Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(275)

Issues Search
    My Issues | Starred     Open | Closed | All

Side by Side Diff: webrtc/api/webrtcsession.cc

Issue 2514883002: Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
Patch Set: Rebase Created 3 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/api/webrtcsession.h ('k') | webrtc/api/webrtcsession_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
(Empty)
1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/api/webrtcsession.h"
12
13 #include <limits.h>
14
15 #include <algorithm>
16 #include <set>
17 #include <utility>
18 #include <vector>
19
20 #include "webrtc/api/call/audio_sink.h"
21 #include "webrtc/api/jsepicecandidate.h"
22 #include "webrtc/api/jsepsessiondescription.h"
23 #include "webrtc/api/peerconnectioninterface.h"
24 #include "webrtc/api/sctputils.h"
25 #include "webrtc/api/webrtcsessiondescriptionfactory.h"
26 #include "webrtc/base/basictypes.h"
27 #include "webrtc/base/bind.h"
28 #include "webrtc/base/checks.h"
29 #include "webrtc/base/helpers.h"
30 #include "webrtc/base/logging.h"
31 #include "webrtc/base/stringencode.h"
32 #include "webrtc/base/stringutils.h"
33 #include "webrtc/call/call.h"
34 #include "webrtc/media/base/mediaconstants.h"
35 #include "webrtc/media/base/videocapturer.h"
36 #include "webrtc/media/sctp/sctptransportinternal.h"
37 #include "webrtc/p2p/base/portallocator.h"
38 #include "webrtc/pc/channel.h"
39 #include "webrtc/pc/channelmanager.h"
40 #include "webrtc/pc/mediasession.h"
41
42 #ifdef HAVE_QUIC
43 #include "webrtc/p2p/quic/quictransportchannel.h"
44 #endif // HAVE_QUIC
45
46 using cricket::ContentInfo;
47 using cricket::ContentInfos;
48 using cricket::MediaContentDescription;
49 using cricket::SessionDescription;
50 using cricket::TransportInfo;
51
52 using cricket::LOCAL_PORT_TYPE;
53 using cricket::STUN_PORT_TYPE;
54 using cricket::RELAY_PORT_TYPE;
55 using cricket::PRFLX_PORT_TYPE;
56
57 namespace webrtc {
58
59 // Error messages
60 const char kBundleWithoutRtcpMux[] = "RTCP-MUX must be enabled when BUNDLE "
61 "is enabled.";
62 const char kCreateChannelFailed[] = "Failed to create channels.";
63 const char kInvalidCandidates[] = "Description contains invalid candidates.";
64 const char kInvalidSdp[] = "Invalid session description.";
65 const char kMlineMismatch[] =
66 "Offer and answer descriptions m-lines are not matching. Rejecting answer.";
67 const char kPushDownTDFailed[] =
68 "Failed to push down transport description:";
69 const char kSdpWithoutDtlsFingerprint[] =
70 "Called with SDP without DTLS fingerprint.";
71 const char kSdpWithoutSdesCrypto[] =
72 "Called with SDP without SDES crypto.";
73 const char kSdpWithoutIceUfragPwd[] =
74 "Called with SDP without ice-ufrag and ice-pwd.";
75 const char kSessionError[] = "Session error code: ";
76 const char kSessionErrorDesc[] = "Session error description: ";
77 const char kDtlsSrtpSetupFailureRtp[] =
78 "Couldn't set up DTLS-SRTP on RTP channel.";
79 const char kDtlsSrtpSetupFailureRtcp[] =
80 "Couldn't set up DTLS-SRTP on RTCP channel.";
81 const char kEnableBundleFailed[] = "Failed to enable BUNDLE.";
82
83 IceCandidatePairType GetIceCandidatePairCounter(
84 const cricket::Candidate& local,
85 const cricket::Candidate& remote) {
86 const auto& l = local.type();
87 const auto& r = remote.type();
88 const auto& host = LOCAL_PORT_TYPE;
89 const auto& srflx = STUN_PORT_TYPE;
90 const auto& relay = RELAY_PORT_TYPE;
91 const auto& prflx = PRFLX_PORT_TYPE;
92 if (l == host && r == host) {
93 bool local_private = IPIsPrivate(local.address().ipaddr());
94 bool remote_private = IPIsPrivate(remote.address().ipaddr());
95 if (local_private) {
96 if (remote_private) {
97 return kIceCandidatePairHostPrivateHostPrivate;
98 } else {
99 return kIceCandidatePairHostPrivateHostPublic;
100 }
101 } else {
102 if (remote_private) {
103 return kIceCandidatePairHostPublicHostPrivate;
104 } else {
105 return kIceCandidatePairHostPublicHostPublic;
106 }
107 }
108 }
109 if (l == host && r == srflx)
110 return kIceCandidatePairHostSrflx;
111 if (l == host && r == relay)
112 return kIceCandidatePairHostRelay;
113 if (l == host && r == prflx)
114 return kIceCandidatePairHostPrflx;
115 if (l == srflx && r == host)
116 return kIceCandidatePairSrflxHost;
117 if (l == srflx && r == srflx)
118 return kIceCandidatePairSrflxSrflx;
119 if (l == srflx && r == relay)
120 return kIceCandidatePairSrflxRelay;
121 if (l == srflx && r == prflx)
122 return kIceCandidatePairSrflxPrflx;
123 if (l == relay && r == host)
124 return kIceCandidatePairRelayHost;
125 if (l == relay && r == srflx)
126 return kIceCandidatePairRelaySrflx;
127 if (l == relay && r == relay)
128 return kIceCandidatePairRelayRelay;
129 if (l == relay && r == prflx)
130 return kIceCandidatePairRelayPrflx;
131 if (l == prflx && r == host)
132 return kIceCandidatePairPrflxHost;
133 if (l == prflx && r == srflx)
134 return kIceCandidatePairPrflxSrflx;
135 if (l == prflx && r == relay)
136 return kIceCandidatePairPrflxRelay;
137 return kIceCandidatePairMax;
138 }
139
140 // Compares |answer| against |offer|. Comparision is done
141 // for number of m-lines in answer against offer. If matches true will be
142 // returned otherwise false.
143 static bool VerifyMediaDescriptions(
144 const SessionDescription* answer, const SessionDescription* offer) {
145 if (offer->contents().size() != answer->contents().size())
146 return false;
147
148 for (size_t i = 0; i < offer->contents().size(); ++i) {
149 if ((offer->contents()[i].name) != answer->contents()[i].name) {
150 return false;
151 }
152 const MediaContentDescription* offer_mdesc =
153 static_cast<const MediaContentDescription*>(
154 offer->contents()[i].description);
155 const MediaContentDescription* answer_mdesc =
156 static_cast<const MediaContentDescription*>(
157 answer->contents()[i].description);
158 if (offer_mdesc->type() != answer_mdesc->type()) {
159 return false;
160 }
161 }
162 return true;
163 }
164
165 // Checks that each non-rejected content has SDES crypto keys or a DTLS
166 // fingerprint. Mismatches, such as replying with a DTLS fingerprint to SDES
167 // keys, will be caught in Transport negotiation, and backstopped by Channel's
168 // |srtp_required| check.
169 static bool VerifyCrypto(const SessionDescription* desc,
170 bool dtls_enabled,
171 std::string* error) {
172 const ContentInfos& contents = desc->contents();
173 for (size_t index = 0; index < contents.size(); ++index) {
174 const ContentInfo* cinfo = &contents[index];
175 if (cinfo->rejected) {
176 continue;
177 }
178
179 // If the content isn't rejected, crypto must be present.
180 const MediaContentDescription* media =
181 static_cast<const MediaContentDescription*>(cinfo->description);
182 const TransportInfo* tinfo = desc->GetTransportInfoByName(cinfo->name);
183 if (!media || !tinfo) {
184 // Something is not right.
185 LOG(LS_ERROR) << kInvalidSdp;
186 *error = kInvalidSdp;
187 return false;
188 }
189 if (dtls_enabled) {
190 if (!tinfo->description.identity_fingerprint) {
191 LOG(LS_WARNING) <<
192 "Session description must have DTLS fingerprint if DTLS enabled.";
193 *error = kSdpWithoutDtlsFingerprint;
194 return false;
195 }
196 } else {
197 if (media->cryptos().empty()) {
198 LOG(LS_WARNING) <<
199 "Session description must have SDES when DTLS disabled.";
200 *error = kSdpWithoutSdesCrypto;
201 return false;
202 }
203 }
204 }
205
206 return true;
207 }
208
209 // Checks that each non-rejected content has ice-ufrag and ice-pwd set.
210 static bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
211 const ContentInfos& contents = desc->contents();
212 for (size_t index = 0; index < contents.size(); ++index) {
213 const ContentInfo* cinfo = &contents[index];
214 if (cinfo->rejected) {
215 continue;
216 }
217
218 // If the content isn't rejected, ice-ufrag and ice-pwd must be present.
219 const TransportInfo* tinfo = desc->GetTransportInfoByName(cinfo->name);
220 if (!tinfo) {
221 // Something is not right.
222 LOG(LS_ERROR) << kInvalidSdp;
223 return false;
224 }
225 if (tinfo->description.ice_ufrag.empty() ||
226 tinfo->description.ice_pwd.empty()) {
227 LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
228 return false;
229 }
230 }
231 return true;
232 }
233
234 static bool GetAudioSsrcByTrackId(const SessionDescription* session_description,
235 const std::string& track_id,
236 uint32_t* ssrc) {
237 const cricket::ContentInfo* audio_info =
238 cricket::GetFirstAudioContent(session_description);
239 if (!audio_info) {
240 LOG(LS_ERROR) << "Audio not used in this call";
241 return false;
242 }
243
244 const cricket::MediaContentDescription* audio_content =
245 static_cast<const cricket::MediaContentDescription*>(
246 audio_info->description);
247 const cricket::StreamParams* stream =
248 cricket::GetStreamByIds(audio_content->streams(), "", track_id);
249 if (!stream) {
250 return false;
251 }
252
253 *ssrc = stream->first_ssrc();
254 return true;
255 }
256
257 static bool GetTrackIdBySsrc(const SessionDescription* session_description,
258 uint32_t ssrc,
259 std::string* track_id) {
260 RTC_DCHECK(track_id != NULL);
261
262 const cricket::ContentInfo* audio_info =
263 cricket::GetFirstAudioContent(session_description);
264 if (audio_info) {
265 const cricket::MediaContentDescription* audio_content =
266 static_cast<const cricket::MediaContentDescription*>(
267 audio_info->description);
268
269 const auto* found =
270 cricket::GetStreamBySsrc(audio_content->streams(), ssrc);
271 if (found) {
272 *track_id = found->id;
273 return true;
274 }
275 }
276
277 const cricket::ContentInfo* video_info =
278 cricket::GetFirstVideoContent(session_description);
279 if (video_info) {
280 const cricket::MediaContentDescription* video_content =
281 static_cast<const cricket::MediaContentDescription*>(
282 video_info->description);
283
284 const auto* found =
285 cricket::GetStreamBySsrc(video_content->streams(), ssrc);
286 if (found) {
287 *track_id = found->id;
288 return true;
289 }
290 }
291 return false;
292 }
293
294 // Get the SCTP port out of a SessionDescription.
295 // Return -1 if not found.
296 static int GetSctpPort(const SessionDescription* session_description) {
297 const ContentInfo* content_info = GetFirstDataContent(session_description);
298 RTC_DCHECK(content_info);
299 if (!content_info) {
300 return -1;
301 }
302 const cricket::DataContentDescription* data =
303 static_cast<const cricket::DataContentDescription*>(
304 (content_info->description));
305 std::string value;
306 cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
307 cricket::kGoogleSctpDataCodecName);
308 for (const cricket::DataCodec& codec : data->codecs()) {
309 if (!codec.Matches(match_pattern)) {
310 continue;
311 }
312 if (codec.GetParam(cricket::kCodecParamPort, &value)) {
313 return rtc::FromString<int>(value);
314 }
315 }
316 return -1;
317 }
318
319 static bool BadSdp(const std::string& source,
320 const std::string& type,
321 const std::string& reason,
322 std::string* err_desc) {
323 std::ostringstream desc;
324 desc << "Failed to set " << source;
325 if (!type.empty()) {
326 desc << " " << type;
327 }
328 desc << " sdp: " << reason;
329
330 if (err_desc) {
331 *err_desc = desc.str();
332 }
333 LOG(LS_ERROR) << desc.str();
334 return false;
335 }
336
337 static bool BadSdp(cricket::ContentSource source,
338 const std::string& type,
339 const std::string& reason,
340 std::string* err_desc) {
341 if (source == cricket::CS_LOCAL) {
342 return BadSdp("local", type, reason, err_desc);
343 } else {
344 return BadSdp("remote", type, reason, err_desc);
345 }
346 }
347
348 static bool BadLocalSdp(const std::string& type,
349 const std::string& reason,
350 std::string* err_desc) {
351 return BadSdp(cricket::CS_LOCAL, type, reason, err_desc);
352 }
353
354 static bool BadRemoteSdp(const std::string& type,
355 const std::string& reason,
356 std::string* err_desc) {
357 return BadSdp(cricket::CS_REMOTE, type, reason, err_desc);
358 }
359
360 static bool BadOfferSdp(cricket::ContentSource source,
361 const std::string& reason,
362 std::string* err_desc) {
363 return BadSdp(source, SessionDescriptionInterface::kOffer, reason, err_desc);
364 }
365
366 static bool BadPranswerSdp(cricket::ContentSource source,
367 const std::string& reason,
368 std::string* err_desc) {
369 return BadSdp(source, SessionDescriptionInterface::kPrAnswer,
370 reason, err_desc);
371 }
372
373 static bool BadAnswerSdp(cricket::ContentSource source,
374 const std::string& reason,
375 std::string* err_desc) {
376 return BadSdp(source, SessionDescriptionInterface::kAnswer, reason, err_desc);
377 }
378
379 #define GET_STRING_OF_STATE(state) \
380 case webrtc::WebRtcSession::state: \
381 result = #state; \
382 break;
383
384 static std::string GetStateString(webrtc::WebRtcSession::State state) {
385 std::string result;
386 switch (state) {
387 GET_STRING_OF_STATE(STATE_INIT)
388 GET_STRING_OF_STATE(STATE_SENTOFFER)
389 GET_STRING_OF_STATE(STATE_RECEIVEDOFFER)
390 GET_STRING_OF_STATE(STATE_SENTPRANSWER)
391 GET_STRING_OF_STATE(STATE_RECEIVEDPRANSWER)
392 GET_STRING_OF_STATE(STATE_INPROGRESS)
393 GET_STRING_OF_STATE(STATE_CLOSED)
394 default:
395 RTC_NOTREACHED();
396 break;
397 }
398 return result;
399 }
400
401 #define GET_STRING_OF_ERROR_CODE(err) \
402 case webrtc::WebRtcSession::err: \
403 result = #err; \
404 break;
405
406 static std::string GetErrorCodeString(webrtc::WebRtcSession::Error err) {
407 std::string result;
408 switch (err) {
409 GET_STRING_OF_ERROR_CODE(ERROR_NONE)
410 GET_STRING_OF_ERROR_CODE(ERROR_CONTENT)
411 GET_STRING_OF_ERROR_CODE(ERROR_TRANSPORT)
412 default:
413 RTC_NOTREACHED();
414 break;
415 }
416 return result;
417 }
418
419 static std::string MakeErrorString(const std::string& error,
420 const std::string& desc) {
421 std::ostringstream ret;
422 ret << error << " " << desc;
423 return ret.str();
424 }
425
426 static std::string MakeTdErrorString(const std::string& desc) {
427 return MakeErrorString(kPushDownTDFailed, desc);
428 }
429
430 // Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
431 bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
432 const SessionDescriptionInterface* new_desc,
433 const std::string& content_name) {
434 if (!old_desc) {
435 return false;
436 }
437 const SessionDescription* new_sd = new_desc->description();
438 const SessionDescription* old_sd = old_desc->description();
439 const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
440 if (!cinfo || cinfo->rejected) {
441 return false;
442 }
443 // If the content isn't rejected, check if ufrag and password has changed.
444 const cricket::TransportDescription* new_transport_desc =
445 new_sd->GetTransportDescriptionByName(content_name);
446 const cricket::TransportDescription* old_transport_desc =
447 old_sd->GetTransportDescriptionByName(content_name);
448 if (!new_transport_desc || !old_transport_desc) {
449 // No transport description exists. This is not an ICE restart.
450 return false;
451 }
452 if (cricket::IceCredentialsChanged(
453 old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
454 new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
455 LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
456 << ".";
457 return true;
458 }
459 return false;
460 }
461
462 WebRtcSession::WebRtcSession(
463 webrtc::MediaControllerInterface* media_controller,
464 rtc::Thread* network_thread,
465 rtc::Thread* worker_thread,
466 rtc::Thread* signaling_thread,
467 cricket::PortAllocator* port_allocator,
468 std::unique_ptr<cricket::TransportController> transport_controller,
469 std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory)
470 : network_thread_(network_thread),
471 worker_thread_(worker_thread),
472 signaling_thread_(signaling_thread),
473 // RFC 3264: The numeric value of the session id and version in the
474 // o line MUST be representable with a "64 bit signed integer".
475 // Due to this constraint session id |sid_| is max limited to LLONG_MAX.
476 sid_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
477 transport_controller_(std::move(transport_controller)),
478 sctp_factory_(std::move(sctp_factory)),
479 media_controller_(media_controller),
480 channel_manager_(media_controller_->channel_manager()),
481 ice_observer_(NULL),
482 ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
483 ice_connection_receiving_(true),
484 older_version_remote_peer_(false),
485 dtls_enabled_(false),
486 data_channel_type_(cricket::DCT_NONE),
487 metrics_observer_(NULL) {
488 transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
489 transport_controller_->SignalConnectionState.connect(
490 this, &WebRtcSession::OnTransportControllerConnectionState);
491 transport_controller_->SignalReceiving.connect(
492 this, &WebRtcSession::OnTransportControllerReceiving);
493 transport_controller_->SignalGatheringState.connect(
494 this, &WebRtcSession::OnTransportControllerGatheringState);
495 transport_controller_->SignalCandidatesGathered.connect(
496 this, &WebRtcSession::OnTransportControllerCandidatesGathered);
497 transport_controller_->SignalCandidatesRemoved.connect(
498 this, &WebRtcSession::OnTransportControllerCandidatesRemoved);
499 transport_controller_->SignalDtlsHandshakeError.connect(
500 this, &WebRtcSession::OnTransportControllerDtlsHandshakeError);
501 }
502
503 WebRtcSession::~WebRtcSession() {
504 RTC_DCHECK(signaling_thread()->IsCurrent());
505 // Destroy video_channel_ first since it may have a pointer to the
506 // voice_channel_.
507 if (video_channel_) {
508 DestroyVideoChannel();
509 }
510 if (voice_channel_) {
511 DestroyVoiceChannel();
512 }
513 if (rtp_data_channel_) {
514 DestroyDataChannel();
515 }
516 if (sctp_transport_) {
517 SignalDataChannelDestroyed();
518 network_thread_->Invoke<void>(
519 RTC_FROM_HERE, rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
520 }
521 #ifdef HAVE_QUIC
522 if (quic_data_transport_) {
523 quic_data_transport_.reset();
524 }
525 #endif
526 SignalDestroyed();
527
528 LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
529 }
530
531 bool WebRtcSession::Initialize(
532 const PeerConnectionFactoryInterface::Options& options,
533 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
534 const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
535 bundle_policy_ = rtc_configuration.bundle_policy;
536 rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
537 transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
538
539 // Obtain a certificate from RTCConfiguration if any were provided (optional).
540 rtc::scoped_refptr<rtc::RTCCertificate> certificate;
541 if (!rtc_configuration.certificates.empty()) {
542 // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
543 // just picking the first one. The decision should be made based on the DTLS
544 // handshake. The DTLS negotiations need to know about all certificates.
545 certificate = rtc_configuration.certificates[0];
546 }
547
548 SetIceConfig(ParseIceConfig(rtc_configuration));
549
550 if (options.disable_encryption) {
551 dtls_enabled_ = false;
552 } else {
553 // Enable DTLS by default if we have an identity store or a certificate.
554 dtls_enabled_ = (cert_generator || certificate);
555 // |rtc_configuration| can override the default |dtls_enabled_| value.
556 if (rtc_configuration.enable_dtls_srtp) {
557 dtls_enabled_ = *(rtc_configuration.enable_dtls_srtp);
558 }
559 }
560
561 // Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
562 // It takes precendence over the disable_sctp_data_channels
563 // PeerConnectionFactoryInterface::Options.
564 if (rtc_configuration.enable_rtp_data_channel) {
565 data_channel_type_ = cricket::DCT_RTP;
566 }
567 #ifdef HAVE_QUIC
568 else if (rtc_configuration.enable_quic) {
569 // Use QUIC instead of DTLS when |enable_quic| is true.
570 data_channel_type_ = cricket::DCT_QUIC;
571 transport_controller_->use_quic();
572 if (dtls_enabled_) {
573 LOG(LS_INFO) << "Using QUIC instead of DTLS";
574 }
575 quic_data_transport_.reset(
576 new QuicDataTransport(signaling_thread(), worker_thread(),
577 network_thread(), transport_controller_.get()));
578 }
579 #endif // HAVE_QUIC
580 else {
581 // DTLS has to be enabled to use SCTP.
582 if (!options.disable_sctp_data_channels && dtls_enabled_) {
583 data_channel_type_ = cricket::DCT_SCTP;
584 }
585 }
586
587 video_options_.screencast_min_bitrate_kbps =
588 rtc_configuration.screencast_min_bitrate;
589 audio_options_.combined_audio_video_bwe =
590 rtc_configuration.combined_audio_video_bwe;
591
592 audio_options_.audio_jitter_buffer_max_packets =
593 rtc::Optional<int>(rtc_configuration.audio_jitter_buffer_max_packets);
594
595 audio_options_.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(
596 rtc_configuration.audio_jitter_buffer_fast_accelerate);
597
598 if (!dtls_enabled_) {
599 // Construct with DTLS disabled.
600 webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
601 signaling_thread(), channel_manager_, this, id(),
602 std::unique_ptr<rtc::RTCCertificateGeneratorInterface>()));
603 } else {
604 // Construct with DTLS enabled.
605 if (!certificate) {
606 webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
607 signaling_thread(), channel_manager_, this, id(),
608 std::move(cert_generator)));
609 } else {
610 // Use the already generated certificate.
611 webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
612 signaling_thread(), channel_manager_, this, id(), certificate));
613 }
614 }
615
616 webrtc_session_desc_factory_->SignalCertificateReady.connect(
617 this, &WebRtcSession::OnCertificateReady);
618
619 if (options.disable_encryption) {
620 webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
621 }
622
623 return true;
624 }
625
626 void WebRtcSession::Close() {
627 SetState(STATE_CLOSED);
628 RemoveUnusedChannels(nullptr);
629 RTC_DCHECK(!voice_channel_);
630 RTC_DCHECK(!video_channel_);
631 RTC_DCHECK(!rtp_data_channel_);
632 RTC_DCHECK(!sctp_transport_);
633 media_controller_->Close();
634 }
635
636 cricket::BaseChannel* WebRtcSession::GetChannel(
637 const std::string& content_name) {
638 if (voice_channel() && voice_channel()->content_name() == content_name) {
639 return voice_channel();
640 }
641 if (video_channel() && video_channel()->content_name() == content_name) {
642 return video_channel();
643 }
644 if (rtp_data_channel() &&
645 rtp_data_channel()->content_name() == content_name) {
646 return rtp_data_channel();
647 }
648 return nullptr;
649 }
650
651 cricket::SecurePolicy WebRtcSession::SdesPolicy() const {
652 return webrtc_session_desc_factory_->SdesPolicy();
653 }
654
655 bool WebRtcSession::GetSctpSslRole(rtc::SSLRole* role) {
656 if (!local_description() || !remote_description()) {
657 LOG(LS_INFO) << "Local and Remote descriptions must be applied to get the "
658 << "SSL Role of the SCTP transport.";
659 return false;
660 }
661 if (!sctp_transport_) {
662 LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
663 << "SSL Role of the SCTP transport.";
664 return false;
665 }
666
667 return transport_controller_->GetSslRole(*sctp_transport_name_, role);
668 }
669
670 bool WebRtcSession::GetSslRole(const std::string& content_name,
671 rtc::SSLRole* role) {
672 if (!local_description() || !remote_description()) {
673 LOG(LS_INFO) << "Local and Remote descriptions must be applied to get the "
674 << "SSL Role of the session.";
675 return false;
676 }
677
678 return transport_controller_->GetSslRole(GetTransportName(content_name),
679 role);
680 }
681
682 void WebRtcSession::CreateOffer(
683 CreateSessionDescriptionObserver* observer,
684 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
685 const cricket::MediaSessionOptions& session_options) {
686 webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
687 }
688
689 void WebRtcSession::CreateAnswer(
690 CreateSessionDescriptionObserver* observer,
691 const cricket::MediaSessionOptions& session_options) {
692 webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
693 }
694
695 bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
696 std::string* err_desc) {
697 RTC_DCHECK(signaling_thread()->IsCurrent());
698
699 // Takes the ownership of |desc| regardless of the result.
700 std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
701
702 // Validate SDP.
703 if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
704 return false;
705 }
706
707 // Update the initial_offerer flag if this session is the initial_offerer.
708 Action action = GetAction(desc->type());
709 if (state() == STATE_INIT && action == kOffer) {
710 initial_offerer_ = true;
711 transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
712 }
713
714 if (action == kAnswer) {
715 current_local_description_.reset(desc_temp.release());
716 pending_local_description_.reset(nullptr);
717 current_remote_description_.reset(pending_remote_description_.release());
718 } else {
719 pending_local_description_.reset(desc_temp.release());
720 }
721
722 // Transport and Media channels will be created only when offer is set.
723 if (action == kOffer && !CreateChannels(local_description()->description())) {
724 // TODO(mallinath) - Handle CreateChannel failure, as new local description
725 // is applied. Restore back to old description.
726 return BadLocalSdp(desc->type(), kCreateChannelFailed, err_desc);
727 }
728
729 // Remove unused channels if MediaContentDescription is rejected.
730 RemoveUnusedChannels(local_description()->description());
731
732 if (!UpdateSessionState(action, cricket::CS_LOCAL, err_desc)) {
733 return false;
734 }
735 if (remote_description()) {
736 // Now that we have a local description, we can push down remote candidates.
737 UseCandidatesInSessionDescription(remote_description());
738 }
739
740 pending_ice_restarts_.clear();
741 if (error() != ERROR_NONE) {
742 return BadLocalSdp(desc->type(), GetSessionErrorMsg(), err_desc);
743 }
744 return true;
745 }
746
747 bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
748 std::string* err_desc) {
749 RTC_DCHECK(signaling_thread()->IsCurrent());
750
751 // Takes the ownership of |desc| regardless of the result.
752 std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
753
754 // Validate SDP.
755 if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
756 return false;
757 }
758
759 const SessionDescriptionInterface* old_remote_description =
760 remote_description();
761 // Grab ownership of the description being replaced for the remainder of this
762 // method, since it's used below.
763 std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
764 Action action = GetAction(desc->type());
765 if (action == kAnswer) {
766 replaced_remote_description.reset(
767 pending_remote_description_ ? pending_remote_description_.release()
768 : current_remote_description_.release());
769 current_remote_description_.reset(desc_temp.release());
770 pending_remote_description_.reset(nullptr);
771 current_local_description_.reset(pending_local_description_.release());
772 } else {
773 replaced_remote_description.reset(pending_remote_description_.release());
774 pending_remote_description_.reset(desc_temp.release());
775 }
776
777 // Transport and Media channels will be created only when offer is set.
778 if (action == kOffer && !CreateChannels(desc->description())) {
779 // TODO(mallinath) - Handle CreateChannel failure, as new local description
780 // is applied. Restore back to old description.
781 return BadRemoteSdp(desc->type(), kCreateChannelFailed, err_desc);
782 }
783
784 // Remove unused channels if MediaContentDescription is rejected.
785 RemoveUnusedChannels(desc->description());
786
787 // NOTE: Candidates allocation will be initiated only when SetLocalDescription
788 // is called.
789 if (!UpdateSessionState(action, cricket::CS_REMOTE, err_desc)) {
790 return false;
791 }
792
793 if (local_description() && !UseCandidatesInSessionDescription(desc)) {
794 return BadRemoteSdp(desc->type(), kInvalidCandidates, err_desc);
795 }
796
797 if (old_remote_description) {
798 for (const cricket::ContentInfo& content :
799 old_remote_description->description()->contents()) {
800 // Check if this new SessionDescription contains new ICE ufrag and
801 // password that indicates the remote peer requests an ICE restart.
802 // TODO(deadbeef): When we start storing both the current and pending
803 // remote description, this should reset pending_ice_restarts and compare
804 // against the current description.
805 if (CheckForRemoteIceRestart(old_remote_description, desc,
806 content.name)) {
807 if (action == kOffer) {
808 pending_ice_restarts_.insert(content.name);
809 }
810 } else {
811 // We retain all received candidates only if ICE is not restarted.
812 // When ICE is restarted, all previous candidates belong to an old
813 // generation and should not be kept.
814 // TODO(deadbeef): This goes against the W3C spec which says the remote
815 // description should only contain candidates from the last set remote
816 // description plus any candidates added since then. We should remove
817 // this once we're sure it won't break anything.
818 WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
819 old_remote_description, content.name, desc);
820 }
821 }
822 }
823
824 if (error() != ERROR_NONE) {
825 return BadRemoteSdp(desc->type(), GetSessionErrorMsg(), err_desc);
826 }
827
828 // Set the the ICE connection state to connecting since the connection may
829 // become writable with peer reflexive candidates before any remote candidate
830 // is signaled.
831 // TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
832 // is to have a new signal the indicates a change in checking state from the
833 // transport and expose a new checking() member from transport that can be
834 // read to determine the current checking state. The existing SignalConnecting
835 // actually means "gathering candidates", so cannot be be used here.
836 if (desc->type() != SessionDescriptionInterface::kOffer &&
837 ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew) {
838 SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
839 }
840 return true;
841 }
842
843 void WebRtcSession::LogState(State old_state, State new_state) {
844 LOG(LS_INFO) << "Session:" << id()
845 << " Old state:" << GetStateString(old_state)
846 << " New state:" << GetStateString(new_state);
847 }
848
849 void WebRtcSession::SetState(State state) {
850 RTC_DCHECK(signaling_thread_->IsCurrent());
851 if (state != state_) {
852 LogState(state_, state);
853 state_ = state;
854 SignalState(this, state_);
855 }
856 }
857
858 void WebRtcSession::SetError(Error error, const std::string& error_desc) {
859 RTC_DCHECK(signaling_thread_->IsCurrent());
860 if (error != error_) {
861 error_ = error;
862 error_desc_ = error_desc;
863 }
864 }
865
866 bool WebRtcSession::UpdateSessionState(
867 Action action, cricket::ContentSource source,
868 std::string* err_desc) {
869 RTC_DCHECK(signaling_thread()->IsCurrent());
870
871 // If there's already a pending error then no state transition should happen.
872 // But all call-sites should be verifying this before calling us!
873 RTC_DCHECK(error() == ERROR_NONE);
874 std::string td_err;
875 if (action == kOffer) {
876 if (!PushdownTransportDescription(source, cricket::CA_OFFER, &td_err)) {
877 return BadOfferSdp(source, MakeTdErrorString(td_err), err_desc);
878 }
879 SetState(source == cricket::CS_LOCAL ? STATE_SENTOFFER
880 : STATE_RECEIVEDOFFER);
881 if (!PushdownMediaDescription(cricket::CA_OFFER, source, err_desc)) {
882 SetError(ERROR_CONTENT, *err_desc);
883 }
884 if (error() != ERROR_NONE) {
885 return BadOfferSdp(source, GetSessionErrorMsg(), err_desc);
886 }
887 } else if (action == kPrAnswer) {
888 if (!PushdownTransportDescription(source, cricket::CA_PRANSWER, &td_err)) {
889 return BadPranswerSdp(source, MakeTdErrorString(td_err), err_desc);
890 }
891 EnableChannels();
892 SetState(source == cricket::CS_LOCAL ? STATE_SENTPRANSWER
893 : STATE_RECEIVEDPRANSWER);
894 if (!PushdownMediaDescription(cricket::CA_PRANSWER, source, err_desc)) {
895 SetError(ERROR_CONTENT, *err_desc);
896 }
897 if (error() != ERROR_NONE) {
898 return BadPranswerSdp(source, GetSessionErrorMsg(), err_desc);
899 }
900 } else if (action == kAnswer) {
901 const cricket::ContentGroup* local_bundle =
902 local_description()->description()->GetGroupByName(
903 cricket::GROUP_TYPE_BUNDLE);
904 const cricket::ContentGroup* remote_bundle =
905 remote_description()->description()->GetGroupByName(
906 cricket::GROUP_TYPE_BUNDLE);
907 if (local_bundle && remote_bundle) {
908 // The answerer decides the transport to bundle on.
909 const cricket::ContentGroup* answer_bundle =
910 (source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
911 if (!EnableBundle(*answer_bundle)) {
912 LOG(LS_WARNING) << "Failed to enable BUNDLE.";
913 return BadAnswerSdp(source, kEnableBundleFailed, err_desc);
914 }
915 }
916 // Only push down the transport description after enabling BUNDLE; we don't
917 // want to push down a description on a transport about to be destroyed.
918 if (!PushdownTransportDescription(source, cricket::CA_ANSWER, &td_err)) {
919 return BadAnswerSdp(source, MakeTdErrorString(td_err), err_desc);
920 }
921 EnableChannels();
922 SetState(STATE_INPROGRESS);
923 if (!PushdownMediaDescription(cricket::CA_ANSWER, source, err_desc)) {
924 SetError(ERROR_CONTENT, *err_desc);
925 }
926 if (error() != ERROR_NONE) {
927 return BadAnswerSdp(source, GetSessionErrorMsg(), err_desc);
928 }
929 }
930 return true;
931 }
932
933 WebRtcSession::Action WebRtcSession::GetAction(const std::string& type) {
934 if (type == SessionDescriptionInterface::kOffer) {
935 return WebRtcSession::kOffer;
936 } else if (type == SessionDescriptionInterface::kPrAnswer) {
937 return WebRtcSession::kPrAnswer;
938 } else if (type == SessionDescriptionInterface::kAnswer) {
939 return WebRtcSession::kAnswer;
940 }
941 RTC_NOTREACHED() << "unknown action type";
942 return WebRtcSession::kOffer;
943 }
944
945 bool WebRtcSession::PushdownMediaDescription(
946 cricket::ContentAction action,
947 cricket::ContentSource source,
948 std::string* err) {
949 auto set_content = [this, action, source, err](cricket::BaseChannel* ch) {
950 if (!ch) {
951 return true;
952 } else if (source == cricket::CS_LOCAL) {
953 return ch->PushdownLocalDescription(local_description()->description(),
954 action, err);
955 } else {
956 return ch->PushdownRemoteDescription(remote_description()->description(),
957 action, err);
958 }
959 };
960
961 bool ret = (set_content(voice_channel()) && set_content(video_channel()) &&
962 set_content(rtp_data_channel()));
963 // Need complete offer/answer with an SCTP m= section before starting SCTP,
964 // according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
965 if (sctp_transport_ && local_description() && remote_description() &&
966 cricket::GetFirstDataContent(local_description()->description()) &&
967 cricket::GetFirstDataContent(remote_description()->description())) {
968 ret &= network_thread_->Invoke<bool>(
969 RTC_FROM_HERE,
970 rtc::Bind(&WebRtcSession::PushdownSctpParameters_n, this, source));
971 }
972 return ret;
973 }
974
975 bool WebRtcSession::PushdownSctpParameters_n(cricket::ContentSource source) {
976 RTC_DCHECK(network_thread_->IsCurrent());
977 RTC_DCHECK(local_description());
978 RTC_DCHECK(remote_description());
979 // Apply the SCTP port (which is hidden inside a DataCodec structure...)
980 // When we support "max-message-size", that would also be pushed down here.
981 return sctp_transport_->Start(
982 GetSctpPort(local_description()->description()),
983 GetSctpPort(remote_description()->description()));
984 }
985
986 bool WebRtcSession::PushdownTransportDescription(cricket::ContentSource source,
987 cricket::ContentAction action,
988 std::string* error_desc) {
989 RTC_DCHECK(signaling_thread()->IsCurrent());
990
991 if (source == cricket::CS_LOCAL) {
992 return PushdownLocalTransportDescription(local_description()->description(),
993 action, error_desc);
994 }
995 return PushdownRemoteTransportDescription(remote_description()->description(),
996 action, error_desc);
997 }
998
999 bool WebRtcSession::PushdownLocalTransportDescription(
1000 const SessionDescription* sdesc,
1001 cricket::ContentAction action,
1002 std::string* err) {
1003 RTC_DCHECK(signaling_thread()->IsCurrent());
1004
1005 if (!sdesc) {
1006 return false;
1007 }
1008
1009 for (const TransportInfo& tinfo : sdesc->transport_infos()) {
1010 if (!transport_controller_->SetLocalTransportDescription(
1011 tinfo.content_name, tinfo.description, action, err)) {
1012 return false;
1013 }
1014 }
1015
1016 return true;
1017 }
1018
1019 bool WebRtcSession::PushdownRemoteTransportDescription(
1020 const SessionDescription* sdesc,
1021 cricket::ContentAction action,
1022 std::string* err) {
1023 RTC_DCHECK(signaling_thread()->IsCurrent());
1024
1025 if (!sdesc) {
1026 return false;
1027 }
1028
1029 for (const TransportInfo& tinfo : sdesc->transport_infos()) {
1030 if (!transport_controller_->SetRemoteTransportDescription(
1031 tinfo.content_name, tinfo.description, action, err)) {
1032 return false;
1033 }
1034 }
1035
1036 return true;
1037 }
1038
1039 bool WebRtcSession::GetTransportDescription(
1040 const SessionDescription* description,
1041 const std::string& content_name,
1042 cricket::TransportDescription* tdesc) {
1043 if (!description || !tdesc) {
1044 return false;
1045 }
1046 const TransportInfo* transport_info =
1047 description->GetTransportInfoByName(content_name);
1048 if (!transport_info) {
1049 return false;
1050 }
1051 *tdesc = transport_info->description;
1052 return true;
1053 }
1054
1055 bool WebRtcSession::EnableBundle(const cricket::ContentGroup& bundle) {
1056 const std::string* first_content_name = bundle.FirstContentName();
1057 if (!first_content_name) {
1058 LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
1059 return false;
1060 }
1061 const std::string& transport_name = *first_content_name;
1062
1063 #ifdef HAVE_QUIC
1064 if (quic_data_transport_ &&
1065 bundle.HasContentName(quic_data_transport_->content_name()) &&
1066 quic_data_transport_->transport_name() != transport_name) {
1067 LOG(LS_ERROR) << "Unable to BUNDLE " << quic_data_transport_->content_name()
1068 << " on " << transport_name << "with QUIC.";
1069 }
1070 #endif
1071 auto maybe_set_transport = [this, bundle,
1072 transport_name](cricket::BaseChannel* ch) {
1073 if (!ch || !bundle.HasContentName(ch->content_name())) {
1074 return true;
1075 }
1076
1077 std::string old_transport_name = ch->transport_name();
1078 if (old_transport_name == transport_name) {
1079 LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
1080 << " on " << transport_name << ".";
1081 return true;
1082 }
1083
1084 cricket::DtlsTransportInternal* rtp_dtls_transport =
1085 transport_controller_->CreateDtlsTransport(
1086 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1087 bool need_rtcp = (ch->rtcp_dtls_transport() != nullptr);
1088 cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
1089 if (need_rtcp) {
1090 rtcp_dtls_transport = transport_controller_->CreateDtlsTransport_n(
1091 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
1092 }
1093
1094 ch->SetTransports(rtp_dtls_transport, rtcp_dtls_transport);
1095 LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
1096 << transport_name << ".";
1097 transport_controller_->DestroyDtlsTransport(
1098 old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1099 // If the channel needs rtcp, it means that the channel used to have a
1100 // rtcp transport which needs to be deleted now.
1101 if (need_rtcp) {
1102 transport_controller_->DestroyDtlsTransport(
1103 old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
1104 }
1105 return true;
1106 };
1107
1108 if (!maybe_set_transport(voice_channel()) ||
1109 !maybe_set_transport(video_channel()) ||
1110 !maybe_set_transport(rtp_data_channel())) {
1111 return false;
1112 }
1113 // For SCTP, transport creation/deletion happens here instead of in the
1114 // object itself.
1115 if (sctp_transport_) {
1116 RTC_DCHECK(sctp_transport_name_);
1117 RTC_DCHECK(sctp_content_name_);
1118 if (transport_name != *sctp_transport_name_ &&
1119 bundle.HasContentName(*sctp_content_name_)) {
1120 network_thread_->Invoke<void>(
1121 RTC_FROM_HERE, rtc::Bind(&WebRtcSession::ChangeSctpTransport_n, this,
1122 transport_name));
1123 }
1124 }
1125
1126 return true;
1127 }
1128
1129 bool WebRtcSession::ProcessIceMessage(const IceCandidateInterface* candidate) {
1130 if (!remote_description()) {
1131 LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
1132 << "without any remote session description.";
1133 return false;
1134 }
1135
1136 if (!candidate) {
1137 LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL.";
1138 return false;
1139 }
1140
1141 bool valid = false;
1142 bool ready = ReadyToUseRemoteCandidate(candidate, NULL, &valid);
1143 if (!valid) {
1144 return false;
1145 }
1146
1147 // Add this candidate to the remote session description.
1148 if (!mutable_remote_description()->AddCandidate(candidate)) {
1149 LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used.";
1150 return false;
1151 }
1152
1153 if (ready) {
1154 return UseCandidate(candidate);
1155 } else {
1156 LOG(LS_INFO) << "ProcessIceMessage: Not ready to use candidate.";
1157 return true;
1158 }
1159 }
1160
1161 bool WebRtcSession::RemoveRemoteIceCandidates(
1162 const std::vector<cricket::Candidate>& candidates) {
1163 if (!remote_description()) {
1164 LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
1165 << "removed without any remote session description.";
1166 return false;
1167 }
1168
1169 if (candidates.empty()) {
1170 LOG(LS_ERROR) << "RemoveRemoteIceCandidates: candidates are empty.";
1171 return false;
1172 }
1173
1174 size_t number_removed =
1175 mutable_remote_description()->RemoveCandidates(candidates);
1176 if (number_removed != candidates.size()) {
1177 LOG(LS_ERROR) << "RemoveRemoteIceCandidates: Failed to remove candidates. "
1178 << "Requested " << candidates.size() << " but only "
1179 << number_removed << " are removed.";
1180 }
1181
1182 // Remove the candidates from the transport controller.
1183 std::string error;
1184 bool res = transport_controller_->RemoveRemoteCandidates(candidates, &error);
1185 if (!res && !error.empty()) {
1186 LOG(LS_ERROR) << "Error when removing remote candidates: " << error;
1187 }
1188 return true;
1189 }
1190
1191 cricket::IceConfig WebRtcSession::ParseIceConfig(
1192 const PeerConnectionInterface::RTCConfiguration& config) const {
1193 cricket::ContinualGatheringPolicy gathering_policy;
1194 // TODO(honghaiz): Add the third continual gathering policy in
1195 // PeerConnectionInterface and map it to GATHER_CONTINUALLY_AND_RECOVER.
1196 switch (config.continual_gathering_policy) {
1197 case PeerConnectionInterface::GATHER_ONCE:
1198 gathering_policy = cricket::GATHER_ONCE;
1199 break;
1200 case PeerConnectionInterface::GATHER_CONTINUALLY:
1201 gathering_policy = cricket::GATHER_CONTINUALLY;
1202 break;
1203 default:
1204 RTC_NOTREACHED();
1205 gathering_policy = cricket::GATHER_ONCE;
1206 }
1207 cricket::IceConfig ice_config;
1208 ice_config.receiving_timeout = config.ice_connection_receiving_timeout;
1209 ice_config.prioritize_most_likely_candidate_pairs =
1210 config.prioritize_most_likely_ice_candidate_pairs;
1211 ice_config.backup_connection_ping_interval =
1212 config.ice_backup_candidate_pair_ping_interval;
1213 ice_config.continual_gathering_policy = gathering_policy;
1214 ice_config.presume_writable_when_fully_relayed =
1215 config.presume_writable_when_fully_relayed;
1216 return ice_config;
1217 }
1218
1219 void WebRtcSession::SetIceConfig(const cricket::IceConfig& config) {
1220 transport_controller_->SetIceConfig(config);
1221 }
1222
1223 void WebRtcSession::MaybeStartGathering() {
1224 transport_controller_->MaybeStartGathering();
1225 }
1226
1227 bool WebRtcSession::GetLocalTrackIdBySsrc(uint32_t ssrc,
1228 std::string* track_id) {
1229 if (!local_description()) {
1230 return false;
1231 }
1232 return webrtc::GetTrackIdBySsrc(local_description()->description(), ssrc,
1233 track_id);
1234 }
1235
1236 bool WebRtcSession::GetRemoteTrackIdBySsrc(uint32_t ssrc,
1237 std::string* track_id) {
1238 if (!remote_description()) {
1239 return false;
1240 }
1241 return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
1242 track_id);
1243 }
1244
1245 std::string WebRtcSession::BadStateErrMsg(State state) {
1246 std::ostringstream desc;
1247 desc << "Called in wrong state: " << GetStateString(state);
1248 return desc.str();
1249 }
1250
1251 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
1252 RTC_DCHECK(signaling_thread()->IsCurrent());
1253 if (!voice_channel_) {
1254 LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
1255 return false;
1256 }
1257 uint32_t send_ssrc = 0;
1258 // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
1259 // exists.
1260 if (!local_description() ||
1261 !GetAudioSsrcByTrackId(local_description()->description(), track_id,
1262 &send_ssrc)) {
1263 LOG(LS_ERROR) << "CanInsertDtmf: Track does not exist: " << track_id;
1264 return false;
1265 }
1266 return voice_channel_->CanInsertDtmf();
1267 }
1268
1269 bool WebRtcSession::InsertDtmf(const std::string& track_id,
1270 int code, int duration) {
1271 RTC_DCHECK(signaling_thread()->IsCurrent());
1272 if (!voice_channel_) {
1273 LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
1274 return false;
1275 }
1276 uint32_t send_ssrc = 0;
1277 if (!VERIFY(local_description() &&
1278 GetAudioSsrcByTrackId(local_description()->description(),
1279 track_id, &send_ssrc))) {
1280 LOG(LS_ERROR) << "InsertDtmf: Track does not exist: " << track_id;
1281 return false;
1282 }
1283 if (!voice_channel_->InsertDtmf(send_ssrc, code, duration)) {
1284 LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
1285 return false;
1286 }
1287 return true;
1288 }
1289
1290 sigslot::signal0<>* WebRtcSession::GetOnDestroyedSignal() {
1291 return &SignalDestroyed;
1292 }
1293
1294 bool WebRtcSession::SendData(const cricket::SendDataParams& params,
1295 const rtc::CopyOnWriteBuffer& payload,
1296 cricket::SendDataResult* result) {
1297 if (!rtp_data_channel_ && !sctp_transport_) {
1298 LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
1299 << "and sctp_transport_ are NULL.";
1300 return false;
1301 }
1302 return rtp_data_channel_
1303 ? rtp_data_channel_->SendData(params, payload, result)
1304 : network_thread_->Invoke<bool>(
1305 RTC_FROM_HERE,
1306 Bind(&cricket::SctpTransportInternal::SendData,
1307 sctp_transport_.get(), params, payload, result));
1308 }
1309
1310 bool WebRtcSession::ConnectDataChannel(DataChannel* webrtc_data_channel) {
1311 if (!rtp_data_channel_ && !sctp_transport_) {
1312 // Don't log an error here, because DataChannels are expected to call
1313 // ConnectDataChannel in this state. It's the only way to initially tell
1314 // whether or not the underlying transport is ready.
1315 return false;
1316 }
1317 if (rtp_data_channel_) {
1318 rtp_data_channel_->SignalReadyToSendData.connect(
1319 webrtc_data_channel, &DataChannel::OnChannelReady);
1320 rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
1321 &DataChannel::OnDataReceived);
1322 } else {
1323 SignalSctpReadyToSendData.connect(webrtc_data_channel,
1324 &DataChannel::OnChannelReady);
1325 SignalSctpDataReceived.connect(webrtc_data_channel,
1326 &DataChannel::OnDataReceived);
1327 SignalSctpStreamClosedRemotely.connect(
1328 webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
1329 }
1330 return true;
1331 }
1332
1333 void WebRtcSession::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
1334 if (!rtp_data_channel_ && !sctp_transport_) {
1335 LOG(LS_ERROR) << "DisconnectDataChannel called when rtp_data_channel_ and "
1336 "sctp_transport_ are NULL.";
1337 return;
1338 }
1339 if (rtp_data_channel_) {
1340 rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
1341 rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
1342 } else {
1343 SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
1344 SignalSctpDataReceived.disconnect(webrtc_data_channel);
1345 SignalSctpStreamClosedRemotely.disconnect(webrtc_data_channel);
1346 }
1347 }
1348
1349 void WebRtcSession::AddSctpDataStream(int sid) {
1350 if (!sctp_transport_) {
1351 LOG(LS_ERROR) << "AddSctpDataStream called when sctp_transport_ is NULL.";
1352 return;
1353 }
1354 network_thread_->Invoke<void>(
1355 RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
1356 sctp_transport_.get(), sid));
1357 }
1358
1359 void WebRtcSession::RemoveSctpDataStream(int sid) {
1360 if (!sctp_transport_) {
1361 LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
1362 << "NULL.";
1363 return;
1364 }
1365 network_thread_->Invoke<void>(
1366 RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
1367 sctp_transport_.get(), sid));
1368 }
1369
1370 bool WebRtcSession::ReadyToSendData() const {
1371 return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
1372 sctp_ready_to_send_data_;
1373 }
1374
1375 std::unique_ptr<SessionStats> WebRtcSession::GetStats_s() {
1376 RTC_DCHECK(signaling_thread()->IsCurrent());
1377 ChannelNamePairs channel_name_pairs;
1378 if (voice_channel()) {
1379 channel_name_pairs.voice = rtc::Optional<ChannelNamePair>(ChannelNamePair(
1380 voice_channel()->content_name(), voice_channel()->transport_name()));
1381 }
1382 if (video_channel()) {
1383 channel_name_pairs.video = rtc::Optional<ChannelNamePair>(ChannelNamePair(
1384 video_channel()->content_name(), video_channel()->transport_name()));
1385 }
1386 if (rtp_data_channel()) {
1387 channel_name_pairs.data = rtc::Optional<ChannelNamePair>(
1388 ChannelNamePair(rtp_data_channel()->content_name(),
1389 rtp_data_channel()->transport_name()));
1390 }
1391 if (sctp_transport_) {
1392 RTC_DCHECK(sctp_content_name_);
1393 RTC_DCHECK(sctp_transport_name_);
1394 channel_name_pairs.data = rtc::Optional<ChannelNamePair>(
1395 ChannelNamePair(*sctp_content_name_, *sctp_transport_name_));
1396 }
1397 return GetStats(channel_name_pairs);
1398 }
1399
1400 std::unique_ptr<SessionStats> WebRtcSession::GetStats(
1401 const ChannelNamePairs& channel_name_pairs) {
1402 if (network_thread()->IsCurrent()) {
1403 return GetStats_n(channel_name_pairs);
1404 }
1405 return network_thread()->Invoke<std::unique_ptr<SessionStats>>(
1406 RTC_FROM_HERE,
1407 rtc::Bind(&WebRtcSession::GetStats_n, this, channel_name_pairs));
1408 }
1409
1410 bool WebRtcSession::GetLocalCertificate(
1411 const std::string& transport_name,
1412 rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
1413 return transport_controller_->GetLocalCertificate(transport_name,
1414 certificate);
1415 }
1416
1417 std::unique_ptr<rtc::SSLCertificate> WebRtcSession::GetRemoteSSLCertificate(
1418 const std::string& transport_name) {
1419 return transport_controller_->GetRemoteSSLCertificate(transport_name);
1420 }
1421
1422 cricket::DataChannelType WebRtcSession::data_channel_type() const {
1423 return data_channel_type_;
1424 }
1425
1426 bool WebRtcSession::IceRestartPending(const std::string& content_name) const {
1427 return pending_ice_restarts_.find(content_name) !=
1428 pending_ice_restarts_.end();
1429 }
1430
1431 void WebRtcSession::SetNeedsIceRestartFlag() {
1432 transport_controller_->SetNeedsIceRestartFlag();
1433 }
1434
1435 bool WebRtcSession::NeedsIceRestart(const std::string& content_name) const {
1436 return transport_controller_->NeedsIceRestart(content_name);
1437 }
1438
1439 void WebRtcSession::OnCertificateReady(
1440 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
1441 transport_controller_->SetLocalCertificate(certificate);
1442 }
1443
1444 void WebRtcSession::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
1445 SetError(ERROR_TRANSPORT,
1446 rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
1447 }
1448
1449 bool WebRtcSession::waiting_for_certificate_for_testing() const {
1450 return webrtc_session_desc_factory_->waiting_for_certificate_for_testing();
1451 }
1452
1453 const rtc::scoped_refptr<rtc::RTCCertificate>&
1454 WebRtcSession::certificate_for_testing() {
1455 return transport_controller_->certificate_for_testing();
1456 }
1457
1458 void WebRtcSession::SetIceConnectionState(
1459 PeerConnectionInterface::IceConnectionState state) {
1460 if (ice_connection_state_ == state) {
1461 return;
1462 }
1463
1464 LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
1465 << " => " << state;
1466 RTC_DCHECK(ice_connection_state_ !=
1467 PeerConnectionInterface::kIceConnectionClosed);
1468 ice_connection_state_ = state;
1469 if (ice_observer_) {
1470 ice_observer_->OnIceConnectionChange(ice_connection_state_);
1471 }
1472 }
1473
1474 void WebRtcSession::OnTransportControllerConnectionState(
1475 cricket::IceConnectionState state) {
1476 switch (state) {
1477 case cricket::kIceConnectionConnecting:
1478 // If the current state is Connected or Completed, then there were
1479 // writable channels but now there are not, so the next state must
1480 // be Disconnected.
1481 // kIceConnectionConnecting is currently used as the default,
1482 // un-connected state by the TransportController, so its only use is
1483 // detecting disconnections.
1484 if (ice_connection_state_ ==
1485 PeerConnectionInterface::kIceConnectionConnected ||
1486 ice_connection_state_ ==
1487 PeerConnectionInterface::kIceConnectionCompleted) {
1488 SetIceConnectionState(
1489 PeerConnectionInterface::kIceConnectionDisconnected);
1490 }
1491 break;
1492 case cricket::kIceConnectionFailed:
1493 SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
1494 break;
1495 case cricket::kIceConnectionConnected:
1496 LOG(LS_INFO) << "Changing to ICE connected state because "
1497 << "all transports are writable.";
1498 SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
1499 break;
1500 case cricket::kIceConnectionCompleted:
1501 LOG(LS_INFO) << "Changing to ICE completed state because "
1502 << "all transports are complete.";
1503 if (ice_connection_state_ !=
1504 PeerConnectionInterface::kIceConnectionConnected) {
1505 // If jumping directly from "checking" to "connected",
1506 // signal "connected" first.
1507 SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
1508 }
1509 SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
1510 if (metrics_observer_) {
1511 ReportTransportStats();
1512 }
1513 break;
1514 default:
1515 RTC_NOTREACHED();
1516 }
1517 }
1518
1519 void WebRtcSession::OnTransportControllerReceiving(bool receiving) {
1520 SetIceConnectionReceiving(receiving);
1521 }
1522
1523 void WebRtcSession::SetIceConnectionReceiving(bool receiving) {
1524 if (ice_connection_receiving_ == receiving) {
1525 return;
1526 }
1527 ice_connection_receiving_ = receiving;
1528 if (ice_observer_) {
1529 ice_observer_->OnIceConnectionReceivingChange(receiving);
1530 }
1531 }
1532
1533 void WebRtcSession::OnTransportControllerCandidatesGathered(
1534 const std::string& transport_name,
1535 const cricket::Candidates& candidates) {
1536 RTC_DCHECK(signaling_thread()->IsCurrent());
1537 int sdp_mline_index;
1538 if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
1539 LOG(LS_ERROR) << "OnTransportControllerCandidatesGathered: content name "
1540 << transport_name << " not found";
1541 return;
1542 }
1543
1544 for (cricket::Candidates::const_iterator citer = candidates.begin();
1545 citer != candidates.end(); ++citer) {
1546 // Use transport_name as the candidate media id.
1547 JsepIceCandidate candidate(transport_name, sdp_mline_index, *citer);
1548 if (ice_observer_) {
1549 ice_observer_->OnIceCandidate(&candidate);
1550 }
1551 if (local_description()) {
1552 mutable_local_description()->AddCandidate(&candidate);
1553 }
1554 }
1555 }
1556
1557 void WebRtcSession::OnTransportControllerCandidatesRemoved(
1558 const std::vector<cricket::Candidate>& candidates) {
1559 RTC_DCHECK(signaling_thread()->IsCurrent());
1560 // Sanity check.
1561 for (const cricket::Candidate& candidate : candidates) {
1562 if (candidate.transport_name().empty()) {
1563 LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
1564 << "empty content name in candidate "
1565 << candidate.ToString();
1566 return;
1567 }
1568 }
1569
1570 if (local_description()) {
1571 mutable_local_description()->RemoveCandidates(candidates);
1572 }
1573 if (ice_observer_) {
1574 ice_observer_->OnIceCandidatesRemoved(candidates);
1575 }
1576 }
1577
1578 void WebRtcSession::OnTransportControllerDtlsHandshakeError(
1579 rtc::SSLHandshakeError error) {
1580 if (metrics_observer_) {
1581 metrics_observer_->IncrementEnumCounter(
1582 webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
1583 static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
1584 }
1585 }
1586
1587 // Enabling voice and video (and RTP data) channel.
1588 void WebRtcSession::EnableChannels() {
1589 if (voice_channel_ && !voice_channel_->enabled())
1590 voice_channel_->Enable(true);
1591
1592 if (video_channel_ && !video_channel_->enabled())
1593 video_channel_->Enable(true);
1594
1595 if (rtp_data_channel_ && !rtp_data_channel_->enabled())
1596 rtp_data_channel_->Enable(true);
1597 }
1598
1599 // Returns the media index for a local ice candidate given the content name.
1600 bool WebRtcSession::GetLocalCandidateMediaIndex(const std::string& content_name,
1601 int* sdp_mline_index) {
1602 if (!local_description() || !sdp_mline_index) {
1603 return false;
1604 }
1605
1606 bool content_found = false;
1607 const ContentInfos& contents = local_description()->description()->contents();
1608 for (size_t index = 0; index < contents.size(); ++index) {
1609 if (contents[index].name == content_name) {
1610 *sdp_mline_index = static_cast<int>(index);
1611 content_found = true;
1612 break;
1613 }
1614 }
1615 return content_found;
1616 }
1617
1618 bool WebRtcSession::UseCandidatesInSessionDescription(
1619 const SessionDescriptionInterface* remote_desc) {
1620 if (!remote_desc) {
1621 return true;
1622 }
1623 bool ret = true;
1624
1625 for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
1626 const IceCandidateCollection* candidates = remote_desc->candidates(m);
1627 for (size_t n = 0; n < candidates->count(); ++n) {
1628 const IceCandidateInterface* candidate = candidates->at(n);
1629 bool valid = false;
1630 if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
1631 if (valid) {
1632 LOG(LS_INFO) << "UseCandidatesInSessionDescription: Not ready to use "
1633 << "candidate.";
1634 }
1635 continue;
1636 }
1637 ret = UseCandidate(candidate);
1638 if (!ret) {
1639 break;
1640 }
1641 }
1642 }
1643 return ret;
1644 }
1645
1646 bool WebRtcSession::UseCandidate(const IceCandidateInterface* candidate) {
1647 size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
1648 size_t remote_content_size =
1649 remote_description()->description()->contents().size();
1650 if (mediacontent_index >= remote_content_size) {
1651 LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
1652 return false;
1653 }
1654
1655 cricket::ContentInfo content =
1656 remote_description()->description()->contents()[mediacontent_index];
1657 std::vector<cricket::Candidate> candidates;
1658 candidates.push_back(candidate->candidate());
1659 // Invoking BaseSession method to handle remote candidates.
1660 std::string error;
1661 if (transport_controller_->AddRemoteCandidates(content.name, candidates,
1662 &error)) {
1663 // Candidates successfully submitted for checking.
1664 if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
1665 ice_connection_state_ ==
1666 PeerConnectionInterface::kIceConnectionDisconnected) {
1667 // If state is New, then the session has just gotten its first remote ICE
1668 // candidates, so go to Checking.
1669 // If state is Disconnected, the session is re-using old candidates or
1670 // receiving additional ones, so go to Checking.
1671 // If state is Connected, stay Connected.
1672 // TODO(bemasc): If state is Connected, and the new candidates are for a
1673 // newly added transport, then the state actually _should_ move to
1674 // checking. Add a way to distinguish that case.
1675 SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
1676 }
1677 // TODO(bemasc): If state is Completed, go back to Connected.
1678 } else {
1679 if (!error.empty()) {
1680 LOG(LS_WARNING) << error;
1681 }
1682 }
1683 return true;
1684 }
1685
1686 void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
1687 // Destroy video_channel_ first since it may have a pointer to the
1688 // voice_channel_.
1689 const cricket::ContentInfo* video_info =
1690 cricket::GetFirstVideoContent(desc);
1691 if ((!video_info || video_info->rejected) && video_channel_) {
1692 DestroyVideoChannel();
1693 }
1694
1695 const cricket::ContentInfo* voice_info =
1696 cricket::GetFirstAudioContent(desc);
1697 if ((!voice_info || voice_info->rejected) && voice_channel_) {
1698 DestroyVoiceChannel();
1699 }
1700
1701 const cricket::ContentInfo* data_info =
1702 cricket::GetFirstDataContent(desc);
1703 if (!data_info || data_info->rejected) {
1704 if (rtp_data_channel_) {
1705 DestroyDataChannel();
1706 }
1707 if (sctp_transport_) {
1708 SignalDataChannelDestroyed();
1709 network_thread_->Invoke<void>(
1710 RTC_FROM_HERE,
1711 rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
1712 }
1713 #ifdef HAVE_QUIC
1714 // Clean up the existing QuicDataTransport and its QuicTransportChannels.
1715 if (quic_data_transport_) {
1716 quic_data_transport_.reset();
1717 }
1718 #endif
1719 }
1720 }
1721
1722 // Returns the name of the transport channel when BUNDLE is enabled, or nullptr
1723 // if the channel is not part of any bundle.
1724 const std::string* WebRtcSession::GetBundleTransportName(
1725 const cricket::ContentInfo* content,
1726 const cricket::ContentGroup* bundle) {
1727 if (!bundle) {
1728 return nullptr;
1729 }
1730 const std::string* first_content_name = bundle->FirstContentName();
1731 if (!first_content_name) {
1732 LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
1733 return nullptr;
1734 }
1735 if (!bundle->HasContentName(content->name)) {
1736 LOG(LS_WARNING) << content->name << " is not part of any bundle group";
1737 return nullptr;
1738 }
1739 LOG(LS_INFO) << "Bundling " << content->name << " on " << *first_content_name;
1740 return first_content_name;
1741 }
1742
1743 bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
1744 const cricket::ContentGroup* bundle_group = nullptr;
1745 if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
1746 bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
1747 if (!bundle_group) {
1748 LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
1749 return false;
1750 }
1751 }
1752 // Creating the media channels and transport proxies.
1753 const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
1754 if (voice && !voice->rejected && !voice_channel_) {
1755 if (!CreateVoiceChannel(voice,
1756 GetBundleTransportName(voice, bundle_group))) {
1757 LOG(LS_ERROR) << "Failed to create voice channel.";
1758 return false;
1759 }
1760 }
1761
1762 const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
1763 if (video && !video->rejected && !video_channel_) {
1764 if (!CreateVideoChannel(video,
1765 GetBundleTransportName(video, bundle_group))) {
1766 LOG(LS_ERROR) << "Failed to create video channel.";
1767 return false;
1768 }
1769 }
1770
1771 const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
1772 if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
1773 !rtp_data_channel_ && !sctp_transport_) {
1774 if (!CreateDataChannel(data, GetBundleTransportName(data, bundle_group))) {
1775 LOG(LS_ERROR) << "Failed to create data channel.";
1776 return false;
1777 }
1778 }
1779
1780 return true;
1781 }
1782
1783 bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content,
1784 const std::string* bundle_transport) {
1785 bool require_rtcp_mux =
1786 rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
1787
1788 std::string transport_name =
1789 bundle_transport ? *bundle_transport : content->name;
1790
1791 cricket::DtlsTransportInternal* rtp_dtls_transport =
1792 transport_controller_->CreateDtlsTransport(
1793 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1794 cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
1795 if (!require_rtcp_mux) {
1796 rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
1797 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
1798 }
1799
1800 voice_channel_.reset(channel_manager_->CreateVoiceChannel(
1801 media_controller_, rtp_dtls_transport, rtcp_dtls_transport,
1802 transport_controller_->signaling_thread(), content->name,
1803 bundle_transport, require_rtcp_mux, SrtpRequired(), audio_options_));
1804 if (!voice_channel_) {
1805 transport_controller_->DestroyDtlsTransport(
1806 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1807 if (rtcp_dtls_transport) {
1808 transport_controller_->DestroyDtlsTransport(
1809 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1810 }
1811 return false;
1812 }
1813
1814 voice_channel_->SignalRtcpMuxFullyActive.connect(
1815 this, &WebRtcSession::DestroyRtcpTransport_n);
1816 voice_channel_->SignalDtlsSrtpSetupFailure.connect(
1817 this, &WebRtcSession::OnDtlsSrtpSetupFailure);
1818
1819 SignalVoiceChannelCreated();
1820 voice_channel_->SignalSentPacket.connect(this,
1821 &WebRtcSession::OnSentPacket_w);
1822 return true;
1823 }
1824
1825 bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
1826 const std::string* bundle_transport) {
1827 bool require_rtcp_mux =
1828 rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
1829
1830 std::string transport_name =
1831 bundle_transport ? *bundle_transport : content->name;
1832
1833 cricket::DtlsTransportInternal* rtp_dtls_transport =
1834 transport_controller_->CreateDtlsTransport(
1835 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1836 cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
1837 if (!require_rtcp_mux) {
1838 rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
1839 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
1840 }
1841
1842 video_channel_.reset(channel_manager_->CreateVideoChannel(
1843 media_controller_, rtp_dtls_transport, rtcp_dtls_transport,
1844 transport_controller_->signaling_thread(), content->name,
1845 bundle_transport, require_rtcp_mux, SrtpRequired(), video_options_));
1846
1847 if (!video_channel_) {
1848 transport_controller_->DestroyDtlsTransport(
1849 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1850 if (rtcp_dtls_transport) {
1851 transport_controller_->DestroyDtlsTransport(
1852 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1853 }
1854 return false;
1855 }
1856
1857 video_channel_->SignalRtcpMuxFullyActive.connect(
1858 this, &WebRtcSession::DestroyRtcpTransport_n);
1859 video_channel_->SignalDtlsSrtpSetupFailure.connect(
1860 this, &WebRtcSession::OnDtlsSrtpSetupFailure);
1861
1862 SignalVideoChannelCreated();
1863 video_channel_->SignalSentPacket.connect(this,
1864 &WebRtcSession::OnSentPacket_w);
1865 return true;
1866 }
1867
1868 bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
1869 const std::string* bundle_transport) {
1870 const std::string transport_name =
1871 bundle_transport ? *bundle_transport : content->name;
1872 #ifdef HAVE_QUIC
1873 if (data_channel_type_ == cricket::DCT_QUIC) {
1874 RTC_DCHECK(transport_controller_->quic());
1875 quic_data_transport_->SetTransports(transport_name);
1876 return true;
1877 }
1878 #endif // HAVE_QUIC
1879 bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
1880 if (sctp) {
1881 if (!sctp_factory_) {
1882 LOG(LS_ERROR)
1883 << "Trying to create SCTP transport, but didn't compile with "
1884 "SCTP support (HAVE_SCTP)";
1885 return false;
1886 }
1887 if (!network_thread_->Invoke<bool>(
1888 RTC_FROM_HERE, rtc::Bind(&WebRtcSession::CreateSctpTransport_n,
1889 this, content->name, transport_name))) {
1890 return false;
1891 };
1892 } else {
1893 bool require_rtcp_mux =
1894 rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
1895
1896 std::string transport_name =
1897 bundle_transport ? *bundle_transport : content->name;
1898 cricket::DtlsTransportInternal* rtp_dtls_transport =
1899 transport_controller_->CreateDtlsTransport(
1900 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1901 cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
1902 if (!require_rtcp_mux) {
1903 rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
1904 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
1905 }
1906
1907 rtp_data_channel_.reset(channel_manager_->CreateRtpDataChannel(
1908 media_controller_, rtp_dtls_transport, rtcp_dtls_transport,
1909 transport_controller_->signaling_thread(), content->name,
1910 bundle_transport, require_rtcp_mux, SrtpRequired()));
1911
1912 if (!rtp_data_channel_) {
1913 transport_controller_->DestroyDtlsTransport(
1914 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1915 if (rtcp_dtls_transport) {
1916 transport_controller_->DestroyDtlsTransport(
1917 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1918 }
1919 return false;
1920 }
1921
1922 rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
1923 this, &WebRtcSession::DestroyRtcpTransport_n);
1924 rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
1925 this, &WebRtcSession::OnDtlsSrtpSetupFailure);
1926 rtp_data_channel_->SignalSentPacket.connect(this,
1927 &WebRtcSession::OnSentPacket_w);
1928 }
1929
1930 SignalDataChannelCreated();
1931
1932 return true;
1933 }
1934
1935 std::unique_ptr<SessionStats> WebRtcSession::GetStats_n(
1936 const ChannelNamePairs& channel_name_pairs) {
1937 RTC_DCHECK(network_thread()->IsCurrent());
1938 std::unique_ptr<SessionStats> session_stats(new SessionStats());
1939 for (const auto channel_name_pair : { &channel_name_pairs.voice,
1940 &channel_name_pairs.video,
1941 &channel_name_pairs.data }) {
1942 if (*channel_name_pair) {
1943 cricket::TransportStats transport_stats;
1944 if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
1945 &transport_stats)) {
1946 return nullptr;
1947 }
1948 session_stats->proxy_to_transport[(*channel_name_pair)->content_name] =
1949 (*channel_name_pair)->transport_name;
1950 session_stats->transport_stats[(*channel_name_pair)->transport_name] =
1951 std::move(transport_stats);
1952 }
1953 }
1954 return session_stats;
1955 }
1956
1957 bool WebRtcSession::CreateSctpTransport_n(const std::string& content_name,
1958 const std::string& transport_name) {
1959 RTC_DCHECK(network_thread_->IsCurrent());
1960 RTC_DCHECK(sctp_factory_);
1961 cricket::DtlsTransportInternal* tc =
1962 transport_controller_->CreateDtlsTransport_n(
1963 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1964 sctp_transport_ = sctp_factory_->CreateSctpTransport(tc);
1965 RTC_DCHECK(sctp_transport_);
1966 sctp_invoker_.reset(new rtc::AsyncInvoker());
1967 sctp_transport_->SignalReadyToSendData.connect(
1968 this, &WebRtcSession::OnSctpTransportReadyToSendData_n);
1969 sctp_transport_->SignalDataReceived.connect(
1970 this, &WebRtcSession::OnSctpTransportDataReceived_n);
1971 sctp_transport_->SignalStreamClosedRemotely.connect(
1972 this, &WebRtcSession::OnSctpStreamClosedRemotely_n);
1973 sctp_transport_name_ = rtc::Optional<std::string>(transport_name);
1974 sctp_content_name_ = rtc::Optional<std::string>(content_name);
1975 return true;
1976 }
1977
1978 void WebRtcSession::ChangeSctpTransport_n(const std::string& transport_name) {
1979 RTC_DCHECK(network_thread_->IsCurrent());
1980 RTC_DCHECK(sctp_transport_);
1981 RTC_DCHECK(sctp_transport_name_);
1982 std::string old_sctp_transport_name = *sctp_transport_name_;
1983 sctp_transport_name_ = rtc::Optional<std::string>(transport_name);
1984 cricket::DtlsTransportInternal* tc =
1985 transport_controller_->CreateDtlsTransport_n(
1986 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1987 sctp_transport_->SetTransportChannel(tc);
1988 transport_controller_->DestroyDtlsTransport_n(
1989 old_sctp_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
1990 }
1991
1992 void WebRtcSession::DestroySctpTransport_n() {
1993 RTC_DCHECK(network_thread_->IsCurrent());
1994 sctp_transport_.reset(nullptr);
1995 sctp_content_name_.reset();
1996 sctp_transport_name_.reset();
1997 sctp_invoker_.reset(nullptr);
1998 sctp_ready_to_send_data_ = false;
1999 }
2000
2001 void WebRtcSession::OnSctpTransportReadyToSendData_n() {
2002 RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
2003 RTC_DCHECK(network_thread_->IsCurrent());
2004 sctp_invoker_->AsyncInvoke<void>(
2005 RTC_FROM_HERE, signaling_thread_,
2006 rtc::Bind(&WebRtcSession::OnSctpTransportReadyToSendData_s, this, true));
2007 }
2008
2009 void WebRtcSession::OnSctpTransportReadyToSendData_s(bool ready) {
2010 RTC_DCHECK(signaling_thread_->IsCurrent());
2011 sctp_ready_to_send_data_ = ready;
2012 SignalSctpReadyToSendData(ready);
2013 }
2014
2015 void WebRtcSession::OnSctpTransportDataReceived_n(
2016 const cricket::ReceiveDataParams& params,
2017 const rtc::CopyOnWriteBuffer& payload) {
2018 RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
2019 RTC_DCHECK(network_thread_->IsCurrent());
2020 sctp_invoker_->AsyncInvoke<void>(
2021 RTC_FROM_HERE, signaling_thread_,
2022 rtc::Bind(&WebRtcSession::OnSctpTransportDataReceived_s, this, params,
2023 payload));
2024 }
2025
2026 void WebRtcSession::OnSctpTransportDataReceived_s(
2027 const cricket::ReceiveDataParams& params,
2028 const rtc::CopyOnWriteBuffer& payload) {
2029 RTC_DCHECK(signaling_thread_->IsCurrent());
2030 if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
2031 // Received OPEN message; parse and signal that a new data channel should
2032 // be created.
2033 std::string label;
2034 InternalDataChannelInit config;
2035 config.id = params.ssrc;
2036 if (!ParseDataChannelOpenMessage(payload, &label, &config)) {
2037 LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
2038 << params.ssrc;
2039 return;
2040 }
2041 config.open_handshake_role = InternalDataChannelInit::kAcker;
2042 SignalDataChannelOpenMessage(label, config);
2043 } else {
2044 // Otherwise just forward the signal.
2045 SignalSctpDataReceived(params, payload);
2046 }
2047 }
2048
2049 void WebRtcSession::OnSctpStreamClosedRemotely_n(int sid) {
2050 RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
2051 RTC_DCHECK(network_thread_->IsCurrent());
2052 sctp_invoker_->AsyncInvoke<void>(
2053 RTC_FROM_HERE, signaling_thread_,
2054 rtc::Bind(&sigslot::signal1<int>::operator(),
2055 &SignalSctpStreamClosedRemotely, sid));
2056 }
2057
2058 // Returns false if bundle is enabled and rtcp_mux is disabled.
2059 bool WebRtcSession::ValidateBundleSettings(const SessionDescription* desc) {
2060 bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
2061 if (!bundle_enabled)
2062 return true;
2063
2064 const cricket::ContentGroup* bundle_group =
2065 desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
2066 RTC_DCHECK(bundle_group != NULL);
2067
2068 const cricket::ContentInfos& contents = desc->contents();
2069 for (cricket::ContentInfos::const_iterator citer = contents.begin();
2070 citer != contents.end(); ++citer) {
2071 const cricket::ContentInfo* content = (&*citer);
2072 RTC_DCHECK(content != NULL);
2073 if (bundle_group->HasContentName(content->name) &&
2074 !content->rejected && content->type == cricket::NS_JINGLE_RTP) {
2075 if (!HasRtcpMuxEnabled(content))
2076 return false;
2077 }
2078 }
2079 // RTCP-MUX is enabled in all the contents.
2080 return true;
2081 }
2082
2083 bool WebRtcSession::HasRtcpMuxEnabled(
2084 const cricket::ContentInfo* content) {
2085 const cricket::MediaContentDescription* description =
2086 static_cast<cricket::MediaContentDescription*>(content->description);
2087 return description->rtcp_mux();
2088 }
2089
2090 bool WebRtcSession::ValidateSessionDescription(
2091 const SessionDescriptionInterface* sdesc,
2092 cricket::ContentSource source, std::string* err_desc) {
2093 std::string type;
2094 if (error() != ERROR_NONE) {
2095 return BadSdp(source, type, GetSessionErrorMsg(), err_desc);
2096 }
2097
2098 if (!sdesc || !sdesc->description()) {
2099 return BadSdp(source, type, kInvalidSdp, err_desc);
2100 }
2101
2102 type = sdesc->type();
2103 Action action = GetAction(sdesc->type());
2104 if (source == cricket::CS_LOCAL) {
2105 if (!ExpectSetLocalDescription(action))
2106 return BadLocalSdp(type, BadStateErrMsg(state()), err_desc);
2107 } else {
2108 if (!ExpectSetRemoteDescription(action))
2109 return BadRemoteSdp(type, BadStateErrMsg(state()), err_desc);
2110 }
2111
2112 // Verify crypto settings.
2113 std::string crypto_error;
2114 if ((webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
2115 dtls_enabled_) &&
2116 !VerifyCrypto(sdesc->description(), dtls_enabled_, &crypto_error)) {
2117 return BadSdp(source, type, crypto_error, err_desc);
2118 }
2119
2120 // Verify ice-ufrag and ice-pwd.
2121 if (!VerifyIceUfragPwdPresent(sdesc->description())) {
2122 return BadSdp(source, type, kSdpWithoutIceUfragPwd, err_desc);
2123 }
2124
2125 if (!ValidateBundleSettings(sdesc->description())) {
2126 return BadSdp(source, type, kBundleWithoutRtcpMux, err_desc);
2127 }
2128
2129 // TODO(skvlad): When the local rtcp-mux policy is Require, reject any
2130 // m-lines that do not rtcp-mux enabled.
2131
2132 // Verify m-lines in Answer when compared against Offer.
2133 if (action == kAnswer) {
2134 const cricket::SessionDescription* offer_desc =
2135 (source == cricket::CS_LOCAL) ? remote_description()->description()
2136 : local_description()->description();
2137 if (!VerifyMediaDescriptions(sdesc->description(), offer_desc)) {
2138 return BadAnswerSdp(source, kMlineMismatch, err_desc);
2139 }
2140 }
2141
2142 return true;
2143 }
2144
2145 bool WebRtcSession::ExpectSetLocalDescription(Action action) {
2146 return ((action == kOffer && state() == STATE_INIT) ||
2147 // update local offer
2148 (action == kOffer && state() == STATE_SENTOFFER) ||
2149 // update the current ongoing session.
2150 (action == kOffer && state() == STATE_INPROGRESS) ||
2151 // accept remote offer
2152 (action == kAnswer && state() == STATE_RECEIVEDOFFER) ||
2153 (action == kAnswer && state() == STATE_SENTPRANSWER) ||
2154 (action == kPrAnswer && state() == STATE_RECEIVEDOFFER) ||
2155 (action == kPrAnswer && state() == STATE_SENTPRANSWER));
2156 }
2157
2158 bool WebRtcSession::ExpectSetRemoteDescription(Action action) {
2159 return ((action == kOffer && state() == STATE_INIT) ||
2160 // update remote offer
2161 (action == kOffer && state() == STATE_RECEIVEDOFFER) ||
2162 // update the current ongoing session
2163 (action == kOffer && state() == STATE_INPROGRESS) ||
2164 // accept local offer
2165 (action == kAnswer && state() == STATE_SENTOFFER) ||
2166 (action == kAnswer && state() == STATE_RECEIVEDPRANSWER) ||
2167 (action == kPrAnswer && state() == STATE_SENTOFFER) ||
2168 (action == kPrAnswer && state() == STATE_RECEIVEDPRANSWER));
2169 }
2170
2171 std::string WebRtcSession::GetSessionErrorMsg() {
2172 std::ostringstream desc;
2173 desc << kSessionError << GetErrorCodeString(error()) << ". ";
2174 desc << kSessionErrorDesc << error_desc() << ".";
2175 return desc.str();
2176 }
2177
2178 // We need to check the local/remote description for the Transport instead of
2179 // the session, because a new Transport added during renegotiation may have
2180 // them unset while the session has them set from the previous negotiation.
2181 // Not doing so may trigger the auto generation of transport description and
2182 // mess up DTLS identity information, ICE credential, etc.
2183 bool WebRtcSession::ReadyToUseRemoteCandidate(
2184 const IceCandidateInterface* candidate,
2185 const SessionDescriptionInterface* remote_desc,
2186 bool* valid) {
2187 *valid = true;
2188
2189 const SessionDescriptionInterface* current_remote_desc =
2190 remote_desc ? remote_desc : remote_description();
2191
2192 if (!current_remote_desc) {
2193 return false;
2194 }
2195
2196 size_t mediacontent_index =
2197 static_cast<size_t>(candidate->sdp_mline_index());
2198 size_t remote_content_size =
2199 current_remote_desc->description()->contents().size();
2200 if (mediacontent_index >= remote_content_size) {
2201 LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate media index "
2202 << mediacontent_index;
2203
2204 *valid = false;
2205 return false;
2206 }
2207
2208 cricket::ContentInfo content =
2209 current_remote_desc->description()->contents()[mediacontent_index];
2210
2211 const std::string transport_name = GetTransportName(content.name);
2212 if (transport_name.empty()) {
2213 return false;
2214 }
2215 return transport_controller_->ReadyForRemoteCandidates(transport_name);
2216 }
2217
2218 bool WebRtcSession::SrtpRequired() const {
2219 return dtls_enabled_ ||
2220 webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED;
2221 }
2222
2223 void WebRtcSession::OnTransportControllerGatheringState(
2224 cricket::IceGatheringState state) {
2225 RTC_DCHECK(signaling_thread()->IsCurrent());
2226 if (state == cricket::kIceGatheringGathering) {
2227 if (ice_observer_) {
2228 ice_observer_->OnIceGatheringChange(
2229 PeerConnectionInterface::kIceGatheringGathering);
2230 }
2231 } else if (state == cricket::kIceGatheringComplete) {
2232 if (ice_observer_) {
2233 ice_observer_->OnIceGatheringChange(
2234 PeerConnectionInterface::kIceGatheringComplete);
2235 }
2236 }
2237 }
2238
2239 void WebRtcSession::ReportTransportStats() {
2240 // Use a set so we don't report the same stats twice if two channels share
2241 // a transport.
2242 std::set<std::string> transport_names;
2243 if (voice_channel()) {
2244 transport_names.insert(voice_channel()->transport_name());
2245 }
2246 if (video_channel()) {
2247 transport_names.insert(video_channel()->transport_name());
2248 }
2249 if (rtp_data_channel()) {
2250 transport_names.insert(rtp_data_channel()->transport_name());
2251 }
2252 if (sctp_transport_name_) {
2253 transport_names.insert(*sctp_transport_name_);
2254 }
2255 for (const auto& name : transport_names) {
2256 cricket::TransportStats stats;
2257 if (transport_controller_->GetStats(name, &stats)) {
2258 ReportBestConnectionState(stats);
2259 ReportNegotiatedCiphers(stats);
2260 }
2261 }
2262 }
2263 // Walk through the ConnectionInfos to gather best connection usage
2264 // for IPv4 and IPv6.
2265 void WebRtcSession::ReportBestConnectionState(
2266 const cricket::TransportStats& stats) {
2267 RTC_DCHECK(metrics_observer_ != NULL);
2268 for (cricket::TransportChannelStatsList::const_iterator it =
2269 stats.channel_stats.begin();
2270 it != stats.channel_stats.end(); ++it) {
2271 for (cricket::ConnectionInfos::const_iterator it_info =
2272 it->connection_infos.begin();
2273 it_info != it->connection_infos.end(); ++it_info) {
2274 if (!it_info->best_connection) {
2275 continue;
2276 }
2277
2278 PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
2279 const cricket::Candidate& local = it_info->local_candidate;
2280 const cricket::Candidate& remote = it_info->remote_candidate;
2281
2282 // Increment the counter for IceCandidatePairType.
2283 if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
2284 (local.type() == RELAY_PORT_TYPE &&
2285 local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
2286 type = kEnumCounterIceCandidatePairTypeTcp;
2287 } else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
2288 type = kEnumCounterIceCandidatePairTypeUdp;
2289 } else {
2290 RTC_CHECK(0);
2291 }
2292 metrics_observer_->IncrementEnumCounter(
2293 type, GetIceCandidatePairCounter(local, remote),
2294 kIceCandidatePairMax);
2295
2296 // Increment the counter for IP type.
2297 if (local.address().family() == AF_INET) {
2298 metrics_observer_->IncrementEnumCounter(
2299 kEnumCounterAddressFamily, kBestConnections_IPv4,
2300 kPeerConnectionAddressFamilyCounter_Max);
2301
2302 } else if (local.address().family() == AF_INET6) {
2303 metrics_observer_->IncrementEnumCounter(
2304 kEnumCounterAddressFamily, kBestConnections_IPv6,
2305 kPeerConnectionAddressFamilyCounter_Max);
2306 } else {
2307 RTC_CHECK(0);
2308 }
2309
2310 return;
2311 }
2312 }
2313 }
2314
2315 void WebRtcSession::ReportNegotiatedCiphers(
2316 const cricket::TransportStats& stats) {
2317 RTC_DCHECK(metrics_observer_ != NULL);
2318 if (!dtls_enabled_ || stats.channel_stats.empty()) {
2319 return;
2320 }
2321
2322 int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
2323 int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
2324 if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
2325 ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
2326 return;
2327 }
2328
2329 PeerConnectionEnumCounterType srtp_counter_type;
2330 PeerConnectionEnumCounterType ssl_counter_type;
2331 if (stats.transport_name == cricket::CN_AUDIO) {
2332 srtp_counter_type = kEnumCounterAudioSrtpCipher;
2333 ssl_counter_type = kEnumCounterAudioSslCipher;
2334 } else if (stats.transport_name == cricket::CN_VIDEO) {
2335 srtp_counter_type = kEnumCounterVideoSrtpCipher;
2336 ssl_counter_type = kEnumCounterVideoSslCipher;
2337 } else if (stats.transport_name == cricket::CN_DATA) {
2338 srtp_counter_type = kEnumCounterDataSrtpCipher;
2339 ssl_counter_type = kEnumCounterDataSslCipher;
2340 } else {
2341 RTC_NOTREACHED();
2342 return;
2343 }
2344
2345 if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
2346 metrics_observer_->IncrementSparseEnumCounter(srtp_counter_type,
2347 srtp_crypto_suite);
2348 }
2349 if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
2350 metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type,
2351 ssl_cipher_suite);
2352 }
2353 }
2354
2355 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
2356 RTC_DCHECK(worker_thread()->IsCurrent());
2357 media_controller_->call_w()->OnSentPacket(sent_packet);
2358 }
2359
2360 const std::string WebRtcSession::GetTransportName(
2361 const std::string& content_name) {
2362 cricket::BaseChannel* channel = GetChannel(content_name);
2363 if (!channel) {
2364 #ifdef HAVE_QUIC
2365 if (data_channel_type_ == cricket::DCT_QUIC && quic_data_transport_ &&
2366 content_name == quic_data_transport_->transport_name()) {
2367 return quic_data_transport_->transport_name();
2368 }
2369 #endif
2370 if (sctp_transport_) {
2371 RTC_DCHECK(sctp_content_name_);
2372 RTC_DCHECK(sctp_transport_name_);
2373 if (content_name == *sctp_content_name_) {
2374 return *sctp_transport_name_;
2375 }
2376 }
2377 // Return an empty string if failed to retrieve the transport name.
2378 return "";
2379 }
2380 return channel->transport_name();
2381 }
2382
2383 void WebRtcSession::DestroyRtcpTransport_n(const std::string& transport_name) {
2384 RTC_DCHECK(network_thread()->IsCurrent());
2385 transport_controller_->DestroyDtlsTransport_n(
2386 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
2387 }
2388
2389 void WebRtcSession::DestroyVideoChannel() {
2390 SignalVideoChannelDestroyed();
2391 RTC_DCHECK(video_channel_->rtp_dtls_transport());
2392 std::string transport_name;
2393 transport_name = video_channel_->rtp_dtls_transport()->transport_name();
2394 bool need_to_delete_rtcp = (video_channel_->rtcp_dtls_transport() != nullptr);
2395 channel_manager_->DestroyVideoChannel(video_channel_.release());
2396 transport_controller_->DestroyDtlsTransport(
2397 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
2398 if (need_to_delete_rtcp) {
2399 transport_controller_->DestroyDtlsTransport(
2400 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
2401 }
2402 }
2403
2404 void WebRtcSession::DestroyVoiceChannel() {
2405 SignalVoiceChannelDestroyed();
2406 RTC_DCHECK(voice_channel_->rtp_dtls_transport());
2407 std::string transport_name;
2408 transport_name = voice_channel_->rtp_dtls_transport()->transport_name();
2409 bool need_to_delete_rtcp = (voice_channel_->rtcp_dtls_transport() != nullptr);
2410 channel_manager_->DestroyVoiceChannel(voice_channel_.release());
2411 transport_controller_->DestroyDtlsTransport(
2412 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
2413 if (need_to_delete_rtcp) {
2414 transport_controller_->DestroyDtlsTransport(
2415 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
2416 }
2417 }
2418
2419 void WebRtcSession::DestroyDataChannel() {
2420 SignalDataChannelDestroyed();
2421 RTC_DCHECK(rtp_data_channel_->rtp_dtls_transport());
2422 std::string transport_name;
2423 transport_name = rtp_data_channel_->rtp_dtls_transport()->transport_name();
2424 bool need_to_delete_rtcp =
2425 (rtp_data_channel_->rtcp_dtls_transport() != nullptr);
2426 channel_manager_->DestroyRtpDataChannel(rtp_data_channel_.release());
2427 transport_controller_->DestroyDtlsTransport(
2428 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
2429 if (need_to_delete_rtcp) {
2430 transport_controller_->DestroyDtlsTransport(
2431 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
2432 }
2433 }
2434 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/api/webrtcsession.h ('k') | webrtc/api/webrtcsession_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698