| /* |
| * Copyright 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/rtp_transport.h" |
| |
| #include <utility> |
| |
| #include "p2p/base/fake_packet_transport.h" |
| #include "pc/test/rtp_transport_test_util.h" |
| #include "rtc_base/buffer.h" |
| #include "rtc_base/containers/flat_set.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "test/explicit_key_value_config.h" |
| #include "test/gtest.h" |
| #include "test/run_loop.h" |
| |
| namespace webrtc { |
| |
| using test::ExplicitKeyValueConfig; |
| |
| constexpr bool kMuxDisabled = false; |
| constexpr bool kMuxEnabled = true; |
| constexpr uint16_t kLocalNetId = 1; |
| constexpr uint16_t kRemoteNetId = 2; |
| constexpr int kLastPacketId = 100; |
| constexpr int kTransportOverheadPerPacket = 28; // Ipv4(20) + UDP(8). |
| |
| class SignalObserver : public sigslot::has_slots<> { |
| public: |
| explicit SignalObserver(RtpTransport* transport) { |
| transport_ = transport; |
| transport->SubscribeReadyToSend( |
| this, [this](bool ready) { OnReadyToSend(ready); }); |
| transport->SubscribeNetworkRouteChanged( |
| this, [this](std::optional<rtc::NetworkRoute> route) { |
| OnNetworkRouteChanged(route); |
| }); |
| if (transport->rtp_packet_transport()) { |
| transport->rtp_packet_transport()->SignalSentPacket.connect( |
| this, &SignalObserver::OnSentPacket); |
| } |
| |
| if (transport->rtcp_packet_transport()) { |
| transport->rtcp_packet_transport()->SignalSentPacket.connect( |
| this, &SignalObserver::OnSentPacket); |
| } |
| } |
| |
| bool ready() const { return ready_; } |
| void OnReadyToSend(bool ready) { ready_ = ready; } |
| |
| std::optional<rtc::NetworkRoute> network_route() { return network_route_; } |
| void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route) { |
| network_route_ = network_route; |
| } |
| |
| void OnSentPacket(rtc::PacketTransportInternal* packet_transport, |
| const rtc::SentPacket& sent_packet) { |
| if (packet_transport == transport_->rtp_packet_transport()) { |
| rtp_transport_sent_count_++; |
| } else { |
| ASSERT_EQ(transport_->rtcp_packet_transport(), packet_transport); |
| rtcp_transport_sent_count_++; |
| } |
| } |
| |
| int rtp_transport_sent_count() { return rtp_transport_sent_count_; } |
| |
| int rtcp_transport_sent_count() { return rtcp_transport_sent_count_; } |
| |
| private: |
| int rtp_transport_sent_count_ = 0; |
| int rtcp_transport_sent_count_ = 0; |
| RtpTransport* transport_ = nullptr; |
| bool ready_ = false; |
| std::optional<rtc::NetworkRoute> network_route_; |
| }; |
| |
| TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtcp("fake_rtcp"); |
| fake_rtcp.SetWritable(true); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetWritable(true); |
| |
| transport.SetRtcpPacketTransport(&fake_rtcp); // rtcp ready |
| EXPECT_FALSE(observer.ready()); |
| transport.SetRtpPacketTransport(&fake_rtp); // rtp ready |
| EXPECT_TRUE(observer.ready()); |
| } |
| |
| TEST(RtpTransportTest, SettingRtpAndRtcpSignalsReady) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtcp("fake_rtcp"); |
| fake_rtcp.SetWritable(true); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetWritable(true); |
| |
| transport.SetRtpPacketTransport(&fake_rtp); // rtp ready |
| EXPECT_FALSE(observer.ready()); |
| transport.SetRtcpPacketTransport(&fake_rtcp); // rtcp ready |
| EXPECT_TRUE(observer.ready()); |
| } |
| |
| TEST(RtpTransportTest, SettingRtpWithRtcpMuxEnabledSignalsReady) { |
| RtpTransport transport(kMuxEnabled, ExplicitKeyValueConfig("")); |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetWritable(true); |
| |
| transport.SetRtpPacketTransport(&fake_rtp); // rtp ready |
| EXPECT_TRUE(observer.ready()); |
| } |
| |
| TEST(RtpTransportTest, DisablingRtcpMuxSignalsNotReady) { |
| RtpTransport transport(kMuxEnabled, ExplicitKeyValueConfig("")); |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetWritable(true); |
| |
| transport.SetRtpPacketTransport(&fake_rtp); // rtp ready |
| EXPECT_TRUE(observer.ready()); |
| |
| transport.SetRtcpMuxEnabled(false); |
| EXPECT_FALSE(observer.ready()); |
| } |
| |
| TEST(RtpTransportTest, EnablingRtcpMuxSignalsReady) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetWritable(true); |
| |
| transport.SetRtpPacketTransport(&fake_rtp); // rtp ready |
| EXPECT_FALSE(observer.ready()); |
| |
| transport.SetRtcpMuxEnabled(true); |
| EXPECT_TRUE(observer.ready()); |
| } |
| |
| // Tests the SignalNetworkRoute is fired when setting a packet transport. |
| TEST(RtpTransportTest, SetRtpTransportWithNetworkRouteChanged) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| |
| EXPECT_FALSE(observer.network_route()); |
| |
| rtc::NetworkRoute network_route; |
| // Set a non-null RTP transport with a new network route. |
| network_route.connected = true; |
| network_route.local = rtc::RouteEndpoint::CreateWithNetworkId(kLocalNetId); |
| network_route.remote = rtc::RouteEndpoint::CreateWithNetworkId(kRemoteNetId); |
| network_route.last_sent_packet_id = kLastPacketId; |
| network_route.packet_overhead = kTransportOverheadPerPacket; |
| fake_rtp.SetNetworkRoute(std::optional<rtc::NetworkRoute>(network_route)); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| ASSERT_TRUE(observer.network_route()); |
| EXPECT_TRUE(observer.network_route()->connected); |
| EXPECT_EQ(kLocalNetId, observer.network_route()->local.network_id()); |
| EXPECT_EQ(kRemoteNetId, observer.network_route()->remote.network_id()); |
| EXPECT_EQ(kTransportOverheadPerPacket, |
| observer.network_route()->packet_overhead); |
| EXPECT_EQ(kLastPacketId, observer.network_route()->last_sent_packet_id); |
| |
| // Set a null RTP transport. |
| transport.SetRtpPacketTransport(nullptr); |
| EXPECT_FALSE(observer.network_route()); |
| } |
| |
| TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| SignalObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtcp("fake_rtcp"); |
| |
| EXPECT_FALSE(observer.network_route()); |
| |
| rtc::NetworkRoute network_route; |
| // Set a non-null RTCP transport with a new network route. |
| network_route.connected = true; |
| network_route.local = rtc::RouteEndpoint::CreateWithNetworkId(kLocalNetId); |
| network_route.remote = rtc::RouteEndpoint::CreateWithNetworkId(kRemoteNetId); |
| network_route.last_sent_packet_id = kLastPacketId; |
| network_route.packet_overhead = kTransportOverheadPerPacket; |
| fake_rtcp.SetNetworkRoute(std::optional<rtc::NetworkRoute>(network_route)); |
| transport.SetRtcpPacketTransport(&fake_rtcp); |
| ASSERT_TRUE(observer.network_route()); |
| EXPECT_TRUE(observer.network_route()->connected); |
| EXPECT_EQ(kLocalNetId, observer.network_route()->local.network_id()); |
| EXPECT_EQ(kRemoteNetId, observer.network_route()->remote.network_id()); |
| EXPECT_EQ(kTransportOverheadPerPacket, |
| observer.network_route()->packet_overhead); |
| EXPECT_EQ(kLastPacketId, observer.network_route()->last_sent_packet_id); |
| |
| // Set a null RTCP transport. |
| transport.SetRtcpPacketTransport(nullptr); |
| EXPECT_FALSE(observer.network_route()); |
| } |
| |
| // Test that RTCP packets are sent over correct transport based on the RTCP-mux |
| // status. |
| TEST(RtpTransportTest, RtcpPacketSentOverCorrectTransport) { |
| // If the RTCP-mux is not enabled, RTCP packets are expected to be sent over |
| // the RtcpPacketTransport. |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| rtc::FakePacketTransport fake_rtcp("fake_rtcp"); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| transport.SetRtcpPacketTransport(&fake_rtcp); // rtcp ready |
| transport.SetRtpPacketTransport(&fake_rtp); // rtp ready |
| SignalObserver observer(&transport); |
| |
| fake_rtp.SetDestination(&fake_rtp, true); |
| fake_rtcp.SetDestination(&fake_rtcp, true); |
| |
| rtc::CopyOnWriteBuffer packet; |
| EXPECT_TRUE(transport.SendRtcpPacket(&packet, rtc::PacketOptions(), 0)); |
| EXPECT_EQ(1, observer.rtcp_transport_sent_count()); |
| |
| // The RTCP packets are expected to be sent over RtpPacketTransport if |
| // RTCP-mux is enabled. |
| transport.SetRtcpMuxEnabled(true); |
| EXPECT_TRUE(transport.SendRtcpPacket(&packet, rtc::PacketOptions(), 0)); |
| EXPECT_EQ(1, observer.rtp_transport_sent_count()); |
| } |
| |
| TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) { |
| RtpTransport transport(kMuxEnabled, ExplicitKeyValueConfig("")); |
| TransportObserver observer(&transport); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetWritable(true); |
| |
| // State changes, so we should signal. |
| transport.SetRtpPacketTransport(&fake_rtp); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 1); |
| |
| // State does not change, so we should not signal. |
| transport.SetRtpPacketTransport(&fake_rtp); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 1); |
| |
| // State does not change, so we should not signal. |
| transport.SetRtcpMuxEnabled(true); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 1); |
| |
| // State changes, so we should signal. |
| transport.SetRtcpMuxEnabled(false); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 2); |
| } |
| |
| // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is |
| // received. |
| TEST(RtpTransportTest, SignalDemuxedRtcp) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetDestination(&fake_rtp, true); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| TransportObserver observer(&transport); |
| |
| // An rtcp packet. |
| const unsigned char data[] = {0x80, 73, 0, 0}; |
| const int len = 4; |
| const rtc::PacketOptions options; |
| const int flags = 0; |
| fake_rtp.SendPacket(reinterpret_cast<const char*>(data), len, options, flags); |
| EXPECT_EQ(0, observer.rtp_count()); |
| EXPECT_EQ(1, observer.rtcp_count()); |
| } |
| |
| static const unsigned char kRtpData[] = {0x80, 0x11, 0, 0, 0, 0, |
| 0, 0, 0, 0, 0, 0}; |
| static const int kRtpLen = 12; |
| |
| // Test that SignalPacketReceived fires with rtcp=false when a RTP packet with a |
| // handled payload type is received. |
| TEST(RtpTransportTest, SignalHandledRtpPayloadType) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetDestination(&fake_rtp, true); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| TransportObserver observer(&transport); |
| RtpDemuxerCriteria demuxer_criteria; |
| // Add a handled payload type. |
| demuxer_criteria.payload_types().insert(0x11); |
| transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer); |
| |
| // An rtp packet. |
| const rtc::PacketOptions options; |
| const int flags = 0; |
| rtc::Buffer rtp_data(kRtpData, kRtpLen); |
| fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags); |
| EXPECT_EQ(1, observer.rtp_count()); |
| EXPECT_EQ(0, observer.un_demuxable_rtp_count()); |
| EXPECT_EQ(0, observer.rtcp_count()); |
| // Remove the sink before destroying the transport. |
| transport.UnregisterRtpDemuxerSink(&observer); |
| } |
| |
| TEST(RtpTransportTest, ReceivedPacketEcnMarkingPropagatedToDemuxedPacket) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| // Setup FakePacketTransport to send packets to itself. |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetDestination(&fake_rtp, true); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| TransportObserver observer(&transport); |
| RtpDemuxerCriteria demuxer_criteria; |
| // Add a payload type of kRtpData. |
| demuxer_criteria.payload_types().insert(0x11); |
| transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer); |
| |
| rtc::PacketOptions options; |
| options.ecn_1 = true; |
| const int flags = 0; |
| rtc::Buffer rtp_data(kRtpData, kRtpLen); |
| fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags); |
| ASSERT_EQ(observer.rtp_count(), 1); |
| EXPECT_EQ(observer.last_recv_rtp_packet().ecn(), rtc::EcnMarking::kEct1); |
| |
| transport.UnregisterRtpDemuxerSink(&observer); |
| } |
| |
| // Test that SignalPacketReceived does not fire when a RTP packet with an |
| // unhandled payload type is received. |
| TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) { |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetDestination(&fake_rtp, true); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| TransportObserver observer(&transport); |
| RtpDemuxerCriteria demuxer_criteria; |
| // Add an unhandled payload type. |
| demuxer_criteria.payload_types().insert(0x12); |
| transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer); |
| |
| const rtc::PacketOptions options; |
| const int flags = 0; |
| rtc::Buffer rtp_data(kRtpData, kRtpLen); |
| fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags); |
| EXPECT_EQ(0, observer.rtp_count()); |
| EXPECT_EQ(1, observer.un_demuxable_rtp_count()); |
| EXPECT_EQ(0, observer.rtcp_count()); |
| // Remove the sink before destroying the transport. |
| transport.UnregisterRtpDemuxerSink(&observer); |
| } |
| |
| TEST(RtpTransportTest, DontChangeReadyToSendStateOnSendFailure) { |
| // ReadyToSendState should only care about if transport is writable unless the |
| // field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/ is set. |
| RtpTransport transport(kMuxEnabled, ExplicitKeyValueConfig("")); |
| TransportObserver observer(&transport); |
| |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| fake_rtp.SetDestination(&fake_rtp, true); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| fake_rtp.SetWritable(true); |
| EXPECT_TRUE(observer.ready_to_send()); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 1); |
| rtc::CopyOnWriteBuffer packet; |
| EXPECT_TRUE(transport.SendRtpPacket(&packet, rtc::PacketOptions(), 0)); |
| |
| // The fake RTP will return -1 due to ENOTCONN. |
| fake_rtp.SetError(ENOTCONN); |
| EXPECT_FALSE(transport.SendRtpPacket(&packet, rtc::PacketOptions(), 0)); |
| // Ready to send state should not have changed. |
| EXPECT_TRUE(observer.ready_to_send()); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 1); |
| } |
| |
| TEST(RtpTransportTest, RecursiveSetSendDoesNotCrash) { |
| const int kShortTimeout = 100; |
| test::RunLoop loop; |
| |
| RtpTransport transport( |
| kMuxEnabled, |
| ExplicitKeyValueConfig("WebRTC-SetReadyToSendFalseIfSendFail/Enabled/")); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| TransportObserver observer(&transport); |
| observer.SetActionOnReadyToSend([&](bool ready) { |
| const rtc::PacketOptions options; |
| const int flags = 0; |
| rtc::CopyOnWriteBuffer rtp_data(kRtpData, kRtpLen); |
| transport.SendRtpPacket(&rtp_data, options, flags); |
| }); |
| // The fake RTP will have no destination, so will return -1. |
| fake_rtp.SetError(ENOTCONN); |
| fake_rtp.SetWritable(true); |
| // At this point, only the initial ready-to-send is observed. |
| EXPECT_TRUE(observer.ready_to_send()); |
| EXPECT_EQ(observer.ready_to_send_signal_count(), 1); |
| // After the wait, the ready-to-send false is observed. |
| EXPECT_EQ_WAIT(observer.ready_to_send_signal_count(), 2, kShortTimeout); |
| EXPECT_FALSE(observer.ready_to_send()); |
| } |
| |
| TEST(RtpTransportTest, RecursiveOnSentPacketDoesNotCrash) { |
| const int kShortTimeout = 100; |
| test::RunLoop loop; |
| RtpTransport transport(kMuxDisabled, ExplicitKeyValueConfig("")); |
| rtc::FakePacketTransport fake_rtp("fake_rtp"); |
| transport.SetRtpPacketTransport(&fake_rtp); |
| fake_rtp.SetDestination(&fake_rtp, true); |
| TransportObserver observer(&transport); |
| const rtc::PacketOptions options; |
| const int flags = 0; |
| |
| fake_rtp.SetWritable(true); |
| observer.SetActionOnSentPacket([&]() { |
| rtc::CopyOnWriteBuffer rtp_data(kRtpData, kRtpLen); |
| if (observer.sent_packet_count() < 2) { |
| transport.SendRtpPacket(&rtp_data, options, flags); |
| } |
| }); |
| rtc::CopyOnWriteBuffer rtp_data(kRtpData, kRtpLen); |
| transport.SendRtpPacket(&rtp_data, options, flags); |
| EXPECT_EQ(observer.sent_packet_count(), 1); |
| EXPECT_EQ_WAIT(observer.sent_packet_count(), 2, kShortTimeout); |
| } |
| |
| } // namespace webrtc |