commit | c2dd59c25da7532df4b4e75d853510a4a11724bf | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Tue Feb 06 10:29:35 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Feb 06 11:30:08 2018 |
tree | 9ed5357162d2f1cf5fed95bfc0c17bf7e6351596 | |
parent | 71d766eb4bd10277faafa1a3a47ebdc1dc4bd716 [diff] |
Skip oversized rtp header extension when parsing Rtp Packet. Rtp Packets in webrtc expected to be less that 1500, i.e. way less that 2^16 bytes for extensions block. This CL explicitly discards longer extension. Bug: chromium:809046 Change-Id: Ibed33b51bafc3fd4804ec135f66110c6d2796734 Reviewed-on: https://webrtc-review.googlesource.com/48061 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21910}
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.