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AlexxIT logogo2rtc

Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.

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Pure Go implementation of the WebRTC API

Quick Overview

go2rtc is an innovative WebRTC streaming server written in Go. It's designed to be a universal stream source, capable of handling various protocols and codecs, making it ideal for video surveillance systems and IoT devices. The project aims to simplify the process of streaming video from different sources to web browsers and mobile applications.

Pros

  • Supports multiple streaming protocols (RTSP, RTMP, HLS, HTTP-FLV, WebRTC)
  • Efficient transcoding capabilities for various video and audio codecs
  • Lightweight and cross-platform, suitable for both powerful servers and small devices
  • Active development and community support

Cons

  • Relatively new project, may have some stability issues
  • Documentation could be more comprehensive for advanced use cases
  • Limited built-in security features compared to some established streaming solutions
  • Requires some technical knowledge to set up and configure properly

Code Examples

Here are a few examples of how to use go2rtc in your Go projects:

  1. Creating a new RTSPServer:
import "github.com/AlexxIT/go2rtc/pkg/rtsp"

server := rtsp.NewServer(":8554")
server.Start()
  1. Adding a stream source:
import "github.com/AlexxIT/go2rtc/pkg/streamer"

src, err := streamer.NewSource("rtsp://example.com/stream")
if err != nil {
    log.Fatal(err)
}
stream := streamer.NewStream(src)
  1. Handling WebRTC connections:
import "github.com/AlexxIT/go2rtc/pkg/webrtc"

conn := webrtc.NewConn()
offer := webrtc.Offer{} // Populate with SDP from client
answer, err := conn.CreateAnswer(offer)
if err != nil {
    log.Fatal(err)
}
// Send answer back to client

Getting Started

To get started with go2rtc, follow these steps:

  1. Install Go on your system (version 1.16 or later)
  2. Clone the repository: 
    git clone https://github.com/AlexxIT/go2rtc.git
  3. Navigate to the project directory and build: 
    cd go2rtc
go build
  4. Run the server: 
    ./go2rtc
  5. Configure your streams in the go2rtc.yaml file
  6. Access the web interface at http://localhost:1984

For more detailed configuration and usage instructions, refer to the project's documentation on GitHub.

Competitor Comparisons

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mediamtx

11,584

Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.

Pros of mediamtx

  • More comprehensive documentation and examples
  • Better support for RTSP and SRT protocols
  • More active development and frequent updates

Cons of mediamtx

  • Larger codebase, potentially more complex to understand and modify
  • Less focus on WebRTC functionality compared to go2rtc
  • May have higher resource usage due to broader feature set

Code Comparison

mediamtx:

type RTSPServer struct {
    rtspAddress string
    protocols   map[conf.Protocol]struct{}
    sessions    map[*gortsplib.ServerSession]*rtspSession
    mutex       sync.Mutex
}

go2rtc:

type RTSPServer struct {
    ln net.Listener
    tracks []*streamer.Track
    clients map[string]*RTSPClient
}

The code snippets show differences in structure and approach. mediamtx appears to have a more complex implementation with additional features, while go2rtc seems more focused on core RTSP functionality.

Both projects serve as media servers with real-time streaming capabilities, but they have different strengths and focuses. mediamtx offers broader protocol support and more extensive documentation, while go2rtc may be simpler to work with for specific use cases, particularly those centered around WebRTC.

bluenviron logo

mediamtx

11,584

Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.

Pros of mediamtx

  • More comprehensive documentation and examples
  • Better support for RTSP and SRT protocols
  • More active development and frequent updates

Cons of mediamtx

  • Larger codebase, potentially more complex to understand and modify
  • Less focus on WebRTC functionality compared to go2rtc
  • May have higher resource usage due to broader feature set

Code Comparison

mediamtx:

type RTSPServer struct {
    rtspAddress string
    protocols   map[conf.Protocol]struct{}
    sessions    map[*gortsplib.ServerSession]*rtspSession
    mutex       sync.Mutex
}

go2rtc:

type RTSPServer struct {
    ln net.Listener
    tracks []*streamer.Track
    clients map[string]*RTSPClient
}

The code snippets show differences in structure and approach. mediamtx appears to have a more complex implementation with additional features, while go2rtc seems more focused on core RTSP functionality.

Both projects serve as media servers with real-time streaming capabilities, but they have different strengths and focuses. mediamtx offers broader protocol support and more extensive documentation, while go2rtc may be simpler to work with for specific use cases, particularly those centered around WebRTC.

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srs

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SRS is a simple, high-efficiency, real-time media server supporting RTMP, WebRTC, HLS, HTTP-FLV, HTTP-TS, SRT, MPEG-DASH, and GB28181.

Pros of srs

  • More comprehensive streaming server with support for multiple protocols (RTMP, HLS, WebRTC, etc.)
  • Highly scalable and suitable for large-scale deployments
  • Extensive documentation and active community support

Cons of srs

  • More complex setup and configuration compared to go2rtc
  • Higher resource requirements due to its comprehensive feature set
  • Steeper learning curve for beginners

Code Comparison

srs (C++):

SrsRtcConnection::SrsRtcConnection(SrsRtcServer* s, const SrsContextId& cid)
{
    server = s;
    cid_ = cid;
    req = NULL;
    sendonly_ = false;
}

go2rtc (Go):

func (c *Conn) WritePacket(packet *rtp.Packet) (err error) {
    if c.mtu > 0 && packet.MarshalSize() > c.mtu {
        return c.writeLargePacket(packet)
    }
    return c.writePacket(packet)
}

The code snippets demonstrate different approaches and languages used in these projects. srs uses C++ for its implementation, while go2rtc is written in Go. The srs example shows a constructor for an RTC connection, while the go2rtc snippet handles packet writing with MTU considerations.

EasyDarwin logo

EasyDarwin

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open source、high performance、industrial rtsp streaming server,a lot of optimization on streaming relay,KeyFrame cache,RESTful,and web management,also EasyDarwin support distributed load balancing,a simple streaming media cloud platform architecture.高性能开源RTSP流媒体服务器，基于go语言研发，维护和优化：RTSP推模式转发、RTSP拉模式转发、录像、检索、回放、关键帧缓存、秒开画面、RESTful接口、WEB后台管理、分布式负载均衡，基于EasyDarwin构建出了一套基础的流媒体云视频平台架构！

Pros of EasyDarwin

  • More mature and established project with a longer development history
  • Supports a wider range of streaming protocols, including RTSP, RTMP, and HLS
  • Offers more comprehensive documentation and user guides

Cons of EasyDarwin

  • Written in C++, which may be less accessible for some developers compared to Go
  • Potentially more complex setup and configuration process
  • Less active development in recent years compared to go2rtc

Code Comparison

EasyDarwin (C++):

int EasyRTSPClient::Start(int nNetType /*= EASY_RTP_OVER_TCP*/)
{
    if(NULL == fRTSPClientHandle)
    {
        fRTSPClientHandle = EasyRTSP_Create();
    }
    if(NULL == fRTSPClientHandle) return -1;

go2rtc (Go):

func (s *Stream) Start() error {
    s.mu.Lock()
    defer s.mu.Unlock()

    if s.source != nil {
        return nil
    }

The code snippets show the initialization process for both projects. EasyDarwin uses C++ and focuses on creating an RTSP client handle, while go2rtc uses Go and implements a more straightforward stream start function.

deepch logo

RTSPtoWeb

1,250

RTSP Stream to WebBrowser

Pros of RTSPtoWeb

  • Simpler setup and configuration for basic RTSP to web streaming
  • Longer development history and potentially more stable
  • Includes a built-in web UI for stream management

Cons of RTSPtoWeb

  • Less flexible in terms of supported protocols and codecs
  • Limited active development and updates
  • Fewer advanced features compared to go2rtc

Code Comparison

RTSPtoWeb:

type Stream struct {
    URL      string    `json:"url"`
    Status   bool      `json:"status"`
    Type     string    `json:"type"`
    Name     string    `json:"name"`
    OnDemand bool      `json:"on_demand"`
    Channels []Channel `json:"channels"`
}

go2rtc:

type Stream struct {
    URL       string
    Media     []*streamer.Media
    Tracks    []*streamer.Track
    Consumers []Consumer
    Producers []Producer
}

The code comparison shows that go2rtc has a more complex stream structure, allowing for greater flexibility in handling different media types and streaming protocols. RTSPtoWeb's structure is simpler, focusing primarily on RTSP streams and basic metadata.

pion logo

webrtc

13,642

Pure Go implementation of the WebRTC API

Pros of webrtc

  • More comprehensive WebRTC implementation with broader feature set
  • Larger community and more frequent updates
  • Better documentation and examples for developers

Cons of webrtc

  • Steeper learning curve for beginners
  • May be overkill for simple streaming applications
  • Requires more setup and configuration for basic use cases

Code Comparison

go2rtc (simplified stream setup):

stream := rtc.NewStream("camera1")
stream.AddSource(rtsp.NewSource("rtsp://example.com/stream"))
stream.AddConsumer(webrtc.NewConsumer())

webrtc (basic peer connection):

peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{})
if err != nil {
    panic(err)
}
// Additional configuration and signaling required

Summary

go2rtc is designed for easy setup of media streaming with WebRTC, RTSP, and other protocols. It's more focused on simplifying the process of creating streaming applications.

webrtc provides a full WebRTC implementation with more flexibility and features, but requires more in-depth knowledge and setup. It's better suited for complex WebRTC applications beyond simple streaming.

Choose go2rtc for quick streaming setups, or webrtc for more comprehensive WebRTC functionality and control.

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README

go2rtc
stars docker pulls releases goreport

Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg, RTMP, etc.

Inspired by:

Fast start

  1. Download binary or use Docker or Home Assistant Add-on or Integration
  2. Open web interface: http://localhost:1984/

Optionally:

Developers:

go2rtc: Binary

Download binary for your OS from latest release:

  • go2rtc_win64.zip - Windows 10+ 64-bit
  • go2rtc_win32.zip - Windows 7+ 32-bit
  • go2rtc_win_arm64.zip - Windows ARM 64-bit
  • go2rtc_linux_amd64 - Linux 64-bit
  • go2rtc_linux_i386 - Linux 32-bit
  • go2rtc_linux_arm64 - Linux ARM 64-bit (ex. Raspberry 64-bit OS)
  • go2rtc_linux_arm - Linux ARM 32-bit (ex. Raspberry 32-bit OS)
  • go2rtc_linux_armv6 - Linux ARMv6 (for old Raspberry 1 and Zero)
  • go2rtc_linux_mipsel - Linux MIPS (ex. Xiaomi Gateway 3, Wyze cameras)
  • go2rtc_mac_amd64.zip - macOS 10.13+ Intel 64-bit
  • go2rtc_mac_arm64.zip - macOS ARM 64-bit

Don't forget to fix the rights chmod +x go2rtc_xxx_xxx on Linux and Mac.

go2rtc: Docker

The Docker container alexxit/go2rtc supports multiple architectures including amd64, 386, arm64, and arm. This container offers the same functionality as the Home Assistant Add-on but is designed to operate independently of Home Assistant. It comes preinstalled with FFmpeg, ngrok, and Python.

go2rtc: Home Assistant Add-on

  1. Install Add-On:
    • Settings > Add-ons > Plus > Repositories > Add https://github.com/AlexxIT/hassio-addons
    • go2rtc > Install > Start
  2. Setup Integration

go2rtc: Home Assistant Integration

WebRTC Camera custom component can be used on any Home Assistant installation, including HassWP on Windows. It can automatically download and use the latest version of go2rtc. Or it can connect to an existing version of go2rtc. Addon installation in this case is optional.

go2rtc: Dev version

Latest, but maybe unstable version:

  • Binary: latest nightly release
  • Docker: alexxit/go2rtc:master or alexxit/go2rtc:master-hardware versions
  • Hass Add-on: go2rtc master or go2rtc master hardware versions

Configuration

  • by default go2rtc will search go2rtc.yaml in the current work directory
  • api server will start on default 1984 port (TCP)
  • rtsp server will start on default 8554 port (TCP)
  • webrtc will use port 8555 (TCP/UDP) for connections
  • ffmpeg will use default transcoding options

Configuration options and a complete list of settings can be found in the wiki.

Available modules:

  • streams
  • api - HTTP API (important for WebRTC support)
  • rtsp - RTSP Server (important for FFmpeg support)
  • webrtc - WebRTC Server
  • mp4 - MSE, MP4 stream and MP4 shapshot Server
  • hls - HLS TS or fMP4 stream Server
  • mjpeg - MJPEG Server
  • ffmpeg - FFmpeg integration
  • ngrok - ngrok integration (external access for private network)
  • hass - Home Assistant integration
  • log - logs config

Module: Streams

go2rtc support different stream source types. You can config one or multiple links of any type as stream source.

Available source types:

  • rtsp - RTSP and RTSPS cameras with two way audio support
  • rtmp - RTMP streams
  • http - HTTP-FLV, MPEG-TS, JPEG (snapshots), MJPEG streams
  • onvif - get camera RTSP link and snapshot link using ONVIF protocol
  • ffmpeg - FFmpeg integration (HLS, files and many others)
  • ffmpeg:device - local USB Camera or Webcam
  • exec - get media from external app output
  • echo - get stream link from bash or python
  • expr - get stream link via built-in expression language
  • homekit - streaming from HomeKit Camera
  • bubble - streaming from ESeeCloud/dvr163 NVR
  • dvrip - streaming from DVR-IP NVR
  • tapo - TP-Link Tapo cameras with two way audio support
  • kasa - TP-Link Kasa cameras
  • gopro - GoPro cameras
  • ivideon - public cameras from Ivideon service
  • hass - Home Assistant integration
  • isapi - two way audio for Hikvision (ISAPI) cameras
  • roborock - Roborock vacuums with cameras
  • webrtc - WebRTC/WHEP sources
  • webtorrent - WebTorrent source from another go2rtc

Read more about incoming sources

Two way audio

Supported for sources:

Two way audio can be used in browser with WebRTC technology. The browser will give access to the microphone only for HTTPS sites (read more).

go2rtc also support play audio files and live streams on this cameras.

Source: RTSP

streams:
  sonoff_camera: rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0
  dahua_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
  amcrest_doorbell:
    - rtsp://username:password@192.168.1.123:554/cam/realmonitor?channel=1&subtype=0#backchannel=0
  unifi_camera: rtspx://192.168.1.123:7441/fD6ouM72bWoFijxK
  glichy_camera: ffmpeg:rtsp://username:password@192.168.1.123/live/ch00_1

Recommendations

  • Amcrest Doorbell users may want to disable two way audio, because with an active stream you won't have a call button working. You need to add #backchannel=0 to the end of your RTSP link in YAML config file
  • Dahua Doorbell users may want to change backchannel audio codec
  • Reolink users may want NOT to use RTSP protocol at all, some camera models have a very awful unusable stream implementation
  • Ubiquiti UniFi users may want to disable HTTPS verification. Use rtspx:// prefix instead of rtsps://. And don't use ?enableSrtp suffix
  • TP-Link Tapo users may skip login and password, because go2rtc support login without them
  • If your camera has 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
  • If the stream from your camera is glitchy, try using ffmpeg source. It will not add CPU load if you won't use transcoding
  • If the stream from your camera is very glitchy, try to use transcoding with ffmpeg source

Other options

Format: rtsp...#{param1}#{param2}#{param3}

  • Add custom timeout #timeout=30 (in seconds)
  • Ignore audio - #media=video or ignore video - #media=audio
  • Ignore two way audio API #backchannel=0 - important for some glitchy cameras
  • Use WebSocket transport #transport=ws...

RTSP over WebSocket

streams:
  # WebSocket with authorization, RTSP - without
  axis-rtsp-ws:  rtsp://192.168.1.123:4567/axis-media/media.amp?overview=0&camera=1&resolution=1280x720&videoframeskipmode=empty&Axis-Orig-Sw=true#transport=ws://user:pass@192.168.1.123:4567/rtsp-over-websocket
  # WebSocket without authorization, RTSP - with
  dahua-rtsp-ws: rtsp://user:pass@192.168.1.123/cam/realmonitor?channel=1&subtype=1&proto=Private3#transport=ws://192.168.1.123/rtspoverwebsocket

Source: RTMP

You can get stream from RTMP server, for example Nginx with nginx-rtmp-module.

streams:
  rtmp_stream: rtmp://192.168.1.123/live/camera1

Source: HTTP

Support Content-Type:

  • HTTP-FLV (video/x-flv) - same as RTMP, but over HTTP
  • HTTP-JPEG (image/jpeg) - camera snapshot link, can be converted by go2rtc to MJPEG stream
  • HTTP-MJPEG (multipart/x) - simple MJPEG stream over HTTP
  • MPEG-TS (video/mpeg) - legacy streaming format

Source also support HTTP and TCP streams with autodetection for different formats: MJPEG, H.264/H.265 bitstream, MPEG-TS.

streams:
  # [HTTP-FLV] stream in video/x-flv format
  http_flv: http://192.168.1.123:20880/api/camera/stream/780900131155/657617
  
  # [JPEG] snapshots from Dahua camera, will be converted to MJPEG stream
  dahua_snap: http://admin:password@192.168.1.123/cgi-bin/snapshot.cgi?channel=1

  # [MJPEG] stream will be proxied without modification
  http_mjpeg: https://mjpeg.sanford.io/count.mjpeg

  # [MJPEG or H.264/H.265 bitstream or MPEG-TS]
  tcp_magic: tcp://192.168.1.123:12345

  # Add custom header
  custom_header: "https://mjpeg.sanford.io/count.mjpeg#header=Authorization: Bearer XXX"

PS. Dahua camera has bug: if you select MJPEG codec for RTSP second stream - snapshot won't work.

Source: ONVIF

New in v1.5.0

The source is not very useful if you already know RTSP and snapshot links for your camera. But it can be useful if you don't.

WebUI > Add webpage support ONVIF autodiscovery. Your server must be on the same subnet as the camera. If you use docker, you must use "network host".

streams:
  dahua1: onvif://admin:password@192.168.1.123
  reolink1: onvif://admin:password@192.168.1.123:8000
  tapo1: onvif://admin:password@192.168.1.123:2020

Source: FFmpeg

You can get any stream or file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.

  • FFmpeg preistalled for Docker and Hass Add-on users
  • Hass Add-on users can target files from /media folder

Format: ffmpeg:{input}#{param1}#{param2}#{param3}. 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#video=h264

  # [RTSP] video with rotation, should be transcoded, so select H264
  rotate: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=h264#rotate=90

All trascoding formats has built-in templates: h264, h265, opus, pcmu, pcmu/16000, pcmu/48000, pcma, pcma/16000, pcma/48000, aac, aac/16000.

But you can override them via YAML config. You can also add your own formats to config and use them with source params.

ffmpeg:
  bin: ffmpeg  # path to ffmpeg binary
  h264: "-codec:v libx264 -g:v 30 -preset:v superfast -tune:v zerolatency -profile:v main -level:v 4.1"
  mycodec: "-any args that supported by ffmpeg..."
  myinput: "-fflags nobuffer -flags low_delay -timeout 5000000 -i {input}"
  myraw: "-ss 00:00:20"
  • You can use go2rtc stream name as ffmpeg input (ex. ffmpeg:camera1#video=h264)
  • You can use video and audio params multiple times (ex. #video=copy#audio=copy#audio=pcmu)
  • You can use rotate param with 90, 180, 270 or -90 values, important with transcoding (ex. #video=h264#rotate=90)
  • You can use width and/or height params, important with transcoding (ex. #video=h264#width=1280)
  • You can use drawtext to add a timestamp (ex. drawtext=x=2:y=2:fontsize=12:fontcolor=white:box=1:boxcolor=black)
    • This will greatly increase the CPU of the server, even with hardware acceleration
  • You can use raw param for any additional FFmpeg arguments (ex. #raw=-vf transpose=1)
  • You can use input param to override default input template (ex. #input=rtsp/udp will change RTSP transport from TCP to UDP+TCP)
    • You can use raw input value (ex. #input=-timeout 5000000 -i {input})
    • You can add your own input templates

Read more about hardware acceleration.

PS. It is recommended to check the available hardware in the WebUI add page.

Source: FFmpeg Device

You can get video from any USB-camera or Webcam as RTSP or WebRTC stream. This is part of FFmpeg integration.

  • check available devices in Web interface
  • video_size and framerate must be supported by your camera!
  • for Linux supported only video for now
  • for macOS you can stream Facetime camera or whole Desktop!
  • for macOS important to set right framerate

Format: ffmpeg:device?{input-params}#{param1}#{param2}#{param3}

streams:
  linux_usbcam:   ffmpeg:device?video=0&video_size=1280x720#video=h264
  windows_webcam: ffmpeg:device?video=0#video=h264
  macos_facetime: ffmpeg:device?video=0&audio=1&video_size=1280x720&framerate=30#video=h264#audio=pcma

PS. It is recommended to check the available devices in the WebUI add page.

Source: Exec

Exec source can run any external application and expect data from it. Two transports are supported - pipe (from v1.5.0) and RTSP.

If you want to use RTSP transport - the command must contain the {output} argument in any place. On launch, it will be replaced by the local address of the RTSP server.

pipe reads data from app stdout in different formats: MJPEG, H.264/H.265 bitstream, MPEG-TS. Also pipe can write data to app stdin in two formats: PCMA and PCM/48000.

The source can be used with:

Pipe commands support parameters (format: exec:{command}#{param1}#{param2}):

  • killsignal - signal which will be send to stop the process (numeric form)
  • killtimeout - time in seconds for forced termination with sigkill
  • backchannel - enable backchannel for two-way audio
streams:
  stream: exec:ffmpeg -re -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
  picam_h264: exec:libcamera-vid -t 0 --inline -o -
  picam_mjpeg: exec:libcamera-vid -t 0 --codec mjpeg -o -
  pi5cam_h264: exec:libcamera-vid -t 0 --libav-format h264 -o -
  canon: exec:gphoto2 --capture-movie --stdout#killsignal=2#killtimeout=5
  play_pcma: exec:ffplay -fflags nobuffer -f alaw -ar 8000 -i -#backchannel=1
  play_pcm48k: exec:ffplay -fflags nobuffer -f s16be -ar 48000 -i -#backchannel=1

Source: Echo

Some sources may have a dynamic link. And you will need to get it using a bash or python script. Your script should echo a link to the source. RTSP, FFmpeg or any of the supported sources.

Docker and Hass Add-on users has preinstalled python3, curl, jq.

Check examples in wiki.

streams:
  apple_hls: echo:python3 hls.py https://developer.apple.com/streaming/examples/basic-stream-osx-ios5.html

Source: Expr

New in v1.8.2

Like echo source, but uses the built-in expr expression language (read more).

Source: HomeKit

Important:

  • You can use HomeKit Cameras without Apple devices (iPhone, iPad, etc.), it's just a yet another protocol
  • HomeKit device can be paired with only one ecosystem. So, if you have paired it to an iPhone (Apple Home) - you can't pair it with Home Assistant or go2rtc. Or if you have paired it to go2rtc - you can't pair it with iPhone
  • HomeKit device should be in same network with working mDNS between device and go2rtc

go2rtc support import paired HomeKit devices from Home Assistant. So you can use HomeKit camera with Hass and go2rtc simultaneously. If you using Hass, I recommend pairing devices with it, it will give you more options.

You can pair device with go2rtc on the HomeKit page. If you can't see your devices - reload the page. Also try reboot your HomeKit device (power off). If you still can't see it - you have a problems with mDNS.

If you see a device but it does not have a pair button - it is paired to some ecosystem (Apple Home, Home Assistant, HomeBridge etc). You need to delete device from that ecosystem, and it will be available for pairing. If you cannot unpair device, you will have to reset it.

Important:

  • HomeKit audio uses very non-standard AAC-ELD codec with very non-standard params and specification violation
  • Audio can't be played in VLC and probably any other player
  • Audio should be transcoded for using with MSE, WebRTC, etc.

Recommended settings for using HomeKit Camera with WebRTC, MSE, MP4, RTSP:

streams:
  aqara_g3:
    - hass:Camera-Hub-G3-AB12
    - ffmpeg:aqara_g3#audio=aac#audio=opus

RTSP link with "normal" audio for any player: rtsp://192.168.1.123:8554/aqara_g3?video&audio=aac

This source is in active development! Tested only with Aqara Camera Hub G3 (both EU and CN versions).

Source: Bubble

New in v1.6.1

Other names: ESeeCloud, dvr163.

  • you can skip username, password, port, ch and stream if they are default
  • setup separate streams for different channels and streams
streams:
  camera1: bubble://username:password@192.168.1.123:34567/bubble/live?ch=0&stream=0

Source: DVRIP

New in v1.2.0

Other names: DVR-IP, NetSurveillance, Sofia protocol (NETsurveillance ActiveX plugin XMeye SDK).

  • you can skip username, password, port, channel and subtype if they are default
  • setup separate streams for different channels
  • use subtype=0 for Main stream, and subtype=1 for Extra1 stream
  • only the TCP protocol is supported
streams:
  only_stream: dvrip://username:password@192.168.1.123:34567?channel=0&subtype=0
  only_tts: dvrip://username:password@192.168.1.123:34567?backchannel=1
  two_way_audio:
    - dvrip://username:password@192.168.1.123:34567?channel=0&subtype=0
    - dvrip://username:password@192.168.1.123:34567?backchannel=1

Source: Tapo

New in v1.2.0

TP-Link Tapo proprietary camera protocol with two way audio support.

  • stream quality is the same as RTSP protocol
  • use the cloud password, this is not the RTSP password! you do not need to add a login!
  • you can also use UPPERCASE MD5 hash from your cloud password with admin username
  • some new camera firmwares requires SHA256 instead of MD5
streams:
  # cloud password without username
  camera1: tapo://cloud-password@192.168.1.123
  # admin username and UPPERCASE MD5 cloud-password hash
  camera2: tapo://admin:UPPERCASE-MD5@192.168.1.123
  # admin username and UPPERCASE SHA256 cloud-password hash
  camera3: tapo://admin:UPPERCASE-SHA256@192.168.1.123

echo -n "cloud password" | md5 | awk '{print toupper($0)}'
echo -n "cloud password" | shasum -a 256 | awk '{print toupper($0)}'

Source: Kasa

New in v1.7.0

TP-Link Kasa non-standard protocol more info.

  • username - urlsafe email, alex@gmail.com -> alex%40gmail.com
  • password - base64password, secret1 -> c2VjcmV0MQ==
streams:
  kc401: kasa://username:password@192.168.1.123:19443/https/stream/mixed

Tested: KD110, KC200, KC401, KC420WS, EC71.

Source: GoPro

New in v1.8.3

Support streaming from GoPro cameras, connected via USB or Wi-Fi to Linux, Mac, Windows. Read more.

Source: Ivideon

Support public cameras from service Ivideon.

streams:
  quailcam: ivideon:100-tu5dkUPct39cTp9oNEN2B6/0

Source: Hass

Support import camera links from Home Assistant config files:

hass:
  config: "/config"  # skip this setting if you Hass Add-on user

streams:
  generic_camera: hass:Camera1  # Settings > Integrations > Integration Name
  aqara_g3: hass:Camera-Hub-G3-AB12

WebRTC Cameras (from v1.6.0)

Any cameras in WebRTC format are supported. But at the moment Home Assistant only supports some Nest cameras in this fomat.

Important. The Nest API only allows you to get a link to a stream for 5 minutes. Do not use this with Frigate! If the stream expires, Frigate will consume all available ram on your machine within seconds. It's recommended to use Nest source - it supports extending the stream.

streams:
  # link to Home Assistant Supervised
  hass-webrtc1: hass://supervisor?entity_id=camera.nest_doorbell
  # link to external Hass with Long-Lived Access Tokens
  hass-webrtc2: hass://192.168.1.123:8123?entity_id=camera.nest_doorbell&token=eyXYZ...

RTSP Cameras

By default, the Home Assistant API does not allow you to get dynamic RTSP link to a camera stream. So more cameras, like Tuya, and possibly others can also be imported by using this method.

Source: ISAPI

New in v1.3.0

This source type support only backchannel audio for Hikvision ISAPI protocol. So it should be used as second source in addition to the RTSP protocol.

streams:
  hikvision1:
    - rtsp://admin:password@192.168.1.123:554/Streaming/Channels/101
    - isapi://admin:password@192.168.1.123:80/

Source: Nest

New in v1.6.0

Currently only WebRTC cameras are supported.

For simplicity, it is recommended to connect the Nest/WebRTC camera to the Home Assistant. But if you can somehow get the below parameters - Nest/WebRTC source will work without Hass.

streams:
  nest-doorbell: nest:?client_id=***&client_secret=***&refresh_token=***&project_id=***&device_id=***

Source: Roborock

New in v1.3.0

This source type support Roborock vacuums with cameras. Known working models:

  • Roborock S6 MaxV - only video (the vacuum has no microphone)
  • Roborock S7 MaxV - video and two way audio

Source support load Roborock credentials from Home Assistant custom integration. Otherwise, you need to log in to your Roborock account (MiHome account is not supported). Go to: go2rtc WebUI > Add webpage. Copy roborock://... source for your vacuum and paste it to go2rtc.yaml config.

If you have graphic pin for your vacuum - add it as numeric pin (lines: 123, 456, 678) to the end of the roborock-link.

Source: WebRTC

New in v1.3.0

This source type support four connection formats.

whep

WebRTC/WHEP - is replaced by WebRTC/WISH standard for WebRTC video/audio viewers. But it may already be supported in some third-party software. It is supported in go2rtc.

go2rtc

This format is only supported in go2rtc. Unlike WHEP it supports asynchronous WebRTC connection and two way audio.

openipc (from v1.7.0)

Support connection to OpenIPC cameras.

wyze (from v1.6.1)

Supports connection to Wyze cameras, using WebRTC protocol. You can use docker-wyze-bridge project to get connection credentials.

kinesis (from v1.6.1)

Supports Amazon Kinesis Video Streams, using WebRTC protocol. You need to specify signalling WebSocket URL with all credentials in query params, client_id and ice_servers list in JSON format.

streams:
  webrtc-whep:    webrtc:http://192.168.1.123:1984/api/webrtc?src=camera1
  webrtc-go2rtc:  webrtc:ws://192.168.1.123:1984/api/ws?src=camera1
  webrtc-openipc: webrtc:ws://192.168.1.123/webrtc_ws#format=openipc#ice_servers=[{"urls":"stun:stun.kinesisvideo.eu-north-1.amazonaws.com:443"}]
  webrtc-wyze:    webrtc:http://192.168.1.123:5000/signaling/camera1?kvs#format=wyze
  webrtc-kinesis: webrtc:wss://...amazonaws.com/?...#format=kinesis#client_id=...#ice_servers=[{...},{...}]

PS. For kinesis sources you can use echo to get connection params using bash/python or any other script language.

Source: WebTorrent

New in v1.3.0

This source can get a stream from another go2rtc via WebTorrent protocol.

streams:
  webtorrent1: webtorrent:?share=huofssuxaty00izc&pwd=k3l2j9djeg8v8r7e

Incoming sources

By default, go2rtc establishes a connection to the source when any client requests it. Go2rtc drops the connection to the source when it has no clients left.

  • Go2rtc also can accepts incoming sources in RTSP, RTMP, HTTP and WebRTC/WHIP formats
  • Go2rtc won't stop such a source if it has no clients
  • You can push data only to existing stream (create stream with empty source in config)
  • You can push multiple incoming sources to same stream
  • You can push data to non empty stream, so it will have additional codecs inside

Examples

  • RTSP with any codec 
    ffmpeg -re -i BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp rtsp://localhost:8554/camera1
  • HTTP-MJPEG with MJPEG codec 
    ffmpeg -re -i BigBuckBunny.mp4 -c mjpeg -f mpjpeg http://localhost:1984/api/stream.mjpeg?dst=camera1
  • HTTP-FLV with H264, AAC codecs 
    ffmpeg -re -i BigBuckBunny.mp4 -c copy -f flv http://localhost:1984/api/stream.flv?dst=camera1
  • MPEG-TS with H264 codec 
    ffmpeg -re -i BigBuckBunny.mp4 -c copy -f mpegts http://localhost:1984/api/stream.ts?dst=camera1

Incoming: Browser

New in v1.3.0

You can turn the browser of any PC or mobile into an IP-camera with support video and two way audio. Or even broadcast your PC screen:

  1. Create empty stream in the go2rtc.yaml
  2. Go to go2rtc WebUI
  3. Open links page for you stream
  4. Select camera+microphone or display+speaker option
  5. Open webrtc local page (your go2rtc should work over HTTPS!) or share link via WebTorrent technology (work over HTTPS by default)

Incoming: WebRTC/WHIP

New in v1.3.0

You can use OBS Studio or any other broadcast software with WHIP protocol support. This standard has not yet been approved. But you can download OBS Studio dev version:

Stream to camera

New in v1.3.0

go2rtc support play audio files (ex. music or TTS) and live streams (ex. radio) on cameras with two way audio support (RTSP/ONVIF cameras, TP-Link Tapo, Hikvision ISAPI, Roborock vacuums, any Browser).

API example:

POST http://localhost:1984/api/streams?dst=camera1&src=ffmpeg:http://example.com/song.mp3#audio=pcma#input=file
  • you can stream: local files, web files, live streams or any format, supported by FFmpeg
  • you should use ffmpeg source for transcoding audio to codec, that your camera supports
  • you can check camera codecs on the go2rtc WebUI info page when the stream is active
  • some cameras support only low quality PCMA/8000 codec (ex. Tapo)
  • it is recommended to choose higher quality formats if your camera supports them (ex. PCMA/48000 for some Dahua cameras)
  • if you play files over http-link, you need to add #input=file params for transcoding, so file will be transcoded and played in real time
  • if you play live streams, you should skip #input param, because it is already in real time
  • you can stop active playback by calling the API with the empty src parameter
  • you will see one active producer and one active consumer in go2rtc WebUI info page during streaming

Publish stream

New in v1.8.0

You can publish any stream to streaming services (YouTube, Telegram, etc.) via RTMP/RTMPS. Important:

  • Supported codecs: H264 for video and AAC for audio
  • AAC audio is required for YouTube, videos without audio will not work
  • You don't need to enable RTMP module listening for this task

You can use API:

POST http://localhost:1984/api/streams?src=camera1&dst=rtmps://...

Or config file:

publish:
  # publish stream "video_audio_transcode" to Telegram
  video_audio_transcode:
    - rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
  # publish stream "audio_transcode" to Telegram and YouTube
  audio_transcode:
    - rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
    - rtmp://xxx.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx-xxxx

streams:
  video_audio_transcode:
    - ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=h264#hardware#audio=aac
  audio_transcode:
    - ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=copy#audio=aac
  • Telegram Desktop App > Any public or private channel or group (where you admin) > Live stream > Start with... > Start streaming.
  • YouTube > Create > Go live > Stream latency: Ultra low-latency > Copy: Stream URL + Stream key.

Module: API

The HTTP API is the main part for interacting with the application. Default address: http://localhost:1984/.

Important! go2rtc passes requests from localhost and from unix socket without HTTP authorisation, even if you have it configured! It is your responsibility to set up secure external access to API. If not properly configured, an attacker can gain access to your cameras and even your server.

API description.

Module config

  • you can disable HTTP API with listen: "" and use, for example, only RTSP client/server protocol
  • you can enable HTTP API only on localhost with listen: "127.0.0.1:1984" setting
  • you can change API base_path and host go2rtc on your main app webserver suburl
  • all files from static_dir hosted on root path: /
  • you can use raw TLS cert/key content or path to files
api:
  listen: ":1984"    # default ":1984", HTTP API port ("" - disabled)
  username: "admin"  # default "", Basic auth for WebUI
  password: "pass"   # default "", Basic auth for WebUI
  base_path: "/rtc"  # default "", API prefix for serve on suburl (/api => /rtc/api)
  static_dir: "www"  # default "", folder for static files (custom web interface)
  origin: "*"        # default "", allow CORS requests (only * supported)
  tls_listen: ":443" # default "", enable HTTPS server
  tls_cert: |        # default "", PEM-encoded fullchain certificate for HTTPS
    -----BEGIN CERTIFICATE-----
    ...
    -----END CERTIFICATE-----
  tls_key: |         # default "", PEM-encoded private key for HTTPS
    -----BEGIN PRIVATE KEY-----
    ...
    -----END PRIVATE KEY-----
  unix_listen: "/tmp/go2rtc.sock"  # default "", unix socket listener for API

PS:

  • MJPEG over WebSocket plays better than native MJPEG because Chrome bug
  • MP4 over WebSocket was created only for Apple iOS because it doesn't support MSE and native MP4

Module: RTSP

You can get any stream as RTSP-stream: rtsp://192.168.1.123:8554/{stream_name}

You can enable external password protection for your RTSP streams. Password protection always disabled for localhost calls (ex. FFmpeg or Hass on same server).

rtsp:
  listen: ":8554"    # RTSP Server TCP port, default - 8554
  username: "admin"  # optional, default - disabled
  password: "pass"   # optional, default - disabled
  default_query: "video&audio"  # optional, default codecs filters

By default go2rtc provide RTSP-stream with only one first video and only one first audio. You can change it with the default_query setting:

  • default_query: "mp4" - MP4 compatible codecs (H264, H265, AAC)
  • default_query: "video=all&audio=all" - all tracks from all source (not all players can handle this)
  • default_query: "video=h264,h265" - only one video track (H264 or H265)
  • default_query: "video&audio=all" - only one first any video and all audio as separate tracks

Read more about codecs filters.

Module: RTMP

New in v1.8.0

You can get any stream as RTMP-stream: rtmp://192.168.1.123/{stream_name}. Only H264/AAC codecs supported right now.

Incoming stream in RTMP-format tested only with OBS Studio and Dahua camera. Different FFmpeg versions has differnt problems with this format.

rtmp:
  listen: ":1935"  # by default - disabled!

Module: WebRTC

In most cases WebRTC uses direct peer-to-peer connection from your browser to go2rtc and sends media data via UDP. It can't pass media data through your Nginx or Cloudflare or Nabu Casa HTTP TCP connection! It can automatically detects your external IP via public STUN server. It can establish a external direct connection via UDP hole punching technology even if you not open your server to the World.

But about 10-20% of users may need to configure additional settings for external access if mobile phone or go2rtc server behing Symmetric NAT.

  • by default, WebRTC uses both TCP and UDP on port 8555 for connections
  • you can use this port for external access
  • you can change the port in YAML config:
webrtc:
  listen: ":8555"  # address of your local server and port (TCP/UDP)

Static public IP

  • forward the port 8555 on your router (you can use same 8555 port or any other as external port)
  • add your external IP-address and external port to YAML config
webrtc:
  candidates:
    - 216.58.210.174:8555  # if you have static public IP-address

Dynamic public IP

  • forward the port 8555 on your router (you can use same 8555 port or any other as the external port)
  • add stun word and external port to YAML config
    • go2rtc automatically detects your external address with STUN-server
webrtc:
  candidates:
    - stun:8555  # if you have dynamic public IP-address

Private IP

ngrok:
  command: ...

Hard tech way 1. Own TCP-tunnel

If you have personal VPS, you can create TCP-tunnel and setup in the same way as "Static public IP". But use your VPS IP-address in YAML config.

Hard tech way 2. Using TURN-server

If you have personal VPS, you can install TURN server (e.g. coturn, config example).

webrtc:
  ice_servers:
    - urls: [stun:stun.l.google.com:19302]
    - urls: [turn:123.123.123.123:3478]
      username: your_user
      credential: your_pass

Module: HomeKit

New in v1.7.0

HomeKit module can work in two modes:

  • export any H264 camera to Apple HomeKit
  • transparent proxy any Apple HomeKit camera (Aqara, Eve, Eufy, etc.) back to Apple HomeKit, so you will have all camera features in Apple Home and also will have RTSP/WebRTC/MP4/etc. from your HomeKit camera

Important

  • HomeKit cameras supports only H264 video and OPUS audio

Minimal config

streams:
  dahua1: rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0
homekit:
  dahua1:  # same stream ID from streams list, default PIN - 19550224

Full config

streams:
  dahua1:
    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0
    - ffmpeg:dahua1#video=h264#hardware  # if your camera doesn't support H264, important for HomeKit
    - ffmpeg:dahua1#audio=opus           # only OPUS audio supported by HomeKit

homekit:
  dahua1:                   # same stream ID from streams list
    pin: 12345678           # custom PIN, default: 19550224
    name: Dahua camera      # custom camera name, default: generated from stream ID
    device_id: dahua1       # custom ID, default: generated from stream ID
    device_private: dahua1  # custom key, default: generated from stream ID

Proxy HomeKit camera

  • Video stream from HomeKit camera to Apple device (iPhone, AppleTV) will be transmitted directly
  • Video stream from HomeKit camera to RTSP/WebRTC/MP4/etc. will be transmitted via go2rtc
streams:
  aqara1:
    - homekit://...
    - ffmpeg:aqara1#audio=aac#audio=opus  # optional audio transcoding

homekit:
  aqara1:  # same stream ID from streams list

Module: WebTorrent

New in v1.3.0

This module support:

Securely and free. You do not need to open a public access to the go2rtc server. But in some cases (Symmetric NAT) you may need to set up external access to WebRTC module.

To generate sharing link or incoming link - goto go2rtc WebUI (stream links page). This link is temporary and will stop working after go2rtc is restarted!

You can create permanent external links in go2rtc config:

webtorrent:
  shares:
    super-secret-share:  # share name, should be unique among all go2rtc users!
      pwd: super-secret-password
      src: rtsp-dahua1   # stream name from streams section

Link example: https://alexxit.github.io/go2rtc/#share=02SNtgjKXY&pwd=wznEQqznxW&media=video+audio

TODO: article how it works...

Module: ngrok

With ngrok integration you can get external access to your streams in situations when you have Internet with private IP-address.

  • ngrok is pre-installed for Docker and Hass Add-on users
  • you may need external access for two different things:
    • WebRTC stream, so you need tunnel WebRTC TCP port (ex. 8555)
    • go2rtc web interface, so you need tunnel API HTTP port (ex. 1984)
  • ngrok support authorization for your web interface
  • ngrok automatically adds HTTPS to your web interface

The ngrok free subscription has the following limitations:

  • You can reserve a free domain for serving the web interface, but the TCP address you get will always be random and change with each restart of the ngrok agent (not a problem for webrtc stream)
  • You can forward multiple ports from a single agent, but you can only run one ngrok agent on the free plan

go2rtc will automatically get your external TCP address (if you enable it in ngrok config) and use it with WebRTC connection (if you enable it in webrtc config).

You need to manually download the ngrok agent app for your OS and register with the ngrok service.

Tunnel for only WebRTC Stream

You need to add your ngrok authtoken and WebRTC TCP port to YAML:

ngrok:
  command: ngrok tcp 8555 --authtoken eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw

Tunnel for WebRTC and Web interface

You need to create ngrok.yaml config file and add it to go2rtc config:

ngrok:
  command: ngrok start --all --config ngrok.yaml

ngrok config example:

version: "2"
authtoken: eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
tunnels:
  api:
    addr: 1984  # use the same port as in go2rtc config
    proto: http
    basic_auth:
      - admin:password  # you can set login/pass for your web interface
  webrtc:
    addr: 8555  # use the same port as in go2rtc config
    proto: tcp

See the ngrok agent documentation for more details on the ngrok configuration file.

Module: Hass

The best and easiest way to use go2rtc inside the Home Assistant is to install the custom integration WebRTC Camera and custom lovelace card.

But go2rtc is also compatible and can be used with RTSPtoWebRTC built-in integration.

You have several options on how to add a camera to Home Assistant:

  1. Camera RTSP source => Generic Camera
  2. Camera any source => go2rtc config => Generic Camera
    • Install any go2rtc
    • Add your stream to go2rtc config
    • Hass > Settings > Integrations > Add Integration > ONVIF > Host: 127.0.0.1, Port: 1984
    • Hass > Settings > Integrations > Add Integration > Generic Camera > Stream Source URL: rtsp://127.0.0.1:8554/camera1 (change to your stream name, leave everything else as is)

You have several options on how to watch the stream from the cameras in Home Assistant:

  1. Camera Entity => Picture Entity Card => Technology HLS, codecs: H264/H265/AAC, poor latency.
  2. Camera Entity => RTSPtoWebRTC => Picture Entity Card => Technology WebRTC, codecs: H264/PCMU/PCMA/OPUS, best latency.
    • Install any go2rtc
    • Hass > Settings > Integrations > Add Integration > RTSPtoWebRTC > http://127.0.0.1:1984/
    • RTSPtoWebRTC > Configure > STUN server: stun.l.google.com:19302
    • Use Picture Entity or Picture Glance lovelace card
  3. Camera Entity or Camera URL => WebRTC Camera => Technology: WebRTC/MSE/MP4/MJPEG, codecs: H264/H265/AAC/PCMU/PCMA/OPUS, best latency, best compatibility.
    • Install and add WebRTC Camera custom integration
    • Use WebRTC Camera custom lovelace card

You can add camera entity_id to go2rtc config if you need transcoding:

streams:
  "camera.hall": ffmpeg:{input}#video=copy#audio=opus

PS. Default Home Assistant lovelace cards don't support 2-way audio. You can use 2-way audio from Add-on Web UI. But you need use HTTPS to access the microphone. This is a browser restriction and cannot be avoided.

PS. There is also another nice card with go2rtc support - Frigate Lovelace Card.

Module: MP4

Provides several features:

  1. MSE stream (fMP4 over WebSocket)
  2. Camera snapshots in MP4 format (single frame), can be sent to Telegram
  3. HTTP progressive streaming (MP4 file stream) - bad format for streaming because of high start delay. This format doesn't work in all Safari browsers, but go2rtc will automatically redirect it to HLS/fMP4 it this case.

API examples:

  • MP4 snapshot: http://192.168.1.123:1984/api/frame.mp4?src=camera1 (H264, H265)
  • MP4 stream: http://192.168.1.123:1984/api/stream.mp4?src=camera1 (H264, H265, AAC)
  • MP4 file: http://192.168.1.123:1984/api/stream.mp4?src=camera1 (H264, H265*, AAC, OPUS, MP3, PCMA, PCMU, PCM)
    • You can use mp4, mp4=flac and mp4=all param for codec filters
    • You can use duration param in seconds (ex. duration=15)
    • You can use filename param (ex. filename=record.mp4)
    • You can use rotate param with 90, 180 or 270 values
    • You can use scale param with positive integer values (ex. scale=4:3)

Read more about codecs filters.

PS. Rotate and scale params don't use transcoding and change video using metadata.

Module: HLS

New in v1.1.0

HLS is the worst technology for real-time streaming. It can only be useful on devices that do not support more modern technology, like WebRTC, MSE/MP4.

The go2rtc implementation differs from the standards and may not work with all players.

API examples:

  • HLS/TS stream: http://192.168.1.123:1984/api/stream.m3u8?src=camera1 (H264)
  • HLS/fMP4 stream: http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4 (H264, H265, AAC)

Read more about codecs filters.

Module: MJPEG

Important. For stream as MJPEG format, your source MUST contain the MJPEG codec. If your stream has a MJPEG codec - you can receive MJPEG stream or JPEG snapshots via API.

You can receive an MJPEG stream in several ways:

  • some cameras support MJPEG codec inside RTSP stream (ex. second stream for Dahua cameras)
  • some cameras has HTTP link with MJPEG stream
  • some cameras has HTTP link with snapshots - go2rtc can convert them to MJPEG stream
  • you can convert H264/H265 stream from your camera via FFmpeg integraion

With this example, your stream will have both H264 and MJPEG codecs:

streams:
  camera1:
    - rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0
    - ffmpeg:camera1#video=mjpeg

API examples:

  • MJPEG stream: http://192.168.1.123:1984/api/stream.mjpeg?src=camera1
  • JPEG snapshots: http://192.168.1.123:1984/api/frame.jpeg?src=camera1
    • You can use width/w and/or height/h params
    • You can use rotate param with 90, 180, 270 or -90 values
    • You can use hardware/hw param read more

PS. This module also supports streaming to the server console (terminal) in the animated ASCII art format (read more):

Module: Log

You can set different log levels for different modules.

log:
  level: info  # default level
  api: trace
  exec: debug
  ngrok: info
  rtsp: warn
  streams: error
  webrtc: fatal

Security

By default go2rtc starts the Web interface on port 1984 and RTSP on port 8554, as well as use port 8555 for WebRTC connections. The three ports are accessible from your local network. So anyone on your local network can watch video from your cameras without authorization. The same rule applies to the Home Assistant Add-on.

This is not a problem if you trust your local network as much as I do. But you can change this behaviour with a go2rtc.yaml config:

api:
  listen: "127.0.0.1:1984" # localhost

rtsp:
  listen: "127.0.0.1:8554" # localhost

webrtc:
  listen: ":8555" # external TCP/UDP port
  • local access to RTSP is not a problem for FFmpeg integration, because it runs locally on your server
  • local access to API is not a problem for Home Assistant Add-on, because Hass runs locally on same server and Add-on Web UI protected with Hass authorization (Ingress feature)
  • external access to WebRTC TCP port is not a problem, because it used only for transmit encrypted media data
    • anyway you need to open this port to your local network and to the Internet in order for WebRTC to work

If you need Web interface protection without Home Assistant Add-on - you need to use reverse proxy, like Nginx, Caddy, ngrok, etc.

PS. Additionally WebRTC will try to use the 8555 UDP port for transmit encrypted media. It works without problems on the local network. And sometimes also works for external access, even if you haven't opened this port on your router (read more). But for stable external WebRTC access, you need to open the 8555 port on your router for both TCP and UDP.

Codecs filters

go2rtc can automatically detect which codecs your device supports for WebRTC and MSE technologies.

But it cannot be done for RTSP, HTTP progressive streaming, HLS technologies. You can manually add a codec filter when you create a link to a stream. The filters work the same for all three technologies. Filters do not create a new codec. They only select the suitable codec from existing sources. You can add new codecs to the stream using the FFmpeg transcoding.

Without filters:

  • RTSP will provide only the first video and only the first audio (any codec)
  • MP4 will include only compatible codecs (H264, H265, AAC)
  • HLS will output in the legacy TS format (H264 without audio)

Some examples:

  • rtsp://192.168.1.123:8554/camera1?mp4 - useful for recording as MP4 files (e.g. Hass or Frigate)
  • rtsp://192.168.1.123:8554/camera1?video=h264,h265&audio=aac - full version of the filter above
  • rtsp://192.168.1.123:8554/camera1?video=h264&audio=aac&audio=opus - H264 video codec and two separate audio tracks
  • rtsp://192.168.1.123:8554/camera1?video&audio=all - any video codec and all audio codecs as separate tracks
  • http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4 - HLS stream with MP4 compatible codecs (HLS/fMP4)
  • http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4=flac - HLS stream with PCMA/PCMU/PCM audio support (HLS/fMP4), won't work on old devices
  • http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=flac - MP4 file with PCMA/PCMU/PCM audio support, won't work on old devices (ex. iOS 12)
  • http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=all - MP4 file with non standard audio codecs, won't work on some players

Codecs madness

AVC/H.264 video can be played almost anywhere. But HEVC/H.265 has a lot of limitations in supporting with different devices and browsers. It's all about patents and money, you can't do anything about it.

DeviceWebRTCMSEHTTP*HLS
latencybestmediumbadbad
- Desktop Chrome 107+
- Desktop Edge
- Android Chrome 107+		H264
PCMU, PCMA
OPUS		H264, H265*
AAC, FLAC*
OPUS		H264, H265*
AAC, FLAC*
OPUS, MP3		no
Desktop FirefoxH264
PCMU, PCMA
OPUS		H264
AAC, FLAC*
OPUS		H264
AAC, FLAC*
OPUS		no
- Desktop Safari 14+
- iPad Safari 14+
- iPhone Safari 17.1+		H264, H265*
PCMU, PCMA
OPUS		H264, H265
AAC, FLAC*		no!H264, H265
AAC, FLAC*
iPhone Safari 14+H264, H265*
PCMU, PCMA
OPUS		no!no!H264, H265
AAC, FLAC*
macOS Hass AppnononoH264, H265
AAC, FLAC*

HTTP* - HTTP Progressive Streaming, not related with Progressive download, because the file has no size and no end

  • Chrome H265: read this and read this
  • Edge H265: read this
  • Desktop Safari H265: Menu > Develop > Experimental > WebRTC H265
  • iOS Safari H265: Settings > Safari > Advanced > Experimental > WebRTC H265

Audio

  • Go2rtc support automatic repack PCMA/PCMU/PCM codecs to FLAC for MSE/MP4/HLS so they will work almost anywhere
  • WebRTC audio codecs: PCMU/8000, PCMA/8000, OPUS/48000/2
  • OPUS and MP3 inside MP4 is part of the standard, but some players do not support them anyway (especially Apple)

Apple devices

  • all Apple devices don't support HTTP progressive streaming
  • iPhones don't support MSE technology because it competes with the HTTP Live Streaming (HLS) technology, invented by Apple
  • HLS is the worst technology for live streaming, it still exists only because of iPhones

Codec names

  • H264 = H.264 = AVC (Advanced Video Coding)
  • H265 = H.265 = HEVC (High Efficiency Video Coding)
  • PCMA = G.711 PCM (A-law) = PCM A-law (alaw)
  • PCMU = G.711 PCM (µ-law) = PCM mu-law (mulaw)
  • PCM = L16 = PCM signed 16-bit big-endian (s16be)
  • AAC = MPEG4-GENERIC
  • MP3 = MPEG-1 Audio Layer III or MPEG-2 Audio Layer III

Built-in transcoding

There are no plans to embed complex transcoding algorithms inside go2rtc. FFmpeg source does a great job with this. Including hardware acceleration support.

But go2rtc has some simple algorithms. They are turned on automatically, you do not need to set them up additionally.

PCM for MSE/MP4/HLS

Go2rtc can pack PCMA, PCMU and PCM codecs into an MP4 container so that they work in all browsers and all built-in players on modern devices. Including Apple QuickTime:

PCMA/PCMU => PCM => FLAC => MSE/MP4/HLS

Resample PCMA/PCMU for WebRTC

By default WebRTC support only PCMA/8000 and PCMU/8000. But go2rtc can automatically resample PCMA and PCMU codec with with a different sample rate. Also go2rtc can transcode PCM codec to PCMA/8000, so WebRTC can play it:

PCM/xxx => PCMA/8000 => WebRTC
PCMA/xxx => PCMA/8000 => WebRTC
PCMU/xxx => PCMU/8000 => WebRTC

Important

  • FLAC codec not supported in a RTSP stream. If you using Frigate or Hass for recording MP4 files with PCMA/PCMU/PCM audio - you should setup transcoding to AAC codec.
  • PCMA and PCMU are VERY low quality codecs. Them support only 256! different sounds. Use them only when you have no other options.

Codecs negotiation

For example, you want to watch RTSP-stream from Dahua IPC-K42 camera in your Chrome browser.

  • this camera support 2-way audio standard ONVIF Profile T
  • this camera support codecs H264, H265 for send video, and you select H264 in camera settings
  • this camera support codecs AAC, PCMU, PCMA for send audio (from mic), and you select AAC/16000 in 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 its 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/48000/2 codec, because it is higher quality than the PCMU/8000 or PCMA/8000

Now you have stream with two sources - RTSP and FFmpeg:

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#audio=opus

go2rtc automatically match codecs for you browser and all your stream sources. This called multi-source 2-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.

Projects using go2rtc

Distributions

Cameras experience

  • Dahua - reference implementation streaming protocols, a lot of settings, high stream quality, multiple streaming clients
  • EZVIZ - awful RTSP protocol realisation, many bugs in SDP
  • Hikvision - a lot of proprietary streaming technologies
  • Reolink - some models has awful unusable RTSP realisation and not best RTMP alternative (I recommend that you contact Reolink support for new firmware), few settings
  • Sonoff - very low stream quality, no settings, not best protocol implementation
  • TP-Link - few streaming clients, packet loss?
  • Chinese cheap noname cameras, Wyze Cams, Xiaomi cameras with hacks (usual has /live/ch00_1 in RTSP URL) - awful but usable RTSP protocol realisation, low stream quality, few settings, packet loss?

TIPS

Using apps for low RTSP delay

  • ffplay -fflags nobuffer -flags low_delay "rtsp://192.168.1.123:8554/camera1"
  • VLC > Preferences > Input / Codecs > Default Caching Level: Lowest Latency

Snapshots to Telegram

read more

FAQ

Q. What's the difference between go2rtc, WebRTC Camera and RTSPtoWebRTC?

go2rtc is a new version of the server-side WebRTC Camera integration, completely rewritten from scratch, with a number of fixes and a huge number of new features. It is compatible with native Home Assistant RTSPtoWebRTC integration. So you can use default lovelace Picture Entity or Picture Glance.

Q. Should I use go2rtc addon or WebRTC Camera integration?

go2rtc is more than just viewing your stream online with WebRTC/MSE/HLS/etc. You can use it all the time for your various tasks. But every time the Hass is rebooted - all integrations are also rebooted. So your streams may be interrupted if you use them in additional tasks.

Basic users can use WebRTC Camera integration. Advanced users can use go2rtc addon or Frigate 12+ addon.

Q. Which RTSP link should I use inside Hass?

You can use direct link to your cameras there (as you always do). go2rtc support zero-config feature. You may leave streams config section empty. And your streams will be created on the fly on first start from Hass. And your cameras will have multiple connections. Some from Hass directly and one from go2rtc.

Also you can specify your streams in go2rtc config file and use RTSP links to this addon. With additional features: multi-source codecs negotiation or FFmpeg transcoding for unsupported codecs. Or use them as source for Frigate. And your cameras will have one connection from go2rtc. And go2rtc will have multiple connection - some from Hass via RTSP protocol, some from your browser via WebRTC/MSE/HLS protocols.

Use any config what you like.

Q. What about lovelace card with support 2-way audio?

At this moment I am focused on improving stability and adding new features to go2rtc. Maybe someone could write such a card themselves. It's not difficult, I have some sketches.