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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 {
45
46class ChannelManager;
47class DataChannel;
48class StatsReport;
49class Transport;
50class VideoCapturer;
51class BaseChannel;
52class VideoChannel;
53class VoiceChannel;
54
55} // namespace cricket
56
57namespace webrtc {
58
59class IceRestartAnswerLatch;
60class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000061class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000062
63extern const char kSetLocalSdpFailed[];
64extern const char kSetRemoteSdpFailed[];
65extern const char kCreateChannelFailed[];
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000066extern const char kBundleWithoutRtcpMux[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000067extern const char kInvalidCandidates[];
68extern const char kInvalidSdp[];
69extern const char kMlineMismatch[];
70extern const char kSdpWithoutCrypto[];
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +000071extern const char kSdpWithoutSdesAndDtlsDisabled[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000072extern const char kSdpWithoutIceUfragPwd[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000073extern const char kSessionError[];
74extern const char kUpdateStateFailed[];
75extern const char kPushDownOfferTDFailed[];
76extern const char kPushDownPranswerTDFailed[];
77extern const char kPushDownAnswerTDFailed[];
78
79// ICE state callback interface.
80class IceObserver {
81 public:
82 // Called any time the IceConnectionState changes
83 virtual void OnIceConnectionChange(
84 PeerConnectionInterface::IceConnectionState new_state) {}
85 // Called any time the IceGatheringState changes
86 virtual void OnIceGatheringChange(
87 PeerConnectionInterface::IceGatheringState new_state) {}
88 // New Ice candidate have been found.
89 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
90 // All Ice candidates have been found.
91 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
92 // (via PeerConnectionObserver)
93 virtual void OnIceComplete() {}
94
95 protected:
96 ~IceObserver() {}
97};
98
99class WebRtcSession : public cricket::BaseSession,
100 public AudioProviderInterface,
101 public DataChannelFactory,
102 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000103 public DtmfProviderInterface,
104 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000105 public:
106 WebRtcSession(cricket::ChannelManager* channel_manager,
107 talk_base::Thread* signaling_thread,
108 talk_base::Thread* worker_thread,
109 cricket::PortAllocator* port_allocator,
110 MediaStreamSignaling* mediastream_signaling);
111 virtual ~WebRtcSession();
112
wu@webrtc.org97077a32013-10-25 21:18:33 +0000113 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
114 const MediaConstraintsInterface* constraints,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000115 DTLSIdentityServiceInterface* dtls_identity_service);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000116 // Deletes the voice, video and data channel and changes the session state
117 // to STATE_RECEIVEDTERMINATE.
118 void Terminate();
119
120 void RegisterIceObserver(IceObserver* observer) {
121 ice_observer_ = observer;
122 }
123
124 virtual cricket::VoiceChannel* voice_channel() {
125 return voice_channel_.get();
126 }
127 virtual cricket::VideoChannel* video_channel() {
128 return video_channel_.get();
129 }
130 virtual cricket::DataChannel* data_channel() {
131 return data_channel_.get();
132 }
133
134 void set_secure_policy(cricket::SecureMediaPolicy secure_policy);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000135 cricket::SecureMediaPolicy secure_policy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000136
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000137 // Get current ssl role from transport.
138 bool GetSslRole(talk_base::SSLRole* role);
139
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000140 // Generic error message callback from WebRtcSession.
141 // TODO - It may be necessary to supply error code as well.
142 sigslot::signal0<> SignalError;
143
wu@webrtc.org91053e72013-08-10 07:18:04 +0000144 void CreateOffer(CreateSessionDescriptionObserver* observer,
145 const MediaConstraintsInterface* constraints);
146 void CreateAnswer(CreateSessionDescriptionObserver* observer,
147 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000148 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000149 bool SetLocalDescription(SessionDescriptionInterface* desc,
150 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000151 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000152 bool SetRemoteDescription(SessionDescriptionInterface* desc,
153 std::string* err_desc);
154 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
155 const SessionDescriptionInterface* local_description() const {
156 return local_desc_.get();
157 }
158 const SessionDescriptionInterface* remote_description() const {
159 return remote_desc_.get();
160 }
161
162 // Get the id used as a media stream track's "id" field from ssrc.
163 virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id);
164
165 // AudioMediaProviderInterface implementation.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000166 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
167 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000168 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000169 const cricket::AudioOptions& options,
170 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000171
172 // Implements VideoMediaProviderInterface.
173 virtual bool SetCaptureDevice(uint32 ssrc,
174 cricket::VideoCapturer* camera) OVERRIDE;
175 virtual void SetVideoPlayout(uint32 ssrc,
176 bool enable,
177 cricket::VideoRenderer* renderer) OVERRIDE;
178 virtual void SetVideoSend(uint32 ssrc, bool enable,
179 const cricket::VideoOptions* options) OVERRIDE;
180
181 // Implements DtmfProviderInterface.
182 virtual bool CanInsertDtmf(const std::string& track_id);
183 virtual bool InsertDtmf(const std::string& track_id,
184 int code, int duration);
185 virtual sigslot::signal0<>* GetOnDestroyedSignal();
186
wu@webrtc.org78187522013-10-07 23:32:02 +0000187 // Implements DataChannelProviderInterface.
188 virtual bool SendData(const cricket::SendDataParams& params,
189 const talk_base::Buffer& payload,
190 cricket::SendDataResult* result) OVERRIDE;
191 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
192 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
193
194
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000195 talk_base::scoped_refptr<DataChannel> CreateDataChannel(
196 const std::string& label,
197 const DataChannelInit* config);
198
199 cricket::DataChannelType data_channel_type() const;
200
wu@webrtc.org91053e72013-08-10 07:18:04 +0000201 bool IceRestartPending() const;
202
203 void ResetIceRestartLatch();
204
205 // Called when an SSLIdentity is generated or retrieved by
206 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
207 void OnIdentityReady(talk_base::SSLIdentity* identity);
208
209 // For unit test.
210 bool waiting_for_identity() const;
211
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000212 private:
213 // Indicates the type of SessionDescription in a call to SetLocalDescription
214 // and SetRemoteDescription.
215 enum Action {
216 kOffer,
217 kPrAnswer,
218 kAnswer,
219 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000220
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000221 // Invokes ConnectChannels() on transport proxies, which initiates ice
222 // candidates allocation.
223 bool StartCandidatesAllocation();
224 bool UpdateSessionState(Action action, cricket::ContentSource source,
225 const cricket::SessionDescription* desc,
226 std::string* err_desc);
227 static Action GetAction(const std::string& type);
228
229 // Transport related callbacks, override from cricket::BaseSession.
230 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
231 virtual void OnTransportConnecting(cricket::Transport* transport);
232 virtual void OnTransportWritable(cricket::Transport* transport);
233 virtual void OnTransportProxyCandidatesReady(
234 cricket::TransportProxy* proxy,
235 const cricket::Candidates& candidates);
236 virtual void OnCandidatesAllocationDone();
237
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 // Creates local session description with audio and video contents.
239 bool CreateDefaultLocalDescription();
240 // Enables media channels to allow sending of media.
241 void EnableChannels();
242 // Creates a JsepIceCandidate and adds it to the local session description
243 // and notify observers. Called when a new local candidate have been found.
244 void ProcessNewLocalCandidate(const std::string& content_name,
245 const cricket::Candidates& candidates);
246 // Returns the media index for a local ice candidate given the content name.
247 // Returns false if the local session description does not have a media
248 // content called |content_name|.
249 bool GetLocalCandidateMediaIndex(const std::string& content_name,
250 int* sdp_mline_index);
251 // Uses all remote candidates in |remote_desc| in this session.
252 bool UseCandidatesInSessionDescription(
253 const SessionDescriptionInterface* remote_desc);
254 // Uses |candidate| in this session.
255 bool UseCandidate(const IceCandidateInterface* candidate);
256 // Deletes the corresponding channel of contents that don't exist in |desc|.
257 // |desc| can be null. This means that all channels are deleted.
258 void RemoveUnusedChannelsAndTransports(
259 const cricket::SessionDescription* desc);
260
261 // Allocates media channels based on the |desc|. If |desc| doesn't have
262 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
263 // This method will also delete any existing media channels before creating.
264 bool CreateChannels(const cricket::SessionDescription* desc);
265
266 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000267 bool CreateVoiceChannel(const cricket::ContentInfo* content);
268 bool CreateVideoChannel(const cricket::ContentInfo* content);
269 bool CreateDataChannel(const cricket::ContentInfo* content);
270
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271 // Copy the candidates from |saved_candidates_| to |dest_desc|.
272 // The |saved_candidates_| will be cleared after this function call.
273 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
274
wu@webrtc.org1d1ffc92013-10-16 18:12:02 +0000275 void OnNewDataChannelReceived(const std::string& label,
276 const DataChannelInit& init);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000277
278 bool GetLocalTrackId(uint32 ssrc, std::string* track_id);
279 bool GetRemoteTrackId(uint32 ssrc, std::string* track_id);
280
281 std::string BadStateErrMsg(const std::string& type, State state);
282 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
283
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000284 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000285 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000286 // Below methods are helper methods which verifies SDP.
287 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
288 cricket::ContentSource source,
289 std::string* error_desc);
290
291 // Check if a call to SetLocalDescription is acceptable with |action|.
292 bool ExpectSetLocalDescription(Action action);
293 // Check if a call to SetRemoteDescription is acceptable with |action|.
294 bool ExpectSetRemoteDescription(Action action);
295 // Verifies a=setup attribute as per RFC 5763.
296 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
297 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000298
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000299 talk_base::scoped_ptr<cricket::VoiceChannel> voice_channel_;
300 talk_base::scoped_ptr<cricket::VideoChannel> video_channel_;
301 talk_base::scoped_ptr<cricket::DataChannel> data_channel_;
302 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000303 MediaStreamSignaling* mediastream_signaling_;
304 IceObserver* ice_observer_;
305 PeerConnectionInterface::IceConnectionState ice_connection_state_;
306 talk_base::scoped_ptr<SessionDescriptionInterface> local_desc_;
307 talk_base::scoped_ptr<SessionDescriptionInterface> remote_desc_;
308 // Candidates that arrived before the remote description was set.
309 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000310 // If the remote peer is using a older version of implementation.
311 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000312 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000313 // Specifies which kind of data channel is allowed. This is controlled
314 // by the chrome command-line flag and constraints:
315 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
316 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
317 // not set or false, SCTP is allowed (DCT_SCTP);
318 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
319 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
320 cricket::DataChannelType data_channel_type_;
321 talk_base::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000322
323 talk_base::scoped_ptr<WebRtcSessionDescriptionFactory>
324 webrtc_session_desc_factory_;
325
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000326 sigslot::signal0<> SignalVoiceChannelDestroyed;
327 sigslot::signal0<> SignalVideoChannelDestroyed;
328 sigslot::signal0<> SignalDataChannelDestroyed;
329};
330
331} // namespace webrtc
332
333#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_