Usage: FFmpeg + MediaMTX

[JackLau1222]

To download and run mediamtx from https://github.com/bluenviron/mediamtx/releases

./mediamtx

To download the code and build FFmpeg, you can use the following command.

cd ~/git 
git clone https://git.ffmpeg.org/ffmpeg.git
cd ffmpeg
./configure --enable-openssl --enable-version3 \
    --enable-libx264 --enable-gpl --enable-libopus
make -j

Note: To enable DTLS handshake, OpenSSL is mandatory. Please install OpenSSL, for instance, brew install openssl, and then configure the environment by running export PKG_CONFIG_PATH="/usr/local/opt/openssl@3/lib/pkgconfig".

Remark: Please use OpenSSL 1.1.1 and newer.

Note: For demonstration purposes, you can install libx264 by running brew install x264 and libopus by running brew install opus.

Although WebRTC has the capability to support x264 main and high profiles without B frames, it is advisable to use the baseline profile for better compatibility. If your stream doesn't have these codecs, you can transcode it using FFmpeg.

~/git/ffmpeg-webrtc/ffmpeg -re -f lavfi -i testsrc=size=1280x720 -f lavfi -i sine=frequency=440 -pix_fmt yuv420p \
    -vcodec libx264 -profile:v baseline -r 25 -g 50 -acodec libopus -ar 48000 -ac 2 \
    -f whip "http://localhost:8889/mystream/whip"
# Or you can also capture your screen, and measure the end-to-end latency.
~/git/ffmpeg-webrtc/ffmpeg -f avfoundation -framerate 25 -pixel_format yuyv422 -i "2:0" \
    -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -preset:v ultrafast \
    -b:v 800k -s 1024x576 -r 25 -g 50 -tune zerolatency -threads 1 -bf 0 \
    -acodec libopus -ar 48000 -ac 2 \
    -f whip "http://localhost:8889/mystream/whip"