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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;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#include "talk/app/webrtc/audiotrack.h"
29#include "talk/app/webrtc/jsepicecandidate.h"
30#include "talk/app/webrtc/jsepsessiondescription.h"
31#include "talk/app/webrtc/mediastreamsignaling.h"
32#include "talk/app/webrtc/streamcollection.h"
33#include "talk/app/webrtc/videotrack.h"
34#include "talk/app/webrtc/test/fakeconstraints.h"
wu@webrtc.org91053e72013-08-10 07:18:04 +000035#include "talk/app/webrtc/test/fakedtlsidentityservice.h"
36#include "talk/app/webrtc/test/fakemediastreamsignaling.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037#include "talk/app/webrtc/webrtcsession.h"
wu@webrtc.org91053e72013-08-10 07:18:04 +000038#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039#include "talk/base/fakenetwork.h"
40#include "talk/base/firewallsocketserver.h"
41#include "talk/base/gunit.h"
42#include "talk/base/logging.h"
43#include "talk/base/network.h"
44#include "talk/base/physicalsocketserver.h"
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000045#include "talk/base/ssladapter.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046#include "talk/base/sslstreamadapter.h"
47#include "talk/base/stringutils.h"
48#include "talk/base/thread.h"
49#include "talk/base/virtualsocketserver.h"
50#include "talk/media/base/fakemediaengine.h"
51#include "talk/media/base/fakevideorenderer.h"
52#include "talk/media/base/mediachannel.h"
53#include "talk/media/devices/fakedevicemanager.h"
54#include "talk/p2p/base/stunserver.h"
55#include "talk/p2p/base/teststunserver.h"
56#include "talk/p2p/client/basicportallocator.h"
57#include "talk/session/media/channelmanager.h"
58#include "talk/session/media/mediasession.h"
59
60#define MAYBE_SKIP_TEST(feature) \
61 if (!(feature())) { \
62 LOG(LS_INFO) << "Feature disabled... skipping"; \
63 return; \
64 }
65
66using cricket::BaseSession;
67using cricket::DF_PLAY;
68using cricket::DF_SEND;
69using cricket::FakeVoiceMediaChannel;
70using cricket::NS_GINGLE_P2P;
71using cricket::NS_JINGLE_ICE_UDP;
72using cricket::TransportInfo;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000073using talk_base::SocketAddress;
74using talk_base::scoped_ptr;
75using webrtc::CreateSessionDescription;
wu@webrtc.org91053e72013-08-10 07:18:04 +000076using webrtc::CreateSessionDescriptionObserver;
77using webrtc::CreateSessionDescriptionRequest;
78using webrtc::DTLSIdentityRequestObserver;
79using webrtc::DTLSIdentityServiceInterface;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080using webrtc::FakeConstraints;
81using webrtc::IceCandidateCollection;
82using webrtc::JsepIceCandidate;
83using webrtc::JsepSessionDescription;
wu@webrtc.org97077a32013-10-25 21:18:33 +000084using webrtc::PeerConnectionFactoryInterface;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085using webrtc::PeerConnectionInterface;
86using webrtc::SessionDescriptionInterface;
87using webrtc::StreamCollection;
wu@webrtc.org91053e72013-08-10 07:18:04 +000088using webrtc::WebRtcSession;
wu@webrtc.org364f2042013-11-20 21:49:41 +000089using webrtc::kBundleWithoutRtcpMux;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000090using webrtc::kCreateChannelFailed;
91using webrtc::kInvalidSdp;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092using webrtc::kMlineMismatch;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000093using webrtc::kPushDownTDFailed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000094using webrtc::kSdpWithoutCrypto;
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000095using webrtc::kSdpWithoutIceUfragPwd;
wu@webrtc.org364f2042013-11-20 21:49:41 +000096using webrtc::kSdpWithoutSdesAndDtlsDisabled;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000097using webrtc::kSessionError;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000098using webrtc::kSessionErrorDesc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099
wu@webrtc.org364f2042013-11-20 21:49:41 +0000100static const int kClientAddrPort = 0;
101static const char kClientAddrHost1[] = "11.11.11.11";
102static const char kClientAddrHost2[] = "22.22.22.22";
103static const char kStunAddrHost[] = "99.99.99.1";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104
105static const char kSessionVersion[] = "1";
106
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107// Media index of candidates belonging to the first media content.
108static const int kMediaContentIndex0 = 0;
109static const char kMediaContentName0[] = "audio";
110
111// Media index of candidates belonging to the second media content.
112static const int kMediaContentIndex1 = 1;
113static const char kMediaContentName1[] = "video";
114
115static const int kIceCandidatesTimeout = 10000;
116
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000117static const char kFakeDtlsFingerprint[] =
118 "BB:CD:72:F7:2F:D0:BA:43:F3:68:B1:0C:23:72:B6:4A:"
119 "0F:DE:34:06:BC:E0:FE:01:BC:73:C8:6D:F4:65:D5:24";
120
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121// Add some extra |newlines| to the |message| after |line|.
122static void InjectAfter(const std::string& line,
123 const std::string& newlines,
124 std::string* message) {
125 const std::string tmp = line + newlines;
126 talk_base::replace_substrs(line.c_str(), line.length(),
127 tmp.c_str(), tmp.length(), message);
128}
129
130class MockIceObserver : public webrtc::IceObserver {
131 public:
132 MockIceObserver()
133 : oncandidatesready_(false),
134 ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
135 ice_gathering_state_(PeerConnectionInterface::kIceGatheringNew) {
136 }
137
138 virtual void OnIceConnectionChange(
139 PeerConnectionInterface::IceConnectionState new_state) {
140 ice_connection_state_ = new_state;
141 }
142 virtual void OnIceGatheringChange(
143 PeerConnectionInterface::IceGatheringState new_state) {
144 // We can never transition back to "new".
145 EXPECT_NE(PeerConnectionInterface::kIceGatheringNew, new_state);
146 ice_gathering_state_ = new_state;
147
148 // oncandidatesready_ really means "ICE gathering is complete".
149 // This if statement ensures that this value remains correct when we
150 // transition from kIceGatheringComplete to kIceGatheringGathering.
151 if (new_state == PeerConnectionInterface::kIceGatheringGathering) {
152 oncandidatesready_ = false;
153 }
154 }
155
156 // Found a new candidate.
157 virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000158 switch (candidate->sdp_mline_index()) {
159 case kMediaContentIndex0:
160 mline_0_candidates_.push_back(candidate->candidate());
161 break;
162 case kMediaContentIndex1:
163 mline_1_candidates_.push_back(candidate->candidate());
164 break;
165 default:
166 ASSERT(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000167 }
wu@webrtc.org364f2042013-11-20 21:49:41 +0000168
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000169 // The ICE gathering state should always be Gathering when a candidate is
170 // received (or possibly Completed in the case of the final candidate).
171 EXPECT_NE(PeerConnectionInterface::kIceGatheringNew, ice_gathering_state_);
172 }
173
174 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
175 virtual void OnIceComplete() {
176 EXPECT_FALSE(oncandidatesready_);
177 oncandidatesready_ = true;
178
179 // OnIceGatheringChange(IceGatheringCompleted) and OnIceComplete() should
180 // be called approximately simultaneously. For ease of testing, this
181 // check additionally requires that they be called in the above order.
182 EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
183 ice_gathering_state_);
184 }
185
186 bool oncandidatesready_;
187 std::vector<cricket::Candidate> mline_0_candidates_;
188 std::vector<cricket::Candidate> mline_1_candidates_;
189 PeerConnectionInterface::IceConnectionState ice_connection_state_;
190 PeerConnectionInterface::IceGatheringState ice_gathering_state_;
191};
192
193class WebRtcSessionForTest : public webrtc::WebRtcSession {
194 public:
195 WebRtcSessionForTest(cricket::ChannelManager* cmgr,
196 talk_base::Thread* signaling_thread,
197 talk_base::Thread* worker_thread,
198 cricket::PortAllocator* port_allocator,
199 webrtc::IceObserver* ice_observer,
200 webrtc::MediaStreamSignaling* mediastream_signaling)
201 : WebRtcSession(cmgr, signaling_thread, worker_thread, port_allocator,
202 mediastream_signaling) {
203 RegisterIceObserver(ice_observer);
204 }
205 virtual ~WebRtcSessionForTest() {}
206
207 using cricket::BaseSession::GetTransportProxy;
208 using webrtc::WebRtcSession::SetAudioPlayout;
209 using webrtc::WebRtcSession::SetAudioSend;
210 using webrtc::WebRtcSession::SetCaptureDevice;
211 using webrtc::WebRtcSession::SetVideoPlayout;
212 using webrtc::WebRtcSession::SetVideoSend;
213};
214
wu@webrtc.org91053e72013-08-10 07:18:04 +0000215class WebRtcSessionCreateSDPObserverForTest
216 : public talk_base::RefCountedObject<CreateSessionDescriptionObserver> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000217 public:
wu@webrtc.org91053e72013-08-10 07:18:04 +0000218 enum State {
219 kInit,
220 kFailed,
221 kSucceeded,
222 };
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000223 WebRtcSessionCreateSDPObserverForTest() : state_(kInit) {}
wu@webrtc.org91053e72013-08-10 07:18:04 +0000224
225 // CreateSessionDescriptionObserver implementation.
226 virtual void OnSuccess(SessionDescriptionInterface* desc) {
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000227 description_.reset(desc);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000228 state_ = kSucceeded;
229 }
230 virtual void OnFailure(const std::string& error) {
231 state_ = kFailed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000232 }
233
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000234 SessionDescriptionInterface* description() { return description_.get(); }
235
236 SessionDescriptionInterface* ReleaseDescription() {
237 return description_.release();
238 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000239
wu@webrtc.org91053e72013-08-10 07:18:04 +0000240 State state() const { return state_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000241
wu@webrtc.org91053e72013-08-10 07:18:04 +0000242 protected:
243 ~WebRtcSessionCreateSDPObserverForTest() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000244
245 private:
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000246 talk_base::scoped_ptr<SessionDescriptionInterface> description_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000247 State state_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000248};
249
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000250class FakeAudioRenderer : public cricket::AudioRenderer {
251 public:
xians@webrtc.orgef221512014-02-21 10:31:29 +0000252 FakeAudioRenderer() : channel_id_(-1) {}
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000253
254 virtual void AddChannel(int channel_id) OVERRIDE {
255 ASSERT(channel_id_ == -1);
256 channel_id_ = channel_id;
257 }
258 virtual void RemoveChannel(int channel_id) OVERRIDE {
259 ASSERT(channel_id == channel_id_);
260 channel_id_ = -1;
261 }
262
263 int channel_id() const { return channel_id_; }
264 private:
265 int channel_id_;
266};
267
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000268class WebRtcSessionTest : public testing::Test {
269 protected:
270 // TODO Investigate why ChannelManager crashes, if it's created
271 // after stun_server.
272 WebRtcSessionTest()
273 : media_engine_(new cricket::FakeMediaEngine()),
274 data_engine_(new cricket::FakeDataEngine()),
275 device_manager_(new cricket::FakeDeviceManager()),
276 channel_manager_(new cricket::ChannelManager(
277 media_engine_, data_engine_, device_manager_,
278 new cricket::CaptureManager(), talk_base::Thread::Current())),
279 tdesc_factory_(new cricket::TransportDescriptionFactory()),
280 desc_factory_(new cricket::MediaSessionDescriptionFactory(
281 channel_manager_.get(), tdesc_factory_.get())),
282 pss_(new talk_base::PhysicalSocketServer),
283 vss_(new talk_base::VirtualSocketServer(pss_.get())),
284 fss_(new talk_base::FirewallSocketServer(vss_.get())),
285 ss_scope_(fss_.get()),
wu@webrtc.org364f2042013-11-20 21:49:41 +0000286 stun_socket_addr_(talk_base::SocketAddress(kStunAddrHost,
287 cricket::STUN_SERVER_PORT)),
288 stun_server_(talk_base::Thread::Current(), stun_socket_addr_),
289 allocator_(&network_manager_, stun_socket_addr_,
wu@webrtc.org967bfff2013-09-19 05:49:50 +0000290 SocketAddress(), SocketAddress(), SocketAddress()),
291 mediastream_signaling_(channel_manager_.get()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000292 tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);
293 allocator_.set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
294 cricket::PORTALLOCATOR_DISABLE_RELAY |
295 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
296 EXPECT_TRUE(channel_manager_->Init());
297 desc_factory_->set_add_legacy_streams(false);
298 }
299
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000300 static void SetUpTestCase() {
301 talk_base::InitializeSSL();
302 }
303
304 static void TearDownTestCase() {
305 talk_base::CleanupSSL();
306 }
307
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000308 void AddInterface(const SocketAddress& addr) {
309 network_manager_.AddInterface(addr);
310 }
311
wu@webrtc.org91053e72013-08-10 07:18:04 +0000312 void Init(DTLSIdentityServiceInterface* identity_service) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000313 ASSERT_TRUE(session_.get() == NULL);
314 session_.reset(new WebRtcSessionForTest(
315 channel_manager_.get(), talk_base::Thread::Current(),
316 talk_base::Thread::Current(), &allocator_,
317 &observer_,
318 &mediastream_signaling_));
319
320 EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
321 observer_.ice_connection_state_);
322 EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
323 observer_.ice_gathering_state_);
324
wu@webrtc.org97077a32013-10-25 21:18:33 +0000325 EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
326 identity_service));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000327 }
328
329 void InitWithDtmfCodec() {
330 // Add kTelephoneEventCodec for dtmf test.
wu@webrtc.org364f2042013-11-20 21:49:41 +0000331 const cricket::AudioCodec kTelephoneEventCodec(
332 106, "telephone-event", 8000, 0, 1, 0);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000333 std::vector<cricket::AudioCodec> codecs;
334 codecs.push_back(kTelephoneEventCodec);
335 media_engine_->SetAudioCodecs(codecs);
336 desc_factory_->set_audio_codecs(codecs);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000337 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000338 }
339
henrike@webrtc.org7666db72013-08-22 14:45:42 +0000340 void InitWithDtls(bool identity_request_should_fail = false) {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000341 FakeIdentityService* identity_service = new FakeIdentityService();
342 identity_service->set_should_fail(identity_request_should_fail);
343 Init(identity_service);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000344 }
345
346 // Creates a local offer and applies it. Starts ice.
347 // Call mediastream_signaling_.UseOptionsWithStreamX() before this function
348 // to decide which streams to create.
349 void InitiateCall() {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000350 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000351 SetLocalDescriptionWithoutError(offer);
352 EXPECT_TRUE_WAIT(PeerConnectionInterface::kIceGatheringNew !=
353 observer_.ice_gathering_state_,
354 kIceCandidatesTimeout);
355 }
356
wu@webrtc.org91053e72013-08-10 07:18:04 +0000357 SessionDescriptionInterface* CreateOffer(
358 const webrtc::MediaConstraintsInterface* constraints) {
359 talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
360 observer = new WebRtcSessionCreateSDPObserverForTest();
361 session_->CreateOffer(observer, constraints);
362 EXPECT_TRUE_WAIT(
363 observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000364 2000);
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000365 return observer->ReleaseDescription();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000366 }
367
368 SessionDescriptionInterface* CreateAnswer(
369 const webrtc::MediaConstraintsInterface* constraints) {
370 talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observer
371 = new WebRtcSessionCreateSDPObserverForTest();
372 session_->CreateAnswer(observer, constraints);
373 EXPECT_TRUE_WAIT(
374 observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000375 2000);
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000376 return observer->ReleaseDescription();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000377 }
378
wu@webrtc.org364f2042013-11-20 21:49:41 +0000379 bool ChannelsExist() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000380 return (session_->voice_channel() != NULL &&
381 session_->video_channel() != NULL);
382 }
383
wu@webrtc.org364f2042013-11-20 21:49:41 +0000384 void CheckTransportChannels() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000385 EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 1) != NULL);
386 EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 2) != NULL);
387 EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 1) != NULL);
388 EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 2) != NULL);
389 }
390
391 void VerifyCryptoParams(const cricket::SessionDescription* sdp) {
392 ASSERT_TRUE(session_.get() != NULL);
393 const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
394 ASSERT_TRUE(content != NULL);
395 const cricket::AudioContentDescription* audio_content =
396 static_cast<const cricket::AudioContentDescription*>(
397 content->description);
398 ASSERT_TRUE(audio_content != NULL);
399 ASSERT_EQ(1U, audio_content->cryptos().size());
400 ASSERT_EQ(47U, audio_content->cryptos()[0].key_params.size());
401 ASSERT_EQ("AES_CM_128_HMAC_SHA1_80",
402 audio_content->cryptos()[0].cipher_suite);
403 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
404 audio_content->protocol());
405
406 content = cricket::GetFirstVideoContent(sdp);
407 ASSERT_TRUE(content != NULL);
408 const cricket::VideoContentDescription* video_content =
409 static_cast<const cricket::VideoContentDescription*>(
410 content->description);
411 ASSERT_TRUE(video_content != NULL);
412 ASSERT_EQ(1U, video_content->cryptos().size());
413 ASSERT_EQ("AES_CM_128_HMAC_SHA1_80",
414 video_content->cryptos()[0].cipher_suite);
415 ASSERT_EQ(47U, video_content->cryptos()[0].key_params.size());
416 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
417 video_content->protocol());
418 }
419
420 void VerifyNoCryptoParams(const cricket::SessionDescription* sdp, bool dtls) {
421 const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
422 ASSERT_TRUE(content != NULL);
423 const cricket::AudioContentDescription* audio_content =
424 static_cast<const cricket::AudioContentDescription*>(
425 content->description);
426 ASSERT_TRUE(audio_content != NULL);
427 ASSERT_EQ(0U, audio_content->cryptos().size());
428
429 content = cricket::GetFirstVideoContent(sdp);
430 ASSERT_TRUE(content != NULL);
431 const cricket::VideoContentDescription* video_content =
432 static_cast<const cricket::VideoContentDescription*>(
433 content->description);
434 ASSERT_TRUE(video_content != NULL);
435 ASSERT_EQ(0U, video_content->cryptos().size());
436
437 if (dtls) {
438 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
439 audio_content->protocol());
440 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
441 video_content->protocol());
442 } else {
443 EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf),
444 audio_content->protocol());
445 EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf),
446 video_content->protocol());
447 }
448 }
449
450 // Set the internal fake description factories to do DTLS-SRTP.
451 void SetFactoryDtlsSrtp() {
452 desc_factory_->set_secure(cricket::SEC_ENABLED);
453 std::string identity_name = "WebRTC" +
454 talk_base::ToString(talk_base::CreateRandomId());
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000455 identity_.reset(talk_base::SSLIdentity::Generate(identity_name));
456 tdesc_factory_->set_identity(identity_.get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000457 tdesc_factory_->set_secure(cricket::SEC_REQUIRED);
458 }
459
460 void VerifyFingerprintStatus(const cricket::SessionDescription* sdp,
461 bool expected) {
462 const TransportInfo* audio = sdp->GetTransportInfoByName("audio");
463 ASSERT_TRUE(audio != NULL);
464 ASSERT_EQ(expected, audio->description.identity_fingerprint.get() != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000465 const TransportInfo* video = sdp->GetTransportInfoByName("video");
466 ASSERT_TRUE(video != NULL);
467 ASSERT_EQ(expected, video->description.identity_fingerprint.get() != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000468 }
469
470 void VerifyAnswerFromNonCryptoOffer() {
471 // Create a SDP without Crypto.
472 cricket::MediaSessionOptions options;
473 options.has_video = true;
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000474 JsepSessionDescription* offer(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000475 CreateRemoteOffer(options, cricket::SEC_DISABLED));
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000476 ASSERT_TRUE(offer != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000477 VerifyNoCryptoParams(offer->description(), false);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000478 SetRemoteDescriptionOfferExpectError(
479 "Called with a SDP without crypto enabled", offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000480 const webrtc::SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000481 // Answer should be NULL as no crypto params in offer.
482 ASSERT_TRUE(answer == NULL);
483 }
484
485 void VerifyAnswerFromCryptoOffer() {
486 cricket::MediaSessionOptions options;
487 options.has_video = true;
488 options.bundle_enabled = true;
489 scoped_ptr<JsepSessionDescription> offer(
490 CreateRemoteOffer(options, cricket::SEC_REQUIRED));
491 ASSERT_TRUE(offer.get() != NULL);
492 VerifyCryptoParams(offer->description());
493 SetRemoteDescriptionWithoutError(offer.release());
wu@webrtc.org91053e72013-08-10 07:18:04 +0000494 scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000495 ASSERT_TRUE(answer.get() != NULL);
496 VerifyCryptoParams(answer->description());
497 }
498
499 void CompareIceUfragAndPassword(const cricket::SessionDescription* desc1,
500 const cricket::SessionDescription* desc2,
501 bool expect_equal) {
502 if (desc1->contents().size() != desc2->contents().size()) {
503 EXPECT_FALSE(expect_equal);
504 return;
505 }
506
507 const cricket::ContentInfos& contents = desc1->contents();
508 cricket::ContentInfos::const_iterator it = contents.begin();
509
510 for (; it != contents.end(); ++it) {
511 const cricket::TransportDescription* transport_desc1 =
512 desc1->GetTransportDescriptionByName(it->name);
513 const cricket::TransportDescription* transport_desc2 =
514 desc2->GetTransportDescriptionByName(it->name);
515 if (!transport_desc1 || !transport_desc2) {
516 EXPECT_FALSE(expect_equal);
517 return;
518 }
519 if (transport_desc1->ice_pwd != transport_desc2->ice_pwd ||
520 transport_desc1->ice_ufrag != transport_desc2->ice_ufrag) {
521 EXPECT_FALSE(expect_equal);
522 return;
523 }
524 }
525 EXPECT_TRUE(expect_equal);
526 }
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000527
528 void RemoveIceUfragPwdLines(const SessionDescriptionInterface* current_desc,
529 std::string *sdp) {
530 const cricket::SessionDescription* desc = current_desc->description();
531 EXPECT_TRUE(current_desc->ToString(sdp));
532
533 const cricket::ContentInfos& contents = desc->contents();
534 cricket::ContentInfos::const_iterator it = contents.begin();
535 // Replace ufrag and pwd lines with empty strings.
536 for (; it != contents.end(); ++it) {
537 const cricket::TransportDescription* transport_desc =
538 desc->GetTransportDescriptionByName(it->name);
539 std::string ufrag_line = "a=ice-ufrag:" + transport_desc->ice_ufrag
540 + "\r\n";
541 std::string pwd_line = "a=ice-pwd:" + transport_desc->ice_pwd
542 + "\r\n";
543 talk_base::replace_substrs(ufrag_line.c_str(), ufrag_line.length(),
544 "", 0,
545 sdp);
546 talk_base::replace_substrs(pwd_line.c_str(), pwd_line.length(),
547 "", 0,
548 sdp);
549 }
550 }
551
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000552 // Creates a remote offer and and applies it as a remote description,
553 // creates a local answer and applies is as a local description.
554 // Call mediastream_signaling_.UseOptionsWithStreamX() before this function
555 // to decide which local and remote streams to create.
556 void CreateAndSetRemoteOfferAndLocalAnswer() {
557 SessionDescriptionInterface* offer = CreateRemoteOffer();
558 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000559 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000560 SetLocalDescriptionWithoutError(answer);
561 }
562 void SetLocalDescriptionWithoutError(SessionDescriptionInterface* desc) {
563 EXPECT_TRUE(session_->SetLocalDescription(desc, NULL));
564 }
565 void SetLocalDescriptionExpectState(SessionDescriptionInterface* desc,
566 BaseSession::State expected_state) {
567 SetLocalDescriptionWithoutError(desc);
568 EXPECT_EQ(expected_state, session_->state());
569 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000570 void SetLocalDescriptionExpectError(const std::string& action,
571 const std::string& expected_error,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000572 SessionDescriptionInterface* desc) {
573 std::string error;
574 EXPECT_FALSE(session_->SetLocalDescription(desc, &error));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000575 std::string sdp_type = "local ";
576 sdp_type.append(action);
577 EXPECT_NE(std::string::npos, error.find(sdp_type));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000578 EXPECT_NE(std::string::npos, error.find(expected_error));
579 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000580 void SetLocalDescriptionOfferExpectError(const std::string& expected_error,
581 SessionDescriptionInterface* desc) {
582 SetLocalDescriptionExpectError(SessionDescriptionInterface::kOffer,
583 expected_error, desc);
584 }
585 void SetLocalDescriptionAnswerExpectError(const std::string& expected_error,
586 SessionDescriptionInterface* desc) {
587 SetLocalDescriptionExpectError(SessionDescriptionInterface::kAnswer,
588 expected_error, desc);
589 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000590 void SetRemoteDescriptionWithoutError(SessionDescriptionInterface* desc) {
591 EXPECT_TRUE(session_->SetRemoteDescription(desc, NULL));
592 }
593 void SetRemoteDescriptionExpectState(SessionDescriptionInterface* desc,
594 BaseSession::State expected_state) {
595 SetRemoteDescriptionWithoutError(desc);
596 EXPECT_EQ(expected_state, session_->state());
597 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000598 void SetRemoteDescriptionExpectError(const std::string& action,
599 const std::string& expected_error,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000600 SessionDescriptionInterface* desc) {
601 std::string error;
602 EXPECT_FALSE(session_->SetRemoteDescription(desc, &error));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000603 std::string sdp_type = "remote ";
604 sdp_type.append(action);
605 EXPECT_NE(std::string::npos, error.find(sdp_type));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000606 EXPECT_NE(std::string::npos, error.find(expected_error));
607 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000608 void SetRemoteDescriptionOfferExpectError(
609 const std::string& expected_error, SessionDescriptionInterface* desc) {
610 SetRemoteDescriptionExpectError(SessionDescriptionInterface::kOffer,
611 expected_error, desc);
612 }
613 void SetRemoteDescriptionPranswerExpectError(
614 const std::string& expected_error, SessionDescriptionInterface* desc) {
615 SetRemoteDescriptionExpectError(SessionDescriptionInterface::kPrAnswer,
616 expected_error, desc);
617 }
618 void SetRemoteDescriptionAnswerExpectError(
619 const std::string& expected_error, SessionDescriptionInterface* desc) {
620 SetRemoteDescriptionExpectError(SessionDescriptionInterface::kAnswer,
621 expected_error, desc);
622 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000623
624 void CreateCryptoOfferAndNonCryptoAnswer(SessionDescriptionInterface** offer,
625 SessionDescriptionInterface** nocrypto_answer) {
626 // Create a SDP without Crypto.
627 cricket::MediaSessionOptions options;
628 options.has_video = true;
629 options.bundle_enabled = true;
630 *offer = CreateRemoteOffer(options, cricket::SEC_ENABLED);
631 ASSERT_TRUE(*offer != NULL);
632 VerifyCryptoParams((*offer)->description());
633
634 *nocrypto_answer = CreateRemoteAnswer(*offer, options,
635 cricket::SEC_DISABLED);
636 EXPECT_TRUE(*nocrypto_answer != NULL);
637 }
638
639 JsepSessionDescription* CreateRemoteOfferWithVersion(
640 cricket::MediaSessionOptions options,
641 cricket::SecurePolicy secure_policy,
642 const std::string& session_version,
643 const SessionDescriptionInterface* current_desc) {
644 std::string session_id = talk_base::ToString(talk_base::CreateRandomId64());
645 const cricket::SessionDescription* cricket_desc = NULL;
646 if (current_desc) {
647 cricket_desc = current_desc->description();
648 session_id = current_desc->session_id();
649 }
650
651 desc_factory_->set_secure(secure_policy);
652 JsepSessionDescription* offer(
653 new JsepSessionDescription(JsepSessionDescription::kOffer));
654 if (!offer->Initialize(desc_factory_->CreateOffer(options, cricket_desc),
655 session_id, session_version)) {
656 delete offer;
657 offer = NULL;
658 }
659 return offer;
660 }
661 JsepSessionDescription* CreateRemoteOffer(
662 cricket::MediaSessionOptions options) {
663 return CreateRemoteOfferWithVersion(options, cricket::SEC_ENABLED,
664 kSessionVersion, NULL);
665 }
666 JsepSessionDescription* CreateRemoteOffer(
667 cricket::MediaSessionOptions options, cricket::SecurePolicy policy) {
668 return CreateRemoteOfferWithVersion(options, policy, kSessionVersion, NULL);
669 }
670 JsepSessionDescription* CreateRemoteOffer(
671 cricket::MediaSessionOptions options,
672 const SessionDescriptionInterface* current_desc) {
673 return CreateRemoteOfferWithVersion(options, cricket::SEC_ENABLED,
674 kSessionVersion, current_desc);
675 }
676
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000677 JsepSessionDescription* CreateRemoteOfferWithSctpPort(
678 const char* sctp_stream_name, int new_port,
679 cricket::MediaSessionOptions options) {
680 options.data_channel_type = cricket::DCT_SCTP;
681 options.AddStream(cricket::MEDIA_TYPE_DATA, "datachannel",
682 sctp_stream_name);
683 return ChangeSDPSctpPort(new_port, CreateRemoteOffer(options));
684 }
685
686 // Takes ownership of offer_basis (and deletes it).
687 JsepSessionDescription* ChangeSDPSctpPort(
688 int new_port, webrtc::SessionDescriptionInterface *offer_basis) {
689 // Stringify the input SDP, swap the 5000 for 'new_port' and create a new
690 // SessionDescription from the mutated string.
691 const char* default_port_str = "5000";
692 char new_port_str[16];
693 talk_base::sprintfn(new_port_str, sizeof(new_port_str), "%d", new_port);
694 std::string offer_str;
695 offer_basis->ToString(&offer_str);
696 talk_base::replace_substrs(default_port_str, strlen(default_port_str),
697 new_port_str, strlen(new_port_str),
698 &offer_str);
699 JsepSessionDescription* offer = new JsepSessionDescription(
700 offer_basis->type());
701 delete offer_basis;
702 offer->Initialize(offer_str, NULL);
703 return offer;
704 }
705
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000706 // Create a remote offer. Call mediastream_signaling_.UseOptionsWithStreamX()
707 // before this function to decide which streams to create.
708 JsepSessionDescription* CreateRemoteOffer() {
709 cricket::MediaSessionOptions options;
710 mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
711 return CreateRemoteOffer(options, session_->remote_description());
712 }
713
714 JsepSessionDescription* CreateRemoteAnswer(
715 const SessionDescriptionInterface* offer,
716 cricket::MediaSessionOptions options,
717 cricket::SecurePolicy policy) {
718 desc_factory_->set_secure(policy);
719 const std::string session_id =
720 talk_base::ToString(talk_base::CreateRandomId64());
721 JsepSessionDescription* answer(
722 new JsepSessionDescription(JsepSessionDescription::kAnswer));
723 if (!answer->Initialize(desc_factory_->CreateAnswer(offer->description(),
724 options, NULL),
725 session_id, kSessionVersion)) {
726 delete answer;
727 answer = NULL;
728 }
729 return answer;
730 }
731
732 JsepSessionDescription* CreateRemoteAnswer(
733 const SessionDescriptionInterface* offer,
734 cricket::MediaSessionOptions options) {
735 return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
736 }
737
738 // Creates an answer session description with streams based on
739 // |mediastream_signaling_|. Call
740 // mediastream_signaling_.UseOptionsWithStreamX() before this function
741 // to decide which streams to create.
742 JsepSessionDescription* CreateRemoteAnswer(
743 const SessionDescriptionInterface* offer) {
744 cricket::MediaSessionOptions options;
745 mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
746 return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
747 }
748
749 void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000750 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +0000751 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000752 mediastream_signaling_.SendAudioVideoStream1();
753 FakeConstraints constraints;
754 constraints.SetMandatoryUseRtpMux(bundle);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000755 SessionDescriptionInterface* offer = CreateOffer(&constraints);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000756 // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
757 // and answer.
758 SetLocalDescriptionWithoutError(offer);
759
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000760 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
761 CreateRemoteAnswer(session_->local_description()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000762 std::string sdp;
763 EXPECT_TRUE(answer->ToString(&sdp));
764
765 size_t expected_candidate_num = 2;
766 if (!rtcp_mux) {
767 // If rtcp_mux is enabled we should expect 4 candidates - host and srflex
768 // for rtp and rtcp.
769 expected_candidate_num = 4;
770 // Disable rtcp-mux from the answer
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000771 const std::string kRtcpMux = "a=rtcp-mux";
772 const std::string kXRtcpMux = "a=xrtcp-mux";
773 talk_base::replace_substrs(kRtcpMux.c_str(), kRtcpMux.length(),
774 kXRtcpMux.c_str(), kXRtcpMux.length(),
775 &sdp);
776 }
777
778 SessionDescriptionInterface* new_answer = CreateSessionDescription(
779 JsepSessionDescription::kAnswer, sdp, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000780
781 // SetRemoteDescription to enable rtcp mux.
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000782 SetRemoteDescriptionWithoutError(new_answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000783 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
784 EXPECT_EQ(expected_candidate_num, observer_.mline_0_candidates_.size());
785 EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
786 for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
787 cricket::Candidate c0 = observer_.mline_0_candidates_[i];
788 cricket::Candidate c1 = observer_.mline_1_candidates_[i];
789 if (bundle) {
790 EXPECT_TRUE(c0.IsEquivalent(c1));
791 } else {
792 EXPECT_FALSE(c0.IsEquivalent(c1));
793 }
794 }
795 }
796 // Tests that we can only send DTMF when the dtmf codec is supported.
797 void TestCanInsertDtmf(bool can) {
798 if (can) {
799 InitWithDtmfCodec();
800 } else {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000801 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000802 }
803 mediastream_signaling_.SendAudioVideoStream1();
804 CreateAndSetRemoteOfferAndLocalAnswer();
805 EXPECT_FALSE(session_->CanInsertDtmf(""));
806 EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
807 }
808
809 // The method sets up a call from the session to itself, in a loopback
810 // arrangement. It also uses a firewall rule to create a temporary
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000811 // disconnection, and then a permanent disconnection.
812 // This code is placed in a method so that it can be invoked
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000813 // by multiple tests with different allocators (e.g. with and without BUNDLE).
814 // While running the call, this method also checks if the session goes through
815 // the correct sequence of ICE states when a connection is established,
816 // broken, and re-established.
817 // The Connection state should go:
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000818 // New -> Checking -> (Connected) -> Completed -> Disconnected -> Completed
819 // -> Failed.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000820 // The Gathering state should go: New -> Gathering -> Completed.
821 void TestLoopbackCall() {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000822 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +0000823 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000824 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000825 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000826
827 EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
828 observer_.ice_gathering_state_);
829 SetLocalDescriptionWithoutError(offer);
830 EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
831 observer_.ice_connection_state_);
832 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringGathering,
833 observer_.ice_gathering_state_,
834 kIceCandidatesTimeout);
835 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
836 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
837 observer_.ice_gathering_state_,
838 kIceCandidatesTimeout);
839
840 std::string sdp;
841 offer->ToString(&sdp);
842 SessionDescriptionInterface* desc =
843 webrtc::CreateSessionDescription(JsepSessionDescription::kAnswer, sdp);
844 ASSERT_TRUE(desc != NULL);
845 SetRemoteDescriptionWithoutError(desc);
846
847 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
848 observer_.ice_connection_state_,
849 kIceCandidatesTimeout);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000850 // The ice connection state is "Connected" too briefly to catch in a test.
851 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000852 observer_.ice_connection_state_,
853 kIceCandidatesTimeout);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000854
855 // Adding firewall rule to block ping requests, which should cause
856 // transport channel failure.
wu@webrtc.org364f2042013-11-20 21:49:41 +0000857 fss_->AddRule(false,
858 talk_base::FP_ANY,
859 talk_base::FD_ANY,
860 talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000861 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
862 observer_.ice_connection_state_,
863 kIceCandidatesTimeout);
864
865 // Clearing the rules, session should move back to completed state.
866 fss_->ClearRules();
867 // Session is automatically calling OnSignalingReady after creation of
868 // new portallocator session which will allocate new set of candidates.
869
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000870 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000871 observer_.ice_connection_state_,
872 kIceCandidatesTimeout);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000873
874 // Now we block ping requests and wait until the ICE connection transitions
875 // to the Failed state. This will take at least 30 seconds because it must
876 // wait for the Port to timeout.
877 int port_timeout = 30000;
878 fss_->AddRule(false,
879 talk_base::FP_ANY,
880 talk_base::FD_ANY,
881 talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
882 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
883 observer_.ice_connection_state_,
884 kIceCandidatesTimeout + port_timeout);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000885 }
886
887 void VerifyTransportType(const std::string& content_name,
888 cricket::TransportProtocol protocol) {
889 const cricket::Transport* transport = session_->GetTransport(content_name);
890 ASSERT_TRUE(transport != NULL);
891 EXPECT_EQ(protocol, transport->protocol());
892 }
893
894 // Adds CN codecs to FakeMediaEngine and MediaDescriptionFactory.
895 void AddCNCodecs() {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000896 const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1, 0);
897 const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1, 0);
898
899 // Add kCNCodec for dtmf test.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000900 std::vector<cricket::AudioCodec> codecs = media_engine_->audio_codecs();;
901 codecs.push_back(kCNCodec1);
902 codecs.push_back(kCNCodec2);
903 media_engine_->SetAudioCodecs(codecs);
904 desc_factory_->set_audio_codecs(codecs);
905 }
906
907 bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
908 const cricket::ContentDescription* description = content->description;
909 ASSERT(description != NULL);
910 const cricket::AudioContentDescription* audio_content_desc =
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000911 static_cast<const cricket::AudioContentDescription*>(description);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000912 ASSERT(audio_content_desc != NULL);
913 for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
914 if (audio_content_desc->codecs()[i].name == "CN")
915 return false;
916 }
917 return true;
918 }
919
920 void SetLocalDescriptionWithDataChannel() {
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000921 webrtc::InternalDataChannelInit dci;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000922 dci.reliable = false;
923 session_->CreateDataChannel("datachannel", &dci);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000924 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000925 SetLocalDescriptionWithoutError(offer);
926 }
927
wu@webrtc.org91053e72013-08-10 07:18:04 +0000928 void VerifyMultipleAsyncCreateDescription(
929 bool success, CreateSessionDescriptionRequest::Type type) {
henrike@webrtc.org7666db72013-08-22 14:45:42 +0000930 InitWithDtls(!success);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000931
932 if (type == CreateSessionDescriptionRequest::kAnswer) {
933 cricket::MediaSessionOptions options;
934 scoped_ptr<JsepSessionDescription> offer(
935 CreateRemoteOffer(options, cricket::SEC_REQUIRED));
936 ASSERT_TRUE(offer.get() != NULL);
937 SetRemoteDescriptionWithoutError(offer.release());
938 }
939
940 const int kNumber = 3;
941 talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
942 observers[kNumber];
943 for (int i = 0; i < kNumber; ++i) {
944 observers[i] = new WebRtcSessionCreateSDPObserverForTest();
945 if (type == CreateSessionDescriptionRequest::kOffer) {
946 session_->CreateOffer(observers[i], NULL);
947 } else {
948 session_->CreateAnswer(observers[i], NULL);
949 }
950 }
951
952 WebRtcSessionCreateSDPObserverForTest::State expected_state =
953 success ? WebRtcSessionCreateSDPObserverForTest::kSucceeded :
954 WebRtcSessionCreateSDPObserverForTest::kFailed;
955
956 for (int i = 0; i < kNumber; ++i) {
957 EXPECT_EQ_WAIT(expected_state, observers[i]->state(), 1000);
958 if (success) {
959 EXPECT_TRUE(observers[i]->description() != NULL);
960 } else {
961 EXPECT_TRUE(observers[i]->description() == NULL);
962 }
963 }
964 }
965
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000966 cricket::FakeMediaEngine* media_engine_;
967 cricket::FakeDataEngine* data_engine_;
968 cricket::FakeDeviceManager* device_manager_;
969 talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
970 talk_base::scoped_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000971 talk_base::scoped_ptr<talk_base::SSLIdentity> identity_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000972 talk_base::scoped_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
973 talk_base::scoped_ptr<talk_base::PhysicalSocketServer> pss_;
974 talk_base::scoped_ptr<talk_base::VirtualSocketServer> vss_;
975 talk_base::scoped_ptr<talk_base::FirewallSocketServer> fss_;
976 talk_base::SocketServerScope ss_scope_;
wu@webrtc.org364f2042013-11-20 21:49:41 +0000977 talk_base::SocketAddress stun_socket_addr_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000978 cricket::TestStunServer stun_server_;
979 talk_base::FakeNetworkManager network_manager_;
980 cricket::BasicPortAllocator allocator_;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000981 PeerConnectionFactoryInterface::Options options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000982 talk_base::scoped_ptr<FakeConstraints> constraints_;
983 FakeMediaStreamSignaling mediastream_signaling_;
984 talk_base::scoped_ptr<WebRtcSessionForTest> session_;
985 MockIceObserver observer_;
986 cricket::FakeVideoMediaChannel* video_channel_;
987 cricket::FakeVoiceMediaChannel* voice_channel_;
988};
989
990TEST_F(WebRtcSessionTest, TestInitialize) {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000991 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000992}
993
994TEST_F(WebRtcSessionTest, TestInitializeWithDtls) {
995 InitWithDtls();
996}
997
wu@webrtc.org91053e72013-08-10 07:18:04 +0000998// Verifies that WebRtcSession uses SEC_REQUIRED by default.
999TEST_F(WebRtcSessionTest, TestDefaultSetSecurePolicy) {
1000 Init(NULL);
wu@webrtc.org364f2042013-11-20 21:49:41 +00001001 EXPECT_EQ(cricket::SEC_REQUIRED, session_->SecurePolicy());
wu@webrtc.org91053e72013-08-10 07:18:04 +00001002}
1003
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001004TEST_F(WebRtcSessionTest, TestSessionCandidates) {
1005 TestSessionCandidatesWithBundleRtcpMux(false, false);
1006}
1007
1008// Below test cases (TestSessionCandidatesWith*) verify the candidates gathered
1009// with rtcp-mux and/or bundle.
1010TEST_F(WebRtcSessionTest, TestSessionCandidatesWithRtcpMux) {
1011 TestSessionCandidatesWithBundleRtcpMux(false, true);
1012}
1013
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001014TEST_F(WebRtcSessionTest, TestSessionCandidatesWithBundleRtcpMux) {
1015 TestSessionCandidatesWithBundleRtcpMux(true, true);
1016}
1017
1018TEST_F(WebRtcSessionTest, TestMultihomeCandidates) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001019 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
1020 AddInterface(talk_base::SocketAddress(kClientAddrHost2, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001021 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001022 mediastream_signaling_.SendAudioVideoStream1();
1023 InitiateCall();
1024 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
1025 EXPECT_EQ(8u, observer_.mline_0_candidates_.size());
1026 EXPECT_EQ(8u, observer_.mline_1_candidates_.size());
1027}
1028
1029TEST_F(WebRtcSessionTest, TestStunError) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001030 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
1031 AddInterface(talk_base::SocketAddress(kClientAddrHost2, kClientAddrPort));
1032 fss_->AddRule(false,
1033 talk_base::FP_UDP,
1034 talk_base::FD_ANY,
1035 talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001036 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001037 mediastream_signaling_.SendAudioVideoStream1();
1038 InitiateCall();
wu@webrtc.org364f2042013-11-20 21:49:41 +00001039 // Since kClientAddrHost1 is blocked, not expecting stun candidates for it.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001040 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
1041 EXPECT_EQ(6u, observer_.mline_0_candidates_.size());
1042 EXPECT_EQ(6u, observer_.mline_1_candidates_.size());
1043}
1044
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001045TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
1046 Init(NULL);
1047 SessionDescriptionInterface* offer = NULL;
1048 // Since |offer| is NULL, there's no way to tell if it's an offer or answer.
1049 std::string unknown_action;
1050 SetLocalDescriptionExpectError(unknown_action, kInvalidSdp, offer);
1051 SetRemoteDescriptionExpectError(unknown_action, kInvalidSdp, offer);
1052}
1053
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001054// Test creating offers and receive answers and make sure the
1055// media engine creates the expected send and receive streams.
1056TEST_F(WebRtcSessionTest, TestCreateOfferReceiveAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001057 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001058 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001059 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001060 const std::string session_id_orig = offer->session_id();
1061 const std::string session_version_orig = offer->session_version();
1062 SetLocalDescriptionWithoutError(offer);
1063
1064 mediastream_signaling_.SendAudioVideoStream2();
1065 SessionDescriptionInterface* answer =
1066 CreateRemoteAnswer(session_->local_description());
1067 SetRemoteDescriptionWithoutError(answer);
1068
1069 video_channel_ = media_engine_->GetVideoChannel(0);
1070 voice_channel_ = media_engine_->GetVoiceChannel(0);
1071
1072 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1073 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
1074
1075 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1076 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
1077
1078 ASSERT_EQ(1u, video_channel_->send_streams().size());
1079 EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
1080 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1081 EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
1082
1083 // Create new offer without send streams.
1084 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001085 offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001086
1087 // Verify the session id is the same and the session version is
1088 // increased.
1089 EXPECT_EQ(session_id_orig, offer->session_id());
1090 EXPECT_LT(talk_base::FromString<uint64>(session_version_orig),
1091 talk_base::FromString<uint64>(offer->session_version()));
1092
1093 SetLocalDescriptionWithoutError(offer);
1094
1095 mediastream_signaling_.SendAudioVideoStream2();
1096 answer = CreateRemoteAnswer(session_->local_description());
1097 SetRemoteDescriptionWithoutError(answer);
1098
1099 EXPECT_EQ(0u, video_channel_->send_streams().size());
1100 EXPECT_EQ(0u, voice_channel_->send_streams().size());
1101
1102 // Make sure the receive streams have not changed.
1103 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1104 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
1105 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1106 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
1107}
1108
1109// Test receiving offers and creating answers and make sure the
1110// media engine creates the expected send and receive streams.
1111TEST_F(WebRtcSessionTest, TestReceiveOfferCreateAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001112 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001113 mediastream_signaling_.SendAudioVideoStream2();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001114 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001115 SetRemoteDescriptionWithoutError(offer);
1116
1117 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001118 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001119 SetLocalDescriptionWithoutError(answer);
1120
1121 const std::string session_id_orig = answer->session_id();
1122 const std::string session_version_orig = answer->session_version();
1123
1124 video_channel_ = media_engine_->GetVideoChannel(0);
1125 voice_channel_ = media_engine_->GetVoiceChannel(0);
1126
1127 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1128 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
1129
1130 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1131 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
1132
1133 ASSERT_EQ(1u, video_channel_->send_streams().size());
1134 EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
1135 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1136 EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
1137
1138 mediastream_signaling_.SendAudioVideoStream1And2();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001139 offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001140 SetRemoteDescriptionWithoutError(offer);
1141
1142 // Answer by turning off all send streams.
1143 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001144 answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001145
1146 // Verify the session id is the same and the session version is
1147 // increased.
1148 EXPECT_EQ(session_id_orig, answer->session_id());
1149 EXPECT_LT(talk_base::FromString<uint64>(session_version_orig),
1150 talk_base::FromString<uint64>(answer->session_version()));
1151 SetLocalDescriptionWithoutError(answer);
1152
1153 ASSERT_EQ(2u, video_channel_->recv_streams().size());
1154 EXPECT_TRUE(kVideoTrack1 == video_channel_->recv_streams()[0].id);
1155 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[1].id);
1156 ASSERT_EQ(2u, voice_channel_->recv_streams().size());
1157 EXPECT_TRUE(kAudioTrack1 == voice_channel_->recv_streams()[0].id);
1158 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[1].id);
1159
1160 // Make sure we have no send streams.
1161 EXPECT_EQ(0u, video_channel_->send_streams().size());
1162 EXPECT_EQ(0u, voice_channel_->send_streams().size());
1163}
1164
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001165TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
1166 Init(NULL);
1167 media_engine_->set_fail_create_channel(true);
1168
1169 SessionDescriptionInterface* offer = CreateOffer(NULL);
1170 ASSERT_TRUE(offer != NULL);
1171 // SetRemoteDescription and SetLocalDescription will take the ownership of
1172 // the offer.
1173 SetRemoteDescriptionOfferExpectError(kCreateChannelFailed, offer);
1174 offer = CreateOffer(NULL);
1175 ASSERT_TRUE(offer != NULL);
1176 SetLocalDescriptionOfferExpectError(kCreateChannelFailed, offer);
1177}
1178
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001179// Test we will return fail when apply an offer that doesn't have
1180// crypto enabled.
1181TEST_F(WebRtcSessionTest, SetNonCryptoOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001182 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001183 cricket::MediaSessionOptions options;
1184 options.has_video = true;
1185 JsepSessionDescription* offer = CreateRemoteOffer(
1186 options, cricket::SEC_DISABLED);
1187 ASSERT_TRUE(offer != NULL);
1188 VerifyNoCryptoParams(offer->description(), false);
1189 // SetRemoteDescription and SetLocalDescription will take the ownership of
1190 // the offer.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001191 SetRemoteDescriptionOfferExpectError(kSdpWithoutCrypto, offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001192 offer = CreateRemoteOffer(options, cricket::SEC_DISABLED);
1193 ASSERT_TRUE(offer != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001194 SetLocalDescriptionOfferExpectError(kSdpWithoutCrypto, offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001195}
1196
1197// Test we will return fail when apply an answer that doesn't have
1198// crypto enabled.
1199TEST_F(WebRtcSessionTest, SetLocalNonCryptoAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001200 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001201 SessionDescriptionInterface* offer = NULL;
1202 SessionDescriptionInterface* answer = NULL;
1203 CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
1204 // SetRemoteDescription and SetLocalDescription will take the ownership of
1205 // the offer.
1206 SetRemoteDescriptionWithoutError(offer);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001207 SetLocalDescriptionAnswerExpectError(kSdpWithoutCrypto, answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001208}
1209
1210// Test we will return fail when apply an answer that doesn't have
1211// crypto enabled.
1212TEST_F(WebRtcSessionTest, SetRemoteNonCryptoAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001213 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001214 SessionDescriptionInterface* offer = NULL;
1215 SessionDescriptionInterface* answer = NULL;
1216 CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
1217 // SetRemoteDescription and SetLocalDescription will take the ownership of
1218 // the offer.
1219 SetLocalDescriptionWithoutError(offer);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001220 SetRemoteDescriptionAnswerExpectError(kSdpWithoutCrypto, answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001221}
1222
1223// Test that we can create and set an offer with a DTLS fingerprint.
1224TEST_F(WebRtcSessionTest, CreateSetDtlsOffer) {
1225 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
1226 InitWithDtls();
1227 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001228 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001229 ASSERT_TRUE(offer != NULL);
1230 VerifyFingerprintStatus(offer->description(), true);
1231 // SetLocalDescription will take the ownership of the offer.
1232 SetLocalDescriptionWithoutError(offer);
1233}
1234
1235// Test that we can process an offer with a DTLS fingerprint
1236// and that we return an answer with a fingerprint.
1237TEST_F(WebRtcSessionTest, ReceiveDtlsOfferCreateAnswer) {
1238 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
1239 InitWithDtls();
1240 SetFactoryDtlsSrtp();
1241 cricket::MediaSessionOptions options;
1242 options.has_video = true;
1243 JsepSessionDescription* offer = CreateRemoteOffer(options);
1244 ASSERT_TRUE(offer != NULL);
1245 VerifyFingerprintStatus(offer->description(), true);
1246
1247 // SetRemoteDescription will take the ownership of the offer.
1248 SetRemoteDescriptionWithoutError(offer);
1249
1250 // Verify that we get a crypto fingerprint in the answer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001251 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001252 ASSERT_TRUE(answer != NULL);
1253 VerifyFingerprintStatus(answer->description(), true);
1254 // Check that we don't have an a=crypto line in the answer.
1255 VerifyNoCryptoParams(answer->description(), true);
1256
1257 // Now set the local description, which should work, even without a=crypto.
1258 SetLocalDescriptionWithoutError(answer);
1259}
1260
1261// Test that even if we support DTLS, if the other side didn't offer a
1262// fingerprint, we don't either.
1263TEST_F(WebRtcSessionTest, ReceiveNoDtlsOfferCreateAnswer) {
1264 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
1265 InitWithDtls();
1266 cricket::MediaSessionOptions options;
1267 options.has_video = true;
1268 JsepSessionDescription* offer = CreateRemoteOffer(
1269 options, cricket::SEC_REQUIRED);
1270 ASSERT_TRUE(offer != NULL);
1271 VerifyFingerprintStatus(offer->description(), false);
1272
1273 // SetRemoteDescription will take the ownership of
1274 // the offer.
1275 SetRemoteDescriptionWithoutError(offer);
1276
1277 // Verify that we don't get a crypto fingerprint in the answer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001278 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001279 ASSERT_TRUE(answer != NULL);
1280 VerifyFingerprintStatus(answer->description(), false);
1281
1282 // Now set the local description.
1283 SetLocalDescriptionWithoutError(answer);
1284}
1285
1286TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001287 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001288 mediastream_signaling_.SendNothing();
1289 // SetLocalDescription take ownership of offer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001290 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001291 SetLocalDescriptionWithoutError(offer);
1292
1293 // SetLocalDescription take ownership of offer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001294 SessionDescriptionInterface* offer2 = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001295 SetLocalDescriptionWithoutError(offer2);
1296}
1297
1298TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001299 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001300 mediastream_signaling_.SendNothing();
1301 // SetLocalDescription take ownership of offer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001302 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001303 SetRemoteDescriptionWithoutError(offer);
1304
wu@webrtc.org91053e72013-08-10 07:18:04 +00001305 SessionDescriptionInterface* offer2 = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001306 SetRemoteDescriptionWithoutError(offer2);
1307}
1308
1309TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001310 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001311 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001312 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001313 SetLocalDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001314 offer = CreateOffer(NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001315 SetRemoteDescriptionOfferExpectError(
1316 "Called in wrong state: STATE_SENTINITIATE", offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001317}
1318
1319TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001320 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001321 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001322 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001323 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001324 offer = CreateOffer(NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001325 SetLocalDescriptionOfferExpectError(
1326 "Called in wrong state: STATE_RECEIVEDINITIATE", offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001327}
1328
1329TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001330 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001331 mediastream_signaling_.SendNothing();
1332 SessionDescriptionInterface* offer = CreateRemoteOffer();
1333 SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE);
1334
1335 JsepSessionDescription* pranswer = static_cast<JsepSessionDescription*>(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001336 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001337 pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
1338 SetLocalDescriptionExpectState(pranswer, BaseSession::STATE_SENTPRACCEPT);
1339
1340 mediastream_signaling_.SendAudioVideoStream1();
1341 JsepSessionDescription* pranswer2 = static_cast<JsepSessionDescription*>(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001342 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001343 pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
1344
1345 SetLocalDescriptionExpectState(pranswer2, BaseSession::STATE_SENTPRACCEPT);
1346
1347 mediastream_signaling_.SendAudioVideoStream2();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001348 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001349 SetLocalDescriptionExpectState(answer, BaseSession::STATE_SENTACCEPT);
1350}
1351
1352TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001353 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001354 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001355 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001356 SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
1357
1358 JsepSessionDescription* pranswer =
1359 CreateRemoteAnswer(session_->local_description());
1360 pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
1361
1362 SetRemoteDescriptionExpectState(pranswer,
1363 BaseSession::STATE_RECEIVEDPRACCEPT);
1364
1365 mediastream_signaling_.SendAudioVideoStream1();
1366 JsepSessionDescription* pranswer2 =
1367 CreateRemoteAnswer(session_->local_description());
1368 pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
1369
1370 SetRemoteDescriptionExpectState(pranswer2,
1371 BaseSession::STATE_RECEIVEDPRACCEPT);
1372
1373 mediastream_signaling_.SendAudioVideoStream2();
1374 SessionDescriptionInterface* answer =
1375 CreateRemoteAnswer(session_->local_description());
1376 SetRemoteDescriptionExpectState(answer, BaseSession::STATE_RECEIVEDACCEPT);
1377}
1378
1379TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001380 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001381 mediastream_signaling_.SendNothing();
1382 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001383 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001384 SessionDescriptionInterface* answer =
1385 CreateRemoteAnswer(offer.get());
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001386 SetLocalDescriptionAnswerExpectError("Called in wrong state: STATE_INIT",
1387 answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001388}
1389
1390TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001391 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001392 mediastream_signaling_.SendNothing();
1393 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001394 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001395 SessionDescriptionInterface* answer =
1396 CreateRemoteAnswer(offer.get());
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001397 SetRemoteDescriptionAnswerExpectError(
1398 "Called in wrong state: STATE_INIT", answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001399}
1400
1401TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001402 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001403 mediastream_signaling_.SendAudioVideoStream1();
1404
1405 cricket::Candidate candidate;
1406 candidate.set_component(1);
1407 JsepIceCandidate ice_candidate1(kMediaContentName0, 0, candidate);
1408
1409 // Fail since we have not set a offer description.
1410 EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1));
1411
wu@webrtc.org91053e72013-08-10 07:18:04 +00001412 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001413 SetLocalDescriptionWithoutError(offer);
1414 // Candidate should be allowed to add before remote description.
1415 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
1416 candidate.set_component(2);
1417 JsepIceCandidate ice_candidate2(kMediaContentName0, 0, candidate);
1418 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
1419
1420 SessionDescriptionInterface* answer = CreateRemoteAnswer(
1421 session_->local_description());
1422 SetRemoteDescriptionWithoutError(answer);
1423
1424 // Verifying the candidates are copied properly from internal vector.
1425 const SessionDescriptionInterface* remote_desc =
1426 session_->remote_description();
1427 ASSERT_TRUE(remote_desc != NULL);
1428 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1429 const IceCandidateCollection* candidates =
1430 remote_desc->candidates(kMediaContentIndex0);
1431 ASSERT_EQ(2u, candidates->count());
1432 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1433 EXPECT_EQ(kMediaContentName0, candidates->at(0)->sdp_mid());
1434 EXPECT_EQ(1, candidates->at(0)->candidate().component());
1435 EXPECT_EQ(2, candidates->at(1)->candidate().component());
1436
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +00001437 // |ice_candidate3| is identical to |ice_candidate2|. It can be added
1438 // successfully, but the total count of candidates will not increase.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001439 candidate.set_component(2);
1440 JsepIceCandidate ice_candidate3(kMediaContentName0, 0, candidate);
1441 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate3));
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +00001442 ASSERT_EQ(2u, candidates->count());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001443
1444 JsepIceCandidate bad_ice_candidate("bad content name", 99, candidate);
1445 EXPECT_FALSE(session_->ProcessIceMessage(&bad_ice_candidate));
1446}
1447
1448// Test that a remote candidate is added to the remote session description and
1449// that it is retained if the remote session description is changed.
1450TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001451 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001452 cricket::Candidate candidate1;
1453 candidate1.set_component(1);
1454 JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
1455 candidate1);
1456 mediastream_signaling_.SendAudioVideoStream1();
1457 CreateAndSetRemoteOfferAndLocalAnswer();
1458
1459 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
1460 const SessionDescriptionInterface* remote_desc =
1461 session_->remote_description();
1462 ASSERT_TRUE(remote_desc != NULL);
1463 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1464 const IceCandidateCollection* candidates =
1465 remote_desc->candidates(kMediaContentIndex0);
1466 ASSERT_EQ(1u, candidates->count());
1467 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1468
1469 // Update the RemoteSessionDescription with a new session description and
1470 // a candidate and check that the new remote session description contains both
1471 // candidates.
1472 SessionDescriptionInterface* offer = CreateRemoteOffer();
1473 cricket::Candidate candidate2;
1474 JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0,
1475 candidate2);
1476 EXPECT_TRUE(offer->AddCandidate(&ice_candidate2));
1477 SetRemoteDescriptionWithoutError(offer);
1478
1479 remote_desc = session_->remote_description();
1480 ASSERT_TRUE(remote_desc != NULL);
1481 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1482 candidates = remote_desc->candidates(kMediaContentIndex0);
1483 ASSERT_EQ(2u, candidates->count());
1484 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1485 // Username and password have be updated with the TransportInfo of the
1486 // SessionDescription, won't be equal to the original one.
1487 candidate2.set_username(candidates->at(0)->candidate().username());
1488 candidate2.set_password(candidates->at(0)->candidate().password());
1489 EXPECT_TRUE(candidate2.IsEquivalent(candidates->at(0)->candidate()));
1490 EXPECT_EQ(kMediaContentIndex0, candidates->at(1)->sdp_mline_index());
1491 // No need to verify the username and password.
1492 candidate1.set_username(candidates->at(1)->candidate().username());
1493 candidate1.set_password(candidates->at(1)->candidate().password());
1494 EXPECT_TRUE(candidate1.IsEquivalent(candidates->at(1)->candidate()));
1495
1496 // Test that the candidate is ignored if we can add the same candidate again.
1497 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
1498}
1499
1500// Test that local candidates are added to the local session description and
1501// that they are retained if the local session description is changed.
1502TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001503 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001504 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001505 mediastream_signaling_.SendAudioVideoStream1();
1506 CreateAndSetRemoteOfferAndLocalAnswer();
1507
1508 const SessionDescriptionInterface* local_desc = session_->local_description();
1509 const IceCandidateCollection* candidates =
1510 local_desc->candidates(kMediaContentIndex0);
1511 ASSERT_TRUE(candidates != NULL);
1512 EXPECT_EQ(0u, candidates->count());
1513
1514 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
1515
1516 local_desc = session_->local_description();
1517 candidates = local_desc->candidates(kMediaContentIndex0);
1518 ASSERT_TRUE(candidates != NULL);
1519 EXPECT_LT(0u, candidates->count());
1520 candidates = local_desc->candidates(1);
1521 ASSERT_TRUE(candidates != NULL);
1522 EXPECT_LT(0u, candidates->count());
1523
1524 // Update the session descriptions.
1525 mediastream_signaling_.SendAudioVideoStream1();
1526 CreateAndSetRemoteOfferAndLocalAnswer();
1527
1528 local_desc = session_->local_description();
1529 candidates = local_desc->candidates(kMediaContentIndex0);
1530 ASSERT_TRUE(candidates != NULL);
1531 EXPECT_LT(0u, candidates->count());
1532 candidates = local_desc->candidates(1);
1533 ASSERT_TRUE(candidates != NULL);
1534 EXPECT_LT(0u, candidates->count());
1535}
1536
1537// Test that we can set a remote session description with remote candidates.
1538TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001539 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001540
1541 cricket::Candidate candidate1;
1542 candidate1.set_component(1);
1543 JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
1544 candidate1);
1545 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001546 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001547
1548 EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
1549 SetRemoteDescriptionWithoutError(offer);
1550
1551 const SessionDescriptionInterface* remote_desc =
1552 session_->remote_description();
1553 ASSERT_TRUE(remote_desc != NULL);
1554 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1555 const IceCandidateCollection* candidates =
1556 remote_desc->candidates(kMediaContentIndex0);
1557 ASSERT_EQ(1u, candidates->count());
1558 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1559
wu@webrtc.org91053e72013-08-10 07:18:04 +00001560 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001561 SetLocalDescriptionWithoutError(answer);
1562}
1563
1564// Test that offers and answers contains ice candidates when Ice candidates have
1565// been gathered.
1566TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001567 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001568 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001569 mediastream_signaling_.SendAudioVideoStream1();
1570 // Ice is started but candidates are not provided until SetLocalDescription
1571 // is called.
1572 EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
1573 EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
1574 CreateAndSetRemoteOfferAndLocalAnswer();
1575 // Wait until at least one local candidate has been collected.
1576 EXPECT_TRUE_WAIT(0u < observer_.mline_0_candidates_.size(),
1577 kIceCandidatesTimeout);
1578 EXPECT_TRUE_WAIT(0u < observer_.mline_1_candidates_.size(),
1579 kIceCandidatesTimeout);
1580
1581 talk_base::scoped_ptr<SessionDescriptionInterface> local_offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001582 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001583 ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
1584 EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
1585 ASSERT_TRUE(local_offer->candidates(kMediaContentIndex1) != NULL);
1586 EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex1)->count());
1587
1588 SessionDescriptionInterface* remote_offer(CreateRemoteOffer());
1589 SetRemoteDescriptionWithoutError(remote_offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001590 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001591 ASSERT_TRUE(answer->candidates(kMediaContentIndex0) != NULL);
1592 EXPECT_LT(0u, answer->candidates(kMediaContentIndex0)->count());
1593 ASSERT_TRUE(answer->candidates(kMediaContentIndex1) != NULL);
1594 EXPECT_LT(0u, answer->candidates(kMediaContentIndex1)->count());
1595 SetLocalDescriptionWithoutError(answer);
1596}
1597
1598// Verifies TransportProxy and media channels are created with content names
1599// present in the SessionDescription.
1600TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001601 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001602 mediastream_signaling_.SendAudioVideoStream1();
1603 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001604 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001605
1606 // CreateOffer creates session description with the content names "audio" and
1607 // "video". Goal is to modify these content names and verify transport channel
1608 // proxy in the BaseSession, as proxies are created with the content names
1609 // present in SDP.
1610 std::string sdp;
1611 EXPECT_TRUE(offer->ToString(&sdp));
1612 const std::string kAudioMid = "a=mid:audio";
1613 const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
1614 const std::string kVideoMid = "a=mid:video";
1615 const std::string kVideoMidReplaceStr = "a=mid:video_content_name";
1616
1617 // Replacing |audio| with |audio_content_name|.
1618 talk_base::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
1619 kAudioMidReplaceStr.c_str(),
1620 kAudioMidReplaceStr.length(),
1621 &sdp);
1622 // Replacing |video| with |video_content_name|.
1623 talk_base::replace_substrs(kVideoMid.c_str(), kVideoMid.length(),
1624 kVideoMidReplaceStr.c_str(),
1625 kVideoMidReplaceStr.length(),
1626 &sdp);
1627
1628 SessionDescriptionInterface* modified_offer =
1629 CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
1630
1631 SetRemoteDescriptionWithoutError(modified_offer);
1632
1633 SessionDescriptionInterface* answer =
wu@webrtc.org91053e72013-08-10 07:18:04 +00001634 CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001635 SetLocalDescriptionWithoutError(answer);
1636
1637 EXPECT_TRUE(session_->GetTransportProxy("audio_content_name") != NULL);
1638 EXPECT_TRUE(session_->GetTransportProxy("video_content_name") != NULL);
1639 EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
1640 EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL);
1641}
1642
1643// Test that an offer contains the correct media content descriptions based on
1644// the send streams when no constraints have been set.
1645TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001646 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001647 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001648 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001649 ASSERT_TRUE(offer != NULL);
1650 const cricket::ContentInfo* content =
1651 cricket::GetFirstAudioContent(offer->description());
1652 EXPECT_TRUE(content != NULL);
1653 content = cricket::GetFirstVideoContent(offer->description());
1654 EXPECT_TRUE(content == NULL);
1655}
1656
1657// Test that an offer contains the correct media content descriptions based on
1658// the send streams when no constraints have been set.
1659TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001660 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001661 // Test Audio only offer.
1662 mediastream_signaling_.UseOptionsAudioOnly();
1663 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001664 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001665 const cricket::ContentInfo* content =
1666 cricket::GetFirstAudioContent(offer->description());
1667 EXPECT_TRUE(content != NULL);
1668 content = cricket::GetFirstVideoContent(offer->description());
1669 EXPECT_TRUE(content == NULL);
1670
1671 // Test Audio / Video offer.
1672 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001673 offer.reset(CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001674 content = cricket::GetFirstAudioContent(offer->description());
1675 EXPECT_TRUE(content != NULL);
1676 content = cricket::GetFirstVideoContent(offer->description());
1677 EXPECT_TRUE(content != NULL);
1678}
1679
1680// Test that an offer contains no media content descriptions if
1681// kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
1682TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001683 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001684 webrtc::FakeConstraints constraints_no_receive;
1685 constraints_no_receive.SetMandatoryReceiveAudio(false);
1686 constraints_no_receive.SetMandatoryReceiveVideo(false);
1687
1688 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001689 CreateOffer(&constraints_no_receive));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001690 ASSERT_TRUE(offer != NULL);
1691 const cricket::ContentInfo* content =
1692 cricket::GetFirstAudioContent(offer->description());
1693 EXPECT_TRUE(content == NULL);
1694 content = cricket::GetFirstVideoContent(offer->description());
1695 EXPECT_TRUE(content == NULL);
1696}
1697
1698// Test that an offer contains only audio media content descriptions if
1699// kOfferToReceiveAudio constraints are set to true.
1700TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001701 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001702 webrtc::FakeConstraints constraints_audio_only;
1703 constraints_audio_only.SetMandatoryReceiveAudio(true);
1704 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001705 CreateOffer(&constraints_audio_only));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001706
1707 const cricket::ContentInfo* content =
1708 cricket::GetFirstAudioContent(offer->description());
1709 EXPECT_TRUE(content != NULL);
1710 content = cricket::GetFirstVideoContent(offer->description());
1711 EXPECT_TRUE(content == NULL);
1712}
1713
1714// Test that an offer contains audio and video media content descriptions if
1715// kOfferToReceiveAudio and kOfferToReceiveVideo constraints are set to true.
1716TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001717 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001718 // Test Audio / Video offer.
1719 webrtc::FakeConstraints constraints_audio_video;
1720 constraints_audio_video.SetMandatoryReceiveAudio(true);
1721 constraints_audio_video.SetMandatoryReceiveVideo(true);
1722 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001723 CreateOffer(&constraints_audio_video));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001724 const cricket::ContentInfo* content =
1725 cricket::GetFirstAudioContent(offer->description());
1726
1727 EXPECT_TRUE(content != NULL);
1728 content = cricket::GetFirstVideoContent(offer->description());
1729 EXPECT_TRUE(content != NULL);
1730
1731 // TODO(perkj): Should the direction be set to SEND_ONLY if
1732 // The constraints is set to not receive audio or video but a track is added?
1733}
1734
1735// Test that an answer can not be created if the last remote description is not
1736// an offer.
1737TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001738 Init(NULL);
1739 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001740 SetLocalDescriptionWithoutError(offer);
1741 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
1742 SetRemoteDescriptionWithoutError(answer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001743 EXPECT_TRUE(CreateAnswer(NULL) == NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001744}
1745
1746// Test that an answer contains the correct media content descriptions when no
1747// constraints have been set.
1748TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001749 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001750 // Create a remote offer with audio and video content.
1751 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1752 SetRemoteDescriptionWithoutError(offer.release());
1753 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001754 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001755 const cricket::ContentInfo* content =
1756 cricket::GetFirstAudioContent(answer->description());
1757 ASSERT_TRUE(content != NULL);
1758 EXPECT_FALSE(content->rejected);
1759
1760 content = cricket::GetFirstVideoContent(answer->description());
1761 ASSERT_TRUE(content != NULL);
1762 EXPECT_FALSE(content->rejected);
1763}
1764
1765// Test that an answer contains the correct media content descriptions when no
1766// constraints have been set and the offer only contain audio.
1767TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001768 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001769 // Create a remote offer with audio only.
1770 cricket::MediaSessionOptions options;
1771 options.has_audio = true;
1772 options.has_video = false;
1773 talk_base::scoped_ptr<JsepSessionDescription> offer(
1774 CreateRemoteOffer(options));
1775 ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
1776 ASSERT_TRUE(cricket::GetFirstAudioContent(offer->description()) != NULL);
1777
1778 SetRemoteDescriptionWithoutError(offer.release());
1779 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001780 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001781 const cricket::ContentInfo* content =
1782 cricket::GetFirstAudioContent(answer->description());
1783 ASSERT_TRUE(content != NULL);
1784 EXPECT_FALSE(content->rejected);
1785
1786 EXPECT_TRUE(cricket::GetFirstVideoContent(answer->description()) == NULL);
1787}
1788
1789// Test that an answer contains the correct media content descriptions when no
1790// constraints have been set.
1791TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001792 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001793 // Create a remote offer with audio and video content.
1794 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1795 SetRemoteDescriptionWithoutError(offer.release());
1796 // Test with a stream with tracks.
1797 mediastream_signaling_.SendAudioVideoStream1();
1798 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001799 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001800 const cricket::ContentInfo* content =
1801 cricket::GetFirstAudioContent(answer->description());
1802 ASSERT_TRUE(content != NULL);
1803 EXPECT_FALSE(content->rejected);
1804
1805 content = cricket::GetFirstVideoContent(answer->description());
1806 ASSERT_TRUE(content != NULL);
1807 EXPECT_FALSE(content->rejected);
1808}
1809
1810// Test that an answer contains the correct media content descriptions when
1811// constraints have been set but no stream is sent.
1812TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001813 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001814 // Create a remote offer with audio and video content.
1815 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1816 SetRemoteDescriptionWithoutError(offer.release());
1817
1818 webrtc::FakeConstraints constraints_no_receive;
1819 constraints_no_receive.SetMandatoryReceiveAudio(false);
1820 constraints_no_receive.SetMandatoryReceiveVideo(false);
1821
1822 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001823 CreateAnswer(&constraints_no_receive));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001824 const cricket::ContentInfo* content =
1825 cricket::GetFirstAudioContent(answer->description());
1826 ASSERT_TRUE(content != NULL);
1827 EXPECT_TRUE(content->rejected);
1828
1829 content = cricket::GetFirstVideoContent(answer->description());
1830 ASSERT_TRUE(content != NULL);
1831 EXPECT_TRUE(content->rejected);
1832}
1833
1834// Test that an answer contains the correct media content descriptions when
1835// constraints have been set and streams are sent.
1836TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001837 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001838 // Create a remote offer with audio and video content.
1839 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1840 SetRemoteDescriptionWithoutError(offer.release());
1841
1842 webrtc::FakeConstraints constraints_no_receive;
1843 constraints_no_receive.SetMandatoryReceiveAudio(false);
1844 constraints_no_receive.SetMandatoryReceiveVideo(false);
1845
1846 // Test with a stream with tracks.
1847 mediastream_signaling_.SendAudioVideoStream1();
1848 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001849 CreateAnswer(&constraints_no_receive));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001850
1851 // TODO(perkj): Should the direction be set to SEND_ONLY?
1852 const cricket::ContentInfo* content =
1853 cricket::GetFirstAudioContent(answer->description());
1854 ASSERT_TRUE(content != NULL);
1855 EXPECT_FALSE(content->rejected);
1856
1857 // TODO(perkj): Should the direction be set to SEND_ONLY?
1858 content = cricket::GetFirstVideoContent(answer->description());
1859 ASSERT_TRUE(content != NULL);
1860 EXPECT_FALSE(content->rejected);
1861}
1862
1863TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
1864 AddCNCodecs();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001865 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001866 webrtc::FakeConstraints constraints;
1867 constraints.SetOptionalVAD(false);
1868 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001869 CreateOffer(&constraints));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001870 const cricket::ContentInfo* content =
1871 cricket::GetFirstAudioContent(offer->description());
1872 EXPECT_TRUE(content != NULL);
1873 EXPECT_TRUE(VerifyNoCNCodecs(content));
1874}
1875
1876TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
1877 AddCNCodecs();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001878 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001879 // Create a remote offer with audio and video content.
1880 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1881 SetRemoteDescriptionWithoutError(offer.release());
1882
1883 webrtc::FakeConstraints constraints;
1884 constraints.SetOptionalVAD(false);
1885 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001886 CreateAnswer(&constraints));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001887 const cricket::ContentInfo* content =
1888 cricket::GetFirstAudioContent(answer->description());
1889 ASSERT_TRUE(content != NULL);
1890 EXPECT_TRUE(VerifyNoCNCodecs(content));
1891}
1892
1893// This test verifies the call setup when remote answer with audio only and
1894// later updates with video.
1895TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001896 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001897 EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
1898 EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
1899
1900 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001901 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001902
1903 cricket::MediaSessionOptions options;
1904 options.has_video = false;
1905 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer, options);
1906
1907 // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
1908 // and answer;
1909 SetLocalDescriptionWithoutError(offer);
1910 SetRemoteDescriptionWithoutError(answer);
1911
1912 video_channel_ = media_engine_->GetVideoChannel(0);
1913 voice_channel_ = media_engine_->GetVoiceChannel(0);
1914
1915 ASSERT_TRUE(video_channel_ == NULL);
1916
1917 ASSERT_EQ(0u, voice_channel_->recv_streams().size());
1918 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1919 EXPECT_EQ(kAudioTrack1, voice_channel_->send_streams()[0].id);
1920
1921 // Let the remote end update the session descriptions, with Audio and Video.
1922 mediastream_signaling_.SendAudioVideoStream2();
1923 CreateAndSetRemoteOfferAndLocalAnswer();
1924
1925 video_channel_ = media_engine_->GetVideoChannel(0);
1926 voice_channel_ = media_engine_->GetVoiceChannel(0);
1927
1928 ASSERT_TRUE(video_channel_ != NULL);
1929 ASSERT_TRUE(voice_channel_ != NULL);
1930
1931 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1932 ASSERT_EQ(1u, video_channel_->send_streams().size());
1933 EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
1934 EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
1935 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1936 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1937 EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
1938 EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
1939
1940 // Change session back to audio only.
1941 mediastream_signaling_.UseOptionsAudioOnly();
1942 CreateAndSetRemoteOfferAndLocalAnswer();
1943
1944 EXPECT_EQ(0u, video_channel_->recv_streams().size());
1945 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1946 EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
1947 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1948 EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
1949}
1950
1951// This test verifies the call setup when remote answer with video only and
1952// later updates with audio.
1953TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001954 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001955 EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
1956 EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
1957 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001958 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001959
1960 cricket::MediaSessionOptions options;
1961 options.has_audio = false;
1962 options.has_video = true;
1963 SessionDescriptionInterface* answer = CreateRemoteAnswer(
1964 offer, options, cricket::SEC_ENABLED);
1965
1966 // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
1967 // and answer.
1968 SetLocalDescriptionWithoutError(offer);
1969 SetRemoteDescriptionWithoutError(answer);
1970
1971 video_channel_ = media_engine_->GetVideoChannel(0);
1972 voice_channel_ = media_engine_->GetVoiceChannel(0);
1973
1974 ASSERT_TRUE(voice_channel_ == NULL);
1975 ASSERT_TRUE(video_channel_ != NULL);
1976
1977 EXPECT_EQ(0u, video_channel_->recv_streams().size());
1978 ASSERT_EQ(1u, video_channel_->send_streams().size());
1979 EXPECT_EQ(kVideoTrack1, video_channel_->send_streams()[0].id);
1980
1981 // Update the session descriptions, with Audio and Video.
1982 mediastream_signaling_.SendAudioVideoStream2();
1983 CreateAndSetRemoteOfferAndLocalAnswer();
1984
1985 voice_channel_ = media_engine_->GetVoiceChannel(0);
1986 ASSERT_TRUE(voice_channel_ != NULL);
1987
1988 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1989 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1990 EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
1991 EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
1992
1993 // Change session back to video only.
1994 mediastream_signaling_.UseOptionsVideoOnly();
1995 CreateAndSetRemoteOfferAndLocalAnswer();
1996
1997 video_channel_ = media_engine_->GetVideoChannel(0);
1998 voice_channel_ = media_engine_->GetVoiceChannel(0);
1999
2000 ASSERT_EQ(1u, video_channel_->recv_streams().size());
2001 EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
2002 ASSERT_EQ(1u, video_channel_->send_streams().size());
2003 EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
2004}
2005
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002006TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002007 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002008 mediastream_signaling_.SendAudioVideoStream1();
2009 scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002010 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002011 VerifyCryptoParams(offer->description());
2012 SetRemoteDescriptionWithoutError(offer.release());
wu@webrtc.org91053e72013-08-10 07:18:04 +00002013 scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002014 VerifyCryptoParams(answer->description());
2015}
2016
2017TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
wu@webrtc.org97077a32013-10-25 21:18:33 +00002018 options_.disable_encryption = true;
wu@webrtc.org91053e72013-08-10 07:18:04 +00002019 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002020 mediastream_signaling_.SendAudioVideoStream1();
2021 scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002022 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002023 VerifyNoCryptoParams(offer->description(), false);
2024}
2025
2026TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002027 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002028 VerifyAnswerFromNonCryptoOffer();
2029}
2030
2031TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002032 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002033 VerifyAnswerFromCryptoOffer();
2034}
2035
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002036// This test verifies that setLocalDescription fails if
2037// no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
2038TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
2039 Init(NULL);
2040 mediastream_signaling_.SendAudioVideoStream1();
2041 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2042 std::string sdp;
2043 RemoveIceUfragPwdLines(offer.get(), &sdp);
2044 SessionDescriptionInterface* modified_offer =
2045 CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002046 SetLocalDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer);
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002047}
2048
2049// This test verifies that setRemoteDescription fails if
2050// no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
2051TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
2052 Init(NULL);
2053 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
2054 std::string sdp;
2055 RemoveIceUfragPwdLines(offer.get(), &sdp);
2056 SessionDescriptionInterface* modified_offer =
2057 CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002058 SetRemoteDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer);
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002059}
2060
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002061TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) {
2062 // This test verifies BUNDLE flag in PortAllocator, if BUNDLE information in
2063 // local description is removed by the application, BUNDLE flag should be
2064 // disabled in PortAllocator. By default BUNDLE is enabled in the WebRtc.
wu@webrtc.org91053e72013-08-10 07:18:04 +00002065 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002066 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2067 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2068 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002069 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002070 cricket::SessionDescription* offer_copy =
2071 offer->description()->Copy();
2072 offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
2073 JsepSessionDescription* modified_offer =
2074 new JsepSessionDescription(JsepSessionDescription::kOffer);
2075 modified_offer->Initialize(offer_copy, "1", "1");
2076
2077 SetLocalDescriptionWithoutError(modified_offer);
2078 EXPECT_FALSE(allocator_.flags() & cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2079}
2080
2081TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002082 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002083 mediastream_signaling_.SendAudioVideoStream1();
2084 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2085 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2086 FakeConstraints constraints;
2087 constraints.SetMandatoryUseRtpMux(true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002088 SessionDescriptionInterface* offer = CreateOffer(&constraints);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002089 SetLocalDescriptionWithoutError(offer);
2090 mediastream_signaling_.SendAudioVideoStream2();
2091 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
2092 CreateRemoteAnswer(session_->local_description()));
2093 cricket::SessionDescription* answer_copy = answer->description()->Copy();
2094 answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
2095 JsepSessionDescription* modified_answer =
2096 new JsepSessionDescription(JsepSessionDescription::kAnswer);
2097 modified_answer->Initialize(answer_copy, "1", "1");
2098 SetRemoteDescriptionWithoutError(modified_answer);
2099 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2100 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2101
2102 video_channel_ = media_engine_->GetVideoChannel(0);
2103 voice_channel_ = media_engine_->GetVoiceChannel(0);
2104
2105 ASSERT_EQ(1u, video_channel_->recv_streams().size());
2106 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
2107
2108 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
2109 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
2110
2111 ASSERT_EQ(1u, video_channel_->send_streams().size());
2112 EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
2113 ASSERT_EQ(1u, voice_channel_->send_streams().size());
2114 EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
2115}
2116
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002117// This test verifies that SetLocalDescription and SetRemoteDescription fails
2118// if BUNDLE is enabled but rtcp-mux is disabled in m-lines.
2119TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002120 WebRtcSessionTest::Init(NULL);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002121 mediastream_signaling_.SendAudioVideoStream1();
2122 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2123 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2124 FakeConstraints constraints;
2125 constraints.SetMandatoryUseRtpMux(true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002126 SessionDescriptionInterface* offer = CreateOffer(&constraints);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002127 std::string offer_str;
2128 offer->ToString(&offer_str);
2129 // Disable rtcp-mux
2130 const std::string rtcp_mux = "rtcp-mux";
2131 const std::string xrtcp_mux = "xrtcp-mux";
2132 talk_base::replace_substrs(rtcp_mux.c_str(), rtcp_mux.length(),
2133 xrtcp_mux.c_str(), xrtcp_mux.length(),
2134 &offer_str);
2135 JsepSessionDescription *local_offer =
2136 new JsepSessionDescription(JsepSessionDescription::kOffer);
2137 EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002138 SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002139 JsepSessionDescription *remote_offer =
2140 new JsepSessionDescription(JsepSessionDescription::kOffer);
2141 EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002142 SetRemoteDescriptionOfferExpectError(kBundleWithoutRtcpMux, remote_offer);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002143 // Trying unmodified SDP.
2144 SetLocalDescriptionWithoutError(offer);
2145}
2146
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002147TEST_F(WebRtcSessionTest, SetAudioPlayout) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002148 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002149 mediastream_signaling_.SendAudioVideoStream1();
2150 CreateAndSetRemoteOfferAndLocalAnswer();
2151 cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2152 ASSERT_TRUE(channel != NULL);
2153 ASSERT_EQ(1u, channel->recv_streams().size());
2154 uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc();
2155 double left_vol, right_vol;
2156 EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
2157 EXPECT_EQ(1, left_vol);
2158 EXPECT_EQ(1, right_vol);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002159 talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
2160 session_->SetAudioPlayout(receive_ssrc, false, renderer.get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002161 EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
2162 EXPECT_EQ(0, left_vol);
2163 EXPECT_EQ(0, right_vol);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002164 EXPECT_EQ(0, renderer->channel_id());
2165 session_->SetAudioPlayout(receive_ssrc, true, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002166 EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
2167 EXPECT_EQ(1, left_vol);
2168 EXPECT_EQ(1, right_vol);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002169 EXPECT_EQ(-1, renderer->channel_id());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002170}
2171
2172TEST_F(WebRtcSessionTest, SetAudioSend) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002173 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002174 mediastream_signaling_.SendAudioVideoStream1();
2175 CreateAndSetRemoteOfferAndLocalAnswer();
2176 cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2177 ASSERT_TRUE(channel != NULL);
2178 ASSERT_EQ(1u, channel->send_streams().size());
2179 uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2180 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2181
2182 cricket::AudioOptions options;
2183 options.echo_cancellation.Set(true);
2184
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002185 talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
2186 session_->SetAudioSend(send_ssrc, false, options, renderer.get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002187 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
2188 EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002189 EXPECT_EQ(0, renderer->channel_id());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002190
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002191 session_->SetAudioSend(send_ssrc, true, options, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002192 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2193 bool value;
2194 EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
2195 EXPECT_TRUE(value);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002196 EXPECT_EQ(-1, renderer->channel_id());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002197}
2198
2199TEST_F(WebRtcSessionTest, SetVideoPlayout) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002200 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002201 mediastream_signaling_.SendAudioVideoStream1();
2202 CreateAndSetRemoteOfferAndLocalAnswer();
2203 cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
2204 ASSERT_TRUE(channel != NULL);
2205 ASSERT_LT(0u, channel->renderers().size());
2206 EXPECT_TRUE(channel->renderers().begin()->second == NULL);
2207 ASSERT_EQ(1u, channel->recv_streams().size());
2208 uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc();
2209 cricket::FakeVideoRenderer renderer;
2210 session_->SetVideoPlayout(receive_ssrc, true, &renderer);
2211 EXPECT_TRUE(channel->renderers().begin()->second == &renderer);
2212 session_->SetVideoPlayout(receive_ssrc, false, &renderer);
2213 EXPECT_TRUE(channel->renderers().begin()->second == NULL);
2214}
2215
2216TEST_F(WebRtcSessionTest, SetVideoSend) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002217 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002218 mediastream_signaling_.SendAudioVideoStream1();
2219 CreateAndSetRemoteOfferAndLocalAnswer();
2220 cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
2221 ASSERT_TRUE(channel != NULL);
2222 ASSERT_EQ(1u, channel->send_streams().size());
2223 uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2224 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2225 cricket::VideoOptions* options = NULL;
2226 session_->SetVideoSend(send_ssrc, false, options);
2227 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
2228 session_->SetVideoSend(send_ssrc, true, options);
2229 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2230}
2231
2232TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
2233 TestCanInsertDtmf(false);
2234}
2235
2236TEST_F(WebRtcSessionTest, CanInsertDtmf) {
2237 TestCanInsertDtmf(true);
2238}
2239
2240TEST_F(WebRtcSessionTest, InsertDtmf) {
2241 // Setup
wu@webrtc.org91053e72013-08-10 07:18:04 +00002242 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002243 mediastream_signaling_.SendAudioVideoStream1();
2244 CreateAndSetRemoteOfferAndLocalAnswer();
2245 FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2246 EXPECT_EQ(0U, channel->dtmf_info_queue().size());
2247
2248 // Insert DTMF
2249 const int expected_flags = DF_SEND;
2250 const int expected_duration = 90;
2251 session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
2252 session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
2253 session_->InsertDtmf(kAudioTrack1, 2, expected_duration);
2254
2255 // Verify
2256 ASSERT_EQ(3U, channel->dtmf_info_queue().size());
2257 const uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2258 EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0,
2259 expected_duration, expected_flags));
2260 EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1,
2261 expected_duration, expected_flags));
2262 EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2,
2263 expected_duration, expected_flags));
2264}
2265
2266// This test verifies the |initiator| flag when session initiates the call.
2267TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002268 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002269 EXPECT_FALSE(session_->initiator());
wu@webrtc.org91053e72013-08-10 07:18:04 +00002270 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002271 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
2272 SetLocalDescriptionWithoutError(offer);
2273 EXPECT_TRUE(session_->initiator());
2274 SetRemoteDescriptionWithoutError(answer);
2275 EXPECT_TRUE(session_->initiator());
2276}
2277
2278// This test verifies the |initiator| flag when session receives the call.
2279TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002280 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002281 EXPECT_FALSE(session_->initiator());
2282 SessionDescriptionInterface* offer = CreateRemoteOffer();
2283 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002284 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002285
2286 EXPECT_FALSE(session_->initiator());
2287 SetLocalDescriptionWithoutError(answer);
2288 EXPECT_FALSE(session_->initiator());
2289}
2290
2291// This test verifies the ice protocol type at initiator of the call
2292// if |a=ice-options:google-ice| is present in answer.
2293TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002294 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002295 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002296 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org723d6832013-07-12 16:04:50 +00002297 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002298 CreateRemoteAnswer(offer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002299 SetLocalDescriptionWithoutError(offer);
2300 std::string sdp;
2301 EXPECT_TRUE(answer->ToString(&sdp));
2302 // Adding ice-options to the session level.
2303 InjectAfter("t=0 0\r\n",
2304 "a=ice-options:google-ice\r\n",
2305 &sdp);
2306 SessionDescriptionInterface* answer_with_gice =
2307 CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
2308 SetRemoteDescriptionWithoutError(answer_with_gice);
2309 VerifyTransportType("audio", cricket::ICEPROTO_GOOGLE);
2310 VerifyTransportType("video", cricket::ICEPROTO_GOOGLE);
2311}
2312
2313// This test verifies the ice protocol type at initiator of the call
2314// if ICE RFC5245 is supported in answer.
2315TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002316 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002317 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002318 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002319 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
2320 SetLocalDescriptionWithoutError(offer);
2321
2322 SetRemoteDescriptionWithoutError(answer);
2323 VerifyTransportType("audio", cricket::ICEPROTO_RFC5245);
2324 VerifyTransportType("video", cricket::ICEPROTO_RFC5245);
2325}
2326
2327// This test verifies the ice protocol type at receiver side of the call if
2328// receiver decides to use google-ice.
2329TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002330 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002331 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002332 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002333 SetRemoteDescriptionWithoutError(offer);
henrike@webrtc.org723d6832013-07-12 16:04:50 +00002334 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002335 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002336 std::string sdp;
2337 EXPECT_TRUE(answer->ToString(&sdp));
2338 // Adding ice-options to the session level.
2339 InjectAfter("t=0 0\r\n",
2340 "a=ice-options:google-ice\r\n",
2341 &sdp);
2342 SessionDescriptionInterface* answer_with_gice =
2343 CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
2344 SetLocalDescriptionWithoutError(answer_with_gice);
2345 VerifyTransportType("audio", cricket::ICEPROTO_GOOGLE);
2346 VerifyTransportType("video", cricket::ICEPROTO_GOOGLE);
2347}
2348
2349// This test verifies the ice protocol type at receiver side of the call if
2350// receiver decides to use ice RFC 5245.
2351TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002352 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002353 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002354 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002355 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002356 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002357 SetLocalDescriptionWithoutError(answer);
2358 VerifyTransportType("audio", cricket::ICEPROTO_RFC5245);
2359 VerifyTransportType("video", cricket::ICEPROTO_RFC5245);
2360}
2361
2362// This test verifies the session state when ICE RFC5245 in offer and
2363// ICE google-ice in answer.
2364TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002365 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002366 mediastream_signaling_.SendAudioVideoStream1();
2367 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002368 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002369 std::string offer_str;
2370 offer->ToString(&offer_str);
2371 // Disable google-ice
2372 const std::string gice_option = "google-ice";
2373 const std::string xgoogle_xice = "xgoogle-xice";
2374 talk_base::replace_substrs(gice_option.c_str(), gice_option.length(),
2375 xgoogle_xice.c_str(), xgoogle_xice.length(),
2376 &offer_str);
2377 JsepSessionDescription *ice_only_offer =
2378 new JsepSessionDescription(JsepSessionDescription::kOffer);
2379 EXPECT_TRUE((ice_only_offer)->Initialize(offer_str, NULL));
2380 SetLocalDescriptionWithoutError(ice_only_offer);
2381 std::string original_offer_sdp;
2382 EXPECT_TRUE(offer->ToString(&original_offer_sdp));
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002383 SessionDescriptionInterface* pranswer_with_gice =
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002384 CreateSessionDescription(JsepSessionDescription::kPrAnswer,
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002385 original_offer_sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002386 SetRemoteDescriptionPranswerExpectError(kPushDownTDFailed,
2387 pranswer_with_gice);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002388 SessionDescriptionInterface* answer_with_gice =
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002389 CreateSessionDescription(JsepSessionDescription::kAnswer,
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002390 original_offer_sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002391 SetRemoteDescriptionAnswerExpectError(kPushDownTDFailed,
2392 answer_with_gice);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002393}
2394
2395// Verifing local offer and remote answer have matching m-lines as per RFC 3264.
2396TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002397 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002398 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002399 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002400 SetLocalDescriptionWithoutError(offer);
2401 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
2402 CreateRemoteAnswer(session_->local_description()));
2403
2404 cricket::SessionDescription* answer_copy = answer->description()->Copy();
2405 answer_copy->RemoveContentByName("video");
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002406 JsepSessionDescription* modified_answer =
2407 new JsepSessionDescription(JsepSessionDescription::kAnswer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002408
2409 EXPECT_TRUE(modified_answer->Initialize(answer_copy,
2410 answer->session_id(),
2411 answer->session_version()));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002412 SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002413
2414 // Modifying content names.
2415 std::string sdp;
2416 EXPECT_TRUE(answer->ToString(&sdp));
2417 const std::string kAudioMid = "a=mid:audio";
2418 const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
2419
2420 // Replacing |audio| with |audio_content_name|.
2421 talk_base::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
2422 kAudioMidReplaceStr.c_str(),
2423 kAudioMidReplaceStr.length(),
2424 &sdp);
2425
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002426 SessionDescriptionInterface* modified_answer1 =
2427 CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002428 SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer1);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002429
2430 SetRemoteDescriptionWithoutError(answer.release());
2431}
2432
2433// Verifying remote offer and local answer have matching m-lines as per
2434// RFC 3264.
2435TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002436 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002437 mediastream_signaling_.SendAudioVideoStream1();
2438 SessionDescriptionInterface* offer = CreateRemoteOffer();
2439 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002440 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002441
2442 cricket::SessionDescription* answer_copy = answer->description()->Copy();
2443 answer_copy->RemoveContentByName("video");
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002444 JsepSessionDescription* modified_answer =
2445 new JsepSessionDescription(JsepSessionDescription::kAnswer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002446
2447 EXPECT_TRUE(modified_answer->Initialize(answer_copy,
2448 answer->session_id(),
2449 answer->session_version()));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002450 SetLocalDescriptionAnswerExpectError(kMlineMismatch, modified_answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002451 SetLocalDescriptionWithoutError(answer);
2452}
2453
2454// This test verifies that WebRtcSession does not start candidate allocation
2455// before SetLocalDescription is called.
2456TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002457 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002458 mediastream_signaling_.SendAudioVideoStream1();
2459 SessionDescriptionInterface* offer = CreateRemoteOffer();
2460 cricket::Candidate candidate;
2461 candidate.set_component(1);
2462 JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
2463 candidate);
2464 EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
2465 cricket::Candidate candidate1;
2466 candidate1.set_component(1);
2467 JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1,
2468 candidate1);
2469 EXPECT_TRUE(offer->AddCandidate(&ice_candidate1));
2470 SetRemoteDescriptionWithoutError(offer);
2471 ASSERT_TRUE(session_->GetTransportProxy("audio") != NULL);
2472 ASSERT_TRUE(session_->GetTransportProxy("video") != NULL);
2473
2474 // Pump for 1 second and verify that no candidates are generated.
2475 talk_base::Thread::Current()->ProcessMessages(1000);
2476 EXPECT_TRUE(observer_.mline_0_candidates_.empty());
2477 EXPECT_TRUE(observer_.mline_1_candidates_.empty());
2478
wu@webrtc.org91053e72013-08-10 07:18:04 +00002479 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002480 SetLocalDescriptionWithoutError(answer);
2481 EXPECT_TRUE(session_->GetTransportProxy("audio")->negotiated());
2482 EXPECT_TRUE(session_->GetTransportProxy("video")->negotiated());
2483 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
2484}
2485
2486// This test verifies that crypto parameter is updated in local session
2487// description as per security policy set in MediaSessionDescriptionFactory.
2488TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002489 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002490 mediastream_signaling_.SendAudioVideoStream1();
2491 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002492 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002493
2494 // Making sure SetLocalDescription correctly sets crypto value in
2495 // SessionDescription object after de-serialization of sdp string. The value
2496 // will be set as per MediaSessionDescriptionFactory.
2497 std::string offer_str;
2498 offer->ToString(&offer_str);
2499 SessionDescriptionInterface* jsep_offer_str =
2500 CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
2501 SetLocalDescriptionWithoutError(jsep_offer_str);
2502 EXPECT_TRUE(session_->voice_channel()->secure_required());
2503 EXPECT_TRUE(session_->video_channel()->secure_required());
2504}
2505
2506// This test verifies the crypto parameter when security is disabled.
2507TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
wu@webrtc.org97077a32013-10-25 21:18:33 +00002508 options_.disable_encryption = true;
wu@webrtc.org91053e72013-08-10 07:18:04 +00002509 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002510 mediastream_signaling_.SendAudioVideoStream1();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002511 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002512 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002513
2514 // Making sure SetLocalDescription correctly sets crypto value in
2515 // SessionDescription object after de-serialization of sdp string. The value
2516 // will be set as per MediaSessionDescriptionFactory.
2517 std::string offer_str;
2518 offer->ToString(&offer_str);
2519 SessionDescriptionInterface *jsep_offer_str =
2520 CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
2521 SetLocalDescriptionWithoutError(jsep_offer_str);
2522 EXPECT_FALSE(session_->voice_channel()->secure_required());
2523 EXPECT_FALSE(session_->video_channel()->secure_required());
2524}
2525
2526// This test verifies that an answer contains new ufrag and password if an offer
2527// with new ufrag and password is received.
2528TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002529 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002530 cricket::MediaSessionOptions options;
2531 options.has_audio = true;
2532 options.has_video = true;
2533 talk_base::scoped_ptr<JsepSessionDescription> offer(
2534 CreateRemoteOffer(options));
2535 SetRemoteDescriptionWithoutError(offer.release());
2536
2537 mediastream_signaling_.SendAudioVideoStream1();
2538 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002539 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002540 SetLocalDescriptionWithoutError(answer.release());
2541
2542 // Receive an offer with new ufrag and password.
2543 options.transport_options.ice_restart = true;
2544 talk_base::scoped_ptr<JsepSessionDescription> updated_offer1(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002545 CreateRemoteOffer(options, session_->remote_description()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002546 SetRemoteDescriptionWithoutError(updated_offer1.release());
2547
2548 talk_base::scoped_ptr<SessionDescriptionInterface> updated_answer1(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002549 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002550
2551 CompareIceUfragAndPassword(updated_answer1->description(),
2552 session_->local_description()->description(),
2553 false);
2554
2555 SetLocalDescriptionWithoutError(updated_answer1.release());
wu@webrtc.org91053e72013-08-10 07:18:04 +00002556}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002557
wu@webrtc.org91053e72013-08-10 07:18:04 +00002558// This test verifies that an answer contains old ufrag and password if an offer
2559// with old ufrag and password is received.
2560TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
2561 Init(NULL);
2562 cricket::MediaSessionOptions options;
2563 options.has_audio = true;
2564 options.has_video = true;
2565 talk_base::scoped_ptr<JsepSessionDescription> offer(
2566 CreateRemoteOffer(options));
2567 SetRemoteDescriptionWithoutError(offer.release());
2568
2569 mediastream_signaling_.SendAudioVideoStream1();
2570 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
2571 CreateAnswer(NULL));
2572 SetLocalDescriptionWithoutError(answer.release());
2573
2574 // Receive an offer without changed ufrag or password.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002575 options.transport_options.ice_restart = false;
2576 talk_base::scoped_ptr<JsepSessionDescription> updated_offer2(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002577 CreateRemoteOffer(options, session_->remote_description()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002578 SetRemoteDescriptionWithoutError(updated_offer2.release());
2579
2580 talk_base::scoped_ptr<SessionDescriptionInterface> updated_answer2(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002581 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002582
2583 CompareIceUfragAndPassword(updated_answer2->description(),
2584 session_->local_description()->description(),
2585 true);
2586
2587 SetLocalDescriptionWithoutError(updated_answer2.release());
2588}
2589
2590TEST_F(WebRtcSessionTest, TestSessionContentError) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002591 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002592 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002593 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002594 const std::string session_id_orig = offer->session_id();
2595 const std::string session_version_orig = offer->session_version();
2596 SetLocalDescriptionWithoutError(offer);
2597
2598 video_channel_ = media_engine_->GetVideoChannel(0);
2599 video_channel_->set_fail_set_send_codecs(true);
2600
2601 mediastream_signaling_.SendAudioVideoStream2();
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002602 SessionDescriptionInterface* answer =
2603 CreateRemoteAnswer(session_->local_description());
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002604 SetRemoteDescriptionAnswerExpectError("ERROR_CONTENT", answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002605}
2606
2607// Runs the loopback call test with BUNDLE and STUN disabled.
2608TEST_F(WebRtcSessionTest, TestIceStatesBasic) {
2609 // Lets try with only UDP ports.
2610 allocator_.set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
2611 cricket::PORTALLOCATOR_DISABLE_TCP |
2612 cricket::PORTALLOCATOR_DISABLE_STUN |
2613 cricket::PORTALLOCATOR_DISABLE_RELAY);
2614 TestLoopbackCall();
2615}
2616
mallinath@webrtc.org385857d2014-02-14 00:56:12 +00002617// Runs the loopback call test with BUNDLE, STUN, and TCP enabled.
2618TEST_F(WebRtcSessionTest, TestIceStatesBundle) {
2619 allocator_.set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
2620 cricket::PORTALLOCATOR_ENABLE_BUNDLE |
2621 cricket::PORTALLOCATOR_DISABLE_RELAY);
2622 TestLoopbackCall();
2623}
2624
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002625TEST_F(WebRtcSessionTest, SetSdpFailedOnSessionError) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002626 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002627 cricket::MediaSessionOptions options;
2628 options.has_audio = true;
2629 options.has_video = true;
2630
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002631 cricket::BaseSession::Error error_code = cricket::BaseSession::ERROR_CONTENT;
2632 std::string error_code_str = "ERROR_CONTENT";
2633 std::string error_desc = "Fake session error description.";
2634 session_->SetError(error_code, error_desc);
2635
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002636 SessionDescriptionInterface* offer = CreateRemoteOffer(options);
2637 SessionDescriptionInterface* answer =
2638 CreateRemoteAnswer(offer, options);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002639
2640 std::string action;
2641 std::ostringstream session_error_msg;
2642 session_error_msg << kSessionError << error_code_str << ". ";
2643 session_error_msg << kSessionErrorDesc << error_desc << ".";
2644 SetRemoteDescriptionExpectError(action, session_error_msg.str(), offer);
2645 SetLocalDescriptionExpectError(action, session_error_msg.str(), answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002646}
2647
2648TEST_F(WebRtcSessionTest, TestRtpDataChannel) {
2649 constraints_.reset(new FakeConstraints());
2650 constraints_->AddOptional(
2651 webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002652 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002653
2654 SetLocalDescriptionWithDataChannel();
2655 EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
2656}
2657
2658TEST_F(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
2659 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2660
2661 constraints_.reset(new FakeConstraints());
2662 constraints_->AddOptional(
2663 webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
wu@webrtc.org97077a32013-10-25 21:18:33 +00002664 options_.disable_sctp_data_channels = false;
2665
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002666 InitWithDtls(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002667
2668 SetLocalDescriptionWithDataChannel();
2669 EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
2670}
2671
wu@webrtc.org967bfff2013-09-19 05:49:50 +00002672TEST_F(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
2673 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2674
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002675 InitWithDtls(false);
2676
2677 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2678 EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +00002679 EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL);
2680}
2681
2682TEST_F(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
2683 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2684 SetFactoryDtlsSrtp();
mallinath@webrtc.org1112c302013-09-23 20:34:45 +00002685 InitWithDtls(false);
2686
2687 // Create remote offer with SCTP.
2688 cricket::MediaSessionOptions options;
2689 options.data_channel_type = cricket::DCT_SCTP;
2690 JsepSessionDescription* offer =
2691 CreateRemoteOffer(options, cricket::SEC_ENABLED);
2692 SetRemoteDescriptionWithoutError(offer);
2693
2694 // Verifies the answer contains SCTP.
2695 talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
2696 EXPECT_TRUE(answer != NULL);
2697 EXPECT_TRUE(answer->description()->GetContentByName("data") != NULL);
2698 EXPECT_TRUE(answer->description()->GetTransportInfoByName("data") != NULL);
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002699}
2700
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002701TEST_F(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) {
2702 constraints_.reset(new FakeConstraints());
2703 constraints_->AddOptional(
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002704 webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
2705 InitWithDtls(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002706
2707 SetLocalDescriptionWithDataChannel();
2708 EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
2709}
2710
2711TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
2712 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2713
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002714 InitWithDtls(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002715
2716 SetLocalDescriptionWithDataChannel();
2717 EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
2718}
wu@webrtc.org91053e72013-08-10 07:18:04 +00002719
wu@webrtc.org97077a32013-10-25 21:18:33 +00002720TEST_F(WebRtcSessionTest, TestDisableSctpDataChannels) {
2721 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2722 options_.disable_sctp_data_channels = true;
2723 InitWithDtls(false);
2724
2725 SetLocalDescriptionWithDataChannel();
2726 EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
2727}
2728
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002729TEST_F(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
2730 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2731 const int new_send_port = 9998;
2732 const int new_recv_port = 7775;
2733
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002734 InitWithDtls(false);
2735 SetFactoryDtlsSrtp();
2736
2737 // By default, don't actually add the codecs to desc_factory_; they don't
2738 // actually get serialized for SCTP in BuildMediaDescription(). Instead,
2739 // let the session description get parsed. That'll get the proper codecs
2740 // into the stream.
2741 cricket::MediaSessionOptions options;
2742 JsepSessionDescription* offer = CreateRemoteOfferWithSctpPort(
2743 "stream1", new_send_port, options);
2744
2745 // SetRemoteDescription will take the ownership of the offer.
2746 SetRemoteDescriptionWithoutError(offer);
2747
2748 SessionDescriptionInterface* answer = ChangeSDPSctpPort(
2749 new_recv_port, CreateAnswer(NULL));
2750 ASSERT_TRUE(answer != NULL);
2751
2752 // Now set the local description, which'll take ownership of the answer.
2753 SetLocalDescriptionWithoutError(answer);
2754
2755 // TEST PLAN: Set the port number to something new, set it in the SDP,
2756 // and pass it all the way down.
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +00002757 webrtc::InternalDataChannelInit dci;
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002758 dci.reliable = true;
2759 EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
2760 talk_base::scoped_refptr<webrtc::DataChannel> dc =
2761 session_->CreateDataChannel("datachannel", &dci);
2762
2763 cricket::FakeDataMediaChannel* ch = data_engine_->GetChannel(0);
2764 int portnum = -1;
2765 ASSERT_TRUE(ch != NULL);
2766 ASSERT_EQ(1UL, ch->send_codecs().size());
2767 EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->send_codecs()[0].id);
2768 EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
2769 ch->send_codecs()[0].name.c_str()));
2770 EXPECT_TRUE(ch->send_codecs()[0].GetParam(cricket::kCodecParamPort,
2771 &portnum));
2772 EXPECT_EQ(new_send_port, portnum);
2773
2774 ASSERT_EQ(1UL, ch->recv_codecs().size());
2775 EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->recv_codecs()[0].id);
2776 EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
2777 ch->recv_codecs()[0].name.c_str()));
2778 EXPECT_TRUE(ch->recv_codecs()[0].GetParam(cricket::kCodecParamPort,
2779 &portnum));
2780 EXPECT_EQ(new_recv_port, portnum);
2781}
2782
wu@webrtc.org91053e72013-08-10 07:18:04 +00002783// Verifies that CreateOffer succeeds when CreateOffer is called before async
2784// identity generation is finished.
2785TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
2786 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002787 InitWithDtls(false);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002788
2789 EXPECT_TRUE(session_->waiting_for_identity());
2790 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2791 EXPECT_TRUE(offer != NULL);
2792}
2793
2794// Verifies that CreateAnswer succeeds when CreateOffer is called before async
2795// identity generation is finished.
2796TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
2797 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002798 InitWithDtls(false);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002799
2800 cricket::MediaSessionOptions options;
2801 scoped_ptr<JsepSessionDescription> offer(
2802 CreateRemoteOffer(options, cricket::SEC_REQUIRED));
2803 ASSERT_TRUE(offer.get() != NULL);
2804 SetRemoteDescriptionWithoutError(offer.release());
2805
2806 talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
2807 EXPECT_TRUE(answer != NULL);
2808}
2809
2810// Verifies that CreateOffer succeeds when CreateOffer is called after async
2811// identity generation is finished.
2812TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
2813 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002814 InitWithDtls(false);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002815
2816 EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
2817 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2818 EXPECT_TRUE(offer != NULL);
2819}
2820
2821// Verifies that CreateOffer fails when CreateOffer is called after async
2822// identity generation fails.
2823TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
2824 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002825 InitWithDtls(true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002826
2827 EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
2828 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2829 EXPECT_TRUE(offer == NULL);
2830}
2831
2832// Verifies that CreateOffer succeeds when Multiple CreateOffer calls are made
2833// before async identity generation is finished.
2834TEST_F(WebRtcSessionTest,
2835 TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
2836 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2837 VerifyMultipleAsyncCreateDescription(
2838 true, CreateSessionDescriptionRequest::kOffer);
2839}
2840
2841// Verifies that CreateOffer fails when Multiple CreateOffer calls are made
2842// before async identity generation fails.
2843TEST_F(WebRtcSessionTest,
2844 TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) {
2845 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2846 VerifyMultipleAsyncCreateDescription(
2847 false, CreateSessionDescriptionRequest::kOffer);
2848}
2849
2850// Verifies that CreateAnswer succeeds when Multiple CreateAnswer calls are made
2851// before async identity generation is finished.
2852TEST_F(WebRtcSessionTest,
2853 TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) {
2854 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2855 VerifyMultipleAsyncCreateDescription(
2856 true, CreateSessionDescriptionRequest::kAnswer);
2857}
2858
2859// Verifies that CreateAnswer fails when Multiple CreateAnswer calls are made
2860// before async identity generation fails.
2861TEST_F(WebRtcSessionTest,
2862 TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) {
2863 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2864 VerifyMultipleAsyncCreateDescription(
2865 false, CreateSessionDescriptionRequest::kAnswer);
2866}
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +00002867
2868// Verifies that setRemoteDescription fails when DTLS is disabled and the remote
2869// offer has no SDES crypto but only DTLS fingerprint.
2870TEST_F(WebRtcSessionTest, TestSetRemoteOfferFailIfDtlsDisabledAndNoCrypto) {
2871 // Init without DTLS.
2872 Init(NULL);
2873 // Create a remote offer with secured transport disabled.
2874 cricket::MediaSessionOptions options;
2875 JsepSessionDescription* offer(CreateRemoteOffer(
2876 options, cricket::SEC_DISABLED));
2877 // Adds a DTLS fingerprint to the remote offer.
2878 cricket::SessionDescription* sdp = offer->description();
2879 TransportInfo* audio = sdp->GetTransportInfoByName("audio");
2880 ASSERT_TRUE(audio != NULL);
2881 ASSERT_TRUE(audio->description.identity_fingerprint.get() == NULL);
2882 audio->description.identity_fingerprint.reset(
2883 talk_base::SSLFingerprint::CreateFromRfc4572(
2884 talk_base::DIGEST_SHA_256, kFakeDtlsFingerprint));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002885 SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesAndDtlsDisabled,
2886 offer);
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +00002887}
2888
wu@webrtc.orgde305012013-10-31 15:40:38 +00002889// This test verifies DSCP is properly applied on the media channels.
2890TEST_F(WebRtcSessionTest, TestDscpConstraint) {
2891 constraints_.reset(new FakeConstraints());
2892 constraints_->AddOptional(
2893 webrtc::MediaConstraintsInterface::kEnableDscp, true);
2894 Init(NULL);
2895 mediastream_signaling_.SendAudioVideoStream1();
2896 SessionDescriptionInterface* offer = CreateOffer(NULL);
2897
2898 SetLocalDescriptionWithoutError(offer);
2899
2900 video_channel_ = media_engine_->GetVideoChannel(0);
2901 voice_channel_ = media_engine_->GetVoiceChannel(0);
2902
2903 ASSERT_TRUE(video_channel_ != NULL);
2904 ASSERT_TRUE(voice_channel_ != NULL);
2905 cricket::AudioOptions audio_options;
2906 EXPECT_TRUE(voice_channel_->GetOptions(&audio_options));
2907 cricket::VideoOptions video_options;
2908 EXPECT_TRUE(video_channel_->GetOptions(&video_options));
2909 EXPECT_TRUE(audio_options.dscp.IsSet());
2910 EXPECT_TRUE(audio_options.dscp.GetWithDefaultIfUnset(false));
2911 EXPECT_TRUE(video_options.dscp.IsSet());
2912 EXPECT_TRUE(video_options.dscp.GetWithDefaultIfUnset(false));
2913}
2914
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002915// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
2916// currently fails because upon disconnection and reconnection OnIceComplete is
2917// called more than once without returning to IceGatheringGathering.