webrtc_m130/test/peer_scenario/scenario_connection.cc

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/*
* Copyright (c) 2019 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 "test/peer_scenario/scenario_connection.h"
#include "absl/memory/memory.h"
#include "media/base/rtp_utils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/jsep_transport_controller.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
namespace webrtc {
class ScenarioIceConnectionImpl : public ScenarioIceConnection,
public sigslot::has_slots<>,
private JsepTransportController::Observer,
private RtpPacketSinkInterface {
public:
ScenarioIceConnectionImpl(test::NetworkEmulationManagerImpl* net,
IceConnectionObserver* observer);
~ScenarioIceConnectionImpl() override;
void SendRtpPacket(rtc::ArrayView<const uint8_t> packet_view) override;
void SendRtcpPacket(rtc::ArrayView<const uint8_t> packet_view) override;
void SetRemoteSdp(SdpType type, const std::string& remote_sdp) override;
void SetLocalSdp(SdpType type, const std::string& local_sdp) override;
EmulatedEndpoint* endpoint() override { return endpoint_; }
const cricket::TransportDescription& transport_description() const override {
return transport_description_;
}
private:
JsepTransportController::Config CreateJsepConfig();
bool OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
DataChannelTransportInterface* data_channel_transport) override;
void OnRtpPacket(const RtpPacketReceived& packet) override;
void OnCandidates(const std::string& mid,
const std::vector<cricket::Candidate>& candidates);
IceConnectionObserver* const observer_;
EmulatedEndpoint* const endpoint_;
EmulatedNetworkManagerInterface* const manager_;
rtc::Thread* const signaling_thread_;
rtc::Thread* const network_thread_;
rtc::scoped_refptr<rtc::RTCCertificate> const certificate_
RTC_GUARDED_BY(network_thread_);
cricket::TransportDescription const transport_description_
RTC_GUARDED_BY(signaling_thread_);
std::unique_ptr<cricket::BasicPortAllocator> port_allocator_
RTC_GUARDED_BY(network_thread_);
std::unique_ptr<JsepTransportController> jsep_controller_;
RtpTransportInternal* rtp_transport_ RTC_GUARDED_BY(network_thread_) =
nullptr;
std::unique_ptr<SessionDescriptionInterface> remote_description_
RTC_GUARDED_BY(signaling_thread_);
std::unique_ptr<SessionDescriptionInterface> local_description_
RTC_GUARDED_BY(signaling_thread_);
};
std::unique_ptr<ScenarioIceConnection> ScenarioIceConnection::Create(
webrtc::test::NetworkEmulationManagerImpl* net,
IceConnectionObserver* observer) {
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
return std::make_unique<ScenarioIceConnectionImpl>(net, observer);
}
ScenarioIceConnectionImpl::ScenarioIceConnectionImpl(
test::NetworkEmulationManagerImpl* net,
IceConnectionObserver* observer)
: observer_(observer),
endpoint_(net->CreateEndpoint(EmulatedEndpointConfig())),
manager_(net->CreateEmulatedNetworkManagerInterface({endpoint_})),
signaling_thread_(rtc::Thread::Current()),
network_thread_(manager_->network_thread()),
certificate_(rtc::RTCCertificate::Create(
rtc::SSLIdentity::Create("", ::rtc::KT_DEFAULT))),
transport_description_(
/*transport_options*/ {},
rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
rtc::CreateRandomString(cricket::ICE_PWD_LENGTH),
cricket::IceMode::ICEMODE_FULL,
cricket::ConnectionRole::CONNECTIONROLE_PASSIVE,
rtc::SSLFingerprint::CreateFromCertificate(*certificate_.get())
.get()),
port_allocator_(
new cricket::BasicPortAllocator(manager_->network_manager())),
jsep_controller_(
new JsepTransportController(signaling_thread_,
network_thread_,
port_allocator_.get(),
/*async_resolver_factory*/ nullptr,
CreateJsepConfig())) {
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread_);
uint32_t flags = cricket::PORTALLOCATOR_DISABLE_TCP;
port_allocator_->set_flags(port_allocator_->flags() | flags);
port_allocator_->Initialize();
RTC_CHECK(port_allocator_->SetConfiguration(/*stun_servers*/ {},
/*turn_servers*/ {}, 0,
webrtc::NO_PRUNE));
jsep_controller_->SetLocalCertificate(certificate_);
});
}
ScenarioIceConnectionImpl::~ScenarioIceConnectionImpl() {
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread_);
jsep_controller_.reset();
port_allocator_.reset();
rtp_transport_ = nullptr;
});
}
JsepTransportController::Config ScenarioIceConnectionImpl::CreateJsepConfig() {
JsepTransportController::Config config;
config.transport_observer = this;
config.bundle_policy =
PeerConnectionInterface::BundlePolicy::kBundlePolicyMaxBundle;
config.rtcp_handler = [this](const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(network_thread_);
observer_->OnPacketReceived(packet);
};
return config;
}
void ScenarioIceConnectionImpl::SendRtpPacket(
rtc::ArrayView<const uint8_t> packet_view) {
rtc::CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
::cricket::kMaxRtpPacketLen);
network_thread_->PostTask(
RTC_FROM_HERE, [this, packet = std::move(packet)]() mutable {
RTC_DCHECK_RUN_ON(network_thread_);
if (rtp_transport_ != nullptr)
rtp_transport_->SendRtpPacket(&packet, rtc::PacketOptions(),
cricket::PF_SRTP_BYPASS);
});
}
void ScenarioIceConnectionImpl::SendRtcpPacket(
rtc::ArrayView<const uint8_t> packet_view) {
rtc::CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
::cricket::kMaxRtpPacketLen);
network_thread_->PostTask(
RTC_FROM_HERE, [this, packet = std::move(packet)]() mutable {
RTC_DCHECK_RUN_ON(network_thread_);
if (rtp_transport_ != nullptr)
rtp_transport_->SendRtcpPacket(&packet, rtc::PacketOptions(),
cricket::PF_SRTP_BYPASS);
});
}
void ScenarioIceConnectionImpl::SetRemoteSdp(SdpType type,
const std::string& remote_sdp) {
RTC_DCHECK_RUN_ON(signaling_thread_);
remote_description_ = webrtc::CreateSessionDescription(type, remote_sdp);
Revert "Reland "Replace sigslot usages with robocaller library."" This reverts commit c5f71087589b18bb4df1b78f2c452c4083edf2d9. Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995? Sample logs: STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575 STDERR: # last system error: 0 STDERR: # Check failed: (signaling_thread())->IsCurrent() STDERR: # Received signal 6 STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace() STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace() STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7f81c8d72db1 gsignal STDERR: #5 0x7f81c8d5c537 abort STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL() STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent() STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived() STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket() STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket() STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket() STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket() STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived() STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived() STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept() STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage() STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage() STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage() STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept() STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage() STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages() STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal() STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState() STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady() STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce() STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask() STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run() STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() STDERR: #40 0x7f81d395ae55 base::RunLoop::Run() STDERR: #41 0x7f81d39c87f2 base::Thread::Run() Original change's description: > Reland "Replace sigslot usages with robocaller library." > > This is a reland of 40261c3663fe316cfe40262c59cee993165ccf63 > > Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError > added a new member with a different name and used it in webrtc code. > After this change do two more follow up CLs to completely remove the old code > from google3. > > Original change's description: > > Replace sigslot usages with robocaller library. > > > > - Replace all the top level signals from jsep_transport_controller. > > - There are still sigslot usages in this file so keep the inheritance > > and that is the reason for not having a binary size gain in this CL. > > > > Bug: webrtc:11943 > > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540 > > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#32321} > > Bug: webrtc:11943 > Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946 > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32359} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11943 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32372}
2020-10-09 18:07:37 +00:00
jsep_controller_->SignalIceCandidatesGathered.connect(
this, &ScenarioIceConnectionImpl::OnCandidates);
auto res = jsep_controller_->SetRemoteDescription(
remote_description_->GetType(), remote_description_->description());
RTC_CHECK(res.ok()) << res.message();
RtpDemuxerCriteria criteria;
for (const auto& content : remote_description_->description()->contents()) {
if (content.media_description()->as_audio()) {
for (const auto& codec :
content.media_description()->as_audio()->codecs()) {
criteria.payload_types.insert(codec.id);
}
}
if (content.media_description()->as_video()) {
for (const auto& codec :
content.media_description()->as_video()->codecs()) {
criteria.payload_types.insert(codec.id);
}
}
}
network_thread_->PostTask(RTC_FROM_HERE, [this, criteria]() {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(rtp_transport_);
rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
});
}
void ScenarioIceConnectionImpl::SetLocalSdp(SdpType type,
const std::string& local_sdp) {
RTC_DCHECK_RUN_ON(signaling_thread_);
local_description_ = webrtc::CreateSessionDescription(type, local_sdp);
auto res = jsep_controller_->SetLocalDescription(
local_description_->GetType(), local_description_->description());
RTC_CHECK(res.ok()) << res.message();
jsep_controller_->MaybeStartGathering();
}
bool ScenarioIceConnectionImpl::OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread_);
if (rtp_transport == nullptr) {
rtp_transport_->UnregisterRtpDemuxerSink(this);
} else {
RTC_DCHECK(rtp_transport_ == nullptr || rtp_transport_ == rtp_transport);
if (rtp_transport_ != rtp_transport) {
rtp_transport_ = rtp_transport;
}
RtpDemuxerCriteria criteria;
criteria.mid = mid;
rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
}
return true;
}
void ScenarioIceConnectionImpl::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(network_thread_);
observer_->OnPacketReceived(packet.Buffer());
}
void ScenarioIceConnectionImpl::OnCandidates(
const std::string& mid,
const std::vector<cricket::Candidate>& candidates) {
RTC_DCHECK_RUN_ON(signaling_thread_);
observer_->OnIceCandidates(mid, candidates);
}
} // namespace webrtc