Revert "srtp: add UseCryptex API to SrtpSession and SrtpTransport" This reverts commit b9386f3f09ba0f38320f6a9e65dfe2992f433cb5. Reason for revert: Downstream libSRTP is not updated yet Original change's description: > srtp: add UseCryptex API to SrtpSession and SrtpTransport > > Adds the UseCryptex(enable, require, sending_session) API on SrtpSession > and UseCryptex(enable, require) on SrtpTransport. Wires libsrtp's > srtp_set_stream_use_cryptex through DoSetKey, and adds a require-cryptex > post-decrypt check via ParseRtpExtensionProfile (new helper in > modules/rtp_rtcp/source/rtp_util) for the cases libsrtp does not yet > enforce; see https://github.com/cisco/libsrtp/pull/805. > > No callers yet; the cryptex SDP negotiation wires this up in a follow-up. > > Bug: webrtc:455813732 > Change-Id: I407400d42912fb303c93c715e7110dafcc3d0cab > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/470600 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> > Cr-Commit-Position: refs/heads/main@{#47707} Bug: webrtc:455813732 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I496ebf60d95e1bbd3d117f5b30b4e4fe69e9a2ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/472440 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#47708}
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.
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