blob: cda98fdd9d945694d218ac21d67ef19433bfcfb0 [file] [log] [blame]
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// This file contains the PeerConnection interface as defined in
29// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
30// Applications must use this interface to implement peerconnection.
31// PeerConnectionFactory class provides factory methods to create
32// peerconnection, mediastream and media tracks objects.
33//
34// The Following steps are needed to setup a typical call using Jsep.
35// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
36// information about input parameters.
37// 2. Create a PeerConnection object. Provide a configuration string which
38// points either to stun or turn server to generate ICE candidates and provide
39// an object that implements the PeerConnectionObserver interface.
40// 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory
41// and add it to PeerConnection by calling AddStream.
42// 4. Create an offer and serialize it and send it to the remote peer.
43// 5. Once an ice candidate have been found PeerConnection will call the
44// observer function OnIceCandidate. The candidates must also be serialized and
45// sent to the remote peer.
46// 6. Once an answer is received from the remote peer, call
47// SetLocalSessionDescription with the offer and SetRemoteSessionDescription
48// with the remote answer.
49// 7. Once a remote candidate is received from the remote peer, provide it to
50// the peerconnection by calling AddIceCandidate.
51
52
53// The Receiver of a call can decide to accept or reject the call.
54// This decision will be taken by the application not peerconnection.
55// If application decides to accept the call
56// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
57// 2. Create a new PeerConnection.
58// 3. Provide the remote offer to the new PeerConnection object by calling
59// SetRemoteSessionDescription.
60// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
61// back to the remote peer.
62// 5. Provide the local answer to the new PeerConnection by calling
63// SetLocalSessionDescription with the answer.
64// 6. Provide the remote ice candidates by calling AddIceCandidate.
65// 7. Once a candidate have been found PeerConnection will call the observer
66// function OnIceCandidate. Send these candidates to the remote peer.
67
68#ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
69#define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
70
71#include <string>
72#include <vector>
73
74#include "talk/app/webrtc/datachannelinterface.h"
75#include "talk/app/webrtc/dtmfsenderinterface.h"
76#include "talk/app/webrtc/jsep.h"
77#include "talk/app/webrtc/mediastreaminterface.h"
78#include "talk/app/webrtc/statstypes.h"
wu@webrtc.orga8910d22014-01-23 22:12:45 +000079#include "talk/base/fileutils.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080#include "talk/base/socketaddress.h"
81
82namespace talk_base {
83class Thread;
84}
85
86namespace cricket {
87class PortAllocator;
88class WebRtcVideoDecoderFactory;
89class WebRtcVideoEncoderFactory;
90}
91
92namespace webrtc {
93class AudioDeviceModule;
94class MediaConstraintsInterface;
95
96// MediaStream container interface.
97class StreamCollectionInterface : public talk_base::RefCountInterface {
98 public:
99 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
100 virtual size_t count() = 0;
101 virtual MediaStreamInterface* at(size_t index) = 0;
102 virtual MediaStreamInterface* find(const std::string& label) = 0;
103 virtual MediaStreamTrackInterface* FindAudioTrack(
104 const std::string& id) = 0;
105 virtual MediaStreamTrackInterface* FindVideoTrack(
106 const std::string& id) = 0;
107
108 protected:
109 // Dtor protected as objects shouldn't be deleted via this interface.
110 ~StreamCollectionInterface() {}
111};
112
113class StatsObserver : public talk_base::RefCountInterface {
114 public:
115 virtual void OnComplete(const std::vector<StatsReport>& reports) = 0;
116
117 protected:
118 virtual ~StatsObserver() {}
119};
120
121class PeerConnectionInterface : public talk_base::RefCountInterface {
122 public:
123 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
124 enum SignalingState {
125 kStable,
126 kHaveLocalOffer,
127 kHaveLocalPrAnswer,
128 kHaveRemoteOffer,
129 kHaveRemotePrAnswer,
130 kClosed,
131 };
132
133 // TODO(bemasc): Remove IceState when callers are changed to
134 // IceConnection/GatheringState.
135 enum IceState {
136 kIceNew,
137 kIceGathering,
138 kIceWaiting,
139 kIceChecking,
140 kIceConnected,
141 kIceCompleted,
142 kIceFailed,
143 kIceClosed,
144 };
145
146 enum IceGatheringState {
147 kIceGatheringNew,
148 kIceGatheringGathering,
149 kIceGatheringComplete
150 };
151
152 enum IceConnectionState {
153 kIceConnectionNew,
154 kIceConnectionChecking,
155 kIceConnectionConnected,
156 kIceConnectionCompleted,
157 kIceConnectionFailed,
158 kIceConnectionDisconnected,
159 kIceConnectionClosed,
160 };
161
162 struct IceServer {
163 std::string uri;
164 std::string username;
165 std::string password;
166 };
167 typedef std::vector<IceServer> IceServers;
168
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000169 // Used by GetStats to decide which stats to include in the stats reports.
170 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
171 // |kStatsOutputLevelDebug| includes both the standard stats and additional
172 // stats for debugging purposes.
173 enum StatsOutputLevel {
174 kStatsOutputLevelStandard,
175 kStatsOutputLevelDebug,
176 };
177
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000178 // Accessor methods to active local streams.
179 virtual talk_base::scoped_refptr<StreamCollectionInterface>
180 local_streams() = 0;
181
182 // Accessor methods to remote streams.
183 virtual talk_base::scoped_refptr<StreamCollectionInterface>
184 remote_streams() = 0;
185
186 // Add a new MediaStream to be sent on this PeerConnection.
187 // Note that a SessionDescription negotiation is needed before the
188 // remote peer can receive the stream.
189 virtual bool AddStream(MediaStreamInterface* stream,
190 const MediaConstraintsInterface* constraints) = 0;
191
192 // Remove a MediaStream from this PeerConnection.
193 // Note that a SessionDescription negotiation is need before the
194 // remote peer is notified.
195 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
196
197 // Returns pointer to the created DtmfSender on success.
198 // Otherwise returns NULL.
199 virtual talk_base::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
200 AudioTrackInterface* track) = 0;
201
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000202 virtual bool GetStats(StatsObserver* observer,
203 MediaStreamTrackInterface* track,
204 StatsOutputLevel level) = 0;
205
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000206 virtual talk_base::scoped_refptr<DataChannelInterface> CreateDataChannel(
207 const std::string& label,
208 const DataChannelInit* config) = 0;
209
210 virtual const SessionDescriptionInterface* local_description() const = 0;
211 virtual const SessionDescriptionInterface* remote_description() const = 0;
212
213 // Create a new offer.
214 // The CreateSessionDescriptionObserver callback will be called when done.
215 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
216 const MediaConstraintsInterface* constraints) = 0;
217 // Create an answer to an offer.
218 // The CreateSessionDescriptionObserver callback will be called when done.
219 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
220 const MediaConstraintsInterface* constraints) = 0;
221 // Sets the local session description.
222 // JsepInterface takes the ownership of |desc| even if it fails.
223 // The |observer| callback will be called when done.
224 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
225 SessionDescriptionInterface* desc) = 0;
226 // Sets the remote session description.
227 // JsepInterface takes the ownership of |desc| even if it fails.
228 // The |observer| callback will be called when done.
229 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
230 SessionDescriptionInterface* desc) = 0;
231 // Restarts or updates the ICE Agent process of gathering local candidates
232 // and pinging remote candidates.
233 virtual bool UpdateIce(const IceServers& configuration,
234 const MediaConstraintsInterface* constraints) = 0;
235 // Provides a remote candidate to the ICE Agent.
236 // A copy of the |candidate| will be created and added to the remote
237 // description. So the caller of this method still has the ownership of the
238 // |candidate|.
239 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
240 // take the ownership of the |candidate|.
241 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
242
243 // Returns the current SignalingState.
244 virtual SignalingState signaling_state() = 0;
245
246 // TODO(bemasc): Remove ice_state when callers are changed to
247 // IceConnection/GatheringState.
248 // Returns the current IceState.
249 virtual IceState ice_state() = 0;
250 virtual IceConnectionState ice_connection_state() = 0;
251 virtual IceGatheringState ice_gathering_state() = 0;
252
253 // Terminates all media and closes the transport.
254 virtual void Close() = 0;
255
256 protected:
257 // Dtor protected as objects shouldn't be deleted via this interface.
258 ~PeerConnectionInterface() {}
259};
260
261// PeerConnection callback interface. Application should implement these
262// methods.
263class PeerConnectionObserver {
264 public:
265 enum StateType {
266 kSignalingState,
267 kIceState,
268 };
269
270 virtual void OnError() = 0;
271
272 // Triggered when the SignalingState changed.
273 virtual void OnSignalingChange(
274 PeerConnectionInterface::SignalingState new_state) {}
275
276 // Triggered when SignalingState or IceState have changed.
277 // TODO(bemasc): Remove once callers transition to OnSignalingChange.
278 virtual void OnStateChange(StateType state_changed) {}
279
280 // Triggered when media is received on a new stream from remote peer.
281 virtual void OnAddStream(MediaStreamInterface* stream) = 0;
282
283 // Triggered when a remote peer close a stream.
284 virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
285
286 // Triggered when a remote peer open a data channel.
287 // TODO(perkj): Make pure virtual.
288 virtual void OnDataChannel(DataChannelInterface* data_channel) {}
289
mallinath@webrtc.org0d92ef62014-01-22 02:21:22 +0000290 // Triggered when renegotiation is needed, for example the ICE has restarted.
fischman@webrtc.orgd7568a02014-01-13 22:04:12 +0000291 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000292
293 // Called any time the IceConnectionState changes
294 virtual void OnIceConnectionChange(
295 PeerConnectionInterface::IceConnectionState new_state) {}
296
297 // Called any time the IceGatheringState changes
298 virtual void OnIceGatheringChange(
299 PeerConnectionInterface::IceGatheringState new_state) {}
300
301 // New Ice candidate have been found.
302 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
303
304 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
305 // All Ice candidates have been found.
306 virtual void OnIceComplete() {}
307
308 protected:
309 // Dtor protected as objects shouldn't be deleted via this interface.
310 ~PeerConnectionObserver() {}
311};
312
313// Factory class used for creating cricket::PortAllocator that is used
314// for ICE negotiation.
315class PortAllocatorFactoryInterface : public talk_base::RefCountInterface {
316 public:
317 struct StunConfiguration {
318 StunConfiguration(const std::string& address, int port)
319 : server(address, port) {}
320 // STUN server address and port.
321 talk_base::SocketAddress server;
322 };
323
324 struct TurnConfiguration {
325 TurnConfiguration(const std::string& address,
326 int port,
327 const std::string& username,
328 const std::string& password,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000329 const std::string& transport_type,
330 bool secure)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000331 : server(address, port),
332 username(username),
333 password(password),
wu@webrtc.org91053e72013-08-10 07:18:04 +0000334 transport_type(transport_type),
335 secure(secure) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000336 talk_base::SocketAddress server;
337 std::string username;
338 std::string password;
339 std::string transport_type;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000340 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000341 };
342
343 virtual cricket::PortAllocator* CreatePortAllocator(
344 const std::vector<StunConfiguration>& stun_servers,
345 const std::vector<TurnConfiguration>& turn_configurations) = 0;
346
347 protected:
348 PortAllocatorFactoryInterface() {}
349 ~PortAllocatorFactoryInterface() {}
350};
351
352// Used to receive callbacks of DTLS identity requests.
353class DTLSIdentityRequestObserver : public talk_base::RefCountInterface {
354 public:
355 virtual void OnFailure(int error) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000356 virtual void OnSuccess(const std::string& der_cert,
357 const std::string& der_private_key) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000358 protected:
359 virtual ~DTLSIdentityRequestObserver() {}
360};
361
362class DTLSIdentityServiceInterface {
363 public:
364 // Asynchronously request a DTLS identity, including a self-signed certificate
365 // and the private key used to sign the certificate, from the identity store
366 // for the given identity name.
367 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
368 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
369 // called with an error code if the request failed.
370 //
371 // Only one request can be made at a time. If a second request is called
372 // before the first one completes, RequestIdentity will abort and return
373 // false.
374 //
375 // |identity_name| is an internal name selected by the client to identify an
376 // identity within an origin. E.g. an web site may cache the certificates used
377 // to communicate with differnent peers under different identity names.
378 //
379 // |common_name| is the common name used to generate the certificate. If the
380 // certificate already exists in the store, |common_name| is ignored.
381 //
382 // |observer| is the object to receive success or failure callbacks.
383 //
384 // Returns true if either OnFailure or OnSuccess will be called.
385 virtual bool RequestIdentity(
386 const std::string& identity_name,
387 const std::string& common_name,
388 DTLSIdentityRequestObserver* observer) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000389
390 virtual ~DTLSIdentityServiceInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000391};
392
393// PeerConnectionFactoryInterface is the factory interface use for creating
394// PeerConnection, MediaStream and media tracks.
395// PeerConnectionFactoryInterface will create required libjingle threads,
396// socket and network manager factory classes for networking.
397// If an application decides to provide its own threads and network
398// implementation of these classes it should use the alternate
399// CreatePeerConnectionFactory method which accepts threads as input and use the
400// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
401// argument.
402class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
403 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +0000404 class Options {
405 public:
406 Options() :
wu@webrtc.org97077a32013-10-25 21:18:33 +0000407 disable_encryption(false),
408 disable_sctp_data_channels(false) {
409 }
wu@webrtc.org97077a32013-10-25 21:18:33 +0000410 bool disable_encryption;
411 bool disable_sctp_data_channels;
412 };
413
414 virtual void SetOptions(const Options& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000415 virtual talk_base::scoped_refptr<PeerConnectionInterface>
416 CreatePeerConnection(
417 const PeerConnectionInterface::IceServers& configuration,
418 const MediaConstraintsInterface* constraints,
419 DTLSIdentityServiceInterface* dtls_identity_service,
420 PeerConnectionObserver* observer) = 0;
421 virtual talk_base::scoped_refptr<PeerConnectionInterface>
422 CreatePeerConnection(
423 const PeerConnectionInterface::IceServers& configuration,
424 const MediaConstraintsInterface* constraints,
425 PortAllocatorFactoryInterface* allocator_factory,
426 DTLSIdentityServiceInterface* dtls_identity_service,
427 PeerConnectionObserver* observer) = 0;
428 virtual talk_base::scoped_refptr<MediaStreamInterface>
429 CreateLocalMediaStream(const std::string& label) = 0;
430
431 // Creates a AudioSourceInterface.
432 // |constraints| decides audio processing settings but can be NULL.
433 virtual talk_base::scoped_refptr<AudioSourceInterface> CreateAudioSource(
434 const MediaConstraintsInterface* constraints) = 0;
435
436 // Creates a VideoSourceInterface. The new source take ownership of
437 // |capturer|. |constraints| decides video resolution and frame rate but can
438 // be NULL.
439 virtual talk_base::scoped_refptr<VideoSourceInterface> CreateVideoSource(
440 cricket::VideoCapturer* capturer,
441 const MediaConstraintsInterface* constraints) = 0;
442
443 // Creates a new local VideoTrack. The same |source| can be used in several
444 // tracks.
445 virtual talk_base::scoped_refptr<VideoTrackInterface>
446 CreateVideoTrack(const std::string& label,
447 VideoSourceInterface* source) = 0;
448
449 // Creates an new AudioTrack. At the moment |source| can be NULL.
450 virtual talk_base::scoped_refptr<AudioTrackInterface>
451 CreateAudioTrack(const std::string& label,
452 AudioSourceInterface* source) = 0;
453
wu@webrtc.orga9890802013-12-13 00:21:03 +0000454 // Starts AEC dump using existing file. Takes ownership of |file| and passes
455 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000456 // the ownerhip. If the operation fails, the file will be closed.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000457 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000458 // http://crbug.com/264611.
459 virtual bool StartAecDump(talk_base::PlatformFile file) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03 +0000460
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000461 protected:
462 // Dtor and ctor protected as objects shouldn't be created or deleted via
463 // this interface.
464 PeerConnectionFactoryInterface() {}
465 ~PeerConnectionFactoryInterface() {} // NOLINT
466};
467
468// Create a new instance of PeerConnectionFactoryInterface.
469talk_base::scoped_refptr<PeerConnectionFactoryInterface>
470CreatePeerConnectionFactory();
471
472// Create a new instance of PeerConnectionFactoryInterface.
473// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
474// |decoder_factory| transferred to the returned factory.
475talk_base::scoped_refptr<PeerConnectionFactoryInterface>
476CreatePeerConnectionFactory(
477 talk_base::Thread* worker_thread,
478 talk_base::Thread* signaling_thread,
479 AudioDeviceModule* default_adm,
480 cricket::WebRtcVideoEncoderFactory* encoder_factory,
481 cricket::WebRtcVideoDecoderFactory* decoder_factory);
482
483} // namespace webrtc
484
485#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_