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henrike@webrtc.org28e20752013-07-10 00:45:361/*
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
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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"
79#include "talk/base/socketaddress.h"
80
81namespace talk_base {
82class Thread;
83}
84
85namespace cricket {
86class PortAllocator;
87class WebRtcVideoDecoderFactory;
88class WebRtcVideoEncoderFactory;
89}
90
91namespace webrtc {
92class AudioDeviceModule;
93class MediaConstraintsInterface;
94
95// MediaStream container interface.
96class StreamCollectionInterface : public talk_base::RefCountInterface {
97 public:
98 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
99 virtual size_t count() = 0;
100 virtual MediaStreamInterface* at(size_t index) = 0;
101 virtual MediaStreamInterface* find(const std::string& label) = 0;
102 virtual MediaStreamTrackInterface* FindAudioTrack(
103 const std::string& id) = 0;
104 virtual MediaStreamTrackInterface* FindVideoTrack(
105 const std::string& id) = 0;
106
107 protected:
108 // Dtor protected as objects shouldn't be deleted via this interface.
109 ~StreamCollectionInterface() {}
110};
111
112class StatsObserver : public talk_base::RefCountInterface {
113 public:
114 virtual void OnComplete(const std::vector<StatsReport>& reports) = 0;
115
116 protected:
117 virtual ~StatsObserver() {}
118};
119
120class PeerConnectionInterface : public talk_base::RefCountInterface {
121 public:
122 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
123 enum SignalingState {
124 kStable,
125 kHaveLocalOffer,
126 kHaveLocalPrAnswer,
127 kHaveRemoteOffer,
128 kHaveRemotePrAnswer,
129 kClosed,
130 };
131
132 // TODO(bemasc): Remove IceState when callers are changed to
133 // IceConnection/GatheringState.
134 enum IceState {
135 kIceNew,
136 kIceGathering,
137 kIceWaiting,
138 kIceChecking,
139 kIceConnected,
140 kIceCompleted,
141 kIceFailed,
142 kIceClosed,
143 };
144
145 enum IceGatheringState {
146 kIceGatheringNew,
147 kIceGatheringGathering,
148 kIceGatheringComplete
149 };
150
151 enum IceConnectionState {
152 kIceConnectionNew,
153 kIceConnectionChecking,
154 kIceConnectionConnected,
155 kIceConnectionCompleted,
156 kIceConnectionFailed,
157 kIceConnectionDisconnected,
158 kIceConnectionClosed,
159 };
160
161 struct IceServer {
162 std::string uri;
163 std::string username;
164 std::string password;
165 };
166 typedef std::vector<IceServer> IceServers;
167
168 // Accessor methods to active local streams.
169 virtual talk_base::scoped_refptr<StreamCollectionInterface>
170 local_streams() = 0;
171
172 // Accessor methods to remote streams.
173 virtual talk_base::scoped_refptr<StreamCollectionInterface>
174 remote_streams() = 0;
175
176 // Add a new MediaStream to be sent on this PeerConnection.
177 // Note that a SessionDescription negotiation is needed before the
178 // remote peer can receive the stream.
179 virtual bool AddStream(MediaStreamInterface* stream,
180 const MediaConstraintsInterface* constraints) = 0;
181
182 // Remove a MediaStream from this PeerConnection.
183 // Note that a SessionDescription negotiation is need before the
184 // remote peer is notified.
185 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
186
187 // Returns pointer to the created DtmfSender on success.
188 // Otherwise returns NULL.
189 virtual talk_base::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
190 AudioTrackInterface* track) = 0;
191
192 virtual bool GetStats(StatsObserver* observer,
193 MediaStreamTrackInterface* track) = 0;
194
195 virtual talk_base::scoped_refptr<DataChannelInterface> CreateDataChannel(
196 const std::string& label,
197 const DataChannelInit* config) = 0;
198
199 virtual const SessionDescriptionInterface* local_description() const = 0;
200 virtual const SessionDescriptionInterface* remote_description() const = 0;
201
202 // Create a new offer.
203 // The CreateSessionDescriptionObserver callback will be called when done.
204 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
205 const MediaConstraintsInterface* constraints) = 0;
206 // Create an answer to an offer.
207 // The CreateSessionDescriptionObserver callback will be called when done.
208 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
209 const MediaConstraintsInterface* constraints) = 0;
210 // Sets the local session description.
211 // JsepInterface takes the ownership of |desc| even if it fails.
212 // The |observer| callback will be called when done.
213 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
214 SessionDescriptionInterface* desc) = 0;
215 // Sets the remote session description.
216 // JsepInterface takes the ownership of |desc| even if it fails.
217 // The |observer| callback will be called when done.
218 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
219 SessionDescriptionInterface* desc) = 0;
220 // Restarts or updates the ICE Agent process of gathering local candidates
221 // and pinging remote candidates.
222 virtual bool UpdateIce(const IceServers& configuration,
223 const MediaConstraintsInterface* constraints) = 0;
224 // Provides a remote candidate to the ICE Agent.
225 // A copy of the |candidate| will be created and added to the remote
226 // description. So the caller of this method still has the ownership of the
227 // |candidate|.
228 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
229 // take the ownership of the |candidate|.
230 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
231
232 // Returns the current SignalingState.
233 virtual SignalingState signaling_state() = 0;
234
235 // TODO(bemasc): Remove ice_state when callers are changed to
236 // IceConnection/GatheringState.
237 // Returns the current IceState.
238 virtual IceState ice_state() = 0;
239 virtual IceConnectionState ice_connection_state() = 0;
240 virtual IceGatheringState ice_gathering_state() = 0;
241
242 // Terminates all media and closes the transport.
243 virtual void Close() = 0;
244
245 protected:
246 // Dtor protected as objects shouldn't be deleted via this interface.
247 ~PeerConnectionInterface() {}
248};
249
250// PeerConnection callback interface. Application should implement these
251// methods.
252class PeerConnectionObserver {
253 public:
254 enum StateType {
255 kSignalingState,
256 kIceState,
257 };
258
259 virtual void OnError() = 0;
260
261 // Triggered when the SignalingState changed.
262 virtual void OnSignalingChange(
263 PeerConnectionInterface::SignalingState new_state) {}
264
265 // Triggered when SignalingState or IceState have changed.
266 // TODO(bemasc): Remove once callers transition to OnSignalingChange.
267 virtual void OnStateChange(StateType state_changed) {}
268
269 // Triggered when media is received on a new stream from remote peer.
270 virtual void OnAddStream(MediaStreamInterface* stream) = 0;
271
272 // Triggered when a remote peer close a stream.
273 virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
274
275 // Triggered when a remote peer open a data channel.
276 // TODO(perkj): Make pure virtual.
277 virtual void OnDataChannel(DataChannelInterface* data_channel) {}
278
279 // Triggered when renegotation is needed, for example the ICE has restarted.
280 virtual void OnRenegotiationNeeded() {}
281
282 // Called any time the IceConnectionState changes
283 virtual void OnIceConnectionChange(
284 PeerConnectionInterface::IceConnectionState new_state) {}
285
286 // Called any time the IceGatheringState changes
287 virtual void OnIceGatheringChange(
288 PeerConnectionInterface::IceGatheringState new_state) {}
289
290 // New Ice candidate have been found.
291 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
292
293 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
294 // All Ice candidates have been found.
295 virtual void OnIceComplete() {}
296
297 protected:
298 // Dtor protected as objects shouldn't be deleted via this interface.
299 ~PeerConnectionObserver() {}
300};
301
302// Factory class used for creating cricket::PortAllocator that is used
303// for ICE negotiation.
304class PortAllocatorFactoryInterface : public talk_base::RefCountInterface {
305 public:
306 struct StunConfiguration {
307 StunConfiguration(const std::string& address, int port)
308 : server(address, port) {}
309 // STUN server address and port.
310 talk_base::SocketAddress server;
311 };
312
313 struct TurnConfiguration {
314 TurnConfiguration(const std::string& address,
315 int port,
316 const std::string& username,
317 const std::string& password,
wu@webrtc.org91053e72013-08-10 07:18:04318 const std::string& transport_type,
319 bool secure)
henrike@webrtc.org28e20752013-07-10 00:45:36320 : server(address, port),
321 username(username),
322 password(password),
wu@webrtc.org91053e72013-08-10 07:18:04323 transport_type(transport_type),
324 secure(secure) {}
henrike@webrtc.org28e20752013-07-10 00:45:36325 talk_base::SocketAddress server;
326 std::string username;
327 std::string password;
328 std::string transport_type;
wu@webrtc.org91053e72013-08-10 07:18:04329 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36330 };
331
332 virtual cricket::PortAllocator* CreatePortAllocator(
333 const std::vector<StunConfiguration>& stun_servers,
334 const std::vector<TurnConfiguration>& turn_configurations) = 0;
335
336 protected:
337 PortAllocatorFactoryInterface() {}
338 ~PortAllocatorFactoryInterface() {}
339};
340
341// Used to receive callbacks of DTLS identity requests.
342class DTLSIdentityRequestObserver : public talk_base::RefCountInterface {
343 public:
344 virtual void OnFailure(int error) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04345 virtual void OnSuccess(const std::string& der_cert,
346 const std::string& der_private_key) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36347 protected:
348 virtual ~DTLSIdentityRequestObserver() {}
349};
350
351class DTLSIdentityServiceInterface {
352 public:
353 // Asynchronously request a DTLS identity, including a self-signed certificate
354 // and the private key used to sign the certificate, from the identity store
355 // for the given identity name.
356 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
357 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
358 // called with an error code if the request failed.
359 //
360 // Only one request can be made at a time. If a second request is called
361 // before the first one completes, RequestIdentity will abort and return
362 // false.
363 //
364 // |identity_name| is an internal name selected by the client to identify an
365 // identity within an origin. E.g. an web site may cache the certificates used
366 // to communicate with differnent peers under different identity names.
367 //
368 // |common_name| is the common name used to generate the certificate. If the
369 // certificate already exists in the store, |common_name| is ignored.
370 //
371 // |observer| is the object to receive success or failure callbacks.
372 //
373 // Returns true if either OnFailure or OnSuccess will be called.
374 virtual bool RequestIdentity(
375 const std::string& identity_name,
376 const std::string& common_name,
377 DTLSIdentityRequestObserver* observer) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04378
379 virtual ~DTLSIdentityServiceInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36380};
381
382// PeerConnectionFactoryInterface is the factory interface use for creating
383// PeerConnection, MediaStream and media tracks.
384// PeerConnectionFactoryInterface will create required libjingle threads,
385// socket and network manager factory classes for networking.
386// If an application decides to provide its own threads and network
387// implementation of these classes it should use the alternate
388// CreatePeerConnectionFactory method which accepts threads as input and use the
389// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
390// argument.
391class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
392 public:
wu@webrtc.org97077a32013-10-25 21:18:33393 class Options {
394 public:
395 Options() :
wu@webrtc.org97077a32013-10-25 21:18:33396 disable_encryption(false),
397 disable_sctp_data_channels(false) {
398 }
wu@webrtc.org97077a32013-10-25 21:18:33399 bool disable_encryption;
400 bool disable_sctp_data_channels;
401 };
402
403 virtual void SetOptions(const Options& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36404 virtual talk_base::scoped_refptr<PeerConnectionInterface>
405 CreatePeerConnection(
406 const PeerConnectionInterface::IceServers& configuration,
407 const MediaConstraintsInterface* constraints,
408 DTLSIdentityServiceInterface* dtls_identity_service,
409 PeerConnectionObserver* observer) = 0;
410 virtual talk_base::scoped_refptr<PeerConnectionInterface>
411 CreatePeerConnection(
412 const PeerConnectionInterface::IceServers& configuration,
413 const MediaConstraintsInterface* constraints,
414 PortAllocatorFactoryInterface* allocator_factory,
415 DTLSIdentityServiceInterface* dtls_identity_service,
416 PeerConnectionObserver* observer) = 0;
417 virtual talk_base::scoped_refptr<MediaStreamInterface>
418 CreateLocalMediaStream(const std::string& label) = 0;
419
420 // Creates a AudioSourceInterface.
421 // |constraints| decides audio processing settings but can be NULL.
422 virtual talk_base::scoped_refptr<AudioSourceInterface> CreateAudioSource(
423 const MediaConstraintsInterface* constraints) = 0;
424
425 // Creates a VideoSourceInterface. The new source take ownership of
426 // |capturer|. |constraints| decides video resolution and frame rate but can
427 // be NULL.
428 virtual talk_base::scoped_refptr<VideoSourceInterface> CreateVideoSource(
429 cricket::VideoCapturer* capturer,
430 const MediaConstraintsInterface* constraints) = 0;
431
432 // Creates a new local VideoTrack. The same |source| can be used in several
433 // tracks.
434 virtual talk_base::scoped_refptr<VideoTrackInterface>
435 CreateVideoTrack(const std::string& label,
436 VideoSourceInterface* source) = 0;
437
438 // Creates an new AudioTrack. At the moment |source| can be NULL.
439 virtual talk_base::scoped_refptr<AudioTrackInterface>
440 CreateAudioTrack(const std::string& label,
441 AudioSourceInterface* source) = 0;
442
wu@webrtc.orga9890802013-12-13 00:21:03443 // Starts AEC dump using existing file. Takes ownership of |file| and passes
444 // it on to VoiceEngine (via other objects) immediately, which will take
445 // the ownerhip.
446 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
447 virtual bool StartAecDump(FILE* file) = 0;
448
henrike@webrtc.org28e20752013-07-10 00:45:36449 protected:
450 // Dtor and ctor protected as objects shouldn't be created or deleted via
451 // this interface.
452 PeerConnectionFactoryInterface() {}
453 ~PeerConnectionFactoryInterface() {} // NOLINT
454};
455
456// Create a new instance of PeerConnectionFactoryInterface.
457talk_base::scoped_refptr<PeerConnectionFactoryInterface>
458CreatePeerConnectionFactory();
459
460// Create a new instance of PeerConnectionFactoryInterface.
461// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
462// |decoder_factory| transferred to the returned factory.
463talk_base::scoped_refptr<PeerConnectionFactoryInterface>
464CreatePeerConnectionFactory(
465 talk_base::Thread* worker_thread,
466 talk_base::Thread* signaling_thread,
467 AudioDeviceModule* default_adm,
468 cricket::WebRtcVideoEncoderFactory* encoder_factory,
469 cricket::WebRtcVideoDecoderFactory* decoder_factory);
470
471} // namespace webrtc
472
473#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_