437077dd45
This reverts commit 487f9a17e426fd14bb06b13e861071b3f15d119b. Reason for revert: speculative revert 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} TBR=steveanton@webrtc.org,mellem@webrtc.org,benwright@webrtc.org Change-Id: Ibd1a7f30931c114212c90824fec414d276d3f915 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9719 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152421 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29141}
42 lines
1.3 KiB
C++
42 lines
1.3 KiB
C++
/*
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* Copyright 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_SCTP_UTILS_H_
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#define PC_SCTP_UTILS_H_
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#include <string>
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#include "api/data_channel_interface.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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} // namespace rtc
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namespace webrtc {
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struct DataChannelInit;
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// Read the message type and return true if it's an OPEN message.
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bool IsOpenMessage(const rtc::CopyOnWriteBuffer& payload);
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bool ParseDataChannelOpenMessage(const rtc::CopyOnWriteBuffer& payload,
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std::string* label,
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DataChannelInit* config);
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bool ParseDataChannelOpenAckMessage(const rtc::CopyOnWriteBuffer& payload);
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bool WriteDataChannelOpenMessage(const std::string& label,
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const DataChannelInit& config,
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rtc::CopyOnWriteBuffer* payload);
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void WriteDataChannelOpenAckMessage(rtc::CopyOnWriteBuffer* payload);
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} // namespace webrtc
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#endif // PC_SCTP_UTILS_H_
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