webrtc_m130/pc/data_channel_controller.cc
Victor Boivie fea41f540c pc: Include SCTP queued bytes in buffered_amount
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.

Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.

This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.

This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.

Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
2024-03-13 15:44:17 +00:00

458 lines
16 KiB
C++

/*
* Copyright 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 "pc/data_channel_controller.h"
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "pc/peer_connection_internal.h"
#include "pc/sctp_utils.h"
#include "rtc_base/logging.h"
namespace webrtc {
DataChannelController::~DataChannelController() {
RTC_DCHECK(sctp_data_channels_n_.empty())
<< "Missing call to TeardownDataChannelTransport_n?";
RTC_DCHECK(!signaling_safety_.flag()->alive())
<< "Missing call to PrepareForShutdown?";
}
bool DataChannelController::HasDataChannels() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return channel_usage_ == DataChannelUsage::kInUse;
}
bool DataChannelController::HasUsedDataChannels() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return channel_usage_ != DataChannelUsage::kNeverUsed;
}
RTCError DataChannelController::SendData(
StreamId sid,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK_RUN_ON(network_thread());
if (!data_channel_transport_) {
RTC_LOG(LS_ERROR) << "SendData called before transport is ready";
return RTCError(RTCErrorType::INVALID_STATE);
}
return data_channel_transport_->SendData(sid.stream_id_int(), params,
payload);
}
void DataChannelController::AddSctpDataStream(StreamId sid) {
RTC_DCHECK_RUN_ON(network_thread());
if (data_channel_transport_) {
data_channel_transport_->OpenChannel(sid.stream_id_int());
}
}
void DataChannelController::RemoveSctpDataStream(StreamId sid) {
RTC_DCHECK_RUN_ON(network_thread());
if (data_channel_transport_) {
data_channel_transport_->CloseChannel(sid.stream_id_int());
}
}
void DataChannelController::OnChannelStateChanged(
SctpDataChannel* channel,
DataChannelInterface::DataState state) {
RTC_DCHECK_RUN_ON(network_thread());
// Stash away the internal id here in case `OnSctpDataChannelClosed` ends up
// releasing the last reference to the channel.
const int channel_id = channel->internal_id();
if (state == DataChannelInterface::DataState::kClosed)
OnSctpDataChannelClosed(channel);
DataChannelUsage channel_usage = sctp_data_channels_n_.empty()
? DataChannelUsage::kHaveBeenUsed
: DataChannelUsage::kInUse;
signaling_thread()->PostTask(SafeTask(
signaling_safety_.flag(), [this, channel_id, state, channel_usage] {
RTC_DCHECK_RUN_ON(signaling_thread());
channel_usage_ = channel_usage;
pc_->OnSctpDataChannelStateChanged(channel_id, state);
}));
}
size_t DataChannelController::buffered_amount(StreamId sid) const {
RTC_DCHECK_RUN_ON(network_thread());
if (!data_channel_transport_) {
return 0;
}
return data_channel_transport_->buffered_amount(sid.stream_id_int());
}
void DataChannelController::OnDataReceived(
int channel_id,
DataMessageType type,
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread());
if (HandleOpenMessage_n(channel_id, type, buffer))
return;
auto it = absl::c_find_if(sctp_data_channels_n_, [&](const auto& c) {
return c->sid_n().has_value() && c->sid_n()->stream_id_int() == channel_id;
});
if (it != sctp_data_channels_n_.end())
(*it)->OnDataReceived(type, buffer);
}
void DataChannelController::OnChannelClosing(int channel_id) {
RTC_DCHECK_RUN_ON(network_thread());
auto it = absl::c_find_if(sctp_data_channels_n_, [&](const auto& c) {
return c->sid_n().has_value() && c->sid_n()->stream_id_int() == channel_id;
});
if (it != sctp_data_channels_n_.end())
(*it)->OnClosingProcedureStartedRemotely();
}
void DataChannelController::OnChannelClosed(int channel_id) {
RTC_DCHECK_RUN_ON(network_thread());
StreamId sid(channel_id);
sid_allocator_.ReleaseSid(sid);
auto it = absl::c_find_if(sctp_data_channels_n_,
[&](const auto& c) { return c->sid_n() == sid; });
if (it != sctp_data_channels_n_.end()) {
rtc::scoped_refptr<SctpDataChannel> channel = std::move(*it);
sctp_data_channels_n_.erase(it);
channel->OnClosingProcedureComplete();
}
}
void DataChannelController::OnReadyToSend() {
RTC_DCHECK_RUN_ON(network_thread());
auto copy = sctp_data_channels_n_;
for (const auto& channel : copy) {
if (channel->sid_n().has_value()) {
channel->OnTransportReady();
} else {
// This happens for role==SSL_SERVER channels when we get notified by
// the transport *before* the SDP code calls `AllocateSctpSids` to
// trigger assignment of sids. In this case OnTransportReady() will be
// called from within `AllocateSctpSids` below.
RTC_LOG(LS_INFO) << "OnReadyToSend: Still waiting for an id for channel.";
}
}
}
void DataChannelController::OnTransportClosed(RTCError error) {
RTC_DCHECK_RUN_ON(network_thread());
// This loop will close all data channels and trigger a callback to
// `OnSctpDataChannelClosed`. We'll empty `sctp_data_channels_n_`, first
// and `OnSctpDataChannelClosed` will become a noop but we'll release the
// StreamId here.
std::vector<rtc::scoped_refptr<SctpDataChannel>> temp_sctp_dcs;
temp_sctp_dcs.swap(sctp_data_channels_n_);
for (const auto& channel : temp_sctp_dcs) {
channel->OnTransportChannelClosed(error);
if (channel->sid_n().has_value()) {
sid_allocator_.ReleaseSid(*channel->sid_n());
}
}
}
void DataChannelController::SetupDataChannelTransport_n(
DataChannelTransportInterface* transport) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(transport);
set_data_channel_transport(transport);
}
void DataChannelController::PrepareForShutdown() {
RTC_DCHECK_RUN_ON(signaling_thread());
signaling_safety_.reset(PendingTaskSafetyFlag::CreateDetachedInactive());
if (channel_usage_ != DataChannelUsage::kNeverUsed)
channel_usage_ = DataChannelUsage::kHaveBeenUsed;
}
void DataChannelController::TeardownDataChannelTransport_n(RTCError error) {
RTC_DCHECK_RUN_ON(network_thread());
OnTransportClosed(error);
set_data_channel_transport(nullptr);
RTC_DCHECK(sctp_data_channels_n_.empty());
weak_factory_.InvalidateWeakPtrs();
}
void DataChannelController::OnTransportChanged(
DataChannelTransportInterface* new_data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
if (data_channel_transport_ &&
data_channel_transport_ != new_data_channel_transport) {
// Changed which data channel transport is used for `sctp_mid_` (eg. now
// it's bundled).
set_data_channel_transport(new_data_channel_transport);
}
}
std::vector<DataChannelStats> DataChannelController::GetDataChannelStats()
const {
RTC_DCHECK_RUN_ON(network_thread());
std::vector<DataChannelStats> stats;
stats.reserve(sctp_data_channels_n_.size());
for (const auto& channel : sctp_data_channels_n_)
stats.push_back(channel->GetStats());
return stats;
}
bool DataChannelController::HandleOpenMessage_n(
int channel_id,
DataMessageType type,
const rtc::CopyOnWriteBuffer& buffer) {
if (type != DataMessageType::kControl || !IsOpenMessage(buffer))
return false;
// Received OPEN message; parse and signal that a new data channel should
// be created.
std::string label;
InternalDataChannelInit config;
config.id = channel_id;
if (!ParseDataChannelOpenMessage(buffer, &label, &config)) {
RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
<< channel_id;
} else {
config.open_handshake_role = InternalDataChannelInit::kAcker;
auto channel_or_error = CreateDataChannel(label, config);
if (channel_or_error.ok()) {
signaling_thread()->PostTask(SafeTask(
signaling_safety_.flag(),
[this, channel = channel_or_error.MoveValue(),
ready_to_send = data_channel_transport_->IsReadyToSend()] {
RTC_DCHECK_RUN_ON(signaling_thread());
OnDataChannelOpenMessage(std::move(channel), ready_to_send);
}));
} else {
RTC_LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message."
<< ToString(channel_or_error.error().type());
}
}
return true;
}
void DataChannelController::OnDataChannelOpenMessage(
rtc::scoped_refptr<SctpDataChannel> channel,
bool ready_to_send) {
channel_usage_ = DataChannelUsage::kInUse;
auto proxy = SctpDataChannel::CreateProxy(channel, signaling_safety_.flag());
pc_->Observer()->OnDataChannel(proxy);
pc_->NoteDataAddedEvent();
if (ready_to_send) {
network_thread()->PostTask([channel = std::move(channel)] {
if (channel->state() != DataChannelInterface::DataState::kClosed)
channel->OnTransportReady();
});
}
}
// RTC_RUN_ON(network_thread())
RTCError DataChannelController::ReserveOrAllocateSid(
absl::optional<StreamId>& sid,
absl::optional<rtc::SSLRole> fallback_ssl_role) {
if (sid.has_value()) {
return sid_allocator_.ReserveSid(*sid)
? RTCError::OK()
: RTCError(RTCErrorType::INVALID_RANGE, "StreamId reserved.");
}
// Attempt to allocate an ID based on the negotiated role.
absl::optional<rtc::SSLRole> role = pc_->GetSctpSslRole_n();
if (!role)
role = fallback_ssl_role;
if (role) {
sid = sid_allocator_.AllocateSid(*role);
if (!sid.has_value())
return RTCError(RTCErrorType::RESOURCE_EXHAUSTED);
}
// When we get here, we may still not have an ID, but that's a supported case
// whereby an id will be assigned later.
RTC_DCHECK(sid.has_value() || !role);
return RTCError::OK();
}
// RTC_RUN_ON(network_thread())
RTCErrorOr<rtc::scoped_refptr<SctpDataChannel>>
DataChannelController::CreateDataChannel(const std::string& label,
InternalDataChannelInit& config) {
absl::optional<StreamId> sid = absl::nullopt;
if (config.id != -1) {
if (config.id < 0 || config.id > cricket::kMaxSctpSid) {
return RTCError(RTCErrorType::INVALID_RANGE, "StreamId out of range.");
}
sid = StreamId(config.id);
}
RTCError err = ReserveOrAllocateSid(sid, config.fallback_ssl_role);
if (!err.ok())
return err;
// In case `sid` has changed. Update `config` accordingly.
if (sid.has_value()) {
config.id = sid->stream_id_int();
}
rtc::scoped_refptr<SctpDataChannel> channel = SctpDataChannel::Create(
weak_factory_.GetWeakPtr(), label, data_channel_transport_ != nullptr,
config, signaling_thread(), network_thread());
RTC_DCHECK(channel);
sctp_data_channels_n_.push_back(channel);
// If we have an id already, notify the transport.
if (sid.has_value())
AddSctpDataStream(*sid);
return channel;
}
RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
DataChannelController::InternalCreateDataChannelWithProxy(
const std::string& label,
const InternalDataChannelInit& config) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!pc_->IsClosed());
if (!config.IsValid()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Invalid DataChannelInit");
}
bool ready_to_send = false;
InternalDataChannelInit new_config = config;
auto ret = network_thread()->BlockingCall(
[&]() -> RTCErrorOr<rtc::scoped_refptr<SctpDataChannel>> {
RTC_DCHECK_RUN_ON(network_thread());
auto channel = CreateDataChannel(label, new_config);
if (!channel.ok())
return channel;
ready_to_send =
data_channel_transport_ && data_channel_transport_->IsReadyToSend();
if (ready_to_send) {
// If the transport is ready to send because the initial channel
// ready signal may have been sent before the DataChannel creation.
// This has to be done async because the upper layer objects (e.g.
// Chrome glue and WebKit) are not wired up properly until after
// `InternalCreateDataChannelWithProxy` returns.
network_thread()->PostTask([channel = channel.value()] {
if (channel->state() != DataChannelInterface::DataState::kClosed)
channel->OnTransportReady();
});
}
return channel;
});
if (!ret.ok())
return ret.MoveError();
channel_usage_ = DataChannelUsage::kInUse;
return SctpDataChannel::CreateProxy(ret.MoveValue(),
signaling_safety_.flag());
}
void DataChannelController::AllocateSctpSids(rtc::SSLRole role) {
RTC_DCHECK_RUN_ON(network_thread());
const bool ready_to_send =
data_channel_transport_ && data_channel_transport_->IsReadyToSend();
std::vector<std::pair<SctpDataChannel*, StreamId>> channels_to_update;
std::vector<rtc::scoped_refptr<SctpDataChannel>> channels_to_close;
for (auto it = sctp_data_channels_n_.begin();
it != sctp_data_channels_n_.end();) {
if (!(*it)->sid_n().has_value()) {
absl::optional<StreamId> sid = sid_allocator_.AllocateSid(role);
if (sid.has_value()) {
(*it)->SetSctpSid_n(*sid);
AddSctpDataStream(*sid);
if (ready_to_send) {
RTC_LOG(LS_INFO) << "AllocateSctpSids: Id assigned, ready to send.";
(*it)->OnTransportReady();
}
channels_to_update.push_back(std::make_pair((*it).get(), *sid));
} else {
channels_to_close.push_back(std::move(*it));
it = sctp_data_channels_n_.erase(it);
continue;
}
}
++it;
}
// Since closing modifies the list of channels, we have to do the actual
// closing outside the loop.
for (const auto& channel : channels_to_close) {
channel->CloseAbruptlyWithDataChannelFailure("Failed to allocate SCTP SID");
}
}
void DataChannelController::OnSctpDataChannelClosed(SctpDataChannel* channel) {
RTC_DCHECK_RUN_ON(network_thread());
// After the closing procedure is done, it's safe to use this ID for
// another data channel.
if (channel->sid_n().has_value()) {
sid_allocator_.ReleaseSid(*channel->sid_n());
}
auto it = absl::c_find_if(sctp_data_channels_n_,
[&](const auto& c) { return c.get() == channel; });
if (it != sctp_data_channels_n_.end())
sctp_data_channels_n_.erase(it);
}
void DataChannelController::set_data_channel_transport(
DataChannelTransportInterface* transport) {
RTC_DCHECK_RUN_ON(network_thread());
if (data_channel_transport_)
data_channel_transport_->SetDataSink(nullptr);
data_channel_transport_ = transport;
if (data_channel_transport_) {
// There's a new data channel transport. This needs to be signaled to the
// `sctp_data_channels_n_` so that they can reopen and reconnect. This is
// necessary when bundling is applied.
NotifyDataChannelsOfTransportCreated();
data_channel_transport_->SetDataSink(this);
}
}
void DataChannelController::NotifyDataChannelsOfTransportCreated() {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(data_channel_transport_);
for (const auto& channel : sctp_data_channels_n_) {
if (channel->sid_n().has_value())
AddSctpDataStream(*channel->sid_n());
channel->OnTransportChannelCreated();
}
}
rtc::Thread* DataChannelController::network_thread() const {
return pc_->network_thread();
}
rtc::Thread* DataChannelController::signaling_thread() const {
return pc_->signaling_thread();
}
} // namespace webrtc