# Spec: Record video (kamera/screenshare) orang lain — WebRTC receive (Request 2) Date: 2026-08-30. Status: Phase A + B DONE (capture → playable MP4); Phase C (UI) open. ## Why this is hard (ground truth, verified from @discordjs/voice 0.19.2 source) `VoiceReceiver.onUdpMessage` (dist/index.mjs:2059) drops EVERY non-opus RTP packet at line 2068: `if ((msg[1] & 127) !== RTP_OPUS_PAYLOAD_TYPE) return;`. So video (kamera H264? actually Discord uses VP8/H264; screenshare combines with video SSRC) is decrypted-capable but never forwarded. `receiver.parsePacket` (2033) DOES decrypt any payload type generically (audio + video) using `connectionData.{encryptionMode, nonceBuffer, secretKey}` — the only audio gate is the opus check inside onUdpMessage. => FIX: wrap `receiver.onUdpMessage` (like screenShareAudio.ts already does for screen-share AUDIO SSRCs): for packets whose payload type is a VIDEO type (payload 96 VP8, 101/102 H264, 106/116/126/127 AV1, VP9 98...), call `receiver.parsePacket(...)` myself to decrypt, then depacketize + write frames. Delegate opus (120) to the original handler. Delegate audio to original. ## Audio already works (screenShareAudio.ts). We add VIDEO. ## Science-of-the-changes below. ## Phase A — capture + decrypt + depacketize to AnnexB h264 (THIS change) Files (new): `src/modules/voice-recording/videoReceiver.ts` - Hook into `recorder.startRecording` alongside `hookScreenShareAudio`. - Wrap `receiver.onUdpMessage`: - read ssrc = msg.readUInt32BE(8); userData = receiver.ssrcMap.get(ssrc) - if payload type is video AND we have a "watching" subscription for that user (videoSSRC present), decrypt via receiver.parsePacket(...), then: - H264 (101/102 + payload 120 not): strip RTP header, reassemble FU-A fragments into AnnexB NALs (start-code prefixed), buffer until we have a full access unit (keyframe SPS/PPS/IDR or slices), append to a per- user-per-burst `.h264` file. - else delegate to original onUdpMessage. - Watch `receiver.ssrcMap` "create"/"update" for `videoSSRC !== undefined` → signal a video burst started for that user (like screenShareAudio does). - Per-user video files written to `config.RECORDINGS_DIR//video-.h264`. - Guard: skip bot's own video (client.user.id). Dependencies: NO new npm deps for Phase A (only crypto already in @discordjs/voice via parsePacket + Buffer). ffmpeg-headless (already in Nix buildInputs) used in Phase B for decode+mux. ## Phase B — decode + mux to playable MP4/WebM (DONE, commit 999c054b) - `closeBurst` waits for the WriteStream `finish` (full flush/fd close), then `muxToMp4(rawPath)`: `ffmpeg -f h264 -i raw.h264 -c copy -movflags +faststart out.mp4`, deletes raw on success (>=1B mp4), keeps it on failure. - Output: `//video--.mp4`. - ffmpeg is on the gateway runtime PATH (pkgs.ffmpeg-headless, already in the Nix buildInputs for the music/GoLive players). - Test: `tests/videoReceiver.test.ts` muxToMp4 case (real ffmpeg, generates a tiny baseline h264, asserts mp4 non-empty + raw deleted; skipped if no ffmpeg). ## Phase C — frontend playback + session grouping (follow-up) - Backend oRPC list video files; FE video player, group by call session like audio. ## Verification - Phase A: join voice, have a member screen-share/camera, confirm `.h264` file grows with NAL frames + keyframes; journal shows "video burst" logs. - Run vitest unit: RTP header strip + FU-A reassembly gives correct bytes. ## Open questions / risks - Discord codec for camera = H264(101/102); screenshare uses H264 (101/103?) and can also be VP8/VP9. Handle H264 first (depacketize proven), VP8/VP9 in Phase B via ffmpeg RTP input. - Encryption: DAVE (dave_protocol_version) adds a session layer; parsePacket already applies daveSession.decrypt for audio — we must call the SAME parsePacket path so DAVE/encryption is handled identically. - ssrc↔user mapping during a broadcast: videoSSRC is in ssrcMap after the voice state; may need the STREAM_CREATE network events to key reliably.