diff --git a/.hermes/plans/2026-08-30_video-record-receive-spec.md b/.hermes/plans/2026-08-30_video-record-receive-spec.md
new file mode 100644
index 00000000..cde5f390
--- /dev/null
+++ b/.hermes/plans/2026-08-30_video-record-receive-spec.md
@@ -0,0 +1,66 @@
+# Spec: Record video (kamera/screenshare) orang lain — WebRTC receive (Request 2)
+
+Date: 2026-08-30. Status: IN PROGRESS (Phase A).
+
+## 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/<uid>/video-<ts>.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 (follow-up)
+- Pipe AnnexB h264 → `ffmpeg -f h264 -i pipe:0 -c copy out.mp4` (or transcode).
+- Reuse muxer patterns from voice (muxer.ts) + session recording.
+- Persist in voice_recordings / new video_recordings table; upload via teleuploader.
+
+## 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.
diff --git a/services/discord-gateway/src/modules/voice-recording/recorder.ts b/services/discord-gateway/src/modules/voice-recording/recorder.ts
index f82c82f6..95c64322 100644
--- a/services/discord-gateway/src/modules/voice-recording/recorder.ts
+++ b/services/discord-gateway/src/modules/voice-recording/recorder.ts
@@ -19,6 +19,7 @@ import {
 } from "./recorder/sessionRecording.js";
 import { createSpeakingHandler } from "./recorder/speakingHandler.js";
 import { hookScreenShareAudio } from "./screenShareAudio.js";
+import { hookVideoReceiver } from "./videoReceiver.js";
 
 const logger = createChildLogger("recorder");
 
@@ -192,6 +193,16 @@ export async function startRecording(
   // to discover and register those SSRCs.
   hookScreenShareAudio(receiver, speakingHandler);
 
+  // ── Video capture (camera + screen share) ───────────────────────────
+  // @discordjs/voice only decrypts/forwards audio (opus). Video RTP (H264,
+  // VP8/VP9/AV1) arrives on the same UDP socket but is dropped by the opus
+  // gate. hookVideoReceiver wraps onUdpMessage, decrypts video payload types
+  // with the connection secret key (reusing receiver.parsePacket), then
+  // depacketizes H264 to AnnexB and writes a raw .h264 stream per user.
+  // Order matters: video hook must be installed AFTER the screen-share audio
+  // hook so non-video packets delegate down the existing wrapper chain.
+  hookVideoReceiver(receiver, client, recordingsDir);
+
   // Handle unexpected disconnection
   connection.on(VoiceConnectionStatus.Disconnected, async () => {
     if (config.VERBOSE) {
diff --git a/services/discord-gateway/src/modules/voice-recording/videoReceiver.ts b/services/discord-gateway/src/modules/voice-recording/videoReceiver.ts
new file mode 100644
index 00000000..d9d2bcbf
--- /dev/null
+++ b/services/discord-gateway/src/modules/voice-recording/videoReceiver.ts
@@ -0,0 +1,369 @@
+import { createWriteStream, mkdirSync, type WriteStream } from "node:fs";
+import path from "node:path";
+import type { VoiceReceiver } from "@discordjs/voice";
+import type { Client } from "discord.js-selfbot-v13";
+import { createChildLogger } from "@/shared/logger/index";
+
+const logger = createChildLogger("video-receiver");
+
+// ─── Discord RTP video payload types (real Discord clients) ───────────────
+// H264 is what Discord uses for the user's CAMERA; screen-share (GoLive) also
+// uses H264. VP8/VP9/AV1 appear for some share types / forced codecs.
+export const VIDEO_PAYLOAD_TYPES = new Set<number>([
+  96, 98, 101, 102, 106, 116, 126, 127,
+]);
+
+// H264 NAL unit types (single byte, after the 1-byte NAL header)
+const NAL_SPS = 7;
+const NAL_PPS = 8;
+const NAL_IDR = 5;
+
+// RTP H264 packet types (byte 0 of the payload)
+const _H264_SINGLE_NAL = 1; // 0-23 → single NAL unit, type = nal_unit_type
+const H264_STAP_A = 24;
+const H264_FU_A = 28;
+
+/**
+ * Pure RTP-H264 depacketizer → AnnexB (prefix NALs with 00 00 00 01 start codes).
+ *
+ * Input: RTP payload bytes (after the 12-byte RTP header, after decryption).
+ * Output: a list of AnnexB NAL buffers (each with start code) ready to append
+ * to a `.h264` elementary stream file.
+ *
+ * Reassembles FU-A fragmented NALs and STAP-A aggregated NALs into standalone
+ * NALs with start codes. Fragments that don't form a complete NAL are buffered.
+ */
+export class H264Depacketizer {
+  /** True after we've seen an SPS/PPS/IDR so we only start writing on a keyframe. */
+  private sawConfig = false;
+  /** Buffer for an in-progress FU-A NAL. */
+  private fuBuffer: number[] = [];
+
+  /** Handle one RTP payload; returns AnnexB NAL buffers to write (may be []). */
+  push(payload: Buffer): Buffer[] {
+    const out: Buffer[] = [];
+    if (payload.length === 0) return out;
+
+    const packetType = payload[0] & 0x1f;
+
+    if (packetType >= 1 && packetType <= 23) {
+      // Single NAL unit.
+      const nalType = payload[0] & 0x1f;
+      this.maybeEnableConfig(nalType);
+      const annexB = this.toAnnexB(payload);
+      if (this.sawConfig && annexB) out.push(annexB);
+      return out;
+    }
+
+    if (packetType === H264_FU_A) {
+      // Fragmentation unit: payload[0] = FU indicator (F/NRI/type=28),
+      // payload[1] = FU header (S/E/R/type).
+      const fuHeader = payload[1];
+      const start = (fuHeader & 0x80) !== 0;
+      const end = (fuHeader & 0x40) !== 0;
+      const nalType = fuHeader & 0x1f;
+
+      if (start) {
+        // First fragment: reconstruct the full NAL header by preserving the
+        // F and NRI bits from the FU indicator (payload[0]) OR'ed with the
+        // NAL unit type from the FU header (payload[1] & 0x1F).
+        const nalHeader = (payload[0] & 0xe0) | nalType;
+        this.fuBuffer = [nalHeader, ...payload.subarray(2)];
+      } else {
+        // Continuation. (Ignore if no buffer — a lost first fragment.)
+        if (this.fuBuffer.length === 0) {
+          // We missed the start; the NAL type unknown, skip this fragment.
+          return out;
+        }
+        if (payload.length > 2) {
+          this.fuBuffer.push(...payload.subarray(2));
+        }
+      }
+
+      if (end) {
+        const nal = Buffer.from(this.fuBuffer);
+        this.fuBuffer = [];
+        this.maybeEnableConfig(nalType);
+        if (this.sawConfig && nal.length > 0) {
+          out.push(this.toAnnexB(nal));
+        }
+      }
+      return out;
+    }
+
+    if (packetType === H264_STAP_A) {
+      // Aggregation packet: list of (2-byte NAL size, NAL data).
+      let offset = 1;
+      while (offset < payload.length) {
+        if (offset + 2 > payload.length) break;
+        const naluLength = payload.readUInt16BE(offset);
+        offset += 2;
+        if (offset + naluLength > payload.length) break;
+        const nalu = payload.subarray(offset, offset + naluLength);
+        offset += naluLength;
+        const nalType = nalu[0] & 0x1f;
+        this.maybeEnableConfig(nalType);
+        const annexB = this.toAnnexB(nalu);
+        if (this.sawConfig && annexB) out.push(annexB);
+      }
+    }
+    return out;
+  }
+
+  private maybeEnableConfig(nalType: number): void {
+    if (nalType === NAL_SPS || nalType === NAL_PPS || nalType === NAL_IDR) {
+      this.sawConfig = true;
+    }
+  }
+
+  private toAnnexB(nal: Buffer): Buffer {
+    const startCode = Buffer.from([0x00, 0x00, 0x00, 0x01]);
+    return Buffer.concat([startCode, nal]);
+  }
+
+  reset(): void {
+    this.sawConfig = false;
+    this.fuBuffer = [];
+  }
+}
+
+// ─── Per-user video burst state ────────────────────────────────────────────
+
+interface VideoBurst {
+  userId: string;
+  depacketizer: H264Depacketizer;
+  /** video SSRC this burst is sourced from (may be camera or screenshare). */
+  ssrc: number;
+  createdAt: number;
+  filePath: string;
+  out: WriteStream;
+  /** count of AnnexB NAL bytes written (for logging/health). */
+  bytesWritten: number;
+  lastPacketAt: number;
+}
+
+/**
+ * Listens for other users' VIDEO RTP (camera + screen share) on the same UDP
+ * socket @discordjs/voice receives audio from, decrypts each video packet with
+ * the connection's secret key (via receiver.parsePacket), depacketizes H264 to
+ * AnnexB, and writes a raw `.h264` elementary stream per user-per-burst.
+ *
+ * Wraps `receiver.onUdpMessage` like screenShareAudio.ts. Packets with a video
+ * payload type are consumed here; ALL other packets (opus mic + screen-share
+ * audio) are delegated to the current handler (the existing wrapper chain).
+ */
+export function hookVideoReceiver(
+  receiver: VoiceReceiver,
+  client: Client,
+  recordingsDir: string,
+): void {
+  const original = receiver.onUdpMessage;
+  const bursts = new Map<string, VideoBurst>();
+
+  // The UDP socket delivers video on SSRCs that are NOT the audioSSRC key of
+  // ssrcMap. Build a videoSSRC→userId index from VoiceUserData as it updates so
+  // we can attribute packets without SSRC-proximity guessing.
+  const videoSsrcToUser = new Map<number, string>();
+
+  receiver.ssrcMap.on("create", (data) => {
+    if (data.videoSSRC !== undefined) {
+      videoSsrcToUser.set(data.videoSSRC, data.userId);
+      logger.info(
+        { userId: data.userId, videoSSRC: data.videoSSRC },
+        "Video SSRC appeared (camera/screenshare start)",
+      );
+    }
+  });
+  receiver.ssrcMap.on("update", (_old, neu) => {
+    if (neu.videoSSRC !== undefined) {
+      videoSsrcToUser.set(neu.videoSSRC, neu.userId);
+    }
+  });
+
+  /** Close a burst's file (flush + end). */
+  const closeBurst = (userId: string): void => {
+    const burst = bursts.get(userId);
+    if (!burst) return;
+    bursts.delete(userId);
+    try {
+      burst.out.end();
+    } catch {
+      // ignore write errors on close
+    }
+    logger.info(
+      {
+        userId,
+        ssrc: burst.ssrc,
+        bytes: burst.bytesWritten,
+        durationMs: Date.now() - burst.createdAt,
+      },
+      "Video burst closed",
+    );
+  };
+
+  const openBurst = (userId: string, ssrc: number): VideoBurst | null => {
+    try {
+      const userDir = path.join(recordingsDir, userId);
+      mkdirSync(userDir, { recursive: true });
+      const filePath = path.join(userDir, `video-${ssrc}-${Date.now()}.h264`);
+      const out = createWriteStream(filePath);
+      const burst: VideoBurst = {
+        userId,
+        ssrc,
+        depacketizer: new H264Depacketizer(),
+        createdAt: Date.now(),
+        filePath,
+        out,
+        bytesWritten: 0,
+        lastPacketAt: Date.now(),
+      };
+      bursts.set(userId, burst);
+      logger.info({ userId, ssrc, filePath }, "Video burst opened");
+      return burst;
+    } catch (err) {
+      logger.warn(
+        { userId, ssrc, err: err instanceof Error ? err.message : String(err) },
+        "Failed to open video burst file",
+      );
+      return null;
+    }
+  };
+
+  receiver.onUdpMessage = (msg: Buffer) => {
+    if (msg.length <= 8) {
+      original.call(receiver, msg);
+      return;
+    }
+
+    const payloadType = msg[1] & 127;
+    if (!VIDEO_PAYLOAD_TYPES.has(payloadType)) {
+      // Not video — let the existing handler (audio + screen-share audio) take it.
+      original.call(receiver, msg);
+      return;
+    }
+
+    // ── It's a video RTP packet. Decrypt + depacketize + write. ──
+    const ssrc = msg.readUInt32BE(8);
+
+    // Attribute to a user.
+    let userId = videoSsrcToUser.get(ssrc);
+    if (!userId) {
+      // Fallback: maybe we registered the SSRC under an audio entry via the
+      // screen-share-audio hook, or proximity. Skip bot's own stream.
+      const candidate = inferVideoOwner(msg, receiver);
+      if (candidate && candidate.userId !== client.user?.id) {
+        userId = candidate.userId;
+        videoSsrcToUser.set(ssrc, candidate.userId);
+      }
+    }
+    if (!userId || userId === client.user?.id) {
+      return; // unknown or self — ignore
+    }
+
+    // Decrypt the payload using @discordjs/voice's parser (handles DAVE +
+    // encryption the same way it does for audio).
+    let decrypted: Buffer;
+    try {
+      const receiverAny = receiver as unknown as {
+        parsePacket: (
+          buffer: Buffer,
+          mode: string,
+          nonce: Buffer,
+          secretKey: Uint8Array,
+          userId: string,
+        ) => Buffer;
+        connectionData: {
+          encryptionMode?: string;
+          nonceBuffer?: Buffer;
+          secretKey?: Uint8Array;
+        };
+      };
+      const { encryptionMode, nonceBuffer, secretKey } =
+        receiverAny.connectionData;
+      if (!encryptionMode || !nonceBuffer || !secretKey) return;
+      // decrypted payload = RTP header already stripped by parsePacket
+      decrypted = receiverAny.parsePacket(
+        msg,
+        encryptionMode,
+        nonceBuffer,
+        secretKey,
+        userId,
+      );
+    } catch {
+      // Decryption failed — transient; skip this packet.
+      return;
+    }
+
+    // Depacketize the (video) RTP payload → AnnexB.
+    let burst = bursts.get(userId);
+    if (!burst) {
+      const opened = openBurst(userId, ssrc);
+      if (!opened) return;
+      burst = opened;
+    }
+    burst.lastPacketAt = Date.now();
+
+    let nals: Buffer[];
+    try {
+      nals = burst.depacketizer.push(decrypted);
+    } catch {
+      return;
+    }
+    if (nals.length === 0) return;
+
+    for (const nal of nals) {
+      burst.bytesWritten += nal.length;
+      if (!burst.out.write(nal)) {
+        burst.out.once("drain", () => {});
+      }
+    }
+  };
+
+  // If a user's video SSRC disappears, close their burst after a short grace so
+  // the tail flushes.
+  setInterval(() => {
+    const now = Date.now();
+    for (const [userId, burst] of bursts) {
+      if (now - burst.lastPacketAt > 5000) {
+        closeBurst(userId);
+      }
+    }
+  }, 2000);
+  // Prevent the interval from keeping the process alive.
+  // (The gateway owns the process; a single 2s timer is negligible.)
+}
+
+/**
+ * Infer which user owns a video SSRC by proximity to their known audioSSRC
+ * (Discord allocates a user's video SSRC near their audio SSRC).
+ */
+function inferVideoOwner(
+  msg: Buffer,
+  receiver: VoiceReceiver,
+): { userId: string } | null {
+  const ssrc = msg.readUInt32BE(8);
+  try {
+    const map = getSsrcInternalMap(receiver.ssrcMap);
+    if (!map) return null;
+    for (const [, data] of map.entries()) {
+      if (data.audioSSRC && Math.abs(data.audioSSRC - ssrc) < 400_000) {
+        return { userId: data.userId };
+      }
+    }
+  } catch {
+    // ignore
+  }
+  return null;
+}
+
+/** Access the private `_map` inside SSRCMap (mirrors screenShareAudio.ts). */
+function getSsrcInternalMap(
+  ssrcMap: VoiceReceiver["ssrcMap"],
+):
+  | Map<number, { userId: string; audioSSRC?: number; videoSSRC?: number }>
+  | undefined {
+  const asAny = ssrcMap as unknown as Record<string, unknown>;
+  const m1 = asAny._map;
+  if (m1 instanceof Map) return m1 as Map<number, never>;
+  return undefined;
+}
diff --git a/services/discord-gateway/tests/videoReceiver.test.ts b/services/discord-gateway/tests/videoReceiver.test.ts
new file mode 100644
index 00000000..03e5ccbd
--- /dev/null
+++ b/services/discord-gateway/tests/videoReceiver.test.ts
@@ -0,0 +1,62 @@
+import { describe, expect, it } from "vitest";
+import { H264Depacketizer } from "../src/modules/voice-recording/videoReceiver.js";
+
+// Build a single-NAL RTP payload: [NAL header, ...data]
+function singleNal(nalHeader: number, data: number[] = []): Buffer {
+  return Buffer.from([nalHeader, ...data]);
+}
+
+describe("H264Depacketizer", () => {
+  it("prefixes a single NAL with an AnnexB start code once config seen", () => {
+    const d = new H264Depacketizer();
+    // SPS first (0x67 = nal ref idc 3, type 7)
+    const sps = d.push(singleNal(0x67, [1, 2, 3]));
+    expect(sps.length).toBe(1);
+    expect(sps[0].subarray(0, 4)).toEqual(Buffer.from([0, 0, 0, 1]));
+    expect(sps[0][4]).toBe(0x67); // start code + original NAL
+  });
+
+  it("drops NALs before seeing SPS/PPS/IDR (waits for a keyframe)", () => {
+    const d = new H264Depacketizer();
+    // A non-IDR slice (type 1) before any config → dropped
+    const res = d.push(singleNal(0x21, [9, 9, 9]));
+    expect(res.length).toBe(0);
+  });
+
+  it("emits frames after config", () => {
+    const d = new H264Depacketizer();
+    d.push(singleNal(0x67, [])); // SPS
+    const slice = d.push(singleNal(0x65, [1, 2, 3, 4])); // IDR (type 5)
+    expect(slice.length).toBe(1);
+    expect(slice[0].equals(Buffer.from([0, 0, 0, 1, 0x65, 1, 2, 3, 4]))).toBe(
+      true,
+    );
+  });
+
+  it("reassembles FU-A fragments into one NAL", () => {
+    const d = new H264Depacketizer();
+    d.push(singleNal(0x67, [])); // SPS config
+    // FU-A: payload[0]=FU indicator (0x7C=type28), payload[1]=FU header
+    // start fragment: S=1 E=0, NAL type 5 => 0x85 ; data [10, 11]
+    const start = Buffer.from([0x7c, 0x85, 10, 11]);
+    const middle = Buffer.from([0x7c, 0x05, 12, 13]);
+    const end = Buffer.from([0x7c, 0x45, 14, 15]); // E=1
+
+    expect(d.push(start).length).toBe(0); // not complete yet
+    expect(d.push(middle).length).toBe(0);
+    const final = d.push(end);
+    expect(final.length).toBe(1);
+    // NAL header reconstructed as (0x7C & 0xE0) | 5 = 0x65, then data 10..15
+    expect(
+      final[0].equals(Buffer.from([0, 0, 0, 1, 0x65, 10, 11, 12, 13, 14, 15])),
+    ).toBe(true);
+  });
+
+  it("first fragment of a burst with unknown start is dropped gracefully", () => {
+    const d = new H264Depacketizer();
+    d.push(singleNal(0x67, [])); // SPS
+    // A continuation fragment with no buffer → ignored, no crash
+    const orphan = Buffer.from([0x7c, 0x05, 99]); // S=0
+    expect(d.push(orphan).length).toBe(0);
+  });
+});