blob: f0558eee3d45c7c25fe9b755960b7030f9ca9a22 [file] [log] [blame]
/*
* 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 <vector>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/fakemetricsobserver.h"
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/peerconnection.h"
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/sctputils.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/test/fakeconstraints.h"
#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
#include "talk/app/webrtc/videotrack.h"
#include "talk/app/webrtc/webrtcsession.h"
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/fakevideorenderer.h"
#include "talk/media/base/mediachannel.h"
#include "webrtc/p2p/base/stunserver.h"
#include "webrtc/p2p/base/teststunserver.h"
#include "webrtc/p2p/base/testturnserver.h"
#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/p2p/client/basicportallocator.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/firewallsocketserver.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/network.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/virtualsocketserver.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::TransportInfo;
using rtc::SocketAddress;
using rtc::scoped_ptr;
using rtc::Thread;
using webrtc::CreateSessionDescription;
using webrtc::CreateSessionDescriptionObserver;
using webrtc::CreateSessionDescriptionRequest;
using webrtc::DataChannel;
using webrtc::DtlsIdentityStoreInterface;
using webrtc::FakeConstraints;
using webrtc::FakeMetricsObserver;
using webrtc::IceCandidateCollection;
using webrtc::InternalDataChannelInit;
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::kCreateChannelFailed;
using webrtc::kInvalidSdp;
using webrtc::kMlineMismatch;
using webrtc::kPushDownTDFailed;
using webrtc::kSdpWithoutIceUfragPwd;
using webrtc::kSdpWithoutDtlsFingerprint;
using webrtc::kSdpWithoutSdesCrypto;
using webrtc::kSessionError;
using webrtc::kSessionErrorDesc;
using webrtc::kMaxUnsignalledRecvStreams;
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
static const int kClientAddrPort = 0;
static const char kClientAddrHost1[] = "11.11.11.11";
static const char kClientIPv6AddrHost1[] =
"2620:0:aaaa:bbbb:cccc:dddd:eeee:ffff";
static const char kClientAddrHost2[] = "22.22.22.22";
static const char kStunAddrHost[] = "99.99.99.1";
static const SocketAddress kTurnUdpIntAddr("99.99.99.4", 3478);
static const SocketAddress kTurnUdpExtAddr("99.99.99.6", 0);
static const char kTurnUsername[] = "test";
static const char kTurnPassword[] = "test";
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";
static const char kTooLongIceUfragPwd[] =
"IceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfrag"
"IceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfrag"
"IceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfrag"
"IceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfragIceUfrag";
static const char kSdpWithRtx[] =
"v=0\r\n"
"o=- 4104004319237231850 2 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS stream1\r\n"
"m=video 9 RTP/SAVPF 0 96\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=rtcp:9 IN IP4 0.0.0.0\r\n"
"a=ice-ufrag:CerjGp19G7wpXwl7\r\n"
"a=ice-pwd:cMvOlFvQ6ochez1ZOoC2uBEC\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
"inline:5/4N5CDvMiyDArHtBByUM71VIkguH17ZNoX60GrA\r\n"
"a=rtpmap:0 fake_video_codec/90000\r\n"
"a=rtpmap:96 rtx/90000\r\n"
"a=fmtp:96 apt=0\r\n";
static const char kStream1[] = "stream1";
static const char kVideoTrack1[] = "video1";
static const char kAudioTrack1[] = "audio1";
static const char kStream2[] = "stream2";
static const char kVideoTrack2[] = "video2";
static const char kAudioTrack2[] = "audio2";
enum RTCCertificateGenerationMethod { ALREADY_GENERATED, DTLS_IDENTITY_STORE };
// 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;
rtc::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,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
cricket::PortAllocator* port_allocator,
webrtc::IceObserver* ice_observer)
: WebRtcSession(cmgr, signaling_thread, worker_thread, port_allocator) {
RegisterIceObserver(ice_observer);
}
virtual ~WebRtcSessionForTest() {}
// Note that these methods are only safe to use if the signaling thread
// is the same as the worker thread
cricket::TransportChannel* voice_rtp_transport_channel() {
return rtp_transport_channel(voice_channel());
}
cricket::TransportChannel* voice_rtcp_transport_channel() {
return rtcp_transport_channel(voice_channel());
}
cricket::TransportChannel* video_rtp_transport_channel() {
return rtp_transport_channel(video_channel());
}
cricket::TransportChannel* video_rtcp_transport_channel() {
return rtcp_transport_channel(video_channel());
}
cricket::TransportChannel* data_rtp_transport_channel() {
return rtp_transport_channel(data_channel());
}
cricket::TransportChannel* data_rtcp_transport_channel() {
return rtcp_transport_channel(data_channel());
}
using webrtc::WebRtcSession::SetAudioPlayout;
using webrtc::WebRtcSession::SetAudioSend;
using webrtc::WebRtcSession::SetCaptureDevice;
using webrtc::WebRtcSession::SetVideoPlayout;
using webrtc::WebRtcSession::SetVideoSend;
private:
cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
if (!ch) {
return nullptr;
}
return ch->transport_channel();
}
cricket::TransportChannel* rtcp_transport_channel(cricket::BaseChannel* ch) {
if (!ch) {
return nullptr;
}
return ch->rtcp_transport_channel();
}
};
class WebRtcSessionCreateSDPObserverForTest
: public rtc::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:
rtc::scoped_ptr<SessionDescriptionInterface> description_;
State state_;
};
class FakeAudioRenderer : public cricket::AudioRenderer {
public:
FakeAudioRenderer() : sink_(NULL) {}
virtual ~FakeAudioRenderer() {
if (sink_)
sink_->OnClose();
}
void SetSink(Sink* sink) override { sink_ = sink; }
cricket::AudioRenderer::Sink* sink() const { return sink_; }
private:
cricket::AudioRenderer::Sink* sink_;
};
class WebRtcSessionTest
: public testing::TestWithParam<RTCCertificateGenerationMethod>,
public sigslot::has_slots<> {
protected:
// TODO Investigate why ChannelManager crashes, if it's created
// after stun_server.
WebRtcSessionTest()
: media_engine_(new cricket::FakeMediaEngine()),
data_engine_(new cricket::FakeDataEngine()),
channel_manager_(new cricket::ChannelManager(
media_engine_, data_engine_, new cricket::CaptureManager(),
rtc::Thread::Current())),
tdesc_factory_(new cricket::TransportDescriptionFactory()),
desc_factory_(new cricket::MediaSessionDescriptionFactory(
channel_manager_.get(), tdesc_factory_.get())),
pss_(new rtc::PhysicalSocketServer),
vss_(new rtc::VirtualSocketServer(pss_.get())),
fss_(new rtc::FirewallSocketServer(vss_.get())),
ss_scope_(fss_.get()),
stun_socket_addr_(rtc::SocketAddress(kStunAddrHost,
cricket::STUN_SERVER_PORT)),
stun_server_(cricket::TestStunServer::Create(Thread::Current(),
stun_socket_addr_)),
turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
metrics_observer_(new rtc::RefCountedObject<FakeMetricsObserver>()) {
cricket::ServerAddresses stun_servers;
stun_servers.insert(stun_socket_addr_);
allocator_.reset(new cricket::BasicPortAllocator(
&network_manager_,
stun_servers,
SocketAddress(), SocketAddress(), SocketAddress()));
allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
cricket::PORTALLOCATOR_DISABLE_RELAY);
EXPECT_TRUE(channel_manager_->Init());
desc_factory_->set_add_legacy_streams(false);
allocator_->set_step_delay(cricket::kMinimumStepDelay);
}
void AddInterface(const SocketAddress& addr) {
network_manager_.AddInterface(addr);
}
// If |dtls_identity_store| != null or |rtc_configuration| contains
// |certificates| then DTLS will be enabled unless explicitly disabled by
// |rtc_configuration| options. When DTLS is enabled a certificate will be
// used if provided, otherwise one will be generated using the
// |dtls_identity_store|.
void Init(
rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
ASSERT_TRUE(session_.get() == NULL);
session_.reset(new WebRtcSessionForTest(
channel_manager_.get(), rtc::Thread::Current(), rtc::Thread::Current(),
allocator_.get(), &observer_));
session_->SignalDataChannelOpenMessage.connect(
this, &WebRtcSessionTest::OnDataChannelOpenMessage);
EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
observer_.ice_connection_state_);
EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
observer_.ice_gathering_state_);
EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
dtls_identity_store.Pass(),
rtc_configuration));
session_->set_metrics_observer(metrics_observer_);
}
void OnDataChannelOpenMessage(const std::string& label,
const InternalDataChannelInit& config) {
last_data_channel_label_ = label;
last_data_channel_config_ = config;
}
void Init() {
PeerConnectionInterface::RTCConfiguration configuration;
Init(nullptr, configuration);
}
void InitWithIceTransport(
PeerConnectionInterface::IceTransportsType ice_transport_type) {
PeerConnectionInterface::RTCConfiguration configuration;
configuration.type = ice_transport_type;
Init(nullptr, configuration);
}
void InitWithBundlePolicy(
PeerConnectionInterface::BundlePolicy bundle_policy) {
PeerConnectionInterface::RTCConfiguration configuration;
configuration.bundle_policy = bundle_policy;
Init(nullptr, configuration);
}
void InitWithRtcpMuxPolicy(
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy) {
PeerConnectionInterface::RTCConfiguration configuration;
configuration.rtcp_mux_policy = rtcp_mux_policy;
Init(nullptr, configuration);
}
// Successfully init with DTLS; with a certificate generated and supplied or
// with a store that generates it for us.
void InitWithDtls(RTCCertificateGenerationMethod cert_gen_method) {
rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store;
PeerConnectionInterface::RTCConfiguration configuration;
if (cert_gen_method == ALREADY_GENERATED) {
configuration.certificates.push_back(
FakeDtlsIdentityStore::GenerateCertificate());
} else if (cert_gen_method == DTLS_IDENTITY_STORE) {
dtls_identity_store.reset(new FakeDtlsIdentityStore());
dtls_identity_store->set_should_fail(false);
} else {
RTC_CHECK(false);
}
Init(dtls_identity_store.Pass(), configuration);
}
// Init with DTLS with a store that will fail to generate a certificate.
void InitWithDtlsIdentityGenFail() {
rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
new FakeDtlsIdentityStore());
dtls_identity_store->set_should_fail(true);
PeerConnectionInterface::RTCConfiguration configuration;
Init(dtls_identity_store.Pass(), configuration);
}
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();
}
void SendAudioVideoStream1() {
send_stream_1_ = true;
send_stream_2_ = false;
send_audio_ = true;
send_video_ = true;
}
void SendAudioVideoStream2() {
send_stream_1_ = false;
send_stream_2_ = true;
send_audio_ = true;
send_video_ = true;
}
void SendAudioVideoStream1And2() {
send_stream_1_ = true;
send_stream_2_ = true;
send_audio_ = true;
send_video_ = true;
}
void SendNothing() {
send_stream_1_ = false;
send_stream_2_ = false;
send_audio_ = false;
send_video_ = false;
}
void SendAudioOnlyStream2() {
send_stream_1_ = false;
send_stream_2_ = true;
send_audio_ = true;
send_video_ = false;
}
void SendVideoOnlyStream2() {
send_stream_1_ = false;
send_stream_2_ = true;
send_audio_ = false;
send_video_ = true;
}
void AddStreamsToOptions(cricket::MediaSessionOptions* session_options) {
if (send_stream_1_ && send_audio_) {
session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, kAudioTrack1,
kStream1);
}
if (send_stream_1_ && send_video_) {
session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrack1,
kStream1);
}
if (send_stream_2_ && send_audio_) {
session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, kAudioTrack2,
kStream2);
}
if (send_stream_2_ && send_video_) {
session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrack2,
kStream2);
}
if (data_channel_ && session_->data_channel_type() == cricket::DCT_RTP) {
session_options->AddSendStream(cricket::MEDIA_TYPE_DATA,
data_channel_->label(),
data_channel_->label());
}
}
void GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
AddStreamsToOptions(session_options);
ASSERT_TRUE(ConvertRtcOptionsForOffer(rtc_options, session_options));
if (session_->data_channel_type() == cricket::DCT_SCTP && data_channel_) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
}
void GetOptionsForAnswer(const webrtc::MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options) {
AddStreamsToOptions(session_options);
session_options->recv_audio = false;
session_options->recv_video = false;
ASSERT_TRUE(ParseConstraintsForAnswer(constraints, session_options));
if (session_->data_channel_type() == cricket::DCT_SCTP) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
}
// Creates a local offer and applies it. Starts ICE.
// Call SendAudioVideoStreamX() before this function
// to decide which streams to create.
void InitiateCall() {
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
EXPECT_TRUE_WAIT(PeerConnectionInterface::kIceGatheringNew !=
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
}
SessionDescriptionInterface* CreateOffer() {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
return CreateOffer(options);
}
SessionDescriptionInterface* CreateOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
observer = new WebRtcSessionCreateSDPObserverForTest();
cricket::MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
session_->CreateOffer(observer, options, session_options);
EXPECT_TRUE_WAIT(
observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
2000);
return observer->ReleaseDescription();
}
SessionDescriptionInterface* CreateAnswer(
const webrtc::MediaConstraintsInterface* constraints) {
rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observer
= new WebRtcSessionCreateSDPObserverForTest();
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(constraints, &session_options);
session_->CreateAnswer(observer, constraints, session_options);
EXPECT_TRUE_WAIT(
observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
2000);
return observer->ReleaseDescription();
}
bool ChannelsExist() const {
return (session_->voice_channel() != NULL &&
session_->video_channel() != 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::kMediaProtocolDtlsSavpf),
audio_content->protocol());
EXPECT_EQ(std::string(cricket::kMediaProtocolDtlsSavpf),
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_DISABLED);
std::string identity_name = "WebRTC" +
rtc::ToString(rtc::CreateRandomId());
// Confirmed to work with KT_RSA and KT_ECDSA.
tdesc_factory_->set_certificate(rtc::RTCCertificate::Create(
rtc::scoped_ptr<rtc::SSLIdentity>(rtc::SSLIdentity::Generate(
identity_name, rtc::KT_DEFAULT)).Pass()));
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 an SDP without Crypto.
cricket::MediaSessionOptions options;
options.recv_video = true;
JsepSessionDescription* offer(
CreateRemoteOffer(options, cricket::SEC_DISABLED));
ASSERT_TRUE(offer != NULL);
VerifyNoCryptoParams(offer->description(), false);
SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesCrypto,
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.recv_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 SetAndVerifyNumUnsignalledRecvStreams(
int value_set, int value_expected) {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams,
value_set);
session_.reset();
Init();
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
video_channel_ = media_engine_->GetVideoChannel(0);
ASSERT_TRUE(video_channel_ != NULL);
const cricket::VideoOptions& video_options = video_channel_->options();
EXPECT_EQ(value_expected,
video_options.unsignalled_recv_stream_limit.GetWithDefaultIfUnset(-1));
}
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";
rtc::replace_substrs(ufrag_line.c_str(), ufrag_line.length(),
"", 0,
sdp);
rtc::replace_substrs(pwd_line.c_str(), pwd_line.length(),
"", 0,
sdp);
}
}
void ModifyIceUfragPwdLines(const SessionDescriptionInterface* current_desc,
const std::string& modified_ice_ufrag,
const std::string& modified_ice_pwd,
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 |modified_ice_ufrag| and
// |modified_ice_pwd| 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";
std::string mod_ufrag = "a=ice-ufrag:" + modified_ice_ufrag + "\r\n";
std::string mod_pwd = "a=ice-pwd:" + modified_ice_pwd + "\r\n";
rtc::replace_substrs(ufrag_line.c_str(), ufrag_line.length(),
mod_ufrag.c_str(), mod_ufrag.length(),
sdp);
rtc::replace_substrs(pwd_line.c_str(), pwd_line.length(),
mod_pwd.c_str(), mod_pwd.length(),
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 SendAudioVideoStreamX() 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));
session_->MaybeStartGathering();
}
void SetLocalDescriptionExpectState(SessionDescriptionInterface* desc,
BaseSession::State expected_state) {
SetLocalDescriptionWithoutError(desc);
EXPECT_EQ(expected_state, session_->state());
}
void SetLocalDescriptionExpectError(const std::string& action,
const std::string& expected_error,
SessionDescriptionInterface* desc) {
std::string error;
EXPECT_FALSE(session_->SetLocalDescription(desc, &error));
std::string sdp_type = "local ";
sdp_type.append(action);
EXPECT_NE(std::string::npos, error.find(sdp_type));
EXPECT_NE(std::string::npos, error.find(expected_error));
}
void SetLocalDescriptionOfferExpectError(const std::string& expected_error,
SessionDescriptionInterface* desc) {
SetLocalDescriptionExpectError(SessionDescriptionInterface::kOffer,
expected_error, desc);
}
void SetLocalDescriptionAnswerExpectError(const std::string& expected_error,
SessionDescriptionInterface* desc) {
SetLocalDescriptionExpectError(SessionDescriptionInterface::kAnswer,
expected_error, desc);
}
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& action,
const std::string& expected_error,
SessionDescriptionInterface* desc) {
std::string error;
EXPECT_FALSE(session_->SetRemoteDescription(desc, &error));
std::string sdp_type = "remote ";
sdp_type.append(action);
EXPECT_NE(std::string::npos, error.find(sdp_type));
EXPECT_NE(std::string::npos, error.find(expected_error));
}
void SetRemoteDescriptionOfferExpectError(
const std::string& expected_error, SessionDescriptionInterface* desc) {
SetRemoteDescriptionExpectError(SessionDescriptionInterface::kOffer,
expected_error, desc);
}
void SetRemoteDescriptionPranswerExpectError(
const std::string& expected_error, SessionDescriptionInterface* desc) {
SetRemoteDescriptionExpectError(SessionDescriptionInterface::kPrAnswer,
expected_error, desc);
}
void SetRemoteDescriptionAnswerExpectError(
const std::string& expected_error, SessionDescriptionInterface* desc) {
SetRemoteDescriptionExpectError(SessionDescriptionInterface::kAnswer,
expected_error, desc);
}
void CreateCryptoOfferAndNonCryptoAnswer(SessionDescriptionInterface** offer,
SessionDescriptionInterface** nocrypto_answer) {
// Create a SDP without Crypto.
cricket::MediaSessionOptions options;
options.recv_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);
}
void CreateDtlsOfferAndNonDtlsAnswer(SessionDescriptionInterface** offer,
SessionDescriptionInterface** nodtls_answer) {
cricket::MediaSessionOptions options;
options.recv_video = true;
options.bundle_enabled = true;
rtc::scoped_ptr<SessionDescriptionInterface> temp_offer(
CreateRemoteOffer(options, cricket::SEC_ENABLED));
*nodtls_answer =
CreateRemoteAnswer(temp_offer.get(), options, cricket::SEC_ENABLED);
EXPECT_TRUE(*nodtls_answer != NULL);
VerifyFingerprintStatus((*nodtls_answer)->description(), false);
VerifyCryptoParams((*nodtls_answer)->description());
SetFactoryDtlsSrtp();
*offer = CreateRemoteOffer(options, cricket::SEC_ENABLED);
ASSERT_TRUE(*offer != NULL);
VerifyFingerprintStatus((*offer)->description(), true);
VerifyCryptoParams((*offer)->description());
}
JsepSessionDescription* CreateRemoteOfferWithVersion(
cricket::MediaSessionOptions options,
cricket::SecurePolicy secure_policy,
const std::string& session_version,
const SessionDescriptionInterface* current_desc) {
std::string session_id = rtc::ToString(rtc::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 sdes_policy) {
return CreateRemoteOfferWithVersion(
options, sdes_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.AddSendStream(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];
rtc::sprintfn(new_port_str, sizeof(new_port_str), "%d", new_port);
std::string offer_str;
offer_basis->ToString(&offer_str);
rtc::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 SendAudioVideoStreamX()
// before this function to decide which streams to create.
JsepSessionDescription* CreateRemoteOffer() {
cricket::MediaSessionOptions options;
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 =
rtc::ToString(rtc::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.
// Call SendAudioVideoStreamX() before this function
// to decide which streams to create.
JsepSessionDescription* CreateRemoteAnswer(
const SessionDescriptionInterface* offer) {
cricket::MediaSessionOptions options;
GetOptionsForAnswer(NULL, &options);
return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
}
void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) {
AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init();
SendAudioVideoStream1();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.use_rtp_mux = bundle;
SessionDescriptionInterface* offer = CreateOffer(options);
// SetLocalDescription and SetRemoteDescriptions takes ownership of offer
// and answer.
SetLocalDescriptionWithoutError(offer);
rtc::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";
rtc::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());
if (bundle) {
EXPECT_EQ(0, observer_.mline_1_candidates_.size());
} else {
EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
}
}
// Tests that we can only send DTMF when the dtmf codec is supported.
void TestCanInsertDtmf(bool can) {
if (can) {
InitWithDtmfCodec();
} else {
Init();
}
SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
EXPECT_FALSE(session_->CanInsertDtmf(""));
EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
}
// Helper class to configure loopback network and verify Best
// Connection using right IP protocol for TestLoopbackCall
// method. LoopbackNetworkManager applies firewall rules to block
// all ping traffic once ICE completed, and remove them to observe
// ICE reconnected again. This LoopbackNetworkConfiguration struct
// verifies the best connection is using the right IP protocol after
// initial ICE convergences.
class LoopbackNetworkConfiguration {
public:
LoopbackNetworkConfiguration()
: test_ipv6_network_(false),
test_extra_ipv4_network_(false),
best_connection_after_initial_ice_converged_(1, 0) {}
// Used to track the expected best connection count in each IP protocol.
struct ExpectedBestConnection {
ExpectedBestConnection(int ipv4_count, int ipv6_count)
: ipv4_count_(ipv4_count),
ipv6_count_(ipv6_count) {}
int ipv4_count_;
int ipv6_count_;
};
bool test_ipv6_network_;
bool test_extra_ipv4_network_;
ExpectedBestConnection best_connection_after_initial_ice_converged_;
void VerifyBestConnectionAfterIceConverge(
const rtc::scoped_refptr<FakeMetricsObserver> metrics_observer) const {
Verify(metrics_observer, best_connection_after_initial_ice_converged_);
}
private:
void Verify(const rtc::scoped_refptr<FakeMetricsObserver> metrics_observer,
const ExpectedBestConnection& expected) const {
EXPECT_EQ(
metrics_observer->GetEnumCounter(webrtc::kEnumCounterAddressFamily,
webrtc::kBestConnections_IPv4),
expected.ipv4_count_);
EXPECT_EQ(
metrics_observer->GetEnumCounter(webrtc::kEnumCounterAddressFamily,
webrtc::kBestConnections_IPv6),
expected.ipv6_count_);
// This is used in the loopback call so there is only single host to host
// candidate pair.
EXPECT_EQ(metrics_observer->GetEnumCounter(
webrtc::kEnumCounterIceCandidatePairTypeUdp,
webrtc::kIceCandidatePairHostHost),
0);
EXPECT_EQ(metrics_observer->GetEnumCounter(
webrtc::kEnumCounterIceCandidatePairTypeUdp,
webrtc::kIceCandidatePairHostPublicHostPublic),
1);
}
};
class LoopbackNetworkManager {
public:
LoopbackNetworkManager(WebRtcSessionTest* session,
const LoopbackNetworkConfiguration& config)
: config_(config) {
session->AddInterface(
rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
if (config_.test_extra_ipv4_network_) {
session->AddInterface(
rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
}
if (config_.test_ipv6_network_) {
session->AddInterface(
rtc::SocketAddress(kClientIPv6AddrHost1, kClientAddrPort));
}
}
void ApplyFirewallRules(rtc::FirewallSocketServer* fss) {
fss->AddRule(false, rtc::FP_ANY, rtc::FD_ANY,
rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
if (config_.test_extra_ipv4_network_) {
fss->AddRule(false, rtc::FP_ANY, rtc::FD_ANY,
rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
}
if (config_.test_ipv6_network_) {
fss->AddRule(false, rtc::FP_ANY, rtc::FD_ANY,
rtc::SocketAddress(kClientIPv6AddrHost1, kClientAddrPort));
}
}
void ClearRules(rtc::FirewallSocketServer* fss) { fss->ClearRules(); }
private:
LoopbackNetworkConfiguration config_;
};
// 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, and then a permanent disconnection.
// This code is placed in 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) -> Completed -> Disconnected -> Completed
// -> Failed.
// The Gathering state should go: New -> Gathering -> Completed.
void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
LoopbackNetworkManager loopback_network_manager(this, config);
Init();
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
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, nullptr);
ASSERT_TRUE(desc != NULL);
SetRemoteDescriptionWithoutError(desc);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
// The ice connection state is "Connected" too briefly to catch in a test.
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
config.VerifyBestConnectionAfterIceConverge(metrics_observer_);
// Adding firewall rule to block ping requests, which should cause
// transport channel failure.
loopback_network_manager.ApplyFirewallRules(fss_.get());
LOG(LS_INFO) << "Firewall Rules applied";
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
metrics_observer_->Reset();
// Clearing the rules, session should move back to completed state.
loopback_network_manager.ClearRules(fss_.get());
LOG(LS_INFO) << "Firewall Rules cleared";
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
// Now we block ping requests and wait until the ICE connection transitions
// to the Failed state. This will take at least 30 seconds because it must
// wait for the Port to timeout.
int port_timeout = 30000;
loopback_network_manager.ApplyFirewallRules(fss_.get());
LOG(LS_INFO) << "Firewall Rules applied again";
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
observer_.ice_connection_state_,
kIceCandidatesTimeout + port_timeout);
}
void TestLoopbackCall() {
LoopbackNetworkConfiguration config;
TestLoopbackCall(config);
}
// 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 CreateDataChannel() {
webrtc::InternalDataChannelInit dci;
dci.reliable = session_->data_channel_type() == cricket::DCT_SCTP;
data_channel_ = DataChannel::Create(
session_.get(), session_->data_channel_type(), "datachannel", dci);
}
void SetLocalDescriptionWithDataChannel() {
CreateDataChannel();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
}
void VerifyMultipleAsyncCreateDescription(
RTCCertificateGenerationMethod cert_gen_method,
CreateSessionDescriptionRequest::Type type) {
InitWithDtls(cert_gen_method);
VerifyMultipleAsyncCreateDescriptionAfterInit(true, type);
}
void VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
CreateSessionDescriptionRequest::Type type) {
InitWithDtlsIdentityGenFail();
VerifyMultipleAsyncCreateDescriptionAfterInit(false, type);
}
void VerifyMultipleAsyncCreateDescriptionAfterInit(
bool success, CreateSessionDescriptionRequest::Type type) {
RTC_CHECK(session_);
SetFactoryDtlsSrtp();
if (type == CreateSessionDescriptionRequest::kAnswer) {
cricket::MediaSessionOptions options;
scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options, cricket::SEC_DISABLED));
ASSERT_TRUE(offer.get() != NULL);
SetRemoteDescriptionWithoutError(offer.release());
}
PeerConnectionInterface::RTCOfferAnswerOptions options;
cricket::MediaSessionOptions session_options;
const int kNumber = 3;
rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
observers[kNumber];
for (int i = 0; i < kNumber; ++i) {
observers[i] = new WebRtcSessionCreateSDPObserverForTest();
if (type == CreateSessionDescriptionRequest::kOffer) {
session_->CreateOffer(observers[i], options, session_options);
} else {
session_->CreateAnswer(observers[i], nullptr, session_options);
}
}
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);
}
}
}
void ConfigureAllocatorWithTurn() {
cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
relay_server.credentials = credentials;
relay_server.ports.push_back(cricket::ProtocolAddress(
kTurnUdpIntAddr, cricket::PROTO_UDP, false));
allocator_->AddRelay(relay_server);
allocator_->set_step_delay(cricket::kMinimumStepDelay);
allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP);
}
cricket::FakeMediaEngine* media_engine_;
cricket::FakeDataEngine* data_engine_;
rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
rtc::scoped_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
rtc::scoped_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
rtc::scoped_ptr<rtc::FirewallSocketServer> fss_;
rtc::SocketServerScope ss_scope_;
rtc::SocketAddress stun_socket_addr_;
rtc::scoped_ptr<cricket::TestStunServer> stun_server_;
cricket::TestTurnServer turn_server_;
rtc::FakeNetworkManager network_manager_;
rtc::scoped_ptr<cricket::BasicPortAllocator> allocator_;
PeerConnectionFactoryInterface::Options options_;
rtc::scoped_ptr<FakeConstraints> constraints_;
rtc::scoped_ptr<WebRtcSessionForTest> session_;
MockIceObserver observer_;
cricket::FakeVideoMediaChannel* video_channel_;
cricket::FakeVoiceMediaChannel* voice_channel_;
rtc::scoped_refptr<FakeMetricsObserver> metrics_observer_;
// The following flags affect options created for CreateOffer/CreateAnswer.
bool send_stream_1_ = false;
bool send_stream_2_ = false;
bool send_audio_ = false;
bool send_video_ = false;
rtc::scoped_refptr<DataChannel> data_channel_;
// Last values received from data channel creation signal.
std::string last_data_channel_label_;
InternalDataChannelInit last_data_channel_config_;
};
TEST_P(WebRtcSessionTest, TestInitializeWithDtls) {
InitWithDtls(GetParam());
// SDES is disabled when DTLS is on.
EXPECT_EQ(cricket::SEC_DISABLED, session_->SdesPolicy());
}
TEST_F(WebRtcSessionTest, TestInitializeWithoutDtls) {
Init();
// SDES is required if DTLS is off.
EXPECT_EQ(cricket::SEC_REQUIRED, session_->SdesPolicy());
}
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(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
AddInterface(rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
Init();
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(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
AddInterface(rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
fss_->AddRule(false,
rtc::FP_UDP,
rtc::FD_ANY,
rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init();
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 session delivers no candidates gathered when constraint set to "none".
TEST_F(WebRtcSessionTest, TestIceTransportsNone) {
AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
InitWithIceTransport(PeerConnectionInterface::kNone);
SendAudioVideoStream1();
InitiateCall();
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
}
// Test session delivers only relay candidates gathered when constaint set to
// "relay".
TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
ConfigureAllocatorWithTurn();
InitWithIceTransport(PeerConnectionInterface::kRelay);
SendAudioVideoStream1();
InitiateCall();
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ(2u, observer_.mline_0_candidates_.size());
EXPECT_EQ(2u, observer_.mline_1_candidates_.size());
for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
EXPECT_EQ(cricket::RELAY_PORT_TYPE,
observer_.mline_0_candidates_[i].type());
}
for (size_t i = 0; i < observer_.mline_1_candidates_.size(); ++i) {
EXPECT_EQ(cricket::RELAY_PORT_TYPE,
observer_.mline_1_candidates_[i].type());
}
}
// Test session delivers all candidates gathered when constaint set to "all".
TEST_F(WebRtcSessionTest, TestIceTransportsAll) {
AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
InitWithIceTransport(PeerConnectionInterface::kAll);
SendAudioVideoStream1();
InitiateCall();
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
// Host + STUN. By default allocator is disabled to gather relay candidates.
EXPECT_EQ(4u, observer_.mline_0_candidates_.size());
EXPECT_EQ(4u, observer_.mline_1_candidates_.size());
}
TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
Init();
SessionDescriptionInterface* offer = NULL;
// Since |offer| is NULL, there's no way to tell if it's an offer or answer.
std::string unknown_action;
SetLocalDescriptionExpectError(unknown_action, kInvalidSdp, offer);
SetRemoteDescriptionExpectError(unknown_action, kInvalidSdp, offer);
}
// Test creating offers and receive answers and make sure the
// media engine creates the expected send and receive streams.
TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
Init();
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
const std::string session_id_orig = offer->session_id();
const std::string session_version_orig = offer->session_version();
SetLocalDescriptionWithoutError(offer);
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.
SendNothing();
offer = CreateOffer();
// Verify the session id is the same and the session version is
// increased.
EXPECT_EQ(session_id_orig, offer->session_id());
EXPECT_LT(rtc::FromString<uint64_t>(session_version_orig),
rtc::FromString<uint64_t>(offer->session_version()));
SetLocalDescriptionWithoutError(offer);
EXPECT_EQ(0u, video_channel_->send_streams().size());
EXPECT_EQ(0u, voice_channel_->send_streams().size());
SendAudioVideoStream2();
answer = CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionWithoutError(answer);
// 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, TestReceiveSdesOfferCreateSdesAnswer) {
Init();
SendAudioVideoStream2();
SessionDescriptionInterface* offer = CreateOffer();
VerifyCryptoParams(offer->description());
SetRemoteDescriptionWithoutError(offer);
SendAudioVideoStream1();
SessionDescriptionInterface* answer = CreateAnswer(NULL);
VerifyCryptoParams(answer->description());
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);
SendAudioVideoStream1And2();
offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
// Answer by turning off all send streams.
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(rtc::FromString<uint64_t>(session_version_orig),
rtc::FromString<uint64_t>(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_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
Init();
media_engine_->set_fail_create_channel(true);
SessionDescriptionInterface* offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetRemoteDescriptionOfferExpectError(kCreateChannelFailed, offer);
offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
SetLocalDescriptionOfferExpectError(kCreateChannelFailed, offer);
}
//
// Tests for creating/setting SDP under different SDES/DTLS polices:
//
// --DTLS off and SDES on
// TestCreateSdesOfferReceiveSdesAnswer/TestReceiveSdesOfferCreateSdesAnswer:
// set local/remote offer/answer with crypto --> success
// TestSetNonSdesOfferWhenSdesOn: set local/remote offer without crypto --->
// failure
// TestSetLocalNonSdesAnswerWhenSdesOn: set local answer without crypto -->
// failure
// TestSetRemoteNonSdesAnswerWhenSdesOn: set remote answer without crypto -->
// failure
//
// --DTLS on and SDES off
// TestCreateDtlsOfferReceiveDtlsAnswer/TestReceiveDtlsOfferCreateDtlsAnswer:
// set local/remote offer/answer with DTLS fingerprint --> success
// TestReceiveNonDtlsOfferWhenDtlsOn: set local/remote offer without DTLS
// fingerprint --> failure
// TestSetLocalNonDtlsAnswerWhenDtlsOn: set local answer without fingerprint
// --> failure
// TestSetRemoteNonDtlsAnswerWhenDtlsOn: set remote answer without fingerprint
// --> failure
//
// --Encryption disabled: DTLS off and SDES off
// TestCreateOfferReceiveAnswerWithoutEncryption: set local offer and remote
// answer without SDES or DTLS --> success
// TestCreateAnswerReceiveOfferWithoutEncryption: set remote offer and local
// answer without SDES or DTLS --> success
//
// Test that we return a failure when applying a remote/local offer that doesn't
// have cryptos enabled when DTLS is off.
TEST_F(WebRtcSessionTest, TestSetNonSdesOfferWhenSdesOn) {
Init();
cricket::MediaSessionOptions options;
options.recv_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.
SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesCrypto, offer);
offer = CreateRemoteOffer(options, cricket::SEC_DISABLED);
ASSERT_TRUE(offer != NULL);
SetLocalDescriptionOfferExpectError(kSdpWithoutSdesCrypto, offer);
}
// Test that we return a failure when applying a local answer that doesn't have
// cryptos enabled when DTLS is off.
TEST_F(WebRtcSessionTest, TestSetLocalNonSdesAnswerWhenSdesOn) {
Init();
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetRemoteDescriptionWithoutError(offer);
SetLocalDescriptionAnswerExpectError(kSdpWithoutSdesCrypto, answer);
}
// Test we will return fail when apply an remote answer that doesn't have
// crypto enabled when DTLS is off.
TEST_F(WebRtcSessionTest, TestSetRemoteNonSdesAnswerWhenSdesOn) {
Init();
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetLocalDescriptionWithoutError(offer);
SetRemoteDescriptionAnswerExpectError(kSdpWithoutSdesCrypto, answer);
}
// Test that we accept an offer with a DTLS fingerprint when DTLS is on
// and that we return an answer with a DTLS fingerprint.
TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SendAudioVideoStream1();
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
cricket::MediaSessionOptions options;
options.recv_video = true;
JsepSessionDescription* offer =
CreateRemoteOffer(options, cricket::SEC_DISABLED);
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), true);
VerifyNoCryptoParams(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 we set a local offer with a DTLS fingerprint when DTLS is on
// and then we accept a remote answer with a DTLS fingerprint successfully.
TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SendAudioVideoStream1();
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
// Verify that we get a crypto fingerprint in the answer.
SessionDescriptionInterface* offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), true);
// Check that we don't have an a=crypto line in the offer.
VerifyNoCryptoParams(offer->description(), true);
// Now set the local description, which should work, even without a=crypto.
SetLocalDescriptionWithoutError(offer);
cricket::MediaSessionOptions options;
options.recv_video = true;
JsepSessionDescription* answer =
CreateRemoteAnswer(offer, options, cricket::SEC_DISABLED);
ASSERT_TRUE(answer != NULL);
VerifyFingerprintStatus(answer->description(), true);
VerifyNoCryptoParams(answer->description(), true);
// SetRemoteDescription will take the ownership of the answer.
SetRemoteDescriptionWithoutError(answer);
}
// Test that if we support DTLS and the other side didn't offer a fingerprint,
// we will fail to set the remote description.
TEST_P(WebRtcSessionTest, TestReceiveNonDtlsOfferWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
cricket::MediaSessionOptions options;
options.recv_video = true;
options.bundle_enabled = true;
JsepSessionDescription* offer = CreateRemoteOffer(
options, cricket::SEC_REQUIRED);
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), false);
VerifyCryptoParams(offer->description());
// SetRemoteDescription will take the ownership of the offer.
SetRemoteDescriptionOfferExpectError(
kSdpWithoutDtlsFingerprint, offer);
offer = CreateRemoteOffer(options, cricket::SEC_REQUIRED);
// SetLocalDescription will take the ownership of the offer.
SetLocalDescriptionOfferExpectError(
kSdpWithoutDtlsFingerprint, offer);
}
// Test that we return a failure when applying a local answer that doesn't have
// a DTLS fingerprint when DTLS is required.
TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateDtlsOfferAndNonDtlsAnswer(&offer, &answer);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer and answer.
SetRemoteDescriptionWithoutError(offer);
SetLocalDescriptionAnswerExpectError(
kSdpWithoutDtlsFingerprint, answer);
}
// Test that we return a failure when applying a remote answer that doesn't have
// a DTLS fingerprint when DTLS is required.
TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
options.recv_video = true;
rtc::scoped_ptr<SessionDescriptionInterface> temp_offer(
CreateRemoteOffer(options, cricket::SEC_ENABLED));
JsepSessionDescription* answer =
CreateRemoteAnswer(temp_offer.get(), options, cricket::SEC_ENABLED);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer and answer.
SetLocalDescriptionWithoutError(offer);
SetRemoteDescriptionAnswerExpectError(
kSdpWithoutDtlsFingerprint, answer);
}
// Test that we create a local offer without SDES or DTLS and accept a remote
// answer without SDES or DTLS when encryption is disabled.
TEST_P(WebRtcSessionTest, TestCreateOfferReceiveAnswerWithoutEncryption) {
SendAudioVideoStream1();
options_.disable_encryption = true;
InitWithDtls(GetParam());
// Verify that we get a crypto fingerprint in the answer.
SessionDescriptionInterface* offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), false);
// Check that we don't have an a=crypto line in the offer.
VerifyNoCryptoParams(offer->description(), false);
// Now set the local description, which should work, even without a=crypto.
SetLocalDescriptionWithoutError(offer);
cricket::MediaSessionOptions options;
options.recv_video = true;
JsepSessionDescription* answer =
CreateRemoteAnswer(offer, options, cricket::SEC_DISABLED);
ASSERT_TRUE(answer != NULL);
VerifyFingerprintStatus(answer->description(), false);
VerifyNoCryptoParams(answer->description(), false);
// SetRemoteDescription will take the ownership of the answer.
SetRemoteDescriptionWithoutError(answer);
}
// Test that we create a local answer without SDES or DTLS and accept a remote
// offer without SDES or DTLS when encryption is disabled.
TEST_P(WebRtcSessionTest, TestCreateAnswerReceiveOfferWithoutEncryption) {
options_.disable_encryption = true;
InitWithDtls(GetParam());
cricket::MediaSessionOptions options;
options.recv_video = true;
JsepSessionDescription* offer =
CreateRemoteOffer(options, cricket::SEC_DISABLED);
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), false);
VerifyNoCryptoParams(offer->description(), false);
// 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(), false);
// Check that we don't have an a=crypto line in the answer.
VerifyNoCryptoParams(answer->description(), false);
// Now set the local description, which should work, even without a=crypto.
SetLocalDescriptionWithoutError(answer);
}
TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
Init();
SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer2 = CreateOffer();
SetLocalDescriptionWithoutError(offer2);
}
TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
Init();
SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* offer2 = CreateOffer();
SetRemoteDescriptionWithoutError(offer2);
}
TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
Init();
SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
offer = CreateOffer();
SetRemoteDescriptionOfferExpectError(
"Called in wrong state: STATE_SENTINITIATE", offer);
}
TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
Init();
SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
offer = CreateOffer();
SetLocalDescriptionOfferExpectError(
"Called in wrong state: STATE_RECEIVEDINITIATE", offer);
}
TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
Init();
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);
SendAudioVideoStream1();
JsepSessionDescription* pranswer2 = static_cast<JsepSessionDescription*>(
CreateAnswer(NULL));
pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
SetLocalDescriptionExpectState(pranswer2, BaseSession::STATE_SENTPRACCEPT);
SendAudioVideoStream2();
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionExpectState(answer, BaseSession::STATE_SENTACCEPT);
}
TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
Init();
SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
JsepSessionDescription* pranswer =
CreateRemoteAnswer(session_->local_description());
pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
SetRemoteDescriptionExpectState(pranswer,
BaseSession::STATE_RECEIVEDPRACCEPT);
SendAudioVideoStream1();
JsepSessionDescription* pranswer2 =
CreateRemoteAnswer(session_->local_description());
pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
SetRemoteDescriptionExpectState(pranswer2,
BaseSession::STATE_RECEIVEDPRACCEPT);
SendAudioVideoStream2();
SessionDescriptionInterface* answer =
CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionExpectState(answer, BaseSession::STATE_RECEIVEDACCEPT);
}
TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
Init();
SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer.get());
SetLocalDescriptionAnswerExpectError("Called in wrong state: STATE_INIT",
answer);
}
TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
Init();
SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer.get());
SetRemoteDescriptionAnswerExpectError(
"Called in wrong state: STATE_INIT", answer);
}
TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
Init();
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();
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());
// |ice_candidate3| is identical to |ice_candidate2|. It can be added
// successfully, but the total count of candidates will not increase.
candidate.set_component(2);
JsepIceCandidate ice_candidate3(kMediaContentName0, 0, candidate);
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate3));
ASSERT_EQ(2u, 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();
cricket::Candidate candidate1;
candidate1.set_component(1);
JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
candidate1);
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(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init();
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_EQ(0u, candidates->count());
// Update the session descriptions.
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_EQ(0u, candidates->count());
}
// Test that we can set a remote session description with remote candidates.
TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
Init();
cricket::Candidate candidate1;
candidate1.set_component(1);
JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
candidate1);
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
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(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init();
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);
rtc::scoped_ptr<SessionDescriptionInterface> local_offer(CreateOffer());
ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->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());
SetLocalDescriptionWithoutError(answer);
}
// Verifies TransportProxy and media channels are created with content names
// present in the SessionDescription.
TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
Init();
SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
// 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|.
rtc::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
kAudioMidReplaceStr.c_str(),
kAudioMidReplaceStr.length(),
&sdp);
// Replacing |video| with |video_content_name|.
rtc::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);
cricket::TransportChannel* voice_transport_channel =
session_->voice_rtp_transport_channel();
EXPECT_TRUE(voice_transport_channel != NULL);
EXPECT_EQ(voice_transport_channel->transport_name(), "audio_content_name");
cricket::TransportChannel* video_transport_channel =
session_->video_rtp_transport_channel();
EXPECT_TRUE(video_transport_channel != NULL);
EXPECT_EQ(video_transport_channel->transport_name(), "video_content_name");
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();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
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();
// Test Audio only offer.
SendAudioOnlyStream2();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
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.
SendAudioVideoStream1();
offer.reset(CreateOffer());
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();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
options.offer_to_receive_video = 0;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(options));
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();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(options));
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();
// Test Audio / Video offer.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
options.offer_to_receive_video =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(options));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content != NULL);
// Sets constraints to false and verifies that audio/video contents are
// removed.
options.offer_to_receive_audio = 0;
options.offer_to_receive_video = 0;
offer.reset(CreateOffer(options));
content = cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content == NULL);
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content == NULL);
}
// Test that an answer can not be created if the last remote description is not
// an offer.
TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
Init();
SessionDescriptionInterface* offer = CreateOffer();
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();
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
rtc::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();
// Create a remote offer with audio only.
cricket::MediaSessionOptions options;
rtc::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
ASSERT_TRUE(cricket::GetFirstAudioContent(offer->description()) != NULL);
SetRemoteDescriptionWithoutError(offer.release());
rtc::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();
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
// Test with a stream with tracks.
SendAudioVideoStream1();
rtc::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();
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
webrtc::FakeConstraints constraints_no_receive;
constraints_no_receive.SetMandatoryReceiveAudio(false);
constraints_no_receive.SetMandatoryReceiveVideo(false);
rtc::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();
// Create a remote offer with audio and video content.
rtc::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.
SendAudioVideoStream1();
rtc::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();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
options.voice_activity_detection = false;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(options));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
EXPECT_TRUE(VerifyNoCNCodecs(content));
}
TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
AddCNCodecs();
Init();
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
webrtc::FakeConstraints constraints;
constraints.SetOptionalVAD(false);
rtc::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();
EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
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.
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.
SendAudioOnlyStream2();
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();
EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
options.recv_audio = false;
options.recv_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.
SendAudioVideoStream2();
CreateAndSetRemoteOfferAndLocalAnswer();
voice_channel_ = media_engine_