Philipp Hancke 2cfedb277a Remove vestiges of GTURN
and update some usage to use the "correct" stun attribute names

BUG=webrtc:42229250

Change-Id: If0c34d1d9b399766d7073661ea2a5515100256a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42810}
2024-08-19 16:05:37 +00:00
2024-08-19 16:05:37 +00:00
2024-07-09 10:30:26 +00:00
2024-08-19 16:05:37 +00:00
2024-07-11 20:26:16 +00:00
2024-08-14 07:11:57 +00:00
2024-05-27 12:46:57 +00:00

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 here 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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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