blob: b3bd1db2e546b5c6b825a39c88bc21b612dde9a8 [file] [log] [blame]
/*
* 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 <cstdint>
#include <set>
#include <string>
#include <utility>
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "p2p/base/fake_packet_transport.h"
#include "pc/test/rtp_transport_test_util.h"
#include "rtc_base/buffer.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "test/gtest.h"
namespace webrtc {
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->SignalReadyToSend.connect(this, &SignalObserver::OnReadyToSend);
transport->SignalNetworkRouteChanged.connect(
this, &SignalObserver::OnNetworkRouteChanged);
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; }
absl::optional<rtc::NetworkRoute> network_route() { return network_route_; }
void OnNetworkRouteChanged(absl::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;
absl::optional<rtc::NetworkRoute> network_route_;
};
TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
RtpTransport transport(kMuxDisabled);
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);
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);
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);
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);
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);
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(absl::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);
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(absl::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);
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);
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);
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);
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 = {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.rtcp_count());
// Remove the sink before destroying the transport.
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);
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 = {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(0, observer.rtcp_count());
// Remove the sink before destroying the transport.
transport.UnregisterRtpDemuxerSink(&observer);
}
} // namespace webrtc