977b56c9e9
This uses libSRTPs srtp_remove_stream() https://github.com/cisco/libsrtp/blob/main/include/srtp.h#L597 method to remove SSRCs from the libSRTP session when they are removed from the RTP demuxer. This works even when the stream was added automatically via the ssrc_any_inbound mechanism. Only streams for inbound SSRCs that were added explicitly via SDP negotiation are removed. Guarded by WebRTC-SrtpRemoveReceiveStream field trial. BUG=webrtc:15604 Change-Id: I655bde5f8ddf26ac91395ef54bd1b3c598813380 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324720 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#41105}
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
- 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation