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henrike@webrtc.org28e20752013-07-10 00:45:361/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:133 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:364 *
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"
Henrik Boström5b4ce332015-08-05 14:55:2275#include "talk/app/webrtc/dtlsidentitystore.h"
henrike@webrtc.org28e20752013-07-10 00:45:3676#include "talk/app/webrtc/dtmfsenderinterface.h"
Henrik Boström5e56c592015-08-11 08:33:1377#include "talk/app/webrtc/dtlsidentitystore.h"
henrike@webrtc.org28e20752013-07-10 00:45:3678#include "talk/app/webrtc/jsep.h"
79#include "talk/app/webrtc/mediastreaminterface.h"
deadbeef70ab1a12015-09-28 23:53:5580#include "talk/app/webrtc/rtpreceiverinterface.h"
81#include "talk/app/webrtc/rtpsenderinterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:3682#include "talk/app/webrtc/statstypes.h"
buildbot@webrtc.org1567b8c2014-05-08 19:54:1683#include "talk/app/webrtc/umametrics.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5284#include "webrtc/base/fileutils.h"
phoglund@webrtc.org006521d2015-02-12 09:23:5985#include "webrtc/base/network.h"
Henrik Boström87713d02015-08-25 07:53:2186#include "webrtc/base/rtccertificate.h"
Joachim Bauch04e5b492015-05-29 07:40:3987#include "webrtc/base/sslstreamadapter.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5288#include "webrtc/base/socketaddress.h"
henrike@webrtc.org28e20752013-07-10 00:45:3689
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5290namespace rtc {
jiayl@webrtc.org61e00b02015-03-04 22:17:3891class SSLIdentity;
henrike@webrtc.org28e20752013-07-10 00:45:3692class Thread;
93}
94
95namespace cricket {
96class PortAllocator;
97class WebRtcVideoDecoderFactory;
98class WebRtcVideoEncoderFactory;
99}
100
101namespace webrtc {
102class AudioDeviceModule;
103class MediaConstraintsInterface;
104
105// MediaStream container interface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52106class StreamCollectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36107 public:
108 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
109 virtual size_t count() = 0;
110 virtual MediaStreamInterface* at(size_t index) = 0;
111 virtual MediaStreamInterface* find(const std::string& label) = 0;
112 virtual MediaStreamTrackInterface* FindAudioTrack(
113 const std::string& id) = 0;
114 virtual MediaStreamTrackInterface* FindVideoTrack(
115 const std::string& id) = 0;
116
117 protected:
118 // Dtor protected as objects shouldn't be deleted via this interface.
119 ~StreamCollectionInterface() {}
120};
121
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52122class StatsObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36123 public:
tommi@webrtc.orge2e199b2014-12-15 13:22:54124 virtual void OnComplete(const StatsReports& reports) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36125
126 protected:
127 virtual ~StatsObserver() {}
128};
129
guoweis@webrtc.org7169afd2014-12-04 17:59:29130class MetricsObserverInterface : public rtc::RefCountInterface {
buildbot@webrtc.org1567b8c2014-05-08 19:54:16131 public:
Guo-wei Shieh3d564c12015-08-19 23:51:15132
133 // |type| is the type of the enum counter to be incremented. |counter|
134 // is the particular counter in that type. |counter_max| is the next sequence
135 // number after the highest counter.
136 virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
137 int counter,
138 int counter_max) {}
139
Guo-wei Shieh456696a2015-10-01 04:48:54140 // This is used to handle sparse counters like SSL cipher suites.
141 // TODO(guoweis): Remove the implementation once the dependency's interface
142 // definition is updated.
143 virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
144 int counter) {
145 IncrementEnumCounter(type, counter, 0 /* Ignored */);
146 }
147
guoweis@webrtc.org7169afd2014-12-04 17:59:29148 virtual void AddHistogramSample(PeerConnectionMetricsName type,
mallinath@webrtc.orgd37bcfa2014-05-12 23:10:18149 int value) = 0;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16150
151 protected:
guoweis@webrtc.org7169afd2014-12-04 17:59:29152 virtual ~MetricsObserverInterface() {}
buildbot@webrtc.org1567b8c2014-05-08 19:54:16153};
154
guoweis@webrtc.org7169afd2014-12-04 17:59:29155typedef MetricsObserverInterface UMAObserver;
156
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52157class PeerConnectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36158 public:
159 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
160 enum SignalingState {
161 kStable,
162 kHaveLocalOffer,
163 kHaveLocalPrAnswer,
164 kHaveRemoteOffer,
165 kHaveRemotePrAnswer,
166 kClosed,
167 };
168
169 // TODO(bemasc): Remove IceState when callers are changed to
170 // IceConnection/GatheringState.
171 enum IceState {
172 kIceNew,
173 kIceGathering,
174 kIceWaiting,
175 kIceChecking,
176 kIceConnected,
177 kIceCompleted,
178 kIceFailed,
179 kIceClosed,
180 };
181
182 enum IceGatheringState {
183 kIceGatheringNew,
184 kIceGatheringGathering,
185 kIceGatheringComplete
186 };
187
188 enum IceConnectionState {
189 kIceConnectionNew,
190 kIceConnectionChecking,
191 kIceConnectionConnected,
192 kIceConnectionCompleted,
193 kIceConnectionFailed,
194 kIceConnectionDisconnected,
195 kIceConnectionClosed,
Guo-wei Shieh3d564c12015-08-19 23:51:15196 kIceConnectionMax,
henrike@webrtc.org28e20752013-07-10 00:45:36197 };
198
199 struct IceServer {
Joachim Bauch7c4e7452015-05-28 21:06:30200 // TODO(jbauch): Remove uri when all code using it has switched to urls.
henrike@webrtc.org28e20752013-07-10 00:45:36201 std::string uri;
Joachim Bauch7c4e7452015-05-28 21:06:30202 std::vector<std::string> urls;
henrike@webrtc.org28e20752013-07-10 00:45:36203 std::string username;
204 std::string password;
205 };
206 typedef std::vector<IceServer> IceServers;
207
buildbot@webrtc.org41451d42014-05-03 05:39:45208 enum IceTransportsType {
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06209 // TODO(pthatcher): Rename these kTransporTypeXXX, but update
210 // Chromium at the same time.
buildbot@webrtc.org41451d42014-05-03 05:39:45211 kNone,
212 kRelay,
213 kNoHost,
214 kAll
215 };
216
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06217 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
218 enum BundlePolicy {
219 kBundlePolicyBalanced,
220 kBundlePolicyMaxBundle,
221 kBundlePolicyMaxCompat
222 };
buildbot@webrtc.org41451d42014-05-03 05:39:45223
Peter Thatcheraf55ccc2015-05-21 14:48:41224 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
225 enum RtcpMuxPolicy {
226 kRtcpMuxPolicyNegotiate,
227 kRtcpMuxPolicyRequire,
228 };
229
Jiayang Liucac1b382015-04-30 19:35:24230 enum TcpCandidatePolicy {
231 kTcpCandidatePolicyEnabled,
232 kTcpCandidatePolicyDisabled
233 };
234
honghaiz1f429e32015-09-28 14:57:34235 enum ContinualGatheringPolicy {
236 GATHER_ONCE,
237 GATHER_CONTINUALLY
238 };
239
Henrik Boström87713d02015-08-25 07:53:21240 // TODO(hbos): Change into class with private data and public getters.
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06241 struct RTCConfiguration {
honghaiz4edc39c2015-09-01 16:53:56242 static const int kUndefined = -1;
243 // Default maximum number of packets in the audio jitter buffer.
244 static const int kAudioJitterBufferMaxPackets = 50;
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06245 // TODO(pthatcher): Rename this ice_transport_type, but update
246 // Chromium at the same time.
247 IceTransportsType type;
248 // TODO(pthatcher): Rename this ice_servers, but update Chromium
249 // at the same time.
250 IceServers servers;
Guo-wei Shiehfe3bc9d2015-08-20 15:48:20251 // A localhost candidate is signaled whenever a candidate with the any
252 // address is allocated.
253 bool enable_localhost_ice_candidate;
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06254 BundlePolicy bundle_policy;
Peter Thatcheraf55ccc2015-05-21 14:48:41255 RtcpMuxPolicy rtcp_mux_policy;
Jiayang Liucac1b382015-04-30 19:35:24256 TcpCandidatePolicy tcp_candidate_policy;
Henrik Lundin64dad832015-05-11 10:44:23257 int audio_jitter_buffer_max_packets;
Henrik Lundin5263b3c2015-06-01 08:29:41258 bool audio_jitter_buffer_fast_accelerate;
honghaiz4edc39c2015-09-01 16:53:56259 int ice_connection_receiving_timeout;
honghaiz1f429e32015-09-28 14:57:34260 ContinualGatheringPolicy continual_gathering_policy;
Henrik Boström87713d02015-08-25 07:53:21261 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06262
Jiayang Liucac1b382015-04-30 19:35:24263 RTCConfiguration()
264 : type(kAll),
Guo-wei Shiehfe3bc9d2015-08-20 15:48:20265 enable_localhost_ice_candidate(false),
Jiayang Liucac1b382015-04-30 19:35:24266 bundle_policy(kBundlePolicyBalanced),
Peter Thatcheraf55ccc2015-05-21 14:48:41267 rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
Henrik Lundin64dad832015-05-11 10:44:23268 tcp_candidate_policy(kTcpCandidatePolicyEnabled),
honghaiz4edc39c2015-09-01 16:53:56269 audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
270 audio_jitter_buffer_fast_accelerate(false),
honghaiz1f429e32015-09-28 14:57:34271 ice_connection_receiving_timeout(kUndefined),
272 continual_gathering_policy(GATHER_ONCE) {}
buildbot@webrtc.org41451d42014-05-03 05:39:45273 };
274
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16275 struct RTCOfferAnswerOptions {
276 static const int kUndefined = -1;
277 static const int kMaxOfferToReceiveMedia = 1;
278
279 // The default value for constraint offerToReceiveX:true.
280 static const int kOfferToReceiveMediaTrue = 1;
281
282 int offer_to_receive_video;
283 int offer_to_receive_audio;
284 bool voice_activity_detection;
285 bool ice_restart;
286 bool use_rtp_mux;
287
288 RTCOfferAnswerOptions()
289 : offer_to_receive_video(kUndefined),
290 offer_to_receive_audio(kUndefined),
291 voice_activity_detection(true),
292 ice_restart(false),
293 use_rtp_mux(true) {}
294
295 RTCOfferAnswerOptions(int offer_to_receive_video,
296 int offer_to_receive_audio,
297 bool voice_activity_detection,
298 bool ice_restart,
299 bool use_rtp_mux)
300 : offer_to_receive_video(offer_to_receive_video),
301 offer_to_receive_audio(offer_to_receive_audio),
302 voice_activity_detection(voice_activity_detection),
303 ice_restart(ice_restart),
304 use_rtp_mux(use_rtp_mux) {}
305 };
306
wu@webrtc.orgb9a088b2014-02-13 23:18:49307 // Used by GetStats to decide which stats to include in the stats reports.
308 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
309 // |kStatsOutputLevelDebug| includes both the standard stats and additional
310 // stats for debugging purposes.
311 enum StatsOutputLevel {
312 kStatsOutputLevelStandard,
313 kStatsOutputLevelDebug,
314 };
315
henrike@webrtc.org28e20752013-07-10 00:45:36316 // Accessor methods to active local streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52317 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36318 local_streams() = 0;
319
320 // Accessor methods to remote streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52321 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36322 remote_streams() = 0;
323
324 // Add a new MediaStream to be sent on this PeerConnection.
325 // Note that a SessionDescription negotiation is needed before the
326 // remote peer can receive the stream.
perkj@webrtc.orgfd0efb62014-11-06 12:16:36327 virtual bool AddStream(MediaStreamInterface* stream) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36328
329 // Remove a MediaStream from this PeerConnection.
330 // Note that a SessionDescription negotiation is need before the
331 // remote peer is notified.
332 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
333
334 // Returns pointer to the created DtmfSender on success.
335 // Otherwise returns NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52336 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
henrike@webrtc.org28e20752013-07-10 00:45:36337 AudioTrackInterface* track) = 0;
338
deadbeef70ab1a12015-09-28 23:53:55339 // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
340 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
341 const {
342 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
343 }
344
345 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
346 const {
347 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
348 }
349
wu@webrtc.orgb9a088b2014-02-13 23:18:49350 virtual bool GetStats(StatsObserver* observer,
351 MediaStreamTrackInterface* track,
352 StatsOutputLevel level) = 0;
353
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52354 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36355 const std::string& label,
356 const DataChannelInit* config) = 0;
357
358 virtual const SessionDescriptionInterface* local_description() const = 0;
359 virtual const SessionDescriptionInterface* remote_description() const = 0;
360
361 // Create a new offer.
362 // The CreateSessionDescriptionObserver callback will be called when done.
363 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16364 const MediaConstraintsInterface* constraints) {}
365
366 // TODO(jiayl): remove the default impl and the old interface when chromium
367 // code is updated.
368 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
369 const RTCOfferAnswerOptions& options) {}
370
henrike@webrtc.org28e20752013-07-10 00:45:36371 // Create an answer to an offer.
372 // The CreateSessionDescriptionObserver callback will be called when done.
373 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
374 const MediaConstraintsInterface* constraints) = 0;
375 // Sets the local session description.
376 // JsepInterface takes the ownership of |desc| even if it fails.
377 // The |observer| callback will be called when done.
378 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
379 SessionDescriptionInterface* desc) = 0;
380 // Sets the remote session description.
381 // JsepInterface takes the ownership of |desc| even if it fails.
382 // The |observer| callback will be called when done.
383 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
384 SessionDescriptionInterface* desc) = 0;
385 // Restarts or updates the ICE Agent process of gathering local candidates
386 // and pinging remote candidates.
deadbeefa67696b2015-09-29 18:56:26387 // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
henrike@webrtc.org28e20752013-07-10 00:45:36388 virtual bool UpdateIce(const IceServers& configuration,
deadbeefa67696b2015-09-29 18:56:26389 const MediaConstraintsInterface* constraints) {
390 return false;
391 }
392 // Sets the PeerConnection's global configuration to |config|.
393 // Any changes to STUN/TURN servers or ICE candidate policy will affect the
394 // next gathering phase, and cause the next call to createOffer to generate
395 // new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
396 // cannot be changed with this method.
397 // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
398 // PeerConnectionInterface implement it.
399 virtual bool SetConfiguration(
400 const PeerConnectionInterface::RTCConfiguration& config) {
401 return false;
402 }
henrike@webrtc.org28e20752013-07-10 00:45:36403 // Provides a remote candidate to the ICE Agent.
404 // A copy of the |candidate| will be created and added to the remote
405 // description. So the caller of this method still has the ownership of the
406 // |candidate|.
407 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
408 // take the ownership of the |candidate|.
409 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
410
buildbot@webrtc.org1567b8c2014-05-08 19:54:16411 virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
412
henrike@webrtc.org28e20752013-07-10 00:45:36413 // Returns the current SignalingState.
414 virtual SignalingState signaling_state() = 0;
415
416 // TODO(bemasc): Remove ice_state when callers are changed to
417 // IceConnection/GatheringState.
418 // Returns the current IceState.
419 virtual IceState ice_state() = 0;
420 virtual IceConnectionState ice_connection_state() = 0;
421 virtual IceGatheringState ice_gathering_state() = 0;
422
423 // Terminates all media and closes the transport.
424 virtual void Close() = 0;
425
426 protected:
427 // Dtor protected as objects shouldn't be deleted via this interface.
428 ~PeerConnectionInterface() {}
429};
430
431// PeerConnection callback interface. Application should implement these
432// methods.
433class PeerConnectionObserver {
434 public:
435 enum StateType {
436 kSignalingState,
437 kIceState,
438 };
439
henrike@webrtc.org28e20752013-07-10 00:45:36440 // Triggered when the SignalingState changed.
441 virtual void OnSignalingChange(
442 PeerConnectionInterface::SignalingState new_state) {}
443
444 // Triggered when SignalingState or IceState have changed.
445 // TODO(bemasc): Remove once callers transition to OnSignalingChange.
446 virtual void OnStateChange(StateType state_changed) {}
447
448 // Triggered when media is received on a new stream from remote peer.
449 virtual void OnAddStream(MediaStreamInterface* stream) = 0;
450
451 // Triggered when a remote peer close a stream.
452 virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
453
454 // Triggered when a remote peer open a data channel.
perkj@webrtc.orgc2dd5ee2014-11-04 11:31:29455 virtual void OnDataChannel(DataChannelInterface* data_channel) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36456
mallinath@webrtc.org0d92ef62014-01-22 02:21:22457 // Triggered when renegotiation is needed, for example the ICE has restarted.
fischman@webrtc.orgd7568a02014-01-13 22:04:12458 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36459
460 // Called any time the IceConnectionState changes
461 virtual void OnIceConnectionChange(
462 PeerConnectionInterface::IceConnectionState new_state) {}
463
464 // Called any time the IceGatheringState changes
465 virtual void OnIceGatheringChange(
466 PeerConnectionInterface::IceGatheringState new_state) {}
467
468 // New Ice candidate have been found.
469 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
470
471 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
472 // All Ice candidates have been found.
473 virtual void OnIceComplete() {}
474
Peter Thatcher54360512015-07-08 18:08:35475 // Called when the ICE connection receiving status changes.
476 virtual void OnIceConnectionReceivingChange(bool receiving) {}
477
henrike@webrtc.org28e20752013-07-10 00:45:36478 protected:
479 // Dtor protected as objects shouldn't be deleted via this interface.
480 ~PeerConnectionObserver() {}
481};
482
483// Factory class used for creating cricket::PortAllocator that is used
484// for ICE negotiation.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52485class PortAllocatorFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36486 public:
487 struct StunConfiguration {
488 StunConfiguration(const std::string& address, int port)
489 : server(address, port) {}
490 // STUN server address and port.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52491 rtc::SocketAddress server;
henrike@webrtc.org28e20752013-07-10 00:45:36492 };
493
494 struct TurnConfiguration {
495 TurnConfiguration(const std::string& address,
496 int port,
497 const std::string& username,
498 const std::string& password,
wu@webrtc.org91053e72013-08-10 07:18:04499 const std::string& transport_type,
500 bool secure)
henrike@webrtc.org28e20752013-07-10 00:45:36501 : server(address, port),
502 username(username),
503 password(password),
wu@webrtc.org91053e72013-08-10 07:18:04504 transport_type(transport_type),
505 secure(secure) {}
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52506 rtc::SocketAddress server;
henrike@webrtc.org28e20752013-07-10 00:45:36507 std::string username;
508 std::string password;
509 std::string transport_type;
wu@webrtc.org91053e72013-08-10 07:18:04510 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36511 };
512
513 virtual cricket::PortAllocator* CreatePortAllocator(
514 const std::vector<StunConfiguration>& stun_servers,
515 const std::vector<TurnConfiguration>& turn_configurations) = 0;
516
phoglund@webrtc.org006521d2015-02-12 09:23:59517 // TODO(phoglund): Make pure virtual when Chrome's factory implements this.
518 // After this method is called, the port allocator should consider loopback
519 // network interfaces as well.
520 virtual void SetNetworkIgnoreMask(int network_ignore_mask) {
521 }
522
henrike@webrtc.org28e20752013-07-10 00:45:36523 protected:
524 PortAllocatorFactoryInterface() {}
525 ~PortAllocatorFactoryInterface() {}
526};
527
henrike@webrtc.org28e20752013-07-10 00:45:36528// PeerConnectionFactoryInterface is the factory interface use for creating
529// PeerConnection, MediaStream and media tracks.
530// PeerConnectionFactoryInterface will create required libjingle threads,
531// socket and network manager factory classes for networking.
532// If an application decides to provide its own threads and network
533// implementation of these classes it should use the alternate
534// CreatePeerConnectionFactory method which accepts threads as input and use the
535// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
536// argument.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52537class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36538 public:
wu@webrtc.org97077a32013-10-25 21:18:33539 class Options {
540 public:
541 Options() :
wu@webrtc.org97077a32013-10-25 21:18:33542 disable_encryption(false),
phoglund@webrtc.org006521d2015-02-12 09:23:59543 disable_sctp_data_channels(false),
Joachim Bauch04e5b492015-05-29 07:40:39544 network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
545 ssl_max_version(rtc::SSL_PROTOCOL_DTLS_10) {
wu@webrtc.org97077a32013-10-25 21:18:33546 }
wu@webrtc.org97077a32013-10-25 21:18:33547 bool disable_encryption;
548 bool disable_sctp_data_channels;
phoglund@webrtc.org006521d2015-02-12 09:23:59549
550 // Sets the network types to ignore. For instance, calling this with
551 // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
552 // loopback interfaces.
553 int network_ignore_mask;
Joachim Bauch04e5b492015-05-29 07:40:39554
555 // Sets the maximum supported protocol version. The highest version
556 // supported by both ends will be used for the connection, i.e. if one
557 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
558 rtc::SSLProtocolVersion ssl_max_version;
wu@webrtc.org97077a32013-10-25 21:18:33559 };
560
561 virtual void SetOptions(const Options& options) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45562
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52563 virtual rtc::scoped_refptr<PeerConnectionInterface>
buildbot@webrtc.org41451d42014-05-03 05:39:45564 CreatePeerConnection(
565 const PeerConnectionInterface::RTCConfiguration& configuration,
566 const MediaConstraintsInterface* constraints,
567 PortAllocatorFactoryInterface* allocator_factory,
Henrik Boström5e56c592015-08-11 08:33:13568 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
buildbot@webrtc.org41451d42014-05-03 05:39:45569 PeerConnectionObserver* observer) = 0;
570
Henrik Boström5e56c592015-08-11 08:33:13571 // TODO(hbos): Remove below version after clients are updated to above method.
buildbot@webrtc.org41451d42014-05-03 05:39:45572 // In latest W3C WebRTC draft, PC constructor will take RTCConfiguration,
573 // and not IceServers. RTCConfiguration is made up of ice servers and
574 // ice transport type.
575 // http://dev.w3.org/2011/webrtc/editor/webrtc.html
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52576 inline rtc::scoped_refptr<PeerConnectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36577 CreatePeerConnection(
pthatcher@webrtc.org877ac762015-02-04 22:03:09578 const PeerConnectionInterface::IceServers& servers,
henrike@webrtc.org28e20752013-07-10 00:45:36579 const MediaConstraintsInterface* constraints,
580 PortAllocatorFactoryInterface* allocator_factory,
Henrik Boström5e56c592015-08-11 08:33:13581 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
buildbot@webrtc.org41451d42014-05-03 05:39:45582 PeerConnectionObserver* observer) {
583 PeerConnectionInterface::RTCConfiguration rtc_config;
pthatcher@webrtc.org877ac762015-02-04 22:03:09584 rtc_config.servers = servers;
buildbot@webrtc.org41451d42014-05-03 05:39:45585 return CreatePeerConnection(rtc_config, constraints, allocator_factory,
Henrik Boström5e56c592015-08-11 08:33:13586 dtls_identity_store.Pass(), observer);
buildbot@webrtc.org41451d42014-05-03 05:39:45587 }
588
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52589 virtual rtc::scoped_refptr<MediaStreamInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36590 CreateLocalMediaStream(const std::string& label) = 0;
591
592 // Creates a AudioSourceInterface.
593 // |constraints| decides audio processing settings but can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52594 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
henrike@webrtc.org28e20752013-07-10 00:45:36595 const MediaConstraintsInterface* constraints) = 0;
596
597 // Creates a VideoSourceInterface. The new source take ownership of
598 // |capturer|. |constraints| decides video resolution and frame rate but can
599 // be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52600 virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
henrike@webrtc.org28e20752013-07-10 00:45:36601 cricket::VideoCapturer* capturer,
602 const MediaConstraintsInterface* constraints) = 0;
603
604 // Creates a new local VideoTrack. The same |source| can be used in several
605 // tracks.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52606 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36607 CreateVideoTrack(const std::string& label,
608 VideoSourceInterface* source) = 0;
609
610 // Creates an new AudioTrack. At the moment |source| can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52611 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36612 CreateAudioTrack(const std::string& label,
613 AudioSourceInterface* source) = 0;
614
wu@webrtc.orga9890802013-12-13 00:21:03615 // Starts AEC dump using existing file. Takes ownership of |file| and passes
616 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45617 // the ownerhip. If the operation fails, the file will be closed.
wu@webrtc.orga9890802013-12-13 00:21:03618 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
wu@webrtc.orga8910d22014-01-23 22:12:45619 // http://crbug.com/264611.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52620 virtual bool StartAecDump(rtc::PlatformFile file) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03621
ivoc112a3d82015-10-16 09:22:18622 // Starts RtcEventLog using existing file. Takes ownership of |file| and
623 // passes it on to VoiceEngine, which will take the ownership. If the
624 // operation fails the file will be closed. The logging will stop
625 // automatically after 10 minutes have passed, or when the StopRtcEventLog
626 // function is called.
627 // This function as well as the StopRtcEventLog don't really belong on this
628 // interface, this is a temporary solution until we move the logging object
629 // from inside voice engine to webrtc::Call, which will happen when the VoE
630 // restructuring effort is further along.
631 // TODO(ivoc): Move this into being:
632 // PeerConnection => MediaController => webrtc::Call.
633 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
634
635 // Stops logging the RtcEventLog.
636 virtual void StopRtcEventLog() = 0;
637
henrike@webrtc.org28e20752013-07-10 00:45:36638 protected:
639 // Dtor and ctor protected as objects shouldn't be created or deleted via
640 // this interface.
641 PeerConnectionFactoryInterface() {}
642 ~PeerConnectionFactoryInterface() {} // NOLINT
643};
644
645// Create a new instance of PeerConnectionFactoryInterface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52646rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36647CreatePeerConnectionFactory();
648
649// Create a new instance of PeerConnectionFactoryInterface.
650// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
651// |decoder_factory| transferred to the returned factory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52652rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36653CreatePeerConnectionFactory(
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52654 rtc::Thread* worker_thread,
655 rtc::Thread* signaling_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36656 AudioDeviceModule* default_adm,
657 cricket::WebRtcVideoEncoderFactory* encoder_factory,
658 cricket::WebRtcVideoDecoderFactory* decoder_factory);
659
660} // namespace webrtc
661
662#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_