ortc/rtptransport_unittest.cc - src.git - Git at Google

 /*
  *  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 <memory>

 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "media/base/fakemediaengine.h"
 #include "ortc/ortcfactory.h"
 #include "ortc/testrtpparameters.h"
 #include "p2p/base/fakepackettransport.h"
 #include "rtc_base/gunit.h"

 namespace webrtc {

 // This test uses fake packet transports and a fake media engine, in order to
 // test the RtpTransport at only an API level. Any end-to-end test should go in
 // ortcfactory_integrationtest.cc instead.
 class RtpTransportTest : public testing::Test {
  public:
   RtpTransportTest() {
     fake_media_engine_ = new cricket::FakeMediaEngine();
     // Note: This doesn't need to use fake network classes, since it uses
     // FakePacketTransports.
     auto result = OrtcFactory::Create(
         nullptr, nullptr, nullptr, nullptr, nullptr,
         std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_),
         CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
     ortc_factory_ = result.MoveValue();
   }

  protected:
   // Owned by |ortc_factory_|.
   cricket::FakeMediaEngine* fake_media_engine_;
   std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
 };

 // Test GetRtpPacketTransport and GetRtcpPacketTransport, with and without RTCP
 // muxing.
 TEST_F(RtpTransportTest, GetPacketTransports) {
   rtc::FakePacketTransport rtp("rtp");
   rtc::FakePacketTransport rtcp("rtcp");
   // With muxed RTCP.
   RtpTransportParameters parameters;
   parameters.rtcp.mux = true;
   auto result =
       ortc_factory_->CreateRtpTransport(parameters, &rtp, nullptr, nullptr);
   ASSERT_TRUE(result.ok());
   EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport());
   EXPECT_EQ(nullptr, result.value()->GetRtcpPacketTransport());
   result.MoveValue().reset();
   // With non-muxed RTCP.
   parameters.rtcp.mux = false;
   result = ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
   ASSERT_TRUE(result.ok());
   EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport());
   EXPECT_EQ(&rtcp, result.value()->GetRtcpPacketTransport());
 }

 // If an RtpTransport starts out un-muxed and then starts muxing, the RTCP
 // packet transport should be forgotten and GetRtcpPacketTransport should
 // return null.
 TEST_F(RtpTransportTest, EnablingRtcpMuxingUnsetsRtcpTransport) {
   rtc::FakePacketTransport rtp("rtp");
   rtc::FakePacketTransport rtcp("rtcp");

   // Create non-muxed.
   RtpTransportParameters parameters;
   parameters.rtcp.mux = false;
   auto result =
       ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
   ASSERT_TRUE(result.ok());
   auto rtp_transport = result.MoveValue();

   // Enable muxing.
   parameters.rtcp.mux = true;
   EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());
   EXPECT_EQ(nullptr, rtp_transport->GetRtcpPacketTransport());
 }

 TEST_F(RtpTransportTest, GetAndSetRtcpParameters) {
   rtc::FakePacketTransport rtp("rtp");
   rtc::FakePacketTransport rtcp("rtcp");
   // Start with non-muxed RTCP.
   RtpTransportParameters parameters;
   parameters.rtcp.mux = false;
   parameters.rtcp.cname = "teST";
   parameters.rtcp.reduced_size = false;
   auto result =
       ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
   ASSERT_TRUE(result.ok());
   auto transport = result.MoveValue();
   EXPECT_EQ(parameters, transport->GetParameters());

   // Changing the CNAME is currently unsupported.
   parameters.rtcp.cname = "different";
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
             transport->SetParameters(parameters).type());
   parameters.rtcp.cname = "teST";

   // Enable RTCP muxing and reduced-size RTCP.
   parameters.rtcp.mux = true;
   parameters.rtcp.reduced_size = true;
   EXPECT_TRUE(transport->SetParameters(parameters).ok());
   EXPECT_EQ(parameters, transport->GetParameters());

   // Empty CNAME should result in the existing CNAME being used.
   parameters.rtcp.cname.clear();
   EXPECT_TRUE(transport->SetParameters(parameters).ok());
   EXPECT_EQ("teST", transport->GetParameters().rtcp.cname);

   // Disabling RTCP muxing after enabling shouldn't be allowed, since enabling
   // muxing should have made the RTP transport forget about the RTCP packet
   // transport initially passed into it.
   parameters.rtcp.mux = false;
   EXPECT_EQ(RTCErrorType::INVALID_STATE,
             transport->SetParameters(parameters).type());
 }

 // When Send or Receive is called on a sender or receiver, the RTCP parameters
 // from the RtpTransport underneath the sender should be applied to the created
 // media stream. The only relevant parameters (currently) are |cname| and
 // |reduced_size|.
 TEST_F(RtpTransportTest, SendAndReceiveApplyRtcpParametersToMediaEngine) {
   // First, create video transport with reduced-size RTCP.
   rtc::FakePacketTransport fake_packet_transport1("1");
   RtpTransportParameters parameters;
   parameters.rtcp.mux = true;
   parameters.rtcp.reduced_size = true;
   parameters.rtcp.cname = "foo";
   auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
       parameters, &fake_packet_transport1, nullptr, nullptr);
   auto video_transport = rtp_transport_result.MoveValue();

   // Create video sender and call Send, expecting parameters to be applied.
   auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
                                                       video_transport.get());
   auto video_sender = sender_result.MoveValue();
   EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
   cricket::FakeVideoMediaChannel* fake_video_channel =
       fake_media_engine_->GetVideoChannel(0);
   ASSERT_NE(nullptr, fake_video_channel);
   EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size);
   ASSERT_EQ(1u, fake_video_channel->send_streams().size());
   const cricket::StreamParams& video_send_stream =
       fake_video_channel->send_streams()[0];
   EXPECT_EQ("foo", video_send_stream.cname);

   // Create video receiver and call Receive, expecting parameters to be applied
   // (minus |cname|, since that's the sent cname, not received).
   auto receiver_result = ortc_factory_->CreateRtpReceiver(
       cricket::MEDIA_TYPE_VIDEO, video_transport.get());
   auto video_receiver = receiver_result.MoveValue();
   EXPECT_TRUE(
       video_receiver->Receive(MakeMinimalVp8ParametersWithSsrc(0xdeadbeef))
           .ok());
   EXPECT_TRUE(fake_video_channel->recv_rtcp_parameters().reduced_size);

   // Create audio transport with non-reduced size RTCP.
   rtc::FakePacketTransport fake_packet_transport2("2");
   parameters.rtcp.reduced_size = false;
   parameters.rtcp.cname = "bar";
   rtp_transport_result = ortc_factory_->CreateRtpTransport(
       parameters, &fake_packet_transport2, nullptr, nullptr);
   auto audio_transport = rtp_transport_result.MoveValue();

   // Create audio sender and call Send, expecting parameters to be applied.
   sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
                                                  audio_transport.get());
   auto audio_sender = sender_result.MoveValue();
   EXPECT_TRUE(audio_sender->Send(MakeMinimalIsacParameters()).ok());

   cricket::FakeVoiceMediaChannel* fake_voice_channel =
       fake_media_engine_->GetVoiceChannel(0);
   ASSERT_NE(nullptr, fake_voice_channel);
   EXPECT_FALSE(fake_voice_channel->send_rtcp_parameters().reduced_size);
   ASSERT_EQ(1u, fake_voice_channel->send_streams().size());
   const cricket::StreamParams& audio_send_stream =
       fake_voice_channel->send_streams()[0];
   EXPECT_EQ("bar", audio_send_stream.cname);

   // Create audio receiver and call Receive, expecting parameters to be applied
   // (minus |cname|, since that's the sent cname, not received).
   receiver_result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO,
                                                      audio_transport.get());
   auto audio_receiver = receiver_result.MoveValue();
   EXPECT_TRUE(
       audio_receiver->Receive(MakeMinimalOpusParametersWithSsrc(0xbaadf00d))
           .ok());
   EXPECT_FALSE(fake_voice_channel->recv_rtcp_parameters().reduced_size);
 }

 // When SetParameters is called, the modified parameters should be applied
 // to the media engine.
 // TODO(deadbeef): Once the implementation supports changing the CNAME,
 // test that here.
 TEST_F(RtpTransportTest, SetRtcpParametersAppliesParametersToMediaEngine) {
   rtc::FakePacketTransport fake_packet_transport("fake");
   RtpTransportParameters parameters;
   parameters.rtcp.mux = true;
   parameters.rtcp.reduced_size = false;
   auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
       parameters, &fake_packet_transport, nullptr, nullptr);
   auto rtp_transport = rtp_transport_result.MoveValue();

   // Create video sender and call Send, applying an initial set of parameters.
   auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
                                                       rtp_transport.get());
   auto sender = sender_result.MoveValue();
   EXPECT_TRUE(sender->Send(MakeMinimalVp8Parameters()).ok());

   // Modify parameters and expect them to be changed at the media engine level.
   parameters.rtcp.reduced_size = true;
   EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());

   cricket::FakeVideoMediaChannel* fake_video_channel =
       fake_media_engine_->GetVideoChannel(0);
   ASSERT_NE(nullptr, fake_video_channel);
   EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size);
 }

 // SetParameters should set keepalive for all RTP transports.
 // It is impossible to modify keepalive parameters if any streams are created.
 // Note: This is an implementation detail for current way of configuring the
 // keep-alive. It may change in the future.
 TEST_F(RtpTransportTest, CantChangeKeepAliveAfterCreatedSendStreams) {
   rtc::FakePacketTransport fake_packet_transport("fake");
   RtpTransportParameters parameters;
   parameters.keepalive.timeout_interval_ms = 100;
   auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
       parameters, &fake_packet_transport, nullptr, nullptr);
   ASSERT_TRUE(rtp_transport_result.ok());
   std::unique_ptr<RtpTransportInterface> rtp_transport =
       rtp_transport_result.MoveValue();

   // Updating keepalive parameters is ok, since no rtp sender created.
   parameters.keepalive.timeout_interval_ms = 200;
   EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());

   // Create video sender. Note: |sender_result| scope must extend past the
   // SetParameters() call below.
   auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
                                                       rtp_transport.get());
   EXPECT_TRUE(sender_result.ok());

   // Modify parameters second time after video send stream created.
   parameters.keepalive.timeout_interval_ms = 10;
   EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
             rtp_transport->SetParameters(parameters).type());
 }

 // Note: This is an implementation detail for current way of configuring the
 // keep-alive. It may change in the future.
 TEST_F(RtpTransportTest, KeepAliveMustBeSameAcrossTransportController) {
   rtc::FakePacketTransport fake_packet_transport("fake");
   RtpTransportParameters parameters;
   parameters.keepalive.timeout_interval_ms = 100;

   // Manually create a controller, that can be shared by multiple transports.
   auto controller_result = ortc_factory_->CreateRtpTransportController();
   ASSERT_TRUE(controller_result.ok());
   std::unique_ptr<RtpTransportControllerInterface> controller =
       controller_result.MoveValue();

   // Create a first transport.
   auto first_transport_result = ortc_factory_->CreateRtpTransport(
       parameters, &fake_packet_transport, nullptr, controller.get());
   ASSERT_TRUE(first_transport_result.ok());

   // Update the parameters, and create another transport for the same
   // controller.
   parameters.keepalive.timeout_interval_ms = 10;
   auto seconds_transport_result = ortc_factory_->CreateRtpTransport(
       parameters, &fake_packet_transport, nullptr, controller.get());
   EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
             seconds_transport_result.error().type());
 }

 }  // namespace webrtc
	/*
	* 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 <memory>

	#include "api/audio_codecs/builtin_audio_decoder_factory.h"
	#include "api/audio_codecs/builtin_audio_encoder_factory.h"
	#include "media/base/fakemediaengine.h"
	#include "ortc/ortcfactory.h"
	#include "ortc/testrtpparameters.h"
	#include "p2p/base/fakepackettransport.h"
	#include "rtc_base/gunit.h"

	namespace webrtc {

	// This test uses fake packet transports and a fake media engine, in order to
	// test the RtpTransport at only an API level. Any end-to-end test should go in
	// ortcfactory_integrationtest.cc instead.
	class RtpTransportTest : public testing::Test {
	public:
	RtpTransportTest() {
	fake_media_engine_ = new cricket::FakeMediaEngine();
	// Note: This doesn't need to use fake network classes, since it uses
	// FakePacketTransports.
	auto result = OrtcFactory::Create(
	nullptr, nullptr, nullptr, nullptr, nullptr,
	std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_),
	CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
	ortc_factory_ = result.MoveValue();
	}

	protected:
	// Owned by \|ortc_factory_\|.
	cricket::FakeMediaEngine* fake_media_engine_;
	std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
	};

	// Test GetRtpPacketTransport and GetRtcpPacketTransport, with and without RTCP
	// muxing.
	TEST_F(RtpTransportTest, GetPacketTransports) {
	rtc::FakePacketTransport rtp("rtp");
	rtc::FakePacketTransport rtcp("rtcp");
	// With muxed RTCP.
	RtpTransportParameters parameters;
	parameters.rtcp.mux = true;
	auto result =
	ortc_factory_->CreateRtpTransport(parameters, &rtp, nullptr, nullptr);
	ASSERT_TRUE(result.ok());
	EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport());
	EXPECT_EQ(nullptr, result.value()->GetRtcpPacketTransport());
	result.MoveValue().reset();
	// With non-muxed RTCP.
	parameters.rtcp.mux = false;
	result = ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
	ASSERT_TRUE(result.ok());
	EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport());
	EXPECT_EQ(&rtcp, result.value()->GetRtcpPacketTransport());
	}

	// If an RtpTransport starts out un-muxed and then starts muxing, the RTCP
	// packet transport should be forgotten and GetRtcpPacketTransport should
	// return null.
	TEST_F(RtpTransportTest, EnablingRtcpMuxingUnsetsRtcpTransport) {
	rtc::FakePacketTransport rtp("rtp");
	rtc::FakePacketTransport rtcp("rtcp");

	// Create non-muxed.
	RtpTransportParameters parameters;
	parameters.rtcp.mux = false;
	auto result =
	ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
	ASSERT_TRUE(result.ok());
	auto rtp_transport = result.MoveValue();

	// Enable muxing.
	parameters.rtcp.mux = true;
	EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());
	EXPECT_EQ(nullptr, rtp_transport->GetRtcpPacketTransport());
	}

	TEST_F(RtpTransportTest, GetAndSetRtcpParameters) {
	rtc::FakePacketTransport rtp("rtp");
	rtc::FakePacketTransport rtcp("rtcp");
	// Start with non-muxed RTCP.
	RtpTransportParameters parameters;
	parameters.rtcp.mux = false;
	parameters.rtcp.cname = "teST";
	parameters.rtcp.reduced_size = false;
	auto result =
	ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
	ASSERT_TRUE(result.ok());
	auto transport = result.MoveValue();
	EXPECT_EQ(parameters, transport->GetParameters());

	// Changing the CNAME is currently unsupported.
	parameters.rtcp.cname = "different";
	EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
	transport->SetParameters(parameters).type());
	parameters.rtcp.cname = "teST";

	// Enable RTCP muxing and reduced-size RTCP.
	parameters.rtcp.mux = true;
	parameters.rtcp.reduced_size = true;
	EXPECT_TRUE(transport->SetParameters(parameters).ok());
	EXPECT_EQ(parameters, transport->GetParameters());

	// Empty CNAME should result in the existing CNAME being used.
	parameters.rtcp.cname.clear();
	EXPECT_TRUE(transport->SetParameters(parameters).ok());
	EXPECT_EQ("teST", transport->GetParameters().rtcp.cname);

	// Disabling RTCP muxing after enabling shouldn't be allowed, since enabling
	// muxing should have made the RTP transport forget about the RTCP packet
	// transport initially passed into it.
	parameters.rtcp.mux = false;
	EXPECT_EQ(RTCErrorType::INVALID_STATE,
	transport->SetParameters(parameters).type());
	}

	// When Send or Receive is called on a sender or receiver, the RTCP parameters
	// from the RtpTransport underneath the sender should be applied to the created
	// media stream. The only relevant parameters (currently) are \|cname\| and
	// \|reduced_size\|.
	TEST_F(RtpTransportTest, SendAndReceiveApplyRtcpParametersToMediaEngine) {
	// First, create video transport with reduced-size RTCP.
	rtc::FakePacketTransport fake_packet_transport1("1");
	RtpTransportParameters parameters;
	parameters.rtcp.mux = true;
	parameters.rtcp.reduced_size = true;
	parameters.rtcp.cname = "foo";
	auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
	parameters, &fake_packet_transport1, nullptr, nullptr);
	auto video_transport = rtp_transport_result.MoveValue();

	// Create video sender and call Send, expecting parameters to be applied.
	auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
	video_transport.get());
	auto video_sender = sender_result.MoveValue();
	EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
	cricket::FakeVideoMediaChannel* fake_video_channel =
	fake_media_engine_->GetVideoChannel(0);
	ASSERT_NE(nullptr, fake_video_channel);
	EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size);
	ASSERT_EQ(1u, fake_video_channel->send_streams().size());
	const cricket::StreamParams& video_send_stream =
	fake_video_channel->send_streams()[0];
	EXPECT_EQ("foo", video_send_stream.cname);

	// Create video receiver and call Receive, expecting parameters to be applied
	// (minus \|cname\|, since that's the sent cname, not received).
	auto receiver_result = ortc_factory_->CreateRtpReceiver(
	cricket::MEDIA_TYPE_VIDEO, video_transport.get());
	auto video_receiver = receiver_result.MoveValue();
	EXPECT_TRUE(
	video_receiver->Receive(MakeMinimalVp8ParametersWithSsrc(0xdeadbeef))
	.ok());
	EXPECT_TRUE(fake_video_channel->recv_rtcp_parameters().reduced_size);

	// Create audio transport with non-reduced size RTCP.
	rtc::FakePacketTransport fake_packet_transport2("2");
	parameters.rtcp.reduced_size = false;
	parameters.rtcp.cname = "bar";
	rtp_transport_result = ortc_factory_->CreateRtpTransport(
	parameters, &fake_packet_transport2, nullptr, nullptr);
	auto audio_transport = rtp_transport_result.MoveValue();

	// Create audio sender and call Send, expecting parameters to be applied.
	sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
	audio_transport.get());
	auto audio_sender = sender_result.MoveValue();
	EXPECT_TRUE(audio_sender->Send(MakeMinimalIsacParameters()).ok());

	cricket::FakeVoiceMediaChannel* fake_voice_channel =
	fake_media_engine_->GetVoiceChannel(0);
	ASSERT_NE(nullptr, fake_voice_channel);
	EXPECT_FALSE(fake_voice_channel->send_rtcp_parameters().reduced_size);
	ASSERT_EQ(1u, fake_voice_channel->send_streams().size());
	const cricket::StreamParams& audio_send_stream =
	fake_voice_channel->send_streams()[0];
	EXPECT_EQ("bar", audio_send_stream.cname);

	// Create audio receiver and call Receive, expecting parameters to be applied
	// (minus \|cname\|, since that's the sent cname, not received).
	receiver_result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO,
	audio_transport.get());
	auto audio_receiver = receiver_result.MoveValue();
	EXPECT_TRUE(
	audio_receiver->Receive(MakeMinimalOpusParametersWithSsrc(0xbaadf00d))
	.ok());
	EXPECT_FALSE(fake_voice_channel->recv_rtcp_parameters().reduced_size);
	}

	// When SetParameters is called, the modified parameters should be applied
	// to the media engine.
	// TODO(deadbeef): Once the implementation supports changing the CNAME,
	// test that here.
	TEST_F(RtpTransportTest, SetRtcpParametersAppliesParametersToMediaEngine) {
	rtc::FakePacketTransport fake_packet_transport("fake");
	RtpTransportParameters parameters;
	parameters.rtcp.mux = true;
	parameters.rtcp.reduced_size = false;
	auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
	parameters, &fake_packet_transport, nullptr, nullptr);
	auto rtp_transport = rtp_transport_result.MoveValue();

	// Create video sender and call Send, applying an initial set of parameters.
	auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
	rtp_transport.get());
	auto sender = sender_result.MoveValue();
	EXPECT_TRUE(sender->Send(MakeMinimalVp8Parameters()).ok());

	// Modify parameters and expect them to be changed at the media engine level.
	parameters.rtcp.reduced_size = true;
	EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());

	cricket::FakeVideoMediaChannel* fake_video_channel =
	fake_media_engine_->GetVideoChannel(0);
	ASSERT_NE(nullptr, fake_video_channel);
	EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size);
	}

	// SetParameters should set keepalive for all RTP transports.
	// It is impossible to modify keepalive parameters if any streams are created.
	// Note: This is an implementation detail for current way of configuring the
	// keep-alive. It may change in the future.
	TEST_F(RtpTransportTest, CantChangeKeepAliveAfterCreatedSendStreams) {
	rtc::FakePacketTransport fake_packet_transport("fake");
	RtpTransportParameters parameters;
	parameters.keepalive.timeout_interval_ms = 100;
	auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
	parameters, &fake_packet_transport, nullptr, nullptr);
	ASSERT_TRUE(rtp_transport_result.ok());
	std::unique_ptr<RtpTransportInterface> rtp_transport =
	rtp_transport_result.MoveValue();

	// Updating keepalive parameters is ok, since no rtp sender created.
	parameters.keepalive.timeout_interval_ms = 200;
	EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());

	// Create video sender. Note: \|sender_result\| scope must extend past the
	// SetParameters() call below.
	auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
	rtp_transport.get());
	EXPECT_TRUE(sender_result.ok());

	// Modify parameters second time after video send stream created.
	parameters.keepalive.timeout_interval_ms = 10;
	EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
	rtp_transport->SetParameters(parameters).type());
	}

	// Note: This is an implementation detail for current way of configuring the
	// keep-alive. It may change in the future.
	TEST_F(RtpTransportTest, KeepAliveMustBeSameAcrossTransportController) {
	rtc::FakePacketTransport fake_packet_transport("fake");
	RtpTransportParameters parameters;
	parameters.keepalive.timeout_interval_ms = 100;

	// Manually create a controller, that can be shared by multiple transports.
	auto controller_result = ortc_factory_->CreateRtpTransportController();
	ASSERT_TRUE(controller_result.ok());
	std::unique_ptr<RtpTransportControllerInterface> controller =
	controller_result.MoveValue();

	// Create a first transport.
	auto first_transport_result = ortc_factory_->CreateRtpTransport(
	parameters, &fake_packet_transport, nullptr, controller.get());
	ASSERT_TRUE(first_transport_result.ok());

	// Update the parameters, and create another transport for the same
	// controller.
	parameters.keepalive.timeout_interval_ms = 10;
	auto seconds_transport_result = ortc_factory_->CreateRtpTransport(
	parameters, &fake_packet_transport, nullptr, controller.get());
	EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
	seconds_transport_result.error().type());
	}

	} // namespace webrtc