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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.org28e20752013-07-10 00:45:36 +000066extern const char kSdpWithoutCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000067extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000068extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000069extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000070extern const char kSessionErrorDesc[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071
72// ICE state callback interface.
73class IceObserver {
74 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000075 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076 // Called any time the IceConnectionState changes
77 virtual void OnIceConnectionChange(
78 PeerConnectionInterface::IceConnectionState new_state) {}
79 // Called any time the IceGatheringState changes
80 virtual void OnIceGatheringChange(
81 PeerConnectionInterface::IceGatheringState new_state) {}
82 // New Ice candidate have been found.
83 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
84 // All Ice candidates have been found.
85 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
86 // (via PeerConnectionObserver)
87 virtual void OnIceComplete() {}
88
89 protected:
90 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000091
92 private:
93 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000094};
95
96class WebRtcSession : public cricket::BaseSession,
97 public AudioProviderInterface,
98 public DataChannelFactory,
99 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000100 public DtmfProviderInterface,
101 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000102 public:
103 WebRtcSession(cricket::ChannelManager* channel_manager,
104 talk_base::Thread* signaling_thread,
105 talk_base::Thread* worker_thread,
106 cricket::PortAllocator* port_allocator,
107 MediaStreamSignaling* mediastream_signaling);
108 virtual ~WebRtcSession();
109
wu@webrtc.org97077a32013-10-25 21:18:33 +0000110 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
111 const MediaConstraintsInterface* constraints,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000112 DTLSIdentityServiceInterface* dtls_identity_service);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000113 // Deletes the voice, video and data channel and changes the session state
114 // to STATE_RECEIVEDTERMINATE.
115 void Terminate();
116
117 void RegisterIceObserver(IceObserver* observer) {
118 ice_observer_ = observer;
119 }
120
121 virtual cricket::VoiceChannel* voice_channel() {
122 return voice_channel_.get();
123 }
124 virtual cricket::VideoChannel* video_channel() {
125 return video_channel_.get();
126 }
127 virtual cricket::DataChannel* data_channel() {
128 return data_channel_.get();
129 }
130
wu@webrtc.org364f2042013-11-20 21:49:41 +0000131 void SetSecurePolicy(cricket::SecureMediaPolicy secure_policy);
132 cricket::SecureMediaPolicy SecurePolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000134 // Get current ssl role from transport.
135 bool GetSslRole(talk_base::SSLRole* role);
136
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000137 // Generic error message callback from WebRtcSession.
138 // TODO - It may be necessary to supply error code as well.
139 sigslot::signal0<> SignalError;
140
wu@webrtc.org91053e72013-08-10 07:18:04 +0000141 void CreateOffer(CreateSessionDescriptionObserver* observer,
142 const MediaConstraintsInterface* constraints);
143 void CreateAnswer(CreateSessionDescriptionObserver* observer,
144 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000145 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000146 bool SetLocalDescription(SessionDescriptionInterface* desc,
147 std::string* err_desc);
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 SetRemoteDescription(SessionDescriptionInterface* desc,
150 std::string* err_desc);
151 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
152 const SessionDescriptionInterface* local_description() const {
153 return local_desc_.get();
154 }
155 const SessionDescriptionInterface* remote_description() const {
156 return remote_desc_.get();
157 }
158
159 // Get the id used as a media stream track's "id" field from ssrc.
160 virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id);
161
162 // AudioMediaProviderInterface implementation.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000163 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
164 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000166 const cricket::AudioOptions& options,
167 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000168
169 // Implements VideoMediaProviderInterface.
170 virtual bool SetCaptureDevice(uint32 ssrc,
171 cricket::VideoCapturer* camera) OVERRIDE;
172 virtual void SetVideoPlayout(uint32 ssrc,
173 bool enable,
174 cricket::VideoRenderer* renderer) OVERRIDE;
175 virtual void SetVideoSend(uint32 ssrc, bool enable,
176 const cricket::VideoOptions* options) OVERRIDE;
177
178 // Implements DtmfProviderInterface.
179 virtual bool CanInsertDtmf(const std::string& track_id);
180 virtual bool InsertDtmf(const std::string& track_id,
181 int code, int duration);
182 virtual sigslot::signal0<>* GetOnDestroyedSignal();
183
wu@webrtc.org78187522013-10-07 23:32:02 +0000184 // Implements DataChannelProviderInterface.
185 virtual bool SendData(const cricket::SendDataParams& params,
186 const talk_base::Buffer& payload,
187 cricket::SendDataResult* result) OVERRIDE;
188 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
189 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000190 virtual void AddSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000191 virtual void RemoveSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000192 virtual bool ReadyToSendData() const OVERRIDE;
wu@webrtc.org78187522013-10-07 23:32:02 +0000193
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000194 // Implements DataChannelFactory.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000195 talk_base::scoped_refptr<DataChannel> CreateDataChannel(
196 const std::string& label,
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000197 const InternalDataChannelInit* config) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000198
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,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225 std::string* err_desc);
226 static Action GetAction(const std::string& type);
227
228 // Transport related callbacks, override from cricket::BaseSession.
229 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
230 virtual void OnTransportConnecting(cricket::Transport* transport);
231 virtual void OnTransportWritable(cricket::Transport* transport);
232 virtual void OnTransportProxyCandidatesReady(
233 cricket::TransportProxy* proxy,
234 const cricket::Candidates& candidates);
235 virtual void OnCandidatesAllocationDone();
236
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000237 // Creates local session description with audio and video contents.
238 bool CreateDefaultLocalDescription();
239 // Enables media channels to allow sending of media.
240 void EnableChannels();
241 // Creates a JsepIceCandidate and adds it to the local session description
242 // and notify observers. Called when a new local candidate have been found.
243 void ProcessNewLocalCandidate(const std::string& content_name,
244 const cricket::Candidates& candidates);
245 // Returns the media index for a local ice candidate given the content name.
246 // Returns false if the local session description does not have a media
247 // content called |content_name|.
248 bool GetLocalCandidateMediaIndex(const std::string& content_name,
249 int* sdp_mline_index);
250 // Uses all remote candidates in |remote_desc| in this session.
251 bool UseCandidatesInSessionDescription(
252 const SessionDescriptionInterface* remote_desc);
253 // Uses |candidate| in this session.
254 bool UseCandidate(const IceCandidateInterface* candidate);
255 // Deletes the corresponding channel of contents that don't exist in |desc|.
256 // |desc| can be null. This means that all channels are deleted.
257 void RemoveUnusedChannelsAndTransports(
258 const cricket::SessionDescription* desc);
259
260 // Allocates media channels based on the |desc|. If |desc| doesn't have
261 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
262 // This method will also delete any existing media channels before creating.
263 bool CreateChannels(const cricket::SessionDescription* desc);
264
265 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000266 bool CreateVoiceChannel(const cricket::ContentInfo* content);
267 bool CreateVideoChannel(const cricket::ContentInfo* content);
268 bool CreateDataChannel(const cricket::ContentInfo* content);
269
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000270 // Copy the candidates from |saved_candidates_| to |dest_desc|.
271 // The |saved_candidates_| will be cleared after this function call.
272 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
273
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000274 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
275 // messages.
276 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
277 const cricket::ReceiveDataParams& params,
278 const talk_base::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000279
280 bool GetLocalTrackId(uint32 ssrc, std::string* track_id);
281 bool GetRemoteTrackId(uint32 ssrc, std::string* track_id);
282
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000283 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000284 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
285
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000286 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000287 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000288 // Below methods are helper methods which verifies SDP.
289 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
290 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000291 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000292
293 // Check if a call to SetLocalDescription is acceptable with |action|.
294 bool ExpectSetLocalDescription(Action action);
295 // Check if a call to SetRemoteDescription is acceptable with |action|.
296 bool ExpectSetRemoteDescription(Action action);
297 // Verifies a=setup attribute as per RFC 5763.
298 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
299 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000300
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000301 std::string GetSessionErrorMsg();
302
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000303 talk_base::scoped_ptr<cricket::VoiceChannel> voice_channel_;
304 talk_base::scoped_ptr<cricket::VideoChannel> video_channel_;
305 talk_base::scoped_ptr<cricket::DataChannel> data_channel_;
306 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 MediaStreamSignaling* mediastream_signaling_;
308 IceObserver* ice_observer_;
309 PeerConnectionInterface::IceConnectionState ice_connection_state_;
310 talk_base::scoped_ptr<SessionDescriptionInterface> local_desc_;
311 talk_base::scoped_ptr<SessionDescriptionInterface> remote_desc_;
312 // Candidates that arrived before the remote description was set.
313 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000314 // If the remote peer is using a older version of implementation.
315 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000316 bool dtls_enabled_;
wu@webrtc.orgde305012013-10-31 15:40:38 +0000317 // Flag will be set based on the constraint value.
318 bool dscp_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000319 // Specifies which kind of data channel is allowed. This is controlled
320 // by the chrome command-line flag and constraints:
321 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
322 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
323 // not set or false, SCTP is allowed (DCT_SCTP);
324 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
325 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
326 cricket::DataChannelType data_channel_type_;
327 talk_base::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000328
329 talk_base::scoped_ptr<WebRtcSessionDescriptionFactory>
330 webrtc_session_desc_factory_;
331
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000332 sigslot::signal0<> SignalVoiceChannelDestroyed;
333 sigslot::signal0<> SignalVideoChannelDestroyed;
334 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000335
wu@webrtc.org364f2042013-11-20 21:49:41 +0000336 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
337};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000338} // namespace webrtc
339
340#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_