Close data channels when ID assignment fails.

This prevents crashes due to unassigned IDs.

Bug: chromium:945256
Change-Id: I63f3a17cc7dff07dab58a6bc59fe3606b23e8e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129902
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27349}
This commit is contained in:
Harald Alvestrand
2019-03-28 11:29:38 +01:00
committed by Commit Bot
parent 4c7112a27a
commit 1f928d3316
3 changed files with 39 additions and 6 deletions

View File

@ -147,6 +147,13 @@ class DataChannel : public DataChannelInterface, public sigslot::has_slots<> {
virtual uint64_t bytes_received() const { return bytes_received_; }
virtual bool Send(const DataBuffer& buffer);
// Close immediately, ignoring any queued data or closing procedure.
// This is called for RTP data channels when SDP indicates a channel should
// be removed, or SCTP data channels when the underlying SctpTransport is
// being destroyed.
// It is also called by the PeerConnection if SCTP ID assignment fails.
void CloseAbruptly();
// Called when the channel's ready to use. That can happen when the
// underlying DataMediaChannel becomes ready, or when this channel is a new
// stream on an existing DataMediaChannel, and we've finished negotiation.
@ -242,11 +249,6 @@ class DataChannel : public DataChannelInterface, public sigslot::has_slots<> {
};
bool Init(const InternalDataChannelInit& config);
// Close immediately, ignoring any queued data or closing procedure.
// This is called for RTP data channels when SDP indicates a channel should
// be removed, or SCTP data channels when the underlying SctpTransport is
// being destroyed.
void CloseAbruptly();
void UpdateState();
void SetState(DataState state);
void DisconnectFromProvider();

View File

@ -5102,16 +5102,23 @@ bool PeerConnection::HasDataChannels() const {
}
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
std::vector<rtc::scoped_refptr<DataChannel>> channels_to_close;
for (const auto& channel : sctp_data_channels_) {
if (channel->id() < 0) {
int sid;
if (!sid_allocator_.AllocateSid(role, &sid)) {
RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid, closing channel.";
channels_to_close.push_back(channel);
continue;
}
channel->SetSctpSid(sid);
}
}
// 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->CloseAbruptly();
}
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {

View File

@ -19,6 +19,7 @@
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "media/sctp/sctp_transport_internal.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
@ -721,6 +722,29 @@ TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
// close message and be destroyed.
rtc::Thread::Current()->ProcessMessages(100);
}
// Test behavior of creating too many datachannels.
TEST_P(PeerConnectionEndToEndTest, TooManyDataChannelsOpenedBeforeConnecting) {
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
std::vector<rtc::scoped_refptr<DataChannelInterface>> channels;
for (int i = 0; i <= cricket::kMaxSctpStreams / 2; i++) {
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("data", init));
channels.push_back(std::move(caller_dc));
}
Negotiate();
WaitForConnection();
EXPECT_EQ_WAIT(callee_signaled_data_channels_.size(),
static_cast<size_t>(cricket::kMaxSctpStreams / 2), kMaxWait);
EXPECT_EQ(DataChannelInterface::kOpen,
channels[(cricket::kMaxSctpStreams / 2) - 1]->state());
EXPECT_EQ(DataChannelInterface::kClosed,
channels[cricket::kMaxSctpStreams / 2]->state());
}
#endif // HAVE_SCTP
INSTANTIATE_TEST_SUITE_P(PeerConnectionEndToEndTest,