commit | 487f9a17e426fd14bb06b13e861071b3f15d119b | [log] [tgz] |
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author | Bjorn A Mellem <mellem@webrtc.org> | Mon Sep 09 20:11:49 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Sep 09 21:58:36 2019 |
tree | 8e0b07f0d348ebd6aeef060ee15a32304fad3e84 | |
parent | 116ffe7e5be4aef2d5641e71f653118a95cabf0c [diff] |
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}
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.