This setup builds a custom libwebrtc.aar for Android and patches audio E2EE so FrameEncryptor/FrameDecryptor receive non-empty additional_data with RTP timestamp bytes.

Why

Stock io.github.webrtc-sdk:android:125.6422.07 can call audio frame encryptor with empty additional_data (ad=0), so nonce derivation based on timestamp is unavailable.

Desktop uses frame timestamp for nonce. This patch aligns Android with that approach by passing an 8-byte big-endian timestamp payload in additional_data:

bytes 0..3 = 0
bytes 4..7 = RTP timestamp (big-endian)

Files

build_custom_webrtc.sh — reproducible build script
patches/0001-audio-e2ee-pass-rtp-timestamp-as-additional-data.patch — WebRTC patch

Build

Recommended on Linux (macOS can work but is less predictable for long WebRTC builds).

Bootstrap depot_tools first:

cd /path/to/rosetta-android/tools/webrtc-custom
./bootstrap_depot_tools.sh

Then run:

cd /path/to/rosetta-android/tools/webrtc-custom
./build_custom_webrtc.sh

Optional env vars:

WEBRTC_ROOT — checkout root (default: $HOME/webrtc_android)
WEBRTC_SRC — direct path to src/
WEBRTC_BRANCH — default branch-heads/6422
WEBRTC_TAG — use a specific tag/commit instead of branch
OUT_AAR — output AAR path (default: app/libs/libwebrtc-custom.aar)
SYNC_JOBS — gclient sync jobs (default: 1, safer for googlesource limits)
SYNC_RETRIES — sync retry attempts (default: 8)
SYNC_RETRY_BASE_SEC — base retry delay in seconds (default: 20)

Troubleshooting (HTTP 429 / RESOURCE_EXHAUSTED)

If build fails with:

The requested URL returned error: 429
RESOURCE_EXHAUSTED
Short term server-time rate limit exceeded

run with conservative sync settings:

SYNC_JOBS=1 SYNC_RETRIES=12 SYNC_RETRY_BASE_SEC=30 ./build_custom_webrtc.sh

The script now retries fetch, git fetch, and gclient sync with backoff.

Integration in app

app/build.gradle.kts already prefers local app/libs/libwebrtc-custom.aar if present. If file exists, Maven WebRTC dependency is not used.

Maintenance policy

Keep patch small and isolated to audio/channel_send.cc + audio/channel_receive.cc.
Pin WebRTC branch/tag for releases.
Rebuild AAR on version bumps and verify e2ee_diag.txt shows ad=8 (or non-zero).