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webrtc/talk/app/webrtc/webrtcsession_unittest.cc
wu@webrtc.org 364f204d16 Update talk to 56698267. 
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/4119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 21:49:41 +00:00

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/*
* libjingle
* Copyright 2012, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/videotrack.h"
#include "talk/app/webrtc/test/fakeconstraints.h"
#include "talk/app/webrtc/test/fakedtlsidentityservice.h"
#include "talk/app/webrtc/test/fakemediastreamsignaling.h"
#include "talk/app/webrtc/webrtcsession.h"
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
#include "talk/base/fakenetwork.h"
#include "talk/base/firewallsocketserver.h"
#include "talk/base/gunit.h"
#include "talk/base/logging.h"
#include "talk/base/network.h"
#include "talk/base/physicalsocketserver.h"
#include "talk/base/ssladapter.h"
#include "talk/base/sslstreamadapter.h"
#include "talk/base/stringutils.h"
#include "talk/base/thread.h"
#include "talk/base/virtualsocketserver.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/fakevideorenderer.h"
#include "talk/media/base/mediachannel.h"
#include "talk/media/devices/fakedevicemanager.h"
#include "talk/p2p/base/stunserver.h"
#include "talk/p2p/base/teststunserver.h"
#include "talk/p2p/client/basicportallocator.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using cricket::BaseSession;
using cricket::DF_PLAY;
using cricket::DF_SEND;
using cricket::FakeVoiceMediaChannel;
using cricket::NS_GINGLE_P2P;
using cricket::NS_JINGLE_ICE_UDP;
using cricket::TransportInfo;
using talk_base::SocketAddress;
using talk_base::scoped_ptr;
using webrtc::CreateSessionDescription;
using webrtc::CreateSessionDescriptionObserver;
using webrtc::CreateSessionDescriptionRequest;
using webrtc::DTLSIdentityRequestObserver;
using webrtc::DTLSIdentityServiceInterface;
using webrtc::FakeConstraints;
using webrtc::IceCandidateCollection;
using webrtc::JsepIceCandidate;
using webrtc::JsepSessionDescription;
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollection;
using webrtc::WebRtcSession;
using webrtc::kBundleWithoutRtcpMux;
using webrtc::kMlineMismatch;
using webrtc::kPushDownAnswerTDFailed;
using webrtc::kPushDownPranswerTDFailed;
using webrtc::kSdpWithoutCrypto;
using webrtc::kSdpWithoutIceUfragPwd;
using webrtc::kSdpWithoutSdesAndDtlsDisabled;
using webrtc::kSessionError;
using webrtc::kSetLocalSdpFailed;
using webrtc::kSetRemoteSdpFailed;
static const int kClientAddrPort = 0;
static const char kClientAddrHost1[] = "11.11.11.11";
static const char kClientAddrHost2[] = "22.22.22.22";
static const char kStunAddrHost[] = "99.99.99.1";
static const char kSessionVersion[] = "1";
// Media index of candidates belonging to the first media content.
static const int kMediaContentIndex0 = 0;
static const char kMediaContentName0[] = "audio";
// Media index of candidates belonging to the second media content.
static const int kMediaContentIndex1 = 1;
static const char kMediaContentName1[] = "video";
static const int kIceCandidatesTimeout = 10000;
static const char kFakeDtlsFingerprint[] =
"BB:CD:72:F7:2F:D0:BA:43:F3:68:B1:0C:23:72:B6:4A:"
"0F:DE:34:06:BC:E0:FE:01:BC:73:C8:6D:F4:65:D5:24";
// Add some extra |newlines| to the |message| after |line|.
static void InjectAfter(const std::string& line,
const std::string& newlines,
std::string* message) {
const std::string tmp = line + newlines;
talk_base::replace_substrs(line.c_str(), line.length(),
tmp.c_str(), tmp.length(), message);
}
class MockIceObserver : public webrtc::IceObserver {
public:
MockIceObserver()
: oncandidatesready_(false),
ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
ice_gathering_state_(PeerConnectionInterface::kIceGatheringNew) {
}
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
ice_connection_state_ = new_state;
}
virtual void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
// We can never transition back to "new".
EXPECT_NE(PeerConnectionInterface::kIceGatheringNew, new_state);
ice_gathering_state_ = new_state;
// oncandidatesready_ really means "ICE gathering is complete".
// This if statement ensures that this value remains correct when we
// transition from kIceGatheringComplete to kIceGatheringGathering.
if (new_state == PeerConnectionInterface::kIceGatheringGathering) {
oncandidatesready_ = false;
}
}
// Found a new candidate.
virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
switch (candidate->sdp_mline_index()) {
case kMediaContentIndex0:
mline_0_candidates_.push_back(candidate->candidate());
break;
case kMediaContentIndex1:
mline_1_candidates_.push_back(candidate->candidate());
break;
default:
ASSERT(false);
}
// The ICE gathering state should always be Gathering when a candidate is
// received (or possibly Completed in the case of the final candidate).
EXPECT_NE(PeerConnectionInterface::kIceGatheringNew, ice_gathering_state_);
}
// TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
virtual void OnIceComplete() {
EXPECT_FALSE(oncandidatesready_);
oncandidatesready_ = true;
// OnIceGatheringChange(IceGatheringCompleted) and OnIceComplete() should
// be called approximately simultaneously. For ease of testing, this
// check additionally requires that they be called in the above order.
EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
ice_gathering_state_);
}
bool oncandidatesready_;
std::vector<cricket::Candidate> mline_0_candidates_;
std::vector<cricket::Candidate> mline_1_candidates_;
PeerConnectionInterface::IceConnectionState ice_connection_state_;
PeerConnectionInterface::IceGatheringState ice_gathering_state_;
};
class WebRtcSessionForTest : public webrtc::WebRtcSession {
public:
WebRtcSessionForTest(cricket::ChannelManager* cmgr,
talk_base::Thread* signaling_thread,
talk_base::Thread* worker_thread,
cricket::PortAllocator* port_allocator,
webrtc::IceObserver* ice_observer,
webrtc::MediaStreamSignaling* mediastream_signaling)
: WebRtcSession(cmgr, signaling_thread, worker_thread, port_allocator,
mediastream_signaling) {
RegisterIceObserver(ice_observer);
}
virtual ~WebRtcSessionForTest() {}
using cricket::BaseSession::GetTransportProxy;
using webrtc::WebRtcSession::SetAudioPlayout;
using webrtc::WebRtcSession::SetAudioSend;
using webrtc::WebRtcSession::SetCaptureDevice;
using webrtc::WebRtcSession::SetVideoPlayout;
using webrtc::WebRtcSession::SetVideoSend;
};
class WebRtcSessionCreateSDPObserverForTest
: public talk_base::RefCountedObject<CreateSessionDescriptionObserver> {
public:
enum State {
kInit,
kFailed,
kSucceeded,
};
WebRtcSessionCreateSDPObserverForTest() : state_(kInit) {}
// CreateSessionDescriptionObserver implementation.
virtual void OnSuccess(SessionDescriptionInterface* desc) {
description_.reset(desc);
state_ = kSucceeded;
}
virtual void OnFailure(const std::string& error) {
state_ = kFailed;
}
SessionDescriptionInterface* description() { return description_.get(); }
SessionDescriptionInterface* ReleaseDescription() {
return description_.release();
}
State state() const { return state_; }
protected:
~WebRtcSessionCreateSDPObserverForTest() {}
private:
talk_base::scoped_ptr<SessionDescriptionInterface> description_;
State state_;
};
class FakeAudioRenderer : public cricket::AudioRenderer {
public:
FakeAudioRenderer() : channel_id_(-1) {}
virtual void AddChannel(int channel_id) OVERRIDE {
ASSERT(channel_id_ == -1);
channel_id_ = channel_id;
}
virtual void RemoveChannel(int channel_id) OVERRIDE {
ASSERT(channel_id == channel_id_);
channel_id_ = -1;
}
int channel_id() const { return channel_id_; }
private:
int channel_id_;
};
class WebRtcSessionTest : public testing::Test {
protected:
// TODO Investigate why ChannelManager crashes, if it's created
// after stun_server.
WebRtcSessionTest()
: media_engine_(new cricket::FakeMediaEngine()),
data_engine_(new cricket::FakeDataEngine()),
device_manager_(new cricket::FakeDeviceManager()),
channel_manager_(new cricket::ChannelManager(
media_engine_, data_engine_, device_manager_,
new cricket::CaptureManager(), talk_base::Thread::Current())),
tdesc_factory_(new cricket::TransportDescriptionFactory()),
desc_factory_(new cricket::MediaSessionDescriptionFactory(
channel_manager_.get(), tdesc_factory_.get())),
pss_(new talk_base::PhysicalSocketServer),
vss_(new talk_base::VirtualSocketServer(pss_.get())),
fss_(new talk_base::FirewallSocketServer(vss_.get())),
ss_scope_(fss_.get()),
stun_socket_addr_(talk_base::SocketAddress(kStunAddrHost,
cricket::STUN_SERVER_PORT)),
stun_server_(talk_base::Thread::Current(), stun_socket_addr_),
allocator_(&network_manager_, stun_socket_addr_,
SocketAddress(), SocketAddress(), SocketAddress()),
mediastream_signaling_(channel_manager_.get()) {
tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);
allocator_.set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
cricket::PORTALLOCATOR_DISABLE_RELAY |
cricket::PORTALLOCATOR_ENABLE_BUNDLE);
EXPECT_TRUE(channel_manager_->Init());
desc_factory_->set_add_legacy_streams(false);
}
static void SetUpTestCase() {
talk_base::InitializeSSL();
}
static void TearDownTestCase() {
talk_base::CleanupSSL();
}
void AddInterface(const SocketAddress& addr) {
network_manager_.AddInterface(addr);
}
void Init(DTLSIdentityServiceInterface* identity_service) {
ASSERT_TRUE(session_.get() == NULL);
session_.reset(new WebRtcSessionForTest(
channel_manager_.get(), talk_base::Thread::Current(),
talk_base::Thread::Current(), &allocator_,
&observer_,
&mediastream_signaling_));
EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
observer_.ice_connection_state_);
EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
observer_.ice_gathering_state_);
EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
identity_service));
}
void InitWithDtmfCodec() {
// Add kTelephoneEventCodec for dtmf test.
const cricket::AudioCodec kTelephoneEventCodec(
106, "telephone-event", 8000, 0, 1, 0);
std::vector<cricket::AudioCodec> codecs;
codecs.push_back(kTelephoneEventCodec);
media_engine_->SetAudioCodecs(codecs);
desc_factory_->set_audio_codecs(codecs);
Init(NULL);
}
void InitWithDtls(bool identity_request_should_fail = false) {
FakeIdentityService* identity_service = new FakeIdentityService();
identity_service->set_should_fail(identity_request_should_fail);
Init(identity_service);
}
// Creates a local offer and applies it. Starts ice.
// Call mediastream_signaling_.UseOptionsWithStreamX() before this function
// to decide which streams to create.
void InitiateCall() {
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
EXPECT_TRUE_WAIT(PeerConnectionInterface::kIceGatheringNew !=
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
}
SessionDescriptionInterface* CreateOffer(
const webrtc::MediaConstraintsInterface* constraints) {
talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
observer = new WebRtcSessionCreateSDPObserverForTest();
session_->CreateOffer(observer, constraints);
EXPECT_TRUE_WAIT(
observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
2000);
return observer->ReleaseDescription();
}
SessionDescriptionInterface* CreateAnswer(
const webrtc::MediaConstraintsInterface* constraints) {
talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observer
= new WebRtcSessionCreateSDPObserverForTest();
session_->CreateAnswer(observer, constraints);
EXPECT_TRUE_WAIT(
observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
2000);
return observer->ReleaseDescription();
}
bool ChannelsExist() const {
return (session_->voice_channel() != NULL &&
session_->video_channel() != NULL);
}
void CheckTransportChannels() const {
EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 1) != NULL);
EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 2) != NULL);
EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 1) != NULL);
EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 2) != NULL);
}
void VerifyCryptoParams(const cricket::SessionDescription* sdp) {
ASSERT_TRUE(session_.get() != NULL);
const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
ASSERT_TRUE(content != NULL);
const cricket::AudioContentDescription* audio_content =
static_cast<const cricket::AudioContentDescription*>(
content->description);
ASSERT_TRUE(audio_content != NULL);
ASSERT_EQ(1U, audio_content->cryptos().size());
ASSERT_EQ(47U, audio_content->cryptos()[0].key_params.size());
ASSERT_EQ("AES_CM_128_HMAC_SHA1_80",
audio_content->cryptos()[0].cipher_suite);
EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
audio_content->protocol());
content = cricket::GetFirstVideoContent(sdp);
ASSERT_TRUE(content != NULL);
const cricket::VideoContentDescription* video_content =
static_cast<const cricket::VideoContentDescription*>(
content->description);
ASSERT_TRUE(video_content != NULL);
ASSERT_EQ(1U, video_content->cryptos().size());
ASSERT_EQ("AES_CM_128_HMAC_SHA1_80",
video_content->cryptos()[0].cipher_suite);
ASSERT_EQ(47U, video_content->cryptos()[0].key_params.size());
EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
video_content->protocol());
}
void VerifyNoCryptoParams(const cricket::SessionDescription* sdp, bool dtls) {
const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
ASSERT_TRUE(content != NULL);
const cricket::AudioContentDescription* audio_content =
static_cast<const cricket::AudioContentDescription*>(
content->description);
ASSERT_TRUE(audio_content != NULL);
ASSERT_EQ(0U, audio_content->cryptos().size());
content = cricket::GetFirstVideoContent(sdp);
ASSERT_TRUE(content != NULL);
const cricket::VideoContentDescription* video_content =
static_cast<const cricket::VideoContentDescription*>(
content->description);
ASSERT_TRUE(video_content != NULL);
ASSERT_EQ(0U, video_content->cryptos().size());
if (dtls) {
EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
audio_content->protocol());
EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
video_content->protocol());
} else {
EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf),
audio_content->protocol());
EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf),
video_content->protocol());
}
}
// Set the internal fake description factories to do DTLS-SRTP.
void SetFactoryDtlsSrtp() {
desc_factory_->set_secure(cricket::SEC_ENABLED);
std::string identity_name = "WebRTC" +
talk_base::ToString(talk_base::CreateRandomId());
identity_.reset(talk_base::SSLIdentity::Generate(identity_name));
tdesc_factory_->set_identity(identity_.get());
tdesc_factory_->set_secure(cricket::SEC_REQUIRED);
}
void VerifyFingerprintStatus(const cricket::SessionDescription* sdp,
bool expected) {
const TransportInfo* audio = sdp->GetTransportInfoByName("audio");
ASSERT_TRUE(audio != NULL);
ASSERT_EQ(expected, audio->description.identity_fingerprint.get() != NULL);
const TransportInfo* video = sdp->GetTransportInfoByName("video");
ASSERT_TRUE(video != NULL);
ASSERT_EQ(expected, video->description.identity_fingerprint.get() != NULL);
}
void VerifyAnswerFromNonCryptoOffer() {
// Create a SDP without Crypto.
cricket::MediaSessionOptions options;
options.has_video = true;
JsepSessionDescription* offer(
CreateRemoteOffer(options, cricket::SEC_DISABLED));
ASSERT_TRUE(offer != NULL);
VerifyNoCryptoParams(offer->description(), false);
SetRemoteDescriptionExpectError("Called with a SDP without crypto enabled",
offer);
const webrtc::SessionDescriptionInterface* answer = CreateAnswer(NULL);
// Answer should be NULL as no crypto params in offer.
ASSERT_TRUE(answer == NULL);
}
void VerifyAnswerFromCryptoOffer() {
cricket::MediaSessionOptions options;
options.has_video = true;
options.bundle_enabled = true;
scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options, cricket::SEC_REQUIRED));
ASSERT_TRUE(offer.get() != NULL);
VerifyCryptoParams(offer->description());
SetRemoteDescriptionWithoutError(offer.release());
scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
ASSERT_TRUE(answer.get() != NULL);
VerifyCryptoParams(answer->description());
}
void CompareIceUfragAndPassword(const cricket::SessionDescription* desc1,
const cricket::SessionDescription* desc2,
bool expect_equal) {
if (desc1->contents().size() != desc2->contents().size()) {
EXPECT_FALSE(expect_equal);
return;
}
const cricket::ContentInfos& contents = desc1->contents();
cricket::ContentInfos::const_iterator it = contents.begin();
for (; it != contents.end(); ++it) {
const cricket::TransportDescription* transport_desc1 =
desc1->GetTransportDescriptionByName(it->name);
const cricket::TransportDescription* transport_desc2 =
desc2->GetTransportDescriptionByName(it->name);
if (!transport_desc1 || !transport_desc2) {
EXPECT_FALSE(expect_equal);
return;
}
if (transport_desc1->ice_pwd != transport_desc2->ice_pwd ||
transport_desc1->ice_ufrag != transport_desc2->ice_ufrag) {
EXPECT_FALSE(expect_equal);
return;
}
}
EXPECT_TRUE(expect_equal);
}
void RemoveIceUfragPwdLines(const SessionDescriptionInterface* current_desc,
std::string *sdp) {
const cricket::SessionDescription* desc = current_desc->description();
EXPECT_TRUE(current_desc->ToString(sdp));
const cricket::ContentInfos& contents = desc->contents();
cricket::ContentInfos::const_iterator it = contents.begin();
// Replace ufrag and pwd lines with empty strings.
for (; it != contents.end(); ++it) {
const cricket::TransportDescription* transport_desc =
desc->GetTransportDescriptionByName(it->name);
std::string ufrag_line = "a=ice-ufrag:" + transport_desc->ice_ufrag
+ "\r\n";
std::string pwd_line = "a=ice-pwd:" + transport_desc->ice_pwd
+ "\r\n";
talk_base::replace_substrs(ufrag_line.c_str(), ufrag_line.length(),
"", 0,
sdp);
talk_base::replace_substrs(pwd_line.c_str(), pwd_line.length(),
"", 0,
sdp);
}
}
// Creates a remote offer and and applies it as a remote description,
// creates a local answer and applies is as a local description.
// Call mediastream_signaling_.UseOptionsWithStreamX() before this function
// to decide which local and remote streams to create.
void CreateAndSetRemoteOfferAndLocalAnswer() {
SessionDescriptionInterface* offer = CreateRemoteOffer();
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
}
void SetLocalDescriptionWithoutError(SessionDescriptionInterface* desc) {
EXPECT_TRUE(session_->SetLocalDescription(desc, NULL));
}
void SetLocalDescriptionExpectState(SessionDescriptionInterface* desc,
BaseSession::State expected_state) {
SetLocalDescriptionWithoutError(desc);
EXPECT_EQ(expected_state, session_->state());
}
void SetLocalDescriptionExpectError(const std::string& expected_error,
SessionDescriptionInterface* desc) {
std::string error;
EXPECT_FALSE(session_->SetLocalDescription(desc, &error));
EXPECT_NE(std::string::npos, error.find(kSetLocalSdpFailed));
EXPECT_NE(std::string::npos, error.find(expected_error));
}
void SetRemoteDescriptionWithoutError(SessionDescriptionInterface* desc) {
EXPECT_TRUE(session_->SetRemoteDescription(desc, NULL));
}
void SetRemoteDescriptionExpectState(SessionDescriptionInterface* desc,
BaseSession::State expected_state) {
SetRemoteDescriptionWithoutError(desc);
EXPECT_EQ(expected_state, session_->state());
}
void SetRemoteDescriptionExpectError(const std::string& expected_error,
SessionDescriptionInterface* desc) {
std::string error;
EXPECT_FALSE(session_->SetRemoteDescription(desc, &error));
EXPECT_NE(std::string::npos, error.find(kSetRemoteSdpFailed));
EXPECT_NE(std::string::npos, error.find(expected_error));
}
void CreateCryptoOfferAndNonCryptoAnswer(SessionDescriptionInterface** offer,
SessionDescriptionInterface** nocrypto_answer) {
// Create a SDP without Crypto.
cricket::MediaSessionOptions options;
options.has_video = true;
options.bundle_enabled = true;
*offer = CreateRemoteOffer(options, cricket::SEC_ENABLED);
ASSERT_TRUE(*offer != NULL);
VerifyCryptoParams((*offer)->description());
*nocrypto_answer = CreateRemoteAnswer(*offer, options,
cricket::SEC_DISABLED);
EXPECT_TRUE(*nocrypto_answer != NULL);
}
JsepSessionDescription* CreateRemoteOfferWithVersion(
cricket::MediaSessionOptions options,
cricket::SecurePolicy secure_policy,
const std::string& session_version,
const SessionDescriptionInterface* current_desc) {
std::string session_id = talk_base::ToString(talk_base::CreateRandomId64());
const cricket::SessionDescription* cricket_desc = NULL;
if (current_desc) {
cricket_desc = current_desc->description();
session_id = current_desc->session_id();
}
desc_factory_->set_secure(secure_policy);
JsepSessionDescription* offer(
new JsepSessionDescription(JsepSessionDescription::kOffer));
if (!offer->Initialize(desc_factory_->CreateOffer(options, cricket_desc),
session_id, session_version)) {
delete offer;
offer = NULL;
}
return offer;
}
JsepSessionDescription* CreateRemoteOffer(
cricket::MediaSessionOptions options) {
return CreateRemoteOfferWithVersion(options, cricket::SEC_ENABLED,
kSessionVersion, NULL);
}
JsepSessionDescription* CreateRemoteOffer(
cricket::MediaSessionOptions options, cricket::SecurePolicy policy) {
return CreateRemoteOfferWithVersion(options, policy, kSessionVersion, NULL);
}
JsepSessionDescription* CreateRemoteOffer(
cricket::MediaSessionOptions options,
const SessionDescriptionInterface* current_desc) {
return CreateRemoteOfferWithVersion(options, cricket::SEC_ENABLED,
kSessionVersion, current_desc);
}
JsepSessionDescription* CreateRemoteOfferWithSctpPort(
const char* sctp_stream_name, int new_port,
cricket::MediaSessionOptions options) {
options.data_channel_type = cricket::DCT_SCTP;
options.AddStream(cricket::MEDIA_TYPE_DATA, "datachannel",
sctp_stream_name);
return ChangeSDPSctpPort(new_port, CreateRemoteOffer(options));
}
// Takes ownership of offer_basis (and deletes it).
JsepSessionDescription* ChangeSDPSctpPort(
int new_port, webrtc::SessionDescriptionInterface *offer_basis) {
// Stringify the input SDP, swap the 5000 for 'new_port' and create a new
// SessionDescription from the mutated string.
const char* default_port_str = "5000";
char new_port_str[16];
talk_base::sprintfn(new_port_str, sizeof(new_port_str), "%d", new_port);
std::string offer_str;
offer_basis->ToString(&offer_str);
talk_base::replace_substrs(default_port_str, strlen(default_port_str),
new_port_str, strlen(new_port_str),
&offer_str);
JsepSessionDescription* offer = new JsepSessionDescription(
offer_basis->type());
delete offer_basis;
offer->Initialize(offer_str, NULL);
return offer;
}
// Create a remote offer. Call mediastream_signaling_.UseOptionsWithStreamX()
// before this function to decide which streams to create.
JsepSessionDescription* CreateRemoteOffer() {
cricket::MediaSessionOptions options;
mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
return CreateRemoteOffer(options, session_->remote_description());
}
JsepSessionDescription* CreateRemoteAnswer(
const SessionDescriptionInterface* offer,
cricket::MediaSessionOptions options,
cricket::SecurePolicy policy) {
desc_factory_->set_secure(policy);
const std::string session_id =
talk_base::ToString(talk_base::CreateRandomId64());
JsepSessionDescription* answer(
new JsepSessionDescription(JsepSessionDescription::kAnswer));
if (!answer->Initialize(desc_factory_->CreateAnswer(offer->description(),
options, NULL),
session_id, kSessionVersion)) {
delete answer;
answer = NULL;
}
return answer;
}
JsepSessionDescription* CreateRemoteAnswer(
const SessionDescriptionInterface* offer,
cricket::MediaSessionOptions options) {
return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
}
// Creates an answer session description with streams based on
// |mediastream_signaling_|. Call
// mediastream_signaling_.UseOptionsWithStreamX() before this function
// to decide which streams to create.
JsepSessionDescription* CreateRemoteAnswer(
const SessionDescriptionInterface* offer) {
cricket::MediaSessionOptions options;
mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
}
void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) {
AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
FakeConstraints constraints;
constraints.SetMandatoryUseRtpMux(bundle);
SessionDescriptionInterface* offer = CreateOffer(&constraints);
// SetLocalDescription and SetRemoteDescriptions takes ownership of offer
// and answer.
SetLocalDescriptionWithoutError(offer);
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateRemoteAnswer(session_->local_description()));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
size_t expected_candidate_num = 2;
if (!rtcp_mux) {
// If rtcp_mux is enabled we should expect 4 candidates - host and srflex
// for rtp and rtcp.
expected_candidate_num = 4;
// Disable rtcp-mux from the answer
const std::string kRtcpMux = "a=rtcp-mux";
const std::string kXRtcpMux = "a=xrtcp-mux";
talk_base::replace_substrs(kRtcpMux.c_str(), kRtcpMux.length(),
kXRtcpMux.c_str(), kXRtcpMux.length(),
&sdp);
}
SessionDescriptionInterface* new_answer = CreateSessionDescription(
JsepSessionDescription::kAnswer, sdp, NULL);
// SetRemoteDescription to enable rtcp mux.
SetRemoteDescriptionWithoutError(new_answer);
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ(expected_candidate_num, observer_.mline_0_candidates_.size());
EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
cricket::Candidate c0 = observer_.mline_0_candidates_[i];
cricket::Candidate c1 = observer_.mline_1_candidates_[i];
if (bundle) {
EXPECT_TRUE(c0.IsEquivalent(c1));
} else {
EXPECT_FALSE(c0.IsEquivalent(c1));
}
}
}
// Tests that we can only send DTMF when the dtmf codec is supported.
void TestCanInsertDtmf(bool can) {
if (can) {
InitWithDtmfCodec();
} else {
Init(NULL);
}
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
EXPECT_FALSE(session_->CanInsertDtmf(""));
EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
}
// The method sets up a call from the session to itself, in a loopback
// arrangement. It also uses a firewall rule to create a temporary
// disconnection. This code is placed as a method so that it can be invoked
// by multiple tests with different allocators (e.g. with and without BUNDLE).
// While running the call, this method also checks if the session goes through
// the correct sequence of ICE states when a connection is established,
// broken, and re-established.
// The Connection state should go:
// New -> Checking -> Connected -> Disconnected -> Connected.
// The Gathering state should go: New -> Gathering -> Completed.
void TestLoopbackCall() {
AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
observer_.ice_gathering_state_);
SetLocalDescriptionWithoutError(offer);
EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
observer_.ice_connection_state_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringGathering,
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
std::string sdp;
offer->ToString(&sdp);
SessionDescriptionInterface* desc =
webrtc::CreateSessionDescription(JsepSessionDescription::kAnswer, sdp);
ASSERT_TRUE(desc != NULL);
SetRemoteDescriptionWithoutError(desc);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
// TODO(bemasc): EXPECT(Completed) once the details are standardized.
// Adding firewall rule to block ping requests, which should cause
// transport channel failure.
fss_->AddRule(false,
talk_base::FP_ANY,
talk_base::FD_ANY,
talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
// Clearing the rules, session should move back to completed state.
fss_->ClearRules();
// Session is automatically calling OnSignalingReady after creation of
// new portallocator session which will allocate new set of candidates.
// TODO(bemasc): Change this to Completed once the details are standardized.
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
}
void VerifyTransportType(const std::string& content_name,
cricket::TransportProtocol protocol) {
const cricket::Transport* transport = session_->GetTransport(content_name);
ASSERT_TRUE(transport != NULL);
EXPECT_EQ(protocol, transport->protocol());
}
// Adds CN codecs to FakeMediaEngine and MediaDescriptionFactory.
void AddCNCodecs() {
const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1, 0);
const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1, 0);
// Add kCNCodec for dtmf test.
std::vector<cricket::AudioCodec> codecs = media_engine_->audio_codecs();;
codecs.push_back(kCNCodec1);
codecs.push_back(kCNCodec2);
media_engine_->SetAudioCodecs(codecs);
desc_factory_->set_audio_codecs(codecs);
}
bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
const cricket::ContentDescription* description = content->description;
ASSERT(description != NULL);
const cricket::AudioContentDescription* audio_content_desc =
static_cast<const cricket::AudioContentDescription*>(description);
ASSERT(audio_content_desc != NULL);
for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
if (audio_content_desc->codecs()[i].name == "CN")
return false;
}
return true;
}
void SetLocalDescriptionWithDataChannel() {
webrtc::DataChannelInit dci;
dci.reliable = false;
session_->CreateDataChannel("datachannel", &dci);
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
}
void VerifyMultipleAsyncCreateDescription(
bool success, CreateSessionDescriptionRequest::Type type) {
InitWithDtls(!success);
if (type == CreateSessionDescriptionRequest::kAnswer) {
cricket::MediaSessionOptions options;
scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options, cricket::SEC_REQUIRED));
ASSERT_TRUE(offer.get() != NULL);
SetRemoteDescriptionWithoutError(offer.release());
}
const int kNumber = 3;
talk_base::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
observers[kNumber];
for (int i = 0; i < kNumber; ++i) {
observers[i] = new WebRtcSessionCreateSDPObserverForTest();
if (type == CreateSessionDescriptionRequest::kOffer) {
session_->CreateOffer(observers[i], NULL);
} else {
session_->CreateAnswer(observers[i], NULL);
}
}
WebRtcSessionCreateSDPObserverForTest::State expected_state =
success ? WebRtcSessionCreateSDPObserverForTest::kSucceeded :
WebRtcSessionCreateSDPObserverForTest::kFailed;
for (int i = 0; i < kNumber; ++i) {
EXPECT_EQ_WAIT(expected_state, observers[i]->state(), 1000);
if (success) {
EXPECT_TRUE(observers[i]->description() != NULL);
} else {
EXPECT_TRUE(observers[i]->description() == NULL);
}
}
}
cricket::FakeMediaEngine* media_engine_;
cricket::FakeDataEngine* data_engine_;
cricket::FakeDeviceManager* device_manager_;
talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
talk_base::scoped_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
talk_base::scoped_ptr<talk_base::SSLIdentity> identity_;
talk_base::scoped_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
talk_base::scoped_ptr<talk_base::PhysicalSocketServer> pss_;
talk_base::scoped_ptr<talk_base::VirtualSocketServer> vss_;
talk_base::scoped_ptr<talk_base::FirewallSocketServer> fss_;
talk_base::SocketServerScope ss_scope_;
talk_base::SocketAddress stun_socket_addr_;
cricket::TestStunServer stun_server_;
talk_base::FakeNetworkManager network_manager_;
cricket::BasicPortAllocator allocator_;
PeerConnectionFactoryInterface::Options options_;
talk_base::scoped_ptr<FakeConstraints> constraints_;
FakeMediaStreamSignaling mediastream_signaling_;
talk_base::scoped_ptr<WebRtcSessionForTest> session_;
MockIceObserver observer_;
cricket::FakeVideoMediaChannel* video_channel_;
cricket::FakeVoiceMediaChannel* voice_channel_;
};
TEST_F(WebRtcSessionTest, TestInitialize) {
Init(NULL);
}
TEST_F(WebRtcSessionTest, TestInitializeWithDtls) {
InitWithDtls();
}
// Verifies that WebRtcSession uses SEC_REQUIRED by default.
TEST_F(WebRtcSessionTest, TestDefaultSetSecurePolicy) {
Init(NULL);
EXPECT_EQ(cricket::SEC_REQUIRED, session_->SecurePolicy());
}
TEST_F(WebRtcSessionTest, TestSessionCandidates) {
TestSessionCandidatesWithBundleRtcpMux(false, false);
}
// Below test cases (TestSessionCandidatesWith*) verify the candidates gathered
// with rtcp-mux and/or bundle.
TEST_F(WebRtcSessionTest, TestSessionCandidatesWithRtcpMux) {
TestSessionCandidatesWithBundleRtcpMux(false, true);
}
TEST_F(WebRtcSessionTest, TestSessionCandidatesWithBundleRtcpMux) {
TestSessionCandidatesWithBundleRtcpMux(true, true);
}
TEST_F(WebRtcSessionTest, TestMultihomeCandidates) {
AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
AddInterface(talk_base::SocketAddress(kClientAddrHost2, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
InitiateCall();
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ(8u, observer_.mline_0_candidates_.size());
EXPECT_EQ(8u, observer_.mline_1_candidates_.size());
}
TEST_F(WebRtcSessionTest, TestStunError) {
AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
AddInterface(talk_base::SocketAddress(kClientAddrHost2, kClientAddrPort));
fss_->AddRule(false,
talk_base::FP_UDP,
talk_base::FD_ANY,
talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
InitiateCall();
// Since kClientAddrHost1 is blocked, not expecting stun candidates for it.
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ(6u, observer_.mline_0_candidates_.size());
EXPECT_EQ(6u, observer_.mline_1_candidates_.size());
}
// Test creating offers and receive answers and make sure the
// media engine creates the expected send and receive streams.
TEST_F(WebRtcSessionTest, TestCreateOfferReceiveAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
const std::string session_id_orig = offer->session_id();
const std::string session_version_orig = offer->session_version();
SetLocalDescriptionWithoutError(offer);
mediastream_signaling_.SendAudioVideoStream2();
SessionDescriptionInterface* answer =
CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionWithoutError(answer);
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_EQ(1u, video_channel_->recv_streams().size());
EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, video_channel_->send_streams().size());
EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
// Create new offer without send streams.
mediastream_signaling_.SendNothing();
offer = CreateOffer(NULL);
// Verify the session id is the same and the session version is
// increased.
EXPECT_EQ(session_id_orig, offer->session_id());
EXPECT_LT(talk_base::FromString<uint64>(session_version_orig),
talk_base::FromString<uint64>(offer->session_version()));
SetLocalDescriptionWithoutError(offer);
mediastream_signaling_.SendAudioVideoStream2();
answer = CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionWithoutError(answer);
EXPECT_EQ(0u, video_channel_->send_streams().size());
EXPECT_EQ(0u, voice_channel_->send_streams().size());
// Make sure the receive streams have not changed.
ASSERT_EQ(1u, video_channel_->recv_streams().size());
EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
}
// Test receiving offers and creating answers and make sure the
// media engine creates the expected send and receive streams.
TEST_F(WebRtcSessionTest, TestReceiveOfferCreateAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream2();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
const std::string session_id_orig = answer->session_id();
const std::string session_version_orig = answer->session_version();
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_EQ(1u, video_channel_->recv_streams().size());
EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, video_channel_->send_streams().size());
EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
mediastream_signaling_.SendAudioVideoStream1And2();
offer = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer);
// Answer by turning off all send streams.
mediastream_signaling_.SendNothing();
answer = CreateAnswer(NULL);
// Verify the session id is the same and the session version is
// increased.
EXPECT_EQ(session_id_orig, answer->session_id());
EXPECT_LT(talk_base::FromString<uint64>(session_version_orig),
talk_base::FromString<uint64>(answer->session_version()));
SetLocalDescriptionWithoutError(answer);
ASSERT_EQ(2u, video_channel_->recv_streams().size());
EXPECT_TRUE(kVideoTrack1 == video_channel_->recv_streams()[0].id);
EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[1].id);
ASSERT_EQ(2u, voice_channel_->recv_streams().size());
EXPECT_TRUE(kAudioTrack1 == voice_channel_->recv_streams()[0].id);
EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[1].id);
// Make sure we have no send streams.
EXPECT_EQ(0u, video_channel_->send_streams().size());
EXPECT_EQ(0u, voice_channel_->send_streams().size());
}
// Test we will return fail when apply an offer that doesn't have
// crypto enabled.
TEST_F(WebRtcSessionTest, SetNonCryptoOffer) {
Init(NULL);
cricket::MediaSessionOptions options;
options.has_video = true;
JsepSessionDescription* offer = CreateRemoteOffer(
options, cricket::SEC_DISABLED);
ASSERT_TRUE(offer != NULL);
VerifyNoCryptoParams(offer->description(), false);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetRemoteDescriptionExpectError(kSdpWithoutCrypto, offer);
offer = CreateRemoteOffer(options, cricket::SEC_DISABLED);
ASSERT_TRUE(offer != NULL);
SetLocalDescriptionExpectError(kSdpWithoutCrypto, offer);
}
// Test we will return fail when apply an answer that doesn't have
// crypto enabled.
TEST_F(WebRtcSessionTest, SetLocalNonCryptoAnswer) {
Init(NULL);
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetRemoteDescriptionWithoutError(offer);
SetLocalDescriptionExpectError(kSdpWithoutCrypto, answer);
}
// Test we will return fail when apply an answer that doesn't have
// crypto enabled.
TEST_F(WebRtcSessionTest, SetRemoteNonCryptoAnswer) {
Init(NULL);
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetLocalDescriptionWithoutError(offer);
SetRemoteDescriptionExpectError(kSdpWithoutCrypto, answer);
}
// Test that we can create and set an offer with a DTLS fingerprint.
TEST_F(WebRtcSessionTest, CreateSetDtlsOffer) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls();
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), true);
// SetLocalDescription will take the ownership of the offer.
SetLocalDescriptionWithoutError(offer);
}
// Test that we can process an offer with a DTLS fingerprint
// and that we return an answer with a fingerprint.
TEST_F(WebRtcSessionTest, ReceiveDtlsOfferCreateAnswer) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls();
SetFactoryDtlsSrtp();
cricket::MediaSessionOptions options;
options.has_video = true;
JsepSessionDescription* offer = CreateRemoteOffer(options);
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), true);
// SetRemoteDescription will take the ownership of the offer.
SetRemoteDescriptionWithoutError(offer);
// Verify that we get a crypto fingerprint in the answer.
SessionDescriptionInterface* answer = CreateAnswer(NULL);
ASSERT_TRUE(answer != NULL);
VerifyFingerprintStatus(answer->description(), true);
// Check that we don't have an a=crypto line in the answer.
VerifyNoCryptoParams(answer->description(), true);
// Now set the local description, which should work, even without a=crypto.
SetLocalDescriptionWithoutError(answer);
}
// Test that even if we support DTLS, if the other side didn't offer a
// fingerprint, we don't either.
TEST_F(WebRtcSessionTest, ReceiveNoDtlsOfferCreateAnswer) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls();
cricket::MediaSessionOptions options;
options.has_video = true;
JsepSessionDescription* offer = CreateRemoteOffer(
options, cricket::SEC_REQUIRED);
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), false);
// SetRemoteDescription will take the ownership of
// the offer.
SetRemoteDescriptionWithoutError(offer);
// Verify that we don't get a crypto fingerprint in the answer.
SessionDescriptionInterface* answer = CreateAnswer(NULL);
ASSERT_TRUE(answer != NULL);
VerifyFingerprintStatus(answer->description(), false);
// Now set the local description.
SetLocalDescriptionWithoutError(answer);
}
TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
Init(NULL);
mediastream_signaling_.SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer2 = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer2);
}
TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
Init(NULL);
mediastream_signaling_.SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* offer2 = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer2);
}
TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
offer = CreateOffer(NULL);
SetRemoteDescriptionExpectError(
"Called with type in wrong state, type: offer state: STATE_SENTINITIATE",
offer);
}
TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer);
offer = CreateOffer(NULL);
SetLocalDescriptionExpectError(
"Called with type in wrong state, type: "
"offer state: STATE_RECEIVEDINITIATE",
offer);
}
TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateRemoteOffer();
SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE);
JsepSessionDescription* pranswer = static_cast<JsepSessionDescription*>(
CreateAnswer(NULL));
pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
SetLocalDescriptionExpectState(pranswer, BaseSession::STATE_SENTPRACCEPT);
mediastream_signaling_.SendAudioVideoStream1();
JsepSessionDescription* pranswer2 = static_cast<JsepSessionDescription*>(
CreateAnswer(NULL));
pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
SetLocalDescriptionExpectState(pranswer2, BaseSession::STATE_SENTPRACCEPT);
mediastream_signaling_.SendAudioVideoStream2();
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionExpectState(answer, BaseSession::STATE_SENTACCEPT);
}
TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
JsepSessionDescription* pranswer =
CreateRemoteAnswer(session_->local_description());
pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
SetRemoteDescriptionExpectState(pranswer,
BaseSession::STATE_RECEIVEDPRACCEPT);
mediastream_signaling_.SendAudioVideoStream1();
JsepSessionDescription* pranswer2 =
CreateRemoteAnswer(session_->local_description());
pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
SetRemoteDescriptionExpectState(pranswer2,
BaseSession::STATE_RECEIVEDPRACCEPT);
mediastream_signaling_.SendAudioVideoStream2();
SessionDescriptionInterface* answer =
CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionExpectState(answer, BaseSession::STATE_RECEIVEDACCEPT);
}
TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer.get());
SetLocalDescriptionExpectError(
"Called with type in wrong state, type: answer state: STATE_INIT",
answer);
}
TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer.get());
SetRemoteDescriptionExpectError(
"Called with type in wrong state, type: answer state: STATE_INIT",
answer);
}
TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
cricket::Candidate candidate;
candidate.set_component(1);
JsepIceCandidate ice_candidate1(kMediaContentName0, 0, candidate);
// Fail since we have not set a offer description.
EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1));
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
// Candidate should be allowed to add before remote description.
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
candidate.set_component(2);
JsepIceCandidate ice_candidate2(kMediaContentName0, 0, candidate);
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
SessionDescriptionInterface* answer = CreateRemoteAnswer(
session_->local_description());
SetRemoteDescriptionWithoutError(answer);
// Verifying the candidates are copied properly from internal vector.
const SessionDescriptionInterface* remote_desc =
session_->remote_description();
ASSERT_TRUE(remote_desc != NULL);
ASSERT_EQ(2u, remote_desc->number_of_mediasections());
const IceCandidateCollection* candidates =
remote_desc->candidates(kMediaContentIndex0);
ASSERT_EQ(2u, candidates->count());
EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
EXPECT_EQ(kMediaContentName0, candidates->at(0)->sdp_mid());
EXPECT_EQ(1, candidates->at(0)->candidate().component());
EXPECT_EQ(2, candidates->at(1)->candidate().component());
candidate.set_component(2);
JsepIceCandidate ice_candidate3(kMediaContentName0, 0, candidate);
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate3));
ASSERT_EQ(3u, candidates->count());
JsepIceCandidate bad_ice_candidate("bad content name", 99, candidate);
EXPECT_FALSE(session_->ProcessIceMessage(&bad_ice_candidate));
}
// Test that a remote candidate is added to the remote session description and
// that it is retained if the remote session description is changed.
TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
Init(NULL);
cricket::Candidate candidate1;
candidate1.set_component(1);
JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
candidate1);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
const SessionDescriptionInterface* remote_desc =
session_->remote_description();
ASSERT_TRUE(remote_desc != NULL);
ASSERT_EQ(2u, remote_desc->number_of_mediasections());
const IceCandidateCollection* candidates =
remote_desc->candidates(kMediaContentIndex0);
ASSERT_EQ(1u, candidates->count());
EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
// Update the RemoteSessionDescription with a new session description and
// a candidate and check that the new remote session description contains both
// candidates.
SessionDescriptionInterface* offer = CreateRemoteOffer();
cricket::Candidate candidate2;
JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0,
candidate2);
EXPECT_TRUE(offer->AddCandidate(&ice_candidate2));
SetRemoteDescriptionWithoutError(offer);
remote_desc = session_->remote_description();
ASSERT_TRUE(remote_desc != NULL);
ASSERT_EQ(2u, remote_desc->number_of_mediasections());
candidates = remote_desc->candidates(kMediaContentIndex0);
ASSERT_EQ(2u, candidates->count());
EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
// Username and password have be updated with the TransportInfo of the
// SessionDescription, won't be equal to the original one.
candidate2.set_username(candidates->at(0)->candidate().username());
candidate2.set_password(candidates->at(0)->candidate().password());
EXPECT_TRUE(candidate2.IsEquivalent(candidates->at(0)->candidate()));
EXPECT_EQ(kMediaContentIndex0, candidates->at(1)->sdp_mline_index());
// No need to verify the username and password.
candidate1.set_username(candidates->at(1)->candidate().username());
candidate1.set_password(candidates->at(1)->candidate().password());
EXPECT_TRUE(candidate1.IsEquivalent(candidates->at(1)->candidate()));
// Test that the candidate is ignored if we can add the same candidate again.
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
}
// Test that local candidates are added to the local session description and
// that they are retained if the local session description is changed.
TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
const SessionDescriptionInterface* local_desc = session_->local_description();
const IceCandidateCollection* candidates =
local_desc->candidates(kMediaContentIndex0);
ASSERT_TRUE(candidates != NULL);
EXPECT_EQ(0u, candidates->count());
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
local_desc = session_->local_description();
candidates = local_desc->candidates(kMediaContentIndex0);
ASSERT_TRUE(candidates != NULL);
EXPECT_LT(0u, candidates->count());
candidates = local_desc->candidates(1);
ASSERT_TRUE(candidates != NULL);
EXPECT_LT(0u, candidates->count());
// Update the session descriptions.
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
local_desc = session_->local_description();
candidates = local_desc->candidates(kMediaContentIndex0);
ASSERT_TRUE(candidates != NULL);
EXPECT_LT(0u, candidates->count());
candidates = local_desc->candidates(1);
ASSERT_TRUE(candidates != NULL);
EXPECT_LT(0u, candidates->count());
}
// Test that we can set a remote session description with remote candidates.
TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
Init(NULL);
cricket::Candidate candidate1;
candidate1.set_component(1);
JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
candidate1);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
SetRemoteDescriptionWithoutError(offer);
const SessionDescriptionInterface* remote_desc =
session_->remote_description();
ASSERT_TRUE(remote_desc != NULL);
ASSERT_EQ(2u, remote_desc->number_of_mediasections());
const IceCandidateCollection* candidates =
remote_desc->candidates(kMediaContentIndex0);
ASSERT_EQ(1u, candidates->count());
EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
}
// Test that offers and answers contains ice candidates when Ice candidates have
// been gathered.
TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
AddInterface(talk_base::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
// Ice is started but candidates are not provided until SetLocalDescription
// is called.
EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
CreateAndSetRemoteOfferAndLocalAnswer();
// Wait until at least one local candidate has been collected.
EXPECT_TRUE_WAIT(0u < observer_.mline_0_candidates_.size(),
kIceCandidatesTimeout);
EXPECT_TRUE_WAIT(0u < observer_.mline_1_candidates_.size(),
kIceCandidatesTimeout);
talk_base::scoped_ptr<SessionDescriptionInterface> local_offer(
CreateOffer(NULL));
ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
ASSERT_TRUE(local_offer->candidates(kMediaContentIndex1) != NULL);
EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex1)->count());
SessionDescriptionInterface* remote_offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(remote_offer);
SessionDescriptionInterface* answer = CreateAnswer(NULL);
ASSERT_TRUE(answer->candidates(kMediaContentIndex0) != NULL);
EXPECT_LT(0u, answer->candidates(kMediaContentIndex0)->count());
ASSERT_TRUE(answer->candidates(kMediaContentIndex1) != NULL);
EXPECT_LT(0u, answer->candidates(kMediaContentIndex1)->count());
SetLocalDescriptionWithoutError(answer);
}
// Verifies TransportProxy and media channels are created with content names
// present in the SessionDescription.
TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
// CreateOffer creates session description with the content names "audio" and
// "video". Goal is to modify these content names and verify transport channel
// proxy in the BaseSession, as proxies are created with the content names
// present in SDP.
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
const std::string kAudioMid = "a=mid:audio";
const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
const std::string kVideoMid = "a=mid:video";
const std::string kVideoMidReplaceStr = "a=mid:video_content_name";
// Replacing |audio| with |audio_content_name|.
talk_base::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
kAudioMidReplaceStr.c_str(),
kAudioMidReplaceStr.length(),
&sdp);
// Replacing |video| with |video_content_name|.
talk_base::replace_substrs(kVideoMid.c_str(), kVideoMid.length(),
kVideoMidReplaceStr.c_str(),
kVideoMidReplaceStr.length(),
&sdp);
SessionDescriptionInterface* modified_offer =
CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
SetRemoteDescriptionWithoutError(modified_offer);
SessionDescriptionInterface* answer =
CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
EXPECT_TRUE(session_->GetTransportProxy("audio_content_name") != NULL);
EXPECT_TRUE(session_->GetTransportProxy("video_content_name") != NULL);
EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL);
}
// Test that an offer contains the correct media content descriptions based on
// the send streams when no constraints have been set.
TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
Init(NULL);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
ASSERT_TRUE(offer != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content == NULL);
}
// Test that an offer contains the correct media content descriptions based on
// the send streams when no constraints have been set.
TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
Init(NULL);
// Test Audio only offer.
mediastream_signaling_.UseOptionsAudioOnly();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content == NULL);
// Test Audio / Video offer.
mediastream_signaling_.SendAudioVideoStream1();
offer.reset(CreateOffer(NULL));
content = cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content != NULL);
}
// Test that an offer contains no media content descriptions if
// kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
Init(NULL);
webrtc::FakeConstraints constraints_no_receive;
constraints_no_receive.SetMandatoryReceiveAudio(false);
constraints_no_receive.SetMandatoryReceiveVideo(false);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints_no_receive));
ASSERT_TRUE(offer != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content == NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content == NULL);
}
// Test that an offer contains only audio media content descriptions if
// kOfferToReceiveAudio constraints are set to true.
TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
Init(NULL);
webrtc::FakeConstraints constraints_audio_only;
constraints_audio_only.SetMandatoryReceiveAudio(true);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints_audio_only));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content == NULL);
}
// Test that an offer contains audio and video media content descriptions if
// kOfferToReceiveAudio and kOfferToReceiveVideo constraints are set to true.
TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
Init(NULL);
// Test Audio / Video offer.
webrtc::FakeConstraints constraints_audio_video;
constraints_audio_video.SetMandatoryReceiveAudio(true);
constraints_audio_video.SetMandatoryReceiveVideo(true);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints_audio_video));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content != NULL);
// TODO(perkj): Should the direction be set to SEND_ONLY if
// The constraints is set to not receive audio or video but a track is added?
}
// Test that an answer can not be created if the last remote description is not
// an offer.
TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
Init(NULL);
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
SetRemoteDescriptionWithoutError(answer);
EXPECT_TRUE(CreateAnswer(NULL) == NULL);
}
// Test that an answer contains the correct media content descriptions when no
// constraints have been set.
TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
Init(NULL);
// Create a remote offer with audio and video content.
talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
content = cricket::GetFirstVideoContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
}
// Test that an answer contains the correct media content descriptions when no
// constraints have been set and the offer only contain audio.
TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
Init(NULL);
// Create a remote offer with audio only.
cricket::MediaSessionOptions options;
options.has_audio = true;
options.has_video = false;
talk_base::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
ASSERT_TRUE(cricket::GetFirstAudioContent(offer->description()) != NULL);
SetRemoteDescriptionWithoutError(offer.release());
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
EXPECT_TRUE(cricket::GetFirstVideoContent(answer->description()) == NULL);
}
// Test that an answer contains the correct media content descriptions when no
// constraints have been set.
TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
Init(NULL);
// Create a remote offer with audio and video content.
talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
// Test with a stream with tracks.
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
content = cricket::GetFirstVideoContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
}
// Test that an answer contains the correct media content descriptions when
// constraints have been set but no stream is sent.
TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
Init(NULL);
// Create a remote offer with audio and video content.
talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
webrtc::FakeConstraints constraints_no_receive;
constraints_no_receive.SetMandatoryReceiveAudio(false);
constraints_no_receive.SetMandatoryReceiveVideo(false);
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(&constraints_no_receive));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_TRUE(content->rejected);
content = cricket::GetFirstVideoContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_TRUE(content->rejected);
}
// Test that an answer contains the correct media content descriptions when
// constraints have been set and streams are sent.
TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
Init(NULL);
// Create a remote offer with audio and video content.
talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
webrtc::FakeConstraints constraints_no_receive;
constraints_no_receive.SetMandatoryReceiveAudio(false);
constraints_no_receive.SetMandatoryReceiveVideo(false);
// Test with a stream with tracks.
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(&constraints_no_receive));
// TODO(perkj): Should the direction be set to SEND_ONLY?
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
// TODO(perkj): Should the direction be set to SEND_ONLY?
content = cricket::GetFirstVideoContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
}
TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
AddCNCodecs();
Init(NULL);
webrtc::FakeConstraints constraints;
constraints.SetOptionalVAD(false);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
EXPECT_TRUE(VerifyNoCNCodecs(content));
}
TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
AddCNCodecs();
Init(NULL);
// Create a remote offer with audio and video content.
talk_base::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
webrtc::FakeConstraints constraints;
constraints.SetOptionalVAD(false);
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(&constraints));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(answer->description());
ASSERT_TRUE(content != NULL);
EXPECT_TRUE(VerifyNoCNCodecs(content));
}
// This test verifies the call setup when remote answer with audio only and
// later updates with video.
TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
Init(NULL);
EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
cricket::MediaSessionOptions options;
options.has_video = false;
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer, options);
// SetLocalDescription and SetRemoteDescriptions takes ownership of offer
// and answer;
SetLocalDescriptionWithoutError(offer);
SetRemoteDescriptionWithoutError(answer);
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(video_channel_ == NULL);
ASSERT_EQ(0u, voice_channel_->recv_streams().size());
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_EQ(kAudioTrack1, voice_channel_->send_streams()[0].id);
// Let the remote end update the session descriptions, with Audio and Video.
mediastream_signaling_.SendAudioVideoStream2();
CreateAndSetRemoteOfferAndLocalAnswer();
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(video_channel_ != NULL);
ASSERT_TRUE(voice_channel_ != NULL);
ASSERT_EQ(1u, video_channel_->recv_streams().size());
ASSERT_EQ(1u, video_channel_->send_streams().size());
EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
// Change session back to audio only.
mediastream_signaling_.UseOptionsAudioOnly();
CreateAndSetRemoteOfferAndLocalAnswer();
EXPECT_EQ(0u, video_channel_->recv_streams().size());
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
}
// This test verifies the call setup when remote answer with video only and
// later updates with audio.
TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
Init(NULL);
EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
cricket::MediaSessionOptions options;
options.has_audio = false;
options.has_video = true;
SessionDescriptionInterface* answer = CreateRemoteAnswer(
offer, options, cricket::SEC_ENABLED);
// SetLocalDescription and SetRemoteDescriptions takes ownership of offer
// and answer.
SetLocalDescriptionWithoutError(offer);
SetRemoteDescriptionWithoutError(answer);
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(voice_channel_ == NULL);
ASSERT_TRUE(video_channel_ != NULL);
EXPECT_EQ(0u, video_channel_->recv_streams().size());
ASSERT_EQ(1u, video_channel_->send_streams().size());
EXPECT_EQ(kVideoTrack1, video_channel_->send_streams()[0].id);
// Update the session descriptions, with Audio and Video.
mediastream_signaling_.SendAudioVideoStream2();
CreateAndSetRemoteOfferAndLocalAnswer();
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(voice_channel_ != NULL);
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
// Change session back to video only.
mediastream_signaling_.UseOptionsVideoOnly();
CreateAndSetRemoteOfferAndLocalAnswer();
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_EQ(1u, video_channel_->recv_streams().size());
EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, video_channel_->send_streams().size());
EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
}
TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
VerifyCryptoParams(offer->description());
SetRemoteDescriptionWithoutError(offer.release());
scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
VerifyCryptoParams(answer->description());
}
TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
options_.disable_encryption = true;
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
VerifyNoCryptoParams(offer->description(), false);
}
TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) {
Init(NULL);
VerifyAnswerFromNonCryptoOffer();
}
TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
Init(NULL);
VerifyAnswerFromCryptoOffer();
}
// This test verifies that setLocalDescription fails if
// no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
std::string sdp;
RemoveIceUfragPwdLines(offer.get(), &sdp);
SessionDescriptionInterface* modified_offer =
CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
SetLocalDescriptionExpectError(kSdpWithoutIceUfragPwd, modified_offer);
}
// This test verifies that setRemoteDescription fails if
// no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
Init(NULL);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
std::string sdp;
RemoveIceUfragPwdLines(offer.get(), &sdp);
SessionDescriptionInterface* modified_offer =
CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
SetRemoteDescriptionExpectError(kSdpWithoutIceUfragPwd, modified_offer);
}
TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) {
// This test verifies BUNDLE flag in PortAllocator, if BUNDLE information in
// local description is removed by the application, BUNDLE flag should be
// disabled in PortAllocator. By default BUNDLE is enabled in the WebRtc.
Init(NULL);
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
cricket::PORTALLOCATOR_ENABLE_BUNDLE);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
cricket::SessionDescription* offer_copy =
offer->description()->Copy();
offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
JsepSessionDescription* modified_offer =
new JsepSessionDescription(JsepSessionDescription::kOffer);
modified_offer->Initialize(offer_copy, "1", "1");
SetLocalDescriptionWithoutError(modified_offer);
EXPECT_FALSE(allocator_.flags() & cricket::PORTALLOCATOR_ENABLE_BUNDLE);
}
TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
cricket::PORTALLOCATOR_ENABLE_BUNDLE);
FakeConstraints constraints;
constraints.SetMandatoryUseRtpMux(true);
SessionDescriptionInterface* offer = CreateOffer(&constraints);
SetLocalDescriptionWithoutError(offer);
mediastream_signaling_.SendAudioVideoStream2();
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateRemoteAnswer(session_->local_description()));
cricket::SessionDescription* answer_copy = answer->description()->Copy();
answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
JsepSessionDescription* modified_answer =
new JsepSessionDescription(JsepSessionDescription::kAnswer);
modified_answer->Initialize(answer_copy, "1", "1");
SetRemoteDescriptionWithoutError(modified_answer);
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
cricket::PORTALLOCATOR_ENABLE_BUNDLE);
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_EQ(1u, video_channel_->recv_streams().size());
EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->recv_streams().size());
EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
ASSERT_EQ(1u, video_channel_->send_streams().size());
EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
ASSERT_EQ(1u, voice_channel_->send_streams().size());
EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
}
// This test verifies that SetLocalDescription and SetRemoteDescription fails
// if BUNDLE is enabled but rtcp-mux is disabled in m-lines.
TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
WebRtcSessionTest::Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_.flags()) ==
cricket::PORTALLOCATOR_ENABLE_BUNDLE);
FakeConstraints constraints;
constraints.SetMandatoryUseRtpMux(true);
SessionDescriptionInterface* offer = CreateOffer(&constraints);
std::string offer_str;
offer->ToString(&offer_str);
// Disable rtcp-mux
const std::string rtcp_mux = "rtcp-mux";
const std::string xrtcp_mux = "xrtcp-mux";
talk_base::replace_substrs(rtcp_mux.c_str(), rtcp_mux.length(),
xrtcp_mux.c_str(), xrtcp_mux.length(),
&offer_str);
JsepSessionDescription *local_offer =
new JsepSessionDescription(JsepSessionDescription::kOffer);
EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
SetLocalDescriptionExpectError(kBundleWithoutRtcpMux, local_offer);
JsepSessionDescription *remote_offer =
new JsepSessionDescription(JsepSessionDescription::kOffer);
EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL));
SetRemoteDescriptionExpectError(kBundleWithoutRtcpMux, remote_offer);
// Trying unmodified SDP.
SetLocalDescriptionWithoutError(offer);
}
TEST_F(WebRtcSessionTest, SetAudioPlayout) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(channel != NULL);
ASSERT_EQ(1u, channel->recv_streams().size());
uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc();
double left_vol, right_vol;
EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
EXPECT_EQ(1, left_vol);
EXPECT_EQ(1, right_vol);
talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
session_->SetAudioPlayout(receive_ssrc, false, renderer.get());
EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
EXPECT_EQ(0, left_vol);
EXPECT_EQ(0, right_vol);
EXPECT_EQ(0, renderer->channel_id());
session_->SetAudioPlayout(receive_ssrc, true, NULL);
EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
EXPECT_EQ(1, left_vol);
EXPECT_EQ(1, right_vol);
EXPECT_EQ(-1, renderer->channel_id());
}
TEST_F(WebRtcSessionTest, SetAudioSend) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(channel != NULL);
ASSERT_EQ(1u, channel->send_streams().size());
uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
cricket::AudioOptions options;
options.echo_cancellation.Set(true);
talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
session_->SetAudioSend(send_ssrc, false, options, renderer.get());
EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
EXPECT_EQ(0, renderer->channel_id());
session_->SetAudioSend(send_ssrc, true, options, NULL);
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
bool value;
EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
EXPECT_TRUE(value);
EXPECT_EQ(-1, renderer->channel_id());
}
TEST_F(WebRtcSessionTest, SetVideoPlayout) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
ASSERT_TRUE(channel != NULL);
ASSERT_LT(0u, channel->renderers().size());
EXPECT_TRUE(channel->renderers().begin()->second == NULL);
ASSERT_EQ(1u, channel->recv_streams().size());
uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc();
cricket::FakeVideoRenderer renderer;
session_->SetVideoPlayout(receive_ssrc, true, &renderer);
EXPECT_TRUE(channel->renderers().begin()->second == &renderer);
session_->SetVideoPlayout(receive_ssrc, false, &renderer);
EXPECT_TRUE(channel->renderers().begin()->second == NULL);
}
TEST_F(WebRtcSessionTest, SetVideoSend) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
ASSERT_TRUE(channel != NULL);
ASSERT_EQ(1u, channel->send_streams().size());
uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
cricket::VideoOptions* options = NULL;
session_->SetVideoSend(send_ssrc, false, options);
EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
session_->SetVideoSend(send_ssrc, true, options);
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
}
TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
TestCanInsertDtmf(false);
}
TEST_F(WebRtcSessionTest, CanInsertDtmf) {
TestCanInsertDtmf(true);
}
TEST_F(WebRtcSessionTest, InsertDtmf) {
// Setup
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
EXPECT_EQ(0U, channel->dtmf_info_queue().size());
// Insert DTMF
const int expected_flags = DF_SEND;
const int expected_duration = 90;
session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
session_->InsertDtmf(kAudioTrack1, 2, expected_duration);
// Verify
ASSERT_EQ(3U, channel->dtmf_info_queue().size());
const uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0,
expected_duration, expected_flags));
EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1,
expected_duration, expected_flags));
EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2,
expected_duration, expected_flags));
}
// This test verifies the |initiator| flag when session initiates the call.
TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
Init(NULL);
EXPECT_FALSE(session_->initiator());
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
SetLocalDescriptionWithoutError(offer);
EXPECT_TRUE(session_->initiator());
SetRemoteDescriptionWithoutError(answer);
EXPECT_TRUE(session_->initiator());
}
// This test verifies the |initiator| flag when session receives the call.
TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
Init(NULL);
EXPECT_FALSE(session_->initiator());
SessionDescriptionInterface* offer = CreateRemoteOffer();
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateAnswer(NULL);
EXPECT_FALSE(session_->initiator());
SetLocalDescriptionWithoutError(answer);
EXPECT_FALSE(session_->initiator());
}
// This test verifies the ice protocol type at initiator of the call
// if |a=ice-options:google-ice| is present in answer.
TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateRemoteAnswer(offer));
SetLocalDescriptionWithoutError(offer);
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
// Adding ice-options to the session level.
InjectAfter("t=0 0\r\n",
"a=ice-options:google-ice\r\n",
&sdp);
SessionDescriptionInterface* answer_with_gice =
CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
SetRemoteDescriptionWithoutError(answer_with_gice);
VerifyTransportType("audio", cricket::ICEPROTO_GOOGLE);
VerifyTransportType("video", cricket::ICEPROTO_GOOGLE);
}
// This test verifies the ice protocol type at initiator of the call
// if ICE RFC5245 is supported in answer.
TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
SetLocalDescriptionWithoutError(offer);
SetRemoteDescriptionWithoutError(answer);
VerifyTransportType("audio", cricket::ICEPROTO_RFC5245);
VerifyTransportType("video", cricket::ICEPROTO_RFC5245);
}
// This test verifies the ice protocol type at receiver side of the call if
// receiver decides to use google-ice.
TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer);
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
// Adding ice-options to the session level.
InjectAfter("t=0 0\r\n",
"a=ice-options:google-ice\r\n",
&sdp);
SessionDescriptionInterface* answer_with_gice =
CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
SetLocalDescriptionWithoutError(answer_with_gice);
VerifyTransportType("audio", cricket::ICEPROTO_GOOGLE);
VerifyTransportType("video", cricket::ICEPROTO_GOOGLE);
}
// This test verifies the ice protocol type at receiver side of the call if
// receiver decides to use ice RFC 5245.
TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
VerifyTransportType("audio", cricket::ICEPROTO_RFC5245);
VerifyTransportType("video", cricket::ICEPROTO_RFC5245);
}
// This test verifies the session state when ICE RFC5245 in offer and
// ICE google-ice in answer.
TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
std::string offer_str;
offer->ToString(&offer_str);
// Disable google-ice
const std::string gice_option = "google-ice";
const std::string xgoogle_xice = "xgoogle-xice";
talk_base::replace_substrs(gice_option.c_str(), gice_option.length(),
xgoogle_xice.c_str(), xgoogle_xice.length(),
&offer_str);
JsepSessionDescription *ice_only_offer =
new JsepSessionDescription(JsepSessionDescription::kOffer);
EXPECT_TRUE((ice_only_offer)->Initialize(offer_str, NULL));
SetLocalDescriptionWithoutError(ice_only_offer);
std::string original_offer_sdp;
EXPECT_TRUE(offer->ToString(&original_offer_sdp));
SessionDescriptionInterface* pranswer_with_gice =
CreateSessionDescription(JsepSessionDescription::kPrAnswer,
original_offer_sdp, NULL);
SetRemoteDescriptionExpectError(kPushDownPranswerTDFailed,
pranswer_with_gice);
SessionDescriptionInterface* answer_with_gice =
CreateSessionDescription(JsepSessionDescription::kAnswer,
original_offer_sdp, NULL);
SetRemoteDescriptionExpectError(kPushDownAnswerTDFailed,
answer_with_gice);
}
// Verifing local offer and remote answer have matching m-lines as per RFC 3264.
TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateRemoteAnswer(session_->local_description()));
cricket::SessionDescription* answer_copy = answer->description()->Copy();
answer_copy->RemoveContentByName("video");
JsepSessionDescription* modified_answer =
new JsepSessionDescription(JsepSessionDescription::kAnswer);
EXPECT_TRUE(modified_answer->Initialize(answer_copy,
answer->session_id(),
answer->session_version()));
SetRemoteDescriptionExpectError(kMlineMismatch, modified_answer);
// Modifying content names.
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
const std::string kAudioMid = "a=mid:audio";
const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
// Replacing |audio| with |audio_content_name|.
talk_base::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
kAudioMidReplaceStr.c_str(),
kAudioMidReplaceStr.length(),
&sdp);
SessionDescriptionInterface* modified_answer1 =
CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
SetRemoteDescriptionExpectError(kMlineMismatch, modified_answer1);
SetRemoteDescriptionWithoutError(answer.release());
}
// Verifying remote offer and local answer have matching m-lines as per
// RFC 3264.
TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateRemoteOffer();
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateAnswer(NULL);
cricket::SessionDescription* answer_copy = answer->description()->Copy();
answer_copy->RemoveContentByName("video");
JsepSessionDescription* modified_answer =
new JsepSessionDescription(JsepSessionDescription::kAnswer);
EXPECT_TRUE(modified_answer->Initialize(answer_copy,
answer->session_id(),
answer->session_version()));
SetLocalDescriptionExpectError(kMlineMismatch, modified_answer);
SetLocalDescriptionWithoutError(answer);
}
// This test verifies that WebRtcSession does not start candidate allocation
// before SetLocalDescription is called.
TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateRemoteOffer();
cricket::Candidate candidate;
candidate.set_component(1);
JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
candidate);
EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
cricket::Candidate candidate1;
candidate1.set_component(1);
JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1,
candidate1);
EXPECT_TRUE(offer->AddCandidate(&ice_candidate1));
SetRemoteDescriptionWithoutError(offer);
ASSERT_TRUE(session_->GetTransportProxy("audio") != NULL);
ASSERT_TRUE(session_->GetTransportProxy("video") != NULL);
// Pump for 1 second and verify that no candidates are generated.
talk_base::Thread::Current()->ProcessMessages(1000);
EXPECT_TRUE(observer_.mline_0_candidates_.empty());
EXPECT_TRUE(observer_.mline_1_candidates_.empty());
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
EXPECT_TRUE(session_->GetTransportProxy("audio")->negotiated());
EXPECT_TRUE(session_->GetTransportProxy("video")->negotiated());
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
}
// This test verifies that crypto parameter is updated in local session
// description as per security policy set in MediaSessionDescriptionFactory.
TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
// Making sure SetLocalDescription correctly sets crypto value in
// SessionDescription object after de-serialization of sdp string. The value
// will be set as per MediaSessionDescriptionFactory.
std::string offer_str;
offer->ToString(&offer_str);
SessionDescriptionInterface* jsep_offer_str =
CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
SetLocalDescriptionWithoutError(jsep_offer_str);
EXPECT_TRUE(session_->voice_channel()->secure_required());
EXPECT_TRUE(session_->video_channel()->secure_required());
}
// This test verifies the crypto parameter when security is disabled.
TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
options_.disable_encryption = true;
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
// Making sure SetLocalDescription correctly sets crypto value in
// SessionDescription object after de-serialization of sdp string. The value
// will be set as per MediaSessionDescriptionFactory.
std::string offer_str;
offer->ToString(&offer_str);
SessionDescriptionInterface *jsep_offer_str =
CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
SetLocalDescriptionWithoutError(jsep_offer_str);
EXPECT_FALSE(session_->voice_channel()->secure_required());
EXPECT_FALSE(session_->video_channel()->secure_required());
}
// This test verifies that an answer contains new ufrag and password if an offer
// with new ufrag and password is received.
TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
Init(NULL);
cricket::MediaSessionOptions options;
options.has_audio = true;
options.has_video = true;
talk_base::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
SetRemoteDescriptionWithoutError(offer.release());
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
SetLocalDescriptionWithoutError(answer.release());
// Receive an offer with new ufrag and password.
options.transport_options.ice_restart = true;
talk_base::scoped_ptr<JsepSessionDescription> updated_offer1(
CreateRemoteOffer(options, session_->remote_description()));
SetRemoteDescriptionWithoutError(updated_offer1.release());
talk_base::scoped_ptr<SessionDescriptionInterface> updated_answer1(
CreateAnswer(NULL));
CompareIceUfragAndPassword(updated_answer1->description(),
session_->local_description()->description(),
false);
SetLocalDescriptionWithoutError(updated_answer1.release());
}
// This test verifies that an answer contains old ufrag and password if an offer
// with old ufrag and password is received.
TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
Init(NULL);
cricket::MediaSessionOptions options;
options.has_audio = true;
options.has_video = true;
talk_base::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
SetRemoteDescriptionWithoutError(offer.release());
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
SetLocalDescriptionWithoutError(answer.release());
// Receive an offer without changed ufrag or password.
options.transport_options.ice_restart = false;
talk_base::scoped_ptr<JsepSessionDescription> updated_offer2(
CreateRemoteOffer(options, session_->remote_description()));
SetRemoteDescriptionWithoutError(updated_offer2.release());
talk_base::scoped_ptr<SessionDescriptionInterface> updated_answer2(
CreateAnswer(NULL));
CompareIceUfragAndPassword(updated_answer2->description(),
session_->local_description()->description(),
true);
SetLocalDescriptionWithoutError(updated_answer2.release());
}
TEST_F(WebRtcSessionTest, TestSessionContentError) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
const std::string session_id_orig = offer->session_id();
const std::string session_version_orig = offer->session_version();
SetLocalDescriptionWithoutError(offer);
video_channel_ = media_engine_->GetVideoChannel(0);
video_channel_->set_fail_set_send_codecs(true);
mediastream_signaling_.SendAudioVideoStream2();
SessionDescriptionInterface* answer =
CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionExpectError("ERROR_CONTENT", answer);
}
// Runs the loopback call test with BUNDLE and STUN disabled.
TEST_F(WebRtcSessionTest, TestIceStatesBasic) {
// Lets try with only UDP ports.
allocator_.set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
cricket::PORTALLOCATOR_DISABLE_TCP |
cricket::PORTALLOCATOR_DISABLE_STUN |
cricket::PORTALLOCATOR_DISABLE_RELAY);
TestLoopbackCall();
}
// Regression-test for a crash which should have been an error.
TEST_F(WebRtcSessionTest, TestNoStateTransitionPendingError) {
Init(NULL);
cricket::MediaSessionOptions options;
options.has_audio = true;
options.has_video = true;
session_->SetError(cricket::BaseSession::ERROR_CONTENT);
SessionDescriptionInterface* offer = CreateRemoteOffer(options);
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer, options);
SetRemoteDescriptionExpectError(kSessionError, offer);
SetLocalDescriptionExpectError(kSessionError, answer);
// Not crashing is our success.
}
TEST_F(WebRtcSessionTest, TestRtpDataChannel) {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
Init(NULL);
SetLocalDescriptionWithDataChannel();
EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
}
TEST_F(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
options_.disable_sctp_data_channels = false;
InitWithDtls(false);
SetLocalDescriptionWithDataChannel();
EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
}
TEST_F(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(false);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL);
EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL);
}
TEST_F(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
SetFactoryDtlsSrtp();
InitWithDtls(false);
// Create remote offer with SCTP.
cricket::MediaSessionOptions options;
options.data_channel_type = cricket::DCT_SCTP;
JsepSessionDescription* offer =
CreateRemoteOffer(options, cricket::SEC_ENABLED);
SetRemoteDescriptionWithoutError(offer);
// Verifies the answer contains SCTP.
talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
EXPECT_TRUE(answer != NULL);
EXPECT_TRUE(answer->description()->GetContentByName("data") != NULL);
EXPECT_TRUE(answer->description()->GetTransportInfoByName("data") != NULL);
}
TEST_F(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
InitWithDtls(false);
SetLocalDescriptionWithDataChannel();
EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
}
TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(false);
SetLocalDescriptionWithDataChannel();
EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
}
TEST_F(WebRtcSessionTest, TestDisableSctpDataChannels) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
options_.disable_sctp_data_channels = true;
InitWithDtls(false);
SetLocalDescriptionWithDataChannel();
EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
}
TEST_F(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
const int new_send_port = 9998;
const int new_recv_port = 7775;
InitWithDtls(false);
SetFactoryDtlsSrtp();
// By default, don't actually add the codecs to desc_factory_; they don't
// actually get serialized for SCTP in BuildMediaDescription(). Instead,
// let the session description get parsed. That'll get the proper codecs
// into the stream.
cricket::MediaSessionOptions options;
JsepSessionDescription* offer = CreateRemoteOfferWithSctpPort(
"stream1", new_send_port, options);
// SetRemoteDescription will take the ownership of the offer.
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = ChangeSDPSctpPort(
new_recv_port, CreateAnswer(NULL));
ASSERT_TRUE(answer != NULL);
// Now set the local description, which'll take ownership of the answer.
SetLocalDescriptionWithoutError(answer);
// TEST PLAN: Set the port number to something new, set it in the SDP,
// and pass it all the way down.
webrtc::DataChannelInit dci;
dci.reliable = true;
EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
talk_base::scoped_refptr<webrtc::DataChannel> dc =
session_->CreateDataChannel("datachannel", &dci);
cricket::FakeDataMediaChannel* ch = data_engine_->GetChannel(0);
int portnum = -1;
ASSERT_TRUE(ch != NULL);
ASSERT_EQ(1UL, ch->send_codecs().size());
EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->send_codecs()[0].id);
EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
ch->send_codecs()[0].name.c_str()));
EXPECT_TRUE(ch->send_codecs()[0].GetParam(cricket::kCodecParamPort,
&portnum));
EXPECT_EQ(new_send_port, portnum);
ASSERT_EQ(1UL, ch->recv_codecs().size());
EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->recv_codecs()[0].id);
EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
ch->recv_codecs()[0].name.c_str()));
EXPECT_TRUE(ch->recv_codecs()[0].GetParam(cricket::kCodecParamPort,
&portnum));
EXPECT_EQ(new_recv_port, portnum);
}
// Verifies that CreateOffer succeeds when CreateOffer is called before async
// identity generation is finished.
TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(false);
EXPECT_TRUE(session_->waiting_for_identity());
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
EXPECT_TRUE(offer != NULL);
}
// Verifies that CreateAnswer succeeds when CreateOffer is called before async
// identity generation is finished.
TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(false);
cricket::MediaSessionOptions options;
scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options, cricket::SEC_REQUIRED));
ASSERT_TRUE(offer.get() != NULL);
SetRemoteDescriptionWithoutError(offer.release());
talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
EXPECT_TRUE(answer != NULL);
}
// Verifies that CreateOffer succeeds when CreateOffer is called after async
// identity generation is finished.
TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(false);
EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
EXPECT_TRUE(offer != NULL);
}
// Verifies that CreateOffer fails when CreateOffer is called after async
// identity generation fails.
TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(true);
EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
EXPECT_TRUE(offer == NULL);
}
// Verifies that CreateOffer succeeds when Multiple CreateOffer calls are made
// before async identity generation is finished.
TEST_F(WebRtcSessionTest,
TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(
true, CreateSessionDescriptionRequest::kOffer);
}
// Verifies that CreateOffer fails when Multiple CreateOffer calls are made
// before async identity generation fails.
TEST_F(WebRtcSessionTest,
TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(
false, CreateSessionDescriptionRequest::kOffer);
}
// Verifies that CreateAnswer succeeds when Multiple CreateAnswer calls are made
// before async identity generation is finished.
TEST_F(WebRtcSessionTest,
TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(
true, CreateSessionDescriptionRequest::kAnswer);
}
// Verifies that CreateAnswer fails when Multiple CreateAnswer calls are made
// before async identity generation fails.
TEST_F(WebRtcSessionTest,
TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(
false, CreateSessionDescriptionRequest::kAnswer);
}
// Verifies that setRemoteDescription fails when DTLS is disabled and the remote
// offer has no SDES crypto but only DTLS fingerprint.
TEST_F(WebRtcSessionTest, TestSetRemoteOfferFailIfDtlsDisabledAndNoCrypto) {
// Init without DTLS.
Init(NULL);
// Create a remote offer with secured transport disabled.
cricket::MediaSessionOptions options;
JsepSessionDescription* offer(CreateRemoteOffer(
options, cricket::SEC_DISABLED));
// Adds a DTLS fingerprint to the remote offer.
cricket::SessionDescription* sdp = offer->description();
TransportInfo* audio = sdp->GetTransportInfoByName("audio");
ASSERT_TRUE(audio != NULL);
ASSERT_TRUE(audio->description.identity_fingerprint.get() == NULL);
audio->description.identity_fingerprint.reset(
talk_base::SSLFingerprint::CreateFromRfc4572(
talk_base::DIGEST_SHA_256, kFakeDtlsFingerprint));
SetRemoteDescriptionExpectError(kSdpWithoutSdesAndDtlsDisabled,
offer);
}
// This test verifies DSCP is properly applied on the media channels.
TEST_F(WebRtcSessionTest, TestDscpConstraint) {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDscp, true);
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SetLocalDescriptionWithoutError(offer);
video_channel_ = media_engine_->GetVideoChannel(0);
voice_channel_ = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(video_channel_ != NULL);
ASSERT_TRUE(voice_channel_ != NULL);
cricket::AudioOptions audio_options;
EXPECT_TRUE(voice_channel_->GetOptions(&audio_options));
cricket::VideoOptions video_options;
EXPECT_TRUE(video_channel_->GetOptions(&video_options));
EXPECT_TRUE(audio_options.dscp.IsSet());
EXPECT_TRUE(audio_options.dscp.GetWithDefaultIfUnset(false));
EXPECT_TRUE(video_options.dscp.IsSet());
EXPECT_TRUE(video_options.dscp.GetWithDefaultIfUnset(false));
}
// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
// currently fails because upon disconnection and reconnection OnIceComplete is
// called more than once without returning to IceGatheringGathering.
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