bool CreateDefaultLocalDescription();
243 // Enables media channels to allow sending of media.
244 void EnableChannels();
245 // Creates a JsepIceCandidate and adds it to the local session description
246 // and notify observers. Called when a new local candidate have been found.
247 void ProcessNewLocalCandidate(const std::string& content_name,
248 const cricket::Candidates& candidates);
249 // Returns the media index for a local ice candidate given the content name.
250 // Returns false if the local session description does not have a media
251 // content called |content_name|.
252 bool GetLocalCandidateMediaIndex(const std::string& content_name,
253 int* sdp_mline_index);
254 // Uses all remote candidates in |remote_desc| in this session.
255 bool UseCandidatesInSessionDescription(
256 const SessionDescriptionInterface* remote_desc);
257 // Uses |candidate| in this session.
258 bool UseCandidate(const IceCandidateInterface* candidate);
259 // Deletes the corresponding channel of contents that don't exist in |desc|.
260 // |desc| can be null. This means that all channels are deleted.
261 void RemoveUnusedChannelsAndTransports(
262 const cricket::SessionDescription* desc);
263
264 // Allocates media channels based on the |desc|. If |desc| doesn't have
265 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
266 // This method will also delete any existing media channels before creating.
267 bool CreateChannels(const cricket::SessionDescription* desc);
268
269 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000270 bool CreateVoiceChannel(const cricket::ContentInfo* content);
271 bool CreateVideoChannel(const cricket::ContentInfo* content);
272 bool CreateDataChannel(const cricket::ContentInfo* content);
273
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274 // Copy the candidates from |saved_candidates_| to |dest_desc|.
275 // The |saved_candidates_| will be cleared after this function call.
276 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
277
henrika@webrtc.org44461fa2014-01-13 09:35:02 +0000278 void OnNewDataChannelReceived(const std::string& label,
279 const DataChannelInit& init);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000280
281 bool GetLocalTrackId(uint32 ssrc, std::string* track_id);
282 bool GetRemoteTrackId(uint32 ssrc, std::string* track_id);
283
284 std::string BadStateErrMsg(const std::string& type, State state);
285 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
286
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000287 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000288 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000289 // Below methods are helper methods which verifies SDP.
290 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
291 cricket::ContentSource source,
292 std::string* error_desc);
293
294 // Check if a call to SetLocalDescription is acceptable with |action|.
295 bool ExpectSetLocalDescription(Action action);
296 // Check if a call to SetRemoteDescription is acceptable with |action|.
297 bool ExpectSetRemoteDescription(Action action);
298 // Verifies a=setup attribute as per RFC 5763.
299 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
300 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000301
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000302 talk_base::scoped_ptr<cricket::VoiceChannel> voice_channel_;
303 talk_base::scoped_ptr<cricket::VideoChannel> video_channel_;
304 talk_base::scoped_ptr<cricket::DataChannel> data_channel_;
305 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000306 MediaStreamSignaling* mediastream_signaling_;
307 IceObserver* ice_observer_;
308 PeerConnectionInterface::IceConnectionState ice_connection_state_;
309 talk_base::scoped_ptr<SessionDescriptionInterface> local_desc_;
310 talk_base::scoped_ptr<SessionDescriptionInterface> remote_desc_;
311 // Candidates that arrived before the remote description was set.
312 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000313 // If the remote peer is using a older version of implementation.
314 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000315 bool dtls_enabled_;
wu@webrtc.orgde305012013-10-31 15:40:38 +0000316 // Flag will be set based on the constraint value.
317 bool dscp_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000318 // Specifies which kind of data channel is allowed. This is controlled
319 // by the chrome command-line flag and constraints:
320 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
321 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
322 // not set or false, SCTP is allowed (DCT_SCTP);
323 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
324 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
325 cricket::DataChannelType data_channel_type_;
326 talk_base::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000327
328 talk_base::scoped_ptr<WebRtcSessionDescriptionFactory>
329 webrtc_session_desc_factory_;
330
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000331 sigslot::signal0<> SignalVoiceChannelDestroyed;
332 sigslot::signal0<> SignalVideoChannelDestroyed;
333 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000334
wu@webrtc.org364f2042013-11-20 21:49:41 +0000335 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
336};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000337} // namespace webrtc
338
339#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_