Webrtc - firefox codecs not supported by client - minimal example

[Dronakurl]

Question

I am stuck with setting up mediamtx to stream via webrtc.
Below, you find a minimal example as a Dockerfile.
Build this with docker build -t mediatest . and run it with

docker run -t --name mediatest -p 8554:8554 -p 8888:8888 -p 8889:8889 mediatest`

Entering "http:/localhost:8888/bunny", I get the expected video.
But 8889 gives "Error: codecs not supported by client".
As you can see in the ffmpeg command, I converted it to h264, and also checked that my browser (firefox on ubuntu linux) supports h264 also using the tool linked in mediamtx's readme.

Google chrome works fine, but not firefox.
As suggested in #3710, I had a look at about:support, too:

Codec Name	Software Decoding	Hardware Decoding
H264	Supported	Supported
VP9	Supported	Supported
VP8	Supported	Unsupported
AV1	Supported	Supported
HEVC	Unsupported	Unsupported
AAC	Supported	Unsupported
MP3	Supported	Unsupported
Opus	Supported	Unsupported
Vorbis	Supported	Unsupported
FLAC	Supported	Unsupported
Wave	Supported	Unsupported

FROM bluenviron/mediamtx:latest-ffmpeg AS server
RUN apk add wget

RUN mkdir -p /media && wget --progress=bar -O /media/bunny.mp4 https://sample-videos.com/video321/mp4/240/big_buck_bunny_240p_1mb.mp4

RUN cat <<EOF >> /mediamtx.yml
  bunny:
    runOnDemand: >
      ffmpeg -re
        -stream_loop -1
        -i /media/bunny.mp4
        -c:v libx264
        -pix_fmt yuv420p
        -preset ultrafast
        -b:v 600k
        -an
        -async 50
        -f rtsp rtsp://localhost:\$RTSP_PORT/\$MTX_PATH
    runOnInitRestart: yes
EOF

ENTRYPOINT [ "/mediamtx" ]

Here is an extract from the log after first streaming with HLS and then with WEBRTC.

[libx264 @ 0x79d99ba14880] using SAR=1/1
[libx264 @ 0x79d99ba14880] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x79d99ba14880] profile Constrained Baseline, level 2.0, 4:2:0, 8-bit
[libx264 @ 0x79d99ba14880] 264 - core 164 - H.264/MPEG-4 AVC codec - Copyleft 2003-2023 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=7 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=15 scenecut=0 intra_refresh=0 rc=abr mbtree=0 bitrate=600 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
2025/03/03 19:55:44 INF [RTSP] [conn [::1]:41674] opened
2025/03/03 19:55:44 INF [RTSP] [session b2df941e] created by [::1]:41674
2025/03/03 19:55:44 INF [RTSP] [session b2df941e] is publishing to path 'bunny', 2 tracks (H264, MPEG-4 Audio)
2025/03/03 19:55:44 INF [HLS] [muxer bunny] is converting into HLS, 2 tracks (MPEG-4 Audio, H264)
Output #0, rtsp, to 'rtsp://localhost:8554/bunny':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf60.16.100
  Stream #0:0(und): Video: h264, yuv420p(progressive), 320x240 [SAR 1:1 DAR 4:3], q=2-31, 600 kb/s, 15 fps, 90k tbn (default)
    Metadata:
      creation_time   : 1970-01-01T00:00:00.000000Z
      handler_name    : VideoHandler
      vendor_id       : [0][0][0][0]
      encoder         : Lavc60.31.102 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/600000 buffer size: 0 vbv_delay: N/A
  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 382 kb/s (default)
    Metadata:
      creation_time   : 1970-01-01T00:00:00.000000Z
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]
2025/03/03 19:56:22 INF [WebRTC] [session 27a26ab0] created by 172.17.0.1:56936
2025/03/03 19:56:22 WAR [WebRTC] [session 27a26ab0] skipping track 2 (MPEG-4 Audio)
2025/03/03 19:56:22 INF [WebRTC] [session 27a26ab0] closed: codecs not supported by client

[dishmaker]

Looks like Chromium does POST whep H264, but firefox does not.

Chromium:

a=rtpmap:103 H264/90000
a=rtcp-fb:103 goog-remb
a=rtcp-fb:103 transport-cc
a=rtcp-fb:103 ccm fir
a=rtcp-fb:103 nack
a=rtcp-fb:103 nack pli
a=fmtp:103 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f

Firefox:

(not found)

BTW firefox probably needs profile=baseline in OBS (Custom Encoder Settings), as stated in
https://developer.mozilla.org/en-US/docs/Web/Media/Guides/Formats/WebRTC_codecs

[aler9]

This is impossible to answer if you don't provide server logs with logLevel set to debug, since they contain the SDPs exchanged between browser and sever, that contain supported codecs. Open a bug report and provide such log.

[Dronakurl]

I made an effort to provide a minimal reproducible example. Can you please help me to set the log level?

[aler9]

"issues", "new issue", "Bug Report" and follow instructions. The logLevel parameter is in the configuration file. Also specifying the browser version, OS version and device model is mandatory.

[dishmaker]

Issue exists:

• Issues loading WebRTC and HLS in Firefox #4408