go2rtc
go2rtc - ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
- zero-dependency and zero-config small app for all OS (Windows, macOS, Linux, ARM, etc.)
- zero-delay for all supported protocols (lowest possible streaming latency)
- zero-load on CPU for supported codecs
- on the fly transcoding for unsupported codecs via FFmpeg
- multi-source two-way codecs negotiation
- streaming from private networks via Ngrok or SSH-tunnels
Codecs negotiation
For example, you want to watch stream from Dahua IPC-K42 camera in your browser.
- this camera support codecs H264, H265 for send video, and you select
H264in camera settings - this camera support codecs AAC, PCMU, PCMA for send audio (from mic), and you select
AAC/16000in camera settings - this camera support codecs AAC, PCMU, PCMA for receive audio (to speaker), you don't need to select them
- your browser support codecs H264, VP8, VP9, AV1 for receive video, you don't need to select them
- your browser support codecs OPUS, PCMU, PCMA for send and receive audio, you don't need to select them
- you can't get camera audio directly, because their audio codecs doesn't match with your browser codecs
- so you decide to use transcoding via FFmpeg and add this setting to config YAML file
- you have chosen
OPUS/48000/2codec, because it is higher quality than the PCMU/8000 or PCMA/8000
- now you have stream with two sources - RTSP and FFmpeg
go2rtc automatically match codecs for you browser and all your stream sources. This called multi-source two-way codecs negotiation. And this is one of the main features of this app.
PS. You can select PCMU or PCMA codec in camera setting and don't use transcoding at all. Or you can select AAC codec for main stream and PCMU codec for second stream and add both RTSP to YAML config, this also will work fine.
streams:
dahua:
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
- ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif#audio=opus
Configuration
Create file go2rtc.yaml next to the app. Modules:
Streams
go2rtc support different stream source types. You can setup only one link as stream source or multiple.
- RTSP/RTSPS - most cameras on market
- RTMP
- FFmpeg/Exec - FFmpeg integration
- Hass - Home Assistant integration
RTSP source
- Support RTSP and RTSPS links with multiple video and audio tracks
- Support 2 way audio ONLY for ONVIF Profile T cameras (back channel connection)
Attention: proprietary 2 way audio standards are not supported!
streams:
rtsp_camera: rtsp://rtsp:12345678@192.168.1.123:554/av_stream/ch0
If your camera support two RTSP links - you can add both of them as sources. This is useful when streams has different codecs, as example AAC audio with main stream and PCMU/PCMA audio with second stream:
Attention: Dahua cameras has different capabilities for different RTSP links. For example, it has support multiple codecs for two way audio with &proto=Onvif in link and only one coded without it.
streams:
onvif_camera:
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=1
RTMP source
You can get stream from RTMP server, for example Frigate. Support ONLY H264 video codec without audio.
streams:
rtmp_stream: rtmp://192.168.1.123/live/camera1
FFmpeg source
You can get any stream or file or device via FFmpeg and push it to go2rtc via RTSP protocol.
Format: ffmpeg:{input}#{params}. Examples:
streams:
# [FILE] all tracks will be copied without transcoding codecs
file1: ffmpeg:~/media/BigBuckBunny.mp4
# [FILE] video will be transcoded to H264, audio will be skipped
file2: ffmpeg:~/media/BigBuckBunny.mp4#video=h264
# [FILE] video will be copied, audio will be transcoded to pcmu
file3: ffmpeg:~/media/BigBuckBunny.mp4#video=copy&audio=pcmu
# [HLS] video will be copied, audio will be skipped
hls: ffmpeg:https://devstreaming-cdn.apple.com/videos/streaming/examples/bipbop_16x9/gear5/prog_index.m3u8#video=copy
# [MJPEG] video will be transcoded to H264
mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg?stream=half&fps=15#video=h264
# [RTSP] video and audio will be copied
rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123:554/av_stream/ch0#video=copy&audio=copy
All trascoding formats has built-in templates. But you can override them via YAML config:
ffmpeg:
bin: ffmpeg # path to ffmpeg binary
link: -hide_banner -i {input} # if input is link
file: -hide_banner -re -stream_loop -1 -i {input} # if input not link
rtsp: -hide_banner -fflags nobuffer -flags low_delay -rtsp_transport tcp -i {input} # if input is RTSP link
output: -rtsp_transport tcp -f rtsp {output} # output
h264: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
h264/ultra: "-codec:v libx264 -g 30 -preset ultrafast -tune zerolatency"
h264/high: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency"
h265: "-codec:v libx265 -g 30 -preset ultrafast -tune zerolatency"
opus: "-codec:a libopus -ar 48000 -ac 2"
pcmu: "-codec:a pcm_mulaw -ar 8000 -ac 1"
pcmu/16000: "-codec:a pcm_mulaw -ar 16000 -ac 1"
pcmu/48000: "-codec:a pcm_mulaw -ar 48000 -ac 1"
pcma: "-codec:a pcm_alaw -ar 8000 -ac 1"
pcma/16000: "-codec:a pcm_alaw -ar 16000 -ac 1"
pcma/48000: "-codec:a pcm_alaw -ar 48000 -ac 1"
aac/16000: "-codec:a aac -ar 16000 -ac 1"
Exec source
FFmpeg source just a shortcut to exec source. You can get any stream or file or device via FFmpeg or GStreamer and push it to go2rtc via RTSP protocol:
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i ~/media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
Hass source
Support import camera links from Home Assistant config files.
- support ONLY Generic Camera, setup via GUI
hass:
config: "~/.homeassistant"
streams:
generic_camera: hass:Camera1 # Settings > Integrations > Integration Name
API server
api:
listen: ":3000" # HTTP API port
base_path: "" # API prefix for serve on suburl
static_dir: "www" # folder for static files
RTSP server
rtsp:
listen: ":554"
WebRTC server
webrtc:
listen: ":8555" # address of your local server (TCP)
candidates:
- 216.58.210.174:8555 # if you have static public IP-address
- 192.168.1.123:8555 # ip you have problems with UDP in LAN
- stun # if you have dynamic public IP-address (auto discovery via STUN)
ice_servers:
- urls: [stun:stun.l.google.com:19302]
- urls: [turn:123.123.123.123:3478]
username: your_user
credential: your_pass
Ngrok
ngrok:
command: ngrok tcp 8555 --authtoken eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
or
ngrok:
command: ngrok start --all --config ngrok.yml
Log
log:
level: info # default level
api: trace
exec: debug
ngrok: info
rtsp: warn
streams: error
webrtc: fatal