webrtc_m130/pc/rtp_transport_unittest.cc
Piotr (Peter) Slatala 042bb00838 Fix RTP transport accepting invalid RTCP headers.
Currently, the RtpTransport checks that the packet is either RTP or
RTCP. However, the RTCP check does not verify that the packet is a valid RTP,
and therefore invalid RTCP packets were allowed in the RtpTransport::OnReadPacket.

This change makes sure that the test for RTCP header (IsRtcpPacket) checks that it has the valid RTP version (2).

So far if the packet had the second byte that looked like
RTCP, it would ignore the first byte.


Bug: None
Change-Id: I5d07d497b9ef609c74b6e507c5f3e19e4bf10194
Reviewed-on: https://webrtc-review.googlesource.com/c/120646
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26480}
2019-01-31 03:00:58 +00:00

357 lines
13 KiB
C++

/*
* 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 <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/rtp_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).
TEST(RtpTransportTest, SetRtcpParametersCantDisableRtcpMux) {
RtpTransport transport(kMuxDisabled);
RtpTransportParameters params;
transport.SetParameters(params);
params.rtcp.mux = false;
EXPECT_FALSE(transport.SetParameters(params).ok());
}
TEST(RtpTransportTest, SetRtcpParametersEmptyCnameUsesExisting) {
static const char kName[] = "name";
RtpTransport transport(kMuxDisabled);
RtpTransportParameters params_with_name;
params_with_name.rtcp.cname = kName;
transport.SetParameters(params_with_name);
EXPECT_EQ(transport.GetParameters().rtcp.cname, kName);
RtpTransportParameters params_without_name;
transport.SetParameters(params_without_name);
EXPECT_EQ(transport.GetParameters().rtcp.cname, kName);
}
TEST(RtpTransportTest, SetRtpTransportKeepAliveNotSupported) {
// Tests that we warn users that keep-alive isn't supported yet.
// TODO(sprang): Wire up keep-alive and remove this test.
RtpTransport transport(kMuxDisabled);
RtpTransportParameters params;
params.keepalive.timeout_interval_ms = 1;
auto result = transport.SetParameters(params);
EXPECT_FALSE(result.ok());
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
}
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_ = std::move(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_network_id = kLocalNetId;
network_route.remote_network_id = 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_network_id = kLocalNetId;
network_route.remote_network_id = 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