webrtc/pc/sctp_utils.h
Bjorn A Mellem 487f9a17e4 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}
2019-09-09 21:58:36 +00:00

48 lines
1.5 KiB
C++

/*
* Copyright 2013 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.
*/
#ifndef PC_SCTP_UTILS_H_
#define PC_SCTP_UTILS_H_
#include <string>
#include "api/data_channel_interface.h"
#include "api/data_channel_transport_interface.h"
#include "media/base/media_channel.h"
namespace rtc {
class CopyOnWriteBuffer;
} // namespace rtc
namespace webrtc {
struct DataChannelInit;
// Read the message type and return true if it's an OPEN message.
bool IsOpenMessage(const rtc::CopyOnWriteBuffer& payload);
bool ParseDataChannelOpenMessage(const rtc::CopyOnWriteBuffer& payload,
std::string* label,
DataChannelInit* config);
bool ParseDataChannelOpenAckMessage(const rtc::CopyOnWriteBuffer& payload);
bool WriteDataChannelOpenMessage(const std::string& label,
const DataChannelInit& config,
rtc::CopyOnWriteBuffer* payload);
void WriteDataChannelOpenAckMessage(rtc::CopyOnWriteBuffer* payload);
cricket::DataMessageType ToCricketDataMessageType(DataMessageType type);
DataMessageType ToWebrtcDataMessageType(cricket::DataMessageType type);
} // namespace webrtc
#endif // PC_SCTP_UTILS_H_