Autoadd cameras from Hass config

.github/workflows/builder.yml

@@ -0,0 +1,59 @@
+# https://github.com/home-assistant/builder
+name: 'Builder'
+
+on:
+  push:
+    tags: [ 'v*' ]
+  workflow_dispatch:
+
+jobs:
+  hassio:
+    name: Hassio Addon
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout the repository
+        uses: actions/checkout@v3
+
+      - name: Login to DockerHub
+        uses: docker/login-action@v2
+        with:
+          username: ${{ secrets.DOCKERHUB_USERNAME }}
+          password: ${{ secrets.DOCKERHUB_TOKEN }}
+
+      - name: Branch name
+        run: |
+          VERSION="${GITHUB_REF#refs/tags/v}"
+          echo "REPO=alexxit/go2rtc" >> $GITHUB_ENV
+          echo "TAG=${VERSION}" >> $GITHUB_ENV
+          echo "IMAGE=alexxit/go2rtc:${VERSION}" >> $GITHUB_ENV
+
+      - name: Build amd64
+        uses: home-assistant/builder@master
+        with:
+          args: --amd64 --target build/hassio --version $TAG-amd64 --no-latest --docker-hub-check
+
+      - name: Build i386
+        uses: home-assistant/builder@master
+        with:
+          args: --i386 --target build/hassio --version $TAG-i386 --no-latest --docker-hub-check
+
+      - name: Build aarch64
+        uses: home-assistant/builder@master
+        with:
+          args: --aarch64 --target build/hassio --version $TAG-aarch64 --no-latest --docker-hub-check
+
+      - name: Build armv7
+        uses: home-assistant/builder@master
+        with:
+          args: --armv7 --target build/hassio --version $TAG-armv7 --no-latest --docker-hub-check
+
+      - name: Docker manifest
+        run: |
+          # thanks to https://github.com/aler9/rtsp-simple-server/blob/main/Makefile
+          docker manifest create "${IMAGE}" \
+            "${IMAGE}-amd64" "${IMAGE}-i386" "${IMAGE}-aarch64" "${IMAGE}-armv7"
+          docker manifest push "${IMAGE}"
+
+          docker manifest create "${REPO}:latest" \
+            "${IMAGE}-amd64" "${IMAGE}-i386" "${IMAGE}-aarch64" "${IMAGE}-armv7"
+          docker manifest push "${REPO}:latest"

README.md

@@ -1,20 +1,20 @@
 # go2rtc
 
-**go2rtc** - ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
+Ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
 
-- zero-dependency and zero-config small [app for all OS](#installation) (Windows, macOS, Linux, ARM)
+- zero-dependency and zero-config small [app for all OS](#go2rtc-binary) (Windows, macOS, Linux, ARM)
 - 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](#source-ffmpeg)
+- low CPU load for supported codecs
+- on the fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
 - multi-source 2-way [codecs negotiation](#codecs-negotiation)
-- streaming from private networks via [Ngrok or SSH-tunnels](#module-webrtc)
+- streaming from private networks via [Ngrok](#module-webrtc)
 
 **Inspired by:**
 
 - [webrtc](https://github.com/pion/webrtc) go library and whole [@pion](https://github.com/pion) team
 - series of streaming projects from [@deepch](https://github.com/deepch)
 - [rtsp-simple-server](https://github.com/aler9/rtsp-simple-server) idea from [@aler9](https://github.com/aler9)
-- [GStreamer](https://gstreamer.freedesktop.org/) multimedia framework pipeline idea
+- [GStreamer](https://gstreamer.freedesktop.org/) framework pipeline idea
 - [MediaSoup](https://mediasoup.org/) framework routing idea
 
 ## Codecs negotiation
@@ -45,7 +45,24 @@ streams:
 
 ![](codecs.svg)
 
-## Installation
+## Fast start
+
+1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+2. Open web interface [http://localhost:1984/](http://localhost:1984/)
+
+**Optionally:**
+
+- add your [streams](#module-streams) to [config](#configuration) file
+- setup [external access](#module-webrtc) to webrtc
+- setup [external access](#module-ngrok) to web interface
+- install [ffmpeg](#source-ffmpeg) for transcoding
+
+**Developers:**
+
+- write your own [web interface](#module-api)
+- integrate [web api](#module-api) into your smart home platform
+
+### go2rtc: Binary
 
 Download binary for your OS from [latest release](https://github.com/AlexxIT/go2rtc/releases/):
 
@@ -59,7 +76,34 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
 - `go2rtc_mac_amd64` - Mac with Intel
 - `go2rtc_mac_arm64` - Mac with M1
 
-Don't forget to fix the rights `chmod +x go2rtc_linux_xxx` on Linux and Mac.
+Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
+
+### go2rtc: Home Assistant Add-on
+
+[![](https://my.home-assistant.io/badges/supervisor_addon.svg)](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons)
+
+1. Install Add-On:
+    - Settings > Add-ons > Plus > Repositories > Add `https://github.com/AlexxIT/hassio-addons`
+    - go2rtc > Install > Start
+2. Setup [Integration](#module-hass)
+
+**Optionally:**
+
+- create `go2rtc.yaml` in your Home Assistant [config](https://www.home-assistant.io/docs/configuration) folder
+
+### go2rtc: Docker
+
+Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support `amd64`, `386`, `arm64`, `arm`. This container same as [Home Assistant Add-on](#go2rtc-home-assistant-add-on), but can be used separately from the Home Assistant. Container has preinstalled [FFmpeg](#source-ffmpeg) and [Ngrok](#module-ngrok) applications.
+
+```yaml
+services:
+  go2rtc:
+    image: alexxit/go2rtc
+    network_mode: host
+    restart: always
+    volumes:
+      - "~/go2rtc.yaml:/config/go2rtc.yaml"
+```
 
 ## Configuration
 
@@ -69,14 +113,14 @@ Create file `go2rtc.yaml` next to the app.
 - `api` server will start on default **1984 port**
 - `rtsp` server will start on default **8554 port**
 - `webrtc` will use random UDP port for each connection
-- `ffmpeg` will use default transcoding options (you need to install it [manually](https://ffmpeg.org/))
+- `ffmpeg` will use default transcoding options (you may install it [manually](https://ffmpeg.org/))
 
 Available modules:
 
 - [streams](#module-streams)
 - [api](#module-api) - HTTP API (important for WebRTC support)
 - [rtsp](#module-rtsp) - RTSP Server (important for FFmpeg support)
-- [webrtc](#module-webrtc) - WebRTC Server (important for external access)
+- [webrtc](#module-webrtc) - WebRTC Server
 - [ngrok](#module-ngrok) - Ngrok integration (external access for private network)
 - [ffmpeg](#source-ffmpeg) - FFmpeg integration
 - [hass](#module-hass) - Home Assistant integration
@@ -84,7 +128,7 @@ Available modules:
 
 ### Module: Streams
 
-**go2rtc** support different stream source types. You can config only one link as stream source or multiple.
+**go2rtc** support different stream source types. You can config one or multiple links of any type as stream source.
 
 Available source types:
 
@@ -94,12 +138,14 @@ Available source types:
 - [exec](#source-exec) - advanced FFmpeg and GStreamer integration
 - [hass](#source-hass) - Home Assistant integration
 
+**PS.** You can use sources like `MJPEG`, `HLS` and others via FFmpeg integration.
+
 #### Source: RTSP
 
 - Support **RTSP and RTSPS** links with multiple video and audio tracks
 - Support **2-way audio** ONLY for [ONVIF Profile T](https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf) cameras (back channel connection)
 
-**Attention:** proprietary 2-way audio standards are not supported!
+**Attention:** other 2-way audio standards are not supported! ONVIF without Profile T is not supported!
 
 ```yaml
 streams:
@@ -130,7 +176,7 @@ streams:
 
 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.
 
-Format: `ffmpeg:{input}#{params}`. Examples:
+Format: `ffmpeg:{input}#{param1}#{param2}#{param3}`. Examples:
 
 ```yaml
 streams:
@@ -141,7 +187,7 @@ streams:
   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
+  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
@@ -150,31 +196,18 @@ streams:
   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/av_stream/ch0#video=copy&audio=copy
+  rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=copy#audio=copy
 ```
 
-All trascoding formats has built-in templates. But you can override them via YAML config. You can also add your own formats to config and use them with source params.
+All trascoding formats has [built-in templates](https://github.com/AlexxIT/go2rtc/blob/master/cmd/ffmpeg/ffmpeg.go): `h264`, `h264/ultra`, `h264/high`, `h265`, `opus`, `pcmu`, `pcmu/16000`, `pcmu/48000`, `pcma`, `pcma/16000`, `pcma/48000`, `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.
 
 ```yaml
 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"
+  h264: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
+  mycodec: "-any args that support ffmpeg..."
 ```
 
 #### Source: Exec
@@ -202,7 +235,7 @@ streams:
 
 ### Module: API
 
-The HTTP API is the main part for interacting with the application.
+The HTTP API is the main part for interacting with the application. Default address: `http://127.0.0.1:1984/`.
 
 - you can use WebRTC only when HTTP API enabled
 - you can disable HTTP API with `listen: ""` and use, for example, only RTSP client/server protocol
@@ -212,11 +245,15 @@ The HTTP API is the main part for interacting with the application.
 
 ```yaml
 api:
-  listen: ":1984"    # HTTP API port ("" - disabled)
-  base_path: ""      # API prefix for serve on suburl
-  static_dir: "www"  # folder for static files ("" - disabled)
+  listen: ":1984"  # HTTP API port ("" - disabled)
+  base_path: ""    # API prefix for serve on suburl
+  static_dir: ""   # folder for static files (custom web interface)
 ```
 
+**PS. go2rtc** don't provide HTTPS or password protection. Use [Nginx](https://nginx.org/) or [Ngrok](#module-ngrok) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on) for this tasks.
+
+**PS2.** You can access microphone (for 2-way audio) only with HTTPS
+
 ### Module: RTSP
 
 You can get any stream as RTSP-stream with codecs filter:
@@ -236,9 +273,9 @@ rtsp:
 
 ### Module: WebRTC
 
-WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of internet do you have.
+WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of Internet do you have.
 
-- by default, WebRTC use two random UDP ports for each connection (for video and audio)
+- by default, WebRTC use two random UDP ports for each connection (video and audio)
 - you can enable one additional TCP port for all connections and use it for external access
 
 **Static public IP**
@@ -353,14 +390,24 @@ tunnels:
 
 ### Module: Hass
 
-go2rtc compatible with Home Assistant [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration API.
+**go2rtc** compatible with Home Assistant [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration.
 
-- add integration with link to go2rtc HTTP API:
-  - Hass > Settings > Integrations > Add Integration > RTSPtoWebRTC > `http://192.168.1.123:1984/`
-- add generic camera with RTSP link:
-  - Hass > Settings > Integrations > Add Integration > Generic Camera > `rtsp://...`
-- use Picture Entity or Picture Glance lovelace card
-- open full screen card - this is should be WebRTC stream
+If you install **go2rtc** as [Hass Add-on](#go2rtc-home-assistant-add-on) - you need to use localhost IP-address, example:
+
+- `http://127.0.0.1:1984/` to web interface
+- `rtsp://127.0.0.1:8554/camera1` to RTSP streams
+
+In other cases you need to use IP-address of server with **go2rtc** application.
+
+1. Add integration with link to go2rtc HTTP API:
+   - Hass > Settings > Integrations > Add Integration > [RTSPtoWebRTC](https://my.home-assistant.io/redirect/config_flow_start/?domain=rtsp_to_webrtc) > `http://127.0.0.1:1984/`
+2. Add generic camera with RTSP link:
+   - Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://...` or `rtmp://...`
+3. Use Picture Entity or Picture Glance lovelace card
+   - you can use either direct RTSP links to cameras or take RTSP streams from **go2rtc**
+4. Open full screen card - this is should be WebRTC stream
+
+PS. Default Home Assistant lovelace cards don't support 2-way audio. You can use 2-way audio from [Add-on Web UI](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons). But you need use HTTPS to access the microphone. This is a browser restriction and cannot be avoided.
 
 ### Module: Log
 
@@ -376,3 +423,29 @@ log:
   streams: error
   webrtc: fatal
 ```
+
+## Security
+
+By default `go2rtc` start Web interface on port `1984` and RTSP on port `8554`. Both 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:
+
+```yaml
+api:
+  listen: "127.0.0.1:1984" # localhost
+
+rtsp:
+  listen: "127.0.0.1:8554" # localhost
+
+webrtc:
+  listen: ":8555" # external TCP port
+```
+
+- local access to RTSP is not a problem for [FFmpeg](#source-ffmpeg) integration, because it runs locally on your server
+- local access to API is not a problem for [Home Assistant Add-on](#go2rtc-home-assistant-add-on), because Hass runs locally on same server and Add-on Web UI protected with Hass authorization ([Ingress feature](https://www.home-assistant.io/blog/2019/04/15/hassio-ingress/))
+- 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](https://nginx.org/), [Caddy](https://caddyserver.com/), [Ngrok](https://ngrok.com/), etc.
+
+PS. Additionally WebRTC opens a lot of random UDP ports for transmit encrypted media. They work without problems on the local network. And sometimes work for external access, even if you haven't opened ports on your router. But for stable external WebRTC access, you need to configure the TCP port.

build/hassio/Dockerfile

@@ -0,0 +1,24 @@
+ARG BUILD_FROM
+FROM $BUILD_FROM
+
+RUN apk add --no-cache git go ffmpeg
+
+ARG BUILD_ARCH
+
+RUN git clone https://github.com/AlexxIT/go2rtc \
+    && cd go2rtc \
+    && CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath -o /usr/local/bin
+
+# https://github.com/home-assistant/docker-base/blob/master/alpine/Dockerfile
+RUN if [ "${BUILD_ARCH}" = "aarch64" ]; then BUILD_ARCH="arm64"; \
+    elif [ "${BUILD_ARCH}" = "armv7" ]; then BUILD_ARCH="arm"; fi \
+    && cd go2rtc \
+    && curl $(curl -s "https://raw.githubusercontent.com/ngrok/docker-ngrok/main/releases.json" | jq -r ".${BUILD_ARCH}.url") -o ngrok.zip \
+    && unzip ngrok -d /usr/local/bin
+
+RUN rm -r /go2rtc
+
+COPY run.sh /
+RUN chmod a+x /run.sh
+
+CMD [ "/run.sh" ]

build/hassio/config.yaml

@@ -0,0 +1,6 @@
+# https://github.com/home-assistant/builder/blob/master/builder.sh
+name: go2rtc
+description: Ultimate camera streaming application
+url: https://github.com/AlexxIT/go2rtc
+image: alexxit/go2rtc
+arch: [ amd64, aarch64, i386, armv7 ]

build/hassio/run.sh

@@ -0,0 +1,14 @@
+#!/usr/bin/with-contenv bashio
+
+set +e
+
+# set cwd for go2rtc (for config file, Hass itegration, etc)
+cd /config
+
+# add the feature to override go2rtc binary from Hass config folder
+export PATH="/config:$PATH"
+
+while true; do
+    go2rtc
+    sleep 5
+done

cmd/api/api.go

@@ -37,14 +37,14 @@ func Init() {
 
 	HandleFunc("/api/frame.mp4", frameHandler)
 	HandleFunc("/api/frame.raw", frameHandler)
-	HandleFunc("/api/stack", stackHandler)
-	HandleFunc("/api/stats", statsHandler)
+	HandleFunc("/api/streams", streamsHandler)
 	HandleFunc("/api/ws", apiWS)
 
 	// ensure we can listen without errors
 	listener, err := net.Listen("tcp", cfg.Mod.Listen)
 	if err != nil {
 		log.Fatal().Err(err).Msg("[api] listen")
+		return
 	}
 
 	log.Info().Str("addr", cfg.Mod.Listen).Msg("[api] listen")
@@ -52,7 +52,7 @@ func Init() {
 	go func() {
 		s := http.Server{}
 		if err = s.Serve(listener); err != nil {
-			log.Fatal().Err(err).Msg("[api] Serve")
+			log.Fatal().Err(err).Msg("[api] serve")
 		}
 	}()
 }
@@ -69,16 +69,30 @@ var basePath string
 var log zerolog.Logger
 var wsHandlers = make(map[string]WSHandler)
 
-func statsHandler(w http.ResponseWriter, _ *http.Request) {
-	v := map[string]interface{}{
-		"streams": streams.All(),
+func streamsHandler(w http.ResponseWriter, r *http.Request) {
+	src := r.URL.Query().Get("src")
+
+	switch r.Method {
+	case "PUT":
+		streams.Get(src)
+		return
+	case "DELETE":
+		streams.Delete(src)
+		return
+	}
+
+	var v interface{}
+	if src != "" {
+		v = streams.Get(src)
+	} else {
+		v = streams.All()
 	}
 	data, err := json.Marshal(v)
 	if err != nil {
-		log.Error().Err(err).Msg("[api.stats] marshal")
+		log.Error().Err(err).Msg("[api.streams] marshal")
 	}
 	if _, err = w.Write(data); err != nil {
-		log.Error().Err(err).Msg("[api.stats] write")
+		log.Error().Err(err).Msg("[api.streams] write")
 	}
 }
 

cmd/api/keyframe.go

@@ -8,8 +8,8 @@ import (
 )
 
 func frameHandler(w http.ResponseWriter, r *http.Request) {
-	url := r.URL.Query().Get("url")
-	stream := streams.Get(url)
+	src := r.URL.Query().Get("src")
+	stream := streams.Get(src)
 	if stream == nil {
 		return
 	}

cmd/api/static.go

@@ -8,6 +8,7 @@ import (
 func initStatic(staticDir string) {
 	var root http.FileSystem
 	if staticDir != "" {
+		log.Info().Str("dir", staticDir).Msg("[api] serve static")
 		root = http.Dir(staticDir)
 	} else {
 		root = http.FS(www.Static)

cmd/app/app.go

@@ -1,6 +1,7 @@
 package app
 
 import (
+	"flag"
 	"github.com/rs/zerolog"
 	"gopkg.in/yaml.v3"
 	"io"
@@ -9,13 +10,25 @@ import (
 )
 
 func Init() {
-	data, _ = os.ReadFile("go2rtc.yaml")
+	config := flag.String(
+		"config",
+		"go2rtc.yaml",
+		"Path to go2rtc configuration file",
+	)
+
+	flag.Parse()
+
+	data, _ = os.ReadFile(*config)
 
 	var cfg struct {
 		Mod map[string]string `yaml:"log"`
 	}
 
-	LoadConfig(&cfg)
+	if data != nil {
+		if err := yaml.Unmarshal(data, &cfg); err != nil {
+			println("ERROR: " + err.Error())
+		}
+	}
 
 	var writer io.Writer = os.Stdout
 
@@ -39,7 +52,9 @@ func Init() {
 
 	modules = cfg.Mod
 
-	log.Info().Msgf("go2rtc %s/%s", runtime.GOOS, runtime.GOARCH)
+	path, _ := os.Getwd()
+	log.Debug().Str("os", runtime.GOOS).Str("arch", runtime.GOARCH).
+		Str("cwd", path).Int("conf_size", len(data)).Msgf("[app]")
 }
 
 func LoadConfig(v interface{}) {

cmd/debug/debug.go

@@ -0,0 +1,27 @@
+package debug
+
+import (
+	"github.com/AlexxIT/go2rtc/cmd/api"
+	"github.com/AlexxIT/go2rtc/cmd/streams"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"net/http"
+	"os"
+	"strconv"
+)
+
+func Init() {
+	api.HandleFunc("/api/stack", stackHandler)
+	api.HandleFunc("/api/exit", exitHandler)
+
+	streams.HandleFunc("null", nullHandler)
+}
+
+func exitHandler(_ http.ResponseWriter, r *http.Request) {
+	s := r.URL.Query().Get("code")
+	code, _ := strconv.Atoi(s)
+	os.Exit(code)
+}
+
+func nullHandler(string) (streamer.Producer, error) {
+	return nil, nil
+}

cmd/debug/stack.go

@@ -1,4 +1,4 @@
-package api
+package debug
 
 import (
 	"bytes"
@@ -8,19 +8,22 @@ import (
 )
 
 var stackSkip = [][]byte{
-	// debug.go
-	[]byte("github.com/AlexxIT/go2rtc/cmd/debug.handler"),
-
-	// cmd.go
-	[]byte("github.com/AlexxIT/go2rtc/cmd.Run"),
+	// main.go
+	[]byte("main.main()"),
 	[]byte("created by os/signal.Notify"),
 
-	// api.go
+	// api/stack.go
+	[]byte("github.com/AlexxIT/go2rtc/cmd/api.stackHandler"),
+
+	// api/api.go
 	[]byte("created by github.com/AlexxIT/go2rtc/cmd/api.Init"),
 	[]byte("created by net/http.(*connReader).startBackgroundRead"),
-	[]byte("created by net/http.(*Server).Serve"),
+	[]byte("created by net/http.(*Server).Serve"), // TODO: why two?
 
 	[]byte("created by github.com/AlexxIT/go2rtc/cmd/rtsp.Init"),
+
+	// webrtc/api.go
+	[]byte("created by github.com/pion/ice/v2.NewTCPMuxDefault"),
 }
 
 func stackHandler(w http.ResponseWriter, r *http.Request) {

cmd/exec/exec.go

@@ -70,7 +70,7 @@ func Handle(url string) (streamer.Producer, error) {
 	}
 
 	select {
-	case <-time.After(time.Second * 10):
+	case <-time.After(time.Second * 15):
 		_ = cmd.Process.Kill()
 		log.Error().Str("url", url).Msg("[exec] timeout")
 		return nil, errors.New("timeout")

cmd/ffmpeg/ffmpeg.go

@@ -54,7 +54,7 @@ func Init() {
 
 		var query url.Values
 		if i := strings.IndexByte(s, '#'); i > 0 {
-			query, _ = url.ParseQuery(s[i+1:])
+			query = parseQuery(s[i+1:])
 			s = s[:i]
 		}
 
@@ -110,3 +110,16 @@ func Init() {
 		return exec.Handle(s)
 	})
 }
+
+func parseQuery(s string) map[string][]string {
+	query := map[string][]string{}
+	for _, key := range strings.Split(s, "#") {
+		var value string
+		i := strings.IndexByte(key, '=')
+		if i > 0 {
+			key, value = key[:i], key[i+1:]
+		}
+		query[key] = append(query[key], value)
+	}
+	return query
+}

cmd/hass/hass.go

@@ -6,6 +6,7 @@ import (
 	"fmt"
 	"github.com/AlexxIT/go2rtc/cmd/api"
 	"github.com/AlexxIT/go2rtc/cmd/app"
+	"github.com/AlexxIT/go2rtc/cmd/rtsp"
 	"github.com/AlexxIT/go2rtc/cmd/streams"
 	"github.com/AlexxIT/go2rtc/cmd/webrtc"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
@@ -13,6 +14,7 @@ import (
 	"net/http"
 	"os"
 	"path"
+	"strings"
 )
 
 func Init() {
@@ -39,28 +41,45 @@ func Init() {
 		return
 	}
 
-	ent := new(entries)
-	if err = json.Unmarshal(data, ent); err != nil {
+	storage := new(entries)
+	if err = json.Unmarshal(data, storage); err != nil {
 		return
 	}
 
 	urls := map[string]string{}
 
-	for _, entrie := range ent.Data.Entries {
-		switch entrie.Domain {
-		case "generic":
-			if entrie.Options.StreamSource != "" {
-				urls[entrie.Title] = entrie.Options.StreamSource
-			}
-		}
-	}
-
 	streams.HandleFunc("hass", func(url string) (streamer.Producer, error) {
 		if hurl := urls[url[5:]]; hurl != "" {
 			return streams.GetProducer(hurl)
 		}
 		return nil, fmt.Errorf("can't get url: %s", url)
 	})
+
+	for _, entrie := range storage.Data.Entries {
+		switch entrie.Domain {
+		case "generic":
+			if entrie.Options.StreamSource == "" {
+				continue
+			}
+			urls[entrie.Title] = entrie.Options.StreamSource
+
+		//case "homekit_controller":
+		//	if entrie.Data.ClientID == "" {
+		//		continue
+		//	}
+		//	urls[entrie.Title] = fmt.Sprintf(
+		//		"homekit://%s:%d?client_id=%s&client_private=%s%s&device_id=%s&device_public=%s",
+		//		entrie.Data.DeviceHost, entrie.Data.DevicePort,
+		//		entrie.Data.ClientID, entrie.Data.ClientPrivate, entrie.Data.ClientPublic,
+		//		entrie.Data.DeviceID, entrie.Data.DevicePublic,
+		//	)
+
+		default:
+			continue
+		}
+
+		streams.Get("hass:" + entrie.Title)
+	}
 }
 
 var log zerolog.Logger
@@ -80,6 +99,15 @@ func handler(w http.ResponseWriter, r *http.Request) {
 		return
 	}
 
+	// TODO: fixme
+	if strings.HasPrefix(url, "rtsp://") {
+		port := ":" + rtsp.Port + "/"
+		i := strings.Index(url, port)
+		if i > 0 {
+			url = url[i+len(port):]
+		}
+	}
+
 	stream := streams.Get(url)
 	str, err = webrtc.ExchangeSDP(stream, string(offer), r.UserAgent())
 	if err != nil {

cmd/mse/mse.go

@@ -13,8 +13,8 @@ func Init() {
 }
 
 func handler(ctx *api.Context, msg *streamer.Message) {
-	url := ctx.Request.URL.Query().Get("url")
-	stream := streams.Get(url)
+	src := ctx.Request.URL.Query().Get("src")
+	stream := streams.Get(src)
 	if stream == nil {
 		return
 	}

cmd/rtsp/rtsp.go

@@ -65,8 +65,17 @@ func rtspHandler(url string) (streamer.Producer, error) {
 	if err = conn.Dial(); err != nil {
 		return nil, err
 	}
+
+	conn.Backchannel = true
 	if err = conn.Describe(); err != nil {
-		return nil, err
+		// second try without backchannel, we need to reconnect
+		if err = conn.Dial(); err != nil {
+			return nil, err
+		}
+		conn.Backchannel = false
+		if err = conn.Describe(); err != nil {
+			return nil, err
+		}
 	}
 
 	return conn, nil

cmd/streams/handlers.go

@@ -19,6 +19,9 @@ func HandleFunc(scheme string, handler Handler) {
 
 func HasProducer(url string) bool {
 	i := strings.IndexByte(url, ':')
+	if i <= 0 { // TODO: i < 4 ?
+		return false
+	}
 	return handlers[url[:i]] != nil
 }
 

cmd/streams/producer.go

@@ -2,6 +2,7 @@ package streams
 
 import (
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"sync"
 )
 
 type state byte
@@ -21,15 +22,19 @@ type Producer struct {
 	tracks  []*streamer.Track
 
 	state state
+	mx    sync.Mutex
 }
 
 func (p *Producer) GetMedias() []*streamer.Media {
+	p.mx.Lock()
+	defer p.mx.Unlock()
+
 	if p.state == stateNone {
 		log.Debug().Str("url", p.url).Msg("[streams] probe producer")
 
 		var err error
 		p.element, err = GetProducer(p.url)
-		if err != nil {
+		if err != nil || p.element == nil {
 			log.Error().Err(err).Str("url", p.url).Msg("[streams] probe producer")
 			return nil
 		}
@@ -41,6 +46,9 @@ func (p *Producer) GetMedias() []*streamer.Media {
 }
 
 func (p *Producer) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.Track {
+	p.mx.Lock()
+	defer p.mx.Unlock()
+
 	if p.state == stateMedias {
 		p.state = stateTracks
 	}
@@ -61,6 +69,9 @@ func (p *Producer) GetTrack(media *streamer.Media, codec *streamer.Codec) *strea
 // internals
 
 func (p *Producer) start() {
+	p.mx.Lock()
+	defer p.mx.Unlock()
+
 	if p.state != stateTracks {
 		return
 	}
@@ -72,10 +83,18 @@ func (p *Producer) start() {
 }
 
 func (p *Producer) stop() {
+	p.mx.Lock()
+
 	log.Debug().Str("url", p.url).Msg("[streams] stop producer")
 
-	_ = p.element.Stop()
-	p.element = nil
+	if p.element != nil {
+		_ = p.element.Stop()
+		p.element = nil
+	} else {
+		log.Warn().Str("url", p.url).Msg("[streams] stop empty producer")
+	}
 	p.tracks = nil
 	p.state = stateNone
+
+	p.mx.Unlock()
 }

cmd/streams/stream.go

@@ -2,6 +2,7 @@ package streams
 
 import (
 	"encoding/json"
+	"errors"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 )
 
@@ -61,6 +62,10 @@ func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
 
 					// Step 4. Get producer track
 					prodTrack := prod.GetTrack(prodMedia, prodCodec)
+					if prodTrack == nil {
+						log.Warn().Msg("[stream] can't get track")
+						continue
+					}
 
 					// Step 5. Add track to consumer and get new track
 					consTrack := consumer.element.AddTrack(consMedia, prodTrack)
@@ -74,7 +79,7 @@ func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
 
 	// can't match tracks for consumer
 	if len(consumer.tracks) == 0 {
-		return nil
+		return errors.New("couldn't find the matching tracks")
 	}
 
 	s.consumers = append(s.consumers, consumer)
@@ -121,7 +126,7 @@ func (s *Stream) RemoveProducer(prod streamer.Producer) {
 }
 
 func (s *Stream) Active() bool {
-	if len(s.consumers) > 0{
+	if len(s.consumers) > 0 {
 		return true
 	}
 

cmd/streams/stream_test.go

@@ -2,6 +2,7 @@ package streams
 
 import (
 	"github.com/AlexxIT/go2rtc/pkg/fake"
+	"github.com/AlexxIT/go2rtc/pkg/rtsp"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/stretchr/testify/assert"
 	"testing"
@@ -103,7 +104,7 @@ a=control:streamid=0
 
 func TestRouting(t *testing.T) {
 	prod := &fake.Producer{}
-	prod.Medias, _ = streamer.UnmarshalRTSPSDP([]byte(dahuaSimple))
+	prod.Medias, _ = rtsp.UnmarshalSDP([]byte(dahuaSimple))
 	assert.Len(t, prod.Medias, 3)
 
 	HandleFunc("fake", func(url string) (streamer.Producer, error) {

cmd/streams/streams.go

@@ -34,14 +34,19 @@ func Get(name string) *Stream {
 	return nil
 }
 
+func Delete(name string) {
+	delete(streams, name)
+}
+
 func All() map[string]interface{} {
-	active := map[string]interface{}{}
+	all := map[string]interface{}{}
 	for name, stream := range streams {
-		if stream.Active() {
-			active[name] = stream
-		}
+		all[name] = stream
+		//if stream.Active() {
+		//	all[name] = stream
+		//}
 	}
-	return active
+	return all
 }
 
 var log zerolog.Logger

cmd/webrtc/candidates.go

@@ -0,0 +1,75 @@
+package webrtc
+
+import (
+	"github.com/AlexxIT/go2rtc/cmd/api"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/AlexxIT/go2rtc/pkg/webrtc"
+	"github.com/pion/sdp/v3"
+	"strings"
+)
+
+var candidates []string
+
+func AddCandidate(address string) {
+	candidates = append(candidates, address)
+}
+
+func addCanditates(answer string) (string, error) {
+	if len(candidates) == 0 {
+		return answer, nil
+	}
+
+	sd := &sdp.SessionDescription{}
+	if err := sd.Unmarshal([]byte(answer)); err != nil {
+		return "", err
+	}
+
+	md := sd.MediaDescriptions[0]
+
+	_, end := md.Attribute("end-of-candidates")
+	if end {
+		md.Attributes = md.Attributes[:len(md.Attributes)-1]
+	}
+
+	for _, address := range candidates {
+		if strings.HasPrefix(address, "stun:") {
+			ip, err := webrtc.GetPublicIP()
+			if err != nil {
+				log.Warn().Err(err).Msg("[webrtc] public IP")
+				continue
+			}
+			address = ip.String() + address[4:]
+
+			log.Debug().Str("addr", address).Msg("[webrtc] stun public address")
+		}
+
+		cand, err := webrtc.NewCandidate(address)
+		if err != nil {
+			log.Warn().Err(err).Msg("[webrtc] candidate")
+			continue
+		}
+
+		md.WithPropertyAttribute(cand)
+	}
+
+	if end {
+		md.WithPropertyAttribute("end-of-candidates")
+	}
+
+	data, err := sd.Marshal()
+	if err != nil {
+		return "", err
+	}
+
+	return string(data), nil
+}
+
+func candidateHandler(ctx *api.Context, msg *streamer.Message) {
+	if ctx.Consumer == nil {
+		return
+	}
+	if conn := ctx.Consumer.(*webrtc.Conn); conn != nil {
+		log.Trace().Str("candidate", msg.Value.(string)).Msg("[webrtc] remote")
+		conn.Push(msg)
+	}
+}

cmd/webrtc/webrtc.go

@@ -8,10 +8,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	pion "github.com/pion/webrtc/v3"
 	"github.com/rs/zerolog"
-	"io/ioutil"
 	"net"
-	"net/http"
-	"strings"
 )
 
 func Init() {
@@ -56,120 +53,23 @@ func Init() {
 
 	candidates = cfg.Mod.Candidates
 
-	api.HandleFunc("/api/webrtc", apiHandler)
-	api.HandleFunc("/api/webrtc/camera", cameraHandler)
 	api.HandleWS(webrtc.MsgTypeOffer, offerHandler)
 	api.HandleWS(webrtc.MsgTypeCandidate, candidateHandler)
 }
 
-func AddCandidate(address string) {
-	candidates = append(candidates, address)
-}
-
 var Port string
 var log zerolog.Logger
-var candidates []string
 
 var NewPConn func() (*pion.PeerConnection, error)
 
-func apiHandler(w http.ResponseWriter, r *http.Request) {
-	url := r.URL.Query().Get("url")
-	stream := streams.Get(url)
-	if stream == nil {
-		return
-	}
-
-	// get offer
-	offer, err := ioutil.ReadAll(r.Body)
-	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] read offer")
-		return
-	}
-
-	// create new webrtc instance
-	cons := new(webrtc.Conn)
-	cons.Conn, err = NewPConn()
-	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] new conn")
-		return
-	}
-
-	cons.UserAgent = r.UserAgent()
-	cons.Listen(func(msg interface{}) {
-		if msg == streamer.StateNull {
-			stream.RemoveConsumer(cons)
-		}
-	})
-
-	if err = stream.AddConsumer(cons); err != nil {
-		log.Warn().Err(err).Msg("[api.webrtc] add consumer")
-		return
-	}
-
-	cons.Init()
-
-	// exchange sdp with waiting all candidates
-	answer, err := cons.ExchangeSDP(string(offer), true)
-
-	// send SDP to client
-	if _, err = w.Write([]byte(answer)); err != nil {
-		log.Error().Err(err).Msg("[api.webrtc] send answer")
-	}
-}
-
-func cameraHandler(w http.ResponseWriter, r *http.Request) {
-	url := r.URL.Query().Get("url")
-	stream := streams.Get(url)
-	if stream == nil {
-		return
-	}
-
-	// get offer
-	offer, err := ioutil.ReadAll(r.Body)
-	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] read offer")
-		return
-	}
-
-	// create new webrtc instance
-	conn := new(webrtc.Conn)
-	conn.Conn, err = NewPConn()
-	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] new conn")
-		return
-	}
-
-	conn.UserAgent = r.UserAgent()
-	conn.Listen(func(msg interface{}) {
-		switch msg.(type) {
-		case pion.PeerConnectionState:
-			if msg == pion.PeerConnectionStateDisconnected {
-				stream.RemoveConsumer(conn)
-			}
-		case streamer.Track:
-			//stream.AddProducer(conn)
-		}
-	})
-
-	conn.Init()
-
-	// exchange sdp with waiting all candidates
-	answer, err := conn.ExchangeSDP(string(offer), true)
-
-	// send SDP to client
-	if _, err = w.Write([]byte(answer)); err != nil {
-		log.Error().Err(err).Msg("[api.webrtc] send answer")
-	}
-}
-
 func offerHandler(ctx *api.Context, msg *streamer.Message) {
-	name := ctx.Request.URL.Query().Get("url")
-	stream := streams.Get(name)
+	src := ctx.Request.URL.Query().Get("src")
+	stream := streams.Get(src)
 	if stream == nil {
 		return
 	}
 
-	log.Debug().Str("stream", name).Msg("[webrtc] new consumer")
+	log.Debug().Str("src", src).Msg("[webrtc] new consumer")
 
 	var err error
 
@@ -208,6 +108,7 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
 	// 2. AddConsumer, so we get new tracks
 	if err = stream.AddConsumer(conn); err != nil {
 		log.Warn().Err(err).Msg("[api.webrtc] add consumer")
+		_ = conn.Conn.Close()
 		ctx.Error(err)
 		return
 	}
@@ -216,7 +117,11 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
 
 	// exchange sdp without waiting all candidates
 	//answer, err := conn.ExchangeSDP(offer, false)
-	answer, err := conn.GetAnswer()
+	//answer, err := conn.GetAnswer()
+	answer, err := conn.GetCompleteAnswer()
+	if err == nil {
+		answer, err = addCanditates(answer)
+	}
 	log.Trace().Msgf("[webrtc] answer\n%s", answer)
 
 	if err != nil {
@@ -229,29 +134,6 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
 		Type: webrtc.MsgTypeAnswer, Value: answer,
 	})
 
-	for _, address := range candidates {
-		if strings.HasPrefix(address, "stun:") {
-			ip, err := webrtc.GetPublicIP()
-			if err != nil {
-				log.Warn().Err(err).Msg("[webrtc] public IP")
-				continue
-			}
-			address = ip.String() + address[4:]
-
-			log.Debug().Str("addr", address).Msg("[webrtc] stun public address")
-		}
-
-		cand, err := webrtc.NewCandidate(address)
-		if err != nil {
-			log.Warn().Err(err).Msg("[webrtc] candidate")
-			continue
-		}
-
-		conn.Fire(&streamer.Message{
-			Type: webrtc.MsgTypeCandidate, Value: cand,
-		})
-	}
-
 	ctx.Consumer = conn
 }
 
@@ -287,6 +169,7 @@ func ExchangeSDP(
 	// 2. AddConsumer, so we get new tracks
 	if err = stream.AddConsumer(conn); err != nil {
 		log.Warn().Err(err).Msg("[api.webrtc] add consumer")
+		_ = conn.Conn.Close()
 		return
 	}
 
@@ -295,6 +178,9 @@ func ExchangeSDP(
 	// exchange sdp without waiting all candidates
 	//answer, err := conn.ExchangeSDP(offer, false)
 	answer, err = conn.GetCompleteAnswer()
+	if err == nil {
+		answer, err = addCanditates(answer)
+	}
 	log.Trace().Msgf("[webrtc] answer\n%s", answer)
 
 	if err != nil {
@@ -303,13 +189,3 @@ func ExchangeSDP(
 
 	return
 }
-
-func candidateHandler(ctx *api.Context, msg *streamer.Message) {
-	if ctx.Consumer == nil {
-		return
-	}
-	if conn := ctx.Consumer.(*webrtc.Conn); conn != nil {
-		log.Trace().Str("candidate", msg.Value.(string)).Msg("[webrtc] remote")
-		conn.Push(msg)
-	}
-}

main.go

@@ -3,6 +3,7 @@ package main
 import (
 	"github.com/AlexxIT/go2rtc/cmd/api"
 	"github.com/AlexxIT/go2rtc/cmd/app"
+	"github.com/AlexxIT/go2rtc/cmd/debug"
 	"github.com/AlexxIT/go2rtc/cmd/exec"
 	"github.com/AlexxIT/go2rtc/cmd/ffmpeg"
 	"github.com/AlexxIT/go2rtc/cmd/hass"
@@ -33,6 +34,7 @@ func main() {
 	mse.Init()
 
 	ngrok.Init()
+	debug.Init()
 
 	sigs := make(chan os.Signal, 1)
 	signal.Notify(sigs, syscall.SIGINT, syscall.SIGTERM)

pkg/rtmp/client.go

@@ -3,9 +3,12 @@ package rtmp
 import (
 	"encoding/base64"
 	"encoding/binary"
+	"encoding/hex"
+	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/deepch/vdk/av"
+	"github.com/deepch/vdk/codec/aacparser"
 	"github.com/deepch/vdk/codec/h264parser"
 	"github.com/deepch/vdk/format/rtmp"
 	"github.com/pion/rtp"
@@ -70,9 +73,36 @@ func (c *Client) Dial() (err error) {
 			c.tracks = append(c.tracks, track)
 
 		case av.AAC:
-			panic("not implemented")
+			// TODO: fix support
+			cd := stream.(aacparser.CodecData)
+
+			// a=fmtp:97 streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1588
+			fmtp := fmt.Sprintf(
+				"config=%s",
+				hex.EncodeToString(cd.ConfigBytes),
+			)
+
+			codec := &streamer.Codec{
+				Name:      streamer.CodecAAC,
+				ClockRate: uint32(cd.Config.SampleRate),
+				Channels:  uint16(cd.Config.ChannelConfig),
+				FmtpLine:  fmtp,
+			}
+
+			media := &streamer.Media{
+				Kind:      streamer.KindAudio,
+				Direction: streamer.DirectionSendonly,
+				Codecs:    []*streamer.Codec{codec},
+			}
+			c.medias = append(c.medias, media)
+
+			track := &streamer.Track{
+				Codec: codec, Direction: media.Direction,
+			}
+			c.tracks = append(c.tracks, track)
+
 		default:
-			panic("unsupported codec")
+			fmt.Printf("[rtmp] unsupported codec %+v\n", stream)
 		}
 	}
 

pkg/rtsp/conn.go

@@ -43,11 +43,15 @@ const (
 	ModeServerConsumer
 )
 
+const KeepAlive = time.Second * 25
+
 type Conn struct {
 	streamer.Element
 
 	// public
 
+	Backchannel bool
+
 	Medias    []*streamer.Media
 	Session   string
 	UserAgent string
@@ -104,6 +108,9 @@ func (c *Conn) Dial() (err error) {
 	//if c.state != StateClientInit {
 	//	panic("wrong state")
 	//}
+	if c.conn != nil && c.auth != nil {
+		c.auth.Reset()
+	}
 
 	c.conn, err = net.DialTimeout(
 		"tcp", c.URL.Host, 10*time.Second,
@@ -144,7 +151,9 @@ func (c *Conn) Request(req *tcp.Request) error {
 	}
 
 	c.sequence++
-	req.Header.Set("CSeq", strconv.Itoa(c.sequence))
+	// important to send case sensitive CSeq
+	// https://github.com/AlexxIT/go2rtc/issues/7
+	req.Header["CSeq"] = []string{strconv.Itoa(c.sequence)}
 
 	c.auth.Write(req)
 
@@ -189,7 +198,7 @@ func (c *Conn) Do(req *tcp.Request) (*tcp.Response, error) {
 	}
 
 	if res.StatusCode != http.StatusOK {
-		return nil, fmt.Errorf("wrong response on %s", req.Method)
+		return res, fmt.Errorf("wrong response on %s", req.Method)
 	}
 
 	return res, nil
@@ -254,23 +263,27 @@ func (c *Conn) Describe() error {
 		Method: MethodDescribe,
 		URL:    c.URL,
 		Header: map[string][]string{
-			"Accept":  {"application/sdp"},
-			"Require": {"www.onvif.org/ver20/backchannel"},
+			"Accept": {"application/sdp"},
 		},
 	}
 
+	if c.Backchannel {
+		req.Header.Set("Require", "www.onvif.org/ver20/backchannel")
+	}
+
 	res, err := c.Do(req)
 	if err != nil {
 		return err
 	}
 
-	// fix bug in Sonoff camera SDP "o=- 1 1 IN IP4 rom t_rtsplin"
-	// TODO: make some universal fix
-	if i := bytes.Index(res.Body, []byte("rom t_rtsplin")); i > 0 {
-		res.Body[i+3] = '_'
+	if val := res.Header.Get("Content-Base"); val != "" {
+		c.URL, err = url.Parse(val)
+		if err != nil {
+			return err
+		}
 	}
 
-	c.Medias, err = streamer.UnmarshalRTSPSDP(res.Body)
+	c.Medias, err = UnmarshalSDP(res.Body)
 	if err != nil {
 		return err
 	}
@@ -355,10 +368,23 @@ func (c *Conn) SetupMedia(
 	// we send our `interleaved`, but camera can answer with another
 
 	// Transport: RTP/AVP/TCP;unicast;interleaved=10-11;ssrc=10117CB7
+	// Transport: RTP/AVP/TCP;unicast;destination=192.168.1.123;source=192.168.10.12;interleaved=0
+	// Transport: RTP/AVP/TCP;ssrc=22345682;interleaved=0-1
 	s := res.Header.Get("Transport")
-	s, ok1, ok2 := between(s, "RTP/AVP/TCP;unicast;interleaved=", "-")
-	if !ok1 || !ok2 {
-		panic("wrong response")
+	// TODO: rewrite
+	if !strings.HasPrefix(s, "RTP/AVP/TCP;") {
+		return nil, fmt.Errorf("wrong transport: %s", s)
+	}
+
+	i := strings.Index(s, "interleaved=")
+	if i < 0 {
+		return nil, fmt.Errorf("wrong transport: %s", s)
+	}
+
+	s = s[i+len("interleaved="):]
+	i = strings.IndexAny(s, "-;")
+	if i > 0 {
+		s = s[:i]
 	}
 
 	ch, err = strconv.Atoi(s)
@@ -425,15 +451,17 @@ func (c *Conn) Accept() error {
 			return err
 		}
 
+		if c.URL == nil {
+			c.URL = req.URL
+			c.UserAgent = req.Header.Get("User-Agent")
+		}
+
 		c.Fire(req)
 
 		// Receiver: OPTIONS > DESCRIBE > SETUP... > PLAY > TEARDOWN
 		// Sender: OPTIONS > ANNOUNCE > SETUP... > RECORD > TEARDOWN
 		switch req.Method {
 		case MethodOptions:
-			c.URL = req.URL
-			c.UserAgent = req.Header.Get("User-Agent")
-
 			res := &tcp.Response{
 				Header: map[string][]string{
 					"Public": {"OPTIONS, SETUP, TEARDOWN, DESCRIBE, PLAY, PAUSE, ANNOUNCE, RECORD"},
@@ -449,7 +477,7 @@ func (c *Conn) Accept() error {
 				return errors.New("wrong content type")
 			}
 
-			c.Medias, err = streamer.UnmarshalRTSPSDP(req.Body)
+			c.Medias, err = UnmarshalSDP(req.Body)
 			if err != nil {
 				return err
 			}
@@ -549,6 +577,7 @@ func (c *Conn) Handle() (err error) {
 	}()
 
 	//c.Fire(streamer.StatePlaying)
+	ts := time.Now().Add(KeepAlive)
 
 	for {
 		// we can read:
@@ -603,7 +632,7 @@ func (c *Conn) Handle() (err error) {
 		if channelID&1 == 0 {
 			packet := &rtp.Packet{}
 			if err = packet.Unmarshal(buf); err != nil {
-				return errors.New("wrong RTP data")
+				return
 			}
 
 			track := c.channels[channelID]
@@ -611,22 +640,34 @@ func (c *Conn) Handle() (err error) {
 				_ = track.WriteRTP(packet)
 				//return fmt.Errorf("wrong channelID: %d", channelID)
 			} else {
-				panic("wrong channelID")
+				continue // TODO: maybe fix this
+				//panic("wrong channelID")
 			}
 		} else {
 			msg := &RTCP{Channel: channelID}
 
 			if err = msg.Header.Unmarshal(buf); err != nil {
-				return errors.New("wrong RTCP data")
+				return
 			}
 
 			msg.Packets, err = rtcp.Unmarshal(buf)
 			if err != nil {
-				return errors.New("wrong RTCP data")
+				return
 			}
 
 			c.Fire(msg)
 		}
+
+		// keep-alive
+		now := time.Now()
+		if now.After(ts) {
+			req := &tcp.Request{Method: MethodOptions, URL: c.URL}
+			// don't need to wait respose on this request
+			if err = c.Request(req); err != nil {
+				return err
+			}
+			ts = now.Add(KeepAlive)
+		}
 	}
 }
 
@@ -686,17 +727,35 @@ type RTCP struct {
 	Packets []rtcp.Packet
 }
 
-func between(s, sub1, sub2 string) (res string, ok1 bool, ok2 bool) {
-	i := strings.Index(s, sub1)
-	if i >= 0 {
-		ok1 = true
-		s = s[i+len(sub1):]
+const sdpHeader = `v=0
+o=- 0 0 IN IP4 0.0.0.0
+s=-
+t=0 0`
+
+func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
+	medias, err := streamer.UnmarshalSDP(rawSDP)
+	if err != nil {
+		// fix SDP header for some cameras
+		i := bytes.Index(rawSDP, []byte("\nm="))
+		if i > 0 {
+			rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
+			medias, err = streamer.UnmarshalSDP(rawSDP)
+		}
+		if err != nil {
+			return nil, err
+		}
 	}
 
-	i = strings.Index(s, sub2)
-	if i >= 0 {
-		return s[:i], ok1, true
+	// fix bug in ONVIF spec
+	// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
+	for _, media := range medias {
+		switch media.Direction {
+		case streamer.DirectionRecvonly, "":
+			media.Direction = streamer.DirectionSendonly
+		case streamer.DirectionSendonly:
+			media.Direction = streamer.DirectionRecvonly
+		}
 	}
 
-	return s, ok1, false
+	return medias, nil
 }

pkg/streamer/media.go

@@ -180,26 +180,6 @@ func UnmarshalSDP(rawSDP []byte) ([]*Media, error) {
 	return medias, nil
 }
 
-func UnmarshalRTSPSDP(rawSDP []byte) ([]*Media, error) {
-	medias, err := UnmarshalSDP(rawSDP)
-	if err != nil {
-		return nil, err
-	}
-
-	// fix bug in ONVIF spec
-	// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
-	for _, media := range medias {
-		switch media.Direction {
-		case DirectionRecvonly, "":
-			media.Direction = DirectionSendonly
-		case DirectionSendonly:
-			media.Direction = DirectionRecvonly
-		}
-	}
-
-	return medias, nil
-}
-
 func MarshalSDP(medias []*Media) ([]byte, error) {
 	sd := &sdp.SessionDescription{}
 

pkg/streamer/track.go

@@ -3,6 +3,7 @@ package streamer
 import (
 	"fmt"
 	"github.com/pion/rtp"
+	"sync"
 )
 
 type WriterFunc func(packet *rtp.Packet) error
@@ -12,6 +13,7 @@ type Track struct {
 	Codec     *Codec
 	Direction string
 	Sink      map[*Track]WriterFunc
+	mx        sync.Mutex
 }
 
 func (t *Track) String() string {
@@ -21,9 +23,11 @@ func (t *Track) String() string {
 }
 
 func (t *Track) WriteRTP(p *rtp.Packet) error {
+	t.mx.Lock()
 	for _, f := range t.Sink {
 		_ = f(p)
 	}
+	t.mx.Unlock()
 	return nil
 }
 
@@ -35,10 +39,14 @@ func (t *Track) Bind(w WriterFunc) *Track {
 	clone := &Track{
 		Codec: t.Codec, Direction: t.Direction, Sink: t.Sink,
 	}
+	t.mx.Lock()
 	t.Sink[clone] = w
+	t.mx.Unlock()
 	return clone
 }
 
 func (t *Track) Unbind() {
+	t.mx.Lock()
 	delete(t.Sink, t)
+	t.mx.Unlock()
 }

pkg/tcp/auth.go

@@ -80,6 +80,12 @@ func (a *Auth) Write(req *Request) {
 	}
 }
 
+func (a *Auth) Reset() {
+	if a.Method == AuthDigest {
+		a.Method = AuthUnknown
+	}
+}
+
 func Between(s, sub1, sub2 string) string {
 	i := strings.Index(s, sub1)
 	if i < 0 {

pkg/tcp/textproto.go

@@ -47,10 +47,13 @@ func ReadResponse(r *bufio.Reader) (*Response, error) {
 	if err != nil {
 		return nil, err
 	}
+	if line == "" {
+		return nil, errors.New("empty response on RTSP request")
+	}
 
 	ss := strings.SplitN(line, " ", 3)
 	if len(ss) != 3 {
-		return nil, errors.New("malformed response")
+		return nil, fmt.Errorf("malformed response: %s", line)
 	}
 
 	res := &Response{

pkg/webrtc/api.go

@@ -1,6 +1,7 @@
 package webrtc
 
 import (
+	"github.com/pion/ice/v2"
 	"github.com/pion/interceptor"
 	"github.com/pion/webrtc/v3"
 	"net"
@@ -21,31 +22,30 @@ func NewAPI(address string) (*webrtc.API, error) {
 		return nil, err
 	}
 
-	if address == "" {
-		return webrtc.NewAPI(
-			webrtc.WithMediaEngine(m),
-			webrtc.WithInterceptorRegistry(i),
-		), nil
-	}
-
-	ln, err := net.Listen("tcp", address)
-	if err != nil {
-		return webrtc.NewAPI(
-			webrtc.WithMediaEngine(m),
-			webrtc.WithInterceptorRegistry(i),
-		), err
-	}
-
 	s := webrtc.SettingEngine{
 		//LoggerFactory: customLoggerFactory{},
 	}
-	s.SetNetworkTypes([]webrtc.NetworkType{
-		webrtc.NetworkTypeUDP4, webrtc.NetworkTypeUDP6,
-		webrtc.NetworkTypeTCP4, webrtc.NetworkTypeTCP6,
+
+	// disable listen on Hassio docker interfaces
+	s.SetInterfaceFilter(func(name string) bool {
+		return name != "hassio" && name != "docker0"
 	})
 
-	tcpMux := webrtc.NewICETCPMux(nil, ln, 8)
-	s.SetICETCPMux(tcpMux)
+	// disable mDNS listener
+	s.SetICEMulticastDNSMode(ice.MulticastDNSModeDisabled)
+
+	if address != "" {
+		ln, err := net.Listen("tcp", address)
+		if err == nil {
+			s.SetNetworkTypes([]webrtc.NetworkType{
+				webrtc.NetworkTypeUDP4, webrtc.NetworkTypeUDP6,
+				webrtc.NetworkTypeTCP4, webrtc.NetworkTypeTCP6,
+			})
+
+			tcpMux := webrtc.NewICETCPMux(nil, ln, 8)
+			s.SetICETCPMux(tcpMux)
+		}
+	}
 
 	return webrtc.NewAPI(
 		webrtc.WithMediaEngine(m),

pkg/webrtc/conn.go

@@ -1,6 +1,7 @@
 package webrtc
 
 import (
+	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/webrtc/v3"
 )
@@ -57,7 +58,8 @@ func (c *Conn) Init() {
 			}
 		}
 
-		panic("something wrong")
+		fmt.Printf("TODO: webrtc ontrack %+v\n", remote)
+		fmt.Printf("TODO: webrtc ontrack %#v\n", remote)
 	})
 
 	c.Conn.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
@@ -75,76 +77,6 @@ func (c *Conn) Init() {
 	})
 }
 
-func (c *Conn) ExchangeSDP(offer string, complete bool) (answer string, err error) {
-	sdOffer := webrtc.SessionDescription{
-		Type: webrtc.SDPTypeOffer, SDP: offer,
-	}
-	if err = c.Conn.SetRemoteDescription(sdOffer); err != nil {
-		return
-	}
-
-	//for _, tr := range c.Conn.GetTransceivers() {
-	//	switch tr.Direction() {
-	//	case webrtc.RTPTransceiverDirectionSendonly:
-	//		// disable transceivers if we don't have track
-	//		// make direction=inactive
-	//		// don't really necessary, but anyway
-	//		if tr.Sender() == nil {
-	//			if err = tr.Stop(); err != nil {
-	//				return
-	//			}
-	//		}
-	//	case webrtc.RTPTransceiverDirectionRecvonly:
-	//		// TODO: change codecs list
-	//		caps := webrtc.RTPCodecCapability{
-	//			MimeType:  webrtc.MimeTypePCMU,
-	//			ClockRate: 8000,
-	//		}
-	//		codecs := []webrtc.RTPCodecParameters{
-	//			{RTPCodecCapability: caps},
-	//		}
-	//		if err = tr.SetCodecPreferences(codecs); err != nil {
-	//			return
-	//		}
-	//	}
-	//}
-
-	var sdAnswer webrtc.SessionDescription
-	sdAnswer, err = c.Conn.CreateAnswer(nil)
-	if err != nil {
-		return
-	}
-
-	//var sd *sdp.SessionDescription
-	//sd, err = sdAnswer.Unmarshal()
-	//for _, media := range sd.MediaDescriptions {
-	//	if media.MediaName.Media != "audio" {
-	//		continue
-	//	}
-	//	for i, attr := range media.Attributes {
-	//		if attr.Key == "sendonly" {
-	//			attr.Key = "inactive"
-	//			media.Attributes[i] = attr
-	//			break
-	//		}
-	//	}
-	//}
-	//var b []byte
-	//b, err = sd.Marshal()
-	//sdAnswer.SDP = string(b)
-
-	if err = c.Conn.SetLocalDescription(sdAnswer); err != nil {
-		return
-	}
-
-	if complete {
-		<-webrtc.GatheringCompletePromise(c.Conn)
-		return c.Conn.LocalDescription().SDP, nil
-	}
-
-	return sdAnswer.SDP, nil
-}
-
 func (c *Conn) SetOffer(offer string) (err error) {
 	sdOffer := webrtc.SessionDescription{
 		Type: webrtc.SDPTypeOffer, SDP: offer,
@@ -198,7 +130,9 @@ func (c *Conn) GetCompleteAnswer() (answer string, err error) {
 func (c *Conn) remote() string {
 	for _, trans := range c.Conn.GetTransceivers() {
 		pair, _ := trans.Receiver().Transport().ICETransport().GetSelectedCandidatePair()
-		return pair.Remote.String()
+		if pair.Remote != nil {
+			return pair.Remote.String()
+		}
 	}
 	return ""
 }

pkg/webrtc/helper.go

@@ -47,6 +47,9 @@ func GetPublicIP() (net.IP, error) {
 	if err = c.Do(message, func(e stun.Event) { res = e }); err != nil {
 		return nil, err
 	}
+	if err = c.Close(); err != nil {
+		return nil, err
+	}
 
 	if res.Error != nil {
 		return nil, res.Error

scripts/README.md

@@ -2,11 +2,18 @@
 
 - UPX-3.96 pack broken bin for `linux_mipsel`
 - UPX-3.95 pack broken bin for `mac_amd64`
+- UPX windows pack is recognised by anti-viruses as malicious
 - `aarch64` = `arm64`
 - `armv7` = `arm`
 
+## Virus
+
+- https://go.dev/doc/faq#virus
+- https://groups.google.com/g/golang-nuts/c/lPwiWYaApSU
+
 ## Useful links
 
 - https://github.com/golang/go/wiki/GoArm
 - https://gist.github.com/asukakenji/f15ba7e588ac42795f421b48b8aede63
 - https://en.wikipedia.org/wiki/AArch64
+- https://stackoverflow.com/questions/22267189/what-does-the-w-flag-mean-when-passed-in-via-the-ldflags-option-to-the-go-comman

scripts/build.cmd

@@ -2,13 +2,13 @@
 
 @SET GOOS=windows
 @SET GOARCH=amd64
-@SET FILENAME=go2rtc_win64.exe
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+@SET FILENAME=go2rtc_win64.zip
+go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
 
 @SET GOOS=windows
 @SET GOARCH=386
-@SET FILENAME=go2rtc_win32.exe
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+@SET FILENAME=go2rtc_win32.zip
+go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
 
 @SET GOOS=linux
 @SET GOARCH=amd64

scripts/build_linux_amd64.cmd

@@ -0,0 +1,4 @@
+@SET GOOS=linux
+@SET GOARCH=amd64
+cd ..
+go build -ldflags "-s -w" -trimpath && upx-3.96 go2rtc

scripts/build_linux_mips.cmd

@@ -0,0 +1,5 @@
+@ECHO OFF
+@SET GOOS=linux
+@SET GOARCH=mipsle
+cd ..
+go build -ldflags "-s -w" -trimpath && upx-3.95 go2rtc

scripts/build_win64.cmd

@@ -0,0 +1,4 @@
+@SET GOOS=windows
+@SET GOARCH=amd64
+cd ..
+go build -ldflags "-w -s" -trimpath

www/README.md

@@ -46,4 +46,8 @@ pc.ontrack = ev => {
 
     video.srcObject = ev.streams[0];
 }
-```
+```
+
+## Useful links
+
+- https://divtable.com/table-styler/

www/index.html

@@ -1,44 +1,106 @@
+<!DOCTYPE html>
 <html>
 <head>
-    <meta charset="UTF-8">
-    <meta name="viewport"
-          content="width=device-width, user-scalable=no, initial-scale=1.0, maximum-scale=1.0, minimum-scale=1.0">
+    <meta charset="utf-8">
+    <meta name="viewport" content="width=device-width, user-scalable=yes, initial-scale=1, maximum-scale=1">
     <meta http-equiv="X-UA-Compatible" content="ie=edge">
+
     <title>go2rtc</title>
+
+    <style>
+        table {
+            background-color: white;
+            text-align: left;
+            border-collapse: collapse;
+        }
+
+        table td, table th {
+            border: 1px solid black;
+            padding: 5px 5px;
+        }
+
+        table tbody td {
+            font-size: 13px;
+        }
+
+        table thead {
+            background: #CFCFCF;
+            background: linear-gradient(to bottom, #dbdbdb 0%, #d3d3d3 66%, #CFCFCF 100%);
+            border-bottom: 3px solid black;
+        }
+
+        table thead th {
+            font-size: 15px;
+            font-weight: bold;
+            color: black;
+            text-align: center;
+        }
+
+        .header {
+            padding: 5px 5px;
+        }
+    </style>
 </head>
 <body>
-<div id="header"></div>
-<table id="items"></table>
+<div class="header">
+    <input id="src" type="text" placeholder="url">
+    <a id="add" href="#">add</a>
+</div>
+<table id="streams">
+    <thead>
+    <tr>
+        <th>Name</th>
+        <th>Online</th>
+        <th>Commands</th>
+    </tr>
+    </thead>
+    <tbody>
+    </tbody>
+</table>
 <script>
     const baseUrl = location.origin + location.pathname.substr(
         0, location.pathname.lastIndexOf("/")
     );
 
-    const header = document.getElementById('header');
-    header.innerHTML = `<a href="api/stats">stats</a>`;
-
     const links = [
-        '<a href="webrtc-async.html?url={name}">webrtc-async</a>',
-        // '<a href="webrtc-sync.html?url={name}">webrtc-sync</a>',
-        '<a href="api/frame.mp4?url={name}">frame.mp4</a>',
-        '<a href="api/frame.raw?url={name}">frame.raw</a>',
-        '<a href="mse.html?url={name}">mse</a>',
+        '<a href="webrtc.html?src={name}">webrtc</a>',
+        '<a href="mse.html?src={name}">mse</a>',
+        '<a href="api/frame.mp4?src={name}">frame.mp4</a>',
+        '<a href="api/streams?src={name}">info</a>',
     ];
 
-    fetch(`${baseUrl}/api/stats`).then(r => {
-        r.json().then(data => {
-            const content = document.getElementById('items');
+    function reload() {
+        fetch(`${baseUrl}/api/streams`).then(r => {
+            r.json().then(data => {
+                let html = '';
 
-            for (let name in data.streams) {
-                let html = `<tr><td>${name || 'default'}</td>`;
-                links.forEach(link => {
-                    html += `<td>${link.replace('{name}', name)}</td>`
-                })
-                html += `</tr>`;
-                content.innerHTML += html
-            }
-        });
-    })
+                for (const [name, value] of Object.entries(data)) {
+                    const online = value !== null ? value.length : 0
+                    html += `<tr><td>${name || 'default'}</td><td>${online}</td><td>`;
+                    links.forEach(link => {
+                        html += link.replace('{name}', encodeURIComponent(name)) + ' ';
+                    })
+                    html += `<a href="#" onclick="deleteStream('${name}')">delete</a>`;
+                    html += `</td></tr>`;
+                }
+
+                let content = document.getElementById('streams').getElementsByTagName('tbody')[0];
+                content.innerHTML = html
+            });
+        })
+    }
+
+    function deleteStream(src) {
+        fetch(`${baseUrl}/api/streams?src=${encodeURIComponent(src)}`, {method: 'DELETE'}).then(reload);
+    }
+
+    const addButton = document.querySelector('a#add');
+    addButton.onclick = () => {
+        let src = document.querySelector('input#src');
+        fetch(`${baseUrl}/api/streams?src=${encodeURIComponent(src.value)}`, {method: 'PUT'}).then(reload);
+    }
+
+    reload();
 </script>
 </body>
 </html>