Update version to 1.1.2

Generate universal macOS binary on release

.github/workflows/builder.yml

@@ -1,59 +0,0 @@
-# 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"

.github/workflows/docker.yml

@@ -0,0 +1,75 @@
+name: docker
+
+on:
+  workflow_dispatch:
+  push:
+    branches:
+      - 'master'
+    tags:
+      - 'v*'
+
+jobs:
+  build-and-push:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v3
+
+      - name: Docker meta
+        id: meta
+        uses: docker/metadata-action@v4
+        with:
+          images: ${{ github.repository }}
+          tags: |
+            type=ref,event=branch
+            type=semver,pattern={{version}},enable=false
+            type=match,pattern=v(.*),group=1
+
+      - name: Docker meta Hardware
+        id: meta-hw
+        uses: docker/metadata-action@v4
+        with:
+          images: ${{ github.repository }}
+          flavor: |
+            suffix=-hardware
+            latest=false
+          tags: |
+            type=ref,event=branch
+            type=semver,pattern={{version}},enable=false
+            type=match,pattern=v(.*),group=1
+
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v2
+
+      - name: Set up Docker Buildx
+        uses: docker/setup-buildx-action@v2
+
+      - name: Login to DockerHub
+        if: github.event_name != 'pull_request'
+        uses: docker/login-action@v2
+        with:
+          username: ${{ secrets.DOCKERHUB_USERNAME }}
+          password: ${{ secrets.DOCKERHUB_TOKEN }}
+
+      - name: Build and push
+        uses: docker/build-push-action@v3
+        with:
+          context: .
+          platforms: |
+            linux/amd64
+            linux/386
+            linux/arm/v7
+            linux/arm64/v8
+          push: ${{ github.event_name != 'pull_request' }}
+          tags: ${{ steps.meta.outputs.tags }}
+          labels: ${{ steps.meta.outputs.labels }}
+
+      - name: Build and push Hardware
+        uses: docker/build-push-action@v3
+        with:
+          context: .
+          file: hardware.Dockerfile
+          platforms: linux/amd64
+          push: ${{ github.event_name != 'pull_request' }}
+          tags: ${{ steps.meta-hw.outputs.tags }}
+          labels: ${{ steps.meta-hw.outputs.labels }}

.github/workflows/release.yml

@@ -0,0 +1,99 @@
+name: release
+
+on:
+  workflow_dispatch:
+#  push:
+#    tags:
+#      - 'v*'
+
+jobs:
+  build-and-release:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v3
+      - name: Generate changelog
+        run: |
+          echo -e "$(git log $(git describe --tags --abbrev=0)..HEAD --oneline | awk '{print "- "$0}')" > CHANGELOG.md
+      - name: install lipo
+        run: |
+          curl -L -o /tmp/lipo https://github.com/konoui/lipo/releases/latest/download/lipo_Linux_amd64
+          chmod +x /tmp/lipo
+          mv /tmp/lipo /usr/local/bin
+      - name: Build Go binaries
+        run: |
+          #!/bin/bash
+          
+          export CGO_ENABLED=0
+          
+          mkdir -p artifacts
+
+          export GOOS=windows
+          export GOARCH=amd64
+          export FILENAME=artifacts/go2rtc_win64.zip
+          go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel "$FILENAME" go2rtc.exe
+          
+          export GOOS=windows
+          export GOARCH=386
+          export FILENAME=artifacts/go2rtc_win32.zip
+          go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel "$FILENAME" go2rtc.exe
+          
+          export GOOS=windows
+          export GOARCH=arm64
+          export FILENAME=artifacts/go2rtc_win_arm64.zip
+          go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel "$FILENAME" go2rtc.exe
+          
+          export GOOS=linux 
+          export GOARCH=amd64
+          export FILENAME=artifacts/go2rtc_linux_amd64
+          go build -ldflags "-s -w" -trimpath -o "$FILENAME"
+          
+          export GOOS=linux
+          export GOARCH=386
+          export FILENAME=artifacts/go2rtc_linux_i386
+          go build -ldflags "-s -w" -trimpath -o "$FILENAME"
+          
+          export GOOS=linux
+          export GOARCH=arm64
+          export FILENAME=artifacts/go2rtc_linux_arm64
+          go build -ldflags "-s -w" -trimpath -o "$FILENAME"
+          
+          export GOOS=linux
+          export GOARCH=arm
+          export GOARM=7
+          export FILENAME=artifacts/go2rtc_linux_arm
+          go build -ldflags "-s -w" -trimpath -o "$FILENAME"
+          
+          export GOOS=linux
+          export GOARCH=mipsle
+          export FILENAME=artifacts/go2rtc_linux_mipsel
+          go build -ldflags "-s -w" -trimpath -o "$FILENAME"
+          
+          export GOOS=darwin
+          export GOARCH=amd64
+          go build -ldflags "-s -w" -trimpath -o go2rtc.amd64
+          
+          export GOOS=darwin
+          export GOARCH=arm64
+          go build -ldflags "-s -w" -trimpath -o go2rtc.arm64
+
+          export FILENAME=artifacts/go2rtc_mac_universal.zip
+          lipo -output go2rtc -create go2rtc.arm64 go2rtc.amd64 && 7z a -mx9 -sdel "$FILENAME" go2rtc
+          
+          parallel --jobs $(nproc) "upx {}" ::: artifacts/go2rtc_linux_*
+      - name: Setup tmate session
+        uses: mxschmitt/action-tmate@v3
+        if: ${{ failure() }}
+      - name: Set env
+        run: echo "RELEASE_VERSION=${GITHUB_REF#refs/*/}" >> $GITHUB_ENV
+      - name: Create GitHub release
+        uses: softprops/action-gh-release@v1
+        env:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        with:
+          files: artifacts/*
+          generate_release_notes: true
+          name: Release ${{ env.RELEASE_VERSION }}
+          body_path: CHANGELOG.md
+          draft: false
+          prerelease: false

Dockerfile

@@ -0,0 +1,61 @@
+# syntax=docker/dockerfile:labs
+
+# 0. Prepare images
+ARG PYTHON_VERSION="3.11"
+ARG GO_VERSION="1.19"
+ARG NGROK_VERSION="3"
+
+FROM python:${PYTHON_VERSION}-alpine AS base
+FROM ngrok/ngrok:${NGROK_VERSION}-alpine AS ngrok
+
+
+# 1. Build go2rtc binary
+FROM --platform=$BUILDPLATFORM golang:${GO_VERSION}-alpine AS build
+ARG TARGETPLATFORM
+ARG TARGETOS
+ARG TARGETARCH
+
+ENV GOOS=${TARGETOS}
+ENV GOARCH=${TARGETARCH}
+
+WORKDIR /build
+
+# Cache dependencies
+COPY go.mod go.sum ./
+RUN --mount=type=cache,target=/root/.cache/go-build go mod download
+
+COPY . .
+RUN --mount=type=cache,target=/root/.cache/go-build CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
+
+
+# 2. Collect all files
+FROM scratch AS rootfs
+
+COPY --from=build /build/go2rtc /usr/local/bin/
+COPY --from=ngrok /bin/ngrok /usr/local/bin/
+
+
+# 3. Final image
+FROM base
+
+# Install ffmpeg, tini (for signal handling),
+# and other common tools for the echo source.
+RUN apk add --no-cache tini ffmpeg bash curl jq
+
+# Hardware Acceleration for Intel CPU (+50MB)
+ARG TARGETARCH
+
+RUN if [ "${TARGETARCH}" = "amd64" ]; then apk add --no-cache libva-intel-driver intel-media-driver; fi
+
+# Hardware: AMD and NVidia VAAPI (not sure about this)
+# RUN libva-glx mesa-va-gallium
+# Hardware: AMD and NVidia VDPAU (not sure about this)
+# RUN libva-vdpau-driver mesa-vdpau-gallium (+150MB total)
+
+COPY --from=rootfs / /
+
+ENTRYPOINT ["/sbin/tini", "--"]
+VOLUME /config
+WORKDIR /config
+
+CMD ["go2rtc", "-config", "/config/go2rtc.yaml"]

README.md

@@ -6,10 +6,10 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
 
 - zero-dependency and zero-config [small app](#go2rtc-binary) for all OS (Windows, macOS, Linux, ARM)
 - zero-delay for many supported protocols (lowest possible streaming latency)
-- streaming from [RTSP](#source-rtsp), [RTMP](#source-rtmp), [MJPEG](#source-ffmpeg), [HLS/HTTP](#source-ffmpeg), [USB Cameras](#source-ffmpeg-device) and [other sources](#module-streams)
-- streaming to [RTSP](#module-rtsp), [WebRTC](#module-webrtc), [MSE/MP4](#module-mp4) or [MJPEG](#module-mjpeg)
+- streaming from [RTSP](#source-rtsp), [RTMP](#source-rtmp), [HTTP](#source-http) (FLV/MJPEG/JPEG), [FFmpeg](#source-ffmpeg), [USB Cameras](#source-ffmpeg-device) and [other sources](#module-streams)
+- streaming to [RTSP](#module-rtsp), [WebRTC](#module-webrtc), [MSE/MP4](#module-mp4), [HLS](#module-hls) or [MJPEG](#module-mjpeg)
 - first project in the World with support streaming from [HomeKit Cameras](#source-homekit)
-- first project in the World with support H265 for WebRTC in browser ([read more](https://github.com/AlexxIT/Blog/issues/5))
+- first project in the World with support H265 for WebRTC in browser (Safari only, [read more](https://github.com/AlexxIT/Blog/issues/5))
 - on the fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
 - multi-source 2-way [codecs negotiation](#codecs-negotiation)
    - mixing tracks from different sources to single stream
@@ -27,17 +27,52 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
 - [MediaSoup](https://mediasoup.org/) framework routing idea
 - HomeKit Accessory Protocol from [@brutella](https://github.com/brutella/hap)
 
+---
+
+* [Fast start](#fast-start)
+  * [go2rtc: Binary](#go2rtc-binary)
+  * [go2rtc: Docker](#go2rtc-docker)
+  * [go2rtc: Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+  * [go2rtc: Home Assistant Integration](#go2rtc-home-assistant-integration)
+* [Configuration](#configuration)
+  * [Module: Streams](#module-streams)
+    * [Source: RTSP](#source-rtsp)
+    * [Source: RTMP](#source-rtmp)
+    * [Source: HTTP](#source-http)
+    * [Source: FFmpeg](#source-ffmpeg)
+    * [Source: FFmpeg Device](#source-ffmpeg-device)
+    * [Source: Exec](#source-exec)
+    * [Source: Echo](#source-echo)
+    * [Source: HomeKit](#source-homekit)
+    * [Source: Ivideon](#source-ivideon)
+    * [Source: Hass](#source-hass)
+  * [Module: API](#module-api)
+  * [Module: RTSP](#module-rtsp)
+  * [Module: WebRTC](#module-webrtc)
+  * [Module: Ngrok](#module-ngrok)
+  * [Module: Hass](#module-hass)
+  * [Module: MP4](#module-mp4)
+  * [Module: HLS](#module-hls)
+  * [Module: MJPEG](#module-mjpeg)
+  * [Module: Log](#module-log)
+* [Security](#security)
+* [Codecs filters](#codecs-filters)
+* [Codecs madness](#codecs-madness)
+* [Codecs negotiation](#codecs-negotiation)
+* [Projects using go2rtc](#projects-using-go2rtc)
+* [Camera experience](#cameras-experience)
+* [TIPS](#tips)
+* [FAQ](#faq)
+
 ## Fast start
 
-1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or Home Assistant [Add-on](#go2rtc-home-assistant-add-on) or [Integration](#go2rtc-home-assistant-integration)
 2. Open web interface: `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:**
 
@@ -54,12 +89,16 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
 - `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_mipsel` - Linux on MIPS (ex. [Xiaomi Gateway 3](https://github.com/AlexxIT/XiaomiGateway3))
-- `go2rtc_mac_amd64` - Mac with Intel
-- `go2rtc_mac_arm64` - Mac with M1
+- `go2rtc_linux_mipsel` - Linux MIPS (ex. [Xiaomi Gateway 3](https://github.com/AlexxIT/XiaomiGateway3))
+- `go2rtc_mac_amd64.zip` - Mac Intel 64-bit
+- `go2rtc_mac_arm64.zip` - Mac ARM 64-bit
 
 Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
 
+### go2rtc: Docker
+
+Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support `amd64`, `386`, `arm64`, `arm`. This container is the same as [Home Assistant Add-on](#go2rtc-home-assistant-add-on), but can be used separately from Home Assistant. Container has preinstalled [FFmpeg](#source-ffmpeg), [Ngrok](#module-ngrok) and [Python](#source-echo).
+
 ### 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)
@@ -69,29 +108,19 @@ Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
     - go2rtc > Install > Start
 2. Setup [Integration](#module-hass)
 
-### go2rtc: Docker
+### go2rtc: Home Assistant Integration
 
-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), [Ngrok](#module-ngrok) and [Python](#source-echo).
-
-```yaml
-services:
-  go2rtc:
-    image: alexxit/go2rtc
-    network_mode: host
-    restart: always
-    volumes:
-      - "~/go2rtc.yaml:/config/go2rtc.yaml"
-```
+[WebRTC Camera](https://github.com/AlexxIT/WebRTC) custom component can be used on any [Home Assistant installation](https://www.home-assistant.io/installation/), including [HassWP](https://github.com/AlexxIT/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.
 
 ## Configuration
 
-Create file `go2rtc.yaml` next to the app.
+- by default go2rtc will search `go2rtc.yaml` in the current work dirrectory
+- `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
 
-- by default, you need to config only your `streams` links
-- `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 may install it [manually](https://ffmpeg.org/))
+Configuration options and a complete list of settings can be found in [the wiki](https://github.com/AlexxIT/go2rtc/wiki/Configuration).
 
 Available modules:
 
@@ -100,6 +129,7 @@ Available modules:
 - [rtsp](#module-rtsp) - RTSP Server (important for FFmpeg support)
 - [webrtc](#module-webrtc) - WebRTC Server
 - [mp4](#module-mp4) - MSE, MP4 stream and MP4 shapshot Server
+- [hls](#module-hls) - HLS TS or fMP4 stream Server
 - [mjpeg](#module-mjpeg) - MJPEG Server
 - [ffmpeg](#source-ffmpeg) - FFmpeg integration
 - [ngrok](#module-ngrok) - Ngrok integration (external access for private network)
@@ -215,7 +245,7 @@ But you can override them via YAML config. You can also add your own formats to
 
 ```yaml
 ffmpeg:
-  bin: ffmpeg                                        # path to ffmpeg binary
+  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 support ffmpeg..."
 ```
@@ -223,8 +253,11 @@ ffmpeg:
 - You can use `video` and `audio` params multiple times (ex. `#video=copy#audio=copy#audio=pcmu`)
 - You can use go2rtc stream name as ffmpeg input (ex. `ffmpeg:camera1#video=h264`)
 - You can use `rotate` params 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 `raw` param for any additional FFmpeg arguments (ex. `#raw=-vf transpose=1`).
 
+Read more about encoding [hardware acceleration](https://github.com/AlexxIT/go2rtc/wiki/Hardware-acceleration).
+
 #### Source: FFmpeg Device
 
 You can get video from any USB-camera or Webcam as RTSP or WebRTC stream. This is part of FFmpeg integration.
@@ -329,7 +362,34 @@ More cameras, like [Tuya](https://www.home-assistant.io/integrations/tuya/), [ON
 
 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
+go2rtc has its own JS video player (`video-rtc.js`) with:
+
+- support technologies:
+   - WebRTC over UDP or TCP
+   - MSE or MP4 or MJPEG over WebSocket 
+- automatic selection best technology according on:
+   - codecs inside your stream
+   - current browser capabilities
+   - current network configuration
+- automatic stop stream while browser or page not active
+- automatic stop stream while player not inside page viewport
+- automatic reconnection
+
+Technology selection based on priorities:
+
+1. Video and Audio better than just Video
+2. H265 better than H264
+3. WebRTC better than MSE, than MP4, than MJPEG
+
+go2rtc has simple HTML page (`stream.html`) with support params in URL:
+
+- multiple streams on page `src=camera1&src=camera2...`
+- stream technology autoselection `mode=webrtc,mse,mp4,mjpeg`
+- stream technology comparison `src=camera1&mode=webrtc&mode=mse&mode=mp4`
+- player width setting in pixels `width=320px` or percents `width=50%`
+
+**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
@@ -337,77 +397,85 @@ The HTTP API is the main part for interacting with the application. Default addr
 
 ```yaml
 api:
-  listen: ":1984"    # HTTP API port ("" - disabled)
-  base_path: "/rtc"  # API prefix for serve on suburl (/api => /rtc/api)
-  static_dir: "www"  # folder for static files (custom web interface)
-  origin: "*"        # allow CORS requests (only * supported)
+  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)
 ```
 
-**PS. go2rtc** doesn'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.
+**PS:**
 
-**PS2.** You can access microphone (for 2-way audio) only with HTTPS ([read more](https://stackoverflow.com/questions/52759992/how-to-access-camera-and-microphone-in-chrome-without-https)).
+- go2rtc doesn't provide HTTPS. Use [Nginx](https://nginx.org/) or [Ngrok](#module-ngrok) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on) for this tasks
+- you can access microphone (for 2-way audio) only with HTTPS ([read more](https://stackoverflow.com/questions/52759992/how-to-access-camera-and-microphone-in-chrome-without-https))
+- MJPEG over WebSocket plays better than native MJPEG because Chrome [bug](https://bugs.chromium.org/p/chromium/issues/detail?id=527446)
+- 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 omit the codec filters, so one first video and one first audio will be selected
-- you can set `?video=copy` or just `?video`, so only one first video without audio will be selected
-- you can set multiple video or audio, so all of them will be selected
-- you can enable external password protection for your RTSP streams
-
-Password protection always disabled for localhost calls (ex. FFmpeg or Hass on same server)
+You can enable external password protection for your RTSP streams. Password protection always disabled for localhost calls (ex. FFmpeg or Hass on same server).
 
 ```yaml
 rtsp:
-  listen: ":8554"  # RTSP Server TCP port, default - 8554
-  username: admin  # optional, default - disabled
-  password: pass   # optional, default - disabled
+  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](#codecs-filters).
+
 ### 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.
 
-- 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
+- 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:
+
+```yaml
+webrtc:
+  listen: ":8555"  # address of your local server and port (TCP/UDP)
+```
 
 **Static public IP**
 
-- add some TCP port to YAML config (ex. 8555)
-- forward this port on your router (you can use same 8555 port or any other)
+- 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
 
 ```yaml
 webrtc:
-  listen: ":8555"  # address of your local server (TCP)
   candidates:
     - 216.58.210.174:8555  # if you have static public IP-address
 ```
 
 **Dynamic public IP**
 
-- add some TCP port to YAML config (ex. 8555)
-- forward this port on your router (you can use same 8555 port or any other)
+- 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
 
 ```yaml
 webrtc:
-  listen: ":8555"  # address of your local server (TCP)
   candidates:
     - stun:8555  # if you have dynamic public IP-address
 ```
 
 **Private IP**
 
-- add some TCP port to YAML config (ex. 8555)
 - setup integration with [Ngrok service](#module-ngrok)
 
 ```yaml
-webrtc:
-  listen: ":8555"  # address of your local server (TCP)
-
 ngrok:
   command: ...
 ```
@@ -485,23 +553,29 @@ tunnels:
 
 ### Module: Hass
 
-If you install **go2rtc** as [Hass Add-on](#go2rtc-home-assistant-add-on) - you need to use localhost IP-address. In other cases you need to use IP-address of server with **go2rtc** application.
+The best and easiest way to use go2rtc inside the Home Assistant is to install the custom integration [WebRTC Camera](#go2rtc-home-assistant-integration) and custom lovelace card.
 
-#### From go2rtc to Hass
+But go2rtc is also compatible and can be used with [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) built-in integration.
 
-Add any supported [stream source](#module-streams) as [Generic Camera](https://www.home-assistant.io/integrations/generic/) and view stream with built-in [Stream](https://www.home-assistant.io/integrations/stream/) integration. Technology `HLS`, supported codecs: `H264`, poor latency.
+You have several options on how to add a camera to Home Assistant:
 
-1. Add your stream to [go2rtc config](#configuration)
-2. Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://127.0.0.1:8554/camera1`
+1. Camera RTSP source => [Generic Camera](https://www.home-assistant.io/integrations/generic/)
+2. Camera [any source](#module-streams) => [go2rtc config](#configuration) => [Generic Camera](https://www.home-assistant.io/integrations/generic/)
+   - Install any [go2rtc](#fast-start)
+   - Add your stream to [go2rtc config](#configuration)
+   - Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://127.0.0.1:8554/camera1` (change to your stream name)
 
-#### From Hass to go2rtc
+You have several options on how to watch the stream from the cameras in Home Assistant:
 
-View almost any Hass camera using `WebRTC` technology, supported codecs `H264`/`PCMU`/`PCMA`/`OPUS`, best latency.
-
-When the stream starts - the camera `entity_id` will be added to go2rtc "on the fly". You don't need to add cameras manually to [go2rtc config](#configuration). Some cameras (like [Nest](https://www.home-assistant.io/integrations/nest/)) have a dynamic link to the stream, it will be updated each time a stream is started from the Hass interface. 
-
-1. 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. Use Picture Entity or Picture Glance lovelace card
+1. `Camera Entity` => `Picture Entity Card` => Technology `HLS`, codecs: `H264/H265/AAC`, poor latency.
+2. `Camera Entity` => [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) => `Picture Entity Card` => Technology `WebRTC`, codecs: `H264/PCMU/PCMA/OPUS`, best latency.
+   - Install any [go2rtc](#fast-start)
+   - 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/`
+   - 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](https://github.com/AlexxIT/WebRTC) => Technology: `WebRTC/MSE/MP4/MJPEG`, codecs: `H264/H265/AAC/PCMU/PCMA/OPUS`, best latency, best compatibility.
+   - Install and add [WebRTC Camera](https://github.com/AlexxIT/WebRTC) custom integration
+   - Use WebRTC Camera custom lovelace card
 
 You can add camera `entity_id` to [go2rtc config](#configuration) if you need transcoding:
 
@@ -517,8 +591,28 @@ PS. Default Home Assistant lovelace cards don't support 2-way audio. You can use
 Provides several features:
 
 1. MSE stream (fMP4 over WebSocket)
-2. Camera snapshots in MP4 format (single frame), can be sent to [Telegram](https://www.telegram.org/)
-3. MP4 "file stream" - bad format for streaming because of high latency, doesn't work in Safari 
+2. Camera snapshots in MP4 format (single frame), can be sent to [Telegram](https://github.com/AlexxIT/go2rtc/wiki/Snapshot-to-Telegram)
+3. 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 stream: `http://192.168.1.123:1984/api/stream.mp4?src=camera1`
+- MP4 snapshot: `http://192.168.1.123:1984/api/frame.mp4?src=camera1`
+
+Read more about [codecs filters](#codecs-filters).
+
+### Module: HLS
+
+[HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming) is the worst technology for real-time streaming. It can only be useful on devices that do not support more modern technology, like [WebRTC](#module-webrtc), [MSE/MP4](#module-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](#codecs-filters).
 
 ### Module: MJPEG
 
@@ -562,7 +656,7 @@ log:
 
 ## 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.
+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:
 
@@ -574,7 +668,7 @@ rtsp:
   listen: "127.0.0.1:8554" # localhost
 
 webrtc:
-  listen: ":8555" # external TCP port
+  listen: ":8555" # external TCP/UDP port
 ```
 
 - local access to RTSP is not a problem for [FFmpeg](#source-ffmpeg) integration, because it runs locally on your server
@@ -584,23 +678,44 @@ webrtc:
 
 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.
+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](https://en.wikipedia.org/wiki/UDP_hole_punching)). 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](#module-webrtc) and [MSE](#module-mp4) technologies.
+
+But it cannot be done for [RTSP](#module-rtsp), [stream.mp4](#module-mp4), [HLS](#module-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](#source-ffmpeg).
+
+Without filters:
+
+- RTSP will provide only the first video and only the first audio
+- 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.mp4?src=camera1&video=h264,h265&audio=aac,opus,mp3,pcma,pcmu` - MP4 file with non standard audio codecs, does not work in some players
 
 ## Codecs madness
 
-`AVC/H.264` codec 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.
+`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.
 
-| Device              | WebRTC      | MSE         | MP4         |
-|---------------------|-------------|-------------|-------------|
-| *latency*           | best        | medium      | bad         |
-| Desktop Chrome 107+ | H264        | H264, H265* | H264, H265* |
-| Desktop Edge        | H264        | H264, H265* | H264, H265* |
-| Desktop Safari      | H264, H265* | H264, H265  | **no!**     |
-| Desktop Firefox     | H264        | H264        | H264        |
-| Android Chrome 107+ | H264        | H264, H265* | H264        |
-| iPad Safari 13+     | H264, H265* | H264, H265  | **no!**     |
-| iPhone Safari 13+   | H264, H265* | **no!**     | **no!**     |
-| masOS Hass App      | no          | no          | no          |
+| Device              | WebRTC                        | MSE                    | stream.mp4                              |
+|---------------------|-------------------------------|------------------------|-----------------------------------------|
+| *latency*           | best                          | medium                 | bad                                     |
+| Desktop Chrome 107+ | H264, OPUS, PCMU, PCMA        | H264, H265*, AAC, OPUS | H264, H265*, AAC, OPUS, PCMU, PCMA, MP3 |
+| Desktop Edge        | H264, OPUS, PCMU, PCMA        | H264, H265*, AAC, OPUS | H264, H265*, AAC, OPUS, PCMU, PCMA, MP3 |
+| Desktop Safari      | H264, H265*, OPUS, PCMU, PCMA | H264, H265, AAC        | **no!**                                 |
+| Desktop Firefox     | H264, OPUS, PCMU, PCMA        | H264, AAC, OPUS        | H264, AAC, OPUS                         |
+| Android Chrome 107+ | H264, OPUS, PCMU, PCMA        | H264, H265*, AAC, OPUS | H264, ?, AAC, OPUS, PCMU, PCMA, MP3     |
+| iPad Safari 13+     | H264, H265*, OPUS, PCMU, PCMA | H264, H265, AAC        | **no!**                                 |
+| iPhone Safari 13+   | H264, H265*, OPUS, PCMU, PCMA | **no!**                | **no!**                                 |
+| masOS Hass App      | no                            | no                     | no                                      |
 
 - Chrome H265: [read this](https://chromestatus.com/feature/5186511939567616) and [read this](https://github.com/StaZhu/enable-chromium-hevc-hardware-decoding)
 - Edge H265: [read this](https://www.reddit.com/r/MicrosoftEdge/comments/v9iw8k/enable_hevc_support_in_edge/)
@@ -609,8 +724,9 @@ PS. Additionally WebRTC opens a lot of random UDP ports for transmit encrypted m
 
 **Audio**
 
-- WebRTC audio codecs: `PCMU/8000`, `PCMA/8000`, `OPUS/48000/2`
-- MSE/MP4 audio codecs: `AAC`
+- **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)
+- `PCMU` and `PCMA` inside **MP4** isn't a standard, but some players support them, for example Chromium browsers
 
 **Apple devices**
 
@@ -647,6 +763,21 @@ streams:
 
 **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
+
+- [Frigate 12+](https://frigate.video/) - open source NVR built around real-time AI object detection
+- [ring-mqtt](https://github.com/tsightler/ring-mqtt) - Ring devices to MQTT Bridge
+- [EufyP2PStream](https://github.com/oischinger/eufyp2pstream) - A small project that provides a Video/Audio Stream from Eufy cameras that don't directly support RTSP
+
+## Cameras experience
+
+- [Dahua](https://www.dahuasecurity.com/) - reference implementation streaming protocols, a lot of settings, high stream quality, multiple streaming clients
+- [Hikvision](https://www.hikvision.com/) - a lot of proprietary streaming technologies
+- [Reolink](https://reolink.com/) - some models has awful unusable RTSP realisation and not best HTTP-FLV alternative (I recommend that you contact Reolink support for new firmware), few settings
+- [Sonoff](https://sonoff.tech/) - very low stream quality, no settings, not best protocol implementation
+- [TP-Link](https://www.tp-link.com/) - 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**
@@ -654,23 +785,27 @@ streams:
 - `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](https://github.com/AlexxIT/go2rtc/wiki/Snapshot-to-Telegram)
+
 ## FAQ
 
 **Q. What's the difference between go2rtc, WebRTC Camera and RTSPtoWebRTC?**
 
 **go2rtc** is a new version of the server-side [WebRTC Camera](https://github.com/AlexxIT/WebRTC) 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](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration. So you [can use](#module-hass) default lovelace Picture Entity or Picture Glance.
 
-**Q. Why go2rtc is an addon and not an integration?**
+**Q. Should I use go2rtc addon or WebRTC Camera integration?**
 
-Because **go2rtc** is more than just viewing your stream online with WebRTC. 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.
+**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.
 
-When **go2rtc** is released, the **WebRTC Camera** integration will be updated. And you can decide whether to use the integration or the addon.
+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](#configuration) and use RTSP links to this addon. With additional features: multi-source [codecs negotiation](#codecs-negotiation) or FFmpeg [transcoding](#source-ffmpeg) 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 protocol.
+Also you can specify your streams in **go2rtc** [config file](#configuration) and use RTSP links to this addon. With additional features: multi-source [codecs negotiation](#codecs-negotiation) or FFmpeg [transcoding](#source-ffmpeg) 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.
 

build/docker/run.sh

@@ -0,0 +1,19 @@
+#!/bin/bash
+
+set -euo pipefail
+
+echo "Starting go2rtc..." >&2
+
+readonly config_path="/config"
+
+if [[ -x "${config_path}/go2rtc" ]]; then
+  readonly binary_path="${config_path}/go2rtc"
+  echo "Using go2rtc binary from '${binary_path}' instead of the embedded one" >&2
+else
+  readonly binary_path="/usr/local/bin/go2rtc"
+fi
+
+# set cwd for go2rtc (for config file, Hass integration, etc)
+cd "${config_path}" || echo "Could not change working directory to '${config_path}'" >&2
+
+exec "${binary_path}"

build/hassio/Dockerfile

@@ -1,41 +0,0 @@
-ARG BUILD_FROM
-
-FROM $BUILD_FROM as build
-
-# 1. Build go2rtc
-RUN apk add --no-cache git go
-
-RUN git clone https://github.com/AlexxIT/go2rtc \
-    && cd go2rtc \
-    && CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
-
-# 2. Download ngrok
-ARG BUILD_ARCH
-
-# https://github.com/home-assistant/docker-base/blob/master/alpine/Dockerfile
-RUN if [ "${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
-
-
-
-# https://devopscube.com/reduce-docker-image-size/
-FROM $BUILD_FROM
-
-# 3. Copy go2rtc and ngrok to release
-COPY --from=build /go2rtc/go2rtc /usr/local/bin
-COPY --from=build /go2rtc/ngrok /usr/local/bin
-
-# 4. Install ffmpeg
-# apk base  OK: 22  MiB in 40  packages
-# ffmpeg    OK: 113 MiB in 110 packages
-# python3   OK: 161 MiB in 114 packages
-RUN apk add --no-cache ffmpeg python3
-
-# 5. Copy run to release
-COPY run.sh /
-RUN chmod a+x /run.sh
-
-CMD [ "/run.sh" ]

build/hassio/config.yaml

@@ -1,6 +0,0 @@
-# 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

@@ -1,14 +0,0 @@
-#!/usr/bin/with-contenv bashio
-
-set +e
-
-# set cwd for go2rtc (for config file, Hass integration, 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

@@ -3,16 +3,21 @@ package api
 import (
 	"encoding/json"
 	"github.com/AlexxIT/go2rtc/cmd/app"
-	"github.com/AlexxIT/go2rtc/cmd/streams"
 	"github.com/rs/zerolog"
 	"net"
 	"net/http"
+	"os"
+	"strconv"
+	"strings"
+	"sync"
 )
 
 func Init() {
 	var cfg struct {
 		Mod struct {
 			Listen    string `yaml:"listen"`
+			Username  string `yaml:"username"`
+			Password  string `yaml:"password"`
 			BasePath  string `yaml:"base_path"`
 			StaticDir string `yaml:"static_dir"`
 			Origin    string `yaml:"origin"`
@@ -35,7 +40,9 @@ func Init() {
 	initStatic(cfg.Mod.StaticDir)
 	initWS(cfg.Mod.Origin)
 
-	HandleFunc("api/streams", streamsHandler)
+	HandleFunc("api", apiHandler)
+	HandleFunc("api/config", configHandler)
+	HandleFunc("api/exit", exitHandler)
 	HandleFunc("api/ws", apiWS)
 
 	// ensure we can listen without errors
@@ -48,14 +55,18 @@ func Init() {
 	log.Info().Str("addr", cfg.Mod.Listen).Msg("[api] listen")
 
 	s := http.Server{}
-	s.Handler = http.DefaultServeMux
-
-	if log.Trace().Enabled() {
-		s.Handler = middlewareLog(s.Handler)
-	}
+	s.Handler = http.DefaultServeMux // 4th
 
 	if cfg.Mod.Origin == "*" {
-		s.Handler = middlewareCORS(s.Handler)
+		s.Handler = middlewareCORS(s.Handler) // 3rd
+	}
+
+	if cfg.Mod.Username != "" {
+		s.Handler = middlewareAuth(cfg.Mod.Username, cfg.Mod.Password, s.Handler) // 2nd
+	}
+
+	if log.Trace().Enabled() {
+		s.Handler = middlewareLog(s.Handler) // 1st
 	}
 
 	go func() {
@@ -83,7 +94,22 @@ var log zerolog.Logger
 
 func middlewareLog(next http.Handler) http.Handler {
 	return http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
-		log.Trace().Msgf("[api] %s %s", r.Method, r.URL)
+		log.Trace().Msgf("[api] %s %s %s", r.Method, r.URL, r.RemoteAddr)
+		next.ServeHTTP(w, r)
+	})
+}
+
+func middlewareAuth(username, password string, next http.Handler) http.Handler {
+	return http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		if !strings.HasPrefix(r.RemoteAddr, "127.") && !strings.HasPrefix(r.RemoteAddr, "[::1]") {
+			user, pass, ok := r.BasicAuth()
+			if !ok || user != username || pass != password {
+				w.Header().Set("Www-Authenticate", `Basic realm="go2rtc"`)
+				http.Error(w, "Unauthorized", http.StatusUnauthorized)
+				return
+			}
+		}
+
 		next.ServeHTTP(w, r)
 	})
 }
@@ -96,31 +122,25 @@ func middlewareCORS(next http.Handler) http.Handler {
 	})
 }
 
-func streamsHandler(w http.ResponseWriter, r *http.Request) {
-	src := r.URL.Query().Get("src")
-	name := r.URL.Query().Get("name")
+var mu sync.Mutex
 
-	if name == "" {
-		name = src
+func apiHandler(w http.ResponseWriter, r *http.Request) {
+	mu.Lock()
+	app.Info["host"] = r.Host
+	mu.Unlock()
+
+	if err := json.NewEncoder(w).Encode(app.Info); err != nil {
+		log.Warn().Err(err).Caller().Send()
 	}
-
-	switch r.Method {
-	case "PUT":
-		streams.New(name, src)
-		return
-	case "DELETE":
-		streams.Delete(src)
-		return
-	}
-
-	var v interface{}
-	if src != "" {
-		v = streams.Get(src)
-	} else {
-		v = streams.All()
-	}
-
-	e := json.NewEncoder(w)
-	e.SetIndent("", "  ")
-	_ = e.Encode(v)
+}
+
+func exitHandler(w http.ResponseWriter, r *http.Request) {
+	if r.Method != "POST" {
+		http.Error(w, "", http.StatusBadRequest)
+		return
+	}
+
+	s := r.URL.Query().Get("code")
+	code, _ := strconv.Atoi(s)
+	os.Exit(code)
 }

cmd/api/config.go

@@ -0,0 +1,102 @@
+package api
+
+import (
+	"github.com/AlexxIT/go2rtc/cmd/app"
+	"gopkg.in/yaml.v3"
+	"io"
+	"net/http"
+	"os"
+)
+
+func configHandler(w http.ResponseWriter, r *http.Request) {
+	if app.ConfigPath == "" {
+		http.Error(w, "", http.StatusGone)
+		return
+	}
+
+	switch r.Method {
+	case "GET":
+		data, err := os.ReadFile(app.ConfigPath)
+		if err != nil {
+			http.Error(w, "", http.StatusNotFound)
+			return
+		}
+		if _, err = w.Write(data); err != nil {
+			log.Warn().Err(err).Caller().Send()
+		}
+
+	case "POST", "PATCH":
+		data, err := io.ReadAll(r.Body)
+		if err != nil {
+			http.Error(w, err.Error(), http.StatusBadRequest)
+			return
+		}
+
+		if r.Method == "PATCH" {
+			// no need to validate after merge
+			data, err = mergeYAML(app.ConfigPath, data)
+			if err != nil {
+				http.Error(w, err.Error(), http.StatusBadRequest)
+				return
+			}
+		} else {
+			// validate config
+			var tmp struct{}
+			if err = yaml.Unmarshal(data, &tmp); err != nil {
+				http.Error(w, err.Error(), http.StatusInternalServerError)
+				return
+			}
+		}
+
+		if err = os.WriteFile(app.ConfigPath, data, 0644); err != nil {
+			http.Error(w, err.Error(), http.StatusInternalServerError)
+			return
+		}
+	}
+}
+
+func mergeYAML(file1 string, yaml2 []byte) ([]byte, error) {
+	// Read the contents of the first YAML file
+	data1, err := os.ReadFile(file1)
+	if err != nil {
+		return nil, err
+	}
+
+	// Unmarshal the first YAML file into a map
+	var config1 map[string]interface{}
+	if err = yaml.Unmarshal(data1, &config1); err != nil {
+		return nil, err
+	}
+
+	// Unmarshal the second YAML document into a map
+	var config2 map[string]interface{}
+	if err = yaml.Unmarshal(yaml2, &config2); err != nil {
+		return nil, err
+	}
+
+	// Merge the two maps
+	config1 = merge(config1, config2)
+
+	// Marshal the merged map into YAML
+	return yaml.Marshal(&config1)
+}
+
+func merge(dst, src map[string]interface{}) map[string]interface{} {
+	for k, v := range src {
+		if vv, ok := dst[k]; ok {
+			switch vv := vv.(type) {
+			case map[string]interface{}:
+				v := v.(map[string]interface{})
+				dst[k] = merge(vv, v)
+			case []interface{}:
+				v := v.([]interface{})
+				dst[k] = v
+			default:
+				dst[k] = v
+			}
+		} else {
+			dst[k] = v
+		}
+	}
+	return dst
+}

cmd/api/ws.go

@@ -6,6 +6,7 @@ import (
 	"net/url"
 	"strings"
 	"sync"
+	"time"
 )
 
 // Message - struct for data exchange in Web API
@@ -25,7 +26,7 @@ var wsHandlers = make(map[string]WSHandler)
 func initWS(origin string) {
 	wsUp = &websocket.Upgrader{
 		ReadBufferSize:  1024,
-		WriteBufferSize: 512000,
+		WriteBufferSize: 2028,
 	}
 
 	switch origin {
@@ -68,6 +69,8 @@ func apiWS(w http.ResponseWriter, r *http.Request) {
 
 	tr := &Transport{Request: r}
 	tr.OnWrite(func(msg interface{}) {
+		_ = ws.SetWriteDeadline(time.Now().Add(time.Second * 5))
+
 		if data, ok := msg.([]byte); ok {
 			_ = ws.WriteMessage(websocket.BinaryMessage, data)
 		} else {
@@ -78,7 +81,9 @@ func apiWS(w http.ResponseWriter, r *http.Request) {
 	for {
 		msg := new(Message)
 		if err = ws.ReadJSON(msg); err != nil {
-			log.Trace().Err(err).Caller().Send()
+			if !websocket.IsCloseError(err, websocket.CloseNoStatusReceived) {
+				log.Trace().Err(err).Caller().Send()
+			}
 			_ = ws.Close()
 			break
 		}
@@ -101,7 +106,9 @@ type Transport struct {
 	Request  *http.Request
 	Consumer interface{} // TODO: rewrite
 
-	mx sync.Mutex
+	closed bool
+	mx     sync.Mutex
+	wrmx   sync.Mutex
 
 	onChange func()
 	onWrite  func(msg interface{})
@@ -118,21 +125,32 @@ func (t *Transport) OnWrite(f func(msg interface{})) {
 }
 
 func (t *Transport) Write(msg interface{}) {
-	t.mx.Lock()
+	t.wrmx.Lock()
 	t.onWrite(msg)
-	t.mx.Unlock()
+	t.wrmx.Unlock()
 }
 
 func (t *Transport) Close() {
+	t.mx.Lock()
 	for _, f := range t.onClose {
 		f()
 	}
+	t.closed = true
+	t.mx.Unlock()
 }
 
 func (t *Transport) OnChange(f func()) {
+	t.mx.Lock()
 	t.onChange = f
+	t.mx.Unlock()
 }
 
 func (t *Transport) OnClose(f func()) {
-	t.onClose = append(t.onClose, f)
+	t.mx.Lock()
+	if t.closed {
+		f()
+	} else {
+		t.onClose = append(t.onClose, f)
+	}
+	t.mx.Unlock()
 }

cmd/app/app.go

@@ -2,46 +2,76 @@ package app
 
 import (
 	"flag"
+	"github.com/AlexxIT/go2rtc/pkg/shell"
 	"github.com/rs/zerolog"
 	"github.com/rs/zerolog/log"
 	"gopkg.in/yaml.v3"
 	"io"
 	"os"
+	"path/filepath"
 	"runtime"
+	"strings"
+	"time"
 )
 
-var Version = "0.1-rc.5"
+var Version = "1.1.2"
 var UserAgent = "go2rtc/" + Version
 
-func Init() {
-	config := flag.String(
-		"config",
-		"go2rtc.yaml",
-		"Path to go2rtc configuration file",
-	)
+var ConfigPath string
+var Info = map[string]interface{}{
+	"version": Version,
+}
 
+func Init() {
+	var confs Config
+
+	flag.Var(&confs, "config", "go2rtc config (path to file or raw text), support multiple")
 	flag.Parse()
 
-	data, _ = os.ReadFile(*config)
+	if confs == nil {
+		confs = []string{"go2rtc.yaml"}
+	}
+
+	for _, conf := range confs {
+		if conf[0] != '{' {
+			// config as file
+			if ConfigPath == "" {
+				ConfigPath = conf
+			}
+
+			data, _ := os.ReadFile(conf)
+			if data == nil {
+				continue
+			}
+
+			data = []byte(shell.ReplaceEnvVars(string(data)))
+			configs = append(configs, data)
+		} else {
+			// config as raw YAML
+			configs = append(configs, []byte(conf))
+		}
+	}
+
+	if ConfigPath != "" {
+		if !filepath.IsAbs(ConfigPath) {
+			if cwd, err := os.Getwd(); err == nil {
+				ConfigPath = filepath.Join(cwd, ConfigPath)
+			}
+		}
+		Info["config_path"] = ConfigPath
+	}
 
 	var cfg struct {
 		Mod map[string]string `yaml:"log"`
 	}
 
-	if data != nil {
-		if err := yaml.Unmarshal(data, &cfg); err != nil {
-			println("ERROR: " + err.Error())
-		}
-	}
+	LoadConfig(&cfg)
 
 	log.Logger = NewLogger(cfg.Mod["format"], cfg.Mod["level"])
 
 	modules = cfg.Mod
 
 	log.Info().Msgf("go2rtc version %s %s/%s", Version, runtime.GOOS, runtime.GOARCH)
-
-	path, _ := os.Getwd()
-	log.Debug().Str("cwd", path).Send()
 }
 
 func NewLogger(format string, level string) zerolog.Logger {
@@ -54,7 +84,7 @@ func NewLogger(format string, level string) zerolog.Logger {
 		}
 	}
 
-	zerolog.TimeFieldFormat = zerolog.TimeFormatUnixMs
+	zerolog.TimeFieldFormat = time.RFC3339Nano
 
 	lvl, err := zerolog.ParseLevel(level)
 	if err != nil || lvl == zerolog.NoLevel {
@@ -65,7 +95,7 @@ func NewLogger(format string, level string) zerolog.Logger {
 }
 
 func LoadConfig(v interface{}) {
-	if data != nil {
+	for _, data := range configs {
 		if err := yaml.Unmarshal(data, v); err != nil {
 			log.Warn().Err(err).Msg("[app] read config")
 		}
@@ -86,8 +116,18 @@ func GetLogger(module string) zerolog.Logger {
 
 // internal
 
-// data - config content
-var data []byte
+type Config []string
+
+func (c *Config) String() string {
+	return strings.Join(*c, " ")
+}
+
+func (c *Config) Set(value string) error {
+	*c = append(*c, value)
+	return nil
+}
+
+var configs [][]byte
 
 // modules log levels
 var modules map[string]string

cmd/debug/debug.go

@@ -4,24 +4,14 @@ 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

@@ -25,6 +25,7 @@ var stackSkip = [][]byte{
 
 	// webrtc/api.go
 	[]byte("created by github.com/pion/ice/v2.NewTCPMuxDefault"),
+	[]byte("created by github.com/pion/ice/v2.NewUDPMuxDefault"),
 }
 
 func stackHandler(w http.ResponseWriter, r *http.Request) {

cmd/exec/exec.go

@@ -34,8 +34,13 @@ func Init() {
 			return false
 		}
 
-		waiter <- conn
-		return true
+		// unblocking write to channel
+		select {
+		case waiter <- conn:
+			return true
+		default:
+			return false
+		}
 	})
 
 	streams.HandleFunc("exec", Handle)
@@ -86,7 +91,13 @@ func Handle(url string) (streamer.Producer, error) {
 	chErr := make(chan error)
 
 	go func() {
-		chErr <- cmd.Wait()
+		err := cmd.Wait()
+		// unblocking write to channel
+		select {
+		case chErr <- err:
+		default:
+			log.Trace().Str("url", url).Msg("[exec] close")
+		}
 	}()
 
 	select {

cmd/ffmpeg/device/device_darwin.go

@@ -15,11 +15,11 @@ func deviceInputSuffix(videoIdx, audioIdx int) string {
 	audio := findMedia(streamer.KindAudio, audioIdx)
 	switch {
 	case video != nil && audio != nil:
-		return `"` + video.Title + `:` + audio.Title + `"`
+		return `"` + video.MID + `:` + audio.MID + `"`
 	case video != nil:
-		return `"` + video.Title + `"`
+		return `"` + video.MID + `"`
 	case audio != nil:
-		return `"` + audio.Title + `"`
+		return `"` + audio.MID + `"`
 	}
 	return ""
 }
@@ -57,7 +57,5 @@ process:
 }
 
 func loadMedia(kind, name string) *streamer.Media {
-	return &streamer.Media{
-		Kind: kind, Title: name,
-	}
+	return &streamer.Media{Kind: kind, MID: name}
 }

cmd/ffmpeg/device/device_linux.go

@@ -13,7 +13,7 @@ const deviceInputPrefix = "-f v4l2"
 
 func deviceInputSuffix(videoIdx, audioIdx int) string {
 	video := findMedia(streamer.KindVideo, videoIdx)
-	return video.Title
+	return video.MID
 }
 
 func loadMedias() {
@@ -44,7 +44,5 @@ func loadMedia(kind, name string) *streamer.Media {
 		return nil
 	}
 
-	return &streamer.Media{
-		Kind: kind, Title: name,
-	}
+	return &streamer.Media{Kind: kind, MID: name}
 }

cmd/ffmpeg/device/device_windows.go

@@ -15,11 +15,11 @@ func deviceInputSuffix(videoIdx, audioIdx int) string {
 	audio := findMedia(streamer.KindAudio, audioIdx)
 	switch {
 	case video != nil && audio != nil:
-		return `video="` + video.Title + `":audio=` + audio.Title + `"`
+		return `video="` + video.MID + `":audio=` + audio.MID + `"`
 	case video != nil:
-		return `video="` + video.Title + `"`
+		return `video="` + video.MID + `"`
 	case audio != nil:
-		return `audio="` + audio.Title + `"`
+		return `audio="` + audio.MID + `"`
 	}
 	return ""
 }
@@ -53,7 +53,5 @@ func loadMedias() {
 }
 
 func loadMedia(kind, name string) *streamer.Media {
-	return &streamer.Media{
-		Kind: kind, Title: name,
-	}
+	return &streamer.Media{Kind: kind, MID: name}
 }

cmd/ffmpeg/ffmpeg.go

@@ -1,6 +1,8 @@
 package ffmpeg
 
 import (
+	"bytes"
+	"errors"
 	"github.com/AlexxIT/go2rtc/cmd/app"
 	"github.com/AlexxIT/go2rtc/cmd/exec"
 	"github.com/AlexxIT/go2rtc/cmd/ffmpeg/device"
@@ -17,190 +19,229 @@ func Init() {
 		Mod map[string]string `yaml:"ffmpeg"`
 	}
 
-	// defaults
-
-	cfg.Mod = map[string]string{
-		"bin": "ffmpeg",
-
-		// inputs
-		"file": "-re -stream_loop -1 -i {input}",
-		"http": "-fflags nobuffer -flags low_delay -i {input}",
-		"rtsp": "-fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_transport tcp -i {input}",
-
-		// output
-		"output": "-user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}",
-
-		// `-g 30` - group of picture, GOP, keyframe interval
-		// `-preset superfast` - we can't use ultrafast because it doesn't support `-profile main -level 4.1`
-		// `-tune zerolatency` - for minimal latency
-		// `-profile main -level 4.1` - most used streaming profile
-		// `-pix_fmt yuv420p` - if input pix format 4:2:2
-		"h264":       "-c:v libx264 -g:v 30 -preset:v superfast -tune:v zerolatency -profile:v main -level:v 4.1 -pix_fmt:v yuv420p",
-		"h264/ultra": "-c:v libx264 -g:v 30 -preset:v ultrafast -tune:v zerolatency",
-		"h264/high":  "-c:v libx264 -g:v 30 -preset:v superfast -tune:v zerolatency",
-		"h265":       "-c:v libx265 -g:v 30 -preset:v superfast -tune:v zerolatency -profile:v main -level:v 5.1 -pix_fmt:v yuv420p",
-		"mjpeg":      "-c:v mjpeg -force_duplicated_matrix:v 1 -huffman:v 0 -pix_fmt:v yuvj420p",
-		"opus":       "-c:a libopus -ar:a 48000 -ac:a 2",
-		"pcmu":       "-c:a pcm_mulaw -ar:a 8000 -ac:a 1",
-		"pcmu/16000": "-c:a pcm_mulaw -ar:a 16000 -ac:a 1",
-		"pcmu/48000": "-c:a pcm_mulaw -ar:a 48000 -ac:a 1",
-		"pcma":       "-c:a pcm_alaw -ar:a 8000 -ac:a 1",
-		"pcma/16000": "-c:a pcm_alaw -ar:a 16000 -ac:a 1",
-		"pcma/48000": "-c:a pcm_alaw -ar:a 48000 -ac:a 1",
-		"aac":        "-c:a aac", // keep sample rate and channels
-		"aac/16000":  "-c:a aac -ar:a 16000 -ac:a 1",
-	}
+	cfg.Mod = defaults // will be overriden from yaml
 
 	app.LoadConfig(&cfg)
 
-	tpl := cfg.Mod
-
-	cmd := "exec:" + tpl["bin"] + " -hide_banner "
-
 	if app.GetLogger("exec").GetLevel() >= 0 {
-		cmd += "-v error "
+		defaults["global"] += " -v error"
 	}
 
-	streams.HandleFunc("ffmpeg", func(s string) (streamer.Producer, error) {
-		s = s[7:] // remove `ffmpeg:`
-
-		var query url.Values
-		var queryVideo, queryAudio bool
-
-		if i := strings.IndexByte(s, '#'); i > 0 {
-			query = parseQuery(s[i+1:])
-			queryVideo = query["video"] != nil
-			queryAudio = query["audio"] != nil
-			s = s[:i]
-		} else {
-			// by default query both video and audio
-			queryVideo = true
-			queryAudio = true
+	streams.HandleFunc("ffmpeg", func(url string) (streamer.Producer, error) {
+		args := parseArgs(url[7:]) // remove `ffmpeg:`
+		if args == nil {
+			return nil, errors.New("can't generate ffmpeg command")
 		}
-
-		var input string
-		if i := strings.Index(s, "://"); i > 0 {
-			switch s[:i] {
-			case "http", "https", "rtmp":
-				input = strings.Replace(tpl["http"], "{input}", s, 1)
-			case "rtsp", "rtsps":
-				// https://ffmpeg.org/ffmpeg-protocols.html#rtsp
-				// skip unnecessary input tracks
-				switch {
-				case queryVideo && queryAudio:
-					input = "-allowed_media_types video+audio "
-				case queryVideo:
-					input = "-allowed_media_types video "
-				case queryAudio:
-					input = "-allowed_media_types audio "
-				}
-
-				input += strings.Replace(tpl["rtsp"], "{input}", s, 1)
-			default:
-				input = "-i " + s
-			}
-		} else if streams.Get(s) != nil {
-			s = "rtsp://localhost:" + rtsp.Port + "/" + s
-			switch {
-			case queryVideo && !queryAudio:
-				s += "?video"
-			case queryAudio && !queryVideo:
-				s += "?audio"
-			}
-			input = strings.Replace(tpl["rtsp"], "{input}", s, 1)
-		} else if strings.HasPrefix(s, "device?") {
-			var err error
-			input, err = device.GetInput(s)
-			if err != nil {
-				return nil, err
-			}
-		} else {
-			input = strings.Replace(tpl["file"], "{input}", s, 1)
-		}
-
-		if _, ok := query["async"]; ok {
-			input = "-use_wallclock_as_timestamps 1 -async 1 " + input
-		}
-
-		s = cmd + input
-
-		if query != nil {
-			for _, raw := range query["raw"] {
-				s += " " + raw
-			}
-
-			for _, rotate := range query["rotate"] {
-				switch rotate {
-				case "90":
-					s += " -vf transpose=1" // 90 degrees clockwise
-				case "180":
-					s += " -vf transpose=1,transpose=1"
-				case "-90", "270":
-					s += " -vf transpose=2" // 90 degrees counterclockwise
-				}
-				break
-			}
-
-			switch len(query["video"]) {
-			case 0:
-				s += " -vn"
-			case 1:
-				if len(query["audio"]) > 1 {
-					s += " -map 0:v:0?"
-				}
-				for _, video := range query["video"] {
-					if video == "copy" {
-						s += " -c:v copy"
-					} else {
-						s += " " + tpl[video]
-					}
-				}
-			default:
-				for i, video := range query["video"] {
-					if video == "copy" {
-						s += " -map 0:v:0? -c:v:" + strconv.Itoa(i) + " copy"
-					} else {
-						s += " -map 0:v:0? " + strings.ReplaceAll(tpl[video], ":v ", ":v:"+strconv.Itoa(i)+" ")
-					}
-				}
-			}
-
-			switch len(query["audio"]) {
-			case 0:
-				s += " -an"
-			case 1:
-				if len(query["video"]) > 1 {
-					s += " -map 0:a:0?"
-				}
-				for _, audio := range query["audio"] {
-					if audio == "copy" {
-						s += " -c:a copy"
-					} else {
-						s += " " + tpl[audio]
-					}
-				}
-			default:
-				for i, audio := range query["audio"] {
-					if audio == "copy" {
-						s += " -map 0:a:0? -c:a:" + strconv.Itoa(i) + " copy"
-					} else {
-						s += " -map 0:a:0? " + strings.ReplaceAll(tpl[audio], ":a ", ":a:"+strconv.Itoa(i)+" ")
-					}
-				}
-			}
-		} else {
-			s += " -c copy"
-		}
-
-		s += " " + tpl["output"]
-
-		return exec.Handle(s)
+		return exec.Handle("exec:" + args.String())
 	})
 
-	device.Bin = cfg.Mod["bin"]
+	device.Bin = defaults["bin"]
 	device.Init()
 }
 
+var defaults = map[string]string{
+	"bin":    "ffmpeg",
+	"global": "-hide_banner",
+
+	// inputs
+	"file": "-re -stream_loop -1 -i {input}",
+	"http": "-fflags nobuffer -flags low_delay -i {input}",
+	"rtsp": "-fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_transport tcp -i {input}",
+
+	// output
+	"output": "-user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}",
+
+	// `-preset superfast` - we can't use ultrafast because it doesn't support `-profile main -level 4.1`
+	// `-tune zerolatency` - for minimal latency
+	// `-profile high -level 4.1` - most used streaming profile
+	"h264":  "-c:v libx264 -g 50 -profile:v high -level:v 4.1 -preset:v superfast -tune:v zerolatency",
+	"h265":  "-c:v libx265 -g 50 -profile:v high -level:v 5.1 -preset:v superfast -tune:v zerolatency",
+	"mjpeg": "-c:v mjpeg -force_duplicated_matrix:v 1 -huffman:v 0 -pix_fmt:v yuvj420p",
+
+	"opus":       "-c:a libopus -ar:a 48000 -ac:a 2",
+	"pcmu":       "-c:a pcm_mulaw -ar:a 8000 -ac:a 1",
+	"pcmu/16000": "-c:a pcm_mulaw -ar:a 16000 -ac:a 1",
+	"pcmu/48000": "-c:a pcm_mulaw -ar:a 48000 -ac:a 1",
+	"pcma":       "-c:a pcm_alaw -ar:a 8000 -ac:a 1",
+	"pcma/16000": "-c:a pcm_alaw -ar:a 16000 -ac:a 1",
+	"pcma/48000": "-c:a pcm_alaw -ar:a 48000 -ac:a 1",
+	"aac":        "-c:a aac", // keep sample rate and channels
+	"aac/16000":  "-c:a aac -ar:a 16000 -ac:a 1",
+	"mp3":        "-c:a libmp3lame -q:a 8",
+
+	// hardware Intel and AMD on Linux
+	// better not to set `-async_depth:v 1` like for QSV, because framedrops
+	// `-bf 0` - disable B-frames is very important
+	"h264/vaapi":  "-c:v h264_vaapi -g 50 -bf 0 -profile:v high -level:v 4.1 -sei:v 0",
+	"h265/vaapi":  "-c:v hevc_vaapi -g 50 -bf 0 -profile:v high -level:v 5.1 -sei:v 0",
+	"mjpeg/vaapi": "-c:v mjpeg_vaapi",
+
+	// hardware Raspberry
+	"h264/v4l2m2m": "-c:v h264_v4l2m2m -g 50 -bf 0",
+	"h265/v4l2m2m": "-c:v hevc_v4l2m2m -g 50 -bf 0",
+
+	// hardware NVidia on Linux and Windows
+	// preset=p2 - faster, tune=ll - low latency
+	"h264/cuda": "-c:v h264_nvenc -g 50 -profile:v high -level:v auto -preset:v p2 -tune:v ll",
+	"h265/cuda": "-c:v hevc_nvenc -g 50 -profile:v high -level:v auto",
+
+	// hardware Intel on Windows
+	"h264/dxva2":  "-c:v h264_qsv -g 50 -bf 0 -profile:v high -level:v 4.1 -async_depth:v 1",
+	"h265/dxva2":  "-c:v hevc_qsv -g 50 -bf 0 -profile:v high -level:v 5.1 -async_depth:v 1",
+	"mjpeg/dxva2": "-c:v mjpeg_qsv -profile:v high -level:v 5.1",
+
+	// hardware macOS
+	"h264/videotoolbox": "-c:v h264_videotoolbox -g 50 -bf 0 -profile:v high -level:v 4.1",
+	"h265/videotoolbox": "-c:v hevc_videotoolbox -g 50 -bf 0 -profile:v high -level:v 5.1",
+}
+
+func parseArgs(s string) *Args {
+	// init FFmpeg arguments
+	args := &Args{
+		bin:    defaults["bin"],
+		global: defaults["global"],
+		output: defaults["output"],
+	}
+
+	var query url.Values
+	if i := strings.IndexByte(s, '#'); i > 0 {
+		query = parseQuery(s[i+1:])
+		args.video = len(query["video"])
+		args.audio = len(query["audio"])
+		s = s[:i]
+	}
+
+	// Parse input:
+	//   1. Input as xxxx:// link (http or rtsp or any other)
+	//   2. Input as stream name
+	//   3. Input as FFmpeg device (local USB camera)
+	if i := strings.Index(s, "://"); i > 0 {
+		switch s[:i] {
+		case "http", "https", "rtmp":
+			args.input = strings.Replace(defaults["http"], "{input}", s, 1)
+		case "rtsp", "rtsps":
+			// https://ffmpeg.org/ffmpeg-protocols.html#rtsp
+			// skip unnecessary input tracks
+			switch {
+			case (args.video > 0 && args.audio > 0) || (args.video == 0 && args.audio == 0):
+				args.input = "-allowed_media_types video+audio "
+			case args.video > 0:
+				args.input = "-allowed_media_types video "
+			case args.audio > 0:
+				args.input = "-allowed_media_types audio "
+			}
+
+			args.input += strings.Replace(defaults["rtsp"], "{input}", s, 1)
+		default:
+			args.input = "-i " + s
+		}
+	} else if streams.Get(s) != nil {
+		s = "rtsp://localhost:" + rtsp.Port + "/" + s
+		switch {
+		case args.video > 0 && args.audio == 0:
+			s += "?video"
+		case args.audio > 0 && args.video == 0:
+			s += "?audio"
+		default:
+			s += "?video&audio"
+		}
+		args.input = strings.Replace(defaults["rtsp"], "{input}", s, 1)
+	} else if strings.HasPrefix(s, "device?") {
+		var err error
+		args.input, err = device.GetInput(s)
+		if err != nil {
+			return nil
+		}
+	} else {
+		args.input = strings.Replace(defaults["file"], "{input}", s, 1)
+	}
+
+	if query["async"] != nil {
+		args.input = "-use_wallclock_as_timestamps 1 -async 1 " + args.input
+	}
+
+	// Parse query params:
+	//   1. `width`/`height` params
+	//   2. `rotate` param
+	//   3. `video` params (support multiple)
+	//   4. `audio` params (support multiple)
+	//   5. `hardware` param
+	if query != nil {
+		// 1. Process raw params for FFmpeg
+		for _, raw := range query["raw"] {
+			args.AddCodec(raw)
+		}
+
+		// 2. Process video filters (resize and rotation)
+		if query["width"] != nil || query["height"] != nil {
+			filter := "scale="
+			if query["width"] != nil {
+				filter += query["width"][0]
+			} else {
+				filter += "-1"
+			}
+			filter += ":"
+			if query["height"] != nil {
+				filter += query["height"][0]
+			} else {
+				filter += "-1"
+			}
+			args.AddFilter(filter)
+		}
+
+		if query["rotate"] != nil {
+			var filter string
+			switch query["rotate"][0] {
+			case "90":
+				filter = "transpose=1" // 90 degrees clockwise
+			case "180":
+				filter = "transpose=1,transpose=1"
+			case "-90", "270":
+				filter = "transpose=2" // 90 degrees counterclockwise
+			}
+			if filter != "" {
+				args.AddFilter(filter)
+			}
+		}
+
+		// 3. Process video codecs
+		if args.video > 0 {
+			for _, video := range query["video"] {
+				if video != "copy" {
+					args.AddCodec(defaults[video])
+				} else {
+					args.AddCodec("-c:v copy")
+				}
+			}
+		} else {
+			args.AddCodec("-vn")
+		}
+
+		// 4. Process audio codecs
+		if args.audio > 0 {
+			for _, audio := range query["audio"] {
+				if audio != "copy" {
+					args.AddCodec(defaults[audio])
+				} else {
+					args.AddCodec("-c:a copy")
+				}
+			}
+		} else {
+			args.AddCodec("-an")
+		}
+
+		if query["hardware"] != nil {
+			MakeHardware(args, query["hardware"][0])
+		}
+	}
+
+	if args.codecs == nil {
+		args.AddCodec("-c copy")
+	}
+
+	return args
+}
+
 func parseQuery(s string) map[string][]string {
 	query := map[string][]string{}
 	for _, key := range strings.Split(s, "#") {
@@ -213,3 +254,76 @@ func parseQuery(s string) map[string][]string {
 	}
 	return query
 }
+
+type Args struct {
+	bin     string   // ffmpeg
+	global  string   // -hide_banner -v error
+	input   string   // -re -stream_loop -1 -i /media/bunny.mp4
+	codecs  []string // -c:v libx264 -g:v 30 -preset:v ultrafast -tune:v zerolatency
+	filters []string // scale=1920:1080
+	output  string   // -f rtsp {output}
+
+	video, audio int // count of video and audio params
+}
+
+func (a *Args) AddCodec(codec string) {
+	a.codecs = append(a.codecs, codec)
+}
+
+func (a *Args) AddFilter(filter string) {
+	a.filters = append(a.filters, filter)
+}
+
+func (a *Args) InsertFilter(filter string) {
+	a.filters = append([]string{filter}, a.filters...)
+}
+
+func (a *Args) String() string {
+	b := bytes.NewBuffer(make([]byte, 0, 512))
+
+	b.WriteString(a.bin)
+
+	if a.global != "" {
+		b.WriteByte(' ')
+		b.WriteString(a.global)
+	}
+
+	b.WriteByte(' ')
+	b.WriteString(a.input)
+
+	multimode := a.video > 1 || a.audio > 1
+	var iv, ia int
+
+	for _, codec := range a.codecs {
+		// support multiple video and/or audio codecs
+		if multimode && len(codec) >= 5 {
+			switch codec[:5] {
+			case "-c:v ":
+				codec = "-map 0:v:0? " + strings.ReplaceAll(codec, ":v ", ":v:"+strconv.Itoa(iv)+" ")
+				iv++
+			case "-c:a ":
+				codec = "-map 0:a:0? " + strings.ReplaceAll(codec, ":a ", ":a:"+strconv.Itoa(ia)+" ")
+				ia++
+			}
+		}
+
+		b.WriteByte(' ')
+		b.WriteString(codec)
+	}
+
+	if a.filters != nil {
+		for i, filter := range a.filters {
+			if i == 0 {
+				b.WriteString(" -vf ")
+			} else {
+				b.WriteByte(',')
+			}
+			b.WriteString(filter)
+		}
+	}
+
+	b.WriteByte(' ')
+	b.WriteString(a.output)
+
+	return b.String()
+}

cmd/ffmpeg/hardware.go

@@ -0,0 +1,112 @@
+package ffmpeg
+
+import (
+	"github.com/rs/zerolog/log"
+	"os/exec"
+	"strings"
+)
+
+const (
+	EngineSoftware     = "software"
+	EngineVAAPI        = "vaapi"        // Intel iGPU and AMD GPU
+	EngineV4L2M2M      = "v4l2m2m"      // Raspberry Pi 3 and 4
+	EngineCUDA         = "cuda"         // NVidia on Windows and Linux
+	EngineDXVA2        = "dxva2"        // Intel on Windows
+	EngineVideoToolbox = "videotoolbox" // macOS
+)
+
+var cache = map[string]string{}
+
+// MakeHardware converts software FFmpeg args to hardware args
+// empty engine for autoselect
+func MakeHardware(args *Args, engine string) {
+	for i, codec := range args.codecs {
+		if len(codec) < 12 {
+			continue // skip short line (-c:v libx264...)
+		}
+
+		// get current codec name
+		name := cut(codec, ' ', 1)
+		switch name {
+		case "libx264":
+			name = "h264"
+		case "libx265":
+			name = "h265"
+		case "mjpeg":
+		default:
+			continue // skip unsupported codec
+		}
+
+		// temporary disable probe for H265 and MJPEG
+		if engine == "" && name == "h264" {
+			if engine = cache[name]; engine == "" {
+				engine = ProbeHardware(name)
+				cache[name] = engine
+			}
+		}
+
+		switch engine {
+		case EngineVAAPI:
+			args.input = "-hwaccel vaapi -hwaccel_output_format vaapi " + args.input
+			args.codecs[i] = defaults[name+"/"+engine]
+
+			for i, filter := range args.filters {
+				if strings.HasPrefix(filter, "scale=") {
+					args.filters[i] = "scale_vaapi=" + filter[6:]
+				}
+			}
+
+			// fix if input doesn't support hwaccel, do nothing when support
+			args.InsertFilter("format=vaapi|nv12,hwupload")
+
+		case EngineCUDA:
+			args.input = "-hwaccel cuda -hwaccel_output_format cuda -extra_hw_frames 2 " + args.input
+			args.codecs[i] = defaults[name+"/"+engine]
+
+			for i, filter := range args.filters {
+				if strings.HasPrefix(filter, "scale=") {
+					args.filters[i] = "scale_cuda=" + filter[6:]
+				}
+			}
+
+		case EngineDXVA2:
+			args.input = "-hwaccel dxva2 -hwaccel_output_format dxva2_vld " + args.input
+			args.codecs[i] = defaults[name+"/"+engine]
+
+			for i, filter := range args.filters {
+				if strings.HasPrefix(filter, "scale=") {
+					args.filters[i] = "scale_qsv=" + filter[6:]
+				}
+			}
+
+			args.InsertFilter("hwmap=derive_device=qsv,format=qsv")
+
+		case EngineVideoToolbox:
+			args.input = "-hwaccel videotoolbox -hwaccel_output_format videotoolbox_vld " + args.input
+			args.codecs[i] = defaults[name+"/"+engine]
+
+		case EngineV4L2M2M:
+			args.codecs[i] = defaults[name+"/"+engine]
+		}
+	}
+}
+
+func run(arg ...string) bool {
+	err := exec.Command(defaults["bin"], arg...).Run()
+	log.Printf("%v %v", arg, err)
+	return err == nil
+}
+
+func cut(s string, sep byte, pos int) string {
+	for n := 0; n < pos; n++ {
+		if i := strings.IndexByte(s, sep); i > 0 {
+			s = s[i+1:]
+		} else {
+			return ""
+		}
+	}
+	if i := strings.IndexByte(s, sep); i > 0 {
+		return s[:i]
+	}
+	return s
+}

cmd/ffmpeg/hardware_darwin.go

@@ -0,0 +1,21 @@
+package ffmpeg
+
+func ProbeHardware(name string) string {
+	switch name {
+	case "h264":
+		if run(
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "h264_videotoolbox", "-f", "null", "-") {
+			return EngineVideoToolbox
+		}
+
+	case "h265":
+		if run(
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "hevc_videotoolbox", "-f", "null", "-") {
+			return EngineVideoToolbox
+		}
+	}
+
+	return EngineSoftware
+}

cmd/ffmpeg/hardware_linux.go

@@ -0,0 +1,67 @@
+package ffmpeg
+
+import (
+	"runtime"
+)
+
+func ProbeHardware(name string) string {
+	if runtime.GOARCH == "arm64" || runtime.GOARCH == "arm" {
+		switch name {
+		case "h264":
+			if run(
+				"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+				"-c", "h264_v4l2m2m", "-f", "null", "-") {
+				return EngineV4L2M2M
+			}
+
+		case "h265":
+			if run(
+				"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+				"-c", "hevc_v4l2m2m", "-f", "null", "-") {
+				return EngineV4L2M2M
+			}
+		}
+
+		return EngineSoftware
+	}
+
+	switch name {
+	case "h264":
+		if run("-init_hw_device", "cuda",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "h264_nvenc", "-f", "null", "-") {
+			return EngineCUDA
+		}
+
+		if run("-init_hw_device", "vaapi",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-vf", "format=nv12,hwupload",
+			"-c", "h264_vaapi", "-f", "null", "-") {
+			return EngineVAAPI
+		}
+
+	case "h265":
+		if run("-init_hw_device", "cuda",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "hevc_nvenc", "-f", "null", "-") {
+			return EngineCUDA
+		}
+
+		if run("-init_hw_device", "vaapi",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-vf", "format=nv12,hwupload",
+			"-c", "hevc_vaapi", "-f", "null", "-") {
+			return EngineVAAPI
+		}
+
+	case "mjpeg":
+		if run("-init_hw_device", "vaapi",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-vf", "format=nv12,hwupload",
+			"-c", "mjpeg_vaapi", "-f", "null", "-") {
+			return EngineVAAPI
+		}
+	}
+
+	return EngineSoftware
+}

cmd/ffmpeg/hardware_windows.go

@@ -0,0 +1,40 @@
+package ffmpeg
+
+func ProbeHardware(name string) string {
+	switch name {
+	case "h264":
+		if run("-init_hw_device", "cuda",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "h264_nvenc", "-f", "null", "-") {
+			return EngineCUDA
+		}
+
+		if run("-init_hw_device", "dxva2",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "h264_qsv", "-f", "null", "-") {
+			return EngineDXVA2
+		}
+
+	case "h265":
+		if run("-init_hw_device", "cuda",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "hevc_nvenc", "-f", "null", "-") {
+			return EngineCUDA
+		}
+
+		if run("-init_hw_device", "dxva2",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "hevc_qsv", "-f", "null", "-") {
+			return EngineDXVA2
+		}
+
+	case "mjpeg":
+		if run("-init_hw_device", "dxva2",
+			"-f", "lavfi", "-i", "testsrc2", "-t", "1",
+			"-c", "mjpeg_qsv", "-f", "null", "-") {
+			return EngineDXVA2
+		}
+	}
+
+	return EngineSoftware
+}

cmd/hls/hls.go

@@ -0,0 +1,261 @@
+package hls
+
+import (
+	"fmt"
+	"github.com/AlexxIT/go2rtc/cmd/api"
+	"github.com/AlexxIT/go2rtc/cmd/streams"
+	"github.com/AlexxIT/go2rtc/pkg/mp4"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/AlexxIT/go2rtc/pkg/ts"
+	"github.com/rs/zerolog/log"
+	"net/http"
+	"strconv"
+	"sync"
+	"time"
+)
+
+func Init() {
+	api.HandleFunc("api/stream.m3u8", handlerStream)
+	api.HandleFunc("api/hls/playlist.m3u8", handlerPlaylist)
+
+	// HLS (TS)
+	api.HandleFunc("api/hls/segment.ts", handlerSegmentTS)
+
+	// HLS (fMP4)
+	api.HandleFunc("api/hls/init.mp4", handlerInit)
+	api.HandleFunc("api/hls/segment.m4s", handlerSegmentMP4)
+}
+
+type Consumer interface {
+	streamer.Consumer
+	Init() ([]byte, error)
+	MimeCodecs() string
+	Start()
+}
+
+type Session struct {
+	cons     Consumer
+	playlist string
+	init     []byte
+	segment  []byte
+	seq      int
+	alive    *time.Timer
+	mu       sync.Mutex
+}
+
+const keepalive = 5 * time.Second
+
+var sessions = map[string]*Session{}
+
+func handlerStream(w http.ResponseWriter, r *http.Request) {
+	// CORS important for Chromecast
+	w.Header().Set("Access-Control-Allow-Origin", "*")
+	w.Header().Set("Content-Type", "application/vnd.apple.mpegurl")
+
+	if r.Method == "OPTIONS" {
+		w.Header().Set("Access-Control-Allow-Methods", "GET")
+		return
+	}
+
+	src := r.URL.Query().Get("src")
+	stream := streams.GetOrNew(src)
+	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
+		return
+	}
+
+	var cons Consumer
+
+	// use fMP4 with codecs filter and TS without
+	medias := mp4.ParseQuery(r.URL.Query())
+	if medias != nil {
+		cons = &mp4.Consumer{
+			RemoteAddr: r.RemoteAddr,
+			UserAgent:  r.UserAgent(),
+			Medias:     medias,
+		}
+	} else {
+		cons = &ts.Consumer{
+			RemoteAddr: r.RemoteAddr,
+			UserAgent:  r.UserAgent(),
+		}
+	}
+
+	session := &Session{cons: cons}
+
+	cons.Listen(func(msg interface{}) {
+		if data, ok := msg.([]byte); ok {
+			session.mu.Lock()
+			session.segment = append(session.segment, data...)
+			session.mu.Unlock()
+		}
+	})
+
+	if err := stream.AddConsumer(cons); err != nil {
+		log.Error().Err(err).Caller().Send()
+		return
+	}
+
+	session.alive = time.AfterFunc(keepalive, func() {
+		stream.RemoveConsumer(cons)
+	})
+	session.init, _ = cons.Init()
+
+	cons.Start()
+
+	sid := strconv.FormatInt(time.Now().UnixNano(), 10)
+
+	// two segments important for Chromecast
+	if medias != nil {
+		session.playlist = `#EXTM3U
+#EXT-X-VERSION:6
+#EXT-X-TARGETDURATION:1
+#EXT-X-MEDIA-SEQUENCE:%d
+#EXT-X-MAP:URI="init.mp4?id=` + sid + `"
+#EXTINF:0.500,
+segment.m4s?id=` + sid + `&n=%d
+#EXTINF:0.500,
+segment.m4s?id=` + sid + `&n=%d`
+	} else {
+		session.playlist = `#EXTM3U
+#EXT-X-VERSION:3
+#EXT-X-TARGETDURATION:1
+#EXT-X-MEDIA-SEQUENCE:%d
+#EXTINF:0.500,
+segment.ts?id=` + sid + `&n=%d
+#EXTINF:0.500,
+segment.ts?id=` + sid + `&n=%d`
+	}
+
+	sessions[sid] = session
+
+	// bandwidth important for Safari, codecs useful for smooth playback
+	data := []byte(`#EXTM3U
+#EXT-X-STREAM-INF:BANDWIDTH=1000000,CODECS="` + cons.MimeCodecs() + `"
+hls/playlist.m3u8?id=` + sid)
+
+	if _, err := w.Write(data); err != nil {
+		log.Error().Err(err).Caller().Send()
+	}
+}
+
+func handlerPlaylist(w http.ResponseWriter, r *http.Request) {
+	w.Header().Set("Access-Control-Allow-Origin", "*")
+	w.Header().Set("Content-Type", "application/vnd.apple.mpegurl")
+
+	if r.Method == "OPTIONS" {
+		w.Header().Set("Access-Control-Allow-Methods", "GET")
+		return
+	}
+
+	sid := r.URL.Query().Get("id")
+	session := sessions[sid]
+	if session == nil {
+		http.NotFound(w, r)
+		return
+	}
+
+	s := fmt.Sprintf(session.playlist, session.seq, session.seq, session.seq+1)
+
+	if _, err := w.Write([]byte(s)); err != nil {
+		log.Error().Err(err).Caller().Send()
+	}
+}
+
+func handlerSegmentTS(w http.ResponseWriter, r *http.Request) {
+	w.Header().Set("Access-Control-Allow-Origin", "*")
+	w.Header().Set("Content-Type", "video/mp2t")
+
+	if r.Method == "OPTIONS" {
+		w.Header().Set("Access-Control-Allow-Methods", "GET")
+		return
+	}
+
+	sid := r.URL.Query().Get("id")
+	session := sessions[sid]
+	if session == nil {
+		http.NotFound(w, r)
+		return
+	}
+
+	session.alive.Reset(keepalive)
+
+	var i byte
+	for len(session.segment) == 0 {
+		if i++; i > 10 {
+			http.NotFound(w, r)
+			return
+		}
+		time.Sleep(time.Millisecond * 100)
+	}
+
+	session.mu.Lock()
+	data := session.segment
+	// important to start new segment with init
+	session.segment = session.init
+	session.seq++
+	session.mu.Unlock()
+
+	if _, err := w.Write(data); err != nil {
+		log.Error().Err(err).Caller().Send()
+	}
+}
+
+func handlerInit(w http.ResponseWriter, r *http.Request) {
+	w.Header().Set("Access-Control-Allow-Origin", "*")
+	w.Header().Add("Content-Type", "video/mp4")
+
+	if r.Method == "OPTIONS" {
+		w.Header().Set("Access-Control-Allow-Methods", "GET")
+		return
+	}
+
+	sid := r.URL.Query().Get("id")
+	session := sessions[sid]
+	if session == nil {
+		http.NotFound(w, r)
+		return
+	}
+
+	if _, err := w.Write(session.init); err != nil {
+		log.Error().Err(err).Caller().Send()
+	}
+}
+
+func handlerSegmentMP4(w http.ResponseWriter, r *http.Request) {
+	w.Header().Set("Access-Control-Allow-Origin", "*")
+	w.Header().Add("Content-Type", "video/iso.segment")
+
+	if r.Method == "OPTIONS" {
+		w.Header().Set("Access-Control-Allow-Methods", "GET, OPTIONS")
+		return
+	}
+
+	sid := r.URL.Query().Get("id")
+	session := sessions[sid]
+	if session == nil {
+		http.NotFound(w, r)
+		return
+	}
+
+	session.alive.Reset(keepalive)
+
+	var i byte
+	for len(session.segment) == 0 {
+		if i++; i > 10 {
+			http.NotFound(w, r)
+			return
+		}
+		time.Sleep(time.Millisecond * 100)
+	}
+
+	session.mu.Lock()
+	data := session.segment
+	session.segment = nil
+	session.seq++
+	session.mu.Unlock()
+
+	if _, err := w.Write(data); err != nil {
+		log.Error().Err(err).Caller().Send()
+	}
+}

cmd/mjpeg/mjpeg.go

@@ -27,7 +27,10 @@ func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
 
 	exit := make(chan []byte)
 
-	cons := &mjpeg.Consumer{}
+	cons := &mjpeg.Consumer{
+		RemoteAddr: r.RemoteAddr,
+		UserAgent:  r.UserAgent(),
+	}
 	cons.Listen(func(msg interface{}) {
 		switch msg := msg.(type) {
 		case []byte:
@@ -68,7 +71,10 @@ func handlerStream(w http.ResponseWriter, r *http.Request) {
 
 	flusher := w.(http.Flusher)
 
-	cons := &mjpeg.Consumer{}
+	cons := &mjpeg.Consumer{
+		RemoteAddr: r.RemoteAddr,
+		UserAgent:  r.UserAgent(),
+	}
 	cons.Listen(func(msg interface{}) {
 		switch msg := msg.(type) {
 		case []byte:
@@ -109,7 +115,10 @@ func handlerWS(tr *api.Transport, _ *api.Message) error {
 		return errors.New(api.StreamNotFound)
 	}
 
-	cons := &mjpeg.Consumer{}
+	cons := &mjpeg.Consumer{
+		RemoteAddr: tr.Request.RemoteAddr,
+		UserAgent:  tr.Request.UserAgent(),
+	}
 	cons.Listen(func(msg interface{}) {
 		if data, ok := msg.([]byte); ok {
 			tr.Write(data)
@@ -121,6 +130,8 @@ func handlerWS(tr *api.Transport, _ *api.Message) error {
 		return err
 	}
 
+	tr.Write(&api.Message{Type: "mjpeg"})
+
 	tr.OnClose(func() {
 		stream.RemoveConsumer(cons)
 	})

cmd/mp4/mp4.go

@@ -5,6 +5,7 @@ import (
 	"github.com/AlexxIT/go2rtc/cmd/app"
 	"github.com/AlexxIT/go2rtc/cmd/streams"
 	"github.com/AlexxIT/go2rtc/pkg/mp4"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/rs/zerolog"
 	"net/http"
 	"strconv"
@@ -25,8 +26,14 @@ func Init() {
 var log zerolog.Logger
 
 func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
-	if isChromeFirst(w, r) {
-		return
+	// Chrome 105 does two requests: without Range and with `Range: bytes=0-`
+	ua := r.UserAgent()
+	if strings.Contains(ua, " Chrome/") {
+		if r.Header.Values("Range") == nil {
+			w.Header().Set("Content-Type", "video/mp4")
+			w.WriteHeader(http.StatusOK)
+			return
+		}
 	}
 
 	src := r.URL.Query().Get("src")
@@ -67,7 +74,22 @@ func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
 func handlerMP4(w http.ResponseWriter, r *http.Request) {
 	log.Trace().Msgf("[mp4] %s %+v", r.Method, r.Header)
 
-	if isChromeFirst(w, r) || isSafari(w, r) {
+	// Chrome has Safari in UA, so check first Chrome and later Safari
+	ua := r.UserAgent()
+	if strings.Contains(ua, " Chrome/") {
+		if r.Header.Values("Range") == nil {
+			w.Header().Set("Content-Type", "video/mp4")
+			w.WriteHeader(http.StatusOK)
+			return
+		}
+	} else if strings.Contains(ua, " Safari/") {
+		// auto redirect to HLS/fMP4 format, because Safari not support MP4 stream
+		url := "stream.m3u8?" + r.URL.RawQuery
+		if !r.URL.Query().Has("mp4") {
+			url += "&mp4"
+		}
+
+		http.Redirect(w, r, url, http.StatusMovedPermanently)
 		return
 	}
 
@@ -80,7 +102,12 @@ func handlerMP4(w http.ResponseWriter, r *http.Request) {
 
 	exit := make(chan error)
 
-	cons := &mp4.Consumer{}
+	cons := &mp4.Consumer{
+		RemoteAddr: r.RemoteAddr,
+		UserAgent:  r.UserAgent(),
+		Medias:     streamer.ParseQuery(r.URL.Query()),
+	}
+
 	cons.Listen(func(msg interface{}) {
 		if data, ok := msg.([]byte); ok {
 			if _, err := w.Write(data); err != nil && exit != nil {
@@ -132,23 +159,3 @@ func handlerMP4(w http.ResponseWriter, r *http.Request) {
 		duration.Stop()
 	}
 }
-
-func isChromeFirst(w http.ResponseWriter, r *http.Request) bool {
-	// Chrome 105 does two requests: without Range and with `Range: bytes=0-`
-	if strings.Contains(r.UserAgent(), " Chrome/") {
-		if r.Header.Values("Range") == nil {
-			w.Header().Set("Content-Type", "video/mp4")
-			w.WriteHeader(http.StatusOK)
-			return true
-		}
-	}
-	return false
-}
-
-func isSafari(w http.ResponseWriter, r *http.Request) bool {
-	if r.Header.Get("Range") == "bytes=0-1" {
-		handlerKeyframe(w, r)
-		return true
-	}
-	return false
-}

cmd/mp4/ws.go

@@ -9,7 +9,7 @@ import (
 	"strings"
 )
 
-const packetSize = 8192
+const packetSize = 1400
 
 func handlerWSMSE(tr *api.Transport, msg *api.Message) error {
 	src := tr.Request.URL.Query().Get("src")
@@ -18,9 +18,10 @@ func handlerWSMSE(tr *api.Transport, msg *api.Message) error {
 		return errors.New(api.StreamNotFound)
 	}
 
-	cons := &mp4.Consumer{}
-	cons.UserAgent = tr.Request.UserAgent()
-	cons.RemoteAddr = tr.Request.RemoteAddr
+	cons := &mp4.Consumer{
+		RemoteAddr: tr.Request.RemoteAddr,
+		UserAgent:  tr.Request.UserAgent(),
+	}
 
 	if codecs, ok := msg.Value.(string); ok {
 		log.Trace().Str("codecs", codecs).Msgf("[mp4] new WS/MSE consumer")
@@ -38,7 +39,7 @@ func handlerWSMSE(tr *api.Transport, msg *api.Message) error {
 	})
 
 	if err := stream.AddConsumer(cons); err != nil {
-		log.Warn().Err(err).Caller().Send()
+		log.Debug().Err(err).Msg("[mp4] add consumer")
 		return err
 	}
 
@@ -68,7 +69,11 @@ func handlerWSMP4(tr *api.Transport, msg *api.Message) error {
 		return errors.New(api.StreamNotFound)
 	}
 
-	cons := &mp4.Segment{}
+	cons := &mp4.Segment{
+		RemoteAddr:   tr.Request.RemoteAddr,
+		UserAgent:    tr.Request.UserAgent(),
+		OnlyKeyframe: true,
+	}
 
 	if codecs, ok := msg.Value.(string); ok {
 		log.Trace().Str("codecs", codecs).Msgf("[mp4] new WS/MP4 consumer")
@@ -86,6 +91,8 @@ func handlerWSMP4(tr *api.Transport, msg *api.Message) error {
 		return err
 	}
 
+	tr.Write(&api.Message{Type: "mp4", Value: cons.MimeType})
+
 	tr.OnClose(func() {
 		stream.RemoveConsumer(cons)
 	})
@@ -99,15 +106,18 @@ func parseMedias(codecs string, parseAudio bool) (medias []*streamer.Media) {
 
 	for _, name := range strings.Split(codecs, ",") {
 		switch name {
-		case "avc1.640029":
+		case mp4.MimeH264:
 			codec := &streamer.Codec{Name: streamer.CodecH264}
 			videos = append(videos, codec)
-		case "hvc1.1.6.L153.B0":
+		case mp4.MimeH265:
 			codec := &streamer.Codec{Name: streamer.CodecH265}
 			videos = append(videos, codec)
-		case "mp4a.40.2":
+		case mp4.MimeAAC:
 			codec := &streamer.Codec{Name: streamer.CodecAAC}
 			audios = append(audios, codec)
+		case mp4.MimeOpus:
+			codec := &streamer.Codec{Name: streamer.CodecOpus}
+			audios = append(audios, codec)
 		}
 	}
 

cmd/rtsp/rtsp.go

@@ -3,27 +3,32 @@ package rtsp
 import (
 	"github.com/AlexxIT/go2rtc/cmd/app"
 	"github.com/AlexxIT/go2rtc/cmd/streams"
+	"github.com/AlexxIT/go2rtc/pkg/mp4"
 	"github.com/AlexxIT/go2rtc/pkg/rtsp"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/rs/zerolog"
 	"net"
+	"net/url"
 	"strings"
 )
 
 func Init() {
 	var conf struct {
 		Mod struct {
-			Listen   string `yaml:"listen"`
-			Username string `yaml:"username"`
-			Password string `yaml:"password"`
+			Listen       string `yaml:"listen" json:"listen"`
+			Username     string `yaml:"username" json:"-"`
+			Password     string `yaml:"password" json:"-"`
+			DefaultQuery string `yaml:"default_query" json:"default_query"`
 		} `yaml:"rtsp"`
 	}
 
 	// default config
 	conf.Mod.Listen = ":8554"
+	conf.Mod.DefaultQuery = "video&audio"
 
 	app.LoadConfig(&conf)
+	app.Info["rtsp"] = conf.Mod
 
 	log = app.GetLogger("rtsp")
 
@@ -48,6 +53,10 @@ func Init() {
 
 	log.Info().Str("addr", address).Msg("[rtsp] listen")
 
+	if query, err := url.ParseQuery(conf.Mod.DefaultQuery); err == nil {
+		defaultMedias = mp4.ParseQuery(query)
+	}
+
 	go func() {
 		for {
 			conn, err := ln.Accept()
@@ -77,6 +86,7 @@ var Port string
 
 var log zerolog.Logger
 var handlers []Handler
+var defaultMedias []*streamer.Media
 
 func rtspHandler(url string) (streamer.Producer, error) {
 	backchannel := true
@@ -102,6 +112,8 @@ func rtspHandler(url string) (streamer.Producer, error) {
 				log.Trace().Msgf("[rtsp] client request:\n%s", msg)
 			case *tcp.Response:
 				log.Trace().Msgf("[rtsp] client response:\n%s", msg)
+			case string:
+				log.Trace().Msgf("[rtsp] client msg: %s", msg)
 			}
 		})
 	}
@@ -161,7 +173,14 @@ func tcpHandler(conn *rtsp.Conn) {
 
 			log.Debug().Str("stream", name).Msg("[rtsp] new consumer")
 
-			initMedias(conn)
+			conn.SessionName = app.UserAgent
+
+			conn.Medias = mp4.ParseQuery(conn.URL.Query())
+			if conn.Medias == nil {
+				for _, media := range defaultMedias {
+					conn.Medias = append(conn.Medias, media.Clone())
+				}
+			}
 
 			if err := stream.AddConsumer(conn); err != nil {
 				log.Warn().Err(err).Str("stream", name).Msg("[rtsp]")
@@ -200,6 +219,9 @@ func tcpHandler(conn *rtsp.Conn) {
 
 	if err := conn.Accept(); err != nil {
 		log.Warn().Err(err).Caller().Send()
+		if closer != nil {
+			closer()
+		}
 		_ = conn.Close()
 		return
 	}
@@ -212,7 +234,7 @@ func tcpHandler(conn *rtsp.Conn) {
 
 	if closer != nil {
 		if err := conn.Handle(); err != nil {
-			log.Debug().Err(err).Caller().Send()
+			log.Debug().Msgf("[rtsp] handle=%s", err)
 		}
 
 		closer()
@@ -222,45 +244,3 @@ func tcpHandler(conn *rtsp.Conn) {
 
 	_ = conn.Close()
 }
-
-func initMedias(conn *rtsp.Conn) {
-	// set media candidates from query list
-	for key, value := range conn.URL.Query() {
-		switch key {
-		case streamer.KindVideo, streamer.KindAudio:
-			for _, name := range value {
-				name = strings.ToUpper(name)
-
-				// check aliases
-				switch name {
-				case "COPY":
-					name = "" // pass empty codecs list
-				case "MJPEG":
-					name = streamer.CodecJPEG
-				case "AAC":
-					name = streamer.CodecAAC
-				}
-
-				media := &streamer.Media{
-					Kind: key, Direction: streamer.DirectionRecvonly,
-				}
-
-				// empty codecs match all codecs
-				if name != "" {
-					// empty clock rate and channels match any values
-					media.Codecs = []*streamer.Codec{{Name: name}}
-				}
-
-				conn.Medias = append(conn.Medias, media)
-			}
-		}
-	}
-
-	// set default media candidates if query is empty
-	if conn.Medias == nil {
-		conn.Medias = []*streamer.Media{
-			{Kind: streamer.KindVideo, Direction: streamer.DirectionRecvonly},
-			{Kind: streamer.KindAudio, Direction: streamer.DirectionRecvonly},
-		}
-	}
-}

cmd/streams/consumer.go

@@ -0,0 +1,15 @@
+package streams
+
+import (
+	"encoding/json"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+)
+
+type Consumer struct {
+	element streamer.Consumer
+	tracks  []*streamer.Track
+}
+
+func (c *Consumer) MarshalJSON() ([]byte, error) {
+	return json.Marshal(c.element)
+}

cmd/streams/producer.go

@@ -1,6 +1,7 @@
 package streams
 
 import (
+	"encoding/json"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"strings"
 	"sync"
@@ -24,11 +25,12 @@ type Producer struct {
 	template string
 
 	element streamer.Producer
+	lastErr error
 	tracks  []*streamer.Track
 
-	state   state
-	mu      sync.Mutex
-	restart *time.Timer
+	state    state
+	mu       sync.Mutex
+	workerID int
 }
 
 func (p *Producer) SetSource(s string) {
@@ -45,16 +47,20 @@ func (p *Producer) GetMedias() []*streamer.Media {
 	if p.state == stateNone {
 		log.Debug().Msgf("[streams] probe producer url=%s", p.url)
 
-		var err error
-		p.element, err = GetProducer(p.url)
-		if err != nil || p.element == nil {
-			log.Error().Err(err).Caller().Send()
+		p.element, p.lastErr = GetProducer(p.url)
+		if p.lastErr != nil || p.element == nil {
+			log.Error().Err(p.lastErr).Str("url", p.url).Caller().Send()
 			return nil
 		}
 
 		p.state = stateMedias
 	}
 
+	// if element in reconnect state
+	if p.element == nil {
+		return nil
+	}
+
 	return p.element.GetMedias()
 }
 
@@ -86,6 +92,15 @@ func (p *Producer) GetTrack(media *streamer.Media, codec *streamer.Codec) *strea
 	return track
 }
 
+func (p *Producer) MarshalJSON() ([]byte, error) {
+	if p.element != nil {
+		return json.Marshal(p.element)
+	}
+
+	info := streamer.Info{URL: p.url}
+	return json.Marshal(info)
+}
+
 // internals
 
 func (p *Producer) start() {
@@ -99,32 +114,45 @@ func (p *Producer) start() {
 	log.Debug().Msgf("[streams] start producer url=%s", p.url)
 
 	p.state = stateStart
-	go func() {
-		// safe read element while mu locked
-		if err := p.element.Start(); err != nil {
-			log.Warn().Err(err).Caller().Send()
-		}
-		p.reconnect()
-	}()
+	p.workerID++
+
+	go p.worker(p.element, p.workerID)
 }
 
-func (p *Producer) reconnect() {
+func (p *Producer) worker(element streamer.Producer, workerID int) {
+	if err := element.Start(); err != nil {
+		p.mu.Lock()
+		closed := p.workerID != workerID
+		p.mu.Unlock()
+
+		if closed {
+			return
+		}
+
+		log.Warn().Err(err).Str("url", p.url).Caller().Send()
+	}
+
+	p.reconnect(workerID)
+}
+
+func (p *Producer) reconnect(workerID int) {
 	p.mu.Lock()
 	defer p.mu.Unlock()
 
-	if p.state != stateStart {
+	if p.workerID != workerID {
 		log.Trace().Msgf("[streams] stop reconnect url=%s", p.url)
 		return
 	}
 
 	log.Debug().Msgf("[streams] reconnect to url=%s", p.url)
 
-	var err error
-	p.element, err = GetProducer(p.url)
-	if err != nil || p.element == nil {
-		log.Debug().Err(err).Caller().Send()
+	p.element, p.lastErr = GetProducer(p.url)
+	if p.lastErr != nil || p.element == nil {
+		log.Debug().Msgf("[streams] producer=%s", p.lastErr)
 		// TODO: dynamic timeout
-		p.restart = time.AfterFunc(30*time.Second, p.reconnect)
+		time.AfterFunc(30*time.Second, func() {
+			p.reconnect(workerID)
+		})
 		return
 	}
 
@@ -148,12 +176,7 @@ func (p *Producer) reconnect() {
 		}
 	}
 
-	go func() {
-		if err = p.element.Start(); err != nil {
-			log.Debug().Err(err).Caller().Send()
-		}
-		p.reconnect()
-	}()
+	go p.worker(p.element, workerID)
 }
 
 func (p *Producer) stop() {
@@ -167,6 +190,8 @@ func (p *Producer) stop() {
 	case stateNone:
 		log.Debug().Msgf("[streams] can't stop none producer")
 		return
+	case stateStart:
+		p.workerID++
 	}
 
 	log.Debug().Msgf("[streams] stop producer url=%s", p.url)
@@ -175,10 +200,6 @@ func (p *Producer) stop() {
 		_ = p.element.Stop()
 		p.element = nil
 	}
-	if p.restart != nil {
-		p.restart.Stop()
-		p.restart = nil
-	}
 
 	p.state = stateNone
 	p.tracks = nil

cmd/streams/stream.go

@@ -7,17 +7,14 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"strings"
 	"sync"
+	"sync/atomic"
 )
 
-type Consumer struct {
-	element streamer.Consumer
-	tracks  []*streamer.Track
-}
-
 type Stream struct {
 	producers []*Producer
 	consumers []*Consumer
 	mu        sync.Mutex
+	requests  int32
 }
 
 func NewStream(source interface{}) *Stream {
@@ -52,6 +49,9 @@ func (s *Stream) SetSource(source string) {
 }
 
 func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
+	// support for multiple simultaneous requests from different consumers
+	atomic.AddInt32(&s.requests, 1)
+
 	ic := len(s.consumers)
 
 	consumer := &Consumer{element: cons}
@@ -82,7 +82,7 @@ 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")
+						log.Warn().Str("url", prod.url).Msg("[streams] can't get track")
 						continue
 					}
 
@@ -91,19 +91,30 @@ func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
 
 					consumer.tracks = append(consumer.tracks, consTrack)
 					producers = append(producers, prod)
-					break producers
+					if !consMedia.MatchAll() {
+						break producers
+					}
 				}
 			}
 		}
 	}
 
-	if len(producers) == 0 {
+	if atomic.AddInt32(&s.requests, -1) == 0 {
 		s.stopProducers()
+	}
+
+	if len(producers) == 0 {
 		if len(codecs) > 0 {
 			return errors.New("codecs not match: " + codecs)
-		} else {
-			return fmt.Errorf("sources unavailable: %d", len(s.producers))
 		}
+
+		for i, producer := range s.producers {
+			if producer.lastErr != nil {
+				return fmt.Errorf("source %d error: %w", i, producer.lastErr)
+			}
+		}
+
+		return fmt.Errorf("sources unavailable: %d", len(s.producers))
 	}
 
 	s.mu.Lock()
@@ -183,22 +194,21 @@ producers:
 //}
 
 func (s *Stream) MarshalJSON() ([]byte, error) {
-	var v []interface{}
-	s.mu.Lock()
-	for _, prod := range s.producers {
-		if prod.element != nil {
-			v = append(v, prod.element)
-		}
+	if !s.mu.TryLock() {
+		log.Warn().Msgf("[streams] json locked")
+		return json.Marshal(nil)
 	}
-	for _, cons := range s.consumers {
-		// cons.element always not nil
-		v = append(v, cons.element)
+
+	var info struct {
+		Producers []*Producer `json:"producers"`
+		Consumers []*Consumer `json:"consumers"`
 	}
+	info.Producers = s.producers
+	info.Consumers = s.consumers
+
 	s.mu.Unlock()
-	if len(v) == 0 {
-		v = nil
-	}
-	return json.Marshal(v)
+
+	return json.Marshal(info)
 }
 
 func (s *Stream) removeConsumer(i int) {
@@ -228,6 +238,10 @@ func (s *Stream) removeProducer(i int) {
 }
 
 func collectCodecs(media *streamer.Media, codecs *string) {
+	if media.Direction == streamer.DirectionRecvonly {
+		return
+	}
+
 	for _, codec := range media.Codecs {
 		name := codec.Name
 		if name == streamer.CodecAAC {

cmd/streams/streams.go

@@ -1,9 +1,12 @@
 package streams
 
 import (
+	"encoding/json"
+	"github.com/AlexxIT/go2rtc/cmd/api"
 	"github.com/AlexxIT/go2rtc/cmd/app"
 	"github.com/AlexxIT/go2rtc/cmd/app/store"
 	"github.com/rs/zerolog"
+	"net/http"
 )
 
 func Init() {
@@ -22,6 +25,8 @@ func Init() {
 	for name, item := range store.GetDict("streams") {
 		streams[name] = NewStream(item)
 	}
+
+	api.HandleFunc("api/streams", streamsHandler)
 }
 
 func Get(name string) *Stream {
@@ -48,19 +53,29 @@ func GetOrNew(src string) *Stream {
 	return New(src, src)
 }
 
-func Delete(name string) {
-	delete(streams, name)
-}
+func streamsHandler(w http.ResponseWriter, r *http.Request) {
+	src := r.URL.Query().Get("src")
 
-func All() map[string]interface{} {
-	all := map[string]interface{}{}
-	for name, stream := range streams {
-		all[name] = stream
-		//if stream.Active() {
-		//	all[name] = stream
-		//}
+	switch r.Method {
+	case "PUT":
+		name := r.URL.Query().Get("name")
+		if name == "" {
+			name = src
+		}
+		New(name, src)
+		return
+	case "DELETE":
+		delete(streams, src)
+		return
+	}
+
+	if src != "" {
+		e := json.NewEncoder(w)
+		e.SetIndent("", "  ")
+		_ = e.Encode(streams[src])
+	} else {
+		_ = json.NewEncoder(w).Encode(streams)
 	}
-	return all
 }
 
 var log zerolog.Logger

cmd/webrtc/candidates.go

@@ -7,6 +7,7 @@ import (
 )
 
 var candidates []string
+var networks = []string{"udp", "tcp"}
 
 func AddCandidate(address string) {
 	candidates = append(candidates, address)
@@ -20,15 +21,17 @@ func asyncCandidates(tr *api.Transport) {
 			continue
 		}
 
-		cand, err := webrtc.NewCandidate(address)
-		if err != nil {
-			log.Warn().Err(err).Caller().Send()
-			continue
+		for _, network := range networks {
+			cand, err := webrtc.NewCandidate(network, address)
+			if err != nil {
+				log.Warn().Err(err).Caller().Send()
+				continue
+			}
+
+			log.Trace().Str("candidate", cand).Msg("[webrtc] config")
+
+			tr.Write(&api.Message{Type: "webrtc/candidate", Value: cand})
 		}
-
-		log.Trace().Str("candidate", cand).Msg("[webrtc] config")
-
-		tr.Write(&api.Message{Type: "webrtc/candidate", Value: cand})
 	}
 }
 
@@ -57,13 +60,15 @@ func syncCanditates(answer string) (string, error) {
 			continue
 		}
 
-		cand, err := webrtc.NewCandidate(address)
-		if err != nil {
-			log.Warn().Err(err).Msg("[webrtc] candidate")
-			continue
-		}
+		for _, network := range networks {
+			cand, err := webrtc.NewCandidate(network, address)
+			if err != nil {
+				log.Warn().Err(err).Msg("[webrtc] candidate")
+				continue
+			}
 
-		md.WithPropertyAttribute(cand)
+			md.WithPropertyAttribute(cand)
+		}
 	}
 
 	if end {

cmd/webrtc/webrtc.go

@@ -22,6 +22,7 @@ func Init() {
 		} `yaml:"webrtc"`
 	}
 
+	cfg.Mod.Listen = ":8555"
 	cfg.Mod.IceServers = []pion.ICEServer{
 		{URLs: []string{"stun:stun.l.google.com:19302"}},
 	}
@@ -68,7 +69,7 @@ var NewPConn func() (*pion.PeerConnection, error)
 
 func asyncHandler(tr *api.Transport, msg *api.Message) error {
 	src := tr.Request.URL.Query().Get("src")
-	stream := streams.Get(src)
+	stream := streams.GetOrNew(src)
 	if stream == nil {
 		return errors.New(api.StreamNotFound)
 	}
@@ -112,7 +113,7 @@ func asyncHandler(tr *api.Transport, msg *api.Message) error {
 
 	// 2. AddConsumer, so we get new tracks
 	if err = stream.AddConsumer(conn); err != nil {
-		log.Warn().Err(err).Caller().Send()
+		log.Debug().Err(err).Msg("[webrtc] add consumer")
 		_ = conn.Conn.Close()
 		return err
 	}

hardware.Dockerfile

@@ -0,0 +1,52 @@
+# 0. Prepare images
+# only debian 12 (bookworm) has latest ffmpeg
+ARG DEBIAN_VERSION="bookworm-slim"
+ARG GO_VERSION="1.19-buster"
+ARG NGROK_VERSION="3"
+
+FROM debian:${DEBIAN_VERSION} AS base
+FROM golang:${GO_VERSION} AS go
+FROM ngrok/ngrok:${NGROK_VERSION} AS ngrok
+
+
+# 1. Build go2rtc binary
+FROM go AS build
+
+WORKDIR /build
+
+# Cache dependencies
+COPY go.mod go.sum ./
+RUN go mod download
+
+COPY . .
+RUN CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
+
+
+# 2. Collect all files
+FROM scratch AS rootfs
+
+COPY --from=build /build/go2rtc /usr/local/bin/
+COPY --from=ngrok /bin/ngrok /usr/local/bin/
+COPY ./build/docker/run.sh /
+
+
+# 3. Final image
+FROM base
+
+# Install ffmpeg, bash (for run.sh), tini (for signal handling),
+# and other common tools for the echo source.
+# non-free for Intel QSV support (not used by go2rtc, just for tests)
+RUN echo 'deb http://deb.debian.org/debian bookworm non-free' > /etc/apt/sources.list.d/debian-non-free.list && \
+    apt-get -y update && apt-get -y install tini ffmpeg python3 curl jq intel-media-va-driver-non-free
+
+COPY --from=rootfs / /
+
+RUN chmod a+x /run.sh && mkdir -p /config
+
+ENTRYPOINT ["/usr/bin/tini", "--"]
+
+# https://github.com/NVIDIA/nvidia-docker/wiki/Installation-(Native-GPU-Support)
+ENV NVIDIA_VISIBLE_DEVICES all
+ENV NVIDIA_DRIVER_CAPABILITIES compute,video,utility
+
+CMD ["/run.sh"]

main.go

@@ -8,6 +8,7 @@ import (
 	"github.com/AlexxIT/go2rtc/cmd/exec"
 	"github.com/AlexxIT/go2rtc/cmd/ffmpeg"
 	"github.com/AlexxIT/go2rtc/cmd/hass"
+	"github.com/AlexxIT/go2rtc/cmd/hls"
 	"github.com/AlexxIT/go2rtc/cmd/homekit"
 	"github.com/AlexxIT/go2rtc/cmd/http"
 	"github.com/AlexxIT/go2rtc/cmd/ivideon"
@@ -26,27 +27,25 @@ import (
 
 func main() {
 	app.Init()     // init config and logs
+	api.Init()     // init HTTP API server
 	streams.Init() // load streams list
 
-	api.Init() // init HTTP API server
-
-	echo.Init()
-
 	rtsp.Init()   // add support RTSP client and RTSP server
 	rtmp.Init()   // add support RTMP client
 	exec.Init()   // add support exec scheme (depends on RTSP server)
 	ffmpeg.Init() // add support ffmpeg scheme (depends on exec scheme)
 	hass.Init()   // add support hass scheme
-
-	webrtc.Init()
-	mp4.Init()
-	mjpeg.Init()
-	http.Init()
+	echo.Init()
+	ivideon.Init()
 
 	srtp.Init()
 	homekit.Init()
 
-	ivideon.Init()
+	webrtc.Init()
+	mp4.Init()
+	hls.Init()
+	mjpeg.Init()
+	http.Init()
 
 	ngrok.Init()
 	debug.Init()

pkg/aac/rtp.go

@@ -17,9 +17,14 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 
 			//log.Printf("[RTP/AAC] units: %d, size: %4d, ts: %10d, %t", headersSize/2, len(packet.Payload), packet.Timestamp, packet.Marker)
 
+			data := packet.Payload[2+headersSize:]
+			if IsADTS(data) {
+				data = data[7:]
+			}
+
 			clone := *packet
 			clone.Version = RTPPacketVersionAAC
-			clone.Payload = packet.Payload[2+headersSize:]
+			clone.Payload = data
 			return push(&clone)
 		}
 	}
@@ -55,3 +60,7 @@ func RTPPay(mtu uint16) streamer.WrapperFunc {
 		}
 	}
 }
+
+func IsADTS(b []byte) bool {
+	return len(b) > 7 && b[0] == 0xFF && b[1]&0xF0 == 0xF0
+}

pkg/h264/helper.go

@@ -3,6 +3,8 @@ package h264
 import (
 	"encoding/base64"
 	"encoding/binary"
+	"encoding/hex"
+	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"strings"
 )
@@ -47,11 +49,39 @@ func Join(ps, iframe []byte) []byte {
 	return b
 }
 
+// GetProfileLevelID - get profile from fmtp line
+// Some devices won't play video with high level, so limit max profile and max level.
+// And return some profile even if fmtp line is empty.
 func GetProfileLevelID(fmtp string) string {
-	if fmtp == "" {
-		return ""
+	// avc1.640029 - H.264 high 4.1 (Chromecast 1st and 2nd Gen)
+	profile := byte(0x64)
+	capab := byte(0)
+	level := byte(0x29)
+
+	if fmtp != "" {
+		var conf []byte
+		// some cameras has wrong profile-level-id
+		// https://github.com/AlexxIT/go2rtc/issues/155
+		if s := streamer.Between(fmtp, "sprop-parameter-sets=", ","); s != "" {
+			if sps, _ := base64.StdEncoding.DecodeString(s); len(sps) >= 4 {
+				conf = sps[1:4]
+			}
+		} else if s = streamer.Between(fmtp, "profile-level-id=", ";"); s != "" {
+			conf, _ = hex.DecodeString(s)
+		}
+
+		if conf != nil {
+			if conf[0] < profile {
+				profile = conf[0]
+				capab = conf[1]
+			}
+			if conf[2] < level {
+				level = conf[2]
+			}
+		}
 	}
-	return streamer.Between(fmtp, "profile-level-id=", ";")
+
+	return fmt.Sprintf("%02X%02X%02X", profile, capab, level)
 }
 
 func GetParameterSet(fmtp string) (sps, pps []byte) {

pkg/h264/rtp.go

@@ -1,6 +1,7 @@
 package h264
 
 import (
+	"bytes"
 	"encoding/binary"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/rtp"
@@ -9,6 +10,8 @@ import (
 
 const RTPPacketVersionAVC = 0
 
+const PSMaxSize = 128 // the biggest SPS I've seen is 48 (EZVIZ CS-CV210)
+
 func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 	depack := &codecs.H264Packet{IsAVC: true}
 
@@ -27,35 +30,40 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 			}
 
 			// Fix TP-Link Tapo TC70: sends SPS and PPS with packet.Marker = true
-			if packet.Marker {
+			// Reolink Duo 2: sends SPS with Marker and PPS without
+			if packet.Marker && len(payload) < PSMaxSize {
 				switch NALUType(payload) {
 				case NALUTypeSPS, NALUTypePPS:
 					buf = append(buf, payload...)
 					return nil
+				case NALUTypeSEI:
+					// RtspServer https://github.com/AlexxIT/go2rtc/issues/244
+					// sends, marked SPS, marked PPS, marked SEI, marked IFrame
+					return nil
 				}
 			}
 
 			if len(buf) == 0 {
-				// Amcrest IP4M-1051: 9, 7, 8, 6, 28...
-				// Amcrest IP4M-1051: 9, 6, 1
-				switch NALUType(payload) {
-				case NALUTypeIFrame:
-					// fix IFrame without SPS,PPS
-					buf = append(buf, ps...)
-				case NALUTypeSEI, NALUTypeAUD:
-					// fix ffmpeg with transcoding first frame
-					i := int(4 + binary.BigEndian.Uint32(payload))
-
-					// check if only one NAL (fix ffmpeg transcoding for Reolink RLC-510A)
-					if i == len(payload) {
-						return nil
-					}
-
-					payload = payload[i:]
-
-					if NALUType(payload) == NALUTypeIFrame {
+				for {
+					// Amcrest IP4M-1051: 9, 7, 8, 6, 28...
+					// Amcrest IP4M-1051: 9, 6, 1
+					switch NALUType(payload) {
+					case NALUTypeIFrame:
+						// fix IFrame without SPS,PPS
 						buf = append(buf, ps...)
+					case NALUTypeSEI, NALUTypeAUD:
+						// fix ffmpeg with transcoding first frame
+						i := int(4 + binary.BigEndian.Uint32(payload))
+
+						// check if only one NAL (fix ffmpeg transcoding for Reolink RLC-510A)
+						if i == len(payload) {
+							return nil
+						}
+
+						payload = payload[i:]
+						continue
 					}
+					break
 				}
 			}
 
@@ -70,7 +78,31 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 				buf = buf[:0]
 			}
 
-			//log.Printf("[AVC] %v, len: %d", Types(payload), len(payload))
+			// should not be that huge SPS
+			if NALUType(payload) == NALUTypeSPS && binary.BigEndian.Uint32(payload) >= PSMaxSize {
+				// some Chinese buggy cameras has single packet with SPS+PPS+IFrame separated by 00 00 00 01
+				// https://github.com/AlexxIT/WebRTC/issues/391
+				// https://github.com/AlexxIT/WebRTC/issues/392
+				for i := 0; i < len(payload); {
+					if i+4 >= len(payload) {
+						break
+					}
+
+					size := bytes.Index(payload[i+4:], []byte{0, 0, 0, 1})
+					if size < 0 {
+						if i == 0 {
+							break
+						}
+						size = len(payload) - (i + 4)
+					}
+
+					binary.BigEndian.PutUint32(payload[i:], uint32(size))
+
+					i += size + 4
+				}
+			}
+
+			//log.Printf("[AVC] %v, len: %d, ts: %10d, seq: %d", Types(payload), len(payload), packet.Timestamp, packet.SequenceNumber)
 
 			clone := *packet
 			clone.Version = RTPPacketVersionAVC

pkg/h265/helper.go

@@ -7,14 +7,16 @@ import (
 )
 
 const (
-	NALUTypePFrame  = 1
-	NALUTypeIFrame  = 19
-	NALUTypeIFrame2 = 20
-	NALUTypeIFrame3 = 21
-	NALUTypeVPS     = 32
-	NALUTypeSPS     = 33
-	NALUTypePPS     = 34
-	NALUTypeFU      = 49
+	NALUTypePFrame    = 1
+	NALUTypeIFrame    = 19
+	NALUTypeIFrame2   = 20
+	NALUTypeIFrame3   = 21
+	NALUTypeVPS       = 32
+	NALUTypeSPS       = 33
+	NALUTypePPS       = 34
+	NALUTypePrefixSEI = 39
+	NALUTypeSuffixSEI = 40
+	NALUTypeFU        = 49
 )
 
 func NALUType(b []byte) byte {

pkg/h265/rtp.go

@@ -20,6 +20,16 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 			nuType := (data[0] >> 1) & 0x3F
 			//log.Printf("[RTP] codec: %s, nalu: %2d, size: %6d, ts: %10d, pt: %2d, ssrc: %d, seq: %d, %v", track.Codec.Name, nuType, len(packet.Payload), packet.Timestamp, packet.PayloadType, packet.SSRC, packet.SequenceNumber, packet.Marker)
 
+			// Fix for RtspServer https://github.com/AlexxIT/go2rtc/issues/244
+			if packet.Marker && len(data) < h264.PSMaxSize {
+				switch nuType {
+				case NALUTypeVPS, NALUTypeSPS, NALUTypePPS:
+					packet.Marker = false
+				case NALUTypePrefixSEI, NALUTypeSuffixSEI:
+					return nil
+				}
+			}
+
 			if nuType == NALUTypeFU {
 				switch data[2] >> 6 {
 				case 2: // begin

pkg/homekit/client.go

@@ -1,6 +1,7 @@
 package homekit
 
 import (
+	"encoding/json"
 	"errors"
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/hap"
@@ -11,6 +12,7 @@ import (
 	"github.com/brutella/hap/rtp"
 	"net"
 	"net/url"
+	"sync/atomic"
 )
 
 type Client struct {
@@ -263,3 +265,19 @@ func (c *Client) getMedias() []*streamer.Media {
 
 	return medias
 }
+
+func (c *Client) MarshalJSON() ([]byte, error) {
+	var recv uint32
+	for _, session := range c.sessions {
+		recv += atomic.LoadUint32(&session.Recv)
+	}
+
+	info := &streamer.Info{
+		Type:   "HomeKit source",
+		URL:    c.conn.URL(),
+		Medias: c.medias,
+		Tracks: c.tracks,
+		Recv:   recv,
+	}
+	return json.Marshal(info)
+}

pkg/httpflv/amf0.go

@@ -0,0 +1,165 @@
+package httpflv
+
+import (
+	"encoding/binary"
+	"errors"
+	"math"
+)
+
+const (
+	TypeNumber byte = iota
+	TypeBoolean
+	TypeString
+	TypeObject
+	TypeEcmaArray = 8
+	TypeObjectEnd = 9
+)
+
+var Err = errors.New("amf0 read error")
+
+// AMF0 spec: http://download.macromedia.com/pub/labs/amf/amf0_spec_121207.pdf
+type AMF0 struct {
+	buf []byte
+	pos int
+}
+
+func NewReader(b []byte) *AMF0 {
+	return &AMF0{buf: b}
+}
+
+func (a *AMF0) ReadMetaData() map[string]interface{} {
+	if b, _ := a.ReadByte(); b != TypeString {
+		return nil
+	}
+	if s, _ := a.ReadString(); s != "onMetaData" {
+		return nil
+	}
+
+	b, _ := a.ReadByte()
+	switch b {
+	case TypeObject:
+		v, _ := a.ReadObject()
+		return v
+	case TypeEcmaArray:
+		v, _ := a.ReadEcmaArray()
+		return v
+	}
+
+	return nil
+}
+
+func (a *AMF0) ReadMap() (map[interface{}]interface{}, error) {
+	dict := make(map[interface{}]interface{})
+
+	for a.pos < len(a.buf) {
+		k, err := a.ReadItem()
+		if err != nil {
+			return nil, err
+		}
+		v, err := a.ReadItem()
+		if err != nil {
+			return nil, err
+		}
+		dict[k] = v
+	}
+
+	return dict, nil
+}
+
+func (a *AMF0) ReadItem() (interface{}, error) {
+	dataType, err := a.ReadByte()
+	if err != nil {
+		return nil, err
+	}
+
+	switch dataType {
+	case TypeNumber:
+		return a.ReadNumber()
+
+	case TypeBoolean:
+		v, err := a.ReadByte()
+		return v != 0, err
+
+	case TypeString:
+		return a.ReadString()
+
+	case TypeObject:
+		return a.ReadObject()
+
+	case TypeObjectEnd:
+		return nil, nil
+	}
+
+	return nil, Err
+}
+
+func (a *AMF0) ReadByte() (byte, error) {
+	if a.pos >= len(a.buf) {
+		return 0, Err
+	}
+
+	v := a.buf[a.pos]
+	a.pos++
+	return v, nil
+}
+
+func (a *AMF0) ReadNumber() (float64, error) {
+	if a.pos+8 >= len(a.buf) {
+		return 0, Err
+	}
+
+	v := binary.BigEndian.Uint64(a.buf[a.pos : a.pos+8])
+	a.pos += 8
+	return math.Float64frombits(v), nil
+}
+
+func (a *AMF0) ReadString() (string, error) {
+	if a.pos+2 >= len(a.buf) {
+		return "", Err
+	}
+
+	size := int(binary.BigEndian.Uint16(a.buf[a.pos:]))
+	a.pos += 2
+
+	if a.pos+size >= len(a.buf) {
+		return "", Err
+	}
+
+	s := string(a.buf[a.pos : a.pos+size])
+	a.pos += size
+
+	return s, nil
+}
+
+func (a *AMF0) ReadObject() (map[string]interface{}, error) {
+	obj := make(map[string]interface{})
+
+	for {
+		k, err := a.ReadString()
+		if err != nil {
+			return nil, err
+		}
+
+		v, err := a.ReadItem()
+		if err != nil {
+			return nil, err
+		}
+
+		if k == "" {
+			break
+		}
+
+		obj[k] = v
+	}
+
+	return obj, nil
+}
+
+func (a *AMF0) ReadEcmaArray() (map[string]interface{}, error) {
+	if a.pos+4 >= len(a.buf) {
+		return nil, Err
+	}
+	a.pos += 4 // skip size
+
+	return a.ReadObject()
+}

pkg/httpflv/flvio.go

@@ -0,0 +1,97 @@
+package httpflv
+
+import (
+	"fmt"
+	"github.com/deepch/vdk/format/flv/flvio"
+	"github.com/deepch/vdk/utils/bits/pio"
+	"io"
+)
+
+// TODO: rewrite all of this someday
+
+func ReadTag(r io.Reader, b []byte) (tag flvio.Tag, ts int32, err error) {
+	if _, err = io.ReadFull(r, b[:flvio.TagHeaderLength]); err != nil {
+		return
+	}
+	var datalen int
+	if tag, ts, datalen, err = flvio.ParseTagHeader(b); err != nil {
+		return
+	}
+
+	data := make([]byte, datalen)
+	if _, err = io.ReadFull(r, data); err != nil {
+		return
+	}
+
+	n, err := ParseHeader(&tag, data)
+	if err != nil {
+		return
+	}
+	tag.Data = data[n:]
+
+	if _, err = io.ReadFull(r, b[:4]); err != nil {
+		return
+	}
+	return
+}
+
+func ParseHeader(self *flvio.Tag, b []byte) (n int, err error) {
+	switch self.Type {
+	case flvio.TAG_AUDIO:
+		return audioParseHeader(self, b)
+
+	case flvio.TAG_VIDEO:
+		return videoParseHeader(self, b)
+	}
+
+	return
+}
+
+func audioParseHeader(tag *flvio.Tag, b []byte) (n int, err error) {
+	if len(b) < n+1 {
+		err = fmt.Errorf("audiodata: parse invalid")
+		return
+	}
+
+	flags := b[n]
+	n++
+	tag.SoundFormat = flags >> 4
+	tag.SoundRate = (flags >> 2) & 0x3
+	tag.SoundSize = (flags >> 1) & 0x1
+	tag.SoundType = flags & 0x1
+
+	switch tag.SoundFormat {
+	case flvio.SOUND_AAC:
+		if len(b) < n+1 {
+			err = fmt.Errorf("audiodata: parse invalid")
+			return
+		}
+		tag.AACPacketType = b[n]
+		n++
+	}
+
+	return
+}
+
+func videoParseHeader(tag *flvio.Tag, b []byte) (n int, err error) {
+	if len(b) < n+1 {
+		err = fmt.Errorf("videodata: parse invalid")
+		return
+	}
+	flags := b[n]
+	tag.FrameType = flags >> 4
+	tag.CodecID = flags & 0xf
+	n++
+
+	if len(b) < n+4 {
+		err = fmt.Errorf("videodata: parse invalid")
+		return
+	}
+	tag.AVCPacketType = b[n]
+	n++
+
+	tag.CompositionTime = pio.I24BE(b[n:])
+	n += 3
+
+	return
+}

pkg/httpflv/httpflv.go

@@ -2,8 +2,9 @@ package httpflv
 
 import (
 	"bufio"
-	"errors"
+	"bytes"
 	"github.com/deepch/vdk/av"
+	"github.com/deepch/vdk/codec/aacparser"
 	"github.com/deepch/vdk/codec/h264parser"
 	"github.com/deepch/vdk/format/flv/flvio"
 	"github.com/deepch/vdk/utils/bits/pio"
@@ -41,8 +42,12 @@ func Accept(res *http.Response) (*Conn, error) {
 		return nil, err
 	}
 
-	if flags&flvio.FILE_HAS_VIDEO == 0 {
-		return nil, errors.New("not supported")
+	if flags&flvio.FILE_HAS_VIDEO != 0 {
+		c.videoIdx = -1
+	}
+
+	if flags&flvio.FILE_HAS_AUDIO != 0 {
+		c.audioIdx = -1
 	}
 
 	if _, err = c.reader.Discard(n); err != nil {
@@ -56,49 +61,154 @@ type Conn struct {
 	conn   io.ReadCloser
 	reader *bufio.Reader
 	buf    []byte
+
+	videoIdx int8
+	audioIdx int8
 }
 
 func (c *Conn) Streams() ([]av.CodecData, error) {
-	for {
+	var video, audio av.CodecData
+
+	// Normal software sends:
+	// 1. Video/audio flag in header
+	// 2. MetaData as first tag (with video/audio codec info)
+	// 3. Video/audio headers in 2nd and 3rd tag
+
+	// Reolink camera sends:
+	// 1. Empty video/audio flag
+	// 2. MedaData without stereo key for AAC
+	// 3. Audio header after Video keyframe tag
+
+	waitVideo := c.videoIdx != 0
+	waitAudio := c.audioIdx != 0
+
+	for i := 0; i < 20; i++ {
 		tag, _, err := flvio.ReadTag(c.reader, c.buf)
 		if err != nil {
 			return nil, err
 		}
 
-		if tag.Type != flvio.TAG_VIDEO || tag.AVCPacketType != flvio.AAC_SEQHDR {
-			continue
+		//log.Printf("[FLV] type=%d avc=%d aac=%d video=%t audio=%t", tag.Type, tag.AVCPacketType, tag.AACPacketType, video != nil, audio != nil)
+
+		switch tag.Type {
+		case flvio.TAG_SCRIPTDATA:
+			if meta := NewReader(tag.Data).ReadMetaData(); meta != nil {
+				waitVideo = meta["videocodecid"] != nil
+
+				// don't wait audio tag because parse all info from MetaData
+				waitAudio = false
+
+				audio = parseAudioConfig(meta)
+			} else {
+				waitVideo = bytes.Contains(tag.Data, []byte("videocodecid"))
+				waitAudio = bytes.Contains(tag.Data, []byte("audiocodecid"))
+			}
+
+		case flvio.TAG_VIDEO:
+			if tag.AVCPacketType == flvio.AVC_SEQHDR {
+				video, _ = h264parser.NewCodecDataFromAVCDecoderConfRecord(tag.Data)
+			}
+			waitVideo = false
+
+		case flvio.TAG_AUDIO:
+			if tag.SoundFormat == flvio.SOUND_AAC && tag.AACPacketType == flvio.AAC_SEQHDR {
+				audio, _ = aacparser.NewCodecDataFromMPEG4AudioConfigBytes(tag.Data)
+			}
+			waitAudio = false
 		}
 
-		stream, err := h264parser.NewCodecDataFromAVCDecoderConfRecord(tag.Data)
-		if err != nil {
-			return nil, err
+		if !waitVideo && !waitAudio {
+			break
 		}
-
-		return []av.CodecData{stream}, nil
 	}
+
+	if video != nil && audio != nil {
+		c.videoIdx = 0
+		c.audioIdx = 1
+		return []av.CodecData{video, audio}, nil
+	} else if video != nil {
+		c.videoIdx = 0
+		c.audioIdx = -1
+		return []av.CodecData{video}, nil
+	} else if audio != nil {
+		c.videoIdx = -1
+		c.audioIdx = 0
+		return []av.CodecData{audio}, nil
+	}
+
+	return nil, nil
 }
 
 func (c *Conn) ReadPacket() (av.Packet, error) {
 	for {
-		tag, ts, err := flvio.ReadTag(c.reader, c.buf)
+		tag, ts, err := ReadTag(c.reader, c.buf)
 		if err != nil {
 			return av.Packet{}, err
 		}
 
-		if tag.Type != flvio.TAG_VIDEO || tag.AVCPacketType != flvio.AVC_NALU {
-			continue
-		}
+		switch tag.Type {
+		case flvio.TAG_VIDEO:
+			if c.videoIdx < 0 || tag.AVCPacketType != flvio.AVC_NALU {
+				continue
+			}
 
-		return av.Packet{
-			Idx:             0,
-			Data:            tag.Data,
-			CompositionTime: flvio.TsToTime(tag.CompositionTime),
-			IsKeyFrame:      tag.FrameType == flvio.FRAME_KEY,
-			Time:            flvio.TsToTime(ts),
-		}, nil
+			//log.Printf("[FLV] %v, len: %d, ts: %10d", h264.Types(tag.Data), len(tag.Data), flvio.TsToTime(ts))
+
+			return av.Packet{
+				Idx:             c.videoIdx,
+				Data:            tag.Data,
+				CompositionTime: flvio.TsToTime(tag.CompositionTime),
+				IsKeyFrame:      tag.FrameType == flvio.FRAME_KEY,
+				Time:            flvio.TsToTime(ts),
+			}, nil
+
+		case flvio.TAG_AUDIO:
+			if c.audioIdx < 0 || tag.SoundFormat != flvio.SOUND_AAC || tag.AACPacketType != flvio.AAC_RAW {
+				continue
+			}
+
+			return av.Packet{Idx: c.audioIdx, Data: tag.Data, Time: flvio.TsToTime(ts)}, nil
+		}
 	}
 }
 
 func (c *Conn) Close() (err error) {
 	return c.conn.Close()
 }
+
+func parseAudioConfig(meta map[string]interface{}) av.CodecData {
+	if meta["audiocodecid"] != float64(10) {
+		return nil
+	}
+
+	config := aacparser.MPEG4AudioConfig{
+		ObjectType: aacparser.AOT_AAC_LC,
+	}
+
+	switch v := meta["audiosamplerate"].(type) {
+	case float64:
+		config.SampleRate = int(v)
+	default:
+		return nil
+	}
+
+	switch meta["stereo"] {
+	case true:
+		config.ChannelConfig = 2
+		config.ChannelLayout = av.CH_STEREO
+	default:
+		// Reolink doesn't have this setting
+		config.ChannelConfig = 1
+		config.ChannelLayout = av.CH_MONO
+	}
+
+	buf := &bytes.Buffer{}
+	if err := aacparser.WriteMPEG4AudioConfig(buf, config); err != nil {
+		return nil
+	}
+
+	return aacparser.CodecData{
+		Config:      config,
+		ConfigBytes: buf.Bytes(),
+	}
+}

pkg/iso/atoms.go

@@ -0,0 +1,318 @@
+package iso
+
+const (
+	Ftyp                        = "ftyp"
+	Moov                        = "moov"
+	MoovMvhd                    = "mvhd"
+	MoovTrak                    = "trak"
+	MoovTrakTkhd                = "tkhd"
+	MoovTrakMdia                = "mdia"
+	MoovTrakMdiaMdhd            = "mdhd"
+	MoovTrakMdiaHdlr            = "hdlr"
+	MoovTrakMdiaMinf            = "minf"
+	MoovTrakMdiaMinfVmhd        = "vmhd"
+	MoovTrakMdiaMinfSmhd        = "smhd"
+	MoovTrakMdiaMinfDinf        = "dinf"
+	MoovTrakMdiaMinfDinfDref    = "dref"
+	MoovTrakMdiaMinfDinfDrefUrl = "url "
+	MoovTrakMdiaMinfStbl        = "stbl"
+	MoovTrakMdiaMinfStblStsd    = "stsd"
+	MoovTrakMdiaMinfStblStts    = "stts"
+	MoovTrakMdiaMinfStblStsc    = "stsc"
+	MoovTrakMdiaMinfStblStsz    = "stsz"
+	MoovTrakMdiaMinfStblStco    = "stco"
+	MoovMvex                    = "mvex"
+	MoovMvexTrex                = "trex"
+	Moof                        = "moof"
+	MoofMfhd                    = "mfhd"
+	MoofTraf                    = "traf"
+	MoofTrafTfhd                = "tfhd"
+	MoofTrafTfdt                = "tfdt"
+	MoofTrafTrun                = "trun"
+	Mdat                        = "mdat"
+)
+
+func (m *Movie) WriteFileType() {
+	m.StartAtom(Ftyp)
+	m.WriteString("iso5")
+	m.WriteUint32(512)
+	m.WriteString("iso5")
+	m.WriteString("iso6")
+	m.WriteString("mp41")
+	m.EndAtom()
+}
+
+func (m *Movie) WriteMovieHeader() {
+	m.StartAtom(MoovMvhd)
+	m.Skip(1)           // version
+	m.Skip(3)           // flags
+	m.Skip(4)           // create time
+	m.Skip(4)           // modify time
+	m.WriteUint32(1000) // time scale
+	m.Skip(4)           // duration
+	m.WriteFloat32(1)   // preferred rate
+	m.WriteFloat16(1)   // preferred volume
+	m.Skip(10)          // reserved
+	m.WriteMatrix()
+	m.Skip(6 * 4)             // predefined?
+	m.WriteUint32(0xFFFFFFFF) // next track ID
+	m.EndAtom()
+}
+
+func (m *Movie) WriteTrackHeader(id uint32, width, height uint16) {
+	const (
+		TkhdTrackEnabled   = 0x0001
+		TkhdTrackInMovie   = 0x0002
+		TkhdTrackInPreview = 0x0004
+		TkhdTrackInPoster  = 0x0008
+	)
+
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-32963
+	m.StartAtom(MoovTrakTkhd)
+	m.Skip(1) // version
+	m.WriteUint24(TkhdTrackEnabled | TkhdTrackInMovie)
+	m.Skip(4)         // create time
+	m.Skip(4)         // modify time
+	m.WriteUint32(id) // trackID
+	m.Skip(4)         // reserved
+	m.Skip(4)         // duration
+	m.Skip(8)         // reserved
+	m.Skip(2)         // layer
+	if width > 0 {
+		m.Skip(2)
+		m.Skip(2)
+	} else {
+		m.WriteUint16(1)  // alternate group
+		m.WriteFloat16(1) // volume
+	}
+	m.Skip(2) // reserved
+	m.WriteMatrix()
+	if width > 0 {
+		m.WriteFloat32(float64(width))
+		m.WriteFloat32(float64(height))
+	} else {
+		m.Skip(4)
+		m.Skip(4)
+	}
+	m.EndAtom()
+}
+
+func (m *Movie) WriteMediaHeader(timescale uint32) {
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-32999
+	m.StartAtom(MoovTrakMdiaMdhd)
+	m.Skip(1)                // version
+	m.Skip(3)                // flags
+	m.Skip(4)                // creation time
+	m.Skip(4)                // modification time
+	m.WriteUint32(timescale) // timescale
+	m.Skip(4)                // duration
+	m.WriteUint16(0x55C4)    // language (Unspecified)
+	m.Skip(2)                // quality
+	m.EndAtom()
+}
+
+func (m *Movie) WriteMediaHandler(s, name string) {
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-33004
+	m.StartAtom(MoovTrakMdiaHdlr)
+	m.Skip(1) // version
+	m.Skip(3) // flags
+	m.Skip(4)
+	m.WriteString(s)    // handler type (4 byte!)
+	m.Skip(3 * 4)       // reserved
+	m.WriteString(name) // handler name (any len)
+	m.Skip(1)           // end string
+	m.EndAtom()
+}
+
+func (m *Movie) WriteVideoMediaInfo() {
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-33012
+	m.StartAtom(MoovTrakMdiaMinfVmhd)
+	m.Skip(1)        // version
+	m.WriteUint24(1) // flags (You should always set this flag to 1)
+	m.Skip(2)        // graphics mode
+	m.Skip(3 * 2)    // op color
+	m.EndAtom()
+}
+
+func (m *Movie) WriteAudioMediaInfo() {
+	m.StartAtom(MoovTrakMdiaMinfSmhd)
+	m.Skip(1) // version
+	m.Skip(3) // flags
+	m.Skip(4) // balance
+	m.EndAtom()
+}
+
+func (m *Movie) WriteDataInfo() {
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-25680
+	m.StartAtom(MoovTrakMdiaMinfDinf)
+	m.StartAtom(MoovTrakMdiaMinfDinfDref)
+	m.Skip(1)        // version
+	m.Skip(3)        // flags
+	m.WriteUint32(1) // childrens
+
+	m.StartAtom(MoovTrakMdiaMinfDinfDrefUrl)
+	m.Skip(1)        // version
+	m.WriteUint24(1) // flags (self reference)
+	m.EndAtom()
+
+	m.EndAtom() // DREF
+	m.EndAtom() // DINF
+}
+
+func (m *Movie) WriteSampleTable(writeSampleDesc func()) {
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-33040
+	m.StartAtom(MoovTrakMdiaMinfStbl)
+
+	m.StartAtom(MoovTrakMdiaMinfStblStsd)
+	m.Skip(1)        // version
+	m.Skip(3)        // flags
+	m.WriteUint32(1) // entry count
+	writeSampleDesc()
+	m.EndAtom()
+
+	m.StartAtom(MoovTrakMdiaMinfStblStts)
+	m.Skip(1) // version
+	m.Skip(3) // flags
+	m.Skip(4) // entry count
+	m.EndAtom()
+
+	m.StartAtom(MoovTrakMdiaMinfStblStsc)
+	m.Skip(1) // version
+	m.Skip(3) // flags
+	m.Skip(4) // entry count
+	m.EndAtom()
+
+	m.StartAtom(MoovTrakMdiaMinfStblStsz)
+	m.Skip(1) // version
+	m.Skip(3) // flags
+	m.Skip(4) // sample size
+	m.Skip(4) // entry count
+	m.EndAtom()
+
+	m.StartAtom(MoovTrakMdiaMinfStblStco)
+	m.Skip(1) // version
+	m.Skip(3) // flags
+	m.Skip(4) // entry count
+	m.EndAtom()
+
+	m.EndAtom()
+}
+
+func (m *Movie) WriteTrackExtend(id uint32) {
+	m.StartAtom(MoovMvexTrex)
+	m.Skip(1)         // version
+	m.Skip(3)         // flags
+	m.WriteUint32(id) // trackID
+	m.WriteUint32(1)  // default sample description index
+	m.Skip(4)         // default sample duration
+	m.Skip(4)         // default sample size
+	m.Skip(4)         // default sample flags
+	m.EndAtom()
+}
+
+func (m *Movie) WriteVideoTrack(id uint32, codec string, timescale uint32, width, height uint16, conf []byte) {
+	m.StartAtom(MoovTrak)
+	m.WriteTrackHeader(id, width, height)
+
+	m.StartAtom(MoovTrakMdia)
+	m.WriteMediaHeader(timescale)
+	m.WriteMediaHandler("vide", "VideoHandler")
+
+	m.StartAtom(MoovTrakMdiaMinf)
+	m.WriteVideoMediaInfo()
+	m.WriteDataInfo()
+	m.WriteSampleTable(func() {
+		m.WriteVideo(codec, width, height, conf)
+	})
+	m.EndAtom() // MINF
+
+	m.EndAtom() // MDIA
+	m.EndAtom() // TRAK
+}
+
+func (m *Movie) WriteAudioTrack(id uint32, codec string, timescale uint32, channels uint16, conf []byte) {
+	m.StartAtom(MoovTrak)
+	m.WriteTrackHeader(id, 0, 0)
+
+	m.StartAtom(MoovTrakMdia)
+	m.WriteMediaHeader(timescale)
+	m.WriteMediaHandler("soun", "SoundHandler")
+
+	m.StartAtom(MoovTrakMdiaMinf)
+	m.WriteAudioMediaInfo()
+	m.WriteDataInfo()
+	m.WriteSampleTable(func() {
+		m.WriteAudio(codec, channels, timescale, conf)
+	})
+	m.EndAtom() // MINF
+
+	m.EndAtom() // MDIA
+	m.EndAtom() // TRAK
+}
+
+func (m *Movie) WriteMovieFragment(seq, tid, duration, size uint32, time uint64) {
+	m.StartAtom(Moof)
+
+	m.StartAtom(MoofMfhd)
+	m.Skip(1)          // version
+	m.Skip(3)          // flags
+	m.WriteUint32(seq) // sequence number
+	m.EndAtom()
+
+	m.StartAtom(MoofTraf)
+
+	const (
+		TfhdDefaultSampleDuration = 0x000008
+		TfhdDefaultSampleSize     = 0x000010
+		TfhdDefaultSampleFlags    = 0x000020
+		TfhdDefaultBaseIsMoof     = 0x020000
+	)
+
+	m.StartAtom(MoofTrafTfhd)
+	m.Skip(1) // version
+	m.WriteUint24(
+		TfhdDefaultSampleDuration |
+			TfhdDefaultSampleSize |
+			TfhdDefaultSampleFlags |
+			TfhdDefaultBaseIsMoof,
+	)
+	m.WriteUint32(tid)       // track id
+	m.WriteUint32(duration)  // default sample duration
+	m.WriteUint32(size)      // default sample size
+	m.WriteUint32(0x2000000) // default sample flags
+	m.EndAtom()
+
+	m.StartAtom(MoofTrafTfdt)
+	m.WriteBytes(1)     // version
+	m.Skip(3)           // flags
+	m.WriteUint64(time) // base media decode time
+	m.EndAtom()
+
+	const (
+		TrunDataOffset       = 0x000001
+		TrunFirstSampleFlags = 0x000004
+		TrunSampleDuration   = 0x0000100
+		TrunSampleSize       = 0x0000200
+		TrunSampleFlags      = 0x0000400
+		TrunSampleCTS        = 0x0000800
+	)
+
+	m.StartAtom(MoofTrafTrun)
+	m.Skip(1)                     // version
+	m.WriteUint24(TrunDataOffset) // flags
+	m.WriteUint32(1)              // sample count
+	// data offset: current pos + uint32 len + MDAT header len
+	m.WriteUint32(uint32(len(m.b)) + 4 + 8)
+	m.EndAtom() // TRUN
+
+	m.EndAtom() // TRAF
+
+	m.EndAtom() // MOOF
+}
+
+func (m *Movie) WriteData(b []byte) {
+	m.StartAtom(Mdat)
+	m.Write(b)
+	m.EndAtom()
+
+}

pkg/iso/codecs.go

@@ -0,0 +1,151 @@
+package iso
+
+import "github.com/AlexxIT/go2rtc/pkg/streamer"
+
+func (m *Movie) WriteVideo(codec string, width, height uint16, conf []byte) {
+	// https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html
+	switch codec {
+	case streamer.CodecH264:
+		m.StartAtom("avc1")
+	case streamer.CodecH265:
+		m.StartAtom("hev1")
+	default:
+		panic("unsupported iso video: " + codec)
+	}
+	m.Skip(6)
+	m.WriteUint16(1)      // data_reference_index
+	m.Skip(2)             // version
+	m.Skip(2)             // revision
+	m.Skip(4)             // vendor
+	m.Skip(4)             // temporal quality
+	m.Skip(4)             // spatial quality
+	m.WriteUint16(width)  // width
+	m.WriteUint16(height) // height
+	m.WriteFloat32(72)    // horizontal resolution
+	m.WriteFloat32(72)    // vertical resolution
+	m.Skip(4)             // reserved
+	m.WriteUint16(1)      // frame count
+	m.Skip(32)            // compressor name
+	m.WriteUint16(24)     // depth
+	m.WriteUint16(0xFFFF) // color table id (-1)
+
+	switch codec {
+	case streamer.CodecH264:
+		m.StartAtom("avcC")
+	case streamer.CodecH265:
+		m.StartAtom("hvcC")
+	}
+	m.Write(conf)
+	m.EndAtom() // AVCC
+
+	m.EndAtom() // AVC1
+}
+
+func (m *Movie) WriteAudio(codec string, channels uint16, sampleRate uint32, conf []byte) {
+	switch codec {
+	case streamer.CodecAAC, streamer.CodecMP3:
+		m.StartAtom("mp4a")
+	case streamer.CodecOpus:
+		m.StartAtom("Opus")
+	case streamer.CodecPCMU:
+		m.StartAtom("ulaw")
+	case streamer.CodecPCMA:
+		m.StartAtom("alaw")
+	default:
+		panic("unsupported iso audio: " + codec)
+	}
+	m.Skip(6)
+	m.WriteUint16(1)                    // data_reference_index
+	m.Skip(2)                           // version
+	m.Skip(2)                           // revision
+	m.Skip(4)                           // vendor
+	m.WriteUint16(channels)             // channel_count
+	m.WriteUint16(16)                   // sample_size
+	m.Skip(2)                           // compression id
+	m.Skip(2)                           // reserved
+	m.WriteFloat32(float64(sampleRate)) // sample_rate
+
+	switch codec {
+	case streamer.CodecAAC:
+		m.WriteEsdsAAC(conf)
+	case streamer.CodecMP3:
+		m.WriteEsdsMP3()
+	case streamer.CodecOpus:
+		// don't know what means this magic
+		m.StartAtom("dOps")
+		m.WriteBytes(0, 0x02, 0x01, 0x38, 0, 0, 0xBB, 0x80, 0, 0, 0)
+		m.EndAtom()
+	case streamer.CodecPCMU, streamer.CodecPCMA:
+		// don't know what means this magic
+		m.StartAtom("chan")
+		m.WriteBytes(0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 4, 0, 0, 0, 0)
+		m.EndAtom()
+	}
+
+	m.EndAtom() // MP4A/OPUS
+}
+
+func (m *Movie) WriteEsdsAAC(conf []byte) {
+	m.StartAtom("esds")
+	m.Skip(1) // version
+	m.Skip(3) // flags
+
+	// MP4ESDescrTag[3]:
+	// - MP4DecConfigDescrTag[4]:
+	//   - MP4DecSpecificDescrTag[5]: conf
+	// - Other[6]
+	const header = 5
+	const size3 = 3
+	const size4 = 13
+	size5 := byte(len(conf))
+	const size6 = 1
+
+	m.WriteBytes(3, 0x80, 0x80, 0x80, size3+header+size4+header+size5+header+size6)
+	m.Skip(2) // es id
+	m.Skip(1) // es flags
+
+	m.WriteBytes(4, 0x80, 0x80, 0x80, size4+header+size5)
+	m.WriteBytes(0x40) // object id
+	m.WriteBytes(0x15) // stream type
+	m.Skip(3)          // buffer size db
+	m.Skip(4)          // max bitraga
+	m.Skip(4)          // avg bitraga
+
+	m.WriteBytes(5, 0x80, 0x80, 0x80, size5)
+	m.Write(conf)
+
+	m.WriteBytes(6, 0x80, 0x80, 0x80, 1)
+	m.WriteBytes(2) // ?
+
+	m.EndAtom() // ESDS
+}
+
+func (m *Movie) WriteEsdsMP3() {
+	m.StartAtom("esds")
+	m.Skip(1) // version
+	m.Skip(3) // flags
+
+	// MP4ESDescrTag[3]:
+	// - MP4DecConfigDescrTag[4]:
+	// - Other[6]
+	const header = 5
+	const size3 = 3
+	const size4 = 13
+	const size6 = 1
+
+	m.WriteBytes(3, 0x80, 0x80, 0x80, size3+header+size4+header+size6)
+	m.Skip(2) // es id
+	m.Skip(1) // es flags
+
+	m.WriteBytes(4, 0x80, 0x80, 0x80, size4)
+	m.WriteBytes(0x6B) // object id
+	m.WriteBytes(0x15) // stream type
+	m.Skip(3)          // buffer size db
+	m.Skip(4)          // max bitraga
+	m.Skip(4)          // avg bitraga
+
+	m.WriteBytes(6, 0x80, 0x80, 0x80, 1)
+	m.WriteBytes(2) // ?
+
+	m.EndAtom() // ESDS
+}

pkg/iso/iso.go

@@ -0,0 +1,91 @@
+package iso
+
+import (
+	"encoding/binary"
+	"math"
+)
+
+type Movie struct {
+	b     []byte
+	start []int
+}
+
+func NewMovie(size int) *Movie {
+	return &Movie{b: make([]byte, 0, size)}
+}
+
+func (m *Movie) Bytes() []byte {
+	return m.b
+}
+
+func (m *Movie) StartAtom(name string) {
+	m.start = append(m.start, len(m.b))
+	m.b = append(m.b, 0, 0, 0, 0)
+	m.b = append(m.b, name...)
+}
+
+func (m *Movie) EndAtom() {
+	n := len(m.start) - 1
+
+	i := m.start[n]
+	size := uint32(len(m.b) - i)
+	binary.BigEndian.PutUint32(m.b[i:], size)
+
+	m.start = m.start[:n]
+}
+
+func (m *Movie) Write(b []byte) {
+	m.b = append(m.b, b...)
+}
+
+func (m *Movie) WriteBytes(b ...byte) {
+	m.b = append(m.b, b...)
+}
+
+func (m *Movie) WriteString(s string) {
+	m.b = append(m.b, s...)
+}
+
+func (m *Movie) Skip(n int) {
+	m.b = append(m.b, make([]byte, n)...)
+}
+
+func (m *Movie) WriteUint16(v uint16) {
+	m.b = append(m.b, byte(v>>8), byte(v))
+}
+
+func (m *Movie) WriteUint24(v uint32) {
+	m.b = append(m.b, byte(v>>16), byte(v>>8), byte(v))
+}
+
+func (m *Movie) WriteUint32(v uint32) {
+	m.b = append(m.b, byte(v>>24), byte(v>>16), byte(v>>8), byte(v))
+}
+
+func (m *Movie) WriteUint64(v uint64) {
+	m.b = append(m.b, byte(v>>56), byte(v>>48), byte(v>>40), byte(v>>32), byte(v>>24), byte(v>>16), byte(v>>8), byte(v))
+}
+
+func (m *Movie) WriteFloat16(f float64) {
+	i, f := math.Modf(f)
+	f *= 256
+	m.b = append(m.b, byte(i), byte(f))
+}
+
+func (m *Movie) WriteFloat32(f float64) {
+	i, f := math.Modf(f)
+	f *= 65536
+	m.b = append(m.b, byte(uint16(i)>>8), byte(i), byte(uint16(f)>>8), byte(f))
+}
+
+func (m *Movie) WriteMatrix() {
+	m.WriteUint32(0x00010000)
+	m.Skip(4)
+	m.Skip(4)
+	m.Skip(4)
+	m.WriteUint32(0x00010000)
+	m.Skip(4)
+	m.Skip(4)
+	m.Skip(4)
+	m.WriteUint32(0x40000000)
+}

pkg/ivideon/client.go

@@ -14,9 +14,19 @@ import (
 	"io"
 	"net/http"
 	"strings"
+	"sync"
+	"sync/atomic"
 	"time"
 )
 
+type State byte
+
+const (
+	StateNone State = iota
+	StateConn
+	StateHandle
+)
+
 type Client struct {
 	streamer.Element
 
@@ -26,12 +36,14 @@ type Client struct {
 	medias []*streamer.Media
 	tracks map[byte]*streamer.Track
 
-	closed bool
-
 	msg *message
 	t0  time.Time
 
 	buffer chan []byte
+	state  State
+	mu     sync.Mutex
+
+	recv uint32
 }
 
 func NewClient(id string) *Client {
@@ -69,16 +81,26 @@ func (c *Client) Dial() (err error) {
 		return err
 	}
 
+	c.state = StateConn
+
 	return nil
 }
 
 func (c *Client) Handle() error {
-	c.buffer = make(chan []byte, 5)
 	// add delay to the stream for smooth playing (not a best solution)
 	c.t0 = time.Now().Add(time.Second)
 
-	// processing stream in separate thread for lower delay between packets
-	go c.worker()
+	c.mu.Lock()
+
+	if c.state == StateConn {
+		c.buffer = make(chan []byte, 5)
+		c.state = StateHandle
+
+		// processing stream in separate thread for lower delay between packets
+		go c.worker(c.buffer)
+	}
+
+	c.mu.Unlock()
 
 	_, data, err := c.conn.ReadMessage()
 	if err != nil {
@@ -87,7 +109,12 @@ func (c *Client) Handle() error {
 
 	track := c.tracks[c.msg.Track]
 	if track != nil {
-		c.buffer <- data
+		c.mu.Lock()
+		if c.state == StateHandle {
+			c.buffer <- data
+			atomic.AddUint32(&c.recv, uint32(len(data)))
+		}
+		c.mu.Unlock()
 	}
 
 	// we have one unprocessed msg after getTracks
@@ -114,7 +141,12 @@ func (c *Client) Handle() error {
 
 			track = c.tracks[msg.Track]
 			if track != nil {
-				c.buffer <- data
+				c.mu.Lock()
+				if c.state == StateHandle {
+					c.buffer <- data
+					atomic.AddUint32(&c.recv, uint32(len(data)))
+				}
+				c.mu.Unlock()
 			}
 
 		default:
@@ -124,11 +156,19 @@ func (c *Client) Handle() error {
 }
 
 func (c *Client) Close() error {
-	if c.conn == nil {
+	c.mu.Lock()
+	defer c.mu.Unlock()
+
+	switch c.state {
+	case StateNone:
 		return nil
+	case StateConn:
+	case StateHandle:
+		close(c.buffer)
 	}
-	close(c.buffer)
-	c.closed = true
+
+	c.state = StateNone
+
 	return c.conn.Close()
 }
 
@@ -208,13 +248,13 @@ func (c *Client) getTracks() error {
 	}
 }
 
-func (c *Client) worker() {
+func (c *Client) worker(buffer chan []byte) {
 	var track *streamer.Track
 	for _, track = range c.tracks {
 		break
 	}
 
-	for data := range c.buffer {
+	for data := range buffer {
 		moof := &fmp4io.MovieFrag{}
 		if _, err := moof.Unmarshal(data, 0); err != nil {
 			continue

pkg/ivideon/streamer.go

@@ -1,8 +1,10 @@
 package ivideon
 
 import (
+	"encoding/json"
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"sync/atomic"
 )
 
 func (c *Client) GetMedias() []*streamer.Media {
@@ -20,7 +22,7 @@ func (c *Client) GetTrack(media *streamer.Media, codec *streamer.Codec) *streame
 
 func (c *Client) Start() error {
 	err := c.Handle()
-	if c.closed {
+	if c.buffer == nil {
 		return nil
 	}
 	return err
@@ -29,3 +31,19 @@ func (c *Client) Start() error {
 func (c *Client) Stop() error {
 	return c.Close()
 }
+
+func (c *Client) MarshalJSON() ([]byte, error) {
+	var tracks []*streamer.Track
+	for _, track := range c.tracks {
+		tracks = append(tracks, track)
+	}
+
+	info := &streamer.Info{
+		Type:   "Ivideon source",
+		URL:    c.ID,
+		Medias: c.medias,
+		Tracks: tracks,
+		Recv:   atomic.LoadUint32(&c.recv),
+	}
+	return json.Marshal(info)
+}

pkg/mjpeg/client.go

@@ -2,6 +2,7 @@ package mjpeg
 
 import (
 	"bufio"
+	"encoding/json"
 	"errors"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/AlexxIT/go2rtc/pkg/tcp"
@@ -11,6 +12,7 @@ import (
 	"net/textproto"
 	"strconv"
 	"strings"
+	"sync/atomic"
 	"time"
 )
 
@@ -24,6 +26,7 @@ type Client struct {
 	res    *http.Response
 
 	track *streamer.Track
+	recv  uint32
 }
 
 func NewClient(res *http.Response) *Client {
@@ -64,10 +67,23 @@ func (c *Client) Start() error {
 }
 
 func (c *Client) Stop() error {
+	// important for close reader/writer gorutines
+	_ = c.res.Body.Close()
 	c.closed = true
 	return nil
 }
 
+func (c *Client) MarshalJSON() ([]byte, error) {
+	info := &streamer.Info{
+		Type:       "MJPEG source",
+		URL:        c.res.Request.URL.String(),
+		RemoteAddr: c.RemoteAddr,
+		UserAgent:  c.UserAgent,
+		Recv:       atomic.LoadUint32(&c.recv),
+	}
+	return json.Marshal(info)
+}
+
 func (c *Client) startJPEG() error {
 	buf, err := io.ReadAll(c.res.Body)
 	if err != nil {
@@ -77,6 +93,8 @@ func (c *Client) startJPEG() error {
 	packet := &rtp.Packet{Header: rtp.Header{Timestamp: now()}, Payload: buf}
 	_ = c.track.WriteRTP(packet)
 
+	atomic.AddUint32(&c.recv, uint32(len(buf)))
+
 	req := c.res.Request
 
 	for !c.closed {
@@ -96,13 +114,19 @@ func (c *Client) startJPEG() error {
 
 		packet = &rtp.Packet{Header: rtp.Header{Timestamp: now()}, Payload: buf}
 		_ = c.track.WriteRTP(packet)
+
+		atomic.AddUint32(&c.recv, uint32(len(buf)))
 	}
 
 	return nil
 }
 
 func (c *Client) startMJPEG(boundary string) error {
-	boundary = "--" + boundary
+	// some cameras add prefix to boundary header:
+	// https://github.com/TheTimeWalker/wallpanel-android
+	if !strings.HasPrefix(boundary, "--") {
+		boundary = "--" + boundary
+	}
 
 	r := bufio.NewReader(c.res.Body)
 	tp := textproto.NewReader(r)
@@ -139,6 +163,8 @@ func (c *Client) startMJPEG(boundary string) error {
 		packet := &rtp.Packet{Header: rtp.Header{Timestamp: now()}, Payload: buf}
 		_ = c.track.WriteRTP(packet)
 
+		atomic.AddUint32(&c.recv, uint32(len(buf)))
+
 		if _, err = r.Discard(2); err != nil {
 			return err
 		}

pkg/mjpeg/consumer.go

@@ -1,8 +1,10 @@
 package mjpeg
 
 import (
+	"encoding/json"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/rtp"
+	"sync/atomic"
 )
 
 type Consumer struct {
@@ -14,7 +16,7 @@ type Consumer struct {
 	codecs []*streamer.Codec
 	start  bool
 
-	send int
+	send uint32
 }
 
 func (c *Consumer) GetMedias() []*streamer.Media {
@@ -28,6 +30,7 @@ func (c *Consumer) GetMedias() []*streamer.Media {
 func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
 	push := func(packet *rtp.Packet) error {
 		c.Fire(packet.Payload)
+		atomic.AddUint32(&c.send, uint32(len(packet.Payload)))
 		return nil
 	}
 
@@ -38,3 +41,13 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 
 	return track.Bind(push)
 }
+
+func (c *Consumer) MarshalJSON() ([]byte, error) {
+	info := &streamer.Info{
+		Type:       "MJPEG client",
+		RemoteAddr: c.RemoteAddr,
+		UserAgent:  c.UserAgent,
+		Send:       atomic.LoadUint32(&c.send),
+	}
+	return json.Marshal(info)
+}

pkg/mjpeg/rfc2435.go

@@ -138,9 +138,9 @@ var chm_ac_symbols = []byte{
 	0xf9, 0xfa,
 }
 
-func MakeHeaders(t byte, w, h uint16, lqt, cqt []byte) []byte {
+func MakeHeaders(p []byte, t byte, w, h uint16, lqt, cqt []byte) []byte {
 	// Appendix A from https://www.rfc-editor.org/rfc/rfc2435
-	p := []byte{0xFF, 0xD8}
+	p = append(p, 0xFF, 0xD8)
 
 	p = MakeQuantHeader(p, lqt, 0)
 	p = MakeQuantHeader(p, cqt, 1)

pkg/mjpeg/rtp.go

@@ -6,7 +6,7 @@ import (
 )
 
 func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
-	var header, payload []byte
+	buf := make([]byte, 0, 512*1024) // 512K
 
 	return func(push streamer.WriterFunc) streamer.WriterFunc {
 		return func(packet *rtp.Packet) error {
@@ -25,7 +25,7 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 				b = b[8:]
 			}
 
-			if header == nil {
+			if len(buf) == 0 {
 				var lqt, cqt []byte
 
 				// 3.1.8.  Quantization Table header
@@ -43,32 +43,41 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 				w := uint16(packet.Payload[6]) << 3
 				h := uint16(packet.Payload[7]) << 3
 
-				// fix 2560x1920 and 2560x1440
-				if w == 512 && (h == 1920 || h == 1440) {
+				// fix sizes more than 2040
+				switch {
+				// 512x1920 512x1440
+				case w == cutSize(2560) && (h == 1920 || h == 1440):
 					w = 2560
+				// 1792x112
+				case w == cutSize(3840) && h == cutSize(2160):
+					w = 3840
+					h = 2160
+				// 256x1296
+				case w == cutSize(2304) && h == 1296:
+					w = 2304
 				}
 
 				//fmt.Printf("t: %d, q: %d, w: %d, h: %d\n", t, q, w, h)
-				header = MakeHeaders(t, w, h, lqt, cqt)
+				buf = MakeHeaders(buf, t, w, h, lqt, cqt)
 			}
 
 			// 3.1.9.  JPEG Payload
-			payload = append(payload, b...)
+			buf = append(buf, b...)
 
 			if !packet.Marker {
 				return nil
 			}
 
-			b = append(header, payload...)
-			if end := b[len(b)-2:]; end[0] != 0xFF && end[1] != 0xD9 {
-				b = append(b, 0xFF, 0xD9)
+			if end := buf[len(buf)-2:]; end[0] != 0xFF && end[1] != 0xD9 {
+				buf = append(buf, 0xFF, 0xD9)
 			}
 
-			header = nil
-			payload = nil
+			clone := *packet
+			clone.Payload = buf
 
-			packet.Payload = b
-			return push(packet)
+			buf = buf[:0] // clear buffer
+
+			return push(&clone)
 		}
 	}
 }
@@ -81,6 +90,10 @@ func RTPPay() streamer.WrapperFunc {
 	}
 }
 
+func cutSize(size uint16) uint16 {
+	return ((size >> 3) & 0xFF) << 3
+}
+
 //func RTPPay() streamer.WrapperFunc {
 //	const packetSize = 1436
 //

pkg/mp4/README.md

@@ -1,19 +1,30 @@
+## Fragmented MP4
+
+```
+ffmpeg -i "rtsp://..." -movflags +frag_keyframe+separate_moof+default_base_moof+empty_moov -frag_duration 1 -c copy -t 5 sample.mp4
+```
+
+- movflags frag_keyframe 
+  Start a new fragment at each video keyframe.
+- frag_duration duration
+  Create fragments that are duration microseconds long.
+- movflags separate_moof
+  Write a separate moof (movie fragment) atom for each track.
+- movflags default_base_moof
+  Similarly to the omit_tfhd_offset, this flag avoids writing the absolute base_data_offset field in tfhd atoms, but does so by using the new default-base-is-moof flag instead.
+
+https://ffmpeg.org/ffmpeg-formats.html#Options-13
+
 ## HEVC
 
-Browser     | avc1 | hvc1 | hev1
-------------|------|------|---
-Mac Chrome  | +    | -    | +
-Mac Safari  | +    | +    | -
-iOS 15?     | +    | +    | -
-Mac Firefox | +    | -    | -
-iOS 12      | +    | -    | -
-Android 13  | +    | -    | -
-
-```
-ffmpeg -i input-hev1.mp4 -c:v copy -tag:v hvc1 -c:a copy output-hvc1.mp4
-Stream #0:0(eng): Video: hevc (Main) (hev1 / 0x31766568), yuv420p(tv, progressive), 720x404, 164 kb/s, 29.97 fps,
-Stream #0:0(eng): Video: hevc (Main) (hvc1 / 0x31637668), yuv420p(tv, progressive), 720x404, 164 kb/s, 29.97 fps,
-```
+| Browser     | avc1 | hvc1 | hev1 |
+|-------------|------|------|------|
+ | Mac Chrome  | +    | -    | +    |
+ | Mac Safari  | +    | +    | -    |
+ | iOS 15?     | +    | +    | -    |
+ | Mac Firefox | +    | -    | -    |
+ | iOS 12      | +    | -    | -    |
+ | Android 13  | +    | -    | -    |
 
 ## Useful links
 

pkg/mp4/consumer.go

@@ -7,6 +7,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/h265"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/rtp"
+	"sync/atomic"
 )
 
 type Consumer struct {
@@ -20,7 +21,17 @@ type Consumer struct {
 	codecs []*streamer.Codec
 	wait   byte
 
-	send int
+	send uint32
+}
+
+// ParseQuery - like usual parse, but with mp4 param handler
+func ParseQuery(query map[string][]string) []*streamer.Media {
+	if query["mp4"] != nil {
+		cons := Consumer{}
+		return cons.GetMedias()
+	}
+
+	return streamer.ParseQuery(query)
 }
 
 const (
@@ -76,7 +87,7 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			}
 
 			buf := c.muxer.Marshal(trackID, packet)
-			c.send += len(buf)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(buf)
 
 			return nil
@@ -108,7 +119,7 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			}
 
 			buf := c.muxer.Marshal(trackID, packet)
-			c.send += len(buf)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(buf)
 
 			return nil
@@ -128,7 +139,7 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			}
 
 			buf := c.muxer.Marshal(trackID, packet)
-			c.send += len(buf)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(buf)
 
 			return nil
@@ -139,14 +150,33 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			push = wrapper(push)
 		}
 
+		return track.Bind(push)
+
+	case streamer.CodecOpus, streamer.CodecMP3, streamer.CodecPCMU, streamer.CodecPCMA:
+		push := func(packet *rtp.Packet) error {
+			if c.wait != waitNone {
+				return nil
+			}
+
+			buf := c.muxer.Marshal(trackID, packet)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(buf)
+
+			return nil
+		}
+
 		return track.Bind(push)
 	}
 
 	panic("unsupported codec")
 }
 
+func (c *Consumer) MimeCodecs() string {
+	return c.muxer.MimeCodecs(c.codecs)
+}
+
 func (c *Consumer) MimeType() string {
-	return c.muxer.MimeType(c.codecs)
+	return `video/mp4; codecs="` + c.MimeCodecs() + `"`
 }
 
 func (c *Consumer) Init() ([]byte, error) {
@@ -163,12 +193,11 @@ func (c *Consumer) Start() {
 //
 
 func (c *Consumer) MarshalJSON() ([]byte, error) {
-	v := map[string]interface{}{
-		"type":        "MP4 server consumer",
-		"send":        c.send,
-		"remote_addr": c.RemoteAddr,
-		"user_agent":  c.UserAgent,
+	info := &streamer.Info{
+		Type:       "MP4 client",
+		RemoteAddr: c.RemoteAddr,
+		UserAgent:  c.UserAgent,
+		Send:       atomic.LoadUint32(&c.send),
 	}
-
-	return json.Marshal(v)
+	return json.Marshal(info)
 }

pkg/mp4/muxer.go

@@ -1,17 +1,13 @@
 package mp4
 
 import (
-	"encoding/binary"
 	"encoding/hex"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/h265"
+	"github.com/AlexxIT/go2rtc/pkg/iso"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"github.com/deepch/vdk/av"
 	"github.com/deepch/vdk/codec/h264parser"
 	"github.com/deepch/vdk/codec/h265parser"
-	"github.com/deepch/vdk/format/fmp4/fmp4io"
-	"github.com/deepch/vdk/format/mp4/mp4io"
-	"github.com/deepch/vdk/format/mp4f/mp4fio"
 	"github.com/pion/rtp"
 )
 
@@ -21,8 +17,15 @@ type Muxer struct {
 	pts       []uint32
 }
 
-func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
-	s := `video/mp4; codecs="`
+const (
+	MimeH264 = "avc1.640029"
+	MimeH265 = "hvc1.1.6.L153.B0"
+	MimeAAC  = "mp4a.40.2"
+	MimeOpus = "opus"
+)
+
+func (m *Muxer) MimeCodecs(codecs []*streamer.Codec) string {
+	var s string
 
 	for i, codec := range codecs {
 		if i > 0 {
@@ -35,17 +38,23 @@ func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
 		case streamer.CodecH265:
 			// H.265 profile=main level=5.1
 			// hvc1 - supported in Safari, hev1 - doesn't, both supported in Chrome
-			s += "hvc1.1.6.L153.B0"
+			s += MimeH265
 		case streamer.CodecAAC:
-			s += "mp4a.40.2"
+			s += MimeAAC
+		case streamer.CodecOpus:
+			s += MimeOpus
 		}
 	}
 
-	return s + `"`
+	return s
 }
 
 func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
-	moov := MOOV()
+	mv := iso.NewMovie(1024)
+	mv.WriteFileType()
+
+	mv.StartAtom(iso.Moov)
+	mv.WriteMovieHeader()
 
 	for i, codec := range codecs {
 		switch codec.Name {
@@ -62,35 +71,11 @@ func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
 				return nil, err
 			}
 
-			width := codecData.Width()
-			height := codecData.Height()
-
-			trak := TRAK(i + 1)
-			trak.Header.TrackWidth = float64(width)
-			trak.Header.TrackHeight = float64(height)
-			trak.Media.Header.TimeScale = int32(codec.ClockRate)
-			trak.Media.Handler = &mp4io.HandlerRefer{
-				SubType: [4]byte{'v', 'i', 'd', 'e'},
-				Name:    []byte{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 'm', 'a', 'i', 'n', 0},
-			}
-			trak.Media.Info.Video = &mp4io.VideoMediaInfo{
-				Flags: 0x000001,
-			}
-			trak.Media.Info.Sample.SampleDesc.AVC1Desc = &mp4io.AVC1Desc{
-				DataRefIdx:           1,
-				HorizontalResolution: 72,
-				VorizontalResolution: 72,
-				Width:                int16(width),
-				Height:               int16(height),
-				FrameCount:           1,
-				Depth:                24,
-				ColorTableId:         -1,
-				Conf: &mp4io.AVC1Conf{
-					Data: codecData.AVCDecoderConfRecordBytes(),
-				},
-			}
-
-			moov.Tracks = append(moov.Tracks, trak)
+			mv.WriteVideoTrack(
+				uint32(i+1), codec.Name, codec.ClockRate,
+				uint16(codecData.Width()), uint16(codecData.Height()),
+				codecData.AVCDecoderConfRecordBytes(),
+			)
 
 		case streamer.CodecH265:
 			vps, sps, pps := h265.GetParameterSet(codec.FmtpLine)
@@ -106,35 +91,11 @@ func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
 				return nil, err
 			}
 
-			width := codecData.Width()
-			height := codecData.Height()
-
-			trak := TRAK(i + 1)
-			trak.Header.TrackWidth = float64(width)
-			trak.Header.TrackHeight = float64(height)
-			trak.Media.Header.TimeScale = int32(codec.ClockRate)
-			trak.Media.Handler = &mp4io.HandlerRefer{
-				SubType: [4]byte{'v', 'i', 'd', 'e'},
-				Name:    []byte{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 'm', 'a', 'i', 'n', 0},
-			}
-			trak.Media.Info.Video = &mp4io.VideoMediaInfo{
-				Flags: 0x000001,
-			}
-			trak.Media.Info.Sample.SampleDesc.HV1Desc = &mp4io.HV1Desc{
-				DataRefIdx:           1,
-				HorizontalResolution: 72,
-				VorizontalResolution: 72,
-				Width:                int16(width),
-				Height:               int16(height),
-				FrameCount:           1,
-				Depth:                24,
-				ColorTableId:         -1,
-				Conf: &mp4io.HV1Conf{
-					Data: codecData.AVCDecoderConfRecordBytes(),
-				},
-			}
-
-			moov.Tracks = append(moov.Tracks, trak)
+			mv.WriteVideoTrack(
+				uint32(i+1), codec.Name, codec.ClockRate,
+				uint16(codecData.Width()), uint16(codecData.Height()),
+				codecData.AVCDecoderConfRecordBytes(),
+			)
 
 		case streamer.CodecAAC:
 			s := streamer.Between(codec.FmtpLine, "config=", ";")
@@ -143,44 +104,29 @@ func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
 				return nil, err
 			}
 
-			trak := TRAK(i + 1)
-			trak.Header.AlternateGroup = 1
-			trak.Header.Duration = 0
-			trak.Header.Volume = 1
-			trak.Media.Header.TimeScale = int32(codec.ClockRate)
+			mv.WriteAudioTrack(
+				uint32(i+1), codec.Name, codec.ClockRate, codec.Channels, b,
+			)
 
-			trak.Media.Handler = &mp4io.HandlerRefer{
-				SubType: [4]byte{'s', 'o', 'u', 'n'},
-				Name:    []byte{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 'm', 'a', 'i', 'n', 0},
-			}
-			trak.Media.Info.Sound = &mp4io.SoundMediaInfo{}
-
-			trak.Media.Info.Sample.SampleDesc.MP4ADesc = &mp4io.MP4ADesc{
-				DataRefIdx:       1,
-				NumberOfChannels: int16(codec.Channels),
-				SampleSize:       int16(av.FLTP.BytesPerSample() * 4),
-				SampleRate:       float64(codec.ClockRate),
-				Unknowns:         []mp4io.Atom{ESDS(b)},
-			}
-
-			moov.Tracks = append(moov.Tracks, trak)
+		case streamer.CodecOpus, streamer.CodecMP3, streamer.CodecPCMU, streamer.CodecPCMA:
+			mv.WriteAudioTrack(
+				uint32(i+1), codec.Name, codec.ClockRate, codec.Channels, nil,
+			)
 		}
 
-		trex := &mp4io.TrackExtend{
-			TrackId:               uint32(i + 1),
-			DefaultSampleDescIdx:  1,
-			DefaultSampleDuration: 0,
-		}
-		moov.MovieExtend.Tracks = append(moov.MovieExtend.Tracks, trex)
-
 		m.pts = append(m.pts, 0)
 		m.dts = append(m.dts, 0)
 	}
 
-	data := make([]byte, moov.Len())
-	moov.Marshal(data)
+	mv.StartAtom(iso.MoovMvex)
+	for i := range codecs {
+		mv.WriteTrackExtend(uint32(i + 1))
+	}
+	mv.EndAtom() // MVEX
 
-	return append(FTYP(), data...), nil
+	mv.EndAtom() // MOOV
+
+	return mv.Bytes(), nil
 }
 
 func (m *Muxer) Reset() {
@@ -192,65 +138,28 @@ func (m *Muxer) Reset() {
 }
 
 func (m *Muxer) Marshal(trackID byte, packet *rtp.Packet) []byte {
-	run := &mp4fio.TrackFragRun{
-		Flags:            0x000b05,
-		FirstSampleFlags: uint32(fmp4io.SampleNoDependencies),
-		DataOffset:       0,
-		Entries:          []mp4io.TrackFragRunEntry{},
-	}
-
-	moof := &mp4fio.MovieFrag{
-		Header: &mp4fio.MovieFragHeader{
-			Seqnum: m.fragIndex + 1,
-		},
-		Tracks: []*mp4fio.TrackFrag{
-			{
-				Header: &mp4fio.TrackFragHeader{
-					Data: []byte{0x00, 0x02, 0x00, 0x20, 0x00, 0x00, 0x00, trackID + 1, 0x01, 0x01, 0x00, 0x00},
-				},
-				DecodeTime: &mp4fio.TrackFragDecodeTime{
-					Version: 1,
-					Flags:   0,
-					Time:    m.dts[trackID],
-				},
-				Run: run,
-			},
-		},
-	}
-
-	entry := mp4io.TrackFragRunEntry{
-		Size: uint32(len(packet.Payload)),
-	}
-
-	newTime := packet.Timestamp
-	if m.pts[trackID] > 0 {
-		entry.Duration = newTime - m.pts[trackID]
-		m.dts[trackID] += uint64(entry.Duration)
-	} else {
-		// important, or Safari will fail with first frame
-		entry.Duration = 1
-	}
-	m.pts[trackID] = newTime
-
-	// important before moof.Len()
-	run.Entries = append(run.Entries, entry)
-
-	moofLen := moof.Len()
-	mdatLen := 8 + len(packet.Payload)
-
-	// important after moof.Len()
-	run.DataOffset = uint32(moofLen + 8)
-
-	buf := make([]byte, moofLen+mdatLen)
-	moof.Marshal(buf)
-
-	binary.BigEndian.PutUint32(buf[moofLen:], uint32(mdatLen))
-	copy(buf[moofLen+4:], "mdat")
-	copy(buf[moofLen+8:], packet.Payload)
+	// important before increment
+	time := m.dts[trackID]
 
 	m.fragIndex++
 
-	//m.total += moofLen + mdatLen
+	var duration uint32
+	newTime := packet.Timestamp
+	if m.pts[trackID] > 0 {
+		duration = newTime - m.pts[trackID]
+		m.dts[trackID] += uint64(duration)
+	} else {
+		// important, or Safari will fail with first frame
+		duration = 1
+	}
+	m.pts[trackID] = newTime
 
-	return buf
+	mv := iso.NewMovie(1024 + len(packet.Payload))
+	mv.WriteMovieFragment(
+		m.fragIndex, uint32(trackID+1), duration,
+		uint32(len(packet.Payload)), time,
+	)
+	mv.WriteData(packet.Payload)
+
+	return mv.Bytes()
 }

pkg/mp4/segment.go

@@ -1,18 +1,25 @@
 package mp4
 
 import (
+	"encoding/json"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/h265"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/rtp"
+	"sync/atomic"
 )
 
 type Segment struct {
 	streamer.Element
 
-	Medias       []*streamer.Media
+	Medias     []*streamer.Media
+	UserAgent  string
+	RemoteAddr string
+
 	MimeType     string
 	OnlyKeyframe bool
+
+	send uint32
 }
 
 func (c *Segment) GetMedias() []*streamer.Media {
@@ -43,7 +50,7 @@ func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *stream
 		return nil
 	}
 
-	c.MimeType = muxer.MimeType(codecs)
+	c.MimeType = `video/mp4; codecs="` + muxer.MimeCodecs(codecs) + `"`
 
 	switch track.Codec.Name {
 	case streamer.CodecH264:
@@ -56,6 +63,7 @@ func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *stream
 				}
 
 				buf := muxer.Marshal(0, packet)
+				atomic.AddUint32(&c.send, uint32(len(buf)))
 				c.Fire(append(init, buf...))
 
 				return nil
@@ -73,6 +81,7 @@ func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *stream
 						buf = append(buf, b...)
 					}
 
+					atomic.AddUint32(&c.send, uint32(len(buf)))
 					c.Fire(buf)
 
 					buf = buf[:0]
@@ -106,6 +115,7 @@ func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *stream
 			}
 
 			buf := muxer.Marshal(0, packet)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(append(init, buf...))
 
 			return nil
@@ -121,3 +131,13 @@ func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *stream
 
 	panic("unsupported codec")
 }
+
+func (c *Segment) MarshalJSON() ([]byte, error) {
+	info := &streamer.Info{
+		Type:       "WS/MP4 client",
+		RemoteAddr: c.RemoteAddr,
+		UserAgent:  c.UserAgent,
+		Send:       atomic.LoadUint32(&c.send),
+	}
+	return json.Marshal(info)
+}

pkg/mp4/v1/consumer.go

@@ -1,7 +1,8 @@
-package mp4f
+package mp4
 
 import (
-	"encoding/json"
+	"encoding/hex"
+	"github.com/AlexxIT/go2rtc/pkg/aac"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/deepch/vdk/av"
@@ -15,6 +16,7 @@ import (
 type Consumer struct {
 	streamer.Element
 
+	Medias     []*streamer.Media
 	UserAgent  string
 	RemoteAddr string
 
@@ -27,6 +29,10 @@ type Consumer struct {
 }
 
 func (c *Consumer) GetMedias() []*streamer.Media {
+	if c.Medias != nil {
+		return c.Medias
+	}
+
 	return []*streamer.Media{
 		{
 			Kind:      streamer.KindVideo,
@@ -89,7 +95,7 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			return nil
 		}
 
-		if !codec.IsRAW() {
+		if codec.IsRTP() {
 			wrapper := h264.RTPDepay(track)
 			push = wrapper(push)
 		}
@@ -97,7 +103,17 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 		return track.Bind(push)
 
 	case streamer.CodecAAC:
-		stream, _ := aacparser.NewCodecDataFromMPEG4AudioConfigBytes([]byte{20, 8})
+		s := streamer.Between(codec.FmtpLine, "config=", ";")
+
+		b, err := hex.DecodeString(s)
+		if err != nil {
+			return nil
+		}
+
+		stream, err := aacparser.NewCodecDataFromMPEG4AudioConfigBytes(b)
+		if err != nil {
+			return nil
+		}
 
 		c.mimeType += ",mp4a.40.2"
 		c.streams = append(c.streams, stream)
@@ -127,6 +143,11 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			return nil
 		}
 
+		if codec.IsRTP() {
+			wrapper := aac.RTPDepay(track)
+			push = wrapper(push)
+		}
+
 		return track.Bind(push)
 	}
 
@@ -149,16 +170,3 @@ func (c *Consumer) Init() ([]byte, error) {
 func (c *Consumer) Start() {
 	c.start = true
 }
-
-//
-
-func (c *Consumer) MarshalJSON() ([]byte, error) {
-	v := map[string]interface{}{
-		"type":        "MSE server consumer",
-		"send":        c.send,
-		"remote_addr": c.RemoteAddr,
-		"user_agent":  c.UserAgent,
-	}
-
-	return json.Marshal(v)
-}

pkg/mp4/v2/consumer.go

@@ -0,0 +1,174 @@
+package mp4
+
+import (
+	"encoding/json"
+	"github.com/AlexxIT/go2rtc/pkg/aac"
+	"github.com/AlexxIT/go2rtc/pkg/h264"
+	"github.com/AlexxIT/go2rtc/pkg/h265"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/rtp"
+	"sync/atomic"
+)
+
+type Consumer struct {
+	streamer.Element
+
+	Medias     []*streamer.Media
+	UserAgent  string
+	RemoteAddr string
+
+	muxer  *Muxer
+	codecs []*streamer.Codec
+	wait   byte
+
+	send uint32
+}
+
+const (
+	waitNone byte = iota
+	waitKeyframe
+	waitInit
+)
+
+func (c *Consumer) GetMedias() []*streamer.Media {
+	if c.Medias != nil {
+		return c.Medias
+	}
+
+	// default medias
+	return []*streamer.Media{
+		{
+			Kind:      streamer.KindVideo,
+			Direction: streamer.DirectionRecvonly,
+			Codecs: []*streamer.Codec{
+				{Name: streamer.CodecH264},
+				{Name: streamer.CodecH265},
+			},
+		},
+		{
+			Kind:      streamer.KindAudio,
+			Direction: streamer.DirectionRecvonly,
+			Codecs: []*streamer.Codec{
+				{Name: streamer.CodecAAC},
+			},
+		},
+	}
+}
+
+func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
+	trackID := byte(len(c.codecs))
+	c.codecs = append(c.codecs, track.Codec)
+
+	codec := track.Codec
+	switch codec.Name {
+	case streamer.CodecH264:
+		c.wait = waitInit
+
+		push := func(packet *rtp.Packet) error {
+			if packet.Version != h264.RTPPacketVersionAVC {
+				return nil
+			}
+
+			if c.wait != waitNone {
+				if c.wait == waitInit || !h264.IsKeyframe(packet.Payload) {
+					return nil
+				}
+				c.wait = waitNone
+			}
+
+			buf := c.muxer.Marshal(trackID, packet)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(buf)
+
+			return nil
+		}
+
+		var wrapper streamer.WrapperFunc
+		if codec.IsRTP() {
+			wrapper = h264.RTPDepay(track)
+		} else {
+			wrapper = h264.RepairAVC(track)
+		}
+		push = wrapper(push)
+
+		return track.Bind(push)
+
+	case streamer.CodecH265:
+		c.wait = waitInit
+
+		push := func(packet *rtp.Packet) error {
+			if packet.Version != h264.RTPPacketVersionAVC {
+				return nil
+			}
+
+			if c.wait != waitNone {
+				if c.wait == waitInit || !h265.IsKeyframe(packet.Payload) {
+					return nil
+				}
+				c.wait = waitNone
+			}
+
+			buf := c.muxer.Marshal(trackID, packet)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(buf)
+
+			return nil
+		}
+
+		if codec.IsRTP() {
+			wrapper := h265.RTPDepay(track)
+			push = wrapper(push)
+		}
+
+		return track.Bind(push)
+
+	case streamer.CodecAAC:
+		push := func(packet *rtp.Packet) error {
+			if c.wait != waitNone {
+				return nil
+			}
+
+			buf := c.muxer.Marshal(trackID, packet)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(buf)
+
+			return nil
+		}
+
+		if codec.IsRTP() {
+			wrapper := aac.RTPDepay(track)
+			push = wrapper(push)
+		}
+
+		return track.Bind(push)
+	}
+
+	panic("unsupported codec")
+}
+
+func (c *Consumer) MimeType() string {
+	return c.muxer.MimeType(c.codecs)
+}
+
+func (c *Consumer) Init() ([]byte, error) {
+	c.muxer = &Muxer{}
+	return c.muxer.GetInit(c.codecs)
+}
+
+func (c *Consumer) Start() {
+	if c.wait == waitInit {
+		c.wait = waitKeyframe
+	}
+}
+
+//
+
+func (c *Consumer) MarshalJSON() ([]byte, error) {
+	info := &streamer.Info{
+		Type:       "MP4 client",
+		RemoteAddr: c.RemoteAddr,
+		UserAgent:  c.UserAgent,
+		Send:       atomic.LoadUint32(&c.send),
+	}
+	return json.Marshal(info)
+}

pkg/mp4/v2/muxer.go

@@ -0,0 +1,256 @@
+package mp4
+
+import (
+	"encoding/binary"
+	"encoding/hex"
+	"github.com/AlexxIT/go2rtc/pkg/h264"
+	"github.com/AlexxIT/go2rtc/pkg/h265"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/deepch/vdk/av"
+	"github.com/deepch/vdk/codec/h264parser"
+	"github.com/deepch/vdk/codec/h265parser"
+	"github.com/deepch/vdk/format/fmp4/fmp4io"
+	"github.com/deepch/vdk/format/mp4/mp4io"
+	"github.com/deepch/vdk/format/mp4f/mp4fio"
+	"github.com/pion/rtp"
+)
+
+type Muxer struct {
+	fragIndex uint32
+	dts       []uint64
+	pts       []uint32
+}
+
+func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
+	s := `video/mp4; codecs="`
+
+	for i, codec := range codecs {
+		if i > 0 {
+			s += ","
+		}
+
+		switch codec.Name {
+		case streamer.CodecH264:
+			s += "avc1." + h264.GetProfileLevelID(codec.FmtpLine)
+		case streamer.CodecH265:
+			// H.265 profile=main level=5.1
+			// hvc1 - supported in Safari, hev1 - doesn't, both supported in Chrome
+			s += "hvc1.1.6.L153.B0"
+		case streamer.CodecAAC:
+			s += "mp4a.40.2"
+		}
+	}
+
+	return s + `"`
+}
+
+func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
+	moov := MOOV()
+
+	for i, codec := range codecs {
+		switch codec.Name {
+		case streamer.CodecH264:
+			sps, pps := h264.GetParameterSet(codec.FmtpLine)
+			if sps == nil {
+				// some dummy SPS and PPS not a problem
+				sps = []byte{0x67, 0x42, 0x00, 0x0a, 0xf8, 0x41, 0xa2}
+				pps = []byte{0x68, 0xce, 0x38, 0x80}
+			}
+
+			codecData, err := h264parser.NewCodecDataFromSPSAndPPS(sps, pps)
+			if err != nil {
+				return nil, err
+			}
+
+			width := codecData.Width()
+			height := codecData.Height()
+
+			trak := TRAK(i + 1)
+			trak.Header.TrackWidth = float64(width)
+			trak.Header.TrackHeight = float64(height)
+			trak.Media.Header.TimeScale = int32(codec.ClockRate)
+			trak.Media.Handler = &mp4io.HandlerRefer{
+				SubType: [4]byte{'v', 'i', 'd', 'e'},
+				Name:    []byte{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 'm', 'a', 'i', 'n', 0},
+			}
+			trak.Media.Info.Video = &mp4io.VideoMediaInfo{
+				Flags: 0x000001,
+			}
+			trak.Media.Info.Sample.SampleDesc.AVC1Desc = &mp4io.AVC1Desc{
+				DataRefIdx:           1,
+				HorizontalResolution: 72,
+				VorizontalResolution: 72,
+				Width:                int16(width),
+				Height:               int16(height),
+				FrameCount:           1,
+				Depth:                24,
+				ColorTableId:         -1,
+				Conf: &mp4io.AVC1Conf{
+					Data: codecData.AVCDecoderConfRecordBytes(),
+				},
+			}
+
+			moov.Tracks = append(moov.Tracks, trak)
+
+		case streamer.CodecH265:
+			vps, sps, pps := h265.GetParameterSet(codec.FmtpLine)
+			if sps == nil {
+				// some dummy SPS and PPS not a problem
+				vps = []byte{0x40, 0x01, 0x0c, 0x01, 0xff, 0xff, 0x01, 0x40, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x99, 0xac, 0x09}
+				sps = []byte{0x42, 0x01, 0x01, 0x01, 0x40, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x99, 0xa0, 0x01, 0x40, 0x20, 0x05, 0xa1, 0xfe, 0x5a, 0xee, 0x46, 0xc1, 0xae, 0x55, 0x04}
+				pps = []byte{0x44, 0x01, 0xc0, 0x73, 0xc0, 0x4c, 0x90}
+			}
+
+			codecData, err := h265parser.NewCodecDataFromVPSAndSPSAndPPS(vps, sps, pps)
+			if err != nil {
+				return nil, err
+			}
+
+			width := codecData.Width()
+			height := codecData.Height()
+
+			trak := TRAK(i + 1)
+			trak.Header.TrackWidth = float64(width)
+			trak.Header.TrackHeight = float64(height)
+			trak.Media.Header.TimeScale = int32(codec.ClockRate)
+			trak.Media.Handler = &mp4io.HandlerRefer{
+				SubType: [4]byte{'v', 'i', 'd', 'e'},
+				Name:    []byte{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 'm', 'a', 'i', 'n', 0},
+			}
+			trak.Media.Info.Video = &mp4io.VideoMediaInfo{
+				Flags: 0x000001,
+			}
+			trak.Media.Info.Sample.SampleDesc.HV1Desc = &mp4io.HV1Desc{
+				DataRefIdx:           1,
+				HorizontalResolution: 72,
+				VorizontalResolution: 72,
+				Width:                int16(width),
+				Height:               int16(height),
+				FrameCount:           1,
+				Depth:                24,
+				ColorTableId:         -1,
+				Conf: &mp4io.HV1Conf{
+					Data: codecData.AVCDecoderConfRecordBytes(),
+				},
+			}
+
+			moov.Tracks = append(moov.Tracks, trak)
+
+		case streamer.CodecAAC:
+			s := streamer.Between(codec.FmtpLine, "config=", ";")
+			b, err := hex.DecodeString(s)
+			if err != nil {
+				return nil, err
+			}
+
+			trak := TRAK(i + 1)
+			trak.Header.AlternateGroup = 1
+			trak.Header.Duration = 0
+			trak.Header.Volume = 1
+			trak.Media.Header.TimeScale = int32(codec.ClockRate)
+
+			trak.Media.Handler = &mp4io.HandlerRefer{
+				SubType: [4]byte{'s', 'o', 'u', 'n'},
+				Name:    []byte{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 'm', 'a', 'i', 'n', 0},
+			}
+			trak.Media.Info.Sound = &mp4io.SoundMediaInfo{}
+
+			trak.Media.Info.Sample.SampleDesc.MP4ADesc = &mp4io.MP4ADesc{
+				DataRefIdx:       1,
+				NumberOfChannels: int16(codec.Channels),
+				SampleSize:       int16(av.FLTP.BytesPerSample() * 4),
+				SampleRate:       float64(codec.ClockRate),
+				Unknowns:         []mp4io.Atom{ESDS(b)},
+			}
+
+			moov.Tracks = append(moov.Tracks, trak)
+		}
+
+		trex := &mp4io.TrackExtend{
+			TrackId:               uint32(i + 1),
+			DefaultSampleDescIdx:  1,
+			DefaultSampleDuration: 0,
+		}
+		moov.MovieExtend.Tracks = append(moov.MovieExtend.Tracks, trex)
+
+		m.pts = append(m.pts, 0)
+		m.dts = append(m.dts, 0)
+	}
+
+	data := make([]byte, moov.Len())
+	moov.Marshal(data)
+
+	return append(FTYP(), data...), nil
+}
+
+func (m *Muxer) Reset() {
+	m.fragIndex = 0
+	for i := range m.dts {
+		m.dts[i] = 0
+		m.pts[i] = 0
+	}
+}
+
+func (m *Muxer) Marshal(trackID byte, packet *rtp.Packet) []byte {
+	run := &mp4fio.TrackFragRun{
+		Flags:            0x000b05,
+		FirstSampleFlags: uint32(fmp4io.SampleNoDependencies),
+		DataOffset:       0,
+		Entries:          []mp4io.TrackFragRunEntry{},
+	}
+
+	moof := &mp4fio.MovieFrag{
+		Header: &mp4fio.MovieFragHeader{
+			Seqnum: m.fragIndex + 1,
+		},
+		Tracks: []*mp4fio.TrackFrag{
+			{
+				Header: &mp4fio.TrackFragHeader{
+					Data: []byte{0x00, 0x02, 0x00, 0x20, 0x00, 0x00, 0x00, trackID + 1, 0x01, 0x01, 0x00, 0x00},
+				},
+				DecodeTime: &mp4fio.TrackFragDecodeTime{
+					Version: 1,
+					Flags:   0,
+					Time:    m.dts[trackID],
+				},
+				Run: run,
+			},
+		},
+	}
+
+	entry := mp4io.TrackFragRunEntry{
+		Size: uint32(len(packet.Payload)),
+	}
+
+	newTime := packet.Timestamp
+	if m.pts[trackID] > 0 {
+		entry.Duration = newTime - m.pts[trackID]
+		m.dts[trackID] += uint64(entry.Duration)
+	} else {
+		// important, or Safari will fail with first frame
+		entry.Duration = 1
+	}
+	m.pts[trackID] = newTime
+
+	// important before moof.Len()
+	run.Entries = append(run.Entries, entry)
+
+	moofLen := moof.Len()
+	mdatLen := 8 + len(packet.Payload)
+
+	// important after moof.Len()
+	run.DataOffset = uint32(moofLen + 8)
+
+	buf := make([]byte, moofLen+mdatLen)
+	moof.Marshal(buf)
+
+	binary.BigEndian.PutUint32(buf[moofLen:], uint32(mdatLen))
+	copy(buf[moofLen+4:], "mdat")
+	copy(buf[moofLen+8:], packet.Payload)
+
+	m.fragIndex++
+
+	//m.total += moofLen + mdatLen
+
+	return buf
+}

pkg/mp4/v2/segment.go

@@ -0,0 +1,143 @@
+package mp4
+
+import (
+	"encoding/json"
+	"github.com/AlexxIT/go2rtc/pkg/h264"
+	"github.com/AlexxIT/go2rtc/pkg/h265"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/rtp"
+	"sync/atomic"
+)
+
+type Segment struct {
+	streamer.Element
+
+	Medias     []*streamer.Media
+	UserAgent  string
+	RemoteAddr string
+
+	MimeType     string
+	OnlyKeyframe bool
+
+	send uint32
+}
+
+func (c *Segment) GetMedias() []*streamer.Media {
+	if c.Medias != nil {
+		return c.Medias
+	}
+
+	// default medias
+	return []*streamer.Media{
+		{
+			Kind:      streamer.KindVideo,
+			Direction: streamer.DirectionRecvonly,
+			Codecs: []*streamer.Codec{
+				{Name: streamer.CodecH264},
+				{Name: streamer.CodecH265},
+			},
+		},
+	}
+}
+
+func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
+	muxer := &Muxer{}
+
+	codecs := []*streamer.Codec{track.Codec}
+
+	init, err := muxer.GetInit(codecs)
+	if err != nil {
+		return nil
+	}
+
+	c.MimeType = muxer.MimeType(codecs)
+
+	switch track.Codec.Name {
+	case streamer.CodecH264:
+		var push streamer.WriterFunc
+
+		if c.OnlyKeyframe {
+			push = func(packet *rtp.Packet) error {
+				if !h264.IsKeyframe(packet.Payload) {
+					return nil
+				}
+
+				buf := muxer.Marshal(0, packet)
+				atomic.AddUint32(&c.send, uint32(len(buf)))
+				c.Fire(append(init, buf...))
+
+				return nil
+			}
+		} else {
+			var buf []byte
+
+			push = func(packet *rtp.Packet) error {
+				if h264.IsKeyframe(packet.Payload) {
+					// fist frame - send only IFrame
+					// other frames - send IFrame and all PFrames
+					if buf == nil {
+						buf = append(buf, init...)
+						b := muxer.Marshal(0, packet)
+						buf = append(buf, b...)
+					}
+
+					atomic.AddUint32(&c.send, uint32(len(buf)))
+					c.Fire(buf)
+
+					buf = buf[:0]
+					buf = append(buf, init...)
+					muxer.Reset()
+				}
+
+				if buf != nil {
+					b := muxer.Marshal(0, packet)
+					buf = append(buf, b...)
+				}
+
+				return nil
+			}
+		}
+
+		var wrapper streamer.WrapperFunc
+		if track.Codec.IsRTP() {
+			wrapper = h264.RTPDepay(track)
+		} else {
+			wrapper = h264.RepairAVC(track)
+		}
+		push = wrapper(push)
+
+		return track.Bind(push)
+
+	case streamer.CodecH265:
+		push := func(packet *rtp.Packet) error {
+			if !h265.IsKeyframe(packet.Payload) {
+				return nil
+			}
+
+			buf := muxer.Marshal(0, packet)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(append(init, buf...))
+
+			return nil
+		}
+
+		if track.Codec.IsRTP() {
+			wrapper := h265.RTPDepay(track)
+			push = wrapper(push)
+		}
+
+		return track.Bind(push)
+	}
+
+	panic("unsupported codec")
+}
+
+func (c *Segment) MarshalJSON() ([]byte, error) {
+	info := &streamer.Info{
+		Type:       "WS/MP4 client",
+		RemoteAddr: c.RemoteAddr,
+		UserAgent:  c.UserAgent,
+		Send:       atomic.LoadUint32(&c.send),
+	}
+	return json.Marshal(info)
+}

pkg/rtmp/client.go

@@ -12,6 +12,7 @@ import (
 	"github.com/deepch/vdk/format/rtmp"
 	"github.com/pion/rtp"
 	"net/http"
+	"sync/atomic"
 	"time"
 )
 
@@ -33,7 +34,7 @@ type Client struct {
 	conn   Conn
 	closed bool
 
-	receive int
+	recv uint32
 }
 
 func NewClient(uri string) *Client {
@@ -138,7 +139,7 @@ func (c *Client) Handle() (err error) {
 			return
 		}
 
-		c.receive += len(pkt.Data)
+		atomic.AddUint32(&c.recv, uint32(len(pkt.Data)))
 
 		track := c.tracks[int(pkt.Idx)]
 

pkg/rtmp/streamer.go

@@ -4,7 +4,7 @@ import (
 	"encoding/json"
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"strconv"
+	"sync/atomic"
 )
 
 func (c *Client) GetMedias() []*streamer.Media {
@@ -29,19 +29,12 @@ func (c *Client) Stop() error {
 }
 
 func (c *Client) MarshalJSON() ([]byte, error) {
-	v := map[string]interface{}{
-		streamer.JSONReceive:    c.receive,
-		streamer.JSONType:       "RTMP client producer",
-		//streamer.JSONRemoteAddr: c.conn.NetConn().RemoteAddr().String(),
-		"url":                   c.URI,
+	info := &streamer.Info{
+		Type:   "RTMP source",
+		URL:    c.URI,
+		Medias: c.medias,
+		Tracks: c.tracks,
+		Recv:   atomic.LoadUint32(&c.recv),
 	}
-	for i, media := range c.medias {
-		k := "media:" + strconv.Itoa(i)
-		v[k] = media.String()
-	}
-	for i, track := range c.tracks {
-		k := "track:" + strconv.Itoa(i)
-		v[k] = track.String()
-	}
-	return json.Marshal(v)
+	return json.Marshal(info)
 }

pkg/rtsp/conn.go

@@ -2,7 +2,6 @@ package rtsp
 
 import (
 	"bufio"
-	"bytes"
 	"crypto/tls"
 	"encoding/binary"
 	"errors"
@@ -20,6 +19,7 @@ import (
 	"net/url"
 	"strconv"
 	"strings"
+	"sync"
 	"time"
 )
 
@@ -47,11 +47,28 @@ const (
 
 type State byte
 
+func (s State) String() string {
+	switch s {
+	case StateNone:
+		return "NONE"
+	case StateConn:
+		return "CONN"
+	case StateSetup:
+		return "SETUP"
+	case StatePlay:
+		return "PLAY"
+	case StateHandle:
+		return "HANDLE"
+	}
+	return strconv.Itoa(int(s))
+}
+
 const (
 	StateNone State = iota
 	StateConn
 	StateSetup
 	StatePlay
+	StateHandle
 )
 
 type Conn struct {
@@ -60,6 +77,7 @@ type Conn struct {
 	// public
 
 	Backchannel bool
+	SessionName string
 
 	Medias    []*streamer.Media
 	Session   string
@@ -72,6 +90,7 @@ type Conn struct {
 	conn     net.Conn
 	mode     Mode
 	state    State
+	stateMu  sync.Mutex
 	reader   *bufio.Reader
 	sequence int
 	uri      string
@@ -261,7 +280,7 @@ func (c *Conn) Options() error {
 	}
 
 	if val := res.Header.Get("Content-Base"); val != "" {
-		c.URL, err = url.Parse(val)
+		c.URL, err = urlParse(val)
 		if err != nil {
 			return err
 		}
@@ -285,13 +304,19 @@ func (c *Conn) Describe() error {
 		req.Header.Set("Require", "www.onvif.org/ver20/backchannel")
 	}
 
+	if c.UserAgent != "" {
+		// this camera will answer with 401 on DESCRIBE without User-Agent
+		// https://github.com/AlexxIT/go2rtc/issues/235
+		req.Header.Set("User-Agent", c.UserAgent)
+	}
+
 	res, err := c.Do(req)
 	if err != nil {
 		return err
 	}
 
 	if val := res.Header.Get("Content-Base"); val != "" {
-		c.URL, err = url.Parse(val)
+		c.URL, err = urlParse(val)
 		if err != nil {
 			return err
 		}
@@ -328,7 +353,7 @@ func (c *Conn) Describe() error {
 
 func (c *Conn) Setup() error {
 	for _, media := range c.Medias {
-		_, err := c.SetupMedia(media, media.Codecs[0])
+		_, err := c.SetupMedia(media, media.Codecs[0], true)
 		if err != nil {
 			return err
 		}
@@ -337,9 +362,17 @@ func (c *Conn) Setup() error {
 	return nil
 }
 
-func (c *Conn) SetupMedia(
-	media *streamer.Media, codec *streamer.Codec,
-) (*streamer.Track, error) {
+func (c *Conn) SetupMedia(media *streamer.Media, codec *streamer.Codec, first bool) (*streamer.Track, error) {
+	// TODO: rewrite recoonection and first flag
+	if first {
+		c.stateMu.Lock()
+		defer c.stateMu.Unlock()
+	}
+
+	if c.state != StateConn && c.state != StateSetup {
+		return nil, fmt.Errorf("RTSP SETUP from wrong state: %s", c.state)
+	}
+
 	ch := c.GetChannel(media)
 	if ch < 0 {
 		return nil, fmt.Errorf("wrong media: %v", media)
@@ -353,7 +386,7 @@ func (c *Conn) SetupMedia(
 		}
 		rawURL += media.Control
 	}
-	trackURL, err := url.Parse(rawURL)
+	trackURL, err := urlParse(rawURL)
 	if err != nil {
 		return nil, err
 	}
@@ -386,7 +419,7 @@ func (c *Conn) SetupMedia(
 
 			for _, newMedia := range c.Medias {
 				if newMedia.Control == media.Control {
-					return c.SetupMedia(newMedia, newMedia.Codecs[0])
+					return c.SetupMedia(newMedia, newMedia.Codecs[0], false)
 				}
 			}
 		}
@@ -461,12 +494,19 @@ func (c *Conn) SetupMedia(
 }
 
 func (c *Conn) Play() (err error) {
+	c.stateMu.Lock()
+	defer c.stateMu.Unlock()
+
 	if c.state != StateSetup {
 		return fmt.Errorf("RTSP PLAY from wrong state: %s", c.state)
 	}
 
 	req := &tcp.Request{Method: MethodPlay, URL: c.URL}
-	return c.Request(req)
+	if err = c.Request(req); err == nil {
+		c.state = StatePlay
+	}
+
+	return
 }
 
 func (c *Conn) Teardown() (err error) {
@@ -476,12 +516,14 @@ func (c *Conn) Teardown() (err error) {
 }
 
 func (c *Conn) Close() error {
+	c.stateMu.Lock()
+	defer c.stateMu.Unlock()
+
 	if c.state == StateNone {
 		return nil
 	}
-	if err := c.Teardown(); err != nil {
-		return err
-	}
+
+	_ = c.Teardown()
 	c.state = StateNone
 	return c.conn.Close()
 }
@@ -583,7 +625,7 @@ func (c *Conn) Accept() error {
 				medias = append(medias, media)
 			}
 
-			res.Body, err = streamer.MarshalSDP(medias)
+			res.Body, err = streamer.MarshalSDP(c.SessionName, medias)
 			if err != nil {
 				return err
 			}
@@ -614,7 +656,16 @@ func (c *Conn) Accept() error {
 
 		case MethodRecord, MethodPlay:
 			res := &tcp.Response{Request: req}
-			return c.Response(res)
+			if err = c.Response(res); err == nil {
+				c.state = StatePlay
+			}
+			return err
+
+		case MethodTeardown:
+			res := &tcp.Response{Request: req}
+			_ = c.Response(res)
+			c.state = StateNone
+			return c.conn.Close()
 
 		default:
 			return fmt.Errorf("unsupported method: %s", req.Method)
@@ -623,13 +674,31 @@ func (c *Conn) Accept() error {
 }
 
 func (c *Conn) Handle() (err error) {
-	if c.state != StateSetup {
-		return fmt.Errorf("RTSP Handle from wrong state: %d", c.state)
+	c.stateMu.Lock()
+
+	switch c.state {
+	case StateNone: // Close after PLAY and before Handle is OK (because SETUP after PLAY)
+	case StatePlay:
+		c.state = StateHandle
+	default:
+		err = fmt.Errorf("RTSP HANDLE from wrong state: %s", c.state)
+
+		c.state = StateNone
+		_ = c.conn.Close()
 	}
 
-	c.state = StatePlay
+	ok := c.state == StateHandle
+
+	c.stateMu.Unlock()
+
+	if !ok {
+		return
+	}
 
 	defer func() {
+		c.stateMu.Lock()
+		defer c.stateMu.Unlock()
+
 		if c.state == StateNone {
 			err = nil
 			return
@@ -680,6 +749,9 @@ func (c *Conn) Handle() (err error) {
 			return
 		}
 
+		var channelID byte
+		var size uint16
+
 		if buf4[0] != '$' {
 			switch string(buf4) {
 			case "RTSP":
@@ -688,26 +760,62 @@ func (c *Conn) Handle() (err error) {
 					return
 				}
 				c.Fire(res)
+				continue
+
 			case "OPTI", "TEAR", "DESC", "SETU", "PLAY", "PAUS", "RECO", "ANNO", "GET_", "SET_":
 				var req *tcp.Request
 				if req, err = tcp.ReadRequest(c.reader); err != nil {
 					return
 				}
 				c.Fire(req)
+				continue
+
 			default:
-				return fmt.Errorf("RTSP wrong input")
+				for i := 0; ; i++ {
+					// search next start symbol
+					if _, err = c.reader.ReadBytes('$'); err != nil {
+						return err
+					}
+
+					if channelID, err = c.reader.ReadByte(); err != nil {
+						return err
+					}
+
+					// check if channel ID exists
+					if c.channels[channelID] == nil {
+						continue
+					}
+
+					buf4 = make([]byte, 2)
+					if _, err = io.ReadFull(c.reader, buf4); err != nil {
+						return err
+					}
+
+					// check if size good for RTP
+					size = binary.BigEndian.Uint16(buf4)
+					if size <= 1500 {
+						break
+					}
+
+					// 10 tries to find good packet
+					if i >= 10 {
+						return fmt.Errorf("RTSP wrong input")
+					}
+				}
+
+				c.Fire("RTSP wrong input")
 			}
-			continue
-		}
+		} else {
+			// hope that the odd channels are always RTCP
+			channelID = buf4[1]
 
-		// hope that the odd channels are always RTCP
-		channelID := buf4[1]
+			// get data size
+			size = binary.BigEndian.Uint16(buf4[2:])
 
-		// get data size
-		size := int(binary.BigEndian.Uint16(buf4[2:]))
-
-		if _, err = c.reader.Discard(4); err != nil {
-			return
+			// skip 4 bytes from c.reader.Peek
+			if _, err = c.reader.Discard(4); err != nil {
+				return
+			}
 		}
 
 		// init memory for data
@@ -716,7 +824,7 @@ func (c *Conn) Handle() (err error) {
 			return
 		}
 
-		c.receive += size
+		c.receive += int(size)
 
 		if channelID&1 == 0 {
 			packet := &rtp.Packet{}
@@ -727,21 +835,19 @@ func (c *Conn) Handle() (err error) {
 			track := c.channels[channelID]
 			if track != nil {
 				_ = track.WriteRTP(packet)
-				//return fmt.Errorf("wrong channelID: %d", channelID)
 			} else {
-				continue // TODO: maybe fix this
-				//panic("wrong channelID")
+				//c.Fire("wrong channelID: " + strconv.Itoa(int(channelID)))
 			}
 		} else {
 			msg := &RTCP{Channel: channelID}
 
 			if err = msg.Header.Unmarshal(buf); err != nil {
-				return
+				continue
 			}
 
 			msg.Packets, err = rtcp.Unmarshal(buf)
 			if err != nil {
-				return
+				continue
 			}
 
 			c.Fire(msg)
@@ -819,42 +925,3 @@ func (c *Conn) bindTrack(
 
 	return track.Bind(push)
 }
-
-type RTCP struct {
-	Channel byte
-	Header  rtcp.Header
-	Packets []rtcp.Packet
-}
-
-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
-		}
-	}
-
-	// 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 medias, nil
-}

pkg/rtsp/helpers.go

@@ -0,0 +1,103 @@
+package rtsp
+
+import (
+	"bytes"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/rtcp"
+	"github.com/pion/sdp/v3"
+	"net/url"
+	"regexp"
+	"strconv"
+	"strings"
+)
+
+type RTCP struct {
+	Channel byte
+	Header  rtcp.Header
+	Packets []rtcp.Packet
+}
+
+const sdpHeader = `v=0
+o=- 0 0 IN IP4 0.0.0.0
+s=-
+t=0 0`
+
+func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
+	// fix bug from Reolink Doorbell
+	if i := bytes.Index(rawSDP, []byte("a=sendonlym=")); i > 0 {
+		rawSDP = append(rawSDP[:i+11], rawSDP[i+10:]...)
+		rawSDP[i+10] = '\n'
+	}
+
+	sd := &sdp.SessionDescription{}
+	if err := sd.Unmarshal(rawSDP); err != nil {
+		// fix multiple `s=` https://github.com/AlexxIT/WebRTC/issues/417
+		re, _ := regexp.Compile("\ns=[^\n]+")
+		rawSDP = re.ReplaceAll(rawSDP, nil)
+
+		// fix SDP header for some cameras
+		if i := bytes.Index(rawSDP, []byte("\nm=")); i > 0 {
+			rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
+			sd = &sdp.SessionDescription{}
+			err = sd.Unmarshal(rawSDP)
+		}
+
+		if err != nil {
+			return nil, err
+		}
+	}
+
+	medias := streamer.UnmarshalMedias(sd.MediaDescriptions)
+
+	for _, media := range medias {
+		// Check buggy SDP with fmtp for H264 on another track
+		// https://github.com/AlexxIT/WebRTC/issues/419
+		for _, codec := range media.Codecs {
+			if codec.Name == streamer.CodecH264 && codec.FmtpLine == "" {
+				codec.FmtpLine = findFmtpLine(codec.PayloadType, sd.MediaDescriptions)
+			}
+		}
+
+		// fix bug in ONVIF spec
+		// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
+		switch media.Direction {
+		case streamer.DirectionRecvonly, "":
+			media.Direction = streamer.DirectionSendonly
+		case streamer.DirectionSendonly:
+			media.Direction = streamer.DirectionRecvonly
+		}
+	}
+
+	return medias, nil
+}
+
+func findFmtpLine(payloadType uint8, descriptions []*sdp.MediaDescription) string {
+	s := strconv.Itoa(int(payloadType))
+	for _, md := range descriptions {
+		codec := streamer.UnmarshalCodec(md, s)
+		if codec.FmtpLine != "" {
+			return codec.FmtpLine
+		}
+	}
+	return ""
+}
+
+// urlParse fix bugs:
+// 1. Content-Base: rtsp://::ffff:192.168.1.123/onvif/profile.1/
+// 2. Content-Base: rtsp://rtsp://turret2-cam.lan:554/stream1/
+func urlParse(rawURL string) (*url.URL, error) {
+	if strings.HasPrefix(rawURL, "rtsp://rtsp://") {
+		rawURL = rawURL[7:]
+	}
+
+	u, err := url.Parse(rawURL)
+	if err != nil && strings.HasSuffix(err.Error(), "after host") {
+		if i1 := strings.Index(rawURL, "://"); i1 > 0 {
+			if i2 := strings.IndexByte(rawURL[i1+3:], '/'); i2 > 0 {
+				return urlParse(rawURL[:i1+3+i2] + ":" + rawURL[i1+3+i2:])
+			}
+		}
+	}
+
+	return u, err
+}

pkg/rtsp/rtsp_test.go

@@ -0,0 +1,108 @@
+package rtsp
+
+import (
+	"github.com/stretchr/testify/assert"
+	"testing"
+)
+
+func TestURLParse(t *testing.T) {
+	// https://github.com/AlexxIT/WebRTC/issues/395
+	base := "rtsp://::ffff:192.168.1.123/onvif/profile.1/"
+	u, err := urlParse(base)
+	assert.Empty(t, err)
+	assert.Equal(t, "::ffff:192.168.1.123:", u.Host)
+
+	// https://github.com/AlexxIT/go2rtc/issues/208
+	base = "rtsp://rtsp://turret2-cam.lan:554/stream1/"
+	u, err = urlParse(base)
+	assert.Empty(t, err)
+	assert.Equal(t, "turret2-cam.lan:554", u.Host)
+}
+
+func TestBugSDP1(t *testing.T) {
+	// https://github.com/AlexxIT/WebRTC/issues/417
+	s := `v=0
+o=- 91674849066 1 IN IP4 192.168.1.123
+s=RtspServer
+i=live
+t=0 0
+a=control:*
+a=range:npt=0-
+m=video 0 RTP/AVP 96
+c=IN IP4 0.0.0.0
+s=RtspServer
+i=live
+a=control:track0
+a=range:npt=0-
+a=rtpmap:96 H264/90000
+a=fmtp:96 packetization-mode=1;profile-level-id=42001E;sprop-parameter-sets=Z0IAHvQCgC3I,aM48gA==
+a=control:track0
+m=audio 0 RTP/AVP 97
+c=IN IP4 0.0.0.0
+s=RtspServer
+i=live
+a=control:track1
+a=range:npt=0-
+a=rtpmap:97 MPEG4-GENERIC/8000/1
+a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1588
+a=control:track1
+`
+	medias, err := UnmarshalSDP([]byte(s))
+	assert.Nil(t, err)
+	assert.NotNil(t, medias)
+}
+
+func TestBugSDP2(t *testing.T) {
+	// https://github.com/AlexxIT/WebRTC/issues/419
+	s := `v=0
+o=- 1675628282 1675628283 IN IP4 192.168.1.123
+s=streamed by the RTSP server
+t=0 0
+m=video 0 RTP/AVP 96
+a=rtpmap:96 H264/90000
+a=control:track0
+m=audio 0 RTP/AVP 8
+a=rtpmap:0 pcma/8000/1
+a=control:track1
+a=framerate:25
+a=range:npt=now-
+a=fmtp:96 packetization-mode=1;profile-level-id=64001F;sprop-parameter-sets=Z0IAH5WoFAFuQA==,aM48gA==
+`
+	medias, err := UnmarshalSDP([]byte(s))
+	assert.Nil(t, err)
+	assert.NotNil(t, medias)
+	assert.NotEqual(t, "", medias[0].Codecs[0].FmtpLine)
+}
+
+func TestBugSDP3(t *testing.T) {
+	s := `v=0
+o=- 1675775048103026 1 IN IP4 192.168.1.123
+s=Session streamed by "preview"
+t=0 0
+a=tool:LIVE555 Streaming Media v2020.08.12
+a=type:broadcast
+a=control:*
+a=range:npt=0-
+a=x-qt-text-nam:Session streamed by "preview"
+m=video 0 RTP/AVP 96
+c=IN IP4 0.0.0.0
+b=AS:8192
+a=rtpmap:96 H264/90000
+a=fmtp:96 packetization-mode=1;profile-level-id=640033;sprop-parameter-sets=Z2QAM6wVFKAoAPGQ,aO48sA==
+a=recvonly
+a=control:track1
+m=audio 0 RTP/AVP 8
+a=control:track2
+a=rtpmap:8 PCMA/8000
+a=sendonlym=audio 0 RTP/AVP 98
+c=IN IP4 0.0.0.0
+b=AS:8192
+a=rtpmap:98 MPEG4-GENERIC/16000
+a=fmtp:98 streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1408;
+a=recvonly
+a=control:track3
+`
+	medias, err := UnmarshalSDP([]byte(s))
+	assert.Nil(t, err)
+	assert.Len(t, medias, 3)
+}

pkg/rtsp/streamer.go

@@ -4,13 +4,31 @@ import (
 	"encoding/json"
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"strconv"
 )
 
 // Element Producer
 
 func (c *Conn) GetMedias() []*streamer.Media {
-	return c.Medias
+	if c.Medias != nil {
+		return c.Medias
+	}
+
+	return []*streamer.Media{
+		{
+			Kind:      streamer.KindVideo,
+			Direction: streamer.DirectionRecvonly,
+			Codecs: []*streamer.Codec{
+				{Name: streamer.CodecAll},
+			},
+		},
+		{
+			Kind:      streamer.KindAudio,
+			Direction: streamer.DirectionRecvonly,
+			Codecs: []*streamer.Codec{
+				{Name: streamer.CodecAll},
+			},
+		},
+	}
 }
 
 func (c *Conn) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.Track {
@@ -21,12 +39,13 @@ func (c *Conn) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.
 	}
 
 	// can't setup new tracks from play state - forcing a reconnection feature
-	if c.state == StatePlay {
+	switch c.state {
+	case StatePlay, StateHandle:
 		go c.Close()
 		return streamer.NewTrack(codec, media.Direction)
 	}
 
-	track, err := c.SetupMedia(media, codec)
+	track, err := c.SetupMedia(media, codec, true)
 	if err != nil {
 		return nil
 	}
@@ -63,11 +82,20 @@ func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.
 		codec := track.Codec.Clone()
 		codec.PayloadType = uint8(96 + i)
 
-		for i, m := range c.Medias {
-			if m == media {
-				media.Codecs = []*streamer.Codec{codec}
-				c.Medias[i] = media
-				break
+		if media.MatchAll() {
+			// fill consumer medias list
+			c.Medias = append(c.Medias, &streamer.Media{
+				Kind: media.Kind, Direction: media.Direction,
+				Codecs: []*streamer.Codec{codec},
+			})
+		} else {
+			// find consumer media and replace codec with right one
+			for i, m := range c.Medias {
+				if m == media {
+					media.Codecs = []*streamer.Codec{codec}
+					c.Medias[i] = media
+					break
+				}
 			}
 		}
 
@@ -87,40 +115,30 @@ func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.
 //
 
 func (c *Conn) MarshalJSON() ([]byte, error) {
-	v := map[string]interface{}{
-		streamer.JSONReceive: c.receive,
-		streamer.JSONSend:    c.send,
+	info := &streamer.Info{
+		UserAgent: c.UserAgent,
+		Medias:    c.Medias,
+		Tracks:    c.tracks,
+		Recv:      uint32(c.receive),
+		Send:      uint32(c.send),
 	}
+
 	switch c.mode {
 	case ModeUnknown:
-		v[streamer.JSONType] = "RTSP unknown"
-	case ModeClientProducer:
-		v[streamer.JSONType] = "RTSP client producer"
-	case ModeServerProducer:
-		v[streamer.JSONType] = "RTSP server producer"
+		info.Type = "RTSP unknown"
+	case ModeClientProducer, ModeServerProducer:
+		info.Type = "RTSP source"
 	case ModeServerConsumer:
-		v[streamer.JSONType] = "RTSP server consumer"
+		info.Type = "RTSP client"
 	}
-	//if c.URI != "" {
-	//	v["uri"] = c.URI
-	//}
+
 	if c.URL != nil {
-		v["url"] = c.URL.String()
+		info.URL = c.URL.String()
 	}
 	if c.conn != nil {
-		v[streamer.JSONRemoteAddr] = c.conn.RemoteAddr().String()
-	}
-	if c.UserAgent != "" {
-		v[streamer.JSONUserAgent] = c.UserAgent
-	}
-	for i, media := range c.Medias {
-		k := "media:" + strconv.Itoa(i)
-		v[k] = media.String()
-	}
-	for i, track := range c.tracks {
-		k := "track:" + strconv.Itoa(int(i>>1))
-		v[k] = track.String()
+		info.RemoteAddr = c.conn.RemoteAddr().String()
 	}
+
 	//for i, track := range c.tracks {
 	//	k := "track:" + strconv.Itoa(i+1)
 	//	if track.MimeType() == streamer.MimeTypeH264 {
@@ -129,5 +147,6 @@ func (c *Conn) MarshalJSON() ([]byte, error) {
 	//		v[k] = track.MimeType()
 	//	}
 	//}
-	return json.Marshal(v)
+
+	return json.Marshal(info)
 }

pkg/shell/env.go

@@ -0,0 +1,32 @@
+package shell
+
+import (
+	"os"
+	"regexp"
+	"strings"
+)
+
+func ReplaceEnvVars(text string) string {
+	re := regexp.MustCompile(`\${([^}{]+)}`)
+	return re.ReplaceAllStringFunc(text, func(match string) string {
+		key := match[2 : len(match)-1]
+
+		var def string
+		var dok bool
+
+		if i := strings.IndexByte(key, ':'); i > 0 {
+			key, def = key[:i], key[i+1:]
+			dok = true
+		}
+
+		if value, vok := os.LookupEnv(key); vok {
+			return value
+		}
+
+		if dok {
+			return def
+		}
+
+		return match
+	})
+}

pkg/srtp/server.go

@@ -3,6 +3,7 @@ package srtp
 import (
 	"encoding/binary"
 	"net"
+	"sync/atomic"
 )
 
 // Server using same UDP port for SRTP and for SRTCP as the iPhone does
@@ -55,6 +56,8 @@ func (s *Server) Serve(conn net.PacketConn) error {
 				}
 			}
 
+			atomic.AddUint32(&session.Recv, uint32(n))
+
 			if err = session.HandleRTP(buf[:n]); err != nil {
 				return err
 			}

pkg/srtp/session.go

@@ -17,6 +17,7 @@ type Session struct {
 
 	Write func(b []byte) (int, error)
 	Track *streamer.Track
+	Recv  uint32
 
 	lastSequence  uint32
 	lastTimestamp uint32

pkg/streamer/helpers.go

@@ -4,13 +4,16 @@ import (
 	"strings"
 )
 
-const (
-	JSONType       = "type"
-	JSONRemoteAddr = "remote_addr"
-	JSONUserAgent  = "user_agent"
-	JSONReceive    = "receive"
-	JSONSend       = "send"
-)
+type Info struct {
+	Type       string   `json:"type,omitempty"`
+	URL        string   `json:"url,omitempty"`
+	RemoteAddr string   `json:"remote_addr,omitempty"`
+	UserAgent  string   `json:"user_agent,omitempty"`
+	Medias     []*Media `json:"medias,omitempty"`
+	Tracks     []*Track `json:"tracks,omitempty"`
+	Recv       uint32   `json:"recv,omitempty"`
+	Send       uint32   `json:"send,omitempty"`
+}
 
 func Between(s, sub1, sub2 string) string {
 	i := strings.Index(s, sub1)

pkg/streamer/media.go

@@ -1,6 +1,7 @@
 package streamer
 
 import (
+	"encoding/json"
 	"fmt"
 	"github.com/pion/sdp/v3"
 	"strconv"
@@ -32,9 +33,12 @@ const (
 	CodecAAC  = "MPEG4-GENERIC"
 	CodecOpus = "OPUS" // payloadType: 111
 	CodecG722 = "G722"
-	CodecMPA  = "MPA" // payload: 14
+	CodecMP3  = "MPA" // payload: 14, aka MPEG-1 Layer III
 
 	CodecELD = "ELD" // AAC-ELD
+
+	CodecAll = "ALL"
+	CodecAny = "ANY"
 )
 
 const PayloadTypeRAW byte = 255
@@ -43,7 +47,7 @@ func GetKind(name string) string {
 	switch name {
 	case CodecH264, CodecH265, CodecVP8, CodecVP9, CodecAV1, CodecJPEG:
 		return KindVideo
-	case CodecPCMU, CodecPCMA, CodecAAC, CodecOpus, CodecG722, CodecMPA, CodecELD:
+	case CodecPCMU, CodecPCMA, CodecAAC, CodecOpus, CodecG722, CodecMP3, CodecELD:
 		return KindAudio
 	}
 	return ""
@@ -59,7 +63,6 @@ type Media struct {
 
 	MID     string `json:"mid,omitempty"`     // TODO: fixme?
 	Control string `json:"control,omitempty"` // TODO: fixme?
-	Title   string `json:"title,omitempty"`   // TODO: fixme?
 }
 
 func (m *Media) String() string {
@@ -70,6 +73,10 @@ func (m *Media) String() string {
 	return s
 }
 
+func (m *Media) MarshalJSON() ([]byte, error) {
+	return json.Marshal(m.String())
+}
+
 func (m *Media) Clone() *Media {
 	clone := *m
 	return &clone
@@ -107,10 +114,6 @@ func (m *Media) MatchMedia(media *Media) *Codec {
 	}
 
 	for _, localCodec := range m.Codecs {
-		if media.Codecs == nil {
-			return localCodec
-		}
-
 		for _, remoteCodec := range media.Codecs {
 			if localCodec.Match(remoteCodec) {
 				return localCodec
@@ -120,6 +123,10 @@ func (m *Media) MatchMedia(media *Media) *Codec {
 	return nil
 }
 
+func (m *Media) MatchAll() bool {
+	return len(m.Codecs) > 0 && m.Codecs[0].Name == CodecAll
+}
+
 // Codec take best from:
 // - deepch/vdk/av.CodecData
 // - pion/webrtc.RTPCodecCapability
@@ -149,19 +156,18 @@ func (c *Codec) Clone() *Codec {
 }
 
 func (c *Codec) Match(codec *Codec) bool {
+	switch codec.Name {
+	case CodecAll, CodecAny:
+		return true
+	}
+
 	return c.Name == codec.Name &&
 		(c.ClockRate == codec.ClockRate || codec.ClockRate == 0) &&
 		(c.Channels == codec.Channels || codec.Channels == 0)
 }
 
-func UnmarshalSDP(rawSDP []byte) ([]*Media, error) {
-	sd := &sdp.SessionDescription{}
-	if err := sd.Unmarshal(rawSDP); err != nil {
-		return nil, err
-	}
-
-	var medias []*Media
-	for _, md := range sd.MediaDescriptions {
+func UnmarshalMedias(descriptions []*sdp.MediaDescription) (medias []*Media) {
+	for _, md := range descriptions {
 		media := UnmarshalMedia(md)
 
 		if media.Direction == DirectionSendRecv {
@@ -175,11 +181,25 @@ func UnmarshalSDP(rawSDP []byte) ([]*Media, error) {
 		medias = append(medias, media)
 	}
 
-	return medias, nil
+	return
 }
 
-func MarshalSDP(medias []*Media) ([]byte, error) {
-	sd := &sdp.SessionDescription{}
+func MarshalSDP(name string, medias []*Media) ([]byte, error) {
+	sd := &sdp.SessionDescription{
+		Origin: sdp.Origin{
+			Username: "-", SessionID: 1, SessionVersion: 1,
+			NetworkType: "IN", AddressType: "IP4", UnicastAddress: "0.0.0.0",
+		},
+		SessionName: sdp.SessionName(name),
+		ConnectionInformation: &sdp.ConnectionInformation{
+			NetworkType: "IN", AddressType: "IP4", Address: &sdp.Address{
+				Address: "0.0.0.0",
+			},
+		},
+		TimeDescriptions: []sdp.TimeDescription{
+			{Timing: sdp.Timing{}},
+		},
+	}
 
 	payloadType := uint8(96)
 
@@ -267,7 +287,7 @@ func UnmarshalCodec(md *sdp.MediaDescription, payloadType string) *Codec {
 			c.Name = CodecPCMA
 			c.ClockRate = 8000
 		case "14":
-			c.Name = CodecMPA
+			c.Name = CodecMP3
 			c.ClockRate = 44100
 		case "26":
 			c.Name = CodecJPEG
@@ -280,6 +300,40 @@ func UnmarshalCodec(md *sdp.MediaDescription, payloadType string) *Codec {
 	return c
 }
 
+func ParseQuery(query map[string][]string) (medias []*Media) {
+	// set media candidates from query list
+	for key, values := range query {
+		switch key {
+		case KindVideo, KindAudio:
+			for _, value := range values {
+				media := &Media{Kind: key, Direction: DirectionRecvonly}
+
+				for _, name := range strings.Split(value, ",") {
+					name = strings.ToUpper(name)
+
+					// check aliases
+					switch name {
+					case "", "COPY":
+						name = CodecAny
+					case "MJPEG":
+						name = CodecJPEG
+					case "AAC":
+						name = CodecAAC
+					case "MP3":
+						name = CodecMP3
+					}
+
+					media.Codecs = append(media.Codecs, &Codec{Name: name})
+				}
+
+				medias = append(medias, media)
+			}
+		}
+	}
+
+	return
+}
+
 func atoi(s string) (i int) {
 	i, _ = strconv.Atoi(s)
 	return

pkg/streamer/media_test.go

@@ -0,0 +1,42 @@
+package streamer
+
+import (
+	"github.com/pion/sdp/v3"
+	"github.com/stretchr/testify/assert"
+	"net/url"
+	"testing"
+)
+
+func TestSDP(t *testing.T) {
+	medias := []*Media{{
+		Kind: KindAudio, Direction: DirectionSendonly,
+		Codecs: []*Codec{
+			{Name: CodecPCMU, ClockRate: 8000},
+		},
+	}}
+
+	data, err := MarshalSDP("go2rtc/1.0.0", medias)
+	assert.Empty(t, err)
+
+	sd := &sdp.SessionDescription{}
+	err = sd.Unmarshal(data)
+	assert.Empty(t, err)
+}
+
+func TestParseQuery(t *testing.T) {
+	u, _ := url.Parse("rtsp://localhost:8554/camera1")
+	medias := ParseQuery(u.Query())
+	assert.Nil(t, medias)
+
+	for _, rawULR := range []string{
+		"rtsp://localhost:8554/camera1?video",
+		"rtsp://localhost:8554/camera1?video=copy",
+		"rtsp://localhost:8554/camera1?video=any",
+	} {
+		u, _ = url.Parse(rawULR)
+		medias = ParseQuery(u.Query())
+		assert.Equal(t, []*Media{
+			{Kind: KindVideo, Direction: DirectionRecvonly, Codecs: []*Codec{{Name: CodecAny}}},
+		}, medias)
+	}
+}

pkg/streamer/track.go

@@ -1,6 +1,7 @@
 package streamer
 
 import (
+	"encoding/json"
 	"fmt"
 	"github.com/pion/rtp"
 	"sync"
@@ -22,12 +23,19 @@ func NewTrack(codec *Codec, direction string) *Track {
 
 func (t *Track) String() string {
 	s := t.Codec.String()
-	t.sinkMu.RLock()
-	s += fmt.Sprintf(", sinks=%d", len(t.sink))
-	t.sinkMu.RUnlock()
+	if t.sinkMu.TryRLock() {
+		s += fmt.Sprintf(", sinks=%d", len(t.sink))
+		t.sinkMu.RUnlock()
+	} else {
+		s += fmt.Sprintf(", sinks=?")
+	}
 	return s
 }
 
+func (t *Track) MarshalJSON() ([]byte, error) {
+	return json.Marshal(t.String())
+}
+
 func (t *Track) WriteRTP(p *rtp.Packet) error {
 	t.sinkMu.RLock()
 	for _, f := range t.sink {

pkg/tcp/auth.go

@@ -70,7 +70,9 @@ func (a *Auth) Write(req *Request) {
 	case AuthBasic:
 		req.Header.Set("Authorization", a.header)
 	case AuthDigest:
-		uri := req.URL.RequestURI()
+		// important to use String except RequestURL for RtspServer:
+		// https://github.com/AlexxIT/go2rtc/issues/244
+		uri := req.URL.String()
 		h2 := HexMD5(req.Method, uri)
 		response := HexMD5(a.h1nonce, h2)
 		header := a.header + fmt.Sprintf(

pkg/ts/ts.go

@@ -0,0 +1,194 @@
+package ts
+
+import (
+	"bytes"
+	"encoding/hex"
+	"github.com/AlexxIT/go2rtc/pkg/aac"
+	"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/ts"
+	"github.com/pion/rtp"
+	"sync/atomic"
+	"time"
+)
+
+type Consumer struct {
+	streamer.Element
+
+	UserAgent  string
+	RemoteAddr string
+
+	buf      *bytes.Buffer
+	muxer    *ts.Muxer
+	mimeType string
+	streams  []av.CodecData
+	start    bool
+	init     []byte
+
+	send uint32
+}
+
+func (c *Consumer) GetMedias() []*streamer.Media {
+	return []*streamer.Media{
+		{
+			Kind:      streamer.KindVideo,
+			Direction: streamer.DirectionRecvonly,
+			Codecs: []*streamer.Codec{
+				{Name: streamer.CodecH264},
+			},
+		},
+		//{
+		//	Kind:      streamer.KindAudio,
+		//	Direction: streamer.DirectionRecvonly,
+		//	Codecs: []*streamer.Codec{
+		//		{Name: streamer.CodecAAC},
+		//	},
+		//},
+	}
+}
+
+func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
+	codec := track.Codec
+	trackID := int8(len(c.streams))
+
+	switch codec.Name {
+	case streamer.CodecH264:
+		sps, pps := h264.GetParameterSet(codec.FmtpLine)
+		stream, err := h264parser.NewCodecDataFromSPSAndPPS(sps, pps)
+		if err != nil {
+			return nil
+		}
+
+		if len(c.mimeType) > 0 {
+			c.mimeType += ","
+		}
+
+		c.mimeType += "avc1." + h264.GetProfileLevelID(codec.FmtpLine)
+
+		c.streams = append(c.streams, stream)
+
+		pkt := av.Packet{Idx: trackID, CompositionTime: time.Millisecond}
+
+		ts2time := time.Second / time.Duration(codec.ClockRate)
+
+		push := func(packet *rtp.Packet) error {
+			if packet.Version != h264.RTPPacketVersionAVC {
+				return nil
+			}
+
+			if !c.start {
+				return nil
+			}
+
+			pkt.Data = packet.Payload
+			newTime := time.Duration(packet.Timestamp) * ts2time
+			if pkt.Time > 0 {
+				pkt.Duration = newTime - pkt.Time
+			}
+			pkt.Time = newTime
+
+			if err = c.muxer.WritePacket(pkt); err != nil {
+				return err
+			}
+
+			// clone bytes from buffer, so next packet won't overwrite it
+			buf := append([]byte{}, c.buf.Bytes()...)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(buf)
+
+			c.buf.Reset()
+
+			return nil
+		}
+
+		if codec.IsRTP() {
+			wrapper := h264.RTPDepay(track)
+			push = wrapper(push)
+		}
+
+		return track.Bind(push)
+
+	case streamer.CodecAAC:
+		s := streamer.Between(codec.FmtpLine, "config=", ";")
+
+		b, err := hex.DecodeString(s)
+		if err != nil {
+			return nil
+		}
+
+		stream, err := aacparser.NewCodecDataFromMPEG4AudioConfigBytes(b)
+		if err != nil {
+			return nil
+		}
+
+		if len(c.mimeType) > 0 {
+			c.mimeType += ","
+		}
+
+		c.mimeType += "mp4a.40.2"
+		c.streams = append(c.streams, stream)
+
+		pkt := av.Packet{Idx: trackID, CompositionTime: time.Millisecond}
+
+		ts2time := time.Second / time.Duration(codec.ClockRate)
+
+		push := func(packet *rtp.Packet) error {
+			if !c.start {
+				return nil
+			}
+
+			pkt.Data = packet.Payload
+			newTime := time.Duration(packet.Timestamp) * ts2time
+			if pkt.Time > 0 {
+				pkt.Duration = newTime - pkt.Time
+			}
+			pkt.Time = newTime
+
+			if err := c.muxer.WritePacket(pkt); err != nil {
+				return err
+			}
+
+			// clone bytes from buffer, so next packet won't overwrite it
+			buf := append([]byte{}, c.buf.Bytes()...)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
+			c.Fire(buf)
+
+			c.buf.Reset()
+
+			return nil
+		}
+
+		if codec.IsRTP() {
+			wrapper := aac.RTPDepay(track)
+			push = wrapper(push)
+		}
+
+		return track.Bind(push)
+	}
+
+	panic("unsupported codec")
+}
+
+func (c *Consumer) MimeCodecs() string {
+	return c.mimeType
+}
+
+func (c *Consumer) Init() ([]byte, error) {
+	c.buf = bytes.NewBuffer(nil)
+	c.muxer = ts.NewMuxer(c.buf)
+
+	// first packet will be with header, it's ok
+	if err := c.muxer.WriteHeader(c.streams); err != nil {
+		return nil, err
+	}
+	data := append([]byte{}, c.buf.Bytes()...)
+
+	return data, nil
+}
+
+func (c *Consumer) Start() {
+	c.start = true
+}

pkg/webrtc/api.go

@@ -35,13 +35,17 @@ func NewAPI(address string) (*webrtc.API, error) {
 	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,
-			})
+		s.SetNetworkTypes([]webrtc.NetworkType{
+			webrtc.NetworkTypeUDP4, webrtc.NetworkTypeUDP6,
+			webrtc.NetworkTypeTCP4, webrtc.NetworkTypeTCP6,
+		})
 
+		if ln, err := net.ListenPacket("udp", address); err == nil {
+			udpMux := webrtc.NewICEUDPMux(nil, ln)
+			s.SetICEUDPMux(udpMux)
+		}
+
+		if ln, err := net.Listen("tcp", address); err == nil {
 			tcpMux := webrtc.NewICETCPMux(nil, ln, 8)
 			s.SetICETCPMux(tcpMux)
 		}

pkg/webrtc/conn.go

@@ -2,6 +2,7 @@ package webrtc
 
 import (
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/sdp/v3"
 	"github.com/pion/webrtc/v3"
 )
 
@@ -90,12 +91,15 @@ func (c *Conn) SetOffer(offer string) (err error) {
 	if err = c.Conn.SetRemoteDescription(sdOffer); err != nil {
 		return
 	}
+
 	rawSDP := []byte(c.Conn.RemoteDescription().SDP)
-	medias, err := streamer.UnmarshalSDP(rawSDP)
-	if err != nil {
+	sd := &sdp.SessionDescription{}
+	if err = sd.Unmarshal(rawSDP); err != nil {
 		return
 	}
 
+	medias := streamer.UnmarshalMedias(sd.MediaDescriptions)
+
 	// sort medias, so video will always be before audio
 	// and ignore application media from Hass default lovelace card
 	for _, media := range medias {

pkg/webrtc/consumer.go

@@ -113,20 +113,12 @@ func (c *Conn) AddCandidate(candidate string) {
 }
 
 func (c *Conn) MarshalJSON() ([]byte, error) {
-	v := map[string]interface{}{
-		streamer.JSONType:       "WebRTC server consumer",
-		streamer.JSONRemoteAddr: c.remote(),
+	info := &streamer.Info{
+		Type:       "WebRTC client",
+		RemoteAddr: c.remote(),
+		UserAgent:  c.UserAgent,
+		Recv:       uint32(c.receive),
+		Send:       uint32(c.send),
 	}
-
-	if c.receive > 0 {
-		v[streamer.JSONReceive] = c.receive
-	}
-	if c.send > 0 {
-		v[streamer.JSONSend] = c.send
-	}
-	if c.UserAgent != "" {
-		v[streamer.JSONUserAgent] = c.UserAgent
-	}
-
-	return json.Marshal(v)
+	return json.Marshal(info)
 }

pkg/webrtc/helper.go

@@ -1,6 +1,7 @@
 package webrtc
 
 import (
+	"errors"
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/ice/v2"
@@ -12,24 +13,30 @@ import (
 	"time"
 )
 
-func NewCandidate(address string) (string, error) {
-	host, port, err := net.SplitHostPort(address)
-	if err != nil {
-		return "", err
+func NewCandidate(network, address string) (string, error) {
+	i := strings.LastIndexByte(address, ':')
+	if i < 0 {
+		return "", errors.New("wrong candidate: " + address)
 	}
+	host, port := address[:i], address[i+1:]
 
 	i, err := strconv.Atoi(port)
 	if err != nil {
 		return "", err
 	}
 
-	cand, err := ice.NewCandidateHost(&ice.CandidateHostConfig{
-		Network:   "tcp",
+	config := &ice.CandidateHostConfig{
+		Network:   network,
 		Address:   host,
 		Port:      i,
 		Component: ice.ComponentRTP,
-		TCPType:   ice.TCPTypePassive,
-	})
+	}
+
+	if network == "tcp" {
+		config.TCPType = ice.TCPTypePassive
+	}
+
+	cand, err := ice.NewCandidateHost(config)
 	if err != nil {
 		return "", err
 	}

scripts/build.cmd

@@ -3,17 +3,17 @@
 @SET GOOS=windows
 @SET GOARCH=amd64
 @SET FILENAME=go2rtc_win64.zip
-go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc.exe
 
 @SET GOOS=windows
 @SET GOARCH=386
 @SET FILENAME=go2rtc_win32.zip
-go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc.exe
 
 @SET GOOS=windows
 @SET GOARCH=arm64
 @SET FILENAME=go2rtc_win_arm64.zip
-go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc.exe
 
 @SET GOOS=linux
 @SET GOARCH=amd64
@@ -44,9 +44,9 @@ go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx %FILENAME%
 @SET GOOS=darwin
 @SET GOARCH=amd64
 @SET FILENAME=go2rtc_mac_amd64.zip
-go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc
 
 @SET GOOS=darwin
 @SET GOARCH=arm64
 @SET FILENAME=go2rtc_mac_arm64.zip
-go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc

www/README.md

@@ -48,6 +48,18 @@ pc.ontrack = ev => {
 }
 ```
 
+## Chromecast 1
+
+2023-02-02. Error:
+
+```
+InvalidStateError: Failed to execute 'addTransceiver' on 'RTCPeerConnection': This operation is only supported in 'unified-plan'. 'unified-plan' will become the default behavior in the future, but it is currently experimental. To try it out, construct the RTCPeerConnection with sdpSemantics:'unified-plan' present in the RTCConfiguration argument.
+```
+
+User-Agent: `Mozilla/5.0 (X11; Linux armv7l) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/70.0.3538.47 Safari/537.36 CrKey/1.36.159268`
+
+https://webrtc.org/getting-started/unified-plan-transition-guide?hl=en
+
 ## Useful links
 
 - https://www.webrtc-experiment.com/DetectRTC/
@@ -56,3 +68,6 @@ pc.ontrack = ev => {
 - https://web.dev/i18n/en/fast-playback-with-preload/#manual_buffering
 - https://developer.mozilla.org/en-US/docs/Web/API/Media_Source_Extensions_API
 - https://chromium.googlesource.com/external/w3c/web-platform-tests/+/refs/heads/master/media-source/mediasource-is-type-supported.html
+- https://googlechrome.github.io/samples/media/sourcebuffer-changetype.html
+- https://chromestatus.com/feature/5100845653819392
+- https://developer.apple.com/documentation/webkit/delivering_video_content_for_safari

www/codecs.html

@@ -53,7 +53,7 @@
     const video = document.createElement("video");
     out.innerText += "video.canPlayType\n";
     types.forEach(type => {
-        out.innerText += type + "=" + (video.canPlayType(type) ? "true" : "false") + "\n";
+        out.innerText += `${type} = ${MediaSource.isTypeSupported(type)} / ${video.canPlayType(type)}\n`;
     })
 
 </script>

www/devices.html

@@ -58,7 +58,7 @@
         0, location.pathname.lastIndexOf("/")
     );
 
-    fetch(`${baseUrl}/api/devices`)
+    fetch(`${baseUrl}/api/devices`, {cache: 'no-cache'})
         .then(r => r.json())
         .then(data => {
             document.querySelector("body > table > tbody").innerHTML =

www/editor.html

@@ -0,0 +1,69 @@
+<!DOCTYPE html>
+<html>
+<head>
+    <title>File Editor</title>
+    <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">
+    <script src="https://cdnjs.cloudflare.com/ajax/libs/ace/1.14.0/ace.min.js"></script>
+    <style>
+        body {
+            font-family: Arial, Helvetica, sans-serif;
+        }
+
+        body {
+            margin: 0;
+            padding: 0;
+            display: flex;
+            flex-direction: column;
+        }
+
+        html, body, #config {
+            width: 100%;
+            height: 100%;
+        }
+    </style>
+</head>
+<body>
+<script src="main.js"></script>
+<div>
+    <button id="save">Save & Restart</button>
+</div>
+<br>
+<div id="config"></div>
+<script>
+    ace.config.set('basePath', 'https://cdnjs.cloudflare.com/ajax/libs/ace/1.14.0/');
+    const editor = ace.edit("config", {
+        mode: "ace/mode/yaml",
+    });
+
+    document.getElementById('save').addEventListener('click', () => {
+        fetch('api/config', {
+            method: 'POST', body: editor.getValue()
+        }).then(r => {
+            if (r.ok) {
+                alert('OK');
+                fetch('api/exit', {method: 'POST'});
+            } else {
+                r.text().then(alert);
+            }
+        });
+    });
+
+    window.addEventListener('load', () => {
+        fetch('api/config', {cache: 'no-cache'}).then(r => {
+            if (r.status === 410) {
+                alert('Config file is not set');
+            } else if (r.status === 404) {
+                editor.setValue(''); // config file not exist
+            } else if (r.ok) {
+                r.text().then(data => {
+                    editor.setValue(data);
+                });
+            } else {
+                alert(`Unknown error: ${r.statusText} (${r.status})`);
+            }
+        });
+    })
+</script>
+</body>
+</html>

www/homekit.html

@@ -65,7 +65,7 @@
         0, location.pathname.lastIndexOf("/")
     );
 
-    fetch(`${baseUrl}/api/homekit`)
+    fetch(`${baseUrl}/api/homekit`, {cache: 'no-cache'})
         .then(r => r.json())
         .then(data => {
             document.querySelector("body > table > tbody").innerHTML =

www/index.html

@@ -10,6 +10,7 @@
     <style>
         body {
             font-family: Arial, Helvetica, sans-serif;
+            background-color: white;
         }
 
         table {
@@ -61,6 +62,7 @@
 </head>
 <body>
 <script src="main.js"></script>
+<div class="info"></div>
 <div class="header">
     <input id="src" type="text" placeholder="url">
     <a id="add" href="#">add</a>
@@ -87,10 +89,7 @@
     const templates = [
         '<a href="stream.html?src={name}">stream</a>',
         '<a href="webrtc.html?src={name}">2-way-aud</a>',
-        '<a href="api/stream.mp4?src={name}">mp4</a>',
-        '<a href="api/stream.mjpeg?src={name}">mjpeg</a>',
-        `<a href="rtsp://${location.hostname}:8554/{name}">rtsp</a>`,
-        '<a href="api/streams?src={name}">info</a>',
+        '<a href="links.html?src={name}">links</a>',
         '<a href="#" data-name="{name}">delete</a>',
     ];
 
@@ -132,25 +131,33 @@
 
     function reload() {
         const url = new URL("api/streams", location.href);
-        fetch(url).then(r => r.json()).then(data => {
+        fetch(url, {cache: 'no-cache'}).then(r => r.json()).then(data => {
             tbody.innerHTML = "";
 
             for (const [name, value] of Object.entries(data)) {
-                const online = value ? value.length : 0;
+                const online = value && value.consumers ? value.consumers.length : 0;
+                const src = encodeURIComponent(name);
                 const links = templates.map(link => {
-                    return link.replace("{name}", encodeURIComponent(name));
+                    return link.replace("{name}", src);
                 }).join(" ");
 
                 const tr = document.createElement("tr");
                 tr.dataset["id"] = name;
                 tr.innerHTML =
                     `<td><label><input type="checkbox" name="${name}">${name}</label></td>` +
-                    `<td>${online}</td><td>${links}</td>`;
+                    `<td><a href="api/streams?src=${src}">${online} / info</a></td>` +
+                    `<td>${links}</td>`;
                 tbody.appendChild(tr);
             }
         });
     }
 
+    const url = new URL("api", location.href);
+    fetch(url, {cache: 'no-cache'}).then(r => r.json()).then(data => {
+        const info = document.querySelector(".info");
+        info.innerText = `Version: ${data.version}, Config: ${data.config_path}`;
+    });
+
     reload();
 </script>
 </body>

www/links.html

@@ -0,0 +1,89 @@
+<!DOCTYPE html>
+<html>
+<head>
+    <title>go2rtc - links</title>
+    <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">
+    <style>
+        body {
+            font-family: Arial, Helvetica, sans-serif;
+        }
+
+        body {
+            margin: 0;
+            padding: 0;
+            display: flex;
+            flex-direction: column;
+        }
+
+        html, body, #config {
+            width: 100%;
+            height: 100%;
+        }
+
+        div {
+            padding: 10px;
+        }
+
+        div > li {
+            list-style-type: none;
+            padding-left: 10px;
+            position: relative;
+        }
+
+        div > li:before {
+            content: "-";
+            position: absolute;
+            left: 0;
+        }
+    </style>
+</head>
+<body>
+<script src="main.js"></script>
+<div id="links"></div>
+<script>
+    const params = new URLSearchParams(location.search);
+    const src = params.get("src");
+
+    const links = document.querySelector("#links");
+    links.innerHTML = `
+        <h2>Any codec in source</h2>
+        <li><a href="stream.html?src=${src}">stream.html</a> with auto-select mode / browsers: all / codecs: H264, H265*, MJPEG, JPEG, AAC, PCMU, PCMA, OPUS</li>
+        <li><a href="api/streams?src=${src}">info.json</a> page with active connections</li>
+    `;
+
+    const url = new URL('api', location.href);
+    fetch(url, {cache: 'no-cache'}).then(r => r.json()).then(data => {
+        let rtsp = location.host + ':8554';
+        try {
+            const host = data.host.match(/^[^:]+/)[0];
+            const port = data.rtsp.listen.match(/[0-9]+$/)[0];
+            rtsp = `${host}:${port}`;
+        } catch (e) {
+        }
+
+        const links = document.querySelector("#links");
+        links.innerHTML += `
+            <li><a href="rtsp://${rtsp}/${src}">rtsp</a> with only one video and one audio / codecs: any</li>
+            <li><a href="rtsp://${rtsp}/${src}?mp4">rtsp</a> for MP4 recording (Hass or Frigate) / codecs: H264, H265, AAC</li>
+            <li><a href="rtsp://${rtsp}/${src}?video=all&audio=all">rtsp</a> with all tracks / codecs: any</li>
+
+            <h2>H264/H265 source</h2>
+            <li><a href="stream.html?src=${src}&mode=webrtc">stream.html</a> WebRTC stream / browsers: all / codecs: H264, PCMU, PCMA, OPUS / +H265 in Safari</li>
+            <li><a href="stream.html?src=${src}&mode=mse">stream.html</a> MSE stream / browsers: Chrome, Firefox, Safari Mac/iPad / codecs: H264, H265*, AAC / +OPUS in Chrome and Firefox</li>
+            <li><a href="api/stream.mp4?src=${src}">stream.mp4</a> MP4 stream with AAC audio / browsers: Chrome, Firefox / codecs: H264, H265*, AAC</li>
+            <li><a href="api/stream.mp4?src=${src}&video=h264,h265&audio=aac,opus,mp3,pcma,pcmu">stream.mp4</a> MP4 stream with any audio / browsers: Chrome / codecs: H264, H265*, AAC, OPUS, MP3, PCMU, PCMA</li>
+            <li><a href="api/frame.mp4?src=${src}">frame.mp4</a> snapshot in MP4-format / browsers: all / codecs: H264, H265*</li>
+            <li><a href="api/stream.m3u8?src=${src}">stream.m3u8</a> HLS/TS / browsers: Safari all, Chrome Android / codecs: H264</li>
+            <li><a href="api/stream.m3u8?src=${src}&mp4">stream.m3u8</a> HLS/fMP4 / browsers: Safari all, Chrome Android / codecs: H264, H265*, AAC</li>
+            <li><a href="webrtc.html?src=${src}">webrtc.html</a> with two-way audio for supported cameras / browsers: all / codecs: H264, PCMU, PCMA, OPUS</li>
+
+            <h2>MJPEG source</h2>
+            <li><a href="stream.html?src=${src}&mode=mjpeg">stream.html</a> with MJPEG mode / browsers: all / codecs: MJPEG, JPEG</li>
+            <li><a href="api/stream.mjpeg?src=${src}">stream.mjpeg</a> MJPEG stream / browsers: all / codecs: MJPEG, JPEG</li>
+            <li><a href="api/frame.jpeg?src=${src}">frame.jpeg</a> snapshot in JPEG-format / browsers: all / codecs: MJPEG, JPEG</li>
+        `;
+    });
+</script>
+</body>
+</html>

www/main.js

@@ -47,6 +47,7 @@ nav li {
         <li><a href="index.html">Streams</a></li>
         <li><a href="devices.html">Devices</a></li>
         <li><a href="homekit.html">HomeKit</a></li>
+        <li><a href="editor.html">Config</a></li>
     </ul>
 </nav>
 ` + document.body.innerHTML;

www/stream.html

@@ -3,13 +3,13 @@
 <head>
     <meta name="viewport" content="width=device-width, initial-scale=1">
     <title>go2rtc - Stream</title>
-    <script src="video-rtc.js"></script>
     <style>
         body {
             background: black;
             margin: 0;
             padding: 0;
             display: flex;
+            font-family: Arial, Helvetica, sans-serif;
         }
 
         html, body {
@@ -25,7 +25,8 @@
     </style>
 </head>
 <body>
-<script>
+<script type="module" src="./video-stream.js"></script>
+<script type="module">
     const params = new URLSearchParams(location.search);
 
     // support multiple streams and multiple modes
@@ -44,16 +45,16 @@
         document.body.className = "flex";
     }
 
-    const background = params.get("background") === "true";
+    const background = params.get("background") !== "false";
     const width = "1 0 " + (params.get("width") || "320px");
 
     for (let i = 0; i < streams.length; i++) {
-        /** @type {VideoRTC} */
-        const video = document.createElement("video-rtc");
+        /** @type {VideoStream} */
+        const video = document.createElement("video-stream");
         video.background = background;
         video.mode = modes[i] || video.mode;
         video.style.flex = width;
-        video.src = new URL("api/ws?src=" + streams[i], location.href);
+        video.src = new URL("api/ws?src=" + encodeURIComponent(streams[i]), location.href);
         document.body.appendChild(video);
     }
 </script>

www/video-rtc.js

@@ -1,5 +1,5 @@
 /**
- * Video player for MSE and WebRTC connections.
+ * Video player for go2rtc streaming application.
  *
  * All modern web technologies are supported in almost any browser except Apple Safari.
  *
@@ -11,8 +11,9 @@
  * - MediaSource for Safari iOS all
  * - Customized built-in elements (extends HTMLVideoElement) because all Safari
  * - Public class fields because old Safari (before 14.0)
+ * - Autoplay for Safari
  */
-class VideoRTC extends HTMLElement {
+export class VideoRTC extends HTMLElement {
     constructor() {
         super();
 
@@ -26,37 +27,57 @@ class VideoRTC extends HTMLElement {
             "hvc1.1.6.L153.B0", // H.265 main 5.1 (Chromecast Ultra)
             "mp4a.40.2",        // AAC LC
             "mp4a.40.5",        // AAC HE
-            "mp4a.69",          // MP3
-            "mp4a.6B",          // MP3
+            "opus",             // OPUS Chrome
         ];
 
         /**
-         * Supported modes (webrtc, mse, mp4, mjpeg).
+         * [config] Supported modes (webrtc, mse, mp4, mjpeg).
          * @type {string}
          */
         this.mode = "webrtc,mse,mp4,mjpeg";
 
         /**
-         * Run stream when not displayed on the screen. Default `false`.
+         * [config] Run stream when not displayed on the screen. Default `false`.
          * @type {boolean}
          */
         this.background = false;
 
         /**
-         * Run stream only when player in the viewport. Stop when user scroll out player.
+         * [config] Run stream only when player in the viewport. Stop when user scroll out player.
          * Value is percentage of visibility from `0` (not visible) to `1` (full visible).
          * Default `0` - disable;
          * @type {number}
          */
-        this.intersectionThreshold = 0;
+        this.visibilityThreshold = 0;
 
         /**
-         * Run stream only when browser page on the screen. Stop when user change browser
+         * [config] Run stream only when browser page on the screen. Stop when user change browser
          * tab or minimise browser windows.
          * @type {boolean}
          */
         this.visibilityCheck = true;
 
+        /**
+         * [config] WebRTC configuration
+         * @type {RTCConfiguration}
+         */
+        this.pcConfig = {
+            iceServers: [{urls: 'stun:stun.l.google.com:19302'}],
+            sdpSemantics: 'unified-plan',  // important for Chromecast 1
+        };
+
+        /**
+         * [info] WebSocket connection state. Values: CONNECTING, OPEN, CLOSED
+         * @type {number}
+         */
+        this.wsState = WebSocket.CLOSED;
+
+        /**
+         * [info] WebRTC connection state.
+         * @type {number}
+         */
+        this.pcState = WebSocket.CLOSED;
+
         /**
          * @type {HTMLVideoElement}
          */
@@ -68,16 +89,9 @@ class VideoRTC extends HTMLElement {
         this.ws = null;
 
         /**
-         * Internal WebSocket connection state. Values: CONNECTING, OPEN, CLOSED
-         * @type {number}
-         */
-        this.wsState = WebSocket.CLOSED;
-
-        /**
-         * Internal WebSocket URL.
          * @type {string|URL}
          */
-        this.url = "";
+        this.wsURL = "";
 
         /**
          * @type {RTCPeerConnection}
@@ -87,37 +101,38 @@ class VideoRTC extends HTMLElement {
         /**
          * @type {number}
          */
-        this.pcState = WebSocket.CLOSED;
-
-        this.pcConfig = {iceServers: [{urls: "stun:stun.l.google.com:19302"}]};
+        this.connectTS = 0;
 
         /**
-         * Internal disconnect TimeoutID.
+         * @type {string}
+         */
+        this.mseCodecs = "";
+
+        /**
+         * [internal] Disconnect TimeoutID.
          * @type {number}
          */
-        this.disconnectTimeout = 0;
+        this.disconnectTID = 0;
 
         /**
-         * Internal reconnect TimeoutID.
+         * [internal] Reconnect TimeoutID.
          * @type {number}
          */
-        this.reconnectTimeout = 0;
+        this.reconnectTID = 0;
 
         /**
-         * Handler for receiving Binary from WebSocket
+         * [internal] Handler for receiving Binary from WebSocket.
          * @type {Function}
          */
         this.ondata = null;
 
         /**
-         * Handlers list for receiving JSON from WebSocket
+         * [internal] Handlers list for receiving JSON from WebSocket
          * @type {Object.<string,Function>}}
          */
         this.onmessage = null;
     }
 
-    /** public properties **/
-
     /**
      * Set video source (WebSocket URL). Support relative path.
      * @param {string|URL} value
@@ -130,9 +145,9 @@ class VideoRTC extends HTMLElement {
             value = "ws" + location.origin.substring(4) + value;
         }
 
-        this.url = value;
+        this.wsURL = value;
 
-        if (this.isConnected) this.connectedCallback();
+        this.onconnect();
     }
 
     /**
@@ -156,10 +171,6 @@ class VideoRTC extends HTMLElement {
         if (this.ws) this.ws.send(JSON.stringify(value));
     }
 
-    get closed() {
-        return this.wsState === WebSocket.CLOSED && this.pcState === WebSocket.CLOSED;
-    }
-
     codecs(type) {
         const test = type === "mse"
             ? codec => MediaSource.isTypeSupported(`video/mp4; codecs="${codec}"`)
@@ -172,11 +183,9 @@ class VideoRTC extends HTMLElement {
      * document-connected element.
      */
     connectedCallback() {
-        console.debug("VideoRTC.connectedCallback", this.wsState, this.pcState);
-
-        if (this.disconnectTimeout) {
-            clearTimeout(this.disconnectTimeout);
-            this.disconnectTimeout = 0;
+        if (this.disconnectTID) {
+            clearTimeout(this.disconnectTID);
+            this.disconnectTID = 0;
         }
 
         // because video autopause on disconnected from DOM
@@ -184,17 +193,13 @@ class VideoRTC extends HTMLElement {
             const seek = this.video.seekable;
             if (seek.length > 0) {
                 this.video.currentTime = seek.end(seek.length - 1);
-                this.play();
             }
+            this.play();
+        } else {
+            this.oninit();
         }
 
-        if (!this.url || !this.closed) return;
-
-        // CLOSED => CONNECTING
-        this.wsState = WebSocket.CONNECTING;
-
-        this.internalInit();
-        this.internalWS();
+        this.onconnect();
     }
 
     /**
@@ -202,50 +207,36 @@ class VideoRTC extends HTMLElement {
      * document's DOM.
      */
     disconnectedCallback() {
-        console.debug("VideoRTC.disconnectedCallback", this.wsState, this.pcState);
+        if (this.background || this.disconnectTID) return;
+        if (this.wsState === WebSocket.CLOSED && this.pcState === WebSocket.CLOSED) return;
 
-        if (this.background || this.disconnectTimeout || this.closed) return;
-
-        this.disconnectTimeout = setTimeout(() => {
-            if (this.reconnectTimeout) {
-                clearTimeout(this.reconnectTimeout);
-                this.reconnectTimeout = 0;
+        this.disconnectTID = setTimeout(() => {
+            if (this.reconnectTID) {
+                clearTimeout(this.reconnectTID);
+                this.reconnectTID = 0;
             }
 
-            this.disconnectTimeout = 0;
+            this.disconnectTID = 0;
 
-            this.wsState = WebSocket.CLOSED;
-            if (this.ws) {
-                this.ws.close();
-                this.ws = null;
-            }
-
-            this.pcState = WebSocket.CLOSED;
-            if (this.pc) {
-                this.pc.close();
-                this.pc = null;
-            }
+            this.ondisconnect();
         }, this.DISCONNECT_TIMEOUT);
     }
 
-    internalInit() {
-        if (this.childElementCount) return;
-
+    /**
+     * Creates child DOM elements. Called automatically once on `connectedCallback`.
+     */
+    oninit() {
         this.video = document.createElement("video");
         this.video.controls = true;
         this.video.playsInline = true;
         this.video.preload = "auto";
 
-        this.appendChild(this.video);
-
-        // important for second video for mode MP4
-        this.style.display = "block";
-        this.style.position = "relative";
-
         this.video.style.display = "block"; // fix bottom margin 4px
         this.video.style.width = "100%";
         this.video.style.height = "100%"
 
+        this.appendChild(this.video);
+
         if (this.background) return;
 
         if ("hidden" in document && this.visibilityCheck) {
@@ -258,7 +249,7 @@ class VideoRTC extends HTMLElement {
             })
         }
 
-        if ("IntersectionObserver" in window && this.intersectionThreshold) {
+        if ("IntersectionObserver" in window && this.visibilityThreshold) {
             const observer = new IntersectionObserver(entries => {
                 entries.forEach(entry => {
                     if (!entry.isIntersecting) {
@@ -267,92 +258,120 @@ class VideoRTC extends HTMLElement {
                         this.connectedCallback();
                     }
                 });
-            }, {threshold: this.intersectionThreshold});
+            }, {threshold: this.visibilityThreshold});
             observer.observe(this);
         }
     }
 
-    internalWS() {
-        if (this.wsState !== WebSocket.CONNECTING) return;
-        if (this.ws) throw "connect with non null WebSocket";
+    /**
+     * Connect to WebSocket. Called automatically on `connectedCallback`.
+     * @return {boolean} true if the connection has started.
+     */
+    onconnect() {
+        if (!this.isConnected || !this.wsURL || this.ws || this.pc) return false;
 
-        const ts = Date.now();
+        // CLOSED or CONNECTING => CONNECTING
+        this.wsState = WebSocket.CONNECTING;
 
-        this.ws = new WebSocket(this.url);
+        this.connectTS = Date.now();
+
+        this.ws = new WebSocket(this.wsURL);
         this.ws.binaryType = "arraybuffer";
+        this.ws.addEventListener("open", ev => this.onopen(ev));
+        this.ws.addEventListener("close", ev => this.onclose(ev));
 
-        this.ws.addEventListener("open", () => {
-            console.debug("VideoRTC.ws.open", this.wsState);
-
-            // CONNECTING => OPEN
-            this.wsState = WebSocket.OPEN;
-
-            this.ws.addEventListener("message", ev => {
-                if (typeof ev.data === "string") {
-                    const msg = JSON.parse(ev.data);
-                    for (const mode in this.onmessage) {
-                        this.onmessage[mode](msg);
-                    }
-                } else {
-                    this.ondata(ev.data);
-                }
-            });
-
-            this.ondata = null;
-            this.onmessage = {};
-
-            let firstMode = "";
-
-            if (this.mode.indexOf("mse") >= 0 && "MediaSource" in window) { // iPhone
-                firstMode ||= "mse";
-                this.internalMSE();
-            } else if (this.mode.indexOf("mp4") >= 0) {
-                firstMode ||= "mp4";
-                this.internalMP4();
-            }
-
-            if (this.mode.indexOf("webrtc") >= 0 && "RTCPeerConnection" in window) { // macOS Desktop app
-                firstMode ||= "webrtc";
-                this.internalRTC();
-            }
-
-            if (this.mode.indexOf("mjpeg") >= 0) {
-                if (firstMode) {
-                    this.onmessage["mjpeg"] = msg => {
-                        if (msg.type !== "error" || msg.value.indexOf(firstMode) !== 0) return;
-                        this.internalMJPEG();
-                    }
-                } else {
-                    this.internalMJPEG();
-                }
-            }
-        });
-
-        this.ws.addEventListener("close", () => {
-            console.debug("VideoRTC.ws.close", this.wsState);
-
-            if (this.wsState === WebSocket.CLOSED) return;
-
-            // CONNECTING, OPEN => CONNECTING
-            this.wsState = WebSocket.CONNECTING;
-            this.ws = null;
-
-            // reconnect no more than once every X seconds
-            const delay = Math.max(this.RECONNECT_TIMEOUT - (Date.now() - ts), 0);
-
-            this.reconnectTimeout = setTimeout(() => {
-                this.reconnectTimeout = 0;
-                this.internalWS();
-            }, delay);
-        });
+        return true;
     }
 
-    internalMSE() {
-        console.debug("VideoRTC.internalMSE");
+    ondisconnect() {
+        this.wsState = WebSocket.CLOSED;
+        if (this.ws) {
+            this.ws.close();
+            this.ws = null;
+        }
 
+        this.pcState = WebSocket.CLOSED;
+        if (this.pc) {
+            this.pc.close();
+            this.pc = null;
+        }
+    }
+
+    /**
+     * @returns {Array.<string>} of modes (mse, webrtc, etc.)
+     */
+    onopen() {
+        // CONNECTING => OPEN
+        this.wsState = WebSocket.OPEN;
+
+        this.ws.addEventListener("message", ev => {
+            if (typeof ev.data === "string") {
+                const msg = JSON.parse(ev.data);
+                for (const mode in this.onmessage) {
+                    this.onmessage[mode](msg);
+                }
+            } else {
+                this.ondata(ev.data);
+            }
+        });
+
+        this.ondata = null;
+        this.onmessage = {};
+
+        const modes = [];
+
+        if (this.mode.indexOf("mse") >= 0 && "MediaSource" in window) { // iPhone
+            modes.push("mse");
+            this.onmse();
+        } else if (this.mode.indexOf("mp4") >= 0) {
+            modes.push("mp4");
+            this.onmp4();
+        }
+
+        if (this.mode.indexOf("webrtc") >= 0 && "RTCPeerConnection" in window) { // macOS Desktop app
+            modes.push("webrtc");
+            this.onwebrtc();
+        }
+
+        if (this.mode.indexOf("mjpeg") >= 0) {
+            if (modes.length) {
+                this.onmessage["mjpeg"] = msg => {
+                    if (msg.type !== "error" || msg.value.indexOf(modes[0]) !== 0) return;
+                    this.onmjpeg();
+                }
+            } else {
+                modes.push("mjpeg");
+                this.onmjpeg();
+            }
+        }
+
+        return modes;
+    }
+
+    /**
+     * @return {boolean} true if reconnection has started.
+     */
+    onclose() {
+        if (this.wsState === WebSocket.CLOSED) return false;
+
+        // CONNECTING, OPEN => CONNECTING
+        this.wsState = WebSocket.CONNECTING;
+        this.ws = null;
+
+        // reconnect no more than once every X seconds
+        const delay = Math.max(this.RECONNECT_TIMEOUT - (Date.now() - this.connectTS), 0);
+
+        this.reconnectTID = setTimeout(() => {
+            this.reconnectTID = 0;
+            this.onconnect();
+        }, delay);
+
+        return true;
+    }
+
+    onmse() {
         const ms = new MediaSource();
         ms.addEventListener("sourceopen", () => {
-            console.debug("VideoRTC.ms.sourceopen");
             URL.revokeObjectURL(this.video.src);
             this.send({type: "mse", value: this.codecs("mse")});
         }, {once: true});
@@ -361,28 +380,34 @@ class VideoRTC extends HTMLElement {
         this.video.srcObject = null;
         this.play();
 
+        this.mseCodecs = "";
+
         this.onmessage["mse"] = msg => {
             if (msg.type !== "mse") return;
 
+            this.mseCodecs = msg.value;
+
             const sb = ms.addSourceBuffer(msg.value);
             sb.mode = "segments"; // segments or sequence
             sb.addEventListener("updateend", () => {
                 if (sb.updating) return;
-                if (bufLen > 0) {
-                    try {
-                        sb.appendBuffer(buf.slice(0, bufLen));
-                    } catch (e) {
-                        console.debug(e);
+
+                try {
+                    if (bufLen > 0) {
+                        const data = buf.slice(0, bufLen);
+                        bufLen = 0;
+                        sb.appendBuffer(data);
+                    } else if (sb.buffered && sb.buffered.length) {
+                        const end = sb.buffered.end(sb.buffered.length - 1) - 15;
+                        const start = sb.buffered.start(0);
+                        if (end > start) {
+                            sb.remove(start, end);
+                            ms.setLiveSeekableRange(end, end + 15);
+                        }
+                        // console.debug("VideoRTC.buffered", start, end);
                     }
-                    bufLen = 0;
-                } else if (sb.buffered.length) {
-                    const end = sb.buffered.end(sb.buffered.length - 1) - 5;
-                    const start = sb.buffered.start(0);
-                    if (end > start) {
-                        sb.remove(start, end);
-                        ms.setLiveSeekableRange(end, end + 5);
-                    }
-                    // console.debug("VideoRTC.buffered", start, end);
+                } catch (e) {
+                    // console.debug(e);
                 }
             });
 
@@ -406,55 +431,12 @@ class VideoRTC extends HTMLElement {
         }
     }
 
-    internalRTC() {
-        console.debug("VideoRTC.internalRTC");
-
+    onwebrtc() {
         const pc = new RTCPeerConnection(this.pcConfig);
 
-        let mseCodecs = "";
-
         /** @type {HTMLVideoElement} */
         const video2 = document.createElement("video");
-        video2.addEventListener("loadeddata", () => {
-            console.debug("VideoRTC.video.loadeddata", video2.readyState, pc.connectionState);
-
-            if (pc.connectionState === "connected" || pc.connectionState === "connecting") {
-                // Video+Audio > Video, H265 > H264, Video > Audio, WebRTC > MSE
-                let rtcPriority = 0, msePriority = 0;
-
-                /** @type {MediaStream} */
-                const rtc = video2.srcObject;
-                if (rtc.getVideoTracks().length > 0) rtcPriority += 0x220;
-                if (rtc.getAudioTracks().length > 0) rtcPriority += 0x102;
-
-                if (mseCodecs.indexOf("hvc1.") >= 0) msePriority += 0x230;
-                if (mseCodecs.indexOf("avc1.") >= 0) msePriority += 0x210;
-                if (mseCodecs.indexOf("mp4a.") >= 0) msePriority += 0x101;
-
-                if (rtcPriority >= msePriority) {
-                    console.debug("VideoRTC.select RTC mode", rtcPriority, msePriority);
-
-                    this.video.controls = true;
-                    this.video.srcObject = rtc;
-                    this.play();
-
-                    this.pcState = WebSocket.OPEN;
-
-                    this.wsState = WebSocket.CLOSED;
-                    this.ws.close();
-                    this.ws = null;
-                } else {
-                    console.debug("VideoRTC.select MSE mode", rtcPriority, msePriority);
-
-                    pc.close();
-
-                    this.pcState = WebSocket.CLOSED;
-                    this.pc = null;
-                }
-            }
-
-            video2.srcObject = null;
-        }, {once: true});
+        video2.addEventListener("loadeddata", ev => this.onpcvideo(ev), {once: true});
 
         pc.addEventListener("icecandidate", ev => {
             const candidate = ev.candidate ? ev.candidate.toJSON().candidate : "";
@@ -462,8 +444,6 @@ class VideoRTC extends HTMLElement {
         });
 
         pc.addEventListener("track", ev => {
-            console.debug("VideoRTC.pc.track", ev.streams.length);
-
             // when stream already init
             if (video2.srcObject !== null) return;
 
@@ -477,30 +457,29 @@ class VideoRTC extends HTMLElement {
         });
 
         pc.addEventListener("connectionstatechange", () => {
-            console.debug("VideoRTC.pc.connectionstatechange", this.pc.connectionState);
-
             if (pc.connectionState === "failed" || pc.connectionState === "disconnected") {
                 pc.close(); // stop next events
 
                 this.pcState = WebSocket.CLOSED;
                 this.pc = null;
 
-                if (this.wsState === WebSocket.CLOSED && this.isConnected) {
-                    this.connectedCallback();
-                }
+                this.onconnect();
             }
         });
 
         this.onmessage["webrtc"] = msg => {
             switch (msg.type) {
                 case "webrtc/candidate":
-                    pc.addIceCandidate({candidate: msg.value, sdpMid: ""}).catch(() => console.debug);
+                    pc.addIceCandidate({
+                        candidate: msg.value,
+                        sdpMid: "0"
+                    }).catch(() => console.debug);
                     break;
                 case "webrtc/answer":
-                    pc.setRemoteDescription({type: "answer", sdp: msg.value}).catch(() => console.debug);
-                    break;
-                case "mse":
-                    mseCodecs = msg.value;
+                    pc.setRemoteDescription({
+                        type: "answer",
+                        sdp: msg.value
+                    }).catch(() => console.debug);
                     break;
                 case "error":
                     if (msg.value.indexOf("webrtc/offer") < 0) return;
@@ -522,52 +501,88 @@ class VideoRTC extends HTMLElement {
         this.pc = pc;
     }
 
-    internalMJPEG() {
-        console.debug("VideoRTC.internalMJPEG");
+    /**
+     * @param ev {Event}
+     */
+    onpcvideo(ev) {
+        if (!this.pc) return;
 
+        /** @type {HTMLVideoElement} */
+        const video2 = ev.target;
+        const state = this.pc.connectionState;
+
+        // Firefox doesn't support pc.connectionState
+        if (state === "connected" || state === "connecting" || !state) {
+            // Video+Audio > Video, H265 > H264, Video > Audio, WebRTC > MSE
+            let rtcPriority = 0, msePriority = 0;
+
+            /** @type {MediaStream} */
+            const ms = video2.srcObject;
+            if (ms.getVideoTracks().length > 0) rtcPriority += 0x220;
+            if (ms.getAudioTracks().length > 0) rtcPriority += 0x102;
+
+            if (this.mseCodecs.indexOf("hvc1.") >= 0) msePriority += 0x230;
+            if (this.mseCodecs.indexOf("avc1.") >= 0) msePriority += 0x210;
+            if (this.mseCodecs.indexOf("mp4a.") >= 0) msePriority += 0x101;
+
+            if (rtcPriority >= msePriority) {
+                this.video.srcObject = ms;
+                this.play();
+
+                this.pcState = WebSocket.OPEN;
+
+                this.wsState = WebSocket.CLOSED;
+                this.ws.close();
+                this.ws = null;
+            } else {
+                this.pcState = WebSocket.CLOSED;
+                this.pc.close();
+                this.pc = null;
+            }
+        }
+
+        video2.srcObject = null;
+    }
+
+    onmjpeg() {
         this.ondata = data => {
+            this.video.controls = false;
             this.video.poster = "data:image/jpeg;base64," + VideoRTC.btoa(data);
         };
 
         this.send({type: "mjpeg"});
-        this.video.controls = false;
     }
 
-    internalMP4() {
-        console.debug("VideoRTC.internalMP4");
+    onmp4() {
+        /** @type {HTMLCanvasElement} **/
+        const canvas = document.createElement("canvas");
+        /** @type {CanvasRenderingContext2D} */
+        let context;
 
         /** @type {HTMLVideoElement} */
-        let video2;
+        const video2 = document.createElement("video");
+        video2.autoplay = true;
+        video2.playsInline = true;
+        video2.muted = true;
 
-        this.ondata = data => {
-            // first video with default position (set container size)
-            // second video with position=absolute and top=0px
-            if (video2) {
-                this.removeChild(this.video);
-                this.video.src = "";
-                this.video = video2;
-                video2.style.position = "";
-                video2.style.top = "";
+        video2.addEventListener("loadeddata", ev => {
+            if (!context) {
+                canvas.width = video2.videoWidth;
+                canvas.height = video2.videoHeight;
+                context = canvas.getContext('2d');
             }
 
-            video2 = this.video.cloneNode();
-            video2.style.position = "absolute";
-            video2.style.top = "0px";
-            this.appendChild(video2);
+            context.drawImage(video2, 0, 0, canvas.width, canvas.height);
 
-            video2.src = "data:video/mp4;base64," + VideoRTC.btoa(data);
-            video2.play().catch(() => console.log);
-        };
-
-        this.ws.addEventListener("close", () => {
-            if (!video2) return;
-
-            this.removeChild(video2);
-            video2.src = "";
+            this.video.controls = false;
+            this.video.poster = canvas.toDataURL("image/jpeg");
         });
 
+        this.ondata = data => {
+            video2.src = "data:video/mp4;base64," + VideoRTC.btoa(data);
+        };
+
         this.send({type: "mp4", value: this.codecs("mp4")});
-        this.video.controls = false;
     }
 
     static btoa(buffer) {
@@ -580,5 +595,3 @@ class VideoRTC extends HTMLElement {
         return window.btoa(binary);
     }
 }
-
-customElements.define("video-rtc", VideoRTC);

www/video-stream.js

@@ -0,0 +1,98 @@
+import {VideoRTC} from "./video-rtc.js";
+
+class VideoStream extends VideoRTC {
+    set divMode(value) {
+        this.querySelector(".mode").innerText = value;
+        this.querySelector(".status").innerText = "";
+    }
+
+    set divError(value) {
+        const state = this.querySelector(".mode").innerText;
+        if (state !== "loading") return;
+        this.querySelector(".mode").innerText = "error";
+        this.querySelector(".status").innerText = value;
+    }
+
+    /**
+     * Custom GUI
+     */
+    oninit() {
+        console.debug("stream.oninit");
+        super.oninit();
+
+        this.innerHTML = `
+        <style>
+        video-stream {
+            position: relative;
+        }
+        .info {
+            position: absolute;
+            top: 0;
+            left: 0;
+            right: 0;
+            padding: 12px;
+            color: white;
+            display: flex;
+            justify-content: space-between;
+            pointer-events: none;
+        }
+        </style>
+        <div class="info">
+            <div class="status"></div>
+            <div class="mode"></div>
+        </div>
+        `;
+
+        const info = this.querySelector(".info")
+        this.insertBefore(this.video, info);
+    }
+
+    onconnect() {
+        console.debug("stream.onconnect");
+        const result = super.onconnect();
+        if (result) this.divMode = "loading";
+        return result;
+    }
+
+    ondisconnect() {
+        console.debug("stream.ondisconnect");
+        super.ondisconnect();
+    }
+
+    onopen() {
+        console.debug("stream.onopen");
+        const result = super.onopen();
+
+        this.onmessage["stream"] = msg => {
+            console.debug("stream.onmessge", msg);
+            switch (msg.type) {
+                case "error":
+                    this.divError = msg.value;
+                    break;
+                case "mse":
+                case "mp4":
+                case "mjpeg":
+                    this.divMode = msg.type.toUpperCase();
+                    break;
+            }
+        }
+
+        return result;
+    }
+
+    onclose() {
+        console.debug("stream.onclose");
+        return super.onclose();
+    }
+
+    onpcvideo(ev) {
+        console.debug("stream.onpcvideo");
+        super.onpcvideo(ev);
+
+        if (this.pcState !== WebSocket.CLOSED) {
+            this.divMode = "RTC";
+        }
+    }
+}
+
+customElements.define("video-stream", VideoStream);

www/webrtc.html

@@ -48,7 +48,7 @@
             console.debug('ws.onmessage', msg);
 
             if (msg.type === 'webrtc/candidate') {
-                pc.addIceCandidate({candidate: msg.value, sdpMid: ''});
+                pc.addIceCandidate({candidate: msg.value, sdpMid: '0'});
             } else if (msg.type === 'webrtc/answer') {
                 pc.setRemoteDescription({type: 'answer', sdp: msg.value});
             }