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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
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000060extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000061extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000062extern const char kInvalidCandidates[];
63extern const char kInvalidSdp[];
64extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000065extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000066extern const char kSdpWithoutDtlsFingerprint[];
67extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000068extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000069extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000070extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000071extern const char kSessionErrorDesc[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000072
73// ICE state callback interface.
74class IceObserver {
75 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000076 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077 // Called any time the IceConnectionState changes
78 virtual void OnIceConnectionChange(
79 PeerConnectionInterface::IceConnectionState new_state) {}
80 // Called any time the IceGatheringState changes
81 virtual void OnIceGatheringChange(
82 PeerConnectionInterface::IceGatheringState new_state) {}
83 // New Ice candidate have been found.
84 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
85 // All Ice candidates have been found.
86 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
87 // (via PeerConnectionObserver)
88 virtual void OnIceComplete() {}
89
90 protected:
91 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000092
93 private:
94 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000095};
96
97class WebRtcSession : public cricket::BaseSession,
98 public AudioProviderInterface,
99 public DataChannelFactory,
100 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000101 public DtmfProviderInterface,
102 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103 public:
104 WebRtcSession(cricket::ChannelManager* channel_manager,
105 talk_base::Thread* signaling_thread,
106 talk_base::Thread* worker_thread,
107 cricket::PortAllocator* port_allocator,
108 MediaStreamSignaling* mediastream_signaling);
109 virtual ~WebRtcSession();
110
wu@webrtc.org97077a32013-10-25 21:18:33 +0000111 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
112 const MediaConstraintsInterface* constraints,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000113 DTLSIdentityServiceInterface* dtls_identity_service);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000114 // Deletes the voice, video and data channel and changes the session state
115 // to STATE_RECEIVEDTERMINATE.
116 void Terminate();
117
118 void RegisterIceObserver(IceObserver* observer) {
119 ice_observer_ = observer;
120 }
121
122 virtual cricket::VoiceChannel* voice_channel() {
123 return voice_channel_.get();
124 }
125 virtual cricket::VideoChannel* video_channel() {
126 return video_channel_.get();
127 }
128 virtual cricket::DataChannel* data_channel() {
129 return data_channel_.get();
130 }
131
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000132 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
133 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000134
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000135 // Get current ssl role from transport.
136 bool GetSslRole(talk_base::SSLRole* role);
137
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000138 // Generic error message callback from WebRtcSession.
139 // TODO - It may be necessary to supply error code as well.
140 sigslot::signal0<> SignalError;
141
wu@webrtc.org91053e72013-08-10 07:18:04 +0000142 void CreateOffer(CreateSessionDescriptionObserver* observer,
143 const MediaConstraintsInterface* constraints);
144 void CreateAnswer(CreateSessionDescriptionObserver* observer,
145 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000146 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000147 bool SetLocalDescription(SessionDescriptionInterface* desc,
148 std::string* err_desc);
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 SetRemoteDescription(SessionDescriptionInterface* desc,
151 std::string* err_desc);
152 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
153 const SessionDescriptionInterface* local_description() const {
154 return local_desc_.get();
155 }
156 const SessionDescriptionInterface* remote_description() const {
157 return remote_desc_.get();
158 }
159
160 // Get the id used as a media stream track's "id" field from ssrc.
161 virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id);
162
163 // AudioMediaProviderInterface implementation.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000164 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
165 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000166 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000167 const cricket::AudioOptions& options,
168 cricket::AudioRenderer* renderer) OVERRIDE;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000169 virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000170
171 // Implements VideoMediaProviderInterface.
172 virtual bool SetCaptureDevice(uint32 ssrc,
173 cricket::VideoCapturer* camera) OVERRIDE;
174 virtual void SetVideoPlayout(uint32 ssrc,
175 bool enable,
176 cricket::VideoRenderer* renderer) OVERRIDE;
177 virtual void SetVideoSend(uint32 ssrc, bool enable,
178 const cricket::VideoOptions* options) OVERRIDE;
179
180 // Implements DtmfProviderInterface.
181 virtual bool CanInsertDtmf(const std::string& track_id);
182 virtual bool InsertDtmf(const std::string& track_id,
183 int code, int duration);
184 virtual sigslot::signal0<>* GetOnDestroyedSignal();
185
wu@webrtc.org78187522013-10-07 23:32:02 +0000186 // Implements DataChannelProviderInterface.
187 virtual bool SendData(const cricket::SendDataParams& params,
188 const talk_base::Buffer& payload,
189 cricket::SendDataResult* result) OVERRIDE;
190 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
191 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000192 virtual void AddSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000193 virtual void RemoveSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000194 virtual bool ReadyToSendData() const OVERRIDE;
wu@webrtc.org78187522013-10-07 23:32:02 +0000195
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000196 // Implements DataChannelFactory.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197 talk_base::scoped_refptr<DataChannel> CreateDataChannel(
198 const std::string& label,
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000199 const InternalDataChannelInit* config) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000200
201 cricket::DataChannelType data_channel_type() const;
202
wu@webrtc.org91053e72013-08-10 07:18:04 +0000203 bool IceRestartPending() const;
204
205 void ResetIceRestartLatch();
206
207 // Called when an SSLIdentity is generated or retrieved by
208 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
209 void OnIdentityReady(talk_base::SSLIdentity* identity);
210
211 // For unit test.
212 bool waiting_for_identity() const;
213
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000214 private:
215 // Indicates the type of SessionDescription in a call to SetLocalDescription
216 // and SetRemoteDescription.
217 enum Action {
218 kOffer,
219 kPrAnswer,
220 kAnswer,
221 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000222
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 // Invokes ConnectChannels() on transport proxies, which initiates ice
224 // candidates allocation.
225 bool StartCandidatesAllocation();
226 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000227 std::string* err_desc);
228 static Action GetAction(const std::string& type);
229
230 // Transport related callbacks, override from cricket::BaseSession.
231 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
232 virtual void OnTransportConnecting(cricket::Transport* transport);
233 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000234 virtual void OnTransportCompleted(cricket::Transport* transport);
235 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000236 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.orgaebb1ad2014-01-14 10:00:58 +0000278 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
279 // messages.
280 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
281 const cricket::ReceiveDataParams& params,
282 const talk_base::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000283
284 bool GetLocalTrackId(uint32 ssrc, std::string* track_id);
285 bool GetRemoteTrackId(uint32 ssrc, std::string* track_id);
286
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000287 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000288 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
289
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000290 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000291 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000292 // Below methods are helper methods which verifies SDP.
293 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
294 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000295 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000296
297 // Check if a call to SetLocalDescription is acceptable with |action|.
298 bool ExpectSetLocalDescription(Action action);
299 // Check if a call to SetRemoteDescription is acceptable with |action|.
300 bool ExpectSetRemoteDescription(Action action);
301 // Verifies a=setup attribute as per RFC 5763.
302 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
303 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000304
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000305 std::string GetSessionErrorMsg();
306
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 talk_base::scoped_ptr<cricket::VoiceChannel> voice_channel_;
308 talk_base::scoped_ptr<cricket::VideoChannel> video_channel_;
309 talk_base::scoped_ptr<cricket::DataChannel> data_channel_;
310 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000311 MediaStreamSignaling* mediastream_signaling_;
312 IceObserver* ice_observer_;
313 PeerConnectionInterface::IceConnectionState ice_connection_state_;
314 talk_base::scoped_ptr<SessionDescriptionInterface> local_desc_;
315 talk_base::scoped_ptr<SessionDescriptionInterface> remote_desc_;
316 // Candidates that arrived before the remote description was set.
317 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000318 // If the remote peer is using a older version of implementation.
319 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000320 bool dtls_enabled_;
wu@webrtc.orgde305012013-10-31 15:40:38 +0000321 // Flag will be set based on the constraint value.
322 bool dscp_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000323 // Specifies which kind of data channel is allowed. This is controlled
324 // by the chrome command-line flag and constraints:
325 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
326 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
327 // not set or false, SCTP is allowed (DCT_SCTP);
328 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
329 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
330 cricket::DataChannelType data_channel_type_;
331 talk_base::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000332
333 talk_base::scoped_ptr<WebRtcSessionDescriptionFactory>
334 webrtc_session_desc_factory_;
335
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000336 sigslot::signal0<> SignalVoiceChannelDestroyed;
337 sigslot::signal0<> SignalVideoChannelDestroyed;
338 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000339
wu@webrtc.org364f2042013-11-20 21:49:41 +0000340 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
341};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000342} // namespace webrtc
343
344#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_