commit | 437077dd455fa2ea717b592f07cb4d370271d529 | [log] [tgz] |
---|---|---|
author | Qingsi Wang <qingsi@webrtc.org> | Tue Sep 10 17:52:26 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Sep 10 17:52:36 2019 |
tree | 970ae470d7ea480bce14c0e4a4ef8f257442d4c7 | |
parent | f6aa572e3614221760f545eee3d0ea354accd458 [diff] |
Revert "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" 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}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.