blob: 367bafe47c8a6fdea14e663783a3f36ae5fab38e [file] [log] [blame]
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:13 +00003 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00004 *
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"
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +000079#include "talk/app/webrtc/umametrics.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000080#include "webrtc/base/fileutils.h"
phoglund@webrtc.org006521d2015-02-12 09:23:59 +000081#include "webrtc/base/network.h"
Joachim Bauch04e5b492015-05-29 09:40:39 +020082#include "webrtc/base/sslstreamadapter.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000083#include "webrtc/base/socketaddress.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000085namespace rtc {
jiayl@webrtc.org61e00b02015-03-04 22:17:38 +000086class SSLIdentity;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000087class Thread;
88}
89
90namespace cricket {
91class PortAllocator;
92class WebRtcVideoDecoderFactory;
93class WebRtcVideoEncoderFactory;
94}
95
96namespace webrtc {
97class AudioDeviceModule;
98class MediaConstraintsInterface;
99
100// MediaStream container interface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000101class StreamCollectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000102 public:
103 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
104 virtual size_t count() = 0;
105 virtual MediaStreamInterface* at(size_t index) = 0;
106 virtual MediaStreamInterface* find(const std::string& label) = 0;
107 virtual MediaStreamTrackInterface* FindAudioTrack(
108 const std::string& id) = 0;
109 virtual MediaStreamTrackInterface* FindVideoTrack(
110 const std::string& id) = 0;
111
112 protected:
113 // Dtor protected as objects shouldn't be deleted via this interface.
114 ~StreamCollectionInterface() {}
115};
116
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000117class StatsObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 public:
tommi@webrtc.orge2e199b2014-12-15 13:22:54 +0000119 virtual void OnComplete(const StatsReports& reports) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120
121 protected:
122 virtual ~StatsObserver() {}
123};
124
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000125class MetricsObserverInterface : public rtc::RefCountInterface {
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000126 public:
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000127 virtual void IncrementCounter(PeerConnectionMetricsCounter type) = 0;
128 virtual void AddHistogramSample(PeerConnectionMetricsName type,
mallinath@webrtc.orgd37bcfa2014-05-12 23:10:18 +0000129 int value) = 0;
jbauchac8869e2015-07-03 01:36:14 -0700130 // TODO(jbauch): Make method abstract when it is implemented by Chromium.
131 virtual void AddHistogramSample(PeerConnectionMetricsName type,
132 const std::string& value) {}
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000133
134 protected:
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000135 virtual ~MetricsObserverInterface() {}
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000136};
137
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000138typedef MetricsObserverInterface UMAObserver;
139
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000140class PeerConnectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000141 public:
142 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
143 enum SignalingState {
144 kStable,
145 kHaveLocalOffer,
146 kHaveLocalPrAnswer,
147 kHaveRemoteOffer,
148 kHaveRemotePrAnswer,
149 kClosed,
150 };
151
152 // TODO(bemasc): Remove IceState when callers are changed to
153 // IceConnection/GatheringState.
154 enum IceState {
155 kIceNew,
156 kIceGathering,
157 kIceWaiting,
158 kIceChecking,
159 kIceConnected,
160 kIceCompleted,
161 kIceFailed,
162 kIceClosed,
163 };
164
165 enum IceGatheringState {
166 kIceGatheringNew,
167 kIceGatheringGathering,
168 kIceGatheringComplete
169 };
170
171 enum IceConnectionState {
172 kIceConnectionNew,
173 kIceConnectionChecking,
174 kIceConnectionConnected,
175 kIceConnectionCompleted,
176 kIceConnectionFailed,
177 kIceConnectionDisconnected,
178 kIceConnectionClosed,
179 };
180
181 struct IceServer {
Joachim Bauch7c4e7452015-05-28 23:06:30 +0200182 // TODO(jbauch): Remove uri when all code using it has switched to urls.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000183 std::string uri;
Joachim Bauch7c4e7452015-05-28 23:06:30 +0200184 std::vector<std::string> urls;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000185 std::string username;
186 std::string password;
187 };
188 typedef std::vector<IceServer> IceServers;
189
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000190 enum IceTransportsType {
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000191 // TODO(pthatcher): Rename these kTransporTypeXXX, but update
192 // Chromium at the same time.
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000193 kNone,
194 kRelay,
195 kNoHost,
196 kAll
197 };
198
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000199 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
200 enum BundlePolicy {
201 kBundlePolicyBalanced,
202 kBundlePolicyMaxBundle,
203 kBundlePolicyMaxCompat
204 };
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000205
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700206 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
207 enum RtcpMuxPolicy {
208 kRtcpMuxPolicyNegotiate,
209 kRtcpMuxPolicyRequire,
210 };
211
Jiayang Liucac1b382015-04-30 12:35:24 -0700212 enum TcpCandidatePolicy {
213 kTcpCandidatePolicyEnabled,
214 kTcpCandidatePolicyDisabled
215 };
216
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000217 struct RTCConfiguration {
218 // TODO(pthatcher): Rename this ice_transport_type, but update
219 // Chromium at the same time.
220 IceTransportsType type;
221 // TODO(pthatcher): Rename this ice_servers, but update Chromium
222 // at the same time.
223 IceServers servers;
224 BundlePolicy bundle_policy;
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700225 RtcpMuxPolicy rtcp_mux_policy;
Jiayang Liucac1b382015-04-30 12:35:24 -0700226 TcpCandidatePolicy tcp_candidate_policy;
Henrik Lundin64dad832015-05-11 12:44:23 +0200227 int audio_jitter_buffer_max_packets;
Henrik Lundin5263b3c2015-06-01 10:29:41 +0200228 bool audio_jitter_buffer_fast_accelerate;
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000229
Jiayang Liucac1b382015-04-30 12:35:24 -0700230 RTCConfiguration()
231 : type(kAll),
232 bundle_policy(kBundlePolicyBalanced),
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700233 rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
Henrik Lundin64dad832015-05-11 12:44:23 +0200234 tcp_candidate_policy(kTcpCandidatePolicyEnabled),
Henrik Lundin5263b3c2015-06-01 10:29:41 +0200235 audio_jitter_buffer_max_packets(50),
236 audio_jitter_buffer_fast_accelerate(false) {}
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000237 };
238
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000239 struct RTCOfferAnswerOptions {
240 static const int kUndefined = -1;
241 static const int kMaxOfferToReceiveMedia = 1;
242
243 // The default value for constraint offerToReceiveX:true.
244 static const int kOfferToReceiveMediaTrue = 1;
245
246 int offer_to_receive_video;
247 int offer_to_receive_audio;
248 bool voice_activity_detection;
249 bool ice_restart;
250 bool use_rtp_mux;
251
252 RTCOfferAnswerOptions()
253 : offer_to_receive_video(kUndefined),
254 offer_to_receive_audio(kUndefined),
255 voice_activity_detection(true),
256 ice_restart(false),
257 use_rtp_mux(true) {}
258
259 RTCOfferAnswerOptions(int offer_to_receive_video,
260 int offer_to_receive_audio,
261 bool voice_activity_detection,
262 bool ice_restart,
263 bool use_rtp_mux)
264 : offer_to_receive_video(offer_to_receive_video),
265 offer_to_receive_audio(offer_to_receive_audio),
266 voice_activity_detection(voice_activity_detection),
267 ice_restart(ice_restart),
268 use_rtp_mux(use_rtp_mux) {}
269 };
270
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000271 // Used by GetStats to decide which stats to include in the stats reports.
272 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
273 // |kStatsOutputLevelDebug| includes both the standard stats and additional
274 // stats for debugging purposes.
275 enum StatsOutputLevel {
276 kStatsOutputLevelStandard,
277 kStatsOutputLevelDebug,
278 };
279
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000280 // Accessor methods to active local streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000281 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000282 local_streams() = 0;
283
284 // Accessor methods to remote streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000285 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000286 remote_streams() = 0;
287
288 // Add a new MediaStream to be sent on this PeerConnection.
289 // Note that a SessionDescription negotiation is needed before the
290 // remote peer can receive the stream.
perkj@webrtc.orgfd0efb62014-11-06 12:16:36 +0000291 virtual bool AddStream(MediaStreamInterface* stream) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000292
293 // Remove a MediaStream from this PeerConnection.
294 // Note that a SessionDescription negotiation is need before the
295 // remote peer is notified.
296 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
297
298 // Returns pointer to the created DtmfSender on success.
299 // Otherwise returns NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000300 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000301 AudioTrackInterface* track) = 0;
302
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000303 virtual bool GetStats(StatsObserver* observer,
304 MediaStreamTrackInterface* track,
305 StatsOutputLevel level) = 0;
306
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000307 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000308 const std::string& label,
309 const DataChannelInit* config) = 0;
310
311 virtual const SessionDescriptionInterface* local_description() const = 0;
312 virtual const SessionDescriptionInterface* remote_description() const = 0;
313
314 // Create a new offer.
315 // The CreateSessionDescriptionObserver callback will be called when done.
316 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000317 const MediaConstraintsInterface* constraints) {}
318
319 // TODO(jiayl): remove the default impl and the old interface when chromium
320 // code is updated.
321 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
322 const RTCOfferAnswerOptions& options) {}
323
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000324 // Create an answer to an offer.
325 // The CreateSessionDescriptionObserver callback will be called when done.
326 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
327 const MediaConstraintsInterface* constraints) = 0;
328 // Sets the local session description.
329 // JsepInterface takes the ownership of |desc| even if it fails.
330 // The |observer| callback will be called when done.
331 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
332 SessionDescriptionInterface* desc) = 0;
333 // Sets the remote session description.
334 // JsepInterface takes the ownership of |desc| even if it fails.
335 // The |observer| callback will be called when done.
336 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
337 SessionDescriptionInterface* desc) = 0;
honghaiz90099622015-07-13 12:19:33 -0700338 // Sets the ICE connection receiving timeout value in milliseconds.
339 virtual void SetIceConnectionReceivingTimeout(int timeout_ms) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000340 // Restarts or updates the ICE Agent process of gathering local candidates
341 // and pinging remote candidates.
342 virtual bool UpdateIce(const IceServers& configuration,
343 const MediaConstraintsInterface* constraints) = 0;
344 // Provides a remote candidate to the ICE Agent.
345 // A copy of the |candidate| will be created and added to the remote
346 // description. So the caller of this method still has the ownership of the
347 // |candidate|.
348 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
349 // take the ownership of the |candidate|.
350 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
351
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000352 virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
353
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000354 // Returns the current SignalingState.
355 virtual SignalingState signaling_state() = 0;
356
357 // TODO(bemasc): Remove ice_state when callers are changed to
358 // IceConnection/GatheringState.
359 // Returns the current IceState.
360 virtual IceState ice_state() = 0;
361 virtual IceConnectionState ice_connection_state() = 0;
362 virtual IceGatheringState ice_gathering_state() = 0;
363
364 // Terminates all media and closes the transport.
365 virtual void Close() = 0;
366
367 protected:
368 // Dtor protected as objects shouldn't be deleted via this interface.
369 ~PeerConnectionInterface() {}
370};
371
372// PeerConnection callback interface. Application should implement these
373// methods.
374class PeerConnectionObserver {
375 public:
376 enum StateType {
377 kSignalingState,
378 kIceState,
379 };
380
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000381 // Triggered when the SignalingState changed.
382 virtual void OnSignalingChange(
383 PeerConnectionInterface::SignalingState new_state) {}
384
385 // Triggered when SignalingState or IceState have changed.
386 // TODO(bemasc): Remove once callers transition to OnSignalingChange.
387 virtual void OnStateChange(StateType state_changed) {}
388
389 // Triggered when media is received on a new stream from remote peer.
390 virtual void OnAddStream(MediaStreamInterface* stream) = 0;
391
392 // Triggered when a remote peer close a stream.
393 virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
394
395 // Triggered when a remote peer open a data channel.
perkj@webrtc.orgc2dd5ee2014-11-04 11:31:29 +0000396 virtual void OnDataChannel(DataChannelInterface* data_channel) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000397
mallinath@webrtc.org0d92ef62014-01-22 02:21:22 +0000398 // Triggered when renegotiation is needed, for example the ICE has restarted.
fischman@webrtc.orgd7568a02014-01-13 22:04:12 +0000399 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000400
401 // Called any time the IceConnectionState changes
402 virtual void OnIceConnectionChange(
403 PeerConnectionInterface::IceConnectionState new_state) {}
404
405 // Called any time the IceGatheringState changes
406 virtual void OnIceGatheringChange(
407 PeerConnectionInterface::IceGatheringState new_state) {}
408
409 // New Ice candidate have been found.
410 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
411
412 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
413 // All Ice candidates have been found.
414 virtual void OnIceComplete() {}
415
Peter Thatcher54360512015-07-08 11:08:35 -0700416 // Called when the ICE connection receiving status changes.
417 virtual void OnIceConnectionReceivingChange(bool receiving) {}
418
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000419 protected:
420 // Dtor protected as objects shouldn't be deleted via this interface.
421 ~PeerConnectionObserver() {}
422};
423
424// Factory class used for creating cricket::PortAllocator that is used
425// for ICE negotiation.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000426class PortAllocatorFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000427 public:
428 struct StunConfiguration {
429 StunConfiguration(const std::string& address, int port)
430 : server(address, port) {}
431 // STUN server address and port.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000432 rtc::SocketAddress server;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000433 };
434
435 struct TurnConfiguration {
436 TurnConfiguration(const std::string& address,
437 int port,
438 const std::string& username,
439 const std::string& password,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000440 const std::string& transport_type,
441 bool secure)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000442 : server(address, port),
443 username(username),
444 password(password),
wu@webrtc.org91053e72013-08-10 07:18:04 +0000445 transport_type(transport_type),
446 secure(secure) {}
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000447 rtc::SocketAddress server;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000448 std::string username;
449 std::string password;
450 std::string transport_type;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000451 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000452 };
453
454 virtual cricket::PortAllocator* CreatePortAllocator(
455 const std::vector<StunConfiguration>& stun_servers,
456 const std::vector<TurnConfiguration>& turn_configurations) = 0;
457
phoglund@webrtc.org006521d2015-02-12 09:23:59 +0000458 // TODO(phoglund): Make pure virtual when Chrome's factory implements this.
459 // After this method is called, the port allocator should consider loopback
460 // network interfaces as well.
461 virtual void SetNetworkIgnoreMask(int network_ignore_mask) {
462 }
463
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000464 protected:
465 PortAllocatorFactoryInterface() {}
466 ~PortAllocatorFactoryInterface() {}
467};
468
469// Used to receive callbacks of DTLS identity requests.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000470class DTLSIdentityRequestObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000471 public:
472 virtual void OnFailure(int error) = 0;
jiayl@webrtc.org61e00b02015-03-04 22:17:38 +0000473 // TODO(jiayl): Unify the OnSuccess method once Chrome code is updated.
wu@webrtc.org91053e72013-08-10 07:18:04 +0000474 virtual void OnSuccess(const std::string& der_cert,
475 const std::string& der_private_key) = 0;
jiayl@webrtc.org61e00b02015-03-04 22:17:38 +0000476 // |identity| is a scoped_ptr because rtc::SSLIdentity is not copyable and the
477 // client has to get the ownership of the object to make use of it.
478 virtual void OnSuccessWithIdentityObj(
479 rtc::scoped_ptr<rtc::SSLIdentity> identity) = 0;
480
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000481 protected:
482 virtual ~DTLSIdentityRequestObserver() {}
483};
484
485class DTLSIdentityServiceInterface {
486 public:
487 // Asynchronously request a DTLS identity, including a self-signed certificate
488 // and the private key used to sign the certificate, from the identity store
489 // for the given identity name.
490 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
491 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
492 // called with an error code if the request failed.
493 //
494 // Only one request can be made at a time. If a second request is called
495 // before the first one completes, RequestIdentity will abort and return
496 // false.
497 //
498 // |identity_name| is an internal name selected by the client to identify an
499 // identity within an origin. E.g. an web site may cache the certificates used
500 // to communicate with differnent peers under different identity names.
501 //
502 // |common_name| is the common name used to generate the certificate. If the
503 // certificate already exists in the store, |common_name| is ignored.
504 //
505 // |observer| is the object to receive success or failure callbacks.
506 //
507 // Returns true if either OnFailure or OnSuccess will be called.
508 virtual bool RequestIdentity(
509 const std::string& identity_name,
510 const std::string& common_name,
511 DTLSIdentityRequestObserver* observer) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000512
513 virtual ~DTLSIdentityServiceInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000514};
515
516// PeerConnectionFactoryInterface is the factory interface use for creating
517// PeerConnection, MediaStream and media tracks.
518// PeerConnectionFactoryInterface will create required libjingle threads,
519// socket and network manager factory classes for networking.
520// If an application decides to provide its own threads and network
521// implementation of these classes it should use the alternate
522// CreatePeerConnectionFactory method which accepts threads as input and use the
523// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
524// argument.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000525class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000526 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +0000527 class Options {
528 public:
529 Options() :
wu@webrtc.org97077a32013-10-25 21:18:33 +0000530 disable_encryption(false),
phoglund@webrtc.org006521d2015-02-12 09:23:59 +0000531 disable_sctp_data_channels(false),
Joachim Bauch04e5b492015-05-29 09:40:39 +0200532 network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
533 ssl_max_version(rtc::SSL_PROTOCOL_DTLS_10) {
wu@webrtc.org97077a32013-10-25 21:18:33 +0000534 }
wu@webrtc.org97077a32013-10-25 21:18:33 +0000535 bool disable_encryption;
536 bool disable_sctp_data_channels;
phoglund@webrtc.org006521d2015-02-12 09:23:59 +0000537
538 // Sets the network types to ignore. For instance, calling this with
539 // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
540 // loopback interfaces.
541 int network_ignore_mask;
Joachim Bauch04e5b492015-05-29 09:40:39 +0200542
543 // Sets the maximum supported protocol version. The highest version
544 // supported by both ends will be used for the connection, i.e. if one
545 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
546 rtc::SSLProtocolVersion ssl_max_version;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000547 };
548
549 virtual void SetOptions(const Options& options) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000550
jiayl@webrtc.org61e00b02015-03-04 22:17:38 +0000551 // This method takes the ownership of |dtls_identity_service|.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000552 virtual rtc::scoped_refptr<PeerConnectionInterface>
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000553 CreatePeerConnection(
554 const PeerConnectionInterface::RTCConfiguration& configuration,
555 const MediaConstraintsInterface* constraints,
556 PortAllocatorFactoryInterface* allocator_factory,
557 DTLSIdentityServiceInterface* dtls_identity_service,
558 PeerConnectionObserver* observer) = 0;
559
560 // TODO(mallinath) : Remove below versions after clients are updated
561 // to above method.
562 // In latest W3C WebRTC draft, PC constructor will take RTCConfiguration,
563 // and not IceServers. RTCConfiguration is made up of ice servers and
564 // ice transport type.
565 // http://dev.w3.org/2011/webrtc/editor/webrtc.html
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000566 inline rtc::scoped_refptr<PeerConnectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000567 CreatePeerConnection(
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000568 const PeerConnectionInterface::IceServers& servers,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000569 const MediaConstraintsInterface* constraints,
570 PortAllocatorFactoryInterface* allocator_factory,
571 DTLSIdentityServiceInterface* dtls_identity_service,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000572 PeerConnectionObserver* observer) {
573 PeerConnectionInterface::RTCConfiguration rtc_config;
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000574 rtc_config.servers = servers;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000575 return CreatePeerConnection(rtc_config, constraints, allocator_factory,
576 dtls_identity_service, observer);
577 }
578
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000579 virtual rtc::scoped_refptr<MediaStreamInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000580 CreateLocalMediaStream(const std::string& label) = 0;
581
582 // Creates a AudioSourceInterface.
583 // |constraints| decides audio processing settings but can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000584 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000585 const MediaConstraintsInterface* constraints) = 0;
586
587 // Creates a VideoSourceInterface. The new source take ownership of
588 // |capturer|. |constraints| decides video resolution and frame rate but can
589 // be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000590 virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000591 cricket::VideoCapturer* capturer,
592 const MediaConstraintsInterface* constraints) = 0;
593
594 // Creates a new local VideoTrack. The same |source| can be used in several
595 // tracks.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000596 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000597 CreateVideoTrack(const std::string& label,
598 VideoSourceInterface* source) = 0;
599
600 // Creates an new AudioTrack. At the moment |source| can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000601 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000602 CreateAudioTrack(const std::string& label,
603 AudioSourceInterface* source) = 0;
604
wu@webrtc.orga9890802013-12-13 00:21:03 +0000605 // Starts AEC dump using existing file. Takes ownership of |file| and passes
606 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000607 // the ownerhip. If the operation fails, the file will be closed.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000608 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000609 // http://crbug.com/264611.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000610 virtual bool StartAecDump(rtc::PlatformFile file) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03 +0000611
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000612 protected:
613 // Dtor and ctor protected as objects shouldn't be created or deleted via
614 // this interface.
615 PeerConnectionFactoryInterface() {}
616 ~PeerConnectionFactoryInterface() {} // NOLINT
617};
618
619// Create a new instance of PeerConnectionFactoryInterface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000620rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000621CreatePeerConnectionFactory();
622
623// Create a new instance of PeerConnectionFactoryInterface.
624// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
625// |decoder_factory| transferred to the returned factory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000626rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000627CreatePeerConnectionFactory(
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000628 rtc::Thread* worker_thread,
629 rtc::Thread* signaling_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000630 AudioDeviceModule* default_adm,
631 cricket::WebRtcVideoEncoderFactory* encoder_factory,
632 cricket::WebRtcVideoDecoderFactory* decoder_factory);
633
634} // namespace webrtc
635
636#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_