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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2012, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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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"
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +000079#include "talk/app/webrtc/umametrics.h"
wu@webrtc.orga8910d22014-01-23 22:12:45 +000080#include "talk/base/fileutils.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000081#include "talk/base/socketaddress.h"
82
83namespace talk_base {
84class Thread;
85}
86
87namespace cricket {
88class PortAllocator;
89class WebRtcVideoDecoderFactory;
90class WebRtcVideoEncoderFactory;
91}
92
93namespace webrtc {
94class AudioDeviceModule;
95class MediaConstraintsInterface;
96
97// MediaStream container interface.
98class StreamCollectionInterface : public talk_base::RefCountInterface {
99 public:
100 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
101 virtual size_t count() = 0;
102 virtual MediaStreamInterface* at(size_t index) = 0;
103 virtual MediaStreamInterface* find(const std::string& label) = 0;
104 virtual MediaStreamTrackInterface* FindAudioTrack(
105 const std::string& id) = 0;
106 virtual MediaStreamTrackInterface* FindVideoTrack(
107 const std::string& id) = 0;
108
109 protected:
110 // Dtor protected as objects shouldn't be deleted via this interface.
111 ~StreamCollectionInterface() {}
112};
113
114class StatsObserver : public talk_base::RefCountInterface {
115 public:
henrike@webrtc.org185636c2014-07-25 18:44:42 +0000116 virtual void OnComplete(const std::vector<StatsReport>& reports) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117
118 protected:
119 virtual ~StatsObserver() {}
120};
121
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000122class UMAObserver : public talk_base::RefCountInterface {
123 public:
mallinath@webrtc.orgd37bcfa2014-05-12 23:10:18 +0000124 virtual void IncrementCounter(PeerConnectionUMAMetricsCounter type) = 0;
125 virtual void AddHistogramSample(PeerConnectionUMAMetricsName type,
126 int value) = 0;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000127
128 protected:
129 virtual ~UMAObserver() {}
130};
131
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000132class PeerConnectionInterface : public talk_base::RefCountInterface {
133 public:
134 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
135 enum SignalingState {
136 kStable,
137 kHaveLocalOffer,
138 kHaveLocalPrAnswer,
139 kHaveRemoteOffer,
140 kHaveRemotePrAnswer,
141 kClosed,
142 };
143
144 // TODO(bemasc): Remove IceState when callers are changed to
145 // IceConnection/GatheringState.
146 enum IceState {
147 kIceNew,
148 kIceGathering,
149 kIceWaiting,
150 kIceChecking,
151 kIceConnected,
152 kIceCompleted,
153 kIceFailed,
154 kIceClosed,
155 };
156
157 enum IceGatheringState {
158 kIceGatheringNew,
159 kIceGatheringGathering,
160 kIceGatheringComplete
161 };
162
163 enum IceConnectionState {
164 kIceConnectionNew,
165 kIceConnectionChecking,
166 kIceConnectionConnected,
167 kIceConnectionCompleted,
168 kIceConnectionFailed,
169 kIceConnectionDisconnected,
170 kIceConnectionClosed,
171 };
172
173 struct IceServer {
174 std::string uri;
175 std::string username;
176 std::string password;
177 };
178 typedef std::vector<IceServer> IceServers;
179
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000180 enum IceTransportsType {
181 kNone,
182 kRelay,
183 kNoHost,
184 kAll
185 };
186
187 struct RTCConfiguration {
188 IceTransportsType type;
189 IceServers servers;
190
191 RTCConfiguration() : type(kAll) {}
192 explicit RTCConfiguration(IceTransportsType type) : type(type) {}
193 };
194
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000195 // Used by GetStats to decide which stats to include in the stats reports.
196 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
197 // |kStatsOutputLevelDebug| includes both the standard stats and additional
198 // stats for debugging purposes.
199 enum StatsOutputLevel {
200 kStatsOutputLevelStandard,
201 kStatsOutputLevelDebug,
202 };
203
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 // Accessor methods to active local streams.
205 virtual talk_base::scoped_refptr<StreamCollectionInterface>
206 local_streams() = 0;
207
208 // Accessor methods to remote streams.
209 virtual talk_base::scoped_refptr<StreamCollectionInterface>
210 remote_streams() = 0;
211
212 // Add a new MediaStream to be sent on this PeerConnection.
213 // Note that a SessionDescription negotiation is needed before the
214 // remote peer can receive the stream.
215 virtual bool AddStream(MediaStreamInterface* stream,
216 const MediaConstraintsInterface* constraints) = 0;
217
218 // Remove a MediaStream from this PeerConnection.
219 // Note that a SessionDescription negotiation is need before the
220 // remote peer is notified.
221 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
222
223 // Returns pointer to the created DtmfSender on success.
224 // Otherwise returns NULL.
225 virtual talk_base::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
226 AudioTrackInterface* track) = 0;
227
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000228 virtual bool GetStats(StatsObserver* observer,
229 MediaStreamTrackInterface* track,
230 StatsOutputLevel level) = 0;
231
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000232 virtual talk_base::scoped_refptr<DataChannelInterface> CreateDataChannel(
233 const std::string& label,
234 const DataChannelInit* config) = 0;
235
236 virtual const SessionDescriptionInterface* local_description() const = 0;
237 virtual const SessionDescriptionInterface* remote_description() const = 0;
238
239 // Create a new offer.
240 // The CreateSessionDescriptionObserver callback will be called when done.
241 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
242 const MediaConstraintsInterface* constraints) = 0;
243 // Create an answer to an offer.
244 // The CreateSessionDescriptionObserver callback will be called when done.
245 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
246 const MediaConstraintsInterface* constraints) = 0;
247 // Sets the local session description.
248 // JsepInterface takes the ownership of |desc| even if it fails.
249 // The |observer| callback will be called when done.
250 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
251 SessionDescriptionInterface* desc) = 0;
252 // Sets the remote session description.
253 // JsepInterface takes the ownership of |desc| even if it fails.
254 // The |observer| callback will be called when done.
255 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
256 SessionDescriptionInterface* desc) = 0;
257 // Restarts or updates the ICE Agent process of gathering local candidates
258 // and pinging remote candidates.
259 virtual bool UpdateIce(const IceServers& configuration,
260 const MediaConstraintsInterface* constraints) = 0;
261 // Provides a remote candidate to the ICE Agent.
262 // A copy of the |candidate| will be created and added to the remote
263 // description. So the caller of this method still has the ownership of the
264 // |candidate|.
265 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
266 // take the ownership of the |candidate|.
267 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
268
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000269 virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
270
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271 // Returns the current SignalingState.
272 virtual SignalingState signaling_state() = 0;
273
274 // TODO(bemasc): Remove ice_state when callers are changed to
275 // IceConnection/GatheringState.
276 // Returns the current IceState.
277 virtual IceState ice_state() = 0;
278 virtual IceConnectionState ice_connection_state() = 0;
279 virtual IceGatheringState ice_gathering_state() = 0;
280
281 // Terminates all media and closes the transport.
282 virtual void Close() = 0;
283
284 protected:
285 // Dtor protected as objects shouldn't be deleted via this interface.
286 ~PeerConnectionInterface() {}
287};
288
289// PeerConnection callback interface. Application should implement these
290// methods.
291class PeerConnectionObserver {
292 public:
293 enum StateType {
294 kSignalingState,
295 kIceState,
296 };
297
298 virtual void OnError() = 0;
299
300 // Triggered when the SignalingState changed.
301 virtual void OnSignalingChange(
302 PeerConnectionInterface::SignalingState new_state) {}
303
304 // Triggered when SignalingState or IceState have changed.
305 // TODO(bemasc): Remove once callers transition to OnSignalingChange.
306 virtual void OnStateChange(StateType state_changed) {}
307
308 // Triggered when media is received on a new stream from remote peer.
309 virtual void OnAddStream(MediaStreamInterface* stream) = 0;
310
311 // Triggered when a remote peer close a stream.
312 virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
313
314 // Triggered when a remote peer open a data channel.
315 // TODO(perkj): Make pure virtual.
316 virtual void OnDataChannel(DataChannelInterface* data_channel) {}
317
mallinath@webrtc.org0d92ef62014-01-22 02:21:22 +0000318 // Triggered when renegotiation is needed, for example the ICE has restarted.
fischman@webrtc.orgd7568a02014-01-13 22:04:12 +0000319 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000320
321 // Called any time the IceConnectionState changes
322 virtual void OnIceConnectionChange(
323 PeerConnectionInterface::IceConnectionState new_state) {}
324
325 // Called any time the IceGatheringState changes
326 virtual void OnIceGatheringChange(
327 PeerConnectionInterface::IceGatheringState new_state) {}
328
329 // New Ice candidate have been found.
330 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
331
332 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
333 // All Ice candidates have been found.
334 virtual void OnIceComplete() {}
335
336 protected:
337 // Dtor protected as objects shouldn't be deleted via this interface.
338 ~PeerConnectionObserver() {}
339};
340
341// Factory class used for creating cricket::PortAllocator that is used
342// for ICE negotiation.
343class PortAllocatorFactoryInterface : public talk_base::RefCountInterface {
344 public:
345 struct StunConfiguration {
346 StunConfiguration(const std::string& address, int port)
347 : server(address, port) {}
348 // STUN server address and port.
349 talk_base::SocketAddress server;
350 };
351
352 struct TurnConfiguration {
353 TurnConfiguration(const std::string& address,
354 int port,
355 const std::string& username,
356 const std::string& password,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000357 const std::string& transport_type,
358 bool secure)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000359 : server(address, port),
360 username(username),
361 password(password),
wu@webrtc.org91053e72013-08-10 07:18:04 +0000362 transport_type(transport_type),
363 secure(secure) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000364 talk_base::SocketAddress server;
365 std::string username;
366 std::string password;
367 std::string transport_type;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000368 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000369 };
370
371 virtual cricket::PortAllocator* CreatePortAllocator(
372 const std::vector<StunConfiguration>& stun_servers,
373 const std::vector<TurnConfiguration>& turn_configurations) = 0;
374
375 protected:
376 PortAllocatorFactoryInterface() {}
377 ~PortAllocatorFactoryInterface() {}
378};
379
380// Used to receive callbacks of DTLS identity requests.
381class DTLSIdentityRequestObserver : public talk_base::RefCountInterface {
382 public:
383 virtual void OnFailure(int error) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000384 virtual void OnSuccess(const std::string& der_cert,
385 const std::string& der_private_key) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000386 protected:
387 virtual ~DTLSIdentityRequestObserver() {}
388};
389
390class DTLSIdentityServiceInterface {
391 public:
392 // Asynchronously request a DTLS identity, including a self-signed certificate
393 // and the private key used to sign the certificate, from the identity store
394 // for the given identity name.
395 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
396 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
397 // called with an error code if the request failed.
398 //
399 // Only one request can be made at a time. If a second request is called
400 // before the first one completes, RequestIdentity will abort and return
401 // false.
402 //
403 // |identity_name| is an internal name selected by the client to identify an
404 // identity within an origin. E.g. an web site may cache the certificates used
405 // to communicate with differnent peers under different identity names.
406 //
407 // |common_name| is the common name used to generate the certificate. If the
408 // certificate already exists in the store, |common_name| is ignored.
409 //
410 // |observer| is the object to receive success or failure callbacks.
411 //
412 // Returns true if either OnFailure or OnSuccess will be called.
413 virtual bool RequestIdentity(
414 const std::string& identity_name,
415 const std::string& common_name,
416 DTLSIdentityRequestObserver* observer) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000417
418 virtual ~DTLSIdentityServiceInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000419};
420
421// PeerConnectionFactoryInterface is the factory interface use for creating
422// PeerConnection, MediaStream and media tracks.
423// PeerConnectionFactoryInterface will create required libjingle threads,
424// socket and network manager factory classes for networking.
425// If an application decides to provide its own threads and network
426// implementation of these classes it should use the alternate
427// CreatePeerConnectionFactory method which accepts threads as input and use the
428// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
429// argument.
430class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
431 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +0000432 class Options {
433 public:
434 Options() :
wu@webrtc.org97077a32013-10-25 21:18:33 +0000435 disable_encryption(false),
436 disable_sctp_data_channels(false) {
437 }
wu@webrtc.org97077a32013-10-25 21:18:33 +0000438 bool disable_encryption;
439 bool disable_sctp_data_channels;
440 };
441
442 virtual void SetOptions(const Options& options) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000443
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000444 virtual talk_base::scoped_refptr<PeerConnectionInterface>
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000445 CreatePeerConnection(
446 const PeerConnectionInterface::RTCConfiguration& configuration,
447 const MediaConstraintsInterface* constraints,
448 PortAllocatorFactoryInterface* allocator_factory,
449 DTLSIdentityServiceInterface* dtls_identity_service,
450 PeerConnectionObserver* observer) = 0;
451
452 // TODO(mallinath) : Remove below versions after clients are updated
453 // to above method.
454 // In latest W3C WebRTC draft, PC constructor will take RTCConfiguration,
455 // and not IceServers. RTCConfiguration is made up of ice servers and
456 // ice transport type.
457 // http://dev.w3.org/2011/webrtc/editor/webrtc.html
458 inline talk_base::scoped_refptr<PeerConnectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000459 CreatePeerConnection(
460 const PeerConnectionInterface::IceServers& configuration,
461 const MediaConstraintsInterface* constraints,
462 PortAllocatorFactoryInterface* allocator_factory,
463 DTLSIdentityServiceInterface* dtls_identity_service,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000464 PeerConnectionObserver* observer) {
465 PeerConnectionInterface::RTCConfiguration rtc_config;
466 rtc_config.servers = configuration;
467 return CreatePeerConnection(rtc_config, constraints, allocator_factory,
468 dtls_identity_service, observer);
469 }
470
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000471 virtual talk_base::scoped_refptr<MediaStreamInterface>
472 CreateLocalMediaStream(const std::string& label) = 0;
473
474 // Creates a AudioSourceInterface.
475 // |constraints| decides audio processing settings but can be NULL.
476 virtual talk_base::scoped_refptr<AudioSourceInterface> CreateAudioSource(
477 const MediaConstraintsInterface* constraints) = 0;
478
479 // Creates a VideoSourceInterface. The new source take ownership of
480 // |capturer|. |constraints| decides video resolution and frame rate but can
481 // be NULL.
482 virtual talk_base::scoped_refptr<VideoSourceInterface> CreateVideoSource(
483 cricket::VideoCapturer* capturer,
484 const MediaConstraintsInterface* constraints) = 0;
485
486 // Creates a new local VideoTrack. The same |source| can be used in several
487 // tracks.
488 virtual talk_base::scoped_refptr<VideoTrackInterface>
489 CreateVideoTrack(const std::string& label,
490 VideoSourceInterface* source) = 0;
491
492 // Creates an new AudioTrack. At the moment |source| can be NULL.
493 virtual talk_base::scoped_refptr<AudioTrackInterface>
494 CreateAudioTrack(const std::string& label,
495 AudioSourceInterface* source) = 0;
496
wu@webrtc.orga9890802013-12-13 00:21:03 +0000497 // Starts AEC dump using existing file. Takes ownership of |file| and passes
498 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000499 // the ownerhip. If the operation fails, the file will be closed.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000500 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000501 // http://crbug.com/264611.
502 virtual bool StartAecDump(talk_base::PlatformFile file) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03 +0000503
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000504 protected:
505 // Dtor and ctor protected as objects shouldn't be created or deleted via
506 // this interface.
507 PeerConnectionFactoryInterface() {}
508 ~PeerConnectionFactoryInterface() {} // NOLINT
509};
510
511// Create a new instance of PeerConnectionFactoryInterface.
512talk_base::scoped_refptr<PeerConnectionFactoryInterface>
513CreatePeerConnectionFactory();
514
515// Create a new instance of PeerConnectionFactoryInterface.
516// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
517// |decoder_factory| transferred to the returned factory.
518talk_base::scoped_refptr<PeerConnectionFactoryInterface>
519CreatePeerConnectionFactory(
520 talk_base::Thread* worker_thread,
521 talk_base::Thread* signaling_thread,
522 AudioDeviceModule* default_adm,
523 cricket::WebRtcVideoEncoderFactory* encoder_factory,
524 cricket::WebRtcVideoDecoderFactory* decoder_factory);
525
526} // namespace webrtc
527
528#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_