commit | d2b885fd91909f1b17fb11292a8c989d5d883b22 | [log] [tgz] |
---|---|---|
author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Jul 23 00:41:55 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Jul 23 22:08:00 2021 |
tree | a73b904cc39e4688a3f236a169fc0766bc31f10b | |
parent | d4b087c6cf50f1843200b1514d20dfc23c7b81cc [diff] |
Fix bug where we assume new m= sections will always be bundled. A recent change [1] assumes that all new m= sections will share the first BUNDLE group (if one already exists), which avoids generating ICE candidates that are ultimately unnecessary. This is fine for JSEP endpoints, but it breaks the following scenarios for non-JSEP endpoints: * Remote offer adding a new m= section that's not part of any BUNDLE group. * Remote offer adding an m= section to the second BUNDLE group. The latter is specifically problematic for any application that wants to bundle all audio streams in one group and all video streams in another group when using Unified Plan SDP, to replicate the behavior of using Plan B without bundling. It may try to add a video stream only for WebRTC to bundle it with audio. This is fixed by doing some minor re-factoring, having BundleManager update the bundle groups at offer time. Also: * Added some additional validation for multiple bundle groups in a subsequent offer, since that now becomes relevant. * Improved rollback support, because now rolling back an offer may need to not only remove mid->transport mappings but alter them. [1]: https://webrtc-review.googlesource.com/c/src/+/221601 Bug: webrtc:12906, webrtc:12999 Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34544}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.