0f4f055ca6e0cab863a82c9d7113cfd5d585631d

The main purpose right now of this CL is to avoid the situation where multiple retransmissions are queued for sending (normally after network glitch with increased pacer queue length), and some of those fail sending because the can't be retrieved from the packet history due to too short time since last sent. Bug: webrtc:8975, webrtc:10607 Change-Id: I9f6369d83f0b8208e5f57b2dc2fd3f2db7c6fea1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135164 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27884}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- 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
- Coding style guide
- Code of conduct
Description
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