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}
diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc
index 81ef220..a902c76 100644
--- a/pc/peer_connection_data_channel_unittest.cc
+++ b/pc/peer_connection_data_channel_unittest.cc
@@ -239,6 +239,20 @@
EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
}
+TEST_P(PeerConnectionDataChannelTest, InternalSctpTransportDeletedOnTeardown) {
+ auto caller = CreatePeerConnectionWithDataChannel();
+
+ ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
+ EXPECT_TRUE(caller->sctp_transport_factory()->last_fake_sctp_transport());
+
+ rtc::scoped_refptr<SctpTransportInterface> sctp_transport =
+ caller->GetInternalPeerConnection()->GetSctpTransport();
+
+ caller.reset();
+ EXPECT_EQ(static_cast<SctpTransport*>(sctp_transport.get())->internal(),
+ nullptr);
+}
+
// Test that sctp_content_name/sctp_transport_name (used for stats) are correct
// before and after BUNDLE is negotiated.
TEST_P(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
