Root cause: BaseMediaConnection.sendOpcode is a silent no-op when
ws.readyState !== OPEN. In GoLive, playStream() calls setVideoAttributes(true)
+ setSpeaking(true) the instant createStream() resolves (right after
SELECT_PROTOCOL_ACK), but the StreamConnection WebSocket can still be in
CONNECTING for a few ms — so op 12 (VIDEO, activating the video SSRC) was
silently DROPPED every session. Empirically verified: 0 ops 12/5 ever logged
across the entire journal, yet 10k+ video frames were sent and audio played
(audio SSRC is activated via the VoiceConnection handshake, independent of
GoLive op 12). Discord's media server thus received video RTP on video_ssrc
but was never told to forward it → black/broken shared-screen video with
working voice.
sendOpcodeWhenOpen retries up to ~2s for ws OPEN instead of dropping. Also
emits a=fmtp:101 packetization-mode=1;profile-level-id=42e01f in the answer
SDP (H264 FU-A fragments require packetization-mode=1 to reassemble).
Also removes pre-existing noNonNullAssertion lint (biome 2.5.8 now errors)
that was blocking the deploy CI.
Root cause: BaseMediaConnection.sendOpcode is a silent no-op when
ws.readyState !== OPEN. In GoLive, playStream() calls
setVideoAttributes(true) + setSpeaking(true) the instant createStream()
resolves (right after SELECT_PROTOCOL_ACK), but the StreamConnection WebSocket
can still be in CONNECTING for a few ms — so op 12 (VIDEO, enabling the video
SSRC) was silently DROPPED every session. Empirically verified: 0 ops 12/5 ever
logged across the entire journal, yet 10k+ video frames were sent and audio
played (audio SSRC is activated via the VoiceConnection handshake, independent
of GoLive op 12). Discord's media server thus received video RTP on video_ssrc
but was never told to forward it → black/broken shared-screen video with
working voice.
sendOpcodeWhenOpen retries up to ~2s for ws OPEN instead of dropping. Also
keeps the H264 packetization-mode=1 answer-SVP (defensive SDP correctness).
Also fix: emit a=fmtp:101 packetization-mode=1;profile-level-id=42e01f in the
answer SDP — H264 FU-A fragments require packetization-mode=1 to reassemble.
Screen share showed a single frozen frame: the GoLive pipeline sent video
only (-"-an", h264 muxer cannot carry audio) so the audio SSRC never
transmitted and Discord kept the stream in thumbnail state.
- prepareStream: mux NUT when includeAudio (h264 muxer drops audio) and
return the actual container format
- Demuxer: support NUT input with a second output pipe (fd3) carrying
Ogg Opus; parse OGG pages into opus frames (20ms, 48kHz) emitted as
GoLiveFrames; fix metadata parsing that dropped the audio stream line
when it arrived in a later stderr chunk (early parsedMeta return)
- playStream: pipe audio.stream into AudioStream → RTP on the audio SSRC
- Encoders: -x264-params repeat-headers=1 → SPS/PPS inline before EVERY
IDR (NUT remux drops container extradata; also enables PLI recovery)
- screenShareController: includeAudio true
- tests: demuxerNut.test.ts — OGG parser unit test + real ffmpeg NUT
integration (video access units + parsed opus frames)
Test 00:32 confirmed the video pipeline WORKS (1410 frames @ 1280x720 sent,
ready=true, camera off) but the gateway crashed at stream stop:
unhandledRejection → graceful shutdown → systemd restart (bot offline).
Root cause candidates (both were fire-and-forget promises without .catch):
- BaseMediaConnection.setProtocols().then(...) — rejects when the PC is
closed while setProtocols is in flight (stream teardown)
- void webRtcConn.createOffer().then(...) — rejects when the PC closes
while the offer is still gathering
Fixes:
- .catch on both promise chains (log + continue; teardown is expected)
- unhandledRejection handler now treats transient stream errors (EPIPE,
ERR_STREAM_DESTROYED, ERR_STREAM_WRITE_AFTER_END, ECONNRESET) like
uncaughtException already does — warn + continue instead of shutting
down the whole gateway. Non-transient rejections still log + shutdown
(with String(reason) so the detail actually shows).
asepharyana