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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:
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000252 FakeAudioRenderer() : channel_id_(-1), sink_(NULL) {}
253 virtual ~FakeAudioRenderer() {
254 if (sink_)
255 sink_->OnClose();
256 }
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000257
258 virtual void AddChannel(int channel_id) OVERRIDE {
259 ASSERT(channel_id_ == -1);
260 channel_id_ = channel_id;
261 }
262 virtual void RemoveChannel(int channel_id) OVERRIDE {
263 ASSERT(channel_id == channel_id_);
264 channel_id_ = -1;
265 }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000266 virtual void SetSink(Sink* sink) OVERRIDE {
267 sink_ = sink;
268 }
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000269
270 int channel_id() const { return channel_id_; }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000271 cricket::AudioRenderer::Sink* sink() const { return sink_; }
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000272 private:
273 int channel_id_;
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000274 cricket::AudioRenderer::Sink* sink_;
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000275};
276
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000277class WebRtcSessionTest : public testing::Test {
278 protected:
279 // TODO Investigate why ChannelManager crashes, if it's created
280 // after stun_server.
281 WebRtcSessionTest()
282 : media_engine_(new cricket::FakeMediaEngine()),
283 data_engine_(new cricket::FakeDataEngine()),
284 device_manager_(new cricket::FakeDeviceManager()),
285 channel_manager_(new cricket::ChannelManager(
286 media_engine_, data_engine_, device_manager_,
287 new cricket::CaptureManager(), talk_base::Thread::Current())),
288 tdesc_factory_(new cricket::TransportDescriptionFactory()),
289 desc_factory_(new cricket::MediaSessionDescriptionFactory(
290 channel_manager_.get(), tdesc_factory_.get())),
291 pss_(new talk_base::PhysicalSocketServer),
292 vss_(new talk_base::VirtualSocketServer(pss_.get())),
293 fss_(new talk_base::FirewallSocketServer(vss_.get())),
294 ss_scope_(fss_.get()),
wu@webrtc.org364f2042013-11-20 21:49:41 +0000295 stun_socket_addr_(talk_base::SocketAddress(kStunAddrHost,
296 cricket::STUN_SERVER_PORT)),
297 stun_server_(talk_base::Thread::Current(), stun_socket_addr_),
298 allocator_(&network_manager_, stun_socket_addr_,
wu@webrtc.org967bfff2013-09-19 05:49:50 +0000299 SocketAddress(), SocketAddress(), SocketAddress()),
300 mediastream_signaling_(channel_manager_.get()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000301 tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);
302 allocator_.set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
303 cricket::PORTALLOCATOR_DISABLE_RELAY |
304 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
305 EXPECT_TRUE(channel_manager_->Init());
306 desc_factory_->set_add_legacy_streams(false);
307 }
308
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000309 static void SetUpTestCase() {
310 talk_base::InitializeSSL();
311 }
312
313 static void TearDownTestCase() {
314 talk_base::CleanupSSL();
315 }
316
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000317 void AddInterface(const SocketAddress& addr) {
318 network_manager_.AddInterface(addr);
319 }
320
wu@webrtc.org91053e72013-08-10 07:18:04 +0000321 void Init(DTLSIdentityServiceInterface* identity_service) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000322 ASSERT_TRUE(session_.get() == NULL);
323 session_.reset(new WebRtcSessionForTest(
324 channel_manager_.get(), talk_base::Thread::Current(),
325 talk_base::Thread::Current(), &allocator_,
326 &observer_,
327 &mediastream_signaling_));
328
329 EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
330 observer_.ice_connection_state_);
331 EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
332 observer_.ice_gathering_state_);
333
wu@webrtc.org97077a32013-10-25 21:18:33 +0000334 EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
335 identity_service));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000336 }
337
338 void InitWithDtmfCodec() {
339 // Add kTelephoneEventCodec for dtmf test.
wu@webrtc.org364f2042013-11-20 21:49:41 +0000340 const cricket::AudioCodec kTelephoneEventCodec(
341 106, "telephone-event", 8000, 0, 1, 0);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000342 std::vector<cricket::AudioCodec> codecs;
343 codecs.push_back(kTelephoneEventCodec);
344 media_engine_->SetAudioCodecs(codecs);
345 desc_factory_->set_audio_codecs(codecs);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000346 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000347 }
348
henrike@webrtc.org7666db72013-08-22 14:45:42 +0000349 void InitWithDtls(bool identity_request_should_fail = false) {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000350 FakeIdentityService* identity_service = new FakeIdentityService();
351 identity_service->set_should_fail(identity_request_should_fail);
352 Init(identity_service);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000353 }
354
355 // Creates a local offer and applies it. Starts ice.
356 // Call mediastream_signaling_.UseOptionsWithStreamX() before this function
357 // to decide which streams to create.
358 void InitiateCall() {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000359 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000360 SetLocalDescriptionWithoutError(offer);
361 EXPECT_TRUE_WAIT(PeerConnectionInterface::kIceGatheringNew !=
362 observer_.ice_gathering_state_,
363 kIceCandidatesTimeout);
364 }
365
wu@webrtc.org91053e72013-08-10 07:18:04 +0000366 SessionDescriptionInterface* CreateOffer(
367 const webrtc::MediaConstraintsInterface* constraints) {
368 talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
369 observer = new WebRtcSessionCreateSDPObserverForTest();
370 session_->CreateOffer(observer, constraints);
371 EXPECT_TRUE_WAIT(
372 observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000373 2000);
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000374 return observer->ReleaseDescription();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000375 }
376
377 SessionDescriptionInterface* CreateAnswer(
378 const webrtc::MediaConstraintsInterface* constraints) {
379 talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observer
380 = new WebRtcSessionCreateSDPObserverForTest();
381 session_->CreateAnswer(observer, constraints);
382 EXPECT_TRUE_WAIT(
383 observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000384 2000);
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000385 return observer->ReleaseDescription();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000386 }
387
wu@webrtc.org364f2042013-11-20 21:49:41 +0000388 bool ChannelsExist() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000389 return (session_->voice_channel() != NULL &&
390 session_->video_channel() != NULL);
391 }
392
wu@webrtc.org364f2042013-11-20 21:49:41 +0000393 void CheckTransportChannels() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000394 EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 1) != NULL);
395 EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 2) != NULL);
396 EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 1) != NULL);
397 EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 2) != NULL);
398 }
399
400 void VerifyCryptoParams(const cricket::SessionDescription* sdp) {
401 ASSERT_TRUE(session_.get() != NULL);
402 const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
403 ASSERT_TRUE(content != NULL);
404 const cricket::AudioContentDescription* audio_content =
405 static_cast<const cricket::AudioContentDescription*>(
406 content->description);
407 ASSERT_TRUE(audio_content != NULL);
408 ASSERT_EQ(1U, audio_content->cryptos().size());
409 ASSERT_EQ(47U, audio_content->cryptos()[0].key_params.size());
410 ASSERT_EQ("AES_CM_128_HMAC_SHA1_80",
411 audio_content->cryptos()[0].cipher_suite);
412 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
413 audio_content->protocol());
414
415 content = cricket::GetFirstVideoContent(sdp);
416 ASSERT_TRUE(content != NULL);
417 const cricket::VideoContentDescription* video_content =
418 static_cast<const cricket::VideoContentDescription*>(
419 content->description);
420 ASSERT_TRUE(video_content != NULL);
421 ASSERT_EQ(1U, video_content->cryptos().size());
422 ASSERT_EQ("AES_CM_128_HMAC_SHA1_80",
423 video_content->cryptos()[0].cipher_suite);
424 ASSERT_EQ(47U, video_content->cryptos()[0].key_params.size());
425 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
426 video_content->protocol());
427 }
428
429 void VerifyNoCryptoParams(const cricket::SessionDescription* sdp, bool dtls) {
430 const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
431 ASSERT_TRUE(content != NULL);
432 const cricket::AudioContentDescription* audio_content =
433 static_cast<const cricket::AudioContentDescription*>(
434 content->description);
435 ASSERT_TRUE(audio_content != NULL);
436 ASSERT_EQ(0U, audio_content->cryptos().size());
437
438 content = cricket::GetFirstVideoContent(sdp);
439 ASSERT_TRUE(content != NULL);
440 const cricket::VideoContentDescription* video_content =
441 static_cast<const cricket::VideoContentDescription*>(
442 content->description);
443 ASSERT_TRUE(video_content != NULL);
444 ASSERT_EQ(0U, video_content->cryptos().size());
445
446 if (dtls) {
447 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
448 audio_content->protocol());
449 EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
450 video_content->protocol());
451 } else {
452 EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf),
453 audio_content->protocol());
454 EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf),
455 video_content->protocol());
456 }
457 }
458
459 // Set the internal fake description factories to do DTLS-SRTP.
460 void SetFactoryDtlsSrtp() {
461 desc_factory_->set_secure(cricket::SEC_ENABLED);
462 std::string identity_name = "WebRTC" +
463 talk_base::ToString(talk_base::CreateRandomId());
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000464 identity_.reset(talk_base::SSLIdentity::Generate(identity_name));
465 tdesc_factory_->set_identity(identity_.get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000466 tdesc_factory_->set_secure(cricket::SEC_REQUIRED);
467 }
468
469 void VerifyFingerprintStatus(const cricket::SessionDescription* sdp,
470 bool expected) {
471 const TransportInfo* audio = sdp->GetTransportInfoByName("audio");
472 ASSERT_TRUE(audio != NULL);
473 ASSERT_EQ(expected, audio->description.identity_fingerprint.get() != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000474 const TransportInfo* video = sdp->GetTransportInfoByName("video");
475 ASSERT_TRUE(video != NULL);
476 ASSERT_EQ(expected, video->description.identity_fingerprint.get() != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000477 }
478
479 void VerifyAnswerFromNonCryptoOffer() {
480 // Create a SDP without Crypto.
481 cricket::MediaSessionOptions options;
482 options.has_video = true;
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000483 JsepSessionDescription* offer(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000484 CreateRemoteOffer(options, cricket::SEC_DISABLED));
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000485 ASSERT_TRUE(offer != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000486 VerifyNoCryptoParams(offer->description(), false);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000487 SetRemoteDescriptionOfferExpectError(
488 "Called with a SDP without crypto enabled", offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000489 const webrtc::SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000490 // Answer should be NULL as no crypto params in offer.
491 ASSERT_TRUE(answer == NULL);
492 }
493
494 void VerifyAnswerFromCryptoOffer() {
495 cricket::MediaSessionOptions options;
496 options.has_video = true;
497 options.bundle_enabled = true;
498 scoped_ptr<JsepSessionDescription> offer(
499 CreateRemoteOffer(options, cricket::SEC_REQUIRED));
500 ASSERT_TRUE(offer.get() != NULL);
501 VerifyCryptoParams(offer->description());
502 SetRemoteDescriptionWithoutError(offer.release());
wu@webrtc.org91053e72013-08-10 07:18:04 +0000503 scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000504 ASSERT_TRUE(answer.get() != NULL);
505 VerifyCryptoParams(answer->description());
506 }
507
508 void CompareIceUfragAndPassword(const cricket::SessionDescription* desc1,
509 const cricket::SessionDescription* desc2,
510 bool expect_equal) {
511 if (desc1->contents().size() != desc2->contents().size()) {
512 EXPECT_FALSE(expect_equal);
513 return;
514 }
515
516 const cricket::ContentInfos& contents = desc1->contents();
517 cricket::ContentInfos::const_iterator it = contents.begin();
518
519 for (; it != contents.end(); ++it) {
520 const cricket::TransportDescription* transport_desc1 =
521 desc1->GetTransportDescriptionByName(it->name);
522 const cricket::TransportDescription* transport_desc2 =
523 desc2->GetTransportDescriptionByName(it->name);
524 if (!transport_desc1 || !transport_desc2) {
525 EXPECT_FALSE(expect_equal);
526 return;
527 }
528 if (transport_desc1->ice_pwd != transport_desc2->ice_pwd ||
529 transport_desc1->ice_ufrag != transport_desc2->ice_ufrag) {
530 EXPECT_FALSE(expect_equal);
531 return;
532 }
533 }
534 EXPECT_TRUE(expect_equal);
535 }
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000536
537 void RemoveIceUfragPwdLines(const SessionDescriptionInterface* current_desc,
538 std::string *sdp) {
539 const cricket::SessionDescription* desc = current_desc->description();
540 EXPECT_TRUE(current_desc->ToString(sdp));
541
542 const cricket::ContentInfos& contents = desc->contents();
543 cricket::ContentInfos::const_iterator it = contents.begin();
544 // Replace ufrag and pwd lines with empty strings.
545 for (; it != contents.end(); ++it) {
546 const cricket::TransportDescription* transport_desc =
547 desc->GetTransportDescriptionByName(it->name);
548 std::string ufrag_line = "a=ice-ufrag:" + transport_desc->ice_ufrag
549 + "\r\n";
550 std::string pwd_line = "a=ice-pwd:" + transport_desc->ice_pwd
551 + "\r\n";
552 talk_base::replace_substrs(ufrag_line.c_str(), ufrag_line.length(),
553 "", 0,
554 sdp);
555 talk_base::replace_substrs(pwd_line.c_str(), pwd_line.length(),
556 "", 0,
557 sdp);
558 }
559 }
560
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000561 // Creates a remote offer and and applies it as a remote description,
562 // creates a local answer and applies is as a local description.
563 // Call mediastream_signaling_.UseOptionsWithStreamX() before this function
564 // to decide which local and remote streams to create.
565 void CreateAndSetRemoteOfferAndLocalAnswer() {
566 SessionDescriptionInterface* offer = CreateRemoteOffer();
567 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000568 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000569 SetLocalDescriptionWithoutError(answer);
570 }
571 void SetLocalDescriptionWithoutError(SessionDescriptionInterface* desc) {
572 EXPECT_TRUE(session_->SetLocalDescription(desc, NULL));
573 }
574 void SetLocalDescriptionExpectState(SessionDescriptionInterface* desc,
575 BaseSession::State expected_state) {
576 SetLocalDescriptionWithoutError(desc);
577 EXPECT_EQ(expected_state, session_->state());
578 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000579 void SetLocalDescriptionExpectError(const std::string& action,
580 const std::string& expected_error,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000581 SessionDescriptionInterface* desc) {
582 std::string error;
583 EXPECT_FALSE(session_->SetLocalDescription(desc, &error));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000584 std::string sdp_type = "local ";
585 sdp_type.append(action);
586 EXPECT_NE(std::string::npos, error.find(sdp_type));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000587 EXPECT_NE(std::string::npos, error.find(expected_error));
588 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000589 void SetLocalDescriptionOfferExpectError(const std::string& expected_error,
590 SessionDescriptionInterface* desc) {
591 SetLocalDescriptionExpectError(SessionDescriptionInterface::kOffer,
592 expected_error, desc);
593 }
594 void SetLocalDescriptionAnswerExpectError(const std::string& expected_error,
595 SessionDescriptionInterface* desc) {
596 SetLocalDescriptionExpectError(SessionDescriptionInterface::kAnswer,
597 expected_error, desc);
598 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000599 void SetRemoteDescriptionWithoutError(SessionDescriptionInterface* desc) {
600 EXPECT_TRUE(session_->SetRemoteDescription(desc, NULL));
601 }
602 void SetRemoteDescriptionExpectState(SessionDescriptionInterface* desc,
603 BaseSession::State expected_state) {
604 SetRemoteDescriptionWithoutError(desc);
605 EXPECT_EQ(expected_state, session_->state());
606 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000607 void SetRemoteDescriptionExpectError(const std::string& action,
608 const std::string& expected_error,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000609 SessionDescriptionInterface* desc) {
610 std::string error;
611 EXPECT_FALSE(session_->SetRemoteDescription(desc, &error));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000612 std::string sdp_type = "remote ";
613 sdp_type.append(action);
614 EXPECT_NE(std::string::npos, error.find(sdp_type));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000615 EXPECT_NE(std::string::npos, error.find(expected_error));
616 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000617 void SetRemoteDescriptionOfferExpectError(
618 const std::string& expected_error, SessionDescriptionInterface* desc) {
619 SetRemoteDescriptionExpectError(SessionDescriptionInterface::kOffer,
620 expected_error, desc);
621 }
622 void SetRemoteDescriptionPranswerExpectError(
623 const std::string& expected_error, SessionDescriptionInterface* desc) {
624 SetRemoteDescriptionExpectError(SessionDescriptionInterface::kPrAnswer,
625 expected_error, desc);
626 }
627 void SetRemoteDescriptionAnswerExpectError(
628 const std::string& expected_error, SessionDescriptionInterface* desc) {
629 SetRemoteDescriptionExpectError(SessionDescriptionInterface::kAnswer,
630 expected_error, desc);
631 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000632
633 void CreateCryptoOfferAndNonCryptoAnswer(SessionDescriptionInterface** offer,
634 SessionDescriptionInterface** nocrypto_answer) {
635 // Create a SDP without Crypto.
636 cricket::MediaSessionOptions options;
637 options.has_video = true;
638 options.bundle_enabled = true;
639 *offer = CreateRemoteOffer(options, cricket::SEC_ENABLED);
640 ASSERT_TRUE(*offer != NULL);
641 VerifyCryptoParams((*offer)->description());
642
643 *nocrypto_answer = CreateRemoteAnswer(*offer, options,
644 cricket::SEC_DISABLED);
645 EXPECT_TRUE(*nocrypto_answer != NULL);
646 }
647
648 JsepSessionDescription* CreateRemoteOfferWithVersion(
649 cricket::MediaSessionOptions options,
650 cricket::SecurePolicy secure_policy,
651 const std::string& session_version,
652 const SessionDescriptionInterface* current_desc) {
653 std::string session_id = talk_base::ToString(talk_base::CreateRandomId64());
654 const cricket::SessionDescription* cricket_desc = NULL;
655 if (current_desc) {
656 cricket_desc = current_desc->description();
657 session_id = current_desc->session_id();
658 }
659
660 desc_factory_->set_secure(secure_policy);
661 JsepSessionDescription* offer(
662 new JsepSessionDescription(JsepSessionDescription::kOffer));
663 if (!offer->Initialize(desc_factory_->CreateOffer(options, cricket_desc),
664 session_id, session_version)) {
665 delete offer;
666 offer = NULL;
667 }
668 return offer;
669 }
670 JsepSessionDescription* CreateRemoteOffer(
671 cricket::MediaSessionOptions options) {
672 return CreateRemoteOfferWithVersion(options, cricket::SEC_ENABLED,
673 kSessionVersion, NULL);
674 }
675 JsepSessionDescription* CreateRemoteOffer(
676 cricket::MediaSessionOptions options, cricket::SecurePolicy policy) {
677 return CreateRemoteOfferWithVersion(options, policy, kSessionVersion, NULL);
678 }
679 JsepSessionDescription* CreateRemoteOffer(
680 cricket::MediaSessionOptions options,
681 const SessionDescriptionInterface* current_desc) {
682 return CreateRemoteOfferWithVersion(options, cricket::SEC_ENABLED,
683 kSessionVersion, current_desc);
684 }
685
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000686 JsepSessionDescription* CreateRemoteOfferWithSctpPort(
687 const char* sctp_stream_name, int new_port,
688 cricket::MediaSessionOptions options) {
689 options.data_channel_type = cricket::DCT_SCTP;
690 options.AddStream(cricket::MEDIA_TYPE_DATA, "datachannel",
691 sctp_stream_name);
692 return ChangeSDPSctpPort(new_port, CreateRemoteOffer(options));
693 }
694
695 // Takes ownership of offer_basis (and deletes it).
696 JsepSessionDescription* ChangeSDPSctpPort(
697 int new_port, webrtc::SessionDescriptionInterface *offer_basis) {
698 // Stringify the input SDP, swap the 5000 for 'new_port' and create a new
699 // SessionDescription from the mutated string.
700 const char* default_port_str = "5000";
701 char new_port_str[16];
702 talk_base::sprintfn(new_port_str, sizeof(new_port_str), "%d", new_port);
703 std::string offer_str;
704 offer_basis->ToString(&offer_str);
705 talk_base::replace_substrs(default_port_str, strlen(default_port_str),
706 new_port_str, strlen(new_port_str),
707 &offer_str);
708 JsepSessionDescription* offer = new JsepSessionDescription(
709 offer_basis->type());
710 delete offer_basis;
711 offer->Initialize(offer_str, NULL);
712 return offer;
713 }
714
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000715 // Create a remote offer. Call mediastream_signaling_.UseOptionsWithStreamX()
716 // before this function to decide which streams to create.
717 JsepSessionDescription* CreateRemoteOffer() {
718 cricket::MediaSessionOptions options;
719 mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
720 return CreateRemoteOffer(options, session_->remote_description());
721 }
722
723 JsepSessionDescription* CreateRemoteAnswer(
724 const SessionDescriptionInterface* offer,
725 cricket::MediaSessionOptions options,
726 cricket::SecurePolicy policy) {
727 desc_factory_->set_secure(policy);
728 const std::string session_id =
729 talk_base::ToString(talk_base::CreateRandomId64());
730 JsepSessionDescription* answer(
731 new JsepSessionDescription(JsepSessionDescription::kAnswer));
732 if (!answer->Initialize(desc_factory_->CreateAnswer(offer->description(),
733 options, NULL),
734 session_id, kSessionVersion)) {
735 delete answer;
736 answer = NULL;
737 }
738 return answer;
739 }
740
741 JsepSessionDescription* CreateRemoteAnswer(
742 const SessionDescriptionInterface* offer,
743 cricket::MediaSessionOptions options) {
744 return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
745 }
746
747 // Creates an answer session description with streams based on
748 // |mediastream_signaling_|. Call
749 // mediastream_signaling_.UseOptionsWithStreamX() before this function
750 // to decide which streams to create.
751 JsepSessionDescription* CreateRemoteAnswer(
752 const SessionDescriptionInterface* offer) {
753 cricket::MediaSessionOptions options;
754 mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
755 return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
756 }
757
758 void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000759 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +0000760 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000761 mediastream_signaling_.SendAudioVideoStream1();
762 FakeConstraints constraints;
763 constraints.SetMandatoryUseRtpMux(bundle);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000764 SessionDescriptionInterface* offer = CreateOffer(&constraints);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000765 // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
766 // and answer.
767 SetLocalDescriptionWithoutError(offer);
768
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000769 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
770 CreateRemoteAnswer(session_->local_description()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000771 std::string sdp;
772 EXPECT_TRUE(answer->ToString(&sdp));
773
774 size_t expected_candidate_num = 2;
775 if (!rtcp_mux) {
776 // If rtcp_mux is enabled we should expect 4 candidates - host and srflex
777 // for rtp and rtcp.
778 expected_candidate_num = 4;
779 // Disable rtcp-mux from the answer
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000780 const std::string kRtcpMux = "a=rtcp-mux";
781 const std::string kXRtcpMux = "a=xrtcp-mux";
782 talk_base::replace_substrs(kRtcpMux.c_str(), kRtcpMux.length(),
783 kXRtcpMux.c_str(), kXRtcpMux.length(),
784 &sdp);
785 }
786
787 SessionDescriptionInterface* new_answer = CreateSessionDescription(
788 JsepSessionDescription::kAnswer, sdp, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000789
790 // SetRemoteDescription to enable rtcp mux.
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000791 SetRemoteDescriptionWithoutError(new_answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000792 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
793 EXPECT_EQ(expected_candidate_num, observer_.mline_0_candidates_.size());
794 EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
795 for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
796 cricket::Candidate c0 = observer_.mline_0_candidates_[i];
797 cricket::Candidate c1 = observer_.mline_1_candidates_[i];
798 if (bundle) {
799 EXPECT_TRUE(c0.IsEquivalent(c1));
800 } else {
801 EXPECT_FALSE(c0.IsEquivalent(c1));
802 }
803 }
804 }
805 // Tests that we can only send DTMF when the dtmf codec is supported.
806 void TestCanInsertDtmf(bool can) {
807 if (can) {
808 InitWithDtmfCodec();
809 } else {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000810 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000811 }
812 mediastream_signaling_.SendAudioVideoStream1();
813 CreateAndSetRemoteOfferAndLocalAnswer();
814 EXPECT_FALSE(session_->CanInsertDtmf(""));
815 EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
816 }
817
818 // The method sets up a call from the session to itself, in a loopback
819 // arrangement. It also uses a firewall rule to create a temporary
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000820 // disconnection, and then a permanent disconnection.
821 // This code is placed in a method so that it can be invoked
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000822 // by multiple tests with different allocators (e.g. with and without BUNDLE).
823 // While running the call, this method also checks if the session goes through
824 // the correct sequence of ICE states when a connection is established,
825 // broken, and re-established.
826 // The Connection state should go:
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000827 // New -> Checking -> (Connected) -> Completed -> Disconnected -> Completed
828 // -> Failed.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000829 // The Gathering state should go: New -> Gathering -> Completed.
830 void TestLoopbackCall() {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000831 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +0000832 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000833 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000834 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000835
836 EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
837 observer_.ice_gathering_state_);
838 SetLocalDescriptionWithoutError(offer);
839 EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
840 observer_.ice_connection_state_);
841 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringGathering,
842 observer_.ice_gathering_state_,
843 kIceCandidatesTimeout);
844 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
845 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
846 observer_.ice_gathering_state_,
847 kIceCandidatesTimeout);
848
849 std::string sdp;
850 offer->ToString(&sdp);
851 SessionDescriptionInterface* desc =
852 webrtc::CreateSessionDescription(JsepSessionDescription::kAnswer, sdp);
853 ASSERT_TRUE(desc != NULL);
854 SetRemoteDescriptionWithoutError(desc);
855
856 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
857 observer_.ice_connection_state_,
858 kIceCandidatesTimeout);
mallinath@webrtc.orgd3dc4242014-03-01 00:05:52 +0000859
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000860 // The ice connection state is "Connected" too briefly to catch in a test.
861 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000862 observer_.ice_connection_state_,
863 kIceCandidatesTimeout);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000864
865 // Adding firewall rule to block ping requests, which should cause
866 // transport channel failure.
wu@webrtc.org364f2042013-11-20 21:49:41 +0000867 fss_->AddRule(false,
868 talk_base::FP_ANY,
869 talk_base::FD_ANY,
870 talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000871 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
872 observer_.ice_connection_state_,
873 kIceCandidatesTimeout);
874
875 // Clearing the rules, session should move back to completed state.
876 fss_->ClearRules();
877 // Session is automatically calling OnSignalingReady after creation of
878 // new portallocator session which will allocate new set of candidates.
879
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000880 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000881 observer_.ice_connection_state_,
882 kIceCandidatesTimeout);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000883
884 // Now we block ping requests and wait until the ICE connection transitions
885 // to the Failed state. This will take at least 30 seconds because it must
886 // wait for the Port to timeout.
887 int port_timeout = 30000;
888 fss_->AddRule(false,
889 talk_base::FP_ANY,
890 talk_base::FD_ANY,
891 talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
892 EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
893 observer_.ice_connection_state_,
894 kIceCandidatesTimeout + port_timeout);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000895 }
896
897 void VerifyTransportType(const std::string& content_name,
898 cricket::TransportProtocol protocol) {
899 const cricket::Transport* transport = session_->GetTransport(content_name);
900 ASSERT_TRUE(transport != NULL);
901 EXPECT_EQ(protocol, transport->protocol());
902 }
903
904 // Adds CN codecs to FakeMediaEngine and MediaDescriptionFactory.
905 void AddCNCodecs() {
wu@webrtc.org364f2042013-11-20 21:49:41 +0000906 const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1, 0);
907 const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1, 0);
908
909 // Add kCNCodec for dtmf test.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000910 std::vector<cricket::AudioCodec> codecs = media_engine_->audio_codecs();;
911 codecs.push_back(kCNCodec1);
912 codecs.push_back(kCNCodec2);
913 media_engine_->SetAudioCodecs(codecs);
914 desc_factory_->set_audio_codecs(codecs);
915 }
916
917 bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
918 const cricket::ContentDescription* description = content->description;
919 ASSERT(description != NULL);
920 const cricket::AudioContentDescription* audio_content_desc =
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000921 static_cast<const cricket::AudioContentDescription*>(description);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000922 ASSERT(audio_content_desc != NULL);
923 for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
924 if (audio_content_desc->codecs()[i].name == "CN")
925 return false;
926 }
927 return true;
928 }
929
930 void SetLocalDescriptionWithDataChannel() {
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000931 webrtc::InternalDataChannelInit dci;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000932 dci.reliable = false;
933 session_->CreateDataChannel("datachannel", &dci);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000934 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000935 SetLocalDescriptionWithoutError(offer);
936 }
937
wu@webrtc.org91053e72013-08-10 07:18:04 +0000938 void VerifyMultipleAsyncCreateDescription(
939 bool success, CreateSessionDescriptionRequest::Type type) {
henrike@webrtc.org7666db72013-08-22 14:45:42 +0000940 InitWithDtls(!success);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000941
942 if (type == CreateSessionDescriptionRequest::kAnswer) {
943 cricket::MediaSessionOptions options;
944 scoped_ptr<JsepSessionDescription> offer(
945 CreateRemoteOffer(options, cricket::SEC_REQUIRED));
946 ASSERT_TRUE(offer.get() != NULL);
947 SetRemoteDescriptionWithoutError(offer.release());
948 }
949
950 const int kNumber = 3;
951 talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
952 observers[kNumber];
953 for (int i = 0; i < kNumber; ++i) {
954 observers[i] = new WebRtcSessionCreateSDPObserverForTest();
955 if (type == CreateSessionDescriptionRequest::kOffer) {
956 session_->CreateOffer(observers[i], NULL);
957 } else {
958 session_->CreateAnswer(observers[i], NULL);
959 }
960 }
961
962 WebRtcSessionCreateSDPObserverForTest::State expected_state =
963 success ? WebRtcSessionCreateSDPObserverForTest::kSucceeded :
964 WebRtcSessionCreateSDPObserverForTest::kFailed;
965
966 for (int i = 0; i < kNumber; ++i) {
967 EXPECT_EQ_WAIT(expected_state, observers[i]->state(), 1000);
968 if (success) {
969 EXPECT_TRUE(observers[i]->description() != NULL);
970 } else {
971 EXPECT_TRUE(observers[i]->description() == NULL);
972 }
973 }
974 }
975
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000976 cricket::FakeMediaEngine* media_engine_;
977 cricket::FakeDataEngine* data_engine_;
978 cricket::FakeDeviceManager* device_manager_;
979 talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
980 talk_base::scoped_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
henrike@webrtc.org723d6832013-07-12 16:04:50 +0000981 talk_base::scoped_ptr<talk_base::SSLIdentity> identity_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000982 talk_base::scoped_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
983 talk_base::scoped_ptr<talk_base::PhysicalSocketServer> pss_;
984 talk_base::scoped_ptr<talk_base::VirtualSocketServer> vss_;
985 talk_base::scoped_ptr<talk_base::FirewallSocketServer> fss_;
986 talk_base::SocketServerScope ss_scope_;
wu@webrtc.org364f2042013-11-20 21:49:41 +0000987 talk_base::SocketAddress stun_socket_addr_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000988 cricket::TestStunServer stun_server_;
989 talk_base::FakeNetworkManager network_manager_;
990 cricket::BasicPortAllocator allocator_;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000991 PeerConnectionFactoryInterface::Options options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000992 talk_base::scoped_ptr<FakeConstraints> constraints_;
993 FakeMediaStreamSignaling mediastream_signaling_;
994 talk_base::scoped_ptr<WebRtcSessionForTest> session_;
995 MockIceObserver observer_;
996 cricket::FakeVideoMediaChannel* video_channel_;
997 cricket::FakeVoiceMediaChannel* voice_channel_;
998};
999
1000TEST_F(WebRtcSessionTest, TestInitialize) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001001 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001002}
1003
1004TEST_F(WebRtcSessionTest, TestInitializeWithDtls) {
1005 InitWithDtls();
1006}
1007
wu@webrtc.org91053e72013-08-10 07:18:04 +00001008// Verifies that WebRtcSession uses SEC_REQUIRED by default.
1009TEST_F(WebRtcSessionTest, TestDefaultSetSecurePolicy) {
1010 Init(NULL);
wu@webrtc.org364f2042013-11-20 21:49:41 +00001011 EXPECT_EQ(cricket::SEC_REQUIRED, session_->SecurePolicy());
wu@webrtc.org91053e72013-08-10 07:18:04 +00001012}
1013
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001014TEST_F(WebRtcSessionTest, TestSessionCandidates) {
1015 TestSessionCandidatesWithBundleRtcpMux(false, false);
1016}
1017
1018// Below test cases (TestSessionCandidatesWith*) verify the candidates gathered
1019// with rtcp-mux and/or bundle.
1020TEST_F(WebRtcSessionTest, TestSessionCandidatesWithRtcpMux) {
1021 TestSessionCandidatesWithBundleRtcpMux(false, true);
1022}
1023
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001024TEST_F(WebRtcSessionTest, TestSessionCandidatesWithBundleRtcpMux) {
1025 TestSessionCandidatesWithBundleRtcpMux(true, true);
1026}
1027
1028TEST_F(WebRtcSessionTest, TestMultihomeCandidates) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001029 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
1030 AddInterface(talk_base::SocketAddress(kClientAddrHost2, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001031 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001032 mediastream_signaling_.SendAudioVideoStream1();
1033 InitiateCall();
1034 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
1035 EXPECT_EQ(8u, observer_.mline_0_candidates_.size());
1036 EXPECT_EQ(8u, observer_.mline_1_candidates_.size());
1037}
1038
1039TEST_F(WebRtcSessionTest, TestStunError) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001040 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
1041 AddInterface(talk_base::SocketAddress(kClientAddrHost2, kClientAddrPort));
1042 fss_->AddRule(false,
1043 talk_base::FP_UDP,
1044 talk_base::FD_ANY,
1045 talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001046 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001047 mediastream_signaling_.SendAudioVideoStream1();
1048 InitiateCall();
wu@webrtc.org364f2042013-11-20 21:49:41 +00001049 // Since kClientAddrHost1 is blocked, not expecting stun candidates for it.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001050 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
1051 EXPECT_EQ(6u, observer_.mline_0_candidates_.size());
1052 EXPECT_EQ(6u, observer_.mline_1_candidates_.size());
1053}
1054
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001055TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
1056 Init(NULL);
1057 SessionDescriptionInterface* offer = NULL;
1058 // Since |offer| is NULL, there's no way to tell if it's an offer or answer.
1059 std::string unknown_action;
1060 SetLocalDescriptionExpectError(unknown_action, kInvalidSdp, offer);
1061 SetRemoteDescriptionExpectError(unknown_action, kInvalidSdp, offer);
1062}
1063
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001064// Test creating offers and receive answers and make sure the
1065// media engine creates the expected send and receive streams.
1066TEST_F(WebRtcSessionTest, TestCreateOfferReceiveAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001067 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001068 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001069 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001070 const std::string session_id_orig = offer->session_id();
1071 const std::string session_version_orig = offer->session_version();
1072 SetLocalDescriptionWithoutError(offer);
1073
1074 mediastream_signaling_.SendAudioVideoStream2();
1075 SessionDescriptionInterface* answer =
1076 CreateRemoteAnswer(session_->local_description());
1077 SetRemoteDescriptionWithoutError(answer);
1078
1079 video_channel_ = media_engine_->GetVideoChannel(0);
1080 voice_channel_ = media_engine_->GetVoiceChannel(0);
1081
1082 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1083 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
1084
1085 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1086 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
1087
1088 ASSERT_EQ(1u, video_channel_->send_streams().size());
1089 EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
1090 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1091 EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
1092
1093 // Create new offer without send streams.
1094 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001095 offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001096
1097 // Verify the session id is the same and the session version is
1098 // increased.
1099 EXPECT_EQ(session_id_orig, offer->session_id());
1100 EXPECT_LT(talk_base::FromString<uint64>(session_version_orig),
1101 talk_base::FromString<uint64>(offer->session_version()));
1102
1103 SetLocalDescriptionWithoutError(offer);
1104
1105 mediastream_signaling_.SendAudioVideoStream2();
1106 answer = CreateRemoteAnswer(session_->local_description());
1107 SetRemoteDescriptionWithoutError(answer);
1108
1109 EXPECT_EQ(0u, video_channel_->send_streams().size());
1110 EXPECT_EQ(0u, voice_channel_->send_streams().size());
1111
1112 // Make sure the receive streams have not changed.
1113 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1114 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
1115 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1116 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
1117}
1118
1119// Test receiving offers and creating answers and make sure the
1120// media engine creates the expected send and receive streams.
1121TEST_F(WebRtcSessionTest, TestReceiveOfferCreateAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001122 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001123 mediastream_signaling_.SendAudioVideoStream2();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001124 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001125 SetRemoteDescriptionWithoutError(offer);
1126
1127 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001128 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001129 SetLocalDescriptionWithoutError(answer);
1130
1131 const std::string session_id_orig = answer->session_id();
1132 const std::string session_version_orig = answer->session_version();
1133
1134 video_channel_ = media_engine_->GetVideoChannel(0);
1135 voice_channel_ = media_engine_->GetVoiceChannel(0);
1136
1137 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1138 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
1139
1140 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1141 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
1142
1143 ASSERT_EQ(1u, video_channel_->send_streams().size());
1144 EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
1145 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1146 EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
1147
1148 mediastream_signaling_.SendAudioVideoStream1And2();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001149 offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001150 SetRemoteDescriptionWithoutError(offer);
1151
1152 // Answer by turning off all send streams.
1153 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001154 answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001155
1156 // Verify the session id is the same and the session version is
1157 // increased.
1158 EXPECT_EQ(session_id_orig, answer->session_id());
1159 EXPECT_LT(talk_base::FromString<uint64>(session_version_orig),
1160 talk_base::FromString<uint64>(answer->session_version()));
1161 SetLocalDescriptionWithoutError(answer);
1162
1163 ASSERT_EQ(2u, video_channel_->recv_streams().size());
1164 EXPECT_TRUE(kVideoTrack1 == video_channel_->recv_streams()[0].id);
1165 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[1].id);
1166 ASSERT_EQ(2u, voice_channel_->recv_streams().size());
1167 EXPECT_TRUE(kAudioTrack1 == voice_channel_->recv_streams()[0].id);
1168 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[1].id);
1169
1170 // Make sure we have no send streams.
1171 EXPECT_EQ(0u, video_channel_->send_streams().size());
1172 EXPECT_EQ(0u, voice_channel_->send_streams().size());
1173}
1174
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001175TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
1176 Init(NULL);
1177 media_engine_->set_fail_create_channel(true);
1178
1179 SessionDescriptionInterface* offer = CreateOffer(NULL);
1180 ASSERT_TRUE(offer != NULL);
1181 // SetRemoteDescription and SetLocalDescription will take the ownership of
1182 // the offer.
1183 SetRemoteDescriptionOfferExpectError(kCreateChannelFailed, offer);
1184 offer = CreateOffer(NULL);
1185 ASSERT_TRUE(offer != NULL);
1186 SetLocalDescriptionOfferExpectError(kCreateChannelFailed, offer);
1187}
1188
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001189// Test we will return fail when apply an offer that doesn't have
1190// crypto enabled.
1191TEST_F(WebRtcSessionTest, SetNonCryptoOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001192 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001193 cricket::MediaSessionOptions options;
1194 options.has_video = true;
1195 JsepSessionDescription* offer = CreateRemoteOffer(
1196 options, cricket::SEC_DISABLED);
1197 ASSERT_TRUE(offer != NULL);
1198 VerifyNoCryptoParams(offer->description(), false);
1199 // SetRemoteDescription and SetLocalDescription will take the ownership of
1200 // the offer.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001201 SetRemoteDescriptionOfferExpectError(kSdpWithoutCrypto, offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001202 offer = CreateRemoteOffer(options, cricket::SEC_DISABLED);
1203 ASSERT_TRUE(offer != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001204 SetLocalDescriptionOfferExpectError(kSdpWithoutCrypto, offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001205}
1206
1207// Test we will return fail when apply an answer that doesn't have
1208// crypto enabled.
1209TEST_F(WebRtcSessionTest, SetLocalNonCryptoAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001210 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001211 SessionDescriptionInterface* offer = NULL;
1212 SessionDescriptionInterface* answer = NULL;
1213 CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
1214 // SetRemoteDescription and SetLocalDescription will take the ownership of
1215 // the offer.
1216 SetRemoteDescriptionWithoutError(offer);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001217 SetLocalDescriptionAnswerExpectError(kSdpWithoutCrypto, answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001218}
1219
1220// Test we will return fail when apply an answer that doesn't have
1221// crypto enabled.
1222TEST_F(WebRtcSessionTest, SetRemoteNonCryptoAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001223 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001224 SessionDescriptionInterface* offer = NULL;
1225 SessionDescriptionInterface* answer = NULL;
1226 CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
1227 // SetRemoteDescription and SetLocalDescription will take the ownership of
1228 // the offer.
1229 SetLocalDescriptionWithoutError(offer);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001230 SetRemoteDescriptionAnswerExpectError(kSdpWithoutCrypto, answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001231}
1232
1233// Test that we can create and set an offer with a DTLS fingerprint.
1234TEST_F(WebRtcSessionTest, CreateSetDtlsOffer) {
1235 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
1236 InitWithDtls();
1237 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001238 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001239 ASSERT_TRUE(offer != NULL);
1240 VerifyFingerprintStatus(offer->description(), true);
1241 // SetLocalDescription will take the ownership of the offer.
1242 SetLocalDescriptionWithoutError(offer);
1243}
1244
1245// Test that we can process an offer with a DTLS fingerprint
1246// and that we return an answer with a fingerprint.
1247TEST_F(WebRtcSessionTest, ReceiveDtlsOfferCreateAnswer) {
1248 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
1249 InitWithDtls();
1250 SetFactoryDtlsSrtp();
1251 cricket::MediaSessionOptions options;
1252 options.has_video = true;
1253 JsepSessionDescription* offer = CreateRemoteOffer(options);
1254 ASSERT_TRUE(offer != NULL);
1255 VerifyFingerprintStatus(offer->description(), true);
1256
1257 // SetRemoteDescription will take the ownership of the offer.
1258 SetRemoteDescriptionWithoutError(offer);
1259
1260 // Verify that we get a crypto fingerprint in the answer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001261 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001262 ASSERT_TRUE(answer != NULL);
1263 VerifyFingerprintStatus(answer->description(), true);
1264 // Check that we don't have an a=crypto line in the answer.
1265 VerifyNoCryptoParams(answer->description(), true);
1266
1267 // Now set the local description, which should work, even without a=crypto.
1268 SetLocalDescriptionWithoutError(answer);
1269}
1270
1271// Test that even if we support DTLS, if the other side didn't offer a
1272// fingerprint, we don't either.
1273TEST_F(WebRtcSessionTest, ReceiveNoDtlsOfferCreateAnswer) {
1274 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
1275 InitWithDtls();
1276 cricket::MediaSessionOptions options;
1277 options.has_video = true;
1278 JsepSessionDescription* offer = CreateRemoteOffer(
1279 options, cricket::SEC_REQUIRED);
1280 ASSERT_TRUE(offer != NULL);
1281 VerifyFingerprintStatus(offer->description(), false);
1282
1283 // SetRemoteDescription will take the ownership of
1284 // the offer.
1285 SetRemoteDescriptionWithoutError(offer);
1286
1287 // Verify that we don't get a crypto fingerprint in the answer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001288 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001289 ASSERT_TRUE(answer != NULL);
1290 VerifyFingerprintStatus(answer->description(), false);
1291
1292 // Now set the local description.
1293 SetLocalDescriptionWithoutError(answer);
1294}
1295
1296TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001297 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001298 mediastream_signaling_.SendNothing();
1299 // SetLocalDescription take ownership of offer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001300 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001301 SetLocalDescriptionWithoutError(offer);
1302
1303 // SetLocalDescription take ownership of offer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001304 SessionDescriptionInterface* offer2 = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001305 SetLocalDescriptionWithoutError(offer2);
1306}
1307
1308TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001309 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001310 mediastream_signaling_.SendNothing();
1311 // SetLocalDescription take ownership of offer.
wu@webrtc.org91053e72013-08-10 07:18:04 +00001312 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001313 SetRemoteDescriptionWithoutError(offer);
1314
wu@webrtc.org91053e72013-08-10 07:18:04 +00001315 SessionDescriptionInterface* offer2 = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001316 SetRemoteDescriptionWithoutError(offer2);
1317}
1318
1319TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
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 SetLocalDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001324 offer = CreateOffer(NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001325 SetRemoteDescriptionOfferExpectError(
1326 "Called in wrong state: STATE_SENTINITIATE", offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001327}
1328
1329TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001330 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001331 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001332 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001333 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001334 offer = CreateOffer(NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001335 SetLocalDescriptionOfferExpectError(
1336 "Called in wrong state: STATE_RECEIVEDINITIATE", offer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001337}
1338
1339TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001340 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001341 mediastream_signaling_.SendNothing();
1342 SessionDescriptionInterface* offer = CreateRemoteOffer();
1343 SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE);
1344
1345 JsepSessionDescription* pranswer = static_cast<JsepSessionDescription*>(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001346 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001347 pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
1348 SetLocalDescriptionExpectState(pranswer, BaseSession::STATE_SENTPRACCEPT);
1349
1350 mediastream_signaling_.SendAudioVideoStream1();
1351 JsepSessionDescription* pranswer2 = static_cast<JsepSessionDescription*>(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001352 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001353 pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
1354
1355 SetLocalDescriptionExpectState(pranswer2, BaseSession::STATE_SENTPRACCEPT);
1356
1357 mediastream_signaling_.SendAudioVideoStream2();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001358 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001359 SetLocalDescriptionExpectState(answer, BaseSession::STATE_SENTACCEPT);
1360}
1361
1362TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001363 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001364 mediastream_signaling_.SendNothing();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001365 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001366 SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
1367
1368 JsepSessionDescription* pranswer =
1369 CreateRemoteAnswer(session_->local_description());
1370 pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
1371
1372 SetRemoteDescriptionExpectState(pranswer,
1373 BaseSession::STATE_RECEIVEDPRACCEPT);
1374
1375 mediastream_signaling_.SendAudioVideoStream1();
1376 JsepSessionDescription* pranswer2 =
1377 CreateRemoteAnswer(session_->local_description());
1378 pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
1379
1380 SetRemoteDescriptionExpectState(pranswer2,
1381 BaseSession::STATE_RECEIVEDPRACCEPT);
1382
1383 mediastream_signaling_.SendAudioVideoStream2();
1384 SessionDescriptionInterface* answer =
1385 CreateRemoteAnswer(session_->local_description());
1386 SetRemoteDescriptionExpectState(answer, BaseSession::STATE_RECEIVEDACCEPT);
1387}
1388
1389TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001390 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001391 mediastream_signaling_.SendNothing();
1392 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001393 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001394 SessionDescriptionInterface* answer =
1395 CreateRemoteAnswer(offer.get());
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001396 SetLocalDescriptionAnswerExpectError("Called in wrong state: STATE_INIT",
1397 answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001398}
1399
1400TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001401 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001402 mediastream_signaling_.SendNothing();
1403 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001404 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001405 SessionDescriptionInterface* answer =
1406 CreateRemoteAnswer(offer.get());
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001407 SetRemoteDescriptionAnswerExpectError(
1408 "Called in wrong state: STATE_INIT", answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001409}
1410
1411TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001412 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001413 mediastream_signaling_.SendAudioVideoStream1();
1414
1415 cricket::Candidate candidate;
1416 candidate.set_component(1);
1417 JsepIceCandidate ice_candidate1(kMediaContentName0, 0, candidate);
1418
1419 // Fail since we have not set a offer description.
1420 EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1));
1421
wu@webrtc.org91053e72013-08-10 07:18:04 +00001422 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001423 SetLocalDescriptionWithoutError(offer);
1424 // Candidate should be allowed to add before remote description.
1425 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
1426 candidate.set_component(2);
1427 JsepIceCandidate ice_candidate2(kMediaContentName0, 0, candidate);
1428 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
1429
1430 SessionDescriptionInterface* answer = CreateRemoteAnswer(
1431 session_->local_description());
1432 SetRemoteDescriptionWithoutError(answer);
1433
1434 // Verifying the candidates are copied properly from internal vector.
1435 const SessionDescriptionInterface* remote_desc =
1436 session_->remote_description();
1437 ASSERT_TRUE(remote_desc != NULL);
1438 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1439 const IceCandidateCollection* candidates =
1440 remote_desc->candidates(kMediaContentIndex0);
1441 ASSERT_EQ(2u, candidates->count());
1442 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1443 EXPECT_EQ(kMediaContentName0, candidates->at(0)->sdp_mid());
1444 EXPECT_EQ(1, candidates->at(0)->candidate().component());
1445 EXPECT_EQ(2, candidates->at(1)->candidate().component());
1446
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +00001447 // |ice_candidate3| is identical to |ice_candidate2|. It can be added
1448 // successfully, but the total count of candidates will not increase.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001449 candidate.set_component(2);
1450 JsepIceCandidate ice_candidate3(kMediaContentName0, 0, candidate);
1451 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate3));
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +00001452 ASSERT_EQ(2u, candidates->count());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001453
1454 JsepIceCandidate bad_ice_candidate("bad content name", 99, candidate);
1455 EXPECT_FALSE(session_->ProcessIceMessage(&bad_ice_candidate));
1456}
1457
1458// Test that a remote candidate is added to the remote session description and
1459// that it is retained if the remote session description is changed.
1460TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001461 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001462 cricket::Candidate candidate1;
1463 candidate1.set_component(1);
1464 JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
1465 candidate1);
1466 mediastream_signaling_.SendAudioVideoStream1();
1467 CreateAndSetRemoteOfferAndLocalAnswer();
1468
1469 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
1470 const SessionDescriptionInterface* remote_desc =
1471 session_->remote_description();
1472 ASSERT_TRUE(remote_desc != NULL);
1473 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1474 const IceCandidateCollection* candidates =
1475 remote_desc->candidates(kMediaContentIndex0);
1476 ASSERT_EQ(1u, candidates->count());
1477 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1478
1479 // Update the RemoteSessionDescription with a new session description and
1480 // a candidate and check that the new remote session description contains both
1481 // candidates.
1482 SessionDescriptionInterface* offer = CreateRemoteOffer();
1483 cricket::Candidate candidate2;
1484 JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0,
1485 candidate2);
1486 EXPECT_TRUE(offer->AddCandidate(&ice_candidate2));
1487 SetRemoteDescriptionWithoutError(offer);
1488
1489 remote_desc = session_->remote_description();
1490 ASSERT_TRUE(remote_desc != NULL);
1491 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1492 candidates = remote_desc->candidates(kMediaContentIndex0);
1493 ASSERT_EQ(2u, candidates->count());
1494 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1495 // Username and password have be updated with the TransportInfo of the
1496 // SessionDescription, won't be equal to the original one.
1497 candidate2.set_username(candidates->at(0)->candidate().username());
1498 candidate2.set_password(candidates->at(0)->candidate().password());
1499 EXPECT_TRUE(candidate2.IsEquivalent(candidates->at(0)->candidate()));
1500 EXPECT_EQ(kMediaContentIndex0, candidates->at(1)->sdp_mline_index());
1501 // No need to verify the username and password.
1502 candidate1.set_username(candidates->at(1)->candidate().username());
1503 candidate1.set_password(candidates->at(1)->candidate().password());
1504 EXPECT_TRUE(candidate1.IsEquivalent(candidates->at(1)->candidate()));
1505
1506 // Test that the candidate is ignored if we can add the same candidate again.
1507 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
1508}
1509
1510// Test that local candidates are added to the local session description and
1511// that they are retained if the local session description is changed.
1512TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001513 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001514 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001515 mediastream_signaling_.SendAudioVideoStream1();
1516 CreateAndSetRemoteOfferAndLocalAnswer();
1517
1518 const SessionDescriptionInterface* local_desc = session_->local_description();
1519 const IceCandidateCollection* candidates =
1520 local_desc->candidates(kMediaContentIndex0);
1521 ASSERT_TRUE(candidates != NULL);
1522 EXPECT_EQ(0u, candidates->count());
1523
1524 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
1525
1526 local_desc = session_->local_description();
1527 candidates = local_desc->candidates(kMediaContentIndex0);
1528 ASSERT_TRUE(candidates != NULL);
1529 EXPECT_LT(0u, candidates->count());
1530 candidates = local_desc->candidates(1);
1531 ASSERT_TRUE(candidates != NULL);
1532 EXPECT_LT(0u, candidates->count());
1533
1534 // Update the session descriptions.
1535 mediastream_signaling_.SendAudioVideoStream1();
1536 CreateAndSetRemoteOfferAndLocalAnswer();
1537
1538 local_desc = session_->local_description();
1539 candidates = local_desc->candidates(kMediaContentIndex0);
1540 ASSERT_TRUE(candidates != NULL);
1541 EXPECT_LT(0u, candidates->count());
1542 candidates = local_desc->candidates(1);
1543 ASSERT_TRUE(candidates != NULL);
1544 EXPECT_LT(0u, candidates->count());
1545}
1546
1547// Test that we can set a remote session description with remote candidates.
1548TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001549 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001550
1551 cricket::Candidate candidate1;
1552 candidate1.set_component(1);
1553 JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
1554 candidate1);
1555 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001556 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001557
1558 EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
1559 SetRemoteDescriptionWithoutError(offer);
1560
1561 const SessionDescriptionInterface* remote_desc =
1562 session_->remote_description();
1563 ASSERT_TRUE(remote_desc != NULL);
1564 ASSERT_EQ(2u, remote_desc->number_of_mediasections());
1565 const IceCandidateCollection* candidates =
1566 remote_desc->candidates(kMediaContentIndex0);
1567 ASSERT_EQ(1u, candidates->count());
1568 EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
1569
wu@webrtc.org91053e72013-08-10 07:18:04 +00001570 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001571 SetLocalDescriptionWithoutError(answer);
1572}
1573
1574// Test that offers and answers contains ice candidates when Ice candidates have
1575// been gathered.
1576TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
wu@webrtc.org364f2042013-11-20 21:49:41 +00001577 AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
wu@webrtc.org91053e72013-08-10 07:18:04 +00001578 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001579 mediastream_signaling_.SendAudioVideoStream1();
1580 // Ice is started but candidates are not provided until SetLocalDescription
1581 // is called.
1582 EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
1583 EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
1584 CreateAndSetRemoteOfferAndLocalAnswer();
1585 // Wait until at least one local candidate has been collected.
1586 EXPECT_TRUE_WAIT(0u < observer_.mline_0_candidates_.size(),
1587 kIceCandidatesTimeout);
1588 EXPECT_TRUE_WAIT(0u < observer_.mline_1_candidates_.size(),
1589 kIceCandidatesTimeout);
1590
1591 talk_base::scoped_ptr<SessionDescriptionInterface> local_offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001592 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001593 ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
1594 EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
1595 ASSERT_TRUE(local_offer->candidates(kMediaContentIndex1) != NULL);
1596 EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex1)->count());
1597
1598 SessionDescriptionInterface* remote_offer(CreateRemoteOffer());
1599 SetRemoteDescriptionWithoutError(remote_offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001600 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001601 ASSERT_TRUE(answer->candidates(kMediaContentIndex0) != NULL);
1602 EXPECT_LT(0u, answer->candidates(kMediaContentIndex0)->count());
1603 ASSERT_TRUE(answer->candidates(kMediaContentIndex1) != NULL);
1604 EXPECT_LT(0u, answer->candidates(kMediaContentIndex1)->count());
1605 SetLocalDescriptionWithoutError(answer);
1606}
1607
1608// Verifies TransportProxy and media channels are created with content names
1609// present in the SessionDescription.
1610TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001611 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001612 mediastream_signaling_.SendAudioVideoStream1();
1613 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001614 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001615
1616 // CreateOffer creates session description with the content names "audio" and
1617 // "video". Goal is to modify these content names and verify transport channel
1618 // proxy in the BaseSession, as proxies are created with the content names
1619 // present in SDP.
1620 std::string sdp;
1621 EXPECT_TRUE(offer->ToString(&sdp));
1622 const std::string kAudioMid = "a=mid:audio";
1623 const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
1624 const std::string kVideoMid = "a=mid:video";
1625 const std::string kVideoMidReplaceStr = "a=mid:video_content_name";
1626
1627 // Replacing |audio| with |audio_content_name|.
1628 talk_base::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
1629 kAudioMidReplaceStr.c_str(),
1630 kAudioMidReplaceStr.length(),
1631 &sdp);
1632 // Replacing |video| with |video_content_name|.
1633 talk_base::replace_substrs(kVideoMid.c_str(), kVideoMid.length(),
1634 kVideoMidReplaceStr.c_str(),
1635 kVideoMidReplaceStr.length(),
1636 &sdp);
1637
1638 SessionDescriptionInterface* modified_offer =
1639 CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
1640
1641 SetRemoteDescriptionWithoutError(modified_offer);
1642
1643 SessionDescriptionInterface* answer =
wu@webrtc.org91053e72013-08-10 07:18:04 +00001644 CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001645 SetLocalDescriptionWithoutError(answer);
1646
1647 EXPECT_TRUE(session_->GetTransportProxy("audio_content_name") != NULL);
1648 EXPECT_TRUE(session_->GetTransportProxy("video_content_name") != NULL);
1649 EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
1650 EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL);
1651}
1652
1653// Test that an offer contains the correct media content descriptions based on
1654// the send streams when no constraints have been set.
1655TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001656 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001657 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001658 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001659 ASSERT_TRUE(offer != NULL);
1660 const cricket::ContentInfo* content =
1661 cricket::GetFirstAudioContent(offer->description());
1662 EXPECT_TRUE(content != NULL);
1663 content = cricket::GetFirstVideoContent(offer->description());
1664 EXPECT_TRUE(content == NULL);
1665}
1666
1667// Test that an offer contains the correct media content descriptions based on
1668// the send streams when no constraints have been set.
1669TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001670 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001671 // Test Audio only offer.
1672 mediastream_signaling_.UseOptionsAudioOnly();
1673 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001674 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001675 const cricket::ContentInfo* content =
1676 cricket::GetFirstAudioContent(offer->description());
1677 EXPECT_TRUE(content != NULL);
1678 content = cricket::GetFirstVideoContent(offer->description());
1679 EXPECT_TRUE(content == NULL);
1680
1681 // Test Audio / Video offer.
1682 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001683 offer.reset(CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001684 content = cricket::GetFirstAudioContent(offer->description());
1685 EXPECT_TRUE(content != NULL);
1686 content = cricket::GetFirstVideoContent(offer->description());
1687 EXPECT_TRUE(content != NULL);
1688}
1689
1690// Test that an offer contains no media content descriptions if
1691// kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
1692TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001693 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001694 webrtc::FakeConstraints constraints_no_receive;
1695 constraints_no_receive.SetMandatoryReceiveAudio(false);
1696 constraints_no_receive.SetMandatoryReceiveVideo(false);
1697
1698 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001699 CreateOffer(&constraints_no_receive));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001700 ASSERT_TRUE(offer != NULL);
1701 const cricket::ContentInfo* content =
1702 cricket::GetFirstAudioContent(offer->description());
1703 EXPECT_TRUE(content == NULL);
1704 content = cricket::GetFirstVideoContent(offer->description());
1705 EXPECT_TRUE(content == NULL);
1706}
1707
1708// Test that an offer contains only audio media content descriptions if
1709// kOfferToReceiveAudio constraints are set to true.
1710TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001711 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001712 webrtc::FakeConstraints constraints_audio_only;
1713 constraints_audio_only.SetMandatoryReceiveAudio(true);
1714 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001715 CreateOffer(&constraints_audio_only));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001716
1717 const cricket::ContentInfo* content =
1718 cricket::GetFirstAudioContent(offer->description());
1719 EXPECT_TRUE(content != NULL);
1720 content = cricket::GetFirstVideoContent(offer->description());
1721 EXPECT_TRUE(content == NULL);
1722}
1723
1724// Test that an offer contains audio and video media content descriptions if
1725// kOfferToReceiveAudio and kOfferToReceiveVideo constraints are set to true.
1726TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001727 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001728 // Test Audio / Video offer.
1729 webrtc::FakeConstraints constraints_audio_video;
1730 constraints_audio_video.SetMandatoryReceiveAudio(true);
1731 constraints_audio_video.SetMandatoryReceiveVideo(true);
1732 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001733 CreateOffer(&constraints_audio_video));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001734 const cricket::ContentInfo* content =
1735 cricket::GetFirstAudioContent(offer->description());
1736
1737 EXPECT_TRUE(content != NULL);
1738 content = cricket::GetFirstVideoContent(offer->description());
1739 EXPECT_TRUE(content != NULL);
1740
1741 // TODO(perkj): Should the direction be set to SEND_ONLY if
1742 // The constraints is set to not receive audio or video but a track is added?
1743}
1744
1745// Test that an answer can not be created if the last remote description is not
1746// an offer.
1747TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001748 Init(NULL);
1749 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001750 SetLocalDescriptionWithoutError(offer);
1751 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
1752 SetRemoteDescriptionWithoutError(answer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00001753 EXPECT_TRUE(CreateAnswer(NULL) == NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001754}
1755
1756// Test that an answer contains the correct media content descriptions when no
1757// constraints have been set.
1758TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001759 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001760 // Create a remote offer with audio and video content.
1761 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1762 SetRemoteDescriptionWithoutError(offer.release());
1763 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001764 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001765 const cricket::ContentInfo* content =
1766 cricket::GetFirstAudioContent(answer->description());
1767 ASSERT_TRUE(content != NULL);
1768 EXPECT_FALSE(content->rejected);
1769
1770 content = cricket::GetFirstVideoContent(answer->description());
1771 ASSERT_TRUE(content != NULL);
1772 EXPECT_FALSE(content->rejected);
1773}
1774
1775// Test that an answer contains the correct media content descriptions when no
1776// constraints have been set and the offer only contain audio.
1777TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001778 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001779 // Create a remote offer with audio only.
1780 cricket::MediaSessionOptions options;
1781 options.has_audio = true;
1782 options.has_video = false;
1783 talk_base::scoped_ptr<JsepSessionDescription> offer(
1784 CreateRemoteOffer(options));
1785 ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
1786 ASSERT_TRUE(cricket::GetFirstAudioContent(offer->description()) != NULL);
1787
1788 SetRemoteDescriptionWithoutError(offer.release());
1789 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001790 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001791 const cricket::ContentInfo* content =
1792 cricket::GetFirstAudioContent(answer->description());
1793 ASSERT_TRUE(content != NULL);
1794 EXPECT_FALSE(content->rejected);
1795
1796 EXPECT_TRUE(cricket::GetFirstVideoContent(answer->description()) == NULL);
1797}
1798
1799// Test that an answer contains the correct media content descriptions when no
1800// constraints have been set.
1801TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001802 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001803 // Create a remote offer with audio and video content.
1804 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1805 SetRemoteDescriptionWithoutError(offer.release());
1806 // Test with a stream with tracks.
1807 mediastream_signaling_.SendAudioVideoStream1();
1808 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001809 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001810 const cricket::ContentInfo* content =
1811 cricket::GetFirstAudioContent(answer->description());
1812 ASSERT_TRUE(content != NULL);
1813 EXPECT_FALSE(content->rejected);
1814
1815 content = cricket::GetFirstVideoContent(answer->description());
1816 ASSERT_TRUE(content != NULL);
1817 EXPECT_FALSE(content->rejected);
1818}
1819
1820// Test that an answer contains the correct media content descriptions when
1821// constraints have been set but no stream is sent.
1822TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001823 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001824 // Create a remote offer with audio and video content.
1825 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1826 SetRemoteDescriptionWithoutError(offer.release());
1827
1828 webrtc::FakeConstraints constraints_no_receive;
1829 constraints_no_receive.SetMandatoryReceiveAudio(false);
1830 constraints_no_receive.SetMandatoryReceiveVideo(false);
1831
1832 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001833 CreateAnswer(&constraints_no_receive));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001834 const cricket::ContentInfo* content =
1835 cricket::GetFirstAudioContent(answer->description());
1836 ASSERT_TRUE(content != NULL);
1837 EXPECT_TRUE(content->rejected);
1838
1839 content = cricket::GetFirstVideoContent(answer->description());
1840 ASSERT_TRUE(content != NULL);
1841 EXPECT_TRUE(content->rejected);
1842}
1843
1844// Test that an answer contains the correct media content descriptions when
1845// constraints have been set and streams are sent.
1846TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001847 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001848 // Create a remote offer with audio and video content.
1849 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1850 SetRemoteDescriptionWithoutError(offer.release());
1851
1852 webrtc::FakeConstraints constraints_no_receive;
1853 constraints_no_receive.SetMandatoryReceiveAudio(false);
1854 constraints_no_receive.SetMandatoryReceiveVideo(false);
1855
1856 // Test with a stream with tracks.
1857 mediastream_signaling_.SendAudioVideoStream1();
1858 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001859 CreateAnswer(&constraints_no_receive));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001860
1861 // TODO(perkj): Should the direction be set to SEND_ONLY?
1862 const cricket::ContentInfo* content =
1863 cricket::GetFirstAudioContent(answer->description());
1864 ASSERT_TRUE(content != NULL);
1865 EXPECT_FALSE(content->rejected);
1866
1867 // TODO(perkj): Should the direction be set to SEND_ONLY?
1868 content = cricket::GetFirstVideoContent(answer->description());
1869 ASSERT_TRUE(content != NULL);
1870 EXPECT_FALSE(content->rejected);
1871}
1872
1873TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
1874 AddCNCodecs();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001875 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001876 webrtc::FakeConstraints constraints;
1877 constraints.SetOptionalVAD(false);
1878 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001879 CreateOffer(&constraints));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001880 const cricket::ContentInfo* content =
1881 cricket::GetFirstAudioContent(offer->description());
1882 EXPECT_TRUE(content != NULL);
1883 EXPECT_TRUE(VerifyNoCNCodecs(content));
1884}
1885
1886TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
1887 AddCNCodecs();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001888 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001889 // Create a remote offer with audio and video content.
1890 talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
1891 SetRemoteDescriptionWithoutError(offer.release());
1892
1893 webrtc::FakeConstraints constraints;
1894 constraints.SetOptionalVAD(false);
1895 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00001896 CreateAnswer(&constraints));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001897 const cricket::ContentInfo* content =
1898 cricket::GetFirstAudioContent(answer->description());
1899 ASSERT_TRUE(content != NULL);
1900 EXPECT_TRUE(VerifyNoCNCodecs(content));
1901}
1902
1903// This test verifies the call setup when remote answer with audio only and
1904// later updates with video.
1905TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001906 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001907 EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
1908 EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
1909
1910 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001911 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001912
1913 cricket::MediaSessionOptions options;
1914 options.has_video = false;
1915 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer, options);
1916
1917 // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
1918 // and answer;
1919 SetLocalDescriptionWithoutError(offer);
1920 SetRemoteDescriptionWithoutError(answer);
1921
1922 video_channel_ = media_engine_->GetVideoChannel(0);
1923 voice_channel_ = media_engine_->GetVoiceChannel(0);
1924
1925 ASSERT_TRUE(video_channel_ == NULL);
1926
1927 ASSERT_EQ(0u, voice_channel_->recv_streams().size());
1928 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1929 EXPECT_EQ(kAudioTrack1, voice_channel_->send_streams()[0].id);
1930
1931 // Let the remote end update the session descriptions, with Audio and Video.
1932 mediastream_signaling_.SendAudioVideoStream2();
1933 CreateAndSetRemoteOfferAndLocalAnswer();
1934
1935 video_channel_ = media_engine_->GetVideoChannel(0);
1936 voice_channel_ = media_engine_->GetVoiceChannel(0);
1937
1938 ASSERT_TRUE(video_channel_ != NULL);
1939 ASSERT_TRUE(voice_channel_ != NULL);
1940
1941 ASSERT_EQ(1u, video_channel_->recv_streams().size());
1942 ASSERT_EQ(1u, video_channel_->send_streams().size());
1943 EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
1944 EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
1945 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1946 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1947 EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
1948 EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
1949
1950 // Change session back to audio only.
1951 mediastream_signaling_.UseOptionsAudioOnly();
1952 CreateAndSetRemoteOfferAndLocalAnswer();
1953
1954 EXPECT_EQ(0u, video_channel_->recv_streams().size());
1955 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1956 EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
1957 ASSERT_EQ(1u, voice_channel_->send_streams().size());
1958 EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
1959}
1960
1961// This test verifies the call setup when remote answer with video only and
1962// later updates with audio.
1963TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00001964 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001965 EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
1966 EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
1967 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00001968 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001969
1970 cricket::MediaSessionOptions options;
1971 options.has_audio = false;
1972 options.has_video = true;
1973 SessionDescriptionInterface* answer = CreateRemoteAnswer(
1974 offer, options, cricket::SEC_ENABLED);
1975
1976 // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
1977 // and answer.
1978 SetLocalDescriptionWithoutError(offer);
1979 SetRemoteDescriptionWithoutError(answer);
1980
1981 video_channel_ = media_engine_->GetVideoChannel(0);
1982 voice_channel_ = media_engine_->GetVoiceChannel(0);
1983
1984 ASSERT_TRUE(voice_channel_ == NULL);
1985 ASSERT_TRUE(video_channel_ != NULL);
1986
1987 EXPECT_EQ(0u, video_channel_->recv_streams().size());
1988 ASSERT_EQ(1u, video_channel_->send_streams().size());
1989 EXPECT_EQ(kVideoTrack1, video_channel_->send_streams()[0].id);
1990
1991 // Update the session descriptions, with Audio and Video.
1992 mediastream_signaling_.SendAudioVideoStream2();
1993 CreateAndSetRemoteOfferAndLocalAnswer();
1994
1995 voice_channel_ = media_engine_->GetVoiceChannel(0);
1996 ASSERT_TRUE(voice_channel_ != NULL);
1997
1998 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
1999 ASSERT_EQ(1u, voice_channel_->send_streams().size());
2000 EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
2001 EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
2002
2003 // Change session back to video only.
2004 mediastream_signaling_.UseOptionsVideoOnly();
2005 CreateAndSetRemoteOfferAndLocalAnswer();
2006
2007 video_channel_ = media_engine_->GetVideoChannel(0);
2008 voice_channel_ = media_engine_->GetVoiceChannel(0);
2009
2010 ASSERT_EQ(1u, video_channel_->recv_streams().size());
2011 EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
2012 ASSERT_EQ(1u, video_channel_->send_streams().size());
2013 EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
2014}
2015
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002016TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002017 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002018 mediastream_signaling_.SendAudioVideoStream1();
2019 scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002020 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002021 VerifyCryptoParams(offer->description());
2022 SetRemoteDescriptionWithoutError(offer.release());
wu@webrtc.org91053e72013-08-10 07:18:04 +00002023 scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002024 VerifyCryptoParams(answer->description());
2025}
2026
2027TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
wu@webrtc.org97077a32013-10-25 21:18:33 +00002028 options_.disable_encryption = true;
wu@webrtc.org91053e72013-08-10 07:18:04 +00002029 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002030 mediastream_signaling_.SendAudioVideoStream1();
2031 scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002032 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002033 VerifyNoCryptoParams(offer->description(), false);
2034}
2035
2036TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002037 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002038 VerifyAnswerFromNonCryptoOffer();
2039}
2040
2041TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002042 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002043 VerifyAnswerFromCryptoOffer();
2044}
2045
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002046// This test verifies that setLocalDescription fails if
2047// no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
2048TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
2049 Init(NULL);
2050 mediastream_signaling_.SendAudioVideoStream1();
2051 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2052 std::string sdp;
2053 RemoveIceUfragPwdLines(offer.get(), &sdp);
2054 SessionDescriptionInterface* modified_offer =
2055 CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002056 SetLocalDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer);
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002057}
2058
2059// This test verifies that setRemoteDescription fails if
2060// no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
2061TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
2062 Init(NULL);
2063 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
2064 std::string sdp;
2065 RemoveIceUfragPwdLines(offer.get(), &sdp);
2066 SessionDescriptionInterface* modified_offer =
2067 CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002068 SetRemoteDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer);
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002069}
2070
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002071TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) {
2072 // This test verifies BUNDLE flag in PortAllocator, if BUNDLE information in
2073 // local description is removed by the application, BUNDLE flag should be
2074 // disabled in PortAllocator. By default BUNDLE is enabled in the WebRtc.
wu@webrtc.org91053e72013-08-10 07:18:04 +00002075 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002076 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2077 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2078 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002079 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002080 cricket::SessionDescription* offer_copy =
2081 offer->description()->Copy();
2082 offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
2083 JsepSessionDescription* modified_offer =
2084 new JsepSessionDescription(JsepSessionDescription::kOffer);
2085 modified_offer->Initialize(offer_copy, "1", "1");
2086
2087 SetLocalDescriptionWithoutError(modified_offer);
2088 EXPECT_FALSE(allocator_.flags() & cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2089}
2090
2091TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002092 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002093 mediastream_signaling_.SendAudioVideoStream1();
2094 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2095 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2096 FakeConstraints constraints;
2097 constraints.SetMandatoryUseRtpMux(true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002098 SessionDescriptionInterface* offer = CreateOffer(&constraints);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002099 SetLocalDescriptionWithoutError(offer);
2100 mediastream_signaling_.SendAudioVideoStream2();
2101 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
2102 CreateRemoteAnswer(session_->local_description()));
2103 cricket::SessionDescription* answer_copy = answer->description()->Copy();
2104 answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
2105 JsepSessionDescription* modified_answer =
2106 new JsepSessionDescription(JsepSessionDescription::kAnswer);
2107 modified_answer->Initialize(answer_copy, "1", "1");
2108 SetRemoteDescriptionWithoutError(modified_answer);
2109 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2110 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2111
2112 video_channel_ = media_engine_->GetVideoChannel(0);
2113 voice_channel_ = media_engine_->GetVoiceChannel(0);
2114
2115 ASSERT_EQ(1u, video_channel_->recv_streams().size());
2116 EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
2117
2118 ASSERT_EQ(1u, voice_channel_->recv_streams().size());
2119 EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
2120
2121 ASSERT_EQ(1u, video_channel_->send_streams().size());
2122 EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
2123 ASSERT_EQ(1u, voice_channel_->send_streams().size());
2124 EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
2125}
2126
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002127// This test verifies that SetLocalDescription and SetRemoteDescription fails
2128// if BUNDLE is enabled but rtcp-mux is disabled in m-lines.
2129TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002130 WebRtcSessionTest::Init(NULL);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002131 mediastream_signaling_.SendAudioVideoStream1();
2132 EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
2133 cricket::PORTALLOCATOR_ENABLE_BUNDLE);
2134 FakeConstraints constraints;
2135 constraints.SetMandatoryUseRtpMux(true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002136 SessionDescriptionInterface* offer = CreateOffer(&constraints);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002137 std::string offer_str;
2138 offer->ToString(&offer_str);
2139 // Disable rtcp-mux
2140 const std::string rtcp_mux = "rtcp-mux";
2141 const std::string xrtcp_mux = "xrtcp-mux";
2142 talk_base::replace_substrs(rtcp_mux.c_str(), rtcp_mux.length(),
2143 xrtcp_mux.c_str(), xrtcp_mux.length(),
2144 &offer_str);
2145 JsepSessionDescription *local_offer =
2146 new JsepSessionDescription(JsepSessionDescription::kOffer);
2147 EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002148 SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002149 JsepSessionDescription *remote_offer =
2150 new JsepSessionDescription(JsepSessionDescription::kOffer);
2151 EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002152 SetRemoteDescriptionOfferExpectError(kBundleWithoutRtcpMux, remote_offer);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002153 // Trying unmodified SDP.
2154 SetLocalDescriptionWithoutError(offer);
2155}
2156
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002157TEST_F(WebRtcSessionTest, SetAudioPlayout) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002158 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002159 mediastream_signaling_.SendAudioVideoStream1();
2160 CreateAndSetRemoteOfferAndLocalAnswer();
2161 cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2162 ASSERT_TRUE(channel != NULL);
2163 ASSERT_EQ(1u, channel->recv_streams().size());
2164 uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc();
2165 double left_vol, right_vol;
2166 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 talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
2170 session_->SetAudioPlayout(receive_ssrc, false, renderer.get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002171 EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
2172 EXPECT_EQ(0, left_vol);
2173 EXPECT_EQ(0, right_vol);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002174 EXPECT_EQ(0, renderer->channel_id());
2175 session_->SetAudioPlayout(receive_ssrc, true, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002176 EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
2177 EXPECT_EQ(1, left_vol);
2178 EXPECT_EQ(1, right_vol);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002179 EXPECT_EQ(-1, renderer->channel_id());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002180}
2181
2182TEST_F(WebRtcSessionTest, SetAudioSend) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002183 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002184 mediastream_signaling_.SendAudioVideoStream1();
2185 CreateAndSetRemoteOfferAndLocalAnswer();
2186 cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2187 ASSERT_TRUE(channel != NULL);
2188 ASSERT_EQ(1u, channel->send_streams().size());
2189 uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2190 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2191
2192 cricket::AudioOptions options;
2193 options.echo_cancellation.Set(true);
2194
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002195 talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
2196 session_->SetAudioSend(send_ssrc, false, options, renderer.get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002197 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
2198 EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002199 EXPECT_EQ(0, renderer->channel_id());
henrike@webrtc.orga7b98182014-02-21 15:51:43 +00002200 EXPECT_TRUE(renderer->sink() != NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002201
henrike@webrtc.orga7b98182014-02-21 15:51:43 +00002202 // This will trigger SetSink(NULL) to the |renderer|.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002203 session_->SetAudioSend(send_ssrc, true, options, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002204 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2205 bool value;
2206 EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
2207 EXPECT_TRUE(value);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00002208 EXPECT_EQ(-1, renderer->channel_id());
henrike@webrtc.orga7b98182014-02-21 15:51:43 +00002209 EXPECT_TRUE(renderer->sink() == NULL);
2210}
2211
2212TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
2213 Init(NULL);
2214 mediastream_signaling_.SendAudioVideoStream1();
2215 CreateAndSetRemoteOfferAndLocalAnswer();
2216 cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2217 ASSERT_TRUE(channel != NULL);
2218 ASSERT_EQ(1u, channel->send_streams().size());
2219 uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2220
2221 talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
2222 cricket::AudioOptions options;
2223 session_->SetAudioSend(send_ssrc, true, options, renderer.get());
2224 EXPECT_TRUE(renderer->sink() != NULL);
2225
2226 // Delete the |renderer| and it will trigger OnClose() to the sink, and this
2227 // will invalidate the |renderer_| pointer in the sink and prevent getting a
2228 // SetSink(NULL) callback afterwards.
2229 renderer.reset();
2230
2231 // This will trigger SetSink(NULL) if no OnClose() callback.
2232 session_->SetAudioSend(send_ssrc, true, options, NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002233}
2234
2235TEST_F(WebRtcSessionTest, SetVideoPlayout) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002236 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002237 mediastream_signaling_.SendAudioVideoStream1();
2238 CreateAndSetRemoteOfferAndLocalAnswer();
2239 cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
2240 ASSERT_TRUE(channel != NULL);
2241 ASSERT_LT(0u, channel->renderers().size());
2242 EXPECT_TRUE(channel->renderers().begin()->second == NULL);
2243 ASSERT_EQ(1u, channel->recv_streams().size());
2244 uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc();
2245 cricket::FakeVideoRenderer renderer;
2246 session_->SetVideoPlayout(receive_ssrc, true, &renderer);
2247 EXPECT_TRUE(channel->renderers().begin()->second == &renderer);
2248 session_->SetVideoPlayout(receive_ssrc, false, &renderer);
2249 EXPECT_TRUE(channel->renderers().begin()->second == NULL);
2250}
2251
2252TEST_F(WebRtcSessionTest, SetVideoSend) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002253 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002254 mediastream_signaling_.SendAudioVideoStream1();
2255 CreateAndSetRemoteOfferAndLocalAnswer();
2256 cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
2257 ASSERT_TRUE(channel != NULL);
2258 ASSERT_EQ(1u, channel->send_streams().size());
2259 uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2260 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2261 cricket::VideoOptions* options = NULL;
2262 session_->SetVideoSend(send_ssrc, false, options);
2263 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
2264 session_->SetVideoSend(send_ssrc, true, options);
2265 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
2266}
2267
2268TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
2269 TestCanInsertDtmf(false);
2270}
2271
2272TEST_F(WebRtcSessionTest, CanInsertDtmf) {
2273 TestCanInsertDtmf(true);
2274}
2275
2276TEST_F(WebRtcSessionTest, InsertDtmf) {
2277 // Setup
wu@webrtc.org91053e72013-08-10 07:18:04 +00002278 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002279 mediastream_signaling_.SendAudioVideoStream1();
2280 CreateAndSetRemoteOfferAndLocalAnswer();
2281 FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
2282 EXPECT_EQ(0U, channel->dtmf_info_queue().size());
2283
2284 // Insert DTMF
2285 const int expected_flags = DF_SEND;
2286 const int expected_duration = 90;
2287 session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
2288 session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
2289 session_->InsertDtmf(kAudioTrack1, 2, expected_duration);
2290
2291 // Verify
2292 ASSERT_EQ(3U, channel->dtmf_info_queue().size());
2293 const uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
2294 EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0,
2295 expected_duration, expected_flags));
2296 EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1,
2297 expected_duration, expected_flags));
2298 EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2,
2299 expected_duration, expected_flags));
2300}
2301
2302// This test verifies the |initiator| flag when session initiates the call.
2303TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002304 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002305 EXPECT_FALSE(session_->initiator());
wu@webrtc.org91053e72013-08-10 07:18:04 +00002306 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002307 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
2308 SetLocalDescriptionWithoutError(offer);
2309 EXPECT_TRUE(session_->initiator());
2310 SetRemoteDescriptionWithoutError(answer);
2311 EXPECT_TRUE(session_->initiator());
2312}
2313
2314// This test verifies the |initiator| flag when session receives the call.
2315TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002316 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002317 EXPECT_FALSE(session_->initiator());
2318 SessionDescriptionInterface* offer = CreateRemoteOffer();
2319 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002320 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002321
2322 EXPECT_FALSE(session_->initiator());
2323 SetLocalDescriptionWithoutError(answer);
2324 EXPECT_FALSE(session_->initiator());
2325}
2326
2327// This test verifies the ice protocol type at initiator of the call
2328// if |a=ice-options:google-ice| is present in answer.
2329TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
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.org723d6832013-07-12 16:04:50 +00002333 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002334 CreateRemoteAnswer(offer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002335 SetLocalDescriptionWithoutError(offer);
2336 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 SetRemoteDescriptionWithoutError(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 initiator of the call
2350// if ICE RFC5245 is supported in answer.
2351TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
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 SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
2356 SetLocalDescriptionWithoutError(offer);
2357
2358 SetRemoteDescriptionWithoutError(answer);
2359 VerifyTransportType("audio", cricket::ICEPROTO_RFC5245);
2360 VerifyTransportType("video", cricket::ICEPROTO_RFC5245);
2361}
2362
2363// This test verifies the ice protocol type at receiver side of the call if
2364// receiver decides to use google-ice.
2365TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002366 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002367 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002368 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002369 SetRemoteDescriptionWithoutError(offer);
henrike@webrtc.org723d6832013-07-12 16:04:50 +00002370 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002371 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002372 std::string sdp;
2373 EXPECT_TRUE(answer->ToString(&sdp));
2374 // Adding ice-options to the session level.
2375 InjectAfter("t=0 0\r\n",
2376 "a=ice-options:google-ice\r\n",
2377 &sdp);
2378 SessionDescriptionInterface* answer_with_gice =
2379 CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
2380 SetLocalDescriptionWithoutError(answer_with_gice);
2381 VerifyTransportType("audio", cricket::ICEPROTO_GOOGLE);
2382 VerifyTransportType("video", cricket::ICEPROTO_GOOGLE);
2383}
2384
2385// This test verifies the ice protocol type at receiver side of the call if
2386// receiver decides to use ice RFC 5245.
2387TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002388 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002389 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002390 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002391 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002392 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002393 SetLocalDescriptionWithoutError(answer);
2394 VerifyTransportType("audio", cricket::ICEPROTO_RFC5245);
2395 VerifyTransportType("video", cricket::ICEPROTO_RFC5245);
2396}
2397
2398// This test verifies the session state when ICE RFC5245 in offer and
2399// ICE google-ice in answer.
2400TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002401 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002402 mediastream_signaling_.SendAudioVideoStream1();
2403 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002404 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002405 std::string offer_str;
2406 offer->ToString(&offer_str);
2407 // Disable google-ice
2408 const std::string gice_option = "google-ice";
2409 const std::string xgoogle_xice = "xgoogle-xice";
2410 talk_base::replace_substrs(gice_option.c_str(), gice_option.length(),
2411 xgoogle_xice.c_str(), xgoogle_xice.length(),
2412 &offer_str);
2413 JsepSessionDescription *ice_only_offer =
2414 new JsepSessionDescription(JsepSessionDescription::kOffer);
2415 EXPECT_TRUE((ice_only_offer)->Initialize(offer_str, NULL));
2416 SetLocalDescriptionWithoutError(ice_only_offer);
2417 std::string original_offer_sdp;
2418 EXPECT_TRUE(offer->ToString(&original_offer_sdp));
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002419 SessionDescriptionInterface* pranswer_with_gice =
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002420 CreateSessionDescription(JsepSessionDescription::kPrAnswer,
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002421 original_offer_sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002422 SetRemoteDescriptionPranswerExpectError(kPushDownTDFailed,
2423 pranswer_with_gice);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002424 SessionDescriptionInterface* answer_with_gice =
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002425 CreateSessionDescription(JsepSessionDescription::kAnswer,
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002426 original_offer_sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002427 SetRemoteDescriptionAnswerExpectError(kPushDownTDFailed,
2428 answer_with_gice);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002429}
2430
2431// Verifing local offer and remote answer have matching m-lines as per RFC 3264.
2432TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002433 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002434 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002435 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002436 SetLocalDescriptionWithoutError(offer);
2437 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
2438 CreateRemoteAnswer(session_->local_description()));
2439
2440 cricket::SessionDescription* answer_copy = answer->description()->Copy();
2441 answer_copy->RemoveContentByName("video");
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002442 JsepSessionDescription* modified_answer =
2443 new JsepSessionDescription(JsepSessionDescription::kAnswer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002444
2445 EXPECT_TRUE(modified_answer->Initialize(answer_copy,
2446 answer->session_id(),
2447 answer->session_version()));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002448 SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002449
2450 // Modifying content names.
2451 std::string sdp;
2452 EXPECT_TRUE(answer->ToString(&sdp));
2453 const std::string kAudioMid = "a=mid:audio";
2454 const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
2455
2456 // Replacing |audio| with |audio_content_name|.
2457 talk_base::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
2458 kAudioMidReplaceStr.c_str(),
2459 kAudioMidReplaceStr.length(),
2460 &sdp);
2461
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002462 SessionDescriptionInterface* modified_answer1 =
2463 CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002464 SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer1);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002465
2466 SetRemoteDescriptionWithoutError(answer.release());
2467}
2468
2469// Verifying remote offer and local answer have matching m-lines as per
2470// RFC 3264.
2471TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002472 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002473 mediastream_signaling_.SendAudioVideoStream1();
2474 SessionDescriptionInterface* offer = CreateRemoteOffer();
2475 SetRemoteDescriptionWithoutError(offer);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002476 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002477
2478 cricket::SessionDescription* answer_copy = answer->description()->Copy();
2479 answer_copy->RemoveContentByName("video");
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002480 JsepSessionDescription* modified_answer =
2481 new JsepSessionDescription(JsepSessionDescription::kAnswer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002482
2483 EXPECT_TRUE(modified_answer->Initialize(answer_copy,
2484 answer->session_id(),
2485 answer->session_version()));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002486 SetLocalDescriptionAnswerExpectError(kMlineMismatch, modified_answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002487 SetLocalDescriptionWithoutError(answer);
2488}
2489
2490// This test verifies that WebRtcSession does not start candidate allocation
2491// before SetLocalDescription is called.
2492TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002493 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002494 mediastream_signaling_.SendAudioVideoStream1();
2495 SessionDescriptionInterface* offer = CreateRemoteOffer();
2496 cricket::Candidate candidate;
2497 candidate.set_component(1);
2498 JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
2499 candidate);
2500 EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
2501 cricket::Candidate candidate1;
2502 candidate1.set_component(1);
2503 JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1,
2504 candidate1);
2505 EXPECT_TRUE(offer->AddCandidate(&ice_candidate1));
2506 SetRemoteDescriptionWithoutError(offer);
2507 ASSERT_TRUE(session_->GetTransportProxy("audio") != NULL);
2508 ASSERT_TRUE(session_->GetTransportProxy("video") != NULL);
2509
2510 // Pump for 1 second and verify that no candidates are generated.
2511 talk_base::Thread::Current()->ProcessMessages(1000);
2512 EXPECT_TRUE(observer_.mline_0_candidates_.empty());
2513 EXPECT_TRUE(observer_.mline_1_candidates_.empty());
2514
wu@webrtc.org91053e72013-08-10 07:18:04 +00002515 SessionDescriptionInterface* answer = CreateAnswer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002516 SetLocalDescriptionWithoutError(answer);
2517 EXPECT_TRUE(session_->GetTransportProxy("audio")->negotiated());
2518 EXPECT_TRUE(session_->GetTransportProxy("video")->negotiated());
2519 EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
2520}
2521
2522// This test verifies that crypto parameter is updated in local session
2523// description as per security policy set in MediaSessionDescriptionFactory.
2524TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002525 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002526 mediastream_signaling_.SendAudioVideoStream1();
2527 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002528 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002529
2530 // Making sure SetLocalDescription correctly sets crypto value in
2531 // SessionDescription object after de-serialization of sdp string. The value
2532 // will be set as per MediaSessionDescriptionFactory.
2533 std::string offer_str;
2534 offer->ToString(&offer_str);
2535 SessionDescriptionInterface* jsep_offer_str =
2536 CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
2537 SetLocalDescriptionWithoutError(jsep_offer_str);
2538 EXPECT_TRUE(session_->voice_channel()->secure_required());
2539 EXPECT_TRUE(session_->video_channel()->secure_required());
2540}
2541
2542// This test verifies the crypto parameter when security is disabled.
2543TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
wu@webrtc.org97077a32013-10-25 21:18:33 +00002544 options_.disable_encryption = true;
wu@webrtc.org91053e72013-08-10 07:18:04 +00002545 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002546 mediastream_signaling_.SendAudioVideoStream1();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002547 talk_base::scoped_ptr<SessionDescriptionInterface> offer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002548 CreateOffer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002549
2550 // Making sure SetLocalDescription correctly sets crypto value in
2551 // SessionDescription object after de-serialization of sdp string. The value
2552 // will be set as per MediaSessionDescriptionFactory.
2553 std::string offer_str;
2554 offer->ToString(&offer_str);
2555 SessionDescriptionInterface *jsep_offer_str =
2556 CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
2557 SetLocalDescriptionWithoutError(jsep_offer_str);
2558 EXPECT_FALSE(session_->voice_channel()->secure_required());
2559 EXPECT_FALSE(session_->video_channel()->secure_required());
2560}
2561
2562// This test verifies that an answer contains new ufrag and password if an offer
2563// with new ufrag and password is received.
2564TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002565 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002566 cricket::MediaSessionOptions options;
2567 options.has_audio = true;
2568 options.has_video = true;
2569 talk_base::scoped_ptr<JsepSessionDescription> offer(
2570 CreateRemoteOffer(options));
2571 SetRemoteDescriptionWithoutError(offer.release());
2572
2573 mediastream_signaling_.SendAudioVideoStream1();
2574 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002575 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002576 SetLocalDescriptionWithoutError(answer.release());
2577
2578 // Receive an offer with new ufrag and password.
2579 options.transport_options.ice_restart = true;
2580 talk_base::scoped_ptr<JsepSessionDescription> updated_offer1(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002581 CreateRemoteOffer(options, session_->remote_description()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002582 SetRemoteDescriptionWithoutError(updated_offer1.release());
2583
2584 talk_base::scoped_ptr<SessionDescriptionInterface> updated_answer1(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002585 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002586
2587 CompareIceUfragAndPassword(updated_answer1->description(),
2588 session_->local_description()->description(),
2589 false);
2590
2591 SetLocalDescriptionWithoutError(updated_answer1.release());
wu@webrtc.org91053e72013-08-10 07:18:04 +00002592}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002593
wu@webrtc.org91053e72013-08-10 07:18:04 +00002594// This test verifies that an answer contains old ufrag and password if an offer
2595// with old ufrag and password is received.
2596TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
2597 Init(NULL);
2598 cricket::MediaSessionOptions options;
2599 options.has_audio = true;
2600 options.has_video = true;
2601 talk_base::scoped_ptr<JsepSessionDescription> offer(
2602 CreateRemoteOffer(options));
2603 SetRemoteDescriptionWithoutError(offer.release());
2604
2605 mediastream_signaling_.SendAudioVideoStream1();
2606 talk_base::scoped_ptr<SessionDescriptionInterface> answer(
2607 CreateAnswer(NULL));
2608 SetLocalDescriptionWithoutError(answer.release());
2609
2610 // Receive an offer without changed ufrag or password.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002611 options.transport_options.ice_restart = false;
2612 talk_base::scoped_ptr<JsepSessionDescription> updated_offer2(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002613 CreateRemoteOffer(options, session_->remote_description()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002614 SetRemoteDescriptionWithoutError(updated_offer2.release());
2615
2616 talk_base::scoped_ptr<SessionDescriptionInterface> updated_answer2(
wu@webrtc.org91053e72013-08-10 07:18:04 +00002617 CreateAnswer(NULL));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002618
2619 CompareIceUfragAndPassword(updated_answer2->description(),
2620 session_->local_description()->description(),
2621 true);
2622
2623 SetLocalDescriptionWithoutError(updated_answer2.release());
2624}
2625
2626TEST_F(WebRtcSessionTest, TestSessionContentError) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002627 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002628 mediastream_signaling_.SendAudioVideoStream1();
wu@webrtc.org91053e72013-08-10 07:18:04 +00002629 SessionDescriptionInterface* offer = CreateOffer(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002630 const std::string session_id_orig = offer->session_id();
2631 const std::string session_version_orig = offer->session_version();
2632 SetLocalDescriptionWithoutError(offer);
2633
2634 video_channel_ = media_engine_->GetVideoChannel(0);
2635 video_channel_->set_fail_set_send_codecs(true);
2636
2637 mediastream_signaling_.SendAudioVideoStream2();
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00002638 SessionDescriptionInterface* answer =
2639 CreateRemoteAnswer(session_->local_description());
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002640 SetRemoteDescriptionAnswerExpectError("ERROR_CONTENT", answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002641}
2642
2643// Runs the loopback call test with BUNDLE and STUN disabled.
2644TEST_F(WebRtcSessionTest, TestIceStatesBasic) {
2645 // Lets try with only UDP ports.
2646 allocator_.set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
2647 cricket::PORTALLOCATOR_DISABLE_TCP |
2648 cricket::PORTALLOCATOR_DISABLE_STUN |
2649 cricket::PORTALLOCATOR_DISABLE_RELAY);
2650 TestLoopbackCall();
2651}
2652
mallinath@webrtc.orgd3dc4242014-03-01 00:05:52 +00002653// Runs the loopback call test with BUNDLE and STUN enabled.
mallinath@webrtc.org385857d2014-02-14 00:56:12 +00002654TEST_F(WebRtcSessionTest, TestIceStatesBundle) {
2655 allocator_.set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
2656 cricket::PORTALLOCATOR_ENABLE_BUNDLE |
mallinath@webrtc.orgd3dc4242014-03-01 00:05:52 +00002657 cricket::PORTALLOCATOR_DISABLE_TCP |
mallinath@webrtc.org385857d2014-02-14 00:56:12 +00002658 cricket::PORTALLOCATOR_DISABLE_RELAY);
2659 TestLoopbackCall();
2660}
2661
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002662TEST_F(WebRtcSessionTest, SetSdpFailedOnSessionError) {
wu@webrtc.org91053e72013-08-10 07:18:04 +00002663 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002664 cricket::MediaSessionOptions options;
2665 options.has_audio = true;
2666 options.has_video = true;
2667
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002668 cricket::BaseSession::Error error_code = cricket::BaseSession::ERROR_CONTENT;
2669 std::string error_code_str = "ERROR_CONTENT";
2670 std::string error_desc = "Fake session error description.";
2671 session_->SetError(error_code, error_desc);
2672
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002673 SessionDescriptionInterface* offer = CreateRemoteOffer(options);
2674 SessionDescriptionInterface* answer =
2675 CreateRemoteAnswer(offer, options);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002676
2677 std::string action;
2678 std::ostringstream session_error_msg;
2679 session_error_msg << kSessionError << error_code_str << ". ";
2680 session_error_msg << kSessionErrorDesc << error_desc << ".";
2681 SetRemoteDescriptionExpectError(action, session_error_msg.str(), offer);
2682 SetLocalDescriptionExpectError(action, session_error_msg.str(), answer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002683}
2684
2685TEST_F(WebRtcSessionTest, TestRtpDataChannel) {
2686 constraints_.reset(new FakeConstraints());
2687 constraints_->AddOptional(
2688 webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002689 Init(NULL);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002690
2691 SetLocalDescriptionWithDataChannel();
2692 EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
2693}
2694
2695TEST_F(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
2696 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2697
2698 constraints_.reset(new FakeConstraints());
2699 constraints_->AddOptional(
2700 webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
wu@webrtc.org97077a32013-10-25 21:18:33 +00002701 options_.disable_sctp_data_channels = false;
2702
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002703 InitWithDtls(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002704
2705 SetLocalDescriptionWithDataChannel();
2706 EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
2707}
2708
wu@webrtc.org967bfff2013-09-19 05:49:50 +00002709TEST_F(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
2710 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2711
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002712 InitWithDtls(false);
2713
2714 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2715 EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +00002716 EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL);
2717}
2718
2719TEST_F(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
2720 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2721 SetFactoryDtlsSrtp();
mallinath@webrtc.org1112c302013-09-23 20:34:45 +00002722 InitWithDtls(false);
2723
2724 // Create remote offer with SCTP.
2725 cricket::MediaSessionOptions options;
2726 options.data_channel_type = cricket::DCT_SCTP;
2727 JsepSessionDescription* offer =
2728 CreateRemoteOffer(options, cricket::SEC_ENABLED);
2729 SetRemoteDescriptionWithoutError(offer);
2730
2731 // Verifies the answer contains SCTP.
2732 talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
2733 EXPECT_TRUE(answer != NULL);
2734 EXPECT_TRUE(answer->description()->GetContentByName("data") != NULL);
2735 EXPECT_TRUE(answer->description()->GetTransportInfoByName("data") != NULL);
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002736}
2737
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002738TEST_F(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) {
2739 constraints_.reset(new FakeConstraints());
2740 constraints_->AddOptional(
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002741 webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
2742 InitWithDtls(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002743
2744 SetLocalDescriptionWithDataChannel();
2745 EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
2746}
2747
2748TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
2749 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2750
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +00002751 InitWithDtls(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002752
2753 SetLocalDescriptionWithDataChannel();
2754 EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
2755}
wu@webrtc.org91053e72013-08-10 07:18:04 +00002756
wu@webrtc.org97077a32013-10-25 21:18:33 +00002757TEST_F(WebRtcSessionTest, TestDisableSctpDataChannels) {
2758 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2759 options_.disable_sctp_data_channels = true;
2760 InitWithDtls(false);
2761
2762 SetLocalDescriptionWithDataChannel();
2763 EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
2764}
2765
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002766TEST_F(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
2767 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2768 const int new_send_port = 9998;
2769 const int new_recv_port = 7775;
2770
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002771 InitWithDtls(false);
2772 SetFactoryDtlsSrtp();
2773
2774 // By default, don't actually add the codecs to desc_factory_; they don't
2775 // actually get serialized for SCTP in BuildMediaDescription(). Instead,
2776 // let the session description get parsed. That'll get the proper codecs
2777 // into the stream.
2778 cricket::MediaSessionOptions options;
2779 JsepSessionDescription* offer = CreateRemoteOfferWithSctpPort(
2780 "stream1", new_send_port, options);
2781
2782 // SetRemoteDescription will take the ownership of the offer.
2783 SetRemoteDescriptionWithoutError(offer);
2784
2785 SessionDescriptionInterface* answer = ChangeSDPSctpPort(
2786 new_recv_port, CreateAnswer(NULL));
2787 ASSERT_TRUE(answer != NULL);
2788
2789 // Now set the local description, which'll take ownership of the answer.
2790 SetLocalDescriptionWithoutError(answer);
2791
2792 // TEST PLAN: Set the port number to something new, set it in the SDP,
2793 // and pass it all the way down.
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +00002794 webrtc::InternalDataChannelInit dci;
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002795 dci.reliable = true;
2796 EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
2797 talk_base::scoped_refptr<webrtc::DataChannel> dc =
2798 session_->CreateDataChannel("datachannel", &dci);
2799
2800 cricket::FakeDataMediaChannel* ch = data_engine_->GetChannel(0);
2801 int portnum = -1;
2802 ASSERT_TRUE(ch != NULL);
2803 ASSERT_EQ(1UL, ch->send_codecs().size());
2804 EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->send_codecs()[0].id);
2805 EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
2806 ch->send_codecs()[0].name.c_str()));
2807 EXPECT_TRUE(ch->send_codecs()[0].GetParam(cricket::kCodecParamPort,
2808 &portnum));
2809 EXPECT_EQ(new_send_port, portnum);
2810
2811 ASSERT_EQ(1UL, ch->recv_codecs().size());
2812 EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->recv_codecs()[0].id);
2813 EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
2814 ch->recv_codecs()[0].name.c_str()));
2815 EXPECT_TRUE(ch->recv_codecs()[0].GetParam(cricket::kCodecParamPort,
2816 &portnum));
2817 EXPECT_EQ(new_recv_port, portnum);
2818}
2819
wu@webrtc.org91053e72013-08-10 07:18:04 +00002820// Verifies that CreateOffer succeeds when CreateOffer is called before async
2821// identity generation is finished.
2822TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
2823 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002824 InitWithDtls(false);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002825
2826 EXPECT_TRUE(session_->waiting_for_identity());
2827 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2828 EXPECT_TRUE(offer != NULL);
2829}
2830
2831// Verifies that CreateAnswer succeeds when CreateOffer is called before async
2832// identity generation is finished.
2833TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
2834 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002835 InitWithDtls(false);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002836
2837 cricket::MediaSessionOptions options;
2838 scoped_ptr<JsepSessionDescription> offer(
2839 CreateRemoteOffer(options, cricket::SEC_REQUIRED));
2840 ASSERT_TRUE(offer.get() != NULL);
2841 SetRemoteDescriptionWithoutError(offer.release());
2842
2843 talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
2844 EXPECT_TRUE(answer != NULL);
2845}
2846
2847// Verifies that CreateOffer succeeds when CreateOffer is called after async
2848// identity generation is finished.
2849TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
2850 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002851 InitWithDtls(false);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002852
2853 EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
2854 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2855 EXPECT_TRUE(offer != NULL);
2856}
2857
2858// Verifies that CreateOffer fails when CreateOffer is called after async
2859// identity generation fails.
2860TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
2861 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
henrike@webrtc.org7666db72013-08-22 14:45:42 +00002862 InitWithDtls(true);
wu@webrtc.org91053e72013-08-10 07:18:04 +00002863
2864 EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
2865 talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
2866 EXPECT_TRUE(offer == NULL);
2867}
2868
2869// Verifies that CreateOffer succeeds when Multiple CreateOffer calls are made
2870// before async identity generation is finished.
2871TEST_F(WebRtcSessionTest,
2872 TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
2873 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2874 VerifyMultipleAsyncCreateDescription(
2875 true, CreateSessionDescriptionRequest::kOffer);
2876}
2877
2878// Verifies that CreateOffer fails when Multiple CreateOffer calls are made
2879// before async identity generation fails.
2880TEST_F(WebRtcSessionTest,
2881 TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) {
2882 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2883 VerifyMultipleAsyncCreateDescription(
2884 false, CreateSessionDescriptionRequest::kOffer);
2885}
2886
2887// Verifies that CreateAnswer succeeds when Multiple CreateAnswer calls are made
2888// before async identity generation is finished.
2889TEST_F(WebRtcSessionTest,
2890 TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) {
2891 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2892 VerifyMultipleAsyncCreateDescription(
2893 true, CreateSessionDescriptionRequest::kAnswer);
2894}
2895
2896// Verifies that CreateAnswer fails when Multiple CreateAnswer calls are made
2897// before async identity generation fails.
2898TEST_F(WebRtcSessionTest,
2899 TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) {
2900 MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
2901 VerifyMultipleAsyncCreateDescription(
2902 false, CreateSessionDescriptionRequest::kAnswer);
2903}
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +00002904
2905// Verifies that setRemoteDescription fails when DTLS is disabled and the remote
2906// offer has no SDES crypto but only DTLS fingerprint.
2907TEST_F(WebRtcSessionTest, TestSetRemoteOfferFailIfDtlsDisabledAndNoCrypto) {
2908 // Init without DTLS.
2909 Init(NULL);
2910 // Create a remote offer with secured transport disabled.
2911 cricket::MediaSessionOptions options;
2912 JsepSessionDescription* offer(CreateRemoteOffer(
2913 options, cricket::SEC_DISABLED));
2914 // Adds a DTLS fingerprint to the remote offer.
2915 cricket::SessionDescription* sdp = offer->description();
2916 TransportInfo* audio = sdp->GetTransportInfoByName("audio");
2917 ASSERT_TRUE(audio != NULL);
2918 ASSERT_TRUE(audio->description.identity_fingerprint.get() == NULL);
2919 audio->description.identity_fingerprint.reset(
2920 talk_base::SSLFingerprint::CreateFromRfc4572(
2921 talk_base::DIGEST_SHA_256, kFakeDtlsFingerprint));
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002922 SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesAndDtlsDisabled,
2923 offer);
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +00002924}
2925
wu@webrtc.orgde305012013-10-31 15:40:38 +00002926// This test verifies DSCP is properly applied on the media channels.
2927TEST_F(WebRtcSessionTest, TestDscpConstraint) {
2928 constraints_.reset(new FakeConstraints());
2929 constraints_->AddOptional(
2930 webrtc::MediaConstraintsInterface::kEnableDscp, true);
2931 Init(NULL);
2932 mediastream_signaling_.SendAudioVideoStream1();
2933 SessionDescriptionInterface* offer = CreateOffer(NULL);
2934
2935 SetLocalDescriptionWithoutError(offer);
2936
2937 video_channel_ = media_engine_->GetVideoChannel(0);
2938 voice_channel_ = media_engine_->GetVoiceChannel(0);
2939
2940 ASSERT_TRUE(video_channel_ != NULL);
2941 ASSERT_TRUE(voice_channel_ != NULL);
2942 cricket::AudioOptions audio_options;
2943 EXPECT_TRUE(voice_channel_->GetOptions(&audio_options));
2944 cricket::VideoOptions video_options;
2945 EXPECT_TRUE(video_channel_->GetOptions(&video_options));
2946 EXPECT_TRUE(audio_options.dscp.IsSet());
2947 EXPECT_TRUE(audio_options.dscp.GetWithDefaultIfUnset(false));
2948 EXPECT_TRUE(video_options.dscp.IsSet());
2949 EXPECT_TRUE(video_options.dscp.GetWithDefaultIfUnset(false));
2950}
2951
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002952// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
2953// currently fails because upon disconnection and reconnection OnIceComplete is
2954// called more than once without returning to IceGatheringGathering.