RDOC WebRTC Streaming Server

Low-latency SRT → WebRTC relay for the Raumdock crew. Streamers push an SRT feed into a per-streamer MediaMTX instance; viewers watch in the browser over WebRTC (WHEP) behind a Caddy reverse proxy with automatic TLS via Cloudflare DNS.

Repo: github.com/cccdemon/CC-webrtc-server Deployed on: Proxmox LXC 103 (streamer) at /opt/webrtc, public IP 85.215.253.135.

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Architecture

 Streamer (OBS)                 LXC 103  "streamer"                     Viewer
 ┌───────────┐    SRT/mpegts   ┌──────────────────────────┐   WebRTC   ┌────────┐
 │  encoder  │ ───────────────▶│  mediamtx-<name>  :889x  │◀──────────▶│ browser│
 └───────────┘   :889x udp     │   (one per streamer)     │   WHEP     │  / OBS │
                               └──────────┬───────────────┘            └────────┘
                                          │ http :8889
                                   ┌──────▼───────┐  TLS (Cloudflare DNS-01)
                                   │  Caddy :443  │  <name>.streaming.raumdock.org
                                   └──────────────┘

Each streamer gets a dedicated MediaMTX container so ports, paths and load stay isolated. Caddy terminates TLS and reverse-proxies https://<name>.streaming.raumdock.org → that streamer's MediaMTX WebRTC port (:8889 inside the container network).

Services / streamers

Streamer	Container	SRT (udp)	API (tcp)	WebRTC ICE (udp)	Public host
jazzz	mediamtx-jazzz	8891	9991	8181	jazzz.streaming.raumdock.org
headwig	mediamtx-headwig	8892	9992	8182	headwig.streaming.raumdock.org
jericho	mediamtx-jericho	8893	9993	8183	jericho.streaming.raumdock.org
polder	mediamtx-polder	8894	9994	8184	polder.streaming.raumdock.org *
headlesschris	mediamtx-headlesschris	8895	9995	8185	headlesschris.streaming.raumdock.org

* polder exists in docker-compose.yml but currently has no Caddyfile entry (no public TLS host). Add one to expose it.

All MediaMTX instances share WebRTC media port :8889 inside their own container; only SRT/API/ICE ports are published to the host. RTSP is enabled (:8554), RTMP/HLS are off.

Components

• docker-compose.yml — one bluenviron/mediamtx:latest-ffmpeg service per streamer + the Caddy service. MediaMTX is configured purely via MTX_* env vars.
• caddy/Dockerfile — builds Caddy with the caddy-dns/cloudflare plugin via xcaddy (needed for DNS-01 TLS).
• caddy/Caddyfile — global ACME config + one reverse-proxy site block per streamer.

Configuration

Caddy needs Cloudflare credentials, set in docker-compose.yml under the caddy service:

environment:
  - CF_API_KEY=<cloudflare API token with DNS edit>
  - CF_API_EMAIL=<account email>

⚠️ The deployed compose contains a live Cloudflare API token. Treat it as a secret — rotate it if this repo is ever public, and prefer a .env file (git-ignored) over committing it.

Run

# on LXC 103
cd /opt/webrtc
docker compose up -d --build        # build Caddy plugin + start all streams
docker compose ps
docker compose logs -f mediamtx-jazzz

Add or restart a single streamer without touching the rest:

docker compose up -d mediamtx-headlesschris

Streamer (publish) setup — OBS

OBS → Settings → Stream → Service: SRT (or custom FFmpeg output):

srt://85.215.253.135:889x?streamid=publish:<path>

• 889x = the SRT port for your streamer row above.
• <path> = stream path, e.g. jazzz_game.

Viewer (playback)

Open https://<name>.streaming.raumdock.org/<path> in a browser — MediaMTX serves its built-in WebRTC/WHEP player.

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⚠️ Known issue: no audio in OBS / WebRTC playback

Symptom: video plays, but viewers (incl. OBS browser source) hear nothing when a colleague streams.

Root cause: MediaMTX logs on the host show:

INF [path jazzz_game] stream is available and online, 2 tracks (H264, MPEG-4 Audio)
WAR [WebRTC] [session …] skipping track 2 (MPEG-4 Audio)
INF [WebRTC] is reading from path 'jazzz_game', 1 track (H264)

The incoming SRT/MPEG-TS stream carries AAC ("MPEG-4 Audio"). WebRTC does not support AAC, and MediaMTX does not transcode — so it drops the audio track and only forwards H264 video. WebRTC audio must be Opus (or G711/G722).

Fixes (pick one):

1. Send Opus from the source (best). Configure the encoder to put Opus into the MPEG-TS/SRT stream. In OBS use a Custom Output (FFmpeg):

   • Container/format: mpegts
   • Audio encoder: libopus
   • (Video stays H264.) MediaMTX forwards Opus over WebRTC untouched → audio works.

2. Transcode on the server. Add a MediaMTX runOnReady (or a sidecar ffmpeg) that republishes the path with -c:a libopus, e.g. ffmpeg -i srt://… -c:v copy -c:a libopus -f rtsp rtsp://localhost:8554/<path>_opus, and point viewers at the _opus path. Costs CPU; needed only if sources can't emit Opus.

3. Use a WebRTC-native publish (WHIP) instead of SRT — browsers/OBS WHIP already negotiate Opus.

Note: the MediaMTX HTTP API (:999x) currently returns {"status":"error","error":"authentication error"}, so codecs are confirmed from container logs rather than the API.