Index: talk/app/webrtc/webrtcsession_unittest.cc |
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc |
deleted file mode 100644 |
index 8d8a78253810826b097ee2e35730415b094162f9..0000000000000000000000000000000000000000 |
--- a/talk/app/webrtc/webrtcsession_unittest.cc |
+++ /dev/null |
@@ -1,4302 +0,0 @@ |
-/* |
- * 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 <utility> |
-#include <vector> |
- |
-#include "talk/app/webrtc/audiotrack.h" |
-#include "talk/app/webrtc/fakemediacontroller.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/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/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" |
-#include "webrtc/media/base/fakemediaengine.h" |
-#include "webrtc/media/base/fakevideorenderer.h" |
-#include "webrtc/media/base/mediachannel.h" |
-#include "webrtc/media/webrtc/fakewebrtccall.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" |
- |
-#define MAYBE_SKIP_TEST(feature) \ |
- if (!(feature())) { \ |
- LOG(LS_INFO) << "Feature disabled... skipping"; \ |
- return; \ |
- } |
- |
-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::SessionStats; |
-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 }; |
- |
-class MockIceObserver : public webrtc::IceObserver { |
- public: |
- MockIceObserver() |
- : oncandidatesready_(false), |
- ice_connection_state_(PeerConnectionInterface::kIceConnectionNew), |
- ice_gathering_state_(PeerConnectionInterface::kIceGatheringNew) { |
- } |
- |
- void OnIceConnectionChange( |
- PeerConnectionInterface::IceConnectionState new_state) override { |
- ice_connection_state_ = new_state; |
- } |
- void OnIceGatheringChange( |
- PeerConnectionInterface::IceGatheringState new_state) override { |
- // We can never transition back to "new". |
- EXPECT_NE(PeerConnectionInterface::kIceGatheringNew, new_state); |
- ice_gathering_state_ = new_state; |
- oncandidatesready_ = |
- new_state == PeerConnectionInterface::kIceGatheringComplete; |
- } |
- |
- // Found a new candidate. |
- void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { |
- 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_); |
- } |
- |
- 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(webrtc::MediaControllerInterface* media_controller, |
- rtc::Thread* signaling_thread, |
- rtc::Thread* worker_thread, |
- cricket::PortAllocator* port_allocator, |
- webrtc::IceObserver* ice_observer) |
- : WebRtcSession(media_controller, |
- 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())), |
- fake_call_(webrtc::Call::Config()), |
- media_controller_( |
- webrtc::MediaControllerInterface::Create(rtc::Thread::Current(), |
- channel_manager_.get())), |
- 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( |
- media_controller_.get(), rtc::Thread::Current(), rtc::Thread::Current(), |
- allocator_.get(), &observer_)); |
- session_->SignalDataChannelOpenMessage.connect( |
- this, &WebRtcSessionTest::OnDataChannelOpenMessage); |
- session_->GetOnDestroyedSignal()->connect( |
- this, &WebRtcSessionTest::OnSessionDestroyed); |
- |
- EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew, |
- observer_.ice_connection_state_); |
- EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew, |
- observer_.ice_gathering_state_); |
- |
- EXPECT_TRUE(session_->Initialize(options_, constraints_.get(), |
- std::move(dtls_identity_store), |
- 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 OnSessionDestroyed() { session_destroyed_ = true; } |
- |
- 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(std::move(dtls_identity_store), 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(std::move(dtls_identity_store), 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) { |
- ASSERT_TRUE(ConvertRtcOptionsForOffer(rtc_options, session_options)); |
- |
- AddStreamsToOptions(session_options); |
- if (rtc_options.offer_to_receive_audio == |
- RTCOfferAnswerOptions::kUndefined) { |
- session_options->recv_audio = |
- session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO); |
- } |
- if (rtc_options.offer_to_receive_video == |
- RTCOfferAnswerOptions::kUndefined) { |
- session_options->recv_video = |
- session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO); |
- } |
- session_options->bundle_enabled = |
- session_options->bundle_enabled && |
- (session_options->has_audio() || session_options->has_video() || |
- session_options->has_data()); |
- |
- 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) { |
- session_options->recv_audio = false; |
- session_options->recv_video = false; |
- ASSERT_TRUE(ParseConstraintsForAnswer(constraints, session_options)); |
- |
- AddStreamsToOptions(session_options); |
- session_options->bundle_enabled = |
- session_options->bundle_enabled && |
- (session_options->has_audio() || session_options->has_video() || |
- session_options->has_data()); |
- |
- 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)))); |
- 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 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, |
- WebRtcSession::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, |
- WebRtcSession::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 SetupLoopbackCall() { |
- 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); |
- } |
- |
- void TestLoopbackCall(const LoopbackNetworkConfiguration& config) { |
- LoopbackNetworkManager loopback_network_manager(this, config); |
- SetupLoopbackCall(); |
- 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); |
- } |
- |
- void TestPacketOptions() { |
- media_controller_.reset( |
- new cricket::FakeMediaController(channel_manager_.get(), &fake_call_)); |
- LoopbackNetworkConfiguration config; |
- LoopbackNetworkManager loopback_network_manager(this, config); |
- |
- SetupLoopbackCall(); |
- |
- uint8_t test_packet[15] = {0}; |
- rtc::PacketOptions options; |
- options.packet_id = 10; |
- media_engine_->GetVideoChannel(0) |
- ->SendRtp(test_packet, sizeof(test_packet), options); |
- |
- const int kPacketTimeout = 2000; |
- EXPECT_EQ_WAIT(fake_call_.last_sent_packet().packet_id, 10, kPacketTimeout); |
- EXPECT_GT(fake_call_.last_sent_packet().send_time_ms, -1); |
- } |
- |
- // 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 turn_server(cricket::RELAY_TURN); |
- cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword); |
- turn_server.credentials = credentials; |
- turn_server.ports.push_back( |
- cricket::ProtocolAddress(kTurnUdpIntAddr, cricket::PROTO_UDP, false)); |
- allocator_->AddTurnServer(turn_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_; |
- cricket::FakeCall fake_call_; |
- rtc::scoped_ptr<webrtc::MediaControllerInterface> media_controller_; |
- 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_; |
- bool session_destroyed_ = false; |
-}; |
- |
-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()); |
-} |
- |
-// Crashes on Win only. See webrtc:5411. |
-#if defined(WEBRTC_WIN) |
-#define MAYBE_TestStunError DISABLED_TestStunError |
-#else |
-#define MAYBE_TestStunError TestStunError |
-#endif |
-TEST_F(WebRtcSessionTest, MAYBE_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 that we can create and set an answer correctly when different |
-// SSL roles have been negotiated for different transports. |
-// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4525 |
-TEST_P(WebRtcSessionTest, TestCreateAnswerWithDifferentSslRoles) { |
- SendAudioVideoStream1(); |
- InitWithDtls(GetParam()); |
- SetFactoryDtlsSrtp(); |
- |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionWithoutError(offer); |
- |
- cricket::MediaSessionOptions options; |
- options.recv_video = true; |
- |
- // First, negotiate different SSL roles. |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(offer, options, cricket::SEC_DISABLED); |
- TransportInfo* audio_transport_info = |
- answer->description()->GetTransportInfoByName("audio"); |
- audio_transport_info->description.connection_role = |
- cricket::CONNECTIONROLE_ACTIVE; |
- TransportInfo* video_transport_info = |
- answer->description()->GetTransportInfoByName("video"); |
- video_transport_info->description.connection_role = |
- cricket::CONNECTIONROLE_PASSIVE; |
- SetRemoteDescriptionWithoutError(answer); |
- |
- // Now create an offer in the reverse direction, and ensure the initial |
- // offerer responds with an answer with correct SSL roles. |
- offer = CreateRemoteOfferWithVersion(options, cricket::SEC_DISABLED, |
- kSessionVersion, |
- session_->remote_description()); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- answer = CreateAnswer(nullptr); |
- audio_transport_info = answer->description()->GetTransportInfoByName("audio"); |
- EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE, |
- audio_transport_info->description.connection_role); |
- video_transport_info = answer->description()->GetTransportInfoByName("video"); |
- EXPECT_EQ(cricket::CONNECTIONROLE_ACTIVE, |
- video_transport_info->description.connection_role); |
- SetLocalDescriptionWithoutError(answer); |
- |
- // Lastly, start BUNDLE-ing on "audio", expecting that the "passive" role of |
- // audio is transferred over to video in the answer that completes the BUNDLE |
- // negotiation. |
- options.bundle_enabled = true; |
- offer = CreateRemoteOfferWithVersion(options, cricket::SEC_DISABLED, |
- kSessionVersion, |
- session_->remote_description()); |
- SetRemoteDescriptionWithoutError(offer); |
- answer = CreateAnswer(nullptr); |
- audio_transport_info = answer->description()->GetTransportInfoByName("audio"); |
- EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE, |
- audio_transport_info->description.connection_role); |
- video_transport_info = answer->description()->GetTransportInfoByName("video"); |
- EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE, |
- video_transport_info->description.connection_role); |
- 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_SENTOFFER", |
- offer); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) { |
- Init(); |
- SendNothing(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetRemoteDescriptionWithoutError(offer); |
- offer = CreateOffer(); |
- SetLocalDescriptionOfferExpectError( |
- "Called in wrong state: STATE_RECEIVEDOFFER", offer); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) { |
- Init(); |
- SendNothing(); |
- SessionDescriptionInterface* offer = CreateRemoteOffer(); |
- SetRemoteDescriptionExpectState(offer, WebRtcSession::STATE_RECEIVEDOFFER); |
- |
- JsepSessionDescription* pranswer = static_cast<JsepSessionDescription*>( |
- CreateAnswer(NULL)); |
- pranswer->set_type(SessionDescriptionInterface::kPrAnswer); |
- SetLocalDescriptionExpectState(pranswer, WebRtcSession::STATE_SENTPRANSWER); |
- |
- SendAudioVideoStream1(); |
- JsepSessionDescription* pranswer2 = static_cast<JsepSessionDescription*>( |
- CreateAnswer(NULL)); |
- pranswer2->set_type(SessionDescriptionInterface::kPrAnswer); |
- |
- SetLocalDescriptionExpectState(pranswer2, WebRtcSession::STATE_SENTPRANSWER); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = CreateAnswer(NULL); |
- SetLocalDescriptionExpectState(answer, WebRtcSession::STATE_INPROGRESS); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) { |
- Init(); |
- SendNothing(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionExpectState(offer, WebRtcSession::STATE_SENTOFFER); |
- |
- JsepSessionDescription* pranswer = |
- CreateRemoteAnswer(session_->local_description()); |
- pranswer->set_type(SessionDescriptionInterface::kPrAnswer); |
- |
- SetRemoteDescriptionExpectState(pranswer, |
- WebRtcSession::STATE_RECEIVEDPRANSWER); |
- |
- SendAudioVideoStream1(); |
- JsepSessionDescription* pranswer2 = |
- CreateRemoteAnswer(session_->local_description()); |
- pranswer2->set_type(SessionDescriptionInterface::kPrAnswer); |
- |
- SetRemoteDescriptionExpectState(pranswer2, |
- WebRtcSession::STATE_RECEIVEDPRANSWER); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionExpectState(answer, WebRtcSession::STATE_INPROGRESS); |
-} |
- |
-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 remote description. |
- EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1)); |
- |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionWithoutError(offer); |
- |
- // Fail since we have not set a remote description. |
- EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1)); |
- |
- SessionDescriptionInterface* answer = CreateRemoteAnswer( |
- session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1)); |
- candidate.set_component(2); |
- JsepIceCandidate ice_candidate2(kMediaContentName0, 0, candidate); |
- EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2)); |
- |
- // 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 |
- // channels |
- // in the WebRtcSession, as channels 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_->GetVoiceChannel(0); |
- ASSERT_TRUE(voice_channel_ != NULL); |
- |
- ASSERT_EQ(1u, voice_channel_->recv_streams().size()); |
- ASSERT_EQ(1u, voice_channel_->send_streams().size()); |
- EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id); |
- EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id); |
- |
- // Change session back to video only. |
- SendVideoOnlyStream2(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- |
- video_channel_ = media_engine_->GetVideoChannel(0); |
- voice_channel_ = media_engine_->GetVoiceChannel(0); |
- |
- ASSERT_EQ(1u, video_channel_->recv_streams().size()); |
- EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id); |
- ASSERT_EQ(1u, video_channel_->send_streams().size()); |
- EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id); |
-} |
- |
-TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) { |
- Init(); |
- SendAudioVideoStream1(); |
- scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- VerifyCryptoParams(offer->description()); |
- SetRemoteDescriptionWithoutError(offer.release()); |
- scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL)); |
- VerifyCryptoParams(answer->description()); |
-} |
- |
-TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) { |
- options_.disable_encryption = true; |
- Init(); |
- SendAudioVideoStream1(); |
- scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- VerifyNoCryptoParams(offer->description(), false); |
-} |
- |
-TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) { |
- Init(); |
- VerifyAnswerFromNonCryptoOffer(); |
-} |
- |
-TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) { |
- Init(); |
- VerifyAnswerFromCryptoOffer(); |
-} |
- |
-// This test verifies that setLocalDescription fails if |
-// no a=ice-ufrag and a=ice-pwd lines are present in the SDP. |
-TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) { |
- Init(); |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- |
- std::string sdp; |
- RemoveIceUfragPwdLines(offer.get(), &sdp); |
- SessionDescriptionInterface* modified_offer = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- SetLocalDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer); |
-} |
- |
-// This test verifies that setRemoteDescription fails if |
-// no a=ice-ufrag and a=ice-pwd lines are present in the SDP. |
-TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) { |
- Init(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer()); |
- std::string sdp; |
- RemoveIceUfragPwdLines(offer.get(), &sdp); |
- SessionDescriptionInterface* modified_offer = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- SetRemoteDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer); |
-} |
- |
-// This test verifies that setLocalDescription fails if local offer has |
-// too short ice ufrag and pwd strings. |
-TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) { |
- Init(); |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- |
- std::string sdp; |
- // Modifying ice ufrag and pwd in local offer with strings smaller than the |
- // recommended values of 4 and 22 bytes respectively. |
- ModifyIceUfragPwdLines(offer.get(), "ice", "icepwd", &sdp); |
- SessionDescriptionInterface* modified_offer = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- std::string error; |
- EXPECT_FALSE(session_->SetLocalDescription(modified_offer, &error)); |
- |
- // Test with string greater than 256. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(offer.get(), kTooLongIceUfragPwd, kTooLongIceUfragPwd, |
- &sdp); |
- modified_offer = CreateSessionDescription(JsepSessionDescription::kOffer, sdp, |
- NULL); |
- EXPECT_FALSE(session_->SetLocalDescription(modified_offer, &error)); |
-} |
- |
-// This test verifies that setRemoteDescription fails if remote offer has |
-// too short ice ufrag and pwd strings. |
-TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionInvalidIceCredentials) { |
- Init(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer()); |
- std::string sdp; |
- // Modifying ice ufrag and pwd in remote offer with strings smaller than the |
- // recommended values of 4 and 22 bytes respectively. |
- ModifyIceUfragPwdLines(offer.get(), "ice", "icepwd", &sdp); |
- SessionDescriptionInterface* modified_offer = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- std::string error; |
- EXPECT_FALSE(session_->SetRemoteDescription(modified_offer, &error)); |
- |
- sdp.clear(); |
- ModifyIceUfragPwdLines(offer.get(), kTooLongIceUfragPwd, kTooLongIceUfragPwd, |
- &sdp); |
- modified_offer = CreateSessionDescription(JsepSessionDescription::kOffer, sdp, |
- NULL); |
- EXPECT_FALSE(session_->SetRemoteDescription(modified_offer, &error)); |
-} |
- |
-// Test that if the remote offer indicates the peer requested ICE restart (via |
-// a new ufrag or pwd), the old ICE candidates are not copied, and vice versa. |
-TEST_F(WebRtcSessionTest, TestSetRemoteOfferWithIceRestart) { |
- Init(); |
- scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer()); |
- |
- // Create the first offer. |
- std::string sdp; |
- ModifyIceUfragPwdLines(offer.get(), "0123456789012345", |
- "abcdefghijklmnopqrstuvwx", &sdp); |
- SessionDescriptionInterface* offer1 = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- cricket::Candidate candidate1(1, "udp", rtc::SocketAddress("1.1.1.1", 5000), |
- 0, "", "", "relay", 0, ""); |
- JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0, |
- candidate1); |
- EXPECT_TRUE(offer1->AddCandidate(&ice_candidate1)); |
- SetRemoteDescriptionWithoutError(offer1); |
- EXPECT_EQ(1, session_->remote_description()->candidates(0)->count()); |
- |
- // The second offer has the same ufrag and pwd but different address. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(offer.get(), "0123456789012345", |
- "abcdefghijklmnopqrstuvwx", &sdp); |
- SessionDescriptionInterface* offer2 = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 6000)); |
- JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0, |
- candidate1); |
- EXPECT_TRUE(offer2->AddCandidate(&ice_candidate2)); |
- SetRemoteDescriptionWithoutError(offer2); |
- EXPECT_EQ(2, session_->remote_description()->candidates(0)->count()); |
- |
- // The third offer has a different ufrag and different address. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(offer.get(), "0123456789012333", |
- "abcdefghijklmnopqrstuvwx", &sdp); |
- SessionDescriptionInterface* offer3 = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 7000)); |
- JsepIceCandidate ice_candidate3(kMediaContentName0, kMediaContentIndex0, |
- candidate1); |
- EXPECT_TRUE(offer3->AddCandidate(&ice_candidate3)); |
- SetRemoteDescriptionWithoutError(offer3); |
- EXPECT_EQ(1, session_->remote_description()->candidates(0)->count()); |
- |
- // The fourth offer has no candidate but a different ufrag/pwd. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(offer.get(), "0123456789012444", |
- "abcdefghijklmnopqrstuvyz", &sdp); |
- SessionDescriptionInterface* offer4 = |
- CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL); |
- SetRemoteDescriptionWithoutError(offer4); |
- EXPECT_EQ(0, session_->remote_description()->candidates(0)->count()); |
-} |
- |
-// Test that if the remote answer indicates the peer requested ICE restart (via |
-// a new ufrag or pwd), the old ICE candidates are not copied, and vice versa. |
-TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithIceRestart) { |
- Init(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionWithoutError(offer); |
- scoped_ptr<SessionDescriptionInterface> answer(CreateRemoteAnswer(offer)); |
- |
- // Create the first answer. |
- std::string sdp; |
- ModifyIceUfragPwdLines(answer.get(), "0123456789012345", |
- "abcdefghijklmnopqrstuvwx", &sdp); |
- SessionDescriptionInterface* answer1 = |
- CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL); |
- cricket::Candidate candidate1(1, "udp", rtc::SocketAddress("1.1.1.1", 5000), |
- 0, "", "", "relay", 0, ""); |
- JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0, |
- candidate1); |
- EXPECT_TRUE(answer1->AddCandidate(&ice_candidate1)); |
- SetRemoteDescriptionWithoutError(answer1); |
- EXPECT_EQ(1, session_->remote_description()->candidates(0)->count()); |
- |
- // The second answer has the same ufrag and pwd but different address. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(answer.get(), "0123456789012345", |
- "abcdefghijklmnopqrstuvwx", &sdp); |
- SessionDescriptionInterface* answer2 = |
- CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL); |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 6000)); |
- JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0, |
- candidate1); |
- EXPECT_TRUE(answer2->AddCandidate(&ice_candidate2)); |
- SetRemoteDescriptionWithoutError(answer2); |
- EXPECT_EQ(2, session_->remote_description()->candidates(0)->count()); |
- |
- // The third answer has a different ufrag and different address. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(answer.get(), "0123456789012333", |
- "abcdefghijklmnopqrstuvwx", &sdp); |
- SessionDescriptionInterface* answer3 = |
- CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL); |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 7000)); |
- JsepIceCandidate ice_candidate3(kMediaContentName0, kMediaContentIndex0, |
- candidate1); |
- EXPECT_TRUE(answer3->AddCandidate(&ice_candidate3)); |
- SetRemoteDescriptionWithoutError(answer3); |
- EXPECT_EQ(1, session_->remote_description()->candidates(0)->count()); |
- |
- // The fourth answer has no candidate but a different ufrag/pwd. |
- sdp.clear(); |
- ModifyIceUfragPwdLines(answer.get(), "0123456789012444", |
- "abcdefghijklmnopqrstuvyz", &sdp); |
- SessionDescriptionInterface* offer4 = |
- CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL); |
- SetRemoteDescriptionWithoutError(offer4); |
- EXPECT_EQ(0, session_->remote_description()->candidates(0)->count()); |
-} |
- |
-// Test that candidates sent to the "video" transport do not get pushed down to |
-// the "audio" transport channel when bundling. |
-TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) { |
- AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); |
- |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateRemoteOffer(); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- SessionDescriptionInterface* answer = CreateAnswer(NULL); |
- SetLocalDescriptionWithoutError(answer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- cricket::BaseChannel* voice_channel = session_->voice_channel(); |
- ASSERT(voice_channel != NULL); |
- |
- // Checks if one of the transport channels contains a connection using a given |
- // port. |
- auto connection_with_remote_port = [this, voice_channel](int port) { |
- SessionStats stats; |
- session_->GetChannelTransportStats(voice_channel, &stats); |
- for (auto& kv : stats.transport_stats) { |
- for (auto& chan_stat : kv.second.channel_stats) { |
- for (auto& conn_info : chan_stat.connection_infos) { |
- if (conn_info.remote_candidate.address().port() == port) { |
- return true; |
- } |
- } |
- } |
- } |
- return false; |
- }; |
- |
- EXPECT_FALSE(connection_with_remote_port(5000)); |
- EXPECT_FALSE(connection_with_remote_port(5001)); |
- EXPECT_FALSE(connection_with_remote_port(6000)); |
- |
- // The way the *_WAIT checks work is they only wait if the condition fails, |
- // which does not help in the case where state is not changing. This is |
- // problematic in this test since we want to verify that adding a video |
- // candidate does _not_ change state. So we interleave candidates and assume |
- // that messages are executed in the order they were posted. |
- |
- // First audio candidate. |
- cricket::Candidate candidate0; |
- candidate0.set_address(rtc::SocketAddress("1.1.1.1", 5000)); |
- candidate0.set_component(1); |
- candidate0.set_protocol("udp"); |
- JsepIceCandidate ice_candidate0(kMediaContentName0, kMediaContentIndex0, |
- candidate0); |
- EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate0)); |
- |
- // Video candidate. |
- cricket::Candidate candidate1; |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 6000)); |
- candidate1.set_component(1); |
- candidate1.set_protocol("udp"); |
- JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1, |
- candidate1); |
- EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1)); |
- |
- // Second audio candidate. |
- cricket::Candidate candidate2; |
- candidate2.set_address(rtc::SocketAddress("1.1.1.1", 5001)); |
- candidate2.set_component(1); |
- candidate2.set_protocol("udp"); |
- JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0, |
- candidate2); |
- EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2)); |
- |
- EXPECT_TRUE_WAIT(connection_with_remote_port(5000), 1000); |
- EXPECT_TRUE_WAIT(connection_with_remote_port(5001), 1000); |
- |
- // No need here for a _WAIT check since we are checking that state hasn't |
- // changed: if this is false we would be doing waits for nothing and if this |
- // is true then there will be no messages processed anyways. |
- EXPECT_FALSE(connection_with_remote_port(6000)); |
-} |
- |
-// kBundlePolicyBalanced BUNDLE policy and answer contains BUNDLE. |
-TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyBalanced BUNDLE policy but no BUNDLE in the answer. |
-TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- |
- // Remove BUNDLE from the answer. |
- rtc::scoped_ptr<SessionDescriptionInterface> answer( |
- CreateRemoteAnswer(session_->local_description())); |
- cricket::SessionDescription* answer_copy = answer->description()->Copy(); |
- answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE); |
- JsepSessionDescription* modified_answer = |
- new JsepSessionDescription(JsepSessionDescription::kAnswer); |
- modified_answer->Initialize(answer_copy, "1", "1"); |
- SetRemoteDescriptionWithoutError(modified_answer); // |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxBundle policy with BUNDLE in the answer. |
-TEST_F(WebRtcSessionTest, TestMaxBundleBundleInAnswer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxBundle policy with BUNDLE in the answer, but no |
-// audio content in the answer. |
-TEST_F(WebRtcSessionTest, TestMaxBundleRejectAudio) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- cricket::MediaSessionOptions recv_options; |
- recv_options.recv_audio = false; |
- recv_options.recv_video = true; |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description(), recv_options); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_TRUE(nullptr == session_->voice_channel()); |
- EXPECT_TRUE(nullptr != session_->video_rtp_transport_channel()); |
- |
- session_->Close(); |
- EXPECT_TRUE(nullptr == session_->voice_rtp_transport_channel()); |
- EXPECT_TRUE(nullptr == session_->voice_rtcp_transport_channel()); |
- EXPECT_TRUE(nullptr == session_->video_rtp_transport_channel()); |
- EXPECT_TRUE(nullptr == session_->video_rtcp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxBundle policy but no BUNDLE in the answer. |
-TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- |
- // Remove BUNDLE from the answer. |
- rtc::scoped_ptr<SessionDescriptionInterface> answer( |
- CreateRemoteAnswer(session_->local_description())); |
- cricket::SessionDescription* answer_copy = answer->description()->Copy(); |
- answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE); |
- JsepSessionDescription* modified_answer = |
- new JsepSessionDescription(JsepSessionDescription::kAnswer); |
- modified_answer->Initialize(answer_copy, "1", "1"); |
- SetRemoteDescriptionWithoutError(modified_answer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxBundle policy with BUNDLE in the remote offer. |
-TEST_F(WebRtcSessionTest, TestMaxBundleBundleInRemoteOffer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle); |
- SendAudioVideoStream1(); |
- |
- SessionDescriptionInterface* offer = CreateRemoteOffer(); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = CreateAnswer(nullptr); |
- SetLocalDescriptionWithoutError(answer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxBundle policy but no BUNDLE in the remote offer. |
-TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInRemoteOffer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle); |
- SendAudioVideoStream1(); |
- |
- // Remove BUNDLE from the offer. |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer()); |
- cricket::SessionDescription* offer_copy = offer->description()->Copy(); |
- offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE); |
- JsepSessionDescription* modified_offer = |
- new JsepSessionDescription(JsepSessionDescription::kOffer); |
- modified_offer->Initialize(offer_copy, "1", "1"); |
- |
- // Expect an error when applying the remote description |
- SetRemoteDescriptionExpectError(JsepSessionDescription::kOffer, |
- kCreateChannelFailed, modified_offer); |
-} |
- |
-// kBundlePolicyMaxCompat bundle policy and answer contains BUNDLE. |
-TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxCompat); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- // This should lead to an audio-only call but isn't implemented |
- // correctly yet. |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxCompat BUNDLE policy but no BUNDLE in the answer. |
-TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxCompat); |
- SendAudioVideoStream1(); |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- |
- // Remove BUNDLE from the answer. |
- rtc::scoped_ptr<SessionDescriptionInterface> answer( |
- CreateRemoteAnswer(session_->local_description())); |
- cricket::SessionDescription* answer_copy = answer->description()->Copy(); |
- answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE); |
- JsepSessionDescription* modified_answer = |
- new JsepSessionDescription(JsepSessionDescription::kAnswer); |
- modified_answer->Initialize(answer_copy, "1", "1"); |
- SetRemoteDescriptionWithoutError(modified_answer); // |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-// kBundlePolicyMaxbundle and then we call SetRemoteDescription first. |
-TEST_F(WebRtcSessionTest, TestMaxBundleWithSetRemoteDescriptionFirst) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- EXPECT_EQ(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestRequireRtcpMux) { |
- InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyRequire); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL); |
- EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL); |
- EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) { |
- InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyNegotiate); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- EXPECT_TRUE(session_->voice_rtcp_transport_channel() != NULL); |
- EXPECT_TRUE(session_->video_rtcp_transport_channel() != NULL); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL); |
- EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL); |
-} |
- |
-// This test verifies that SetLocalDescription and SetRemoteDescription fails |
-// if BUNDLE is enabled but rtcp-mux is disabled in m-lines. |
-TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) { |
- Init(); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- std::string offer_str; |
- offer->ToString(&offer_str); |
- // Disable rtcp-mux |
- const std::string rtcp_mux = "rtcp-mux"; |
- const std::string xrtcp_mux = "xrtcp-mux"; |
- rtc::replace_substrs(rtcp_mux.c_str(), rtcp_mux.length(), |
- xrtcp_mux.c_str(), xrtcp_mux.length(), |
- &offer_str); |
- JsepSessionDescription* local_offer = |
- new JsepSessionDescription(JsepSessionDescription::kOffer); |
- EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL)); |
- SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer); |
- JsepSessionDescription* remote_offer = |
- new JsepSessionDescription(JsepSessionDescription::kOffer); |
- EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL)); |
- SetRemoteDescriptionOfferExpectError(kBundleWithoutRtcpMux, remote_offer); |
- // Trying unmodified SDP. |
- SetLocalDescriptionWithoutError(offer); |
-} |
- |
-TEST_F(WebRtcSessionTest, SetAudioPlayout) { |
- Init(); |
- SendAudioVideoStream1(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); |
- ASSERT_TRUE(channel != NULL); |
- ASSERT_EQ(1u, channel->recv_streams().size()); |
- uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc(); |
- double volume; |
- EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume)); |
- EXPECT_EQ(1, volume); |
- session_->SetAudioPlayout(receive_ssrc, false); |
- EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume)); |
- EXPECT_EQ(0, volume); |
- session_->SetAudioPlayout(receive_ssrc, true); |
- EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume)); |
- EXPECT_EQ(1, volume); |
-} |
- |
-TEST_F(WebRtcSessionTest, SetAudioSend) { |
- Init(); |
- SendAudioVideoStream1(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); |
- ASSERT_TRUE(channel != NULL); |
- ASSERT_EQ(1u, channel->send_streams().size()); |
- uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); |
- EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); |
- |
- cricket::AudioOptions options; |
- options.echo_cancellation = rtc::Optional<bool>(true); |
- |
- rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer()); |
- session_->SetAudioSend(send_ssrc, false, options, renderer.get()); |
- EXPECT_TRUE(channel->IsStreamMuted(send_ssrc)); |
- EXPECT_EQ(rtc::Optional<bool>(), channel->options().echo_cancellation); |
- EXPECT_TRUE(renderer->sink() != NULL); |
- |
- // This will trigger SetSink(NULL) to the |renderer|. |
- session_->SetAudioSend(send_ssrc, true, options, NULL); |
- EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); |
- EXPECT_EQ(rtc::Optional<bool>(true), channel->options().echo_cancellation); |
- EXPECT_TRUE(renderer->sink() == NULL); |
-} |
- |
-TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) { |
- Init(); |
- SendAudioVideoStream1(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); |
- ASSERT_TRUE(channel != NULL); |
- ASSERT_EQ(1u, channel->send_streams().size()); |
- uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); |
- |
- rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer()); |
- cricket::AudioOptions options; |
- session_->SetAudioSend(send_ssrc, true, options, renderer.get()); |
- EXPECT_TRUE(renderer->sink() != NULL); |
- |
- // Delete the |renderer| and it will trigger OnClose() to the sink, and this |
- // will invalidate the |renderer_| pointer in the sink and prevent getting a |
- // SetSink(NULL) callback afterwards. |
- renderer.reset(); |
- |
- // This will trigger SetSink(NULL) if no OnClose() callback. |
- session_->SetAudioSend(send_ssrc, true, options, NULL); |
-} |
- |
-TEST_F(WebRtcSessionTest, SetVideoPlayout) { |
- Init(); |
- SendAudioVideoStream1(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0); |
- ASSERT_TRUE(channel != NULL); |
- ASSERT_LT(0u, channel->sinks().size()); |
- EXPECT_TRUE(channel->sinks().begin()->second == NULL); |
- ASSERT_EQ(1u, channel->recv_streams().size()); |
- uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc(); |
- cricket::FakeVideoRenderer renderer; |
- session_->SetVideoPlayout(receive_ssrc, true, &renderer); |
- EXPECT_TRUE(channel->sinks().begin()->second == &renderer); |
- session_->SetVideoPlayout(receive_ssrc, false, &renderer); |
- EXPECT_TRUE(channel->sinks().begin()->second == NULL); |
-} |
- |
-TEST_F(WebRtcSessionTest, SetVideoSend) { |
- Init(); |
- SendAudioVideoStream1(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0); |
- ASSERT_TRUE(channel != NULL); |
- ASSERT_EQ(1u, channel->send_streams().size()); |
- uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); |
- EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); |
- cricket::VideoOptions* options = NULL; |
- session_->SetVideoSend(send_ssrc, false, options); |
- EXPECT_TRUE(channel->IsStreamMuted(send_ssrc)); |
- session_->SetVideoSend(send_ssrc, true, options); |
- EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); |
-} |
- |
-TEST_F(WebRtcSessionTest, CanNotInsertDtmf) { |
- TestCanInsertDtmf(false); |
-} |
- |
-TEST_F(WebRtcSessionTest, CanInsertDtmf) { |
- TestCanInsertDtmf(true); |
-} |
- |
-TEST_F(WebRtcSessionTest, InsertDtmf) { |
- // Setup |
- Init(); |
- SendAudioVideoStream1(); |
- CreateAndSetRemoteOfferAndLocalAnswer(); |
- FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); |
- EXPECT_EQ(0U, channel->dtmf_info_queue().size()); |
- |
- // Insert DTMF |
- const int expected_duration = 90; |
- session_->InsertDtmf(kAudioTrack1, 0, expected_duration); |
- session_->InsertDtmf(kAudioTrack1, 1, expected_duration); |
- session_->InsertDtmf(kAudioTrack1, 2, expected_duration); |
- |
- // Verify |
- ASSERT_EQ(3U, channel->dtmf_info_queue().size()); |
- const uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); |
- EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0, |
- expected_duration)); |
- EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1, |
- expected_duration)); |
- EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2, |
- expected_duration)); |
-} |
- |
-// This test verifies the |initial_offerer| flag when session initiates the |
-// call. |
-TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) { |
- Init(); |
- EXPECT_FALSE(session_->initial_offerer()); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SessionDescriptionInterface* answer = CreateRemoteAnswer(offer); |
- SetLocalDescriptionWithoutError(offer); |
- EXPECT_TRUE(session_->initial_offerer()); |
- SetRemoteDescriptionWithoutError(answer); |
- EXPECT_TRUE(session_->initial_offerer()); |
-} |
- |
-// This test verifies the |initial_offerer| flag when session receives the call. |
-TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) { |
- Init(); |
- EXPECT_FALSE(session_->initial_offerer()); |
- SessionDescriptionInterface* offer = CreateRemoteOffer(); |
- SetRemoteDescriptionWithoutError(offer); |
- SessionDescriptionInterface* answer = CreateAnswer(NULL); |
- |
- EXPECT_FALSE(session_->initial_offerer()); |
- SetLocalDescriptionWithoutError(answer); |
- EXPECT_FALSE(session_->initial_offerer()); |
-} |
- |
-// Verifing local offer and remote answer have matching m-lines as per RFC 3264. |
-TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) { |
- Init(); |
- SendAudioVideoStream1(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionWithoutError(offer); |
- rtc::scoped_ptr<SessionDescriptionInterface> answer( |
- CreateRemoteAnswer(session_->local_description())); |
- |
- cricket::SessionDescription* answer_copy = answer->description()->Copy(); |
- answer_copy->RemoveContentByName("video"); |
- JsepSessionDescription* modified_answer = |
- new JsepSessionDescription(JsepSessionDescription::kAnswer); |
- |
- EXPECT_TRUE(modified_answer->Initialize(answer_copy, |
- answer->session_id(), |
- answer->session_version())); |
- SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer); |
- |
- // Different content names. |
- std::string sdp; |
- EXPECT_TRUE(answer->ToString(&sdp)); |
- const std::string kAudioMid = "a=mid:audio"; |
- const std::string kAudioMidReplaceStr = "a=mid:audio_content_name"; |
- rtc::replace_substrs(kAudioMid.c_str(), kAudioMid.length(), |
- kAudioMidReplaceStr.c_str(), |
- kAudioMidReplaceStr.length(), |
- &sdp); |
- SessionDescriptionInterface* modified_answer1 = |
- CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL); |
- SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer1); |
- |
- // Different media types. |
- EXPECT_TRUE(answer->ToString(&sdp)); |
- const std::string kAudioMline = "m=audio"; |
- const std::string kAudioMlineReplaceStr = "m=video"; |
- rtc::replace_substrs(kAudioMline.c_str(), kAudioMline.length(), |
- kAudioMlineReplaceStr.c_str(), |
- kAudioMlineReplaceStr.length(), |
- &sdp); |
- SessionDescriptionInterface* modified_answer2 = |
- CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL); |
- SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer2); |
- |
- SetRemoteDescriptionWithoutError(answer.release()); |
-} |
- |
-// Verifying remote offer and local answer have matching m-lines as per |
-// RFC 3264. |
-TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) { |
- Init(); |
- SendAudioVideoStream1(); |
- SessionDescriptionInterface* offer = CreateRemoteOffer(); |
- SetRemoteDescriptionWithoutError(offer); |
- SessionDescriptionInterface* answer = CreateAnswer(NULL); |
- |
- cricket::SessionDescription* answer_copy = answer->description()->Copy(); |
- answer_copy->RemoveContentByName("video"); |
- JsepSessionDescription* modified_answer = |
- new JsepSessionDescription(JsepSessionDescription::kAnswer); |
- |
- EXPECT_TRUE(modified_answer->Initialize(answer_copy, |
- answer->session_id(), |
- answer->session_version())); |
- SetLocalDescriptionAnswerExpectError(kMlineMismatch, modified_answer); |
- SetLocalDescriptionWithoutError(answer); |
-} |
- |
-// This test verifies that WebRtcSession does not start candidate allocation |
-// before SetLocalDescription is called. |
-TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) { |
- Init(); |
- SendAudioVideoStream1(); |
- SessionDescriptionInterface* offer = CreateRemoteOffer(); |
- cricket::Candidate candidate; |
- candidate.set_component(1); |
- JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0, |
- candidate); |
- EXPECT_TRUE(offer->AddCandidate(&ice_candidate)); |
- cricket::Candidate candidate1; |
- candidate1.set_component(1); |
- JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1, |
- candidate1); |
- EXPECT_TRUE(offer->AddCandidate(&ice_candidate1)); |
- SetRemoteDescriptionWithoutError(offer); |
- ASSERT_TRUE(session_->voice_rtp_transport_channel() != NULL); |
- ASSERT_TRUE(session_->video_rtp_transport_channel() != NULL); |
- |
- // Pump for 1 second and verify that no candidates are generated. |
- rtc::Thread::Current()->ProcessMessages(1000); |
- EXPECT_TRUE(observer_.mline_0_candidates_.empty()); |
- EXPECT_TRUE(observer_.mline_1_candidates_.empty()); |
- |
- SessionDescriptionInterface* answer = CreateAnswer(NULL); |
- SetLocalDescriptionWithoutError(answer); |
- EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout); |
-} |
- |
-// This test verifies that crypto parameter is updated in local session |
-// description as per security policy set in MediaSessionDescriptionFactory. |
-TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) { |
- Init(); |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- |
- // Making sure SetLocalDescription correctly sets crypto value in |
- // SessionDescription object after de-serialization of sdp string. The value |
- // will be set as per MediaSessionDescriptionFactory. |
- std::string offer_str; |
- offer->ToString(&offer_str); |
- SessionDescriptionInterface* jsep_offer_str = |
- CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL); |
- SetLocalDescriptionWithoutError(jsep_offer_str); |
- EXPECT_TRUE(session_->voice_channel()->secure_required()); |
- EXPECT_TRUE(session_->video_channel()->secure_required()); |
-} |
- |
-// This test verifies the crypto parameter when security is disabled. |
-TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) { |
- options_.disable_encryption = true; |
- Init(); |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- |
- // Making sure SetLocalDescription correctly sets crypto value in |
- // SessionDescription object after de-serialization of sdp string. The value |
- // will be set as per MediaSessionDescriptionFactory. |
- std::string offer_str; |
- offer->ToString(&offer_str); |
- SessionDescriptionInterface* jsep_offer_str = |
- CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL); |
- SetLocalDescriptionWithoutError(jsep_offer_str); |
- EXPECT_FALSE(session_->voice_channel()->secure_required()); |
- EXPECT_FALSE(session_->video_channel()->secure_required()); |
-} |
- |
-// This test verifies that an answer contains new ufrag and password if an offer |
-// with new ufrag and password is received. |
-TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) { |
- Init(); |
- cricket::MediaSessionOptions options; |
- options.recv_video = true; |
- rtc::scoped_ptr<JsepSessionDescription> offer( |
- CreateRemoteOffer(options)); |
- SetRemoteDescriptionWithoutError(offer.release()); |
- |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> answer( |
- CreateAnswer(NULL)); |
- SetLocalDescriptionWithoutError(answer.release()); |
- |
- // Receive an offer with new ufrag and password. |
- options.audio_transport_options.ice_restart = true; |
- options.video_transport_options.ice_restart = true; |
- options.data_transport_options.ice_restart = true; |
- rtc::scoped_ptr<JsepSessionDescription> updated_offer1( |
- CreateRemoteOffer(options, session_->remote_description())); |
- SetRemoteDescriptionWithoutError(updated_offer1.release()); |
- |
- rtc::scoped_ptr<SessionDescriptionInterface> updated_answer1( |
- CreateAnswer(NULL)); |
- |
- CompareIceUfragAndPassword(updated_answer1->description(), |
- session_->local_description()->description(), |
- false); |
- |
- SetLocalDescriptionWithoutError(updated_answer1.release()); |
-} |
- |
-// This test verifies that an answer contains old ufrag and password if an offer |
-// with old ufrag and password is received. |
-TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) { |
- Init(); |
- cricket::MediaSessionOptions options; |
- options.recv_video = true; |
- rtc::scoped_ptr<JsepSessionDescription> offer( |
- CreateRemoteOffer(options)); |
- SetRemoteDescriptionWithoutError(offer.release()); |
- |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> answer( |
- CreateAnswer(NULL)); |
- SetLocalDescriptionWithoutError(answer.release()); |
- |
- // Receive an offer without changed ufrag or password. |
- options.audio_transport_options.ice_restart = false; |
- options.video_transport_options.ice_restart = false; |
- options.data_transport_options.ice_restart = false; |
- rtc::scoped_ptr<JsepSessionDescription> updated_offer2( |
- CreateRemoteOffer(options, session_->remote_description())); |
- SetRemoteDescriptionWithoutError(updated_offer2.release()); |
- |
- rtc::scoped_ptr<SessionDescriptionInterface> updated_answer2( |
- CreateAnswer(NULL)); |
- |
- CompareIceUfragAndPassword(updated_answer2->description(), |
- session_->local_description()->description(), |
- true); |
- |
- SetLocalDescriptionWithoutError(updated_answer2.release()); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestSessionContentError) { |
- Init(); |
- SendAudioVideoStream1(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- const std::string session_id_orig = offer->session_id(); |
- const std::string session_version_orig = offer->session_version(); |
- SetLocalDescriptionWithoutError(offer); |
- |
- video_channel_ = media_engine_->GetVideoChannel(0); |
- video_channel_->set_fail_set_send_codecs(true); |
- |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionAnswerExpectError("ERROR_CONTENT", answer); |
- |
- // Test that after a content error, setting any description will |
- // result in an error. |
- video_channel_->set_fail_set_send_codecs(false); |
- answer = CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionExpectError("", "ERROR_CONTENT", answer); |
- offer = CreateRemoteOffer(); |
- SetLocalDescriptionExpectError("", "ERROR_CONTENT", offer); |
-} |
- |
-// Runs the loopback call test with BUNDLE and STUN disabled. |
-TEST_F(WebRtcSessionTest, TestIceStatesBasic) { |
- // Lets try with only UDP ports. |
- allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP | |
- cricket::PORTALLOCATOR_DISABLE_STUN | |
- cricket::PORTALLOCATOR_DISABLE_RELAY); |
- TestLoopbackCall(); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestIceStatesBasicIPv6) { |
- allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP | |
- cricket::PORTALLOCATOR_DISABLE_STUN | |
- cricket::PORTALLOCATOR_ENABLE_IPV6 | |
- cricket::PORTALLOCATOR_DISABLE_RELAY); |
- |
- // best connection is IPv6 since it has higher network preference. |
- LoopbackNetworkConfiguration config; |
- config.test_ipv6_network_ = true; |
- config.best_connection_after_initial_ice_converged_ = |
- LoopbackNetworkConfiguration::ExpectedBestConnection(0, 1); |
- |
- TestLoopbackCall(config); |
-} |
- |
-// Runs the loopback call test with BUNDLE and STUN enabled. |
-TEST_F(WebRtcSessionTest, TestIceStatesBundle) { |
- allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP | |
- cricket::PORTALLOCATOR_DISABLE_RELAY); |
- TestLoopbackCall(); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestRtpDataChannel) { |
- constraints_.reset(new FakeConstraints()); |
- constraints_->AddOptional( |
- webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true); |
- Init(); |
- |
- SetLocalDescriptionWithDataChannel(); |
- EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type()); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
- constraints_.reset(new FakeConstraints()); |
- constraints_->AddOptional( |
- webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true); |
- options_.disable_sctp_data_channels = false; |
- |
- InitWithDtls(GetParam()); |
- |
- SetLocalDescriptionWithDataChannel(); |
- EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type()); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
- InitWithDtls(GetParam()); |
- |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL); |
- EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- SetFactoryDtlsSrtp(); |
- InitWithDtls(GetParam()); |
- |
- // Create remote offer with SCTP. |
- cricket::MediaSessionOptions options; |
- options.data_channel_type = cricket::DCT_SCTP; |
- JsepSessionDescription* offer = |
- CreateRemoteOffer(options, cricket::SEC_DISABLED); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- // Verifies the answer contains SCTP. |
- rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL)); |
- EXPECT_TRUE(answer != NULL); |
- EXPECT_TRUE(answer->description()->GetContentByName("data") != NULL); |
- EXPECT_TRUE(answer->description()->GetTransportInfoByName("data") != NULL); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) { |
- constraints_.reset(new FakeConstraints()); |
- constraints_->AddOptional( |
- webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false); |
- InitWithDtls(GetParam()); |
- |
- SetLocalDescriptionWithDataChannel(); |
- EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type()); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestSctpDataChannelWithDtls) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
- InitWithDtls(GetParam()); |
- |
- SetLocalDescriptionWithDataChannel(); |
- EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type()); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestDisableSctpDataChannels) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- options_.disable_sctp_data_channels = true; |
- InitWithDtls(GetParam()); |
- |
- SetLocalDescriptionWithDataChannel(); |
- EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type()); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- const int new_send_port = 9998; |
- const int new_recv_port = 7775; |
- |
- InitWithDtls(GetParam()); |
- SetFactoryDtlsSrtp(); |
- |
- // By default, don't actually add the codecs to desc_factory_; they don't |
- // actually get serialized for SCTP in BuildMediaDescription(). Instead, |
- // let the session description get parsed. That'll get the proper codecs |
- // into the stream. |
- cricket::MediaSessionOptions options; |
- JsepSessionDescription* offer = CreateRemoteOfferWithSctpPort( |
- "stream1", new_send_port, options); |
- |
- // SetRemoteDescription will take the ownership of the offer. |
- SetRemoteDescriptionWithoutError(offer); |
- |
- SessionDescriptionInterface* answer = ChangeSDPSctpPort( |
- new_recv_port, CreateAnswer(NULL)); |
- ASSERT_TRUE(answer != NULL); |
- |
- // Now set the local description, which'll take ownership of the answer. |
- SetLocalDescriptionWithoutError(answer); |
- |
- // TEST PLAN: Set the port number to something new, set it in the SDP, |
- // and pass it all the way down. |
- EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type()); |
- CreateDataChannel(); |
- |
- cricket::FakeDataMediaChannel* ch = data_engine_->GetChannel(0); |
- int portnum = -1; |
- ASSERT_TRUE(ch != NULL); |
- ASSERT_EQ(1UL, ch->send_codecs().size()); |
- EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->send_codecs()[0].id); |
- EXPECT_EQ(0, strcmp(cricket::kGoogleSctpDataCodecName, |
- ch->send_codecs()[0].name.c_str())); |
- EXPECT_TRUE(ch->send_codecs()[0].GetParam(cricket::kCodecParamPort, |
- &portnum)); |
- EXPECT_EQ(new_send_port, portnum); |
- |
- ASSERT_EQ(1UL, ch->recv_codecs().size()); |
- EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->recv_codecs()[0].id); |
- EXPECT_EQ(0, strcmp(cricket::kGoogleSctpDataCodecName, |
- ch->recv_codecs()[0].name.c_str())); |
- EXPECT_TRUE(ch->recv_codecs()[0].GetParam(cricket::kCodecParamPort, |
- &portnum)); |
- EXPECT_EQ(new_recv_port, portnum); |
-} |
- |
-// Verifies that when a session's DataChannel receives an OPEN message, |
-// WebRtcSession signals the DataChannel creation request with the expected |
-// config. |
-TEST_P(WebRtcSessionTest, TestSctpDataChannelOpenMessage) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
- InitWithDtls(GetParam()); |
- |
- SetLocalDescriptionWithDataChannel(); |
- EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type()); |
- |
- webrtc::DataChannelInit config; |
- config.id = 1; |
- rtc::Buffer payload; |
- webrtc::WriteDataChannelOpenMessage("a", config, &payload); |
- cricket::ReceiveDataParams params; |
- params.ssrc = config.id; |
- params.type = cricket::DMT_CONTROL; |
- |
- cricket::DataChannel* data_channel = session_->data_channel(); |
- data_channel->SignalDataReceived(data_channel, params, payload); |
- |
- EXPECT_EQ("a", last_data_channel_label_); |
- EXPECT_EQ(config.id, last_data_channel_config_.id); |
- EXPECT_FALSE(last_data_channel_config_.negotiated); |
- EXPECT_EQ(webrtc::InternalDataChannelInit::kAcker, |
- last_data_channel_config_.open_handshake_role); |
-} |
- |
-TEST_P(WebRtcSessionTest, TestUsesProvidedCertificate) { |
- rtc::scoped_refptr<rtc::RTCCertificate> certificate = |
- FakeDtlsIdentityStore::GenerateCertificate(); |
- |
- PeerConnectionInterface::RTCConfiguration configuration; |
- configuration.certificates.push_back(certificate); |
- Init(nullptr, configuration); |
- EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000); |
- |
- EXPECT_EQ(session_->certificate_for_testing(), certificate); |
-} |
- |
-// Verifies that CreateOffer succeeds when CreateOffer is called before async |
-// identity generation is finished (even if a certificate is provided this is |
-// an async op). |
-TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- InitWithDtls(GetParam()); |
- |
- EXPECT_TRUE(session_->waiting_for_certificate_for_testing()); |
- SendAudioVideoStream1(); |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- |
- EXPECT_TRUE(offer != NULL); |
- VerifyNoCryptoParams(offer->description(), true); |
- VerifyFingerprintStatus(offer->description(), true); |
-} |
- |
-// Verifies that CreateAnswer succeeds when CreateOffer is called before async |
-// identity generation is finished (even if a certificate is provided this is |
-// an async op). |
-TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- InitWithDtls(GetParam()); |
- SetFactoryDtlsSrtp(); |
- |
- cricket::MediaSessionOptions options; |
- options.recv_video = true; |
- scoped_ptr<JsepSessionDescription> offer( |
- CreateRemoteOffer(options, cricket::SEC_DISABLED)); |
- ASSERT_TRUE(offer.get() != NULL); |
- SetRemoteDescriptionWithoutError(offer.release()); |
- |
- rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL)); |
- EXPECT_TRUE(answer != NULL); |
- VerifyNoCryptoParams(answer->description(), true); |
- VerifyFingerprintStatus(answer->description(), true); |
-} |
- |
-// Verifies that CreateOffer succeeds when CreateOffer is called after async |
-// identity generation is finished (even if a certificate is provided this is |
-// an async op). |
-TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- InitWithDtls(GetParam()); |
- |
- EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000); |
- |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- EXPECT_TRUE(offer != NULL); |
-} |
- |
-// Verifies that CreateOffer fails when CreateOffer is called after async |
-// identity generation fails. |
-TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- InitWithDtlsIdentityGenFail(); |
- |
- EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000); |
- |
- rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer()); |
- EXPECT_TRUE(offer == NULL); |
-} |
- |
-// Verifies that CreateOffer succeeds when Multiple CreateOffer calls are made |
-// before async identity generation is finished. |
-TEST_P(WebRtcSessionTest, |
- TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- VerifyMultipleAsyncCreateDescription(GetParam(), |
- CreateSessionDescriptionRequest::kOffer); |
-} |
- |
-// Verifies that CreateOffer fails when Multiple CreateOffer calls are made |
-// before async identity generation fails. |
-TEST_F(WebRtcSessionTest, |
- TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- VerifyMultipleAsyncCreateDescriptionIdentityGenFailure( |
- CreateSessionDescriptionRequest::kOffer); |
-} |
- |
-// Verifies that CreateAnswer succeeds when Multiple CreateAnswer calls are made |
-// before async identity generation is finished. |
-TEST_P(WebRtcSessionTest, |
- TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- VerifyMultipleAsyncCreateDescription( |
- GetParam(), CreateSessionDescriptionRequest::kAnswer); |
-} |
- |
-// Verifies that CreateAnswer fails when Multiple CreateAnswer calls are made |
-// before async identity generation fails. |
-TEST_F(WebRtcSessionTest, |
- TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- VerifyMultipleAsyncCreateDescriptionIdentityGenFailure( |
- CreateSessionDescriptionRequest::kAnswer); |
-} |
- |
-// Verifies that setRemoteDescription fails when DTLS is disabled and the remote |
-// offer has no SDES crypto but only DTLS fingerprint. |
-TEST_F(WebRtcSessionTest, TestSetRemoteOfferFailIfDtlsDisabledAndNoCrypto) { |
- // Init without DTLS. |
- Init(); |
- // Create a remote offer with secured transport disabled. |
- cricket::MediaSessionOptions options; |
- JsepSessionDescription* offer(CreateRemoteOffer( |
- options, cricket::SEC_DISABLED)); |
- // Adds a DTLS fingerprint to the remote offer. |
- cricket::SessionDescription* sdp = offer->description(); |
- TransportInfo* audio = sdp->GetTransportInfoByName("audio"); |
- ASSERT_TRUE(audio != NULL); |
- ASSERT_TRUE(audio->description.identity_fingerprint.get() == NULL); |
- audio->description.identity_fingerprint.reset( |
- rtc::SSLFingerprint::CreateFromRfc4572( |
- rtc::DIGEST_SHA_256, kFakeDtlsFingerprint)); |
- SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesCrypto, |
- offer); |
-} |
- |
-// This test verifies DSCP is properly applied on the media channels. |
-TEST_F(WebRtcSessionTest, TestDscpConstraint) { |
- constraints_.reset(new FakeConstraints()); |
- constraints_->AddOptional( |
- webrtc::MediaConstraintsInterface::kEnableDscp, true); |
- Init(); |
- SendAudioVideoStream1(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- |
- SetLocalDescriptionWithoutError(offer); |
- |
- video_channel_ = media_engine_->GetVideoChannel(0); |
- voice_channel_ = media_engine_->GetVoiceChannel(0); |
- |
- ASSERT_TRUE(video_channel_ != NULL); |
- ASSERT_TRUE(voice_channel_ != NULL); |
- const cricket::AudioOptions& audio_options = voice_channel_->options(); |
- const cricket::VideoOptions& video_options = video_channel_->options(); |
- EXPECT_EQ(rtc::Optional<bool>(true), audio_options.dscp); |
- EXPECT_EQ(rtc::Optional<bool>(true), video_options.dscp); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) { |
- constraints_.reset(new FakeConstraints()); |
- constraints_->AddOptional( |
- webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate, |
- true); |
- 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(rtc::Optional<bool>(true), video_options.suspend_below_min_bitrate); |
-} |
- |
-TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) { |
- constraints_.reset(new FakeConstraints()); |
- constraints_->AddOptional( |
- webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe, |
- true); |
- Init(); |
- SendAudioVideoStream1(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- |
- SetLocalDescriptionWithoutError(offer); |
- |
- voice_channel_ = media_engine_->GetVoiceChannel(0); |
- |
- ASSERT_TRUE(voice_channel_ != NULL); |
- const cricket::AudioOptions& audio_options = voice_channel_->options(); |
- EXPECT_EQ(rtc::Optional<bool>(true), audio_options.combined_audio_video_bwe); |
-} |
- |
-// Tests that we can renegotiate new media content with ICE candidates in the |
-// new remote SDP. |
-TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- InitWithDtls(GetParam()); |
- SetFactoryDtlsSrtp(); |
- |
- SendAudioOnlyStream2(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionWithoutError(offer); |
- |
- SessionDescriptionInterface* answer = CreateRemoteAnswer(offer); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- cricket::MediaSessionOptions options; |
- options.recv_video = true; |
- offer = CreateRemoteOffer(options, cricket::SEC_DISABLED); |
- |
- cricket::Candidate candidate1; |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000)); |
- candidate1.set_component(1); |
- JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1, |
- candidate1); |
- EXPECT_TRUE(offer->AddCandidate(&ice_candidate)); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- answer = CreateAnswer(NULL); |
- SetLocalDescriptionWithoutError(answer); |
-} |
- |
-// Tests that we can renegotiate new media content with ICE candidates separated |
-// from the remote SDP. |
-TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- InitWithDtls(GetParam()); |
- SetFactoryDtlsSrtp(); |
- |
- SendAudioOnlyStream2(); |
- SessionDescriptionInterface* offer = CreateOffer(); |
- SetLocalDescriptionWithoutError(offer); |
- |
- SessionDescriptionInterface* answer = CreateRemoteAnswer(offer); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- cricket::MediaSessionOptions options; |
- options.recv_video = true; |
- offer = CreateRemoteOffer(options, cricket::SEC_DISABLED); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- cricket::Candidate candidate1; |
- candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000)); |
- candidate1.set_component(1); |
- JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1, |
- candidate1); |
- EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate)); |
- |
- answer = CreateAnswer(NULL); |
- SetLocalDescriptionWithoutError(answer); |
-} |
- |
-// Flaky on Win and Mac only. See webrtc:4943 |
-#if defined(WEBRTC_WIN) || defined(WEBRTC_MAC) |
-#define MAYBE_TestRtxRemovedByCreateAnswer DISABLED_TestRtxRemovedByCreateAnswer |
-#else |
-#define MAYBE_TestRtxRemovedByCreateAnswer TestRtxRemovedByCreateAnswer |
-#endif |
-// Tests that RTX codec is removed from the answer when it isn't supported |
-// by local side. |
-TEST_F(WebRtcSessionTest, MAYBE_TestRtxRemovedByCreateAnswer) { |
- Init(); |
- SendAudioVideoStream1(); |
- std::string offer_sdp(kSdpWithRtx); |
- |
- SessionDescriptionInterface* offer = |
- CreateSessionDescription(JsepSessionDescription::kOffer, offer_sdp, NULL); |
- EXPECT_TRUE(offer->ToString(&offer_sdp)); |
- |
- // Offer SDP contains the RTX codec. |
- EXPECT_TRUE(offer_sdp.find("rtx") != std::string::npos); |
- SetRemoteDescriptionWithoutError(offer); |
- |
- SessionDescriptionInterface* answer = CreateAnswer(NULL); |
- std::string answer_sdp; |
- answer->ToString(&answer_sdp); |
- // Answer SDP removes the unsupported RTX codec. |
- EXPECT_TRUE(answer_sdp.find("rtx") == std::string::npos); |
- SetLocalDescriptionWithoutError(answer); |
-} |
- |
-// This verifies that the voice channel after bundle has both options from video |
-// and voice channels. |
-TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) { |
- InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced); |
- SendAudioVideoStream1(); |
- |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.use_rtp_mux = true; |
- |
- SessionDescriptionInterface* offer = CreateOffer(options); |
- SetLocalDescriptionWithoutError(offer); |
- |
- session_->video_channel()->SetOption(cricket::BaseChannel::ST_RTP, |
- rtc::Socket::Option::OPT_SNDBUF, 4000); |
- |
- session_->voice_channel()->SetOption(cricket::BaseChannel::ST_RTP, |
- rtc::Socket::Option::OPT_RCVBUF, 8000); |
- |
- int option_val; |
- EXPECT_TRUE(session_->video_rtp_transport_channel()->GetOption( |
- rtc::Socket::Option::OPT_SNDBUF, &option_val)); |
- EXPECT_EQ(4000, option_val); |
- EXPECT_FALSE(session_->voice_rtp_transport_channel()->GetOption( |
- rtc::Socket::Option::OPT_SNDBUF, &option_val)); |
- |
- EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption( |
- rtc::Socket::Option::OPT_RCVBUF, &option_val)); |
- EXPECT_EQ(8000, option_val); |
- EXPECT_FALSE(session_->video_rtp_transport_channel()->GetOption( |
- rtc::Socket::Option::OPT_RCVBUF, &option_val)); |
- |
- EXPECT_NE(session_->voice_rtp_transport_channel(), |
- session_->video_rtp_transport_channel()); |
- |
- SendAudioVideoStream2(); |
- SessionDescriptionInterface* answer = |
- CreateRemoteAnswer(session_->local_description()); |
- SetRemoteDescriptionWithoutError(answer); |
- |
- EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption( |
- rtc::Socket::Option::OPT_SNDBUF, &option_val)); |
- EXPECT_EQ(4000, option_val); |
- |
- EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption( |
- rtc::Socket::Option::OPT_RCVBUF, &option_val)); |
- EXPECT_EQ(8000, option_val); |
-} |
- |
-// Test creating a session, request multiple offers, destroy the session |
-// and make sure we got success/failure callbacks for all of the requests. |
-// Background: crbug.com/507307 |
-TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) { |
- Init(); |
- |
- rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observers[100]; |
- PeerConnectionInterface::RTCOfferAnswerOptions options; |
- options.offer_to_receive_audio = |
- RTCOfferAnswerOptions::kOfferToReceiveMediaTrue; |
- cricket::MediaSessionOptions session_options; |
- session_options.recv_audio = true; |
- |
- for (auto& o : observers) { |
- o = new WebRtcSessionCreateSDPObserverForTest(); |
- session_->CreateOffer(o, options, session_options); |
- } |
- |
- session_.reset(); |
- |
- for (auto& o : observers) { |
- // We expect to have received a notification now even if the session was |
- // terminated. The offer creation may or may not have succeeded, but we |
- // must have received a notification which, so the only invalid state |
- // is kInit. |
- EXPECT_NE(WebRtcSessionCreateSDPObserverForTest::kInit, o->state()); |
- } |
-} |
- |
-TEST_F(WebRtcSessionTest, TestPacketOptionsAndOnPacketSent) { |
- TestPacketOptions(); |
-} |
- |
-// Make sure the signal from "GetOnDestroyedSignal()" fires when the session |
-// is destroyed. |
-TEST_F(WebRtcSessionTest, TestOnDestroyedSignal) { |
- Init(); |
- session_.reset(); |
- EXPECT_TRUE(session_destroyed_); |
-} |
- |
-// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test |
-// currently fails because upon disconnection and reconnection OnIceComplete is |
-// called more than once without returning to IceGatheringGathering. |
- |
-INSTANTIATE_TEST_CASE_P(WebRtcSessionTests, |
- WebRtcSessionTest, |
- testing::Values(ALREADY_GENERATED, |
- DTLS_IDENTITY_STORE)); |