059e183419
Reason for revert: It turns out this revert was not necessary because the connection-state mapping for turn-turn connections was not done in connection. Original issue's description: > Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ ) > > Reason for revert: > ReadyToSendMedia did not consider the new presumed_writable state. > > Original issue's description: > > Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )" > > > > This reverts commit 72d41aa6da94dacb8a8464d1abd4ca7d1afffc65. > > > > New change made: > > Do not reset the BWE when the new network route is not ready to send media. > > > > BUG= > > R=pthatcher@webrtc.org, stefan@webrtc.org > > TBR=pthatcher@webrtc.org,stefan@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG= Review-Url: https://codereview.webrtc.org/2094863003 Cr-Commit-Position: refs/heads/master@{#13282}
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc