Update version to 1.0.0

Fix timezone in logs
Simplify Dockerfile
2025-09-27 04:36:12 +08:00 · 2023-01-20 17:07:43 +03:00 · 2023-01-20 13:45:01 +03:00 · 2023-01-20 11:23:28 +03:00 · 2023-01-20 10:54:26 +03:00 · 2023-01-20 10:26:57 +03:00
94 changed files with 4459 additions and 1542 deletions
--- a/.github/workflows/builder.yml
+++ b/.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"
--- a/.github/workflows/docker.yml
+++ b/.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: ${{ steps.meta-hw.outputs.labels }}
--- a/.github/workflows/release.yml
+++ b/.github/workflows/release.yml
@@ -0,0 +1,92 @@
+name: release
+
+on:
+  workflow_dispatch:
+  push:
+    tags:
+      - 'v*'
+
+jobs:
+  build-and-release:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v3
+      - name: Generate changelog
+        run: |
+          echo -e "$(git log $(git describe --tags --abbrev=0)..HEAD --oneline | awk '{print "- "$0}')" > CHANGELOG.md
+      - name: Build Go binaries
+        run: |
+          #!/bin/bash
+          
+          esport CGO_ENABLED=0
+          
+          mkdir artifacts
+          export GOOS=windows
+          export GOARCH=amd64
+          export FILENAME=artifacts/go2rtc_win64.zip
+          go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel "$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 "$FILENAME"
+          
+          export GOOS=darwin
+          export GOARCH=amd64
+          export FILENAME=artifacts/go2rtc_mac_amd64.zip
+          go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel "$FILENAME" go2rtc
+          
+          export GOOS=darwin
+          export GOARCH=arm64
+          export FILENAME=artifacts/go2rtc_mac_arm64.zip
+          go build -ldflags "-s -w" -trimpath && 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 ${{ env.RELEASE_VERSION }}
+          body_path: CHANGELOG.md
+          draft: false
+          prerelease: false
--- a/61
+++ b/61
@@ -0,0 +1,61 @@
+# syntax=docker/dockerfile:labs
+
+# 0. Prepare images
+ARG PYTHON_VERSION="3.11"
+ARG GO_VERSION="1.19"
+ARG NGROK_VERSION="3"
+
+FROM python:${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"]
--- a/README.md
+++ b/README.md
@@ -6,9 +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 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) 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 (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
@@ -26,35 +27,6 @@ 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)

-## Codecs negotiation
-
-For example, you want to watch RTSP-stream from [Dahua IPC-K42](https://www.dahuasecurity.com/fr/products/All-Products/Network-Cameras/Wireless-Series/Wi-Fi-Series/4MP/IPC-K42) camera in your Chrome browser.
-
- this camera support 2-way audio standard **ONVIF Profile T**
- this camera support codecs **H264, H265** for send video, and you select `H264` in camera settings
- this camera support codecs **AAC, PCMU, PCMA** for send audio (from mic), and you select `AAC/16000` in camera settings
- this camera support codecs **AAC, PCMU, PCMA** for receive audio (to speaker), you don't need to select them
- your browser support codecs **H264, VP8, VP9, AV1** for receive video, you don't need to select them
- your browser support codecs **OPUS, PCMU, PCMA** for send and receive audio, you don't need to select them
- you can't get camera audio directly, because its audio codecs doesn't match with your browser codecs 
-   - so you decide to use transcoding via FFmpeg and add this setting to config YAML file
-   - you have chosen `OPUS/48000/2` codec, because it is higher quality than the `PCMU/8000` or `PCMA/8000`
-
-Now you have stream with two sources - **RTSP and FFmpeg**:
-
-```yaml
-streams:
-  dahua:
-    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
-    - ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0#audio=opus
-```
-
-**go2rtc** automatically match codecs for you browser and all your stream sources. This called **multi-source 2-way codecs negotiation**. And this is one of the main features of this app.
-
-![](assets/codecs.svg)
-
-**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.
-
 ## Fast start

 1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
@@ -65,7 +37,6 @@ streams:
 - 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:**

@@ -82,9 +53,9 @@ 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.

@@ -99,27 +70,17 @@ 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 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"
-```
+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).

 ## Configuration

-Create file `go2rtc.yaml` next to the app.
+Create file `go2rtc.yaml`. go2rtc will search this file in current work dirrectory by default.

 - 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/))
+- `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

 Available modules:

@@ -127,12 +88,15 @@ Available modules:
 - [api](#module-api) - HTTP API (important for WebRTC support)
 - [rtsp](#module-rtsp) - RTSP Server (important for FFmpeg support)
 - [webrtc](#module-webrtc) - WebRTC Server
- [mp4](#module-mp4) - MSE, MP4 stream and MP4 shapshot
+- [mp4](#module-mp4) - MSE, MP4 stream and MP4 shapshot Server
+- [mjpeg](#module-mjpeg) - MJPEG Server
 - [ffmpeg](#source-ffmpeg) - FFmpeg integration
 - [ngrok](#module-ngrok) - Ngrok integration (external access for private network)
 - [hass](#module-hass) - Home Assistant integration
 - [log](#module-log) - logs config

+Full default config [example](https://github.com/AlexxIT/go2rtc/wiki/Configuration).
+
 ### Module: Streams

 **go2rtc** support different stream source types. You can config one or multiple links of any type as stream source.
@@ -140,8 +104,9 @@ Available modules:
 Available source types:

 - [rtsp](#source-rtsp) - `RTSP` and `RTSPS` cameras
- [rtmp](#source-rtmp) - `RTMP` and `HTTP-FLV` streams
- [ffmpeg](#source-ffmpeg) - FFmpeg integration (`MJPEG`, `HLS`, `files` and others)
+- [rtmp](#source-rtmp) - `RTMP` streams
+- [http](#source-http) - `HTTP-FLV`, `JPEG` (snapshots), `MJPEG` streams 
+- [ffmpeg](#source-ffmpeg) - FFmpeg integration (`HLS`, `files` and many others)
 - [ffmpeg:device](#source-ffmpeg-device) - local USB Camera or Webcam
 - [exec](#source-exec) - advanced FFmpeg and GStreamer integration
 - [echo](#source-echo) - get stream link from bash or python
@@ -176,13 +141,35 @@ streams:

 #### Source: RTMP

-You can get stream from RTMP server, for example [Frigate](https://docs.frigate.video/configuration/rtmp). Support ONLY `H264` video codec without audio.
+You can get stream from RTMP server, for example [Frigate](https://docs.frigate.video/configuration/rtmp).

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

+#### Source: HTTP
+
+Support Content-Type:
+
+- **HTTP-FLV** (`video/x-flv`) - same as RTMP, but over HTTP
+- **HTTP-JPEG** (`image/jpeg`) - camera snapshot link, can be converted by go2rtc to MJPEG stream
+- **HTTP-MJPEG** (`multipart/x`) - simple MJPEG stream over HTTP
+
+```yaml
+streams:
+  # [HTTP-FLV] stream in video/x-flv format
+  http_flv: http://192.168.1.123:20880/api/camera/stream/780900131155/657617
+  
+  # [JPEG] snapshots from Dahua camera, will be converted to MJPEG stream
+  dahua_snap: http://admin:password@192.168.1.123/cgi-bin/snapshot.cgi?channel=1
+
+  # [MJPEG] stream will be proxied without modification
+  http_mjpeg: https://mjpeg.sanford.io/count.mjpeg
+```
+
+**PS.** Dahua camera has bug: if you select MJPEG codec for RTSP second stream - snapshot won't work.
+
 #### Source: FFmpeg

 You can get any stream or file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.
@@ -207,24 +194,30 @@ streams:
  hls: ffmpeg:https://devstreaming-cdn.apple.com/videos/streaming/examples/bipbop_16x9/gear5/prog_index.m3u8#video=copy

  # [MJPEG] video will be transcoded to H264
-  mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg?stream=half&fps=15#video=h264
+  mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg#video=h264

  # [RTSP] video with rotation, should be transcoded, so select H264
-  rotate: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#raw=-vf transpose=1#video=h264
+  rotate: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=h264#rotate=90
 ```

-All trascoding formats has [built-in templates](https://github.com/AlexxIT/go2rtc/blob/master/cmd/ffmpeg/ffmpeg.go): `h264`, `h264/ultra`, `h264/high`, `h265`, `opus`, `pcmu`, `pcmu/16000`, `pcmu/48000`, `pcma`, `pcma/16000`, `pcma/48000`, `aac/16000`.
+All trascoding formats has [built-in templates](https://github.com/AlexxIT/go2rtc/blob/master/cmd/ffmpeg/ffmpeg.go): `h264`, `h264/ultra`, `h264/high`, `h265`, `opus`, `pcmu`, `pcmu/16000`, `pcmu/48000`, `pcma`, `pcma/16000`, `pcma/48000`, `aac`, `aac/16000`.

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

 ```yaml
 ffmpeg:
-  bin: ffmpeg                                        # path to ffmpeg binary
-  h264: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
+  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..."
 ```

-Also you can use `raw` param for any additional FFmpeg arguments. As example for video rotation (`#raw=-vf transpose=1`). Remember that rotation is not possible without transcoding, so add supported codec as second param (`#video=h264`).
+- 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

@@ -279,6 +272,24 @@ You can pair device with go2rtc on the HomeKit page. If you can't see your devic

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

+**Important:**
+
+- HomeKit audio uses very non-standard **AAC-ELD** codec with very non-standard params and specification violation
+- Audio can be transcoded by [ffmpeg](#source-ffmpeg) source with `#async` option
+- Audio can be played by `ffplay` with `-use_wallclock_as_timestamps 1 -async 1` options
+- Audio can't be played in `VLC` and probably any other player
+
+Recommended settings for using HomeKit Camera with WebRTC, MSE, MP4, RTSP:
+
+```
+streams:
+  aqara_g3:
+    - hass:Camera-Hub-G3-AB12
+    - ffmpeg:aqara_g3#audio=aac#audio=opus#async
+```
+
+RTSP link with "normal" audio for any player: `rtsp://192.168.1.123:8554/aqara_g3?video&audio=aac`
+
 **This source is in active development!** Tested only with [Aqara Camera Hub G3](https://www.aqara.com/eu/product/camera-hub-g3) (both EU and CN versions).

 #### Source: Ivideon
@@ -312,7 +323,34 @@ 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
@@ -320,14 +358,18 @@ The HTTP API is the main part for interacting with the application. Default addr

 ```yaml
 api:
-  listen: ":1984"  # HTTP API port ("" - disabled)
-  base_path: ""    # API prefix for serve on suburl
-  static_dir: ""   # folder for static files (custom web interface)
+  listen: ":1984"    # default ":1984", HTTP API port ("" - disabled)
+  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 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
+- 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

@@ -336,55 +378,58 @@ 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)

 ```yaml
 rtsp:
-  listen: ":8554"
+  listen: ":8554"  # RTSP Server TCP port, default - 8554
+  username: admin  # optional, default - disabled
+  password: pass   # optional, default - disabled
 ```

 ### 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: ...
 ```
@@ -478,7 +523,8 @@ View almost any Hass camera using `WebRTC` technology, supported codecs `H264`/`
 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
+2. RTSPtoWebRTC > Configure > STUN server: `stun.l.google.com:19302`
+3. Use Picture Entity or Picture Glance lovelace card

 You can add camera `entity_id` to [go2rtc config](#configuration) if you need transcoding:

@@ -494,21 +540,38 @@ 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. Progressive MP4 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, doesn't work in Safari 
+
+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`

 ### Module: MJPEG

-**Important.** For stream as MJPEG format, your source MUST contain the MJPEG codec. If your camera outputs H264/H265 - you SHOULD use transcoding. With this example, your stream will have both H264 and MJPEG codecs:
+**Important.** For stream as MJPEG format, your source MUST contain the MJPEG codec. If your stream has a MJPEG codec - you can receive **MJPEG stream** or **JPEG snapshots** via API.
+
+You can receive an MJPEG stream in several ways:
+
+- some cameras support MJPEG codec inside [RTSP stream](#source-rtsp) (ex. second stream for Dahua cameras)
+- some cameras has HTTP link with [MJPEG stream](#source-http)
+- some cameras has HTTP link with snapshots - go2rtc can convert them to [MJPEG stream](#source-http)
+- you can convert H264/H265 stream from your camera via [FFmpeg integraion](#source-ffmpeg)
+
+With this example, your stream will have both H264 and MJPEG codecs:

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

-Example link to MJPEG: `http://192.168.1.123:1984/api/stream.mjpeg?src=camera1`
+API examples:
+
+- MJPEG stream: `http://192.168.1.123:1984/api/stream.mjpeg?src=camera1`
+- JPEG snapshots: `http://192.168.1.123:1984/api/frame.jpeg?src=camera1`

 ### Module: Log

@@ -527,7 +590,7 @@ 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:

@@ -539,7 +602,7 @@ 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
@@ -549,26 +612,25 @@ 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 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.

-Device | WebRTC | MSE | MP4
-------|--------|-----|----
-*latency*       | best        | medium      | bad
-Desktop Chrome  | H264        | H264, H265* | H264, H265*
-Desktop Safari  | H264, H265* | H264        | no
-Desktop Edge    | H264        | H264, H265* | H264, H265*
-Desktop Firefox | H264        | H264        | H264
-Desktop Opera   | no          | H264        | H264
-iPhone Safari   | H264, H265* | no          | no
-iPad Safari     | H264, H265* | H264        | no
-Android Chrome  | H264        | H264        | H264
-masOS Hass App  | no          | no          | no
+| 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          |

- Chrome H265: [read this](https://github.com/StaZhu/enable-chromium-hevc-hardware-decoding)
+- 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/)
 - Desktop Safari H265: Menu > Develop > Experimental > WebRTC H265
 - iOS Safari H265: Settings > Safari > Advanced > Experimental > WebRTC H265
@@ -578,6 +640,41 @@ masOS Hass App  | no          | no          | no
 - WebRTC audio codecs: `PCMU/8000`, `PCMA/8000`, `OPUS/48000/2`
 - MSE/MP4 audio codecs: `AAC`

+**Apple devices**
+
+- all Apple devices don't support MP4 stream (they only support progressive loading of static files)
+- iPhones don't support MSE technology because it competes with the HLS technology, invented by Apple
+- HLS is the worst technology for **live** streaming, it still exists only because of iPhones
+
+## Codecs negotiation
+
+For example, you want to watch RTSP-stream from [Dahua IPC-K42](https://www.dahuasecurity.com/fr/products/All-Products/Network-Cameras/Wireless-Series/Wi-Fi-Series/4MP/IPC-K42) camera in your Chrome browser.
+
+- this camera support 2-way audio standard **ONVIF Profile T**
+- this camera support codecs **H264, H265** for send video, and you select `H264` in camera settings
+- this camera support codecs **AAC, PCMU, PCMA** for send audio (from mic), and you select `AAC/16000` in camera settings
+- this camera support codecs **AAC, PCMU, PCMA** for receive audio (to speaker), you don't need to select them
+- your browser support codecs **H264, VP8, VP9, AV1** for receive video, you don't need to select them
+- your browser support codecs **OPUS, PCMU, PCMA** for send and receive audio, you don't need to select them
+- you can't get camera audio directly, because its audio codecs doesn't match with your browser codecs
+  - so you decide to use transcoding via FFmpeg and add this setting to config YAML file
+  - you have chosen `OPUS/48000/2` codec, because it is higher quality than the `PCMU/8000` or `PCMA/8000`
+
+Now you have stream with two sources - **RTSP and FFmpeg**:
+
+```yaml
+streams:
+  dahua:
+    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
+    - ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0#audio=opus
+```
+
+**go2rtc** automatically match codecs for you browser and all your stream sources. This called **multi-source 2-way codecs negotiation**. And this is one of the main features of this app.
+
+![](assets/codecs.svg)
+
+**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.
+
 ## TIPS

 **Using apps for low RTSP delay**
@@ -585,6 +682,10 @@ masOS Hass App  | no          | no          | no
 - `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?**
--- a/build/docker/run.sh
+++ b/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}"
--- a/build/hassio/Dockerfile
+++ b/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 $BUILD_FROM
-
-# 3. Copy go2rtc and ngrok to release
-COPY --from=build /go2rtc/go2rtc /usr/local/bin
-COPY --from=build /go2rtc/ngrok /usr/local/bin
-
-# 4. Install ffmpeg
-# apk base  OK: 22  MiB in 40  packages
-# ffmpeg    OK: 113 MiB in 110 packages
-# python3   OK: 161 MiB in 114 packages
-RUN apk add --no-cache ffmpeg python3
-
-# 5. Copy run to release
-COPY run.sh /
-RUN chmod a+x /run.sh
-
-CMD [ "/run.sh" ]
--- a/build/hassio/config.yaml
+++ b/build/hassio/config.yaml
@@ -1,6 +0,0 @@
-# https://github.com/home-assistant/builder/blob/master/builder.sh
-name: go2rtc
-description: Ultimate camera streaming application
-url: https://github.com/AlexxIT/go2rtc
-image: alexxit/go2rtc
-arch: [ amd64, aarch64, i386, armv7 ]
--- a/build/hassio/run.sh
+++ b/build/hassio/run.sh
@@ -1,14 +0,0 @@
-#!/usr/bin/with-contenv bashio
-
-set +e
-
-# set cwd for go2rtc (for config file, Hass integration, etc)
-cd /config
-
-# add the feature to override go2rtc binary from Hass config folder
-export PATH="/config:$PATH"
-
-while true; do
-  go2rtc
-  sleep 5
-done
--- a/cmd/api/api.go
+++ b/cmd/api/api.go
@@ -3,20 +3,24 @@ package api
 import (
 	"encoding/json"
 	"github.com/AlexxIT/go2rtc/cmd/app"
-	"github.com/AlexxIT/go2rtc/cmd/streams"
-	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"github.com/gorilla/websocket"
 	"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"`
 		} `yaml:"api"`
 	}

@@ -34,9 +38,11 @@ func Init() {
 	log = app.GetLogger("api")

 	initStatic(cfg.Mod.StaticDir)
-	initWS()
+	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
@@ -48,16 +54,22 @@ func Init() {

 	log.Info().Str("addr", cfg.Mod.Listen).Msg("[api] listen")

+	s := http.Server{}
+	s.Handler = http.DefaultServeMux // 4th
+
+	if cfg.Mod.Origin == "*" {
+		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() {
-		s := http.Server{}
-
-		if log.Trace().Enabled() {
-			s.Handler = http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
-				log.Trace().Stringer("url", r.URL).Msgf("[api] %s", r.Method)
-				http.DefaultServeMux.ServeHTTP(w, r)
-			})
-		}
-
 		if err = s.Serve(listener); err != nil {
 			log.Fatal().Err(err).Msg("[api] serve")
 		}
@@ -75,65 +87,60 @@ func HandleFunc(pattern string, handler http.HandlerFunc) {
 	http.HandleFunc(pattern, handler)
 }

-func HandleWS(msgType string, handler WSHandler) {
-	wsHandlers[msgType] = handler
-}
+const StreamNotFound = "stream not found"

 var basePath string
 var log zerolog.Logger
-var wsHandlers = make(map[string]WSHandler)

-func streamsHandler(w http.ResponseWriter, r *http.Request) {
-	src := r.URL.Query().Get("src")
-	name := r.URL.Query().Get("name")
-	
-	if name == "" {
-		name = src
-	}
-
-	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 middlewareLog(next http.Handler) http.Handler {
+	return http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		log.Trace().Msgf("[api] %s %s %s", r.Method, r.URL, r.RemoteAddr)
+		next.ServeHTTP(w, r)
+	})
 }

-func apiWS(w http.ResponseWriter, r *http.Request) {
-	ctx := new(Context)
-	if err := ctx.Upgrade(w, r); err != nil {
-		log.Error().Err(err).Msg("[api.ws] upgrade")
-		return
-	}
-	defer ctx.Close()
-
-	for {
-		msg := new(streamer.Message)
-		if err := ctx.Conn.ReadJSON(msg); err != nil {
-			if websocket.IsUnexpectedCloseError(
-				err, websocket.CloseGoingAway, websocket.CloseAbnormalClosure,
-			) {
-				log.Error().Err(err).Msg("[api.ws] readJSON")
+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
 			}
-			return
 		}

-		handler := wsHandlers[msg.Type]
-		if handler != nil {
-			handler(ctx, msg)
-		}
+		next.ServeHTTP(w, r)
+	})
+}
+
+func middlewareCORS(next http.Handler) http.Handler {
+	return http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		w.Header().Set("Access-Control-Allow-Origin", "*")
+		w.Header().Set("Access-Control-Allow-Methods", "GET, POST, PUT, DELETE, OPTIONS")
+		next.ServeHTTP(w, r)
+	})
+}
+
+var mu sync.Mutex
+
+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()
 	}
 }
+
+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)
+}
--- a/cmd/api/config.go
+++ b/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
+}
--- a/cmd/api/ws.go
+++ b/cmd/api/ws.go
@@ -1,85 +1,154 @@
 package api

 import (
-	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/gorilla/websocket"
 	"net/http"
 	"net/url"
 	"strings"
 	"sync"
+	"time"
 )

-func initWS() {
+// Message - struct for data exchange in Web API
+type Message struct {
+	Type  string      `json:"type"`
+	Value interface{} `json:"value,omitempty"`
+}
+
+type WSHandler func(tr *Transport, msg *Message) error
+
+func HandleWS(msgType string, handler WSHandler) {
+	wsHandlers[msgType] = handler
+}
+
+var wsHandlers = make(map[string]WSHandler)
+
+func initWS(origin string) {
 	wsUp = &websocket.Upgrader{
 		ReadBufferSize:  1024,
-		WriteBufferSize: 512000,
+		WriteBufferSize: 2028,
 	}
-	wsUp.CheckOrigin = func(r *http.Request) bool {
-		origin := r.Header["Origin"]
-		if len(origin) == 0 {
-			return true
-		}
-		o, err := url.Parse(origin[0])
-		if err != nil {
+
+	switch origin {
+	case "":
+		// same origin + ignore port
+		wsUp.CheckOrigin = func(r *http.Request) bool {
+			origin := r.Header["Origin"]
+			if len(origin) == 0 {
+				return true
+			}
+			o, err := url.Parse(origin[0])
+			if err != nil {
+				return false
+			}
+			if o.Host == r.Host {
+				return true
+			}
+			log.Trace().Msgf("[api.ws] origin=%s, host=%s", o.Host, r.Host)
+			// https://github.com/AlexxIT/go2rtc/issues/118
+			if i := strings.IndexByte(o.Host, ':'); i > 0 {
+				return o.Host[:i] == r.Host
+			}
 			return false
 		}
-		if o.Host == r.Host {
+	case "*":
+		// any origin
+		wsUp.CheckOrigin = func(r *http.Request) bool {
 			return true
 		}
-		log.Trace().Msgf("[api.ws] origin: %s, host: %s", o.Host, r.Host)
-		// some users change Nginx external port using Docker port
-		// so origin will be with a port and host without
-		if i := strings.IndexByte(o.Host, ':'); i > 0 {
-			return o.Host[:i] == r.Host
-		}
-		return false
 	}
 }

+func apiWS(w http.ResponseWriter, r *http.Request) {
+	ws, err := wsUp.Upgrade(w, r, nil)
+	if err != nil {
+		origin := r.Header.Get("Origin")
+		log.Error().Err(err).Caller().Msgf("host=%s origin=%s", r.Host, origin)
+		return
+	}
+
+	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 {
+			_ = ws.WriteJSON(msg)
+		}
+	})
+
+	for {
+		msg := new(Message)
+		if err = ws.ReadJSON(msg); err != nil {
+			log.Trace().Err(err).Caller().Send()
+			_ = ws.Close()
+			break
+		}
+
+		if handler := wsHandlers[msg.Type]; handler != nil {
+			go func() {
+				if err = handler(tr, msg); err != nil {
+					tr.Write(&Message{Type: "error", Value: msg.Type + ": " + err.Error()})
+				}
+			}()
+		}
+	}
+
+	tr.Close()
+}
+
 var wsUp *websocket.Upgrader

-type WSHandler func(ctx *Context, msg *streamer.Message)
-
-type Context struct {
-	Conn     *websocket.Conn
+type Transport struct {
 	Request  *http.Request
 	Consumer interface{} // TODO: rewrite

-	onClose []func()
-	mu      sync.Mutex
+	closed bool
+	mx     sync.Mutex
+	wrmx   sync.Mutex
+
+	onChange func()
+	onWrite  func(msg interface{})
+	onClose  []func()
 }

-func (ctx *Context) Upgrade(w http.ResponseWriter, r *http.Request) (err error) {
-	ctx.Conn, err = wsUp.Upgrade(w, r, nil)
-	ctx.Request = r
-	return
+func (t *Transport) OnWrite(f func(msg interface{})) {
+	t.mx.Lock()
+	if t.onChange != nil {
+		t.onChange()
+	}
+	t.onWrite = f
+	t.mx.Unlock()
 }

-func (ctx *Context) Close() {
-	for _, f := range ctx.onClose {
+func (t *Transport) Write(msg interface{}) {
+	t.wrmx.Lock()
+	t.onWrite(msg)
+	t.wrmx.Unlock()
+}
+
+func (t *Transport) Close() {
+	t.mx.Lock()
+	for _, f := range t.onClose {
 		f()
 	}
-	_ = ctx.Conn.Close()
+	t.closed = true
+	t.mx.Unlock()
 }

-func (ctx *Context) Write(msg interface{}) {
-	ctx.mu.Lock()
+func (t *Transport) OnChange(f func()) {
+	t.mx.Lock()
+	t.onChange = f
+	t.mx.Unlock()
+}

-	if data, ok := msg.([]byte); ok {
-		_ = ctx.Conn.WriteMessage(websocket.BinaryMessage, data)
+func (t *Transport) OnClose(f func()) {
+	t.mx.Lock()
+	if t.closed {
+		f()
 	} else {
-		_ = ctx.Conn.WriteJSON(msg)
+		t.onClose = append(t.onClose, f)
 	}
-
-	ctx.mu.Unlock()
-}
-
-func (ctx *Context) Error(err error) {
-	ctx.Write(&streamer.Message{
-		Type: "error", Value: err.Error(),
-	})
-}
-
-func (ctx *Context) OnClose(f func()) {
-	ctx.onClose = append(ctx.onClose, f)
+	t.mx.Unlock()
 }
--- a/cmd/app/app.go
+++ b/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.3"
+var Version = "1.0.0"
 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
--- a/cmd/debug/debug.go
+++ b/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
 }
--- a/cmd/debug/stack.go
+++ b/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) {
--- a/cmd/exec/exec.go
+++ b/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 {
--- a/cmd/ffmpeg/ffmpeg.go
+++ b/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 +19,226 @@ 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 ultrafast -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",
-	}
+	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",
+
+	// 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"
+		}
+		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 +251,76 @@ 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()
+}
--- a/cmd/ffmpeg/hardware.go
+++ b/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
+}
--- a/cmd/ffmpeg/hardware_darwin.go
+++ b/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
+}
--- a/cmd/ffmpeg/hardware_linux.go
+++ b/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
+}
--- a/cmd/ffmpeg/hardware_windows.go
+++ b/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
+}
--- a/cmd/http/http.go
+++ b/cmd/http/http.go
@@ -0,0 +1,56 @@
+package http
+
+import (
+	"errors"
+	"fmt"
+	"github.com/AlexxIT/go2rtc/cmd/streams"
+	"github.com/AlexxIT/go2rtc/pkg/mjpeg"
+	"github.com/AlexxIT/go2rtc/pkg/rtmp"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/AlexxIT/go2rtc/pkg/tcp"
+	"net/http"
+	"strings"
+)
+
+func Init() {
+	streams.HandleFunc("http", handle)
+	streams.HandleFunc("https", handle)
+}
+
+func handle(url string) (streamer.Producer, error) {
+	// first we get the Content-Type to define supported producer
+	req, err := http.NewRequest("GET", url, nil)
+	if err != nil {
+		return nil, err
+	}
+
+	res, err := tcp.Do(req)
+	if err != nil {
+		return nil, err
+	}
+
+	if res.StatusCode != http.StatusOK {
+		return nil, errors.New(res.Status)
+	}
+
+	ct := res.Header.Get("Content-Type")
+	if i := strings.IndexByte(ct, ';'); i > 0 {
+		ct = ct[:i]
+	}
+
+	switch ct {
+	case "image/jpeg", "multipart/x-mixed-replace":
+		return mjpeg.NewClient(res), nil
+	case "video/x-flv":
+		var conn *rtmp.Client
+		if conn, err = rtmp.Accept(res); err != nil {
+			return nil, err
+		}
+		if err = conn.Describe(); err != nil {
+			return nil, err
+		}
+		return conn, nil
+	}
+
+	return nil, fmt.Errorf("unsupported Content-Type: %s", ct)
+}
--- a/cmd/mjpeg/mjpeg.go
+++ b/cmd/mjpeg/mjpeg.go
@@ -1,6 +1,7 @@
 package mjpeg

 import (
+	"errors"
 	"github.com/AlexxIT/go2rtc/cmd/api"
 	"github.com/AlexxIT/go2rtc/cmd/streams"
 	"github.com/AlexxIT/go2rtc/pkg/mjpeg"
@@ -12,18 +13,24 @@ import (
 func Init() {
 	api.HandleFunc("api/frame.jpeg", handlerKeyframe)
 	api.HandleFunc("api/stream.mjpeg", handlerStream)
+
+	api.HandleWS("mjpeg", handlerWS)
 }

 func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
 	src := r.URL.Query().Get("src")
 	stream := streams.GetOrNew(src)
 	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
 		return
 	}

 	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:
@@ -40,8 +47,12 @@ func handlerKeyframe(w http.ResponseWriter, r *http.Request) {

 	stream.RemoveConsumer(cons)

-	w.Header().Set("Content-Type", "image/jpeg")
-	w.Header().Set("Content-Length", strconv.Itoa(len(data)))
+	h := w.Header()
+	h.Set("Content-Type", "image/jpeg")
+	h.Set("Content-Length", strconv.Itoa(len(data)))
+	h.Set("Cache-Control", "no-cache")
+	h.Set("Connection", "close")
+	h.Set("Pragma", "no-cache")

 	if _, err := w.Write(data); err != nil {
 		log.Error().Err(err).Caller().Send()
@@ -54,23 +65,28 @@ func handlerStream(w http.ResponseWriter, r *http.Request) {
 	src := r.URL.Query().Get("src")
 	stream := streams.GetOrNew(src)
 	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
 		return
 	}

-	exit := make(chan struct{})
+	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:
 			data := []byte(header + strconv.Itoa(len(msg)))
-			data = append(data, 0x0D, 0x0A, 0x0D, 0x0A)
+			data = append(data, '\r', '\n', '\r', '\n')
 			data = append(data, msg...)
-			data = append(data, 0x0D, 0x0A)
+			data = append(data, '\r', '\n')

-			if _, err := w.Write(data); err != nil {
-				exit <- struct{}{}
-			}
+			// Chrome bug: mjpeg image always shows the second to last image
+			// https://bugs.chromium.org/p/chromium/issues/detail?id=527446
+			_, _ = w.Write(data)
+			flusher.Flush()
 		}
 	})

@@ -79,11 +95,46 @@ func handlerStream(w http.ResponseWriter, r *http.Request) {
 		return
 	}

-	w.Header().Set("Content-Type", `multipart/x-mixed-replace; boundary=frame`)
+	h := w.Header()
+	h.Set("Content-Type", "multipart/x-mixed-replace; boundary=frame")
+	h.Set("Cache-Control", "no-cache")
+	h.Set("Connection", "close")
+	h.Set("Pragma", "no-cache")

-	<-exit
+	<-r.Context().Done()

 	stream.RemoveConsumer(cons)

 	//log.Trace().Msg("[api.mjpeg] close")
 }
+
+func handlerWS(tr *api.Transport, _ *api.Message) error {
+	src := tr.Request.URL.Query().Get("src")
+	stream := streams.GetOrNew(src)
+	if stream == nil {
+		return errors.New(api.StreamNotFound)
+	}
+
+	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)
+		}
+	})
+
+	if err := stream.AddConsumer(cons); err != nil {
+		log.Error().Err(err).Caller().Send()
+		return err
+	}
+
+	tr.Write(&api.Message{Type: "mjpeg"})
+
+	tr.OnClose(func() {
+		stream.RemoveConsumer(cons)
+	})
+
+	return nil
+}
--- a/cmd/mp4/mp4.go
+++ b/cmd/mp4/mp4.go
@@ -15,7 +15,8 @@ import (
 func Init() {
 	log = app.GetLogger("mp4")

-	api.HandleWS(MsgTypeMSE, handlerWS)
+	api.HandleWS("mse", handlerWSMSE)
+	api.HandleWS("mp4", handlerWSMP4)

 	api.HandleFunc("api/frame.mp4", handlerKeyframe)
 	api.HandleFunc("api/stream.mp4", handlerMP4)
@@ -31,12 +32,13 @@ func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
 	src := r.URL.Query().Get("src")
 	stream := streams.GetOrNew(src)
 	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
 		return
 	}

 	exit := make(chan []byte)

-	cons := &mp4.Keyframe{}
+	cons := &mp4.Segment{OnlyKeyframe: true}
 	cons.Listen(func(msg interface{}) {
 		if data, ok := msg.([]byte); ok && exit != nil {
 			exit <- data
@@ -63,21 +65,25 @@ 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) {
 		return
 	}

-	log.Trace().Msgf("[api.mp4] %+v", r)
-
 	src := r.URL.Query().Get("src")
 	stream := streams.GetOrNew(src)
 	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
 		return
 	}

 	exit := make(chan error)

-	cons := &mp4.Consumer{}
+	cons := &mp4.Consumer{
+		RemoteAddr: r.RemoteAddr,
+		UserAgent:  r.UserAgent(),
+	}
 	cons.Listen(func(msg interface{}) {
 		if data, ok := msg.([]byte); ok {
 			if _, err := w.Write(data); err != nil && exit != nil {
--- a/cmd/mp4/mse.go
+++ b/cmd/mp4/mse.go
@@ -1,57 +0,0 @@
-package mp4
-
-import (
-	"github.com/AlexxIT/go2rtc/cmd/api"
-	"github.com/AlexxIT/go2rtc/cmd/streams"
-	"github.com/AlexxIT/go2rtc/pkg/mp4"
-	"github.com/AlexxIT/go2rtc/pkg/streamer"
-)
-
-const MsgTypeMSE = "mse" // fMP4
-
-const packetSize = 8192
-
-func handlerWS(ctx *api.Context, msg *streamer.Message) {
-	src := ctx.Request.URL.Query().Get("src")
-	stream := streams.GetOrNew(src)
-	if stream == nil {
-		return
-	}
-
-	cons := &mp4.Consumer{}
-	cons.UserAgent = ctx.Request.UserAgent()
-	cons.RemoteAddr = ctx.Request.RemoteAddr
-
-	cons.Listen(func(msg interface{}) {
-		if data, ok := msg.([]byte); ok {
-			for len(data) > packetSize {
-				ctx.Write(data[:packetSize])
-				data = data[packetSize:]
-			}
-			ctx.Write(data)
-		}
-	})
-
-	if err := stream.AddConsumer(cons); err != nil {
-		log.Warn().Err(err).Caller().Send()
-		ctx.Error(err)
-		return
-	}
-
-	ctx.OnClose(func() {
-		stream.RemoveConsumer(cons)
-	})
-
-	ctx.Write(&streamer.Message{Type: MsgTypeMSE, Value: cons.MimeType()})
-
-	data, err := cons.Init()
-	if err != nil {
-		log.Warn().Err(err).Caller().Send()
-		ctx.Error(err)
-		return
-	}
-
-	ctx.Write(data)
-
-	cons.Start()
-}
--- a/cmd/mp4/ws.go
+++ b/cmd/mp4/ws.go
@@ -0,0 +1,142 @@
+package mp4
+
+import (
+	"errors"
+	"github.com/AlexxIT/go2rtc/cmd/api"
+	"github.com/AlexxIT/go2rtc/cmd/streams"
+	"github.com/AlexxIT/go2rtc/pkg/mp4"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"strings"
+)
+
+const packetSize = 1400
+
+func handlerWSMSE(tr *api.Transport, msg *api.Message) error {
+	src := tr.Request.URL.Query().Get("src")
+	stream := streams.GetOrNew(src)
+	if stream == nil {
+		return errors.New(api.StreamNotFound)
+	}
+
+	cons := &mp4.Consumer{
+		RemoteAddr: tr.Request.RemoteAddr,
+		UserAgent:  tr.Request.UserAgent(),
+	}
+	cons.UserAgent = tr.Request.UserAgent()
+	cons.RemoteAddr = tr.Request.RemoteAddr
+
+	if codecs, ok := msg.Value.(string); ok {
+		log.Trace().Str("codecs", codecs).Msgf("[mp4] new WS/MSE consumer")
+		cons.Medias = parseMedias(codecs, true)
+	}
+
+	cons.Listen(func(msg interface{}) {
+		if data, ok := msg.([]byte); ok {
+			for len(data) > packetSize {
+				tr.Write(data[:packetSize])
+				data = data[packetSize:]
+			}
+			tr.Write(data)
+		}
+	})
+
+	if err := stream.AddConsumer(cons); err != nil {
+		log.Debug().Err(err).Msg("[mp4] add consumer")
+		return err
+	}
+
+	tr.OnClose(func() {
+		stream.RemoveConsumer(cons)
+	})
+
+	tr.Write(&api.Message{Type: "mse", Value: cons.MimeType()})
+
+	data, err := cons.Init()
+	if err != nil {
+		log.Warn().Err(err).Caller().Send()
+		return err
+	}
+
+	tr.Write(data)
+
+	cons.Start()
+
+	return nil
+}
+
+func handlerWSMP4(tr *api.Transport, msg *api.Message) error {
+	src := tr.Request.URL.Query().Get("src")
+	stream := streams.GetOrNew(src)
+	if stream == nil {
+		return errors.New(api.StreamNotFound)
+	}
+
+	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")
+		cons.Medias = parseMedias(codecs, false)
+	}
+
+	cons.Listen(func(msg interface{}) {
+		if data, ok := msg.([]byte); ok {
+			tr.Write(data)
+		}
+	})
+
+	if err := stream.AddConsumer(cons); err != nil {
+		log.Error().Err(err).Caller().Send()
+		return err
+	}
+
+	tr.Write(&api.Message{Type: "mp4", Value: cons.MimeType})
+
+	tr.OnClose(func() {
+		stream.RemoveConsumer(cons)
+	})
+
+	return nil
+}
+
+func parseMedias(codecs string, parseAudio bool) (medias []*streamer.Media) {
+	var videos []*streamer.Codec
+	var audios []*streamer.Codec
+
+	for _, name := range strings.Split(codecs, ",") {
+		switch name {
+		case "avc1.640029":
+			codec := &streamer.Codec{Name: streamer.CodecH264}
+			videos = append(videos, codec)
+		case "hvc1.1.6.L153.B0":
+			codec := &streamer.Codec{Name: streamer.CodecH265}
+			videos = append(videos, codec)
+		case "mp4a.40.2":
+			codec := &streamer.Codec{Name: streamer.CodecAAC}
+			audios = append(audios, codec)
+		}
+	}
+
+	if videos != nil {
+		media := &streamer.Media{
+			Kind:      streamer.KindVideo,
+			Direction: streamer.DirectionRecvonly,
+			Codecs:    videos,
+		}
+		medias = append(medias, media)
+	}
+
+	if audios != nil && parseAudio {
+		media := &streamer.Media{
+			Kind:      streamer.KindAudio,
+			Direction: streamer.DirectionRecvonly,
+			Codecs:    audios,
+		}
+		medias = append(medias, media)
+	}
+
+	return
+}
--- a/cmd/rtmp/rtmp.go
+++ b/cmd/rtmp/rtmp.go
@@ -8,8 +8,6 @@ import (

 func Init() {
 	streams.HandleFunc("rtmp", handle)
-	streams.HandleFunc("http", handle)
-	streams.HandleFunc("https", handle)
 }

 func handle(url string) (streamer.Producer, error) {
@@ -17,5 +15,8 @@ func handle(url string) (streamer.Producer, error) {
 	if err := conn.Dial(); err != nil {
 		return nil, err
 	}
+	if err := conn.Describe(); err != nil {
+		return nil, err
+	}
 	return conn, nil
 }
--- a/cmd/rtsp/rtsp.go
+++ b/cmd/rtsp/rtsp.go
@@ -14,7 +14,9 @@ import (
 func Init() {
 	var conf struct {
 		Mod struct {
-			Listen string `yaml:"listen"`
+			Listen   string `yaml:"listen" json:"listen"`
+			Username string `yaml:"username" json:"-"`
+			Password string `yaml:"password" json:"-"`
 		} `yaml:"rtsp"`
 	}

@@ -22,6 +24,7 @@ func Init() {
 	conf.Mod.Listen = ":8554"

 	app.LoadConfig(&conf)
+	app.Info["rtsp"] = conf.Mod

 	log = app.GetLogger("rtsp")

@@ -52,7 +55,13 @@ func Init() {
 			if err != nil {
 				return
 			}
-			go tcpHandler(conn)
+
+			c := rtsp.NewServer(conn)
+			// skip check auth for localhost
+			if conf.Mod.Username != "" && !conn.RemoteAddr().(*net.TCPAddr).IP.IsLoopback() {
+				c.Auth(conf.Mod.Username, conf.Mod.Password)
+			}
+			go tcpHandler(c)
 		}
 	}()
 }
@@ -121,13 +130,12 @@ func rtspHandler(url string) (streamer.Producer, error) {
 	return conn, nil
 }

-func tcpHandler(c net.Conn) {
+func tcpHandler(conn *rtsp.Conn) {
 	var name string
 	var closer func()

 	trace := log.Trace().Enabled()

-	conn := rtsp.NewServer(c)
 	conn.Listen(func(msg interface{}) {
 		if trace {
 			switch msg := msg.(type) {
@@ -140,6 +148,11 @@ func tcpHandler(c net.Conn) {

 		switch msg {
 		case rtsp.MethodDescribe:
+			if len(conn.URL.Path) == 0 {
+				log.Warn().Msg("[rtsp] server empty URL on DESCRIBE")
+				return
+			}
+
 			name = conn.URL.Path[1:]

 			stream := streams.Get(name)
@@ -149,6 +162,8 @@ func tcpHandler(c net.Conn) {

 			log.Debug().Str("stream", name).Msg("[rtsp] new consumer")

+			conn.SessionName = app.UserAgent
+
 			initMedias(conn)

 			if err := stream.AddConsumer(conn); err != nil {
@@ -161,6 +176,11 @@ func tcpHandler(c net.Conn) {
 			}

 		case rtsp.MethodAnnounce:
+			if len(conn.URL.Path) == 0 {
+				log.Warn().Msg("[rtsp] server empty URL on ANNOUNCE")
+				return
+			}
+
 			name = conn.URL.Path[1:]

 			stream := streams.Get(name)
@@ -183,6 +203,9 @@ func tcpHandler(c net.Conn) {

 	if err := conn.Accept(); err != nil {
 		log.Warn().Err(err).Caller().Send()
+		if closer != nil {
+			closer()
+		}
 		_ = conn.Close()
 		return
 	}
@@ -195,7 +218,7 @@ func tcpHandler(c net.Conn) {

 	if closer != nil {
 		if err := conn.Handle(); err != nil {
-			log.Debug().Err(err).Caller().Send()
+			log.Debug().Msgf("[rtsp] handle=%s", err)
 		}

 		closer()
--- a/cmd/streams/consumer.go
+++ b/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)
+}
--- a/cmd/streams/producer.go
+++ b/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
--- a/cmd/streams/stream.go
+++ b/cmd/streams/stream.go
@@ -3,19 +3,18 @@ package streams
 import (
 	"encoding/json"
 	"errors"
+	"fmt"
 	"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 {
@@ -50,11 +49,16 @@ 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}
 	var producers []*Producer // matched producers for consumer

+	var codecs string
+
 	// Step 1. Get consumer medias
 	for icc, consMedia := range cons.GetMedias() {
 		log.Trace().Stringer("media", consMedia).
@@ -67,6 +71,8 @@ func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
 				log.Trace().Stringer("media", prodMedia).
 					Msgf("[streams] producer=%d candidate=%d", ip, ipc)

+				collectCodecs(prodMedia, &codecs)
+
 				// Step 3. Match consumer/producer codecs list
 				prodCodec := prodMedia.MatchMedia(consMedia)
 				if prodCodec != nil {
@@ -76,7 +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
 					}

@@ -91,9 +97,22 @@ func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
 		}
 	}

-	if len(producers) == 0 {
+	if atomic.AddInt32(&s.requests, -1) == 0 {
 		s.stopProducers()
-		return errors.New("couldn't find the matching tracks")
+	}
+
+	if len(producers) == 0 {
+		if len(codecs) > 0 {
+			return errors.New("codecs not match: " + codecs)
+		}
+
+		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()
@@ -173,22 +192,21 @@ 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) {
@@ -216,3 +234,23 @@ func (s *Stream) removeProducer(i int) {
 		s.producers = append(s.producers[:i], s.producers[i+1:]...)
 	}
 }
+
+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 {
+			name = "AAC"
+		}
+		if strings.Contains(*codecs, name) {
+			continue
+		}
+		if len(*codecs) > 0 {
+			*codecs += ","
+		}
+		*codecs += name
+	}
+}
--- a/cmd/streams/streams.go
+++ b/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
--- a/cmd/webrtc/candidates.go
+++ b/cmd/webrtc/candidates.go
@@ -2,18 +2,40 @@ package webrtc

 import (
 	"github.com/AlexxIT/go2rtc/cmd/api"
-	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	"github.com/pion/sdp/v3"
 )

 var candidates []string
+var networks = []string{"udp", "tcp"}

 func AddCandidate(address string) {
 	candidates = append(candidates, address)
 }

-func addCanditates(answer string) (string, error) {
+func asyncCandidates(tr *api.Transport) {
+	for _, address := range candidates {
+		address, err := webrtc.LookupIP(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})
+		}
+	}
+}
+
+func syncCanditates(answer string) (string, error) {
 	if len(candidates) == 0 {
 		return answer, nil
 	}
@@ -38,13 +60,15 @@ func addCanditates(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 {
@@ -59,12 +83,14 @@ func addCanditates(answer string) (string, error) {
 	return string(data), nil
 }

-func candidateHandler(ctx *api.Context, msg *streamer.Message) {
-	if ctx.Consumer == nil {
-		return
+func candidateHandler(tr *api.Transport, msg *api.Message) error {
+	if tr.Consumer == nil {
+		return nil
 	}
-	if conn := ctx.Consumer.(*webrtc.Conn); conn != nil {
-		log.Trace().Str("candidate", msg.Value.(string)).Msg("[webrtc] remote")
-		conn.Push(msg)
+	if conn := tr.Consumer.(*webrtc.Conn); conn != nil {
+		s := msg.Value.(string)
+		log.Trace().Str("candidate", s).Msg("[webrtc] remote")
+		conn.AddCandidate(s)
 	}
+	return nil
 }
--- a/cmd/webrtc/webrtc.go
+++ b/cmd/webrtc/webrtc.go
@@ -1,14 +1,14 @@
 package webrtc

 import (
+	"errors"
 	"github.com/AlexxIT/go2rtc/cmd/api"
 	"github.com/AlexxIT/go2rtc/cmd/app"
 	"github.com/AlexxIT/go2rtc/cmd/streams"
-	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	pion "github.com/pion/webrtc/v3"
 	"github.com/rs/zerolog"
-	"io/ioutil"
+	"io"
 	"net"
 	"net/http"
 )
@@ -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"}},
 	}
@@ -33,7 +34,7 @@ func Init() {
 	address := cfg.Mod.Listen
 	pionAPI, err := webrtc.NewAPI(address)
 	if pionAPI == nil {
-		log.Error().Err(err).Msg("[webrtc] init API")
+		log.Error().Err(err).Caller().Msg("webrtc.NewAPI")
 		return
 	}

@@ -55,8 +56,8 @@ func Init() {

 	candidates = cfg.Mod.Candidates

-	api.HandleWS(webrtc.MsgTypeOffer, offerHandler)
-	api.HandleWS(webrtc.MsgTypeCandidate, candidateHandler)
+	api.HandleWS("webrtc/offer", asyncHandler)
+	api.HandleWS("webrtc/candidate", candidateHandler)

 	api.HandleFunc("api/webrtc", syncHandler)
 }
@@ -66,11 +67,11 @@ var log zerolog.Logger

 var NewPConn func() (*pion.PeerConnection, error)

-func offerHandler(ctx *api.Context, msg *streamer.Message) {
-	src := ctx.Request.URL.Query().Get("src")
-	stream := streams.Get(src)
+func asyncHandler(tr *api.Transport, msg *api.Message) error {
+	src := tr.Request.URL.Query().Get("src")
+	stream := streams.GetOrNew(src)
 	if stream == nil {
-		return
+		return errors.New(api.StreamNotFound)
 	}

 	log.Debug().Str("url", src).Msg("[webrtc] new consumer")
@@ -81,21 +82,23 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
 	conn := new(webrtc.Conn)
 	conn.Conn, err = NewPConn()
 	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] new conn")
-		return
+		log.Error().Err(err).Caller().Send()
+		return err
 	}

-	conn.UserAgent = ctx.Request.UserAgent()
+	conn.UserAgent = tr.Request.UserAgent()
 	conn.Listen(func(msg interface{}) {
 		switch msg := msg.(type) {
 		case pion.PeerConnectionState:
 			if msg == pion.PeerConnectionStateClosed {
 				stream.RemoveConsumer(conn)
 			}
-		case *streamer.Message:
-			// subscribe on webrtc server candidates
-			log.Trace().Str("candidate", msg.Value.(string)).Msg("[webrtc] local")
-			ctx.Write(msg)
+		case *pion.ICECandidate:
+			if msg != nil {
+				s := msg.ToJSON().Candidate
+				log.Trace().Str("candidate", s).Msg("[webrtc] local")
+				tr.Write(&api.Message{Type: "webrtc/candidate", Value: s})
+			}
 		}
 	})

@@ -104,41 +107,35 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
 	log.Trace().Msgf("[webrtc] offer:\n%s", offer)

 	if err = conn.SetOffer(offer); err != nil {
-		log.Warn().Err(err).Msg("[api.webrtc] set offer")
-		ctx.Error(err)
-		return
+		log.Warn().Err(err).Caller().Send()
+		return err
 	}

 	// 2. AddConsumer, so we get new tracks
 	if err = stream.AddConsumer(conn); err != nil {
-		log.Warn().Err(err).Msg("[api.webrtc] add consumer")
+		log.Debug().Err(err).Msg("[webrtc] add consumer")
 		_ = conn.Conn.Close()
-		ctx.Error(err)
-		return
+		return err
 	}

 	conn.Init()

-	// exchange sdp without waiting all candidates
-	//answer, err := conn.ExchangeSDP(offer, false)
-	//answer, err := conn.GetAnswer()
-	answer, err := conn.GetCompleteAnswer()
-	if err == nil {
-		answer, err = addCanditates(answer)
-	}
+	// 3. Exchange SDP without waiting all candidates
+	answer, err := conn.GetAnswer()
 	log.Trace().Msgf("[webrtc] answer\n%s", answer)

 	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] get answer")
-		ctx.Error(err)
-		return
+		log.Error().Err(err).Caller().Send()
+		return err
 	}

-	ctx.Write(&streamer.Message{
-		Type: webrtc.MsgTypeAnswer, Value: answer,
-	})
+	tr.Consumer = conn

-	ctx.Consumer = conn
+	tr.Write(&api.Message{Type: "webrtc/answer", Value: answer})
+
+	asyncCandidates(tr)
+
+	return nil
 }

 func syncHandler(w http.ResponseWriter, r *http.Request) {
@@ -149,21 +146,21 @@ func syncHandler(w http.ResponseWriter, r *http.Request) {
 	}

 	// get offer
-	offer, err := ioutil.ReadAll(r.Body)
+	offer, err := io.ReadAll(r.Body)
 	if err != nil {
-		log.Error().Err(err).Caller().Send()
+		log.Error().Err(err).Caller().Msg("ioutil.ReadAll")
 		return
 	}

 	answer, err := ExchangeSDP(stream, string(offer), r.UserAgent())
 	if err != nil {
-		log.Error().Err(err).Caller().Send()
+		log.Error().Err(err).Caller().Msg("ExchangeSDP")
 		return
 	}

 	// send SDP to client
 	if _, err = w.Write([]byte(answer)); err != nil {
-		log.Error().Err(err).Caller().Send()
+		log.Error().Err(err).Caller().Msg("w.Write")
 	}
 }

@@ -174,7 +171,7 @@ func ExchangeSDP(
 	conn := new(webrtc.Conn)
 	conn.Conn, err = NewPConn()
 	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] new conn")
+		log.Error().Err(err).Caller().Msg("NewPConn")
 		return
 	}

@@ -192,13 +189,13 @@ func ExchangeSDP(
 	log.Trace().Msgf("[webrtc] offer:\n%s", offer)

 	if err = conn.SetOffer(offer); err != nil {
-		log.Warn().Err(err).Msg("[api.webrtc] set offer")
+		log.Warn().Err(err).Caller().Msg("conn.SetOffer")
 		return
 	}

 	// 2. AddConsumer, so we get new tracks
 	if err = stream.AddConsumer(conn); err != nil {
-		log.Warn().Err(err).Msg("[api.webrtc] add consumer")
+		log.Warn().Err(err).Caller().Msg("stream.AddConsumer")
 		_ = conn.Conn.Close()
 		return
 	}
@@ -209,12 +206,12 @@ func ExchangeSDP(
 	//answer, err := conn.ExchangeSDP(offer, false)
 	answer, err = conn.GetCompleteAnswer()
 	if err == nil {
-		answer, err = addCanditates(answer)
+		answer, err = syncCanditates(answer)
 	}
 	log.Trace().Msgf("[webrtc] answer\n%s", answer)

 	if err != nil {
-		log.Error().Err(err).Msg("[webrtc] get answer")
+		log.Error().Err(err).Caller().Msg("conn.GetCompleteAnswer")
 	}

 	return
--- a/go.mod
+++ b/go.mod
@@ -1,6 +1,6 @@
 module github.com/AlexxIT/go2rtc

-go 1.17
+go 1.19

 require (
 	github.com/brutella/hap v0.0.17
@@ -26,6 +26,7 @@ require (
 	github.com/davecgh/go-spew v1.1.1 // indirect
 	github.com/go-chi/chi v1.5.4 // indirect
 	github.com/google/uuid v1.3.0 // indirect
+	github.com/kr/pretty v0.2.1 // indirect
 	github.com/mattn/go-colorable v0.1.12 // indirect
 	github.com/mattn/go-isatty v0.0.14 // indirect
 	github.com/miekg/dns v1.1.50 // indirect
@@ -41,12 +42,11 @@ require (
 	github.com/pmezard/go-difflib v1.0.0 // indirect
 	github.com/xiam/to v0.0.0-20200126224905-d60d31e03561 // indirect
 	golang.org/x/crypto v0.0.0-20220516162934-403b01795ae8 // indirect
-	golang.org/x/mod v0.4.2 // indirect
+	golang.org/x/mod v0.6.0-dev.0.20220419223038-86c51ed26bb4 // indirect
 	golang.org/x/net v0.0.0-20220630215102-69896b714898 // indirect
-	golang.org/x/sys v0.0.0-20220622161953-175b2fd9d664 // indirect
+	golang.org/x/sys v0.0.0-20220811171246-fbc7d0a398ab // indirect
 	golang.org/x/text v0.3.7 // indirect
-	golang.org/x/tools v0.1.6-0.20210726203631-07bc1bf47fb2 // indirect
-	golang.org/x/xerrors v0.0.0-20200804184101-5ec99f83aff1 // indirect
+	golang.org/x/tools v0.1.11 // indirect
 )

 replace (
--- a/go.sum
+++ b/go.sum
@@ -4,7 +4,6 @@ github.com/AlexxIT/vdk v0.0.18-0.20221108193131-6168555b4f92 h1:cIeYMGaAirSZnrKR
 github.com/AlexxIT/vdk v0.0.18-0.20221108193131-6168555b4f92/go.mod h1:7ydHfSkflMZxBXfWR79dMjrT54xzvLxnPaByOa9Jpzg=
 github.com/brutella/dnssd v1.2.3 h1:4fBLjZjPH7SbcHhEcIJhZcC9nOhIDZ0m3rn9bjl1/i0=
 github.com/brutella/dnssd v1.2.3/go.mod h1:JoW2sJUrmVIef25G6lrLj7HS6Xdwh6q8WUIvMkkBYXs=
-github.com/cheekybits/genny v1.0.0/go.mod h1:+tQajlRqAUrPI7DOSpB0XAqZYtQakVtB7wXkRAgjxjQ=
 github.com/coreos/go-systemd/v22 v22.3.3-0.20220203105225-a9a7ef127534/go.mod h1:Y58oyj3AT4RCenI/lSvhwexgC+NSVTIJ3seZv2GcEnc=
 github.com/davecgh/go-spew v1.1.0/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
 github.com/davecgh/go-spew v1.1.1 h1:vj9j/u1bqnvCEfJOwUhtlOARqs3+rkHYY13jYWTU97c=
@@ -15,7 +14,6 @@ github.com/go-chi/chi v1.5.4 h1:QHdzF2szwjqVV4wmByUnTcsbIg7UGaQ0tPF2t5GcAIs=
 github.com/go-chi/chi v1.5.4/go.mod h1:uaf8YgoFazUOkPBG7fxPftUylNumIev9awIWOENIuEg=
 github.com/go-task/slim-sprig v0.0.0-20210107165309-348f09dbbbc0/go.mod h1:fyg7847qk6SyHyPtNmDHnmrv/HOrqktSC+C9fM+CJOE=
 github.com/godbus/dbus/v5 v5.0.4/go.mod h1:xhWf0FNVPg57R7Z0UbKHbJfkEywrmjJnf7w5xrFpKfA=
-github.com/golang/mock v1.2.0/go.mod h1:oTYuIxOrZwtPieC+H1uAHpcLFnEyAGVDL/k47Jfbm0A=
 github.com/golang/protobuf v1.2.0/go.mod h1:6lQm79b+lXiMfvg/cZm0SGofjICqVBUtrP5yJMmIC1U=
 github.com/golang/protobuf v1.4.0-rc.1/go.mod h1:ceaxUfeHdC40wWswd/P6IGgMaK3YpKi5j83Wpe3EHw8=
 github.com/golang/protobuf v1.4.0-rc.1.0.20200221234624-67d41d38c208/go.mod h1:xKAWHe0F5eneWXFV3EuXVDTCmh+JuBKY0li0aMyXATA=
@@ -29,7 +27,6 @@ github.com/google/go-cmp v0.3.0/go.mod h1:8QqcDgzrUqlUb/G2PQTWiueGozuR1884gddMyw
 github.com/google/go-cmp v0.3.1/go.mod h1:8QqcDgzrUqlUb/G2PQTWiueGozuR1884gddMywk6iLU=
 github.com/google/go-cmp v0.4.0/go.mod h1:v8dTdLbMG2kIc/vJvl+f65V22dbkXbowE6jgT/gNBxE=
 github.com/google/go-cmp v0.5.5/go.mod h1:v8dTdLbMG2kIc/vJvl+f65V22dbkXbowE6jgT/gNBxE=
-github.com/google/uuid v1.1.1/go.mod h1:TIyPZe4MgqvfeYDBFedMoGGpEw/LqOeaOT+nhxU+yHo=
 github.com/google/uuid v1.3.0 h1:t6JiXgmwXMjEs8VusXIJk2BXHsn+wx8BZdTaoZ5fu7I=
 github.com/google/uuid v1.3.0/go.mod h1:TIyPZe4MgqvfeYDBFedMoGGpEw/LqOeaOT+nhxU+yHo=
 github.com/gorilla/websocket v1.5.0 h1:PPwGk2jz7EePpoHN/+ClbZu8SPxiqlu12wZP/3sWmnc=
@@ -37,13 +34,12 @@ github.com/gorilla/websocket v1.5.0/go.mod h1:YR8l580nyteQvAITg2hZ9XVh4b55+EU/ad
 github.com/hashicorp/mdns v1.0.5 h1:1M5hW1cunYeoXOqHwEb/GBDDHAFo0Yqb/uz/beC6LbE=
 github.com/hashicorp/mdns v1.0.5/go.mod h1:mtBihi+LeNXGtG8L9dX59gAEa12BDtBQSp4v/YAJqrc=
 github.com/hpcloud/tail v1.0.0/go.mod h1:ab1qPbhIpdTxEkNHXyeSf5vhxWSCs/tWer42PpOxQnU=
-github.com/kr/pretty v0.1.0 h1:L/CwN0zerZDmRFUapSPitk6f+Q3+0za1rQkzVuMiMFI=
 github.com/kr/pretty v0.1.0/go.mod h1:dAy3ld7l9f0ibDNOQOHHMYYIIbhfbHSm3C4ZsoJORNo=
+github.com/kr/pretty v0.2.1 h1:Fmg33tUaq4/8ym9TJN1x7sLJnHVwhP33CNkpYV/7rwI=
+github.com/kr/pretty v0.2.1/go.mod h1:ipq/a2n7PKx3OHsz4KJII5eveXtPO4qwEXGdVfWzfnI=
 github.com/kr/pty v1.1.1/go.mod h1:pFQYn66WHrOpPYNljwOMqo10TkYh1fy3cYio2l3bCsQ=
 github.com/kr/text v0.1.0 h1:45sCR5RtlFHMR4UwH9sdQ5TC8v0qDQCHnXt+kaKSTVE=
 github.com/kr/text v0.1.0/go.mod h1:4Jbv+DJW3UT/LiOwJeYQe1efqtUx/iVham/4vfdArNI=
-github.com/lucas-clemente/quic-go v0.7.1-0.20190401152353-907071221cf9/go.mod h1:PpMmPfPKO9nKJ/psF49ESTAGQSdfXxlg1otPbEB2nOw=
-github.com/marten-seemann/qtls v0.2.3/go.mod h1:xzjG7avBwGGbdZ8dTGxlBnLArsVKLvwmjgmPuiQEcYk=
 github.com/mattn/go-colorable v0.1.12 h1:jF+Du6AlPIjs2BiUiQlKOX0rt3SujHxPnksPKZbaA40=
 github.com/mattn/go-colorable v0.1.12/go.mod h1:u5H1YNBxpqRaxsYJYSkiCWKzEfiAb1Gb520KVy5xxl4=
 github.com/mattn/go-isatty v0.0.14 h1:yVuAays6BHfxijgZPzw+3Zlu5yQgKGP2/hcQbHb7S9Y=
@@ -54,74 +50,49 @@ github.com/miekg/dns v1.1.50/go.mod h1:e3IlAVfNqAllflbibAZEWOXOQ+Ynzk/dDozDxY7Xn
 github.com/nxadm/tail v1.4.4/go.mod h1:kenIhsEOeOJmVchQTgglprH7qJGnHDVpk1VPCcaMI8A=
 github.com/nxadm/tail v1.4.8/go.mod h1:+ncqLTQzXmGhMZNUePPaPqPvBxHAIsmXswZKocGu+AU=
 github.com/onsi/ginkgo v1.6.0/go.mod h1:lLunBs/Ym6LB5Z9jYTR76FiuTmxDTDusOGeTQH+WWjE=
-github.com/onsi/ginkgo v1.7.0/go.mod h1:lLunBs/Ym6LB5Z9jYTR76FiuTmxDTDusOGeTQH+WWjE=
 github.com/onsi/ginkgo v1.12.1/go.mod h1:zj2OWP4+oCPe1qIXoGWkgMRwljMUYCdkwsT2108oapk=
 github.com/onsi/ginkgo v1.16.4/go.mod h1:dX+/inL/fNMqNlz0e9LfyB9TswhZpCVdJM/Z6Vvnwo0=
 github.com/onsi/ginkgo v1.16.5/go.mod h1:+E8gABHa3K6zRBolWtd+ROzc/U5bkGt0FwiG042wbpU=
-github.com/onsi/gomega v1.4.3/go.mod h1:ex+gbHU/CVuBBDIJjb2X0qEXbFg53c61hWP/1CpauHY=
 github.com/onsi/gomega v1.7.1/go.mod h1:XdKZgCCFLUoM/7CFJVPcG8C1xQ1AJ0vpAezJrB7JYyY=
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@@ -132,7 +103,6 @@ github.com/rs/zerolog v1.27.0 h1:1T7qCieN22GVc8S4Q2yuexzBb1EqjbgjSH9RohbMjKs=
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@@ -145,28 +115,21 @@ github.com/xiam/to v0.0.0-20200126224905-d60d31e03561 h1:SVoNK97S6JlaYlHcaC+79tg
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@@ -190,13 +153,11 @@ golang.org/x/sync v0.0.0-20210220032951-036812b2e83c h1:5KslGYwFpkhGh+Q16bwMP3cO
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@@ -210,8 +171,9 @@ golang.org/x/sys v0.0.0-20210927094055-39ccf1dd6fa6/go.mod h1:oPkhp1MJrh7nUepCBc
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 golang.org/x/term v0.0.0-20210927222741-03fcf44c2211/go.mod h1:jbD1KX2456YbFQfuXm/mYQcufACuNUgVhRMnK/tPxf8=
 golang.org/x/text v0.3.0/go.mod h1:NqM8EUOU14njkJ3fqMW+pc6Ldnwhi/IjpwHt7yyuwOQ=
@@ -222,12 +184,12 @@ golang.org/x/text v0.3.7/go.mod h1:u+2+/6zg+i71rQMx5EYifcz6MCKuco9NR6JIITiCfzQ=
 golang.org/x/tools v0.0.0-20180917221912-90fa682c2a6e/go.mod h1:n7NCudcB/nEzxVGmLbDWY5pfWTLqBcC2KZ6jyYvM4mQ=
 golang.org/x/tools v0.0.0-20191119224855-298f0cb1881e/go.mod h1:b+2E5dAYhXwXZwtnZ6UAqBI28+e2cm9otk0dWdXHAEo=
 golang.org/x/tools v0.0.0-20201224043029-2b0845dc783e/go.mod h1:emZCQorbCU4vsT4fOWvOPXz4eW1wZW4PmDk9uLelYpA=
-golang.org/x/tools v0.1.6-0.20210726203631-07bc1bf47fb2 h1:BonxutuHCTL0rBDnZlKjpGIQFTjyUVTexFOdWkB6Fg0=
 golang.org/x/tools v0.1.6-0.20210726203631-07bc1bf47fb2/go.mod h1:o0xws9oXOQQZyjljx8fwUC0k7L1pTE6eaCbjGeHmOkk=
+golang.org/x/tools v0.1.11 h1:loJ25fNOEhSXfHrpoGj91eCUThwdNX6u24rO1xnNteY=
+golang.org/x/tools v0.1.11/go.mod h1:SgwaegtQh8clINPpECJMqnxLv9I09HLqnW3RMqW0CA4=
 golang.org/x/xerrors v0.0.0-20190717185122-a985d3407aa7/go.mod h1:I/5z698sn9Ka8TeJc9MKroUUfqBBauWjQqLJ2OPfmY0=
 golang.org/x/xerrors v0.0.0-20191011141410-1b5146add898/go.mod h1:I/5z698sn9Ka8TeJc9MKroUUfqBBauWjQqLJ2OPfmY0=
 golang.org/x/xerrors v0.0.0-20191204190536-9bdfabe68543/go.mod h1:I/5z698sn9Ka8TeJc9MKroUUfqBBauWjQqLJ2OPfmY0=
-golang.org/x/xerrors v0.0.0-20200804184101-5ec99f83aff1 h1:go1bK/D/BFZV2I8cIQd1NKEZ+0owSTG1fDTci4IqFcE=
 golang.org/x/xerrors v0.0.0-20200804184101-5ec99f83aff1/go.mod h1:I/5z698sn9Ka8TeJc9MKroUUfqBBauWjQqLJ2OPfmY0=
 google.golang.org/protobuf v0.0.0-20200109180630-ec00e32a8dfd/go.mod h1:DFci5gLYBciE7Vtevhsrf46CRTquxDuWsQurQQe4oz8=
 google.golang.org/protobuf v0.0.0-20200221191635-4d8936d0db64/go.mod h1:kwYJMbMJ01Woi6D6+Kah6886xMZcty6N08ah7+eCXa0=
@@ -242,7 +204,6 @@ gopkg.in/check.v1 v1.0.0-20190902080502-41f04d3bba15 h1:YR8cESwS4TdDjEe65xsg0ogR
 gopkg.in/check.v1 v1.0.0-20190902080502-41f04d3bba15/go.mod h1:Co6ibVJAznAaIkqp8huTwlJQCZ016jof/cbN4VW5Yz0=
 gopkg.in/fsnotify.v1 v1.4.7/go.mod h1:Tz8NjZHkW78fSQdbUxIjBTcgA1z1m8ZHf0WmKUhAMys=
 gopkg.in/tomb.v1 v1.0.0-20141024135613-dd632973f1e7/go.mod h1:dt/ZhP58zS4L8KSrWDmTeBkI65Dw0HsyUHuEVlX15mw=
-gopkg.in/yaml.v2 v2.2.1/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
 gopkg.in/yaml.v2 v2.2.2/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
 gopkg.in/yaml.v2 v2.2.4/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
 gopkg.in/yaml.v2 v2.2.8/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
--- a/hardware.Dockerfile
+++ b/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"]
--- a/main.go
+++ b/main.go
@@ -9,6 +9,7 @@ import (
 	"github.com/AlexxIT/go2rtc/cmd/ffmpeg"
 	"github.com/AlexxIT/go2rtc/cmd/hass"
 	"github.com/AlexxIT/go2rtc/cmd/homekit"
+	"github.com/AlexxIT/go2rtc/cmd/http"
 	"github.com/AlexxIT/go2rtc/cmd/ivideon"
 	"github.com/AlexxIT/go2rtc/cmd/mjpeg"
 	"github.com/AlexxIT/go2rtc/cmd/mp4"
@@ -40,6 +41,7 @@ func main() {
 	webrtc.Init()
 	mp4.Init()
 	mjpeg.Init()
+	http.Init()

 	srtp.Init()
 	homekit.Init()
--- a/pkg/fake/producer.go
+++ b/pkg/fake/producer.go
@@ -24,7 +24,7 @@ func (p *Producer) GetTrack(media *streamer.Media, codec *streamer.Codec) *strea
 		panic("you shall not pass!")
 	}

-	track := &streamer.Track{Codec: codec, Direction: media.Direction}
+	track := streamer.NewTrack(codec, media.Direction)

 	switch media.Direction {
 	case streamer.DirectionSendonly:
--- a/pkg/h264/helper.go
+++ b/pkg/h264/helper.go
@@ -3,6 +3,7 @@ package h264
 import (
 	"encoding/base64"
 	"encoding/binary"
+	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"strings"
 )
@@ -51,6 +52,16 @@ func GetProfileLevelID(fmtp string) string {
 	if fmtp == "" {
 		return ""
 	}
+
+	// some cameras has wrong profile-level-id
+	// https://github.com/AlexxIT/go2rtc/issues/155
+	if s := streamer.Between(fmtp, "sprop-parameter-sets=", ","); s != "" {
+		sps, _ := base64.StdEncoding.DecodeString(s)
+		if len(sps) >= 4 {
+			return fmt.Sprintf("%06X", sps[1:4])
+		}
+	}
+
 	return streamer.Between(fmtp, "profile-level-id=", ";")
 }

--- a/pkg/h264/rtp.go
+++ b/pkg/h264/rtp.go
@@ -36,26 +36,26 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 			}

 			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 +70,7 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 				buf = buf[:0]
 			}

-			//log.Printf("[AVC] %v, len: %d", Types(payload), len(payload))
+			//log.Printf("[AVC] %v, len: %d, ts: %10d, seq: %d", Types(payload), len(payload), packet.Timestamp, packet.SequenceNumber)

 			clone := *packet
 			clone.Version = RTPPacketVersionAVC
--- a/pkg/h265/README.md
+++ b/pkg/h265/README.md
@@ -1,3 +1,4 @@
 ## Useful links

 - https://datatracker.ietf.org/doc/html/rfc7798
+- [Add initial support for WebRTC HEVC](https://trac.webkit.org/changeset/259452/webkit)
--- a/pkg/h265/helper.go
+++ b/pkg/h265/helper.go
@@ -2,19 +2,57 @@ package h265

 import (
 	"encoding/base64"
+	"encoding/binary"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 )

 const (
-	NALUnitTypeIFrame = 19
+	NALUTypePFrame  = 1
+	NALUTypeIFrame  = 19
+	NALUTypeIFrame2 = 20
+	NALUTypeIFrame3 = 21
+	NALUTypeVPS     = 32
+	NALUTypeSPS     = 33
+	NALUTypePPS     = 34
+	NALUTypeFU      = 49
 )

-func NALUnitType(b []byte) byte {
-	return b[4] >> 1
+func NALUType(b []byte) byte {
+	return (b[4] >> 1) & 0x3F
 }

 func IsKeyframe(b []byte) bool {
-	return NALUnitType(b) == NALUnitTypeIFrame
+	for {
+		switch NALUType(b) {
+		case NALUTypePFrame:
+			return false
+		case NALUTypeIFrame, NALUTypeIFrame2, NALUTypeIFrame3:
+			return true
+		}
+
+		size := int(binary.BigEndian.Uint32(b)) + 4
+		if size < len(b) {
+			b = b[size:]
+			continue
+		} else {
+			return false
+		}
+	}
+}
+
+func Types(data []byte) []byte {
+	var types []byte
+	for {
+		types = append(types, NALUType(data))
+
+		size := 4 + int(binary.BigEndian.Uint32(data))
+		if size < len(data) {
+			data = data[size:]
+		} else {
+			break
+		}
+	}
+	return types
 }

 func GetParameterSet(fmtp string) (vps, sps, pps []byte) {
--- a/pkg/h265/rtp.go
+++ b/pkg/h265/rtp.go
@@ -1,84 +1,66 @@
 package h265

 import (
+	"encoding/binary"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"github.com/deepch/vdk/codec/h265parser"
 	"github.com/pion/rtp"
 )

 func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
-	vps, sps, pps := GetParameterSet(track.Codec.FmtpLine)
+	//vps, sps, pps := GetParameterSet(track.Codec.FmtpLine)
+	//ps := h264.EncodeAVC(vps, sps, pps)

-	var buffer []byte
+	buf := make([]byte, 0, 512*1024) // 512K
+	var nuStart int

 	return func(push streamer.WriterFunc) streamer.WriterFunc {
 		return func(packet *rtp.Packet) error {
-			nut := (packet.Payload[0] >> 1) & 0x3f
-			//fmt.Printf(
-			//	"[RTP] codec: %s, nalu: %2d, size: %6d, ts: %10d, pt: %2d, ssrc: %d, seq: %d\n",
-			//	track.Codec.Name, nut, len(packet.Payload), packet.Timestamp,
-			//	packet.PayloadType, packet.SSRC, packet.SequenceNumber,
-			//)
+			data := packet.Payload
+			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)

-			switch nut {
-			case h265parser.NAL_UNIT_UNSPECIFIED_49:
-				data := packet.Payload
+			if nuType == NALUTypeFU {
 				switch data[2] >> 6 {
 				case 2: // begin
-					buffer = []byte{
-						(data[0] & 0x81) | (data[2] & 0x3f << 1), data[1],
-					}
-					buffer = append(buffer, data[3:]...)
+					nuType = data[2] & 0x3F
+
+					// push PS data before keyframe
+					//if len(buf) == 0 && nuType >= 19 && nuType <= 21 {
+					//	buf = append(buf, ps...)
+					//}
+
+					nuStart = len(buf)
+					buf = append(buf, 0, 0, 0, 0) // NAL unit size
+					buf = append(buf, (data[0]&0x81)|(nuType<<1), data[1])
+					buf = append(buf, data[3:]...)
 					return nil
 				case 0: // continue
-					buffer = append(buffer, data[3:]...)
+					buf = append(buf, data[3:]...)
 					return nil
 				case 1: // end
-					packet.Payload = append(buffer, data[3:]...)
+					buf = append(buf, data[3:]...)
+					binary.BigEndian.PutUint32(buf[nuStart:], uint32(len(buf)-nuStart-4))
 				}
-			case h265parser.NAL_UNIT_VPS:
-				vps = packet.Payload
-				return nil
-			case h265parser.NAL_UNIT_SPS:
-				sps = packet.Payload
-				return nil
-			case h265parser.NAL_UNIT_PPS:
-				pps = packet.Payload
-				return nil
-			default:
-				//panic("not implemented")
+			} else {
+				nuStart = len(buf)
+				buf = append(buf, 0, 0, 0, 0) // NAL unit size
+				buf = append(buf, data...)
+				binary.BigEndian.PutUint32(buf[nuStart:], uint32(len(data)))
 			}

-			var clone rtp.Packet
-
-			nut = (packet.Payload[0] >> 1) & 0x3f
-			if nut >= h265parser.NAL_UNIT_CODED_SLICE_BLA_W_LP && nut <= h265parser.NAL_UNIT_CODED_SLICE_CRA {
-				clone = *packet
-				clone.Version = h264.RTPPacketVersionAVC
-				clone.Payload = h264.EncodeAVC(vps)
-				if err := push(&clone); err != nil {
-					return err
-				}
-
-				clone = *packet
-				clone.Version = h264.RTPPacketVersionAVC
-				clone.Payload = h264.EncodeAVC(sps)
-				if err := push(&clone); err != nil {
-					return err
-				}
-
-				clone = *packet
-				clone.Version = h264.RTPPacketVersionAVC
-				clone.Payload = h264.EncodeAVC(pps)
-				if err := push(&clone); err != nil {
-					return err
-				}
+			// collect all NAL Units for Access Unit
+			if !packet.Marker {
+				return nil
 			}

-			clone = *packet
+			//log.Printf("[HEVC] %v, len: %d", Types(buf), len(buf))
+
+			clone := *packet
 			clone.Version = h264.RTPPacketVersionAVC
-			clone.Payload = h264.EncodeAVC(packet.Payload)
+			clone.Payload = buf
+
+			buf = buf[:0]

 			return push(&clone)
 		}
@@ -86,68 +68,80 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 }

 // SafariPay - generate Safari friendly payload for H265
+// https://github.com/AlexxIT/Blog/issues/5
 func SafariPay(mtu uint16) streamer.WrapperFunc {
 	sequencer := rtp.NewRandomSequencer()
 	size := int(mtu - 12) // rtp.Header size

-	var buffer []byte
-
 	return func(push streamer.WriterFunc) streamer.WriterFunc {
 		return func(packet *rtp.Packet) error {
 			if packet.Version != h264.RTPPacketVersionAVC {
 				return push(packet)
 			}

-			data := packet.Payload
-			data[0] = 0
-			data[1] = 0
-			data[2] = 0
-			data[3] = 1
+			// protect original packets from modification
+			au := make([]byte, len(packet.Payload))
+			copy(au, packet.Payload)

 			var start byte

-			nut := (data[4] >> 1) & 0b111111
-			//fmt.Printf("[H265] nut: %2d, size: %6d, data: %16x\n", nut, len(data), data[4:20])
-			switch {
-			case nut >= h265parser.NAL_UNIT_VPS && nut <= h265parser.NAL_UNIT_PPS:
-				buffer = append(buffer, data...)
-				return nil
-			case nut >= h265parser.NAL_UNIT_CODED_SLICE_BLA_W_LP && nut <= h265parser.NAL_UNIT_CODED_SLICE_CRA:
-				buffer = append([]byte{3}, buffer...)
-				data = append(buffer, data...)
-				start = 1
-			default:
-				data = append([]byte{2}, data...)
-				start = 0
+			for i := 0; i < len(au); {
+				size := int(binary.BigEndian.Uint32(au[i:])) + 4
+
+				// convert AVC to Annex-B
+				au[i] = 0
+				au[i+1] = 0
+				au[i+2] = 0
+				au[i+3] = 1
+
+				switch NALUType(au[i:]) {
+				case NALUTypeIFrame, NALUTypeIFrame2, NALUTypeIFrame3:
+					start = 3
+				default:
+					if start == 0 {
+						start = 2
+					}
+				}
+
+				i += size
 			}

-			for len(data) > size {
+			// rtp.Packet payload
+			b := make([]byte, 1, size)
+			size-- // minus header byte
+
+			for au != nil {
+				b[0] = start
+
+				if start > 1 {
+					start -= 2
+				}
+
+				if len(au) > size {
+					b = append(b, au[:size]...)
+					au = au[size:]
+				} else {
+					b = append(b, au...)
+					au = nil
+				}
+
 				clone := rtp.Packet{
 					Header: rtp.Header{
 						Version:        2,
-						Marker:         false,
+						Marker:         au == nil,
 						SequenceNumber: sequencer.NextSequenceNumber(),
 						Timestamp:      packet.Timestamp,
 					},
-					Payload: data[:size],
+					Payload: b,
 				}
 				if err := push(&clone); err != nil {
 					return err
 				}

-				data = append([]byte{start}, data[size:]...)
+				b = b[:1] // clear buffer
 			}

-			clone := rtp.Packet{
-				Header: rtp.Header{
-					Version:        2,
-					Marker:         true,
-					SequenceNumber: sequencer.NextSequenceNumber(),
-					Timestamp:      packet.Timestamp,
-				},
-				Payload: data,
-			}
-			return push(&clone)
+			return nil
 		}
 	}
 }
--- a/pkg/homekit/client.go
+++ b/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 {
@@ -75,7 +77,7 @@ func (c *Client) GetTrack(media *streamer.Media, codec *streamer.Codec) *streame
 		}
 	}

-	track := &streamer.Track{Codec: codec, Direction: media.Direction}
+	track := streamer.NewTrack(codec, media.Direction)
 	c.tracks = append(c.tracks, track)
 	return track
 }
@@ -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)
+}
--- a/pkg/httpflv/amf0.go
+++ b/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()
+}
--- a/pkg/httpflv/httpflv.go
+++ b/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"
@@ -22,13 +23,17 @@ func Dial(uri string) (*Conn, error) {
 		return nil, err
 	}

+	return Accept(res)
+}
+
+func Accept(res *http.Response) (*Conn, error) {
 	c := Conn{
 		conn:   res.Body,
 		reader: bufio.NewReaderSize(res.Body, pio.RecommendBufioSize),
 		buf:    make([]byte, 256),
 	}

-	if _, err = io.ReadFull(c.reader, c.buf[:flvio.FileHeaderLength]); err != nil {
+	if _, err := io.ReadFull(c.reader, c.buf[:flvio.FileHeaderLength]); err != nil {
 		return nil, err
 	}

@@ -37,8 +42,12 @@ func Dial(uri string) (*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 {
@@ -49,29 +58,83 @@ func Dial(uri string) (*Conn, error) {
 }

 type Conn struct {
-	conn      io.ReadCloser
-	reader    *bufio.Reader
-	buf       []byte
+	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
+		return []av.CodecData{video}, nil
+	} else if audio != nil {
+		c.audioIdx = 0
+		return []av.CodecData{audio}, nil
+	}
+
+	return nil, nil
 }

 func (c *Conn) ReadPacket() (av.Packet, error) {
@@ -81,20 +144,67 @@ func (c *Conn) ReadPacket() (av.Packet, error) {
 			return av.Packet{}, err
 		}

-		if tag.Type != flvio.TAG_VIDEO || tag.AVCPacketType != flvio.AVC_NALU {
-			continue
-		}
+		switch tag.Type {
+		case flvio.TAG_VIDEO:
+			if 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
+			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 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(),
+	}
+}
--- a/pkg/ivideon/client.go
+++ b/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()
 }

@@ -165,7 +205,7 @@ func (c *Client) getTracks() error {
 					Name:        streamer.CodecH264,
 					ClockRate:   90000,
 					FmtpLine:    "profile-level-id=" + msg.CodecString[i+1:],
-					PayloadType: streamer.PayloadTypeMP4,
+					PayloadType: streamer.PayloadTypeRAW,
 				}

 				i = bytes.Index(msg.Data, []byte("avcC")) - 4
@@ -192,10 +232,7 @@ func (c *Client) getTracks() error {
 				}
 				c.medias = append(c.medias, media)

-				track := &streamer.Track{
-					Direction: streamer.DirectionSendonly,
-					Codec:     codec,
-				}
+				track := streamer.NewTrack(codec, streamer.DirectionSendonly)
 				c.tracks[msg.TrackID] = track

 			case "mp4a": // mp4a.40.2
@@ -211,13 +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
--- a/pkg/ivideon/streamer.go
+++ b/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)
+}
--- a/pkg/mjpeg/README.md
+++ b/pkg/mjpeg/README.md
@@ -2,3 +2,4 @@

 - https://www.rfc-editor.org/rfc/rfc2435
 - https://github.com/GStreamer/gst-plugins-good/blob/master/gst/rtp/gstrtpjpegdepay.c
+- https://mjpeg.sanford.io/
--- a/pkg/mjpeg/client.go
+++ b/pkg/mjpeg/client.go
@@ -0,0 +1,174 @@
+package mjpeg
+
+import (
+	"bufio"
+	"encoding/json"
+	"errors"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/AlexxIT/go2rtc/pkg/tcp"
+	"github.com/pion/rtp"
+	"io"
+	"net/http"
+	"net/textproto"
+	"strconv"
+	"strings"
+	"sync/atomic"
+	"time"
+)
+
+type Client struct {
+	streamer.Element
+
+	UserAgent  string
+	RemoteAddr string
+
+	closed bool
+	res    *http.Response
+
+	track *streamer.Track
+	recv  uint32
+}
+
+func NewClient(res *http.Response) *Client {
+	codec := &streamer.Codec{
+		Name: streamer.CodecJPEG, ClockRate: 90000, PayloadType: streamer.PayloadTypeRAW,
+	}
+	return &Client{
+		res:   res,
+		track: streamer.NewTrack(codec, streamer.DirectionSendonly),
+	}
+}
+
+func (c *Client) GetMedias() []*streamer.Media {
+	return []*streamer.Media{{
+		Kind:      streamer.KindVideo,
+		Direction: streamer.DirectionSendonly,
+		Codecs:    []*streamer.Codec{c.track.Codec},
+	}}
+}
+
+func (c *Client) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.Track {
+	return c.track
+}
+
+func (c *Client) Start() error {
+	ct := c.res.Header.Get("Content-Type")
+
+	if ct == "image/jpeg" {
+		return c.startJPEG()
+	}
+
+	// added in go1.18
+	if _, s, ok := strings.Cut(ct, "boundary="); ok {
+		return c.startMJPEG(s)
+	}
+
+	return errors.New("wrong Content-Type: " + ct)
+}
+
+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 {
+		return err
+	}
+
+	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 {
+		res, err := tcp.Do(req)
+		if err != nil {
+			return err
+		}
+
+		if res.StatusCode != http.StatusOK {
+			return errors.New("wrong status: " + res.Status)
+		}
+
+		buf, err = io.ReadAll(res.Body)
+		if err != nil {
+			return err
+		}
+
+		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
+
+	r := bufio.NewReader(c.res.Body)
+	tp := textproto.NewReader(r)
+
+	for !c.closed {
+		s, err := tp.ReadLine()
+		if err != nil {
+			return err
+		}
+		if s != boundary {
+			return errors.New("wrong boundary: " + s)
+		}
+
+		header, err := tp.ReadMIMEHeader()
+		if err != nil {
+			return err
+		}
+
+		s = header.Get("Content-Length")
+		if s == "" {
+			return errors.New("no content length")
+		}
+
+		size, err := strconv.Atoi(s)
+		if err != nil {
+			return err
+		}
+
+		buf := make([]byte, size)
+		if _, err = io.ReadFull(r, buf); err != nil {
+			return err
+		}
+
+		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
+		}
+	}
+
+	return nil
+}
+
+func now() uint32 {
+	return uint32(time.Now().UnixMilli() * 90)
+}
--- a/pkg/mjpeg/consumer.go
+++ b/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 {
@@ -26,70 +28,26 @@ func (c *Consumer) GetMedias() []*streamer.Media {
 }

 func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
-	var header, payload []byte
-
 	push := func(packet *rtp.Packet) error {
-		//fmt.Printf(
-		//	"[RTP] codec: %s, size: %6d, ts: %10d, pt: %2d, ssrc: %d, seq: %d, mark: %v\n",
-		//	track.Codec.Name, len(packet.Payload), packet.Timestamp,
-		//	packet.PayloadType, packet.SSRC, packet.SequenceNumber, packet.Marker,
-		//)
-
-		// https://www.rfc-editor.org/rfc/rfc2435#section-3.1
-		b := packet.Payload
-
-		// 3.1.  JPEG header
-		t := b[4]
-
-		// 3.1.7.  Restart Marker header
-		if 64 <= t && t <= 127 {
-			b = b[12:] // skip it
-		} else {
-			b = b[8:]
-		}
-
-		if header == nil {
-			var lqt, cqt []byte
-
-			// 3.1.8.  Quantization Table header
-			q := packet.Payload[5]
-			if q >= 128 {
-				lqt = b[4:68]
-				cqt = b[68:132]
-				b = b[132:]
-			} else {
-				lqt, cqt = MakeTables(q)
-			}
-
-			// https://www.rfc-editor.org/rfc/rfc2435#section-3.1.5
-			// The maximum width is 2040 pixels.
-			w := uint16(packet.Payload[6]) << 3
-			h := uint16(packet.Payload[7]) << 3
-
-			// fix 2560x1920 and 2560x1440
-			if w == 512 && (h == 1920 || h == 1440) {
-				w = 2560
-			}
-
-			//fmt.Printf("t: %d, q: %d, w: %d, h: %d\n", t, q, w, h)
-			header = MakeHeaders(t, w, h, lqt, cqt)
-		}
-
-		// 3.1.9.  JPEG Payload
-		payload = append(payload, b...)
-
-		if packet.Marker {
-			b = append(header, payload...)
-			if end := b[len(b)-2:]; end[0] != 0xFF && end[1] != 0xD9 {
-				b = append(b, 0xFF, 0xD9)
-			}
-			c.Fire(b)
-
-			header = nil
-			payload = nil
-		}
-
+		c.Fire(packet.Payload)
+		atomic.AddUint32(&c.send, uint32(len(packet.Payload)))
 		return nil
 	}
+
+	if track.Codec.IsRTP() {
+		wrapper := RTPDepay(track)
+		push = wrapper(push)
+	}
+
 	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)
+}
--- a/pkg/mjpeg/rfc2435.go
+++ b/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)
--- a/pkg/mjpeg/rtp.go
+++ b/pkg/mjpeg/rtp.go
@@ -0,0 +1,212 @@
+package mjpeg
+
+import (
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/rtp"
+)
+
+func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
+	buf := make([]byte, 0, 512*1024) // 512K
+
+	return func(push streamer.WriterFunc) streamer.WriterFunc {
+		return func(packet *rtp.Packet) error {
+			//log.Printf("[RTP] codec: %s, size: %6d, ts: %10d, pt: %2d, ssrc: %d, seq: %d, mark: %v", track.Codec.Name, len(packet.Payload), packet.Timestamp, packet.PayloadType, packet.SSRC, packet.SequenceNumber, packet.Marker)
+
+			// https://www.rfc-editor.org/rfc/rfc2435#section-3.1
+			b := packet.Payload
+
+			// 3.1.  JPEG header
+			t := b[4]
+
+			// 3.1.7.  Restart Marker header
+			if 64 <= t && t <= 127 {
+				b = b[12:] // skip it
+			} else {
+				b = b[8:]
+			}
+
+			if len(buf) == 0 {
+				var lqt, cqt []byte
+
+				// 3.1.8.  Quantization Table header
+				q := packet.Payload[5]
+				if q >= 128 {
+					lqt = b[4:68]
+					cqt = b[68:132]
+					b = b[132:]
+				} else {
+					lqt, cqt = MakeTables(q)
+				}
+
+				// https://www.rfc-editor.org/rfc/rfc2435#section-3.1.5
+				// The maximum width is 2040 pixels.
+				w := uint16(packet.Payload[6]) << 3
+				h := uint16(packet.Payload[7]) << 3
+
+				// fix 2560x1920 and 2560x1440
+				if w == 512 && (h == 1920 || h == 1440) {
+					w = 2560
+				}
+
+				//fmt.Printf("t: %d, q: %d, w: %d, h: %d\n", t, q, w, h)
+				buf = MakeHeaders(buf, t, w, h, lqt, cqt)
+			}
+
+			// 3.1.9.  JPEG Payload
+			buf = append(buf, b...)
+
+			if !packet.Marker {
+				return nil
+			}
+
+			if end := buf[len(buf)-2:]; end[0] != 0xFF && end[1] != 0xD9 {
+				buf = append(buf, 0xFF, 0xD9)
+			}
+
+			clone := *packet
+			clone.Payload = buf
+
+			buf = buf[:0] // clear buffer
+
+			return push(&clone)
+		}
+	}
+}
+
+func RTPPay() streamer.WrapperFunc {
+	return func(push streamer.WriterFunc) streamer.WriterFunc {
+		return func(packet *rtp.Packet) error {
+			return nil
+		}
+	}
+}
+
+//func RTPPay() streamer.WrapperFunc {
+//	const packetSize = 1436
+//
+//	sequencer := rtp.NewRandomSequencer()
+//
+//	return func(push streamer.WriterFunc) streamer.WriterFunc {
+//		return func(packet *rtp.Packet) error {
+//			// reincode image to more common form
+//			img, err := jpeg.Decode(bytes.NewReader(packet.Payload))
+//			if err != nil {
+//				return err
+//			}
+//
+//			wh := img.Bounds().Size()
+//			w := wh.X
+//			h := wh.Y
+//
+//			if w > 2040 {
+//				w = 2040
+//			} else if w&3 > 0 {
+//				w &= 3
+//			}
+//			if h > 2040 {
+//				h = 2040
+//			} else if h&3 > 0 {
+//				h &= 3
+//			}
+//
+//			if w != wh.X || h != wh.Y {
+//				x0 := (wh.X - w) / 2
+//				y0 := (wh.Y - h) / 2
+//				rect := image.Rect(x0, y0, x0+w, y0+h)
+//				img = img.(*image.YCbCr).SubImage(rect)
+//			}
+//
+//			buf := bytes.NewBuffer(nil)
+//			if err = jpeg.Encode(buf, img, nil); err != nil {
+//				return err
+//			}
+//
+//			h1 := make([]byte, 8)
+//			h1[4] = 1   // Type
+//			h1[5] = 255 // Q
+//
+//			// MBZ=0, Precision=0, Length=128
+//			h2 := make([]byte, 4, 132)
+//			h2[3] = 128
+//
+//			var jpgData []byte
+//
+//			p := buf.Bytes()
+//
+//			for jpgData == nil {
+//				// 2 bytes h1
+//				if p[0] != 0xFF {
+//					return nil
+//				}
+//
+//				size := binary.BigEndian.Uint16(p[2:]) + 2
+//
+//				// 2 bytes payload size (include 2 bytes)
+//				switch p[1] {
+//				case 0xD8: // 0. Start Of Image (size=0)
+//					p = p[2:]
+//					continue
+//				case 0xDB: // 1. Define Quantization Table (size=130)
+//					for i := uint16(4 + 1); i < size; i += 1 + 64 {
+//						h2 = append(h2, p[i:i+64]...)
+//					}
+//				case 0xC0: // 2. Start Of Frame (size=15)
+//					if p[4] != 8 {
+//						return nil
+//					}
+//					h := binary.BigEndian.Uint16(p[5:])
+//					w := binary.BigEndian.Uint16(p[7:])
+//					h1[6] = uint8(w >> 3)
+//					h1[7] = uint8(h >> 3)
+//				case 0xC4: // 3. Define Huffman Table (size=416)
+//				case 0xDA: // 4. Start Of Scan (size=10)
+//					jpgData = p[size:]
+//				}
+//
+//				p = p[size:]
+//			}
+//
+//			offset := 0
+//			p = make([]byte, 0)
+//
+//			for jpgData != nil {
+//				p = p[:0]
+//
+//				if offset > 0 {
+//					h1[1] = byte(offset >> 16)
+//					h1[2] = byte(offset >> 8)
+//					h1[3] = byte(offset)
+//					p = append(p, h1...)
+//				} else {
+//					p = append(p, h1...)
+//					p = append(p, h2...)
+//				}
+//
+//				dataLen := packetSize - len(p)
+//				if dataLen < len(jpgData) {
+//					p = append(p, jpgData[:dataLen]...)
+//					jpgData = jpgData[dataLen:]
+//					offset += dataLen
+//				} else {
+//					p = append(p, jpgData...)
+//					jpgData = nil
+//				}
+//
+//				clone := rtp.Packet{
+//					Header: rtp.Header{
+//						Version:        2,
+//						Marker:         jpgData == nil,
+//						SequenceNumber: sequencer.NextSequenceNumber(),
+//						Timestamp:      packet.Timestamp,
+//					},
+//					Payload: p,
+//				}
+//				if err := push(&clone); err != nil {
+//					return err
+//				}
+//			}
+//
+//			return nil
+//		}
+//	}
+//}
--- a/pkg/mp4/consumer.go
+++ b/pkg/mp4/consumer.go
@@ -7,22 +7,35 @@ import (
 	"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
-	start  bool
+	wait   byte

-	send int
+	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,
@@ -49,50 +62,60 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 	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.start {
-				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)
-			c.send += len(buf)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(buf)

 			return nil
 		}

 		var wrapper streamer.WrapperFunc
-		if codec.IsMP4() {
-			wrapper = h264.RepairAVC(track)
-		} else {
+		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.start {
-				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)
-			c.send += len(buf)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(buf)

 			return nil
 		}

-		if !codec.IsMP4() {
+		if codec.IsRTP() {
 			wrapper := h265.RTPDepay(track)
 			push = wrapper(push)
 		}
@@ -101,18 +124,18 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea

 	case streamer.CodecAAC:
 		push := func(packet *rtp.Packet) error {
-			if !c.start {
+			if c.wait != waitNone {
 				return nil
 			}

 			buf := c.muxer.Marshal(trackID, packet)
-			c.send += len(buf)
+			atomic.AddUint32(&c.send, uint32(len(buf)))
 			c.Fire(buf)

 			return nil
 		}

-		if !codec.IsMP4() {
+		if codec.IsRTP() {
 			wrapper := aac.RTPDepay(track)
 			push = wrapper(push)
 		}
@@ -133,18 +156,19 @@ func (c *Consumer) Init() ([]byte, error) {
 }

 func (c *Consumer) Start() {
-	c.start = true
+	if c.wait == waitInit {
+		c.wait = waitKeyframe
+	}
 }

 //

 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)
 }
--- a/pkg/mp4/keyframe.go
+++ b/pkg/mp4/keyframe.go
@@ -1,85 +0,0 @@
-package mp4
-
-import (
-	"github.com/AlexxIT/go2rtc/pkg/h264"
-	"github.com/AlexxIT/go2rtc/pkg/h265"
-	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"github.com/pion/rtp"
-)
-
-type Keyframe struct {
-	streamer.Element
-
-	MimeType string
-}
-
-func (c *Keyframe) GetMedias() []*streamer.Media {
-	return []*streamer.Media{
-		{
-			Kind:      streamer.KindVideo,
-			Direction: streamer.DirectionRecvonly,
-			Codecs: []*streamer.Codec{
-				{Name: streamer.CodecH264},
-				{Name: streamer.CodecH265},
-			},
-		},
-	}
-}
-
-func (c *Keyframe) 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:
-		push := func(packet *rtp.Packet) error {
-			if !h264.IsKeyframe(packet.Payload) {
-				return nil
-			}
-
-			buf := muxer.Marshal(0, packet)
-			c.Fire(append(init, buf...))
-
-			return nil
-		}
-
-		var wrapper streamer.WrapperFunc
-		if track.Codec.IsMP4() {
-			wrapper = h264.RepairAVC(track)
-		} else {
-			wrapper = h264.RTPDepay(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)
-			c.Fire(append(init, buf...))
-
-			return nil
-		}
-
-		if !track.Codec.IsMP4() {
-			wrapper := h265.RTPDepay(track)
-			push = wrapper(push)
-		}
-
-		return track.Bind(push)
-	}
-
-	panic("unsupported codec")
-}
--- a/pkg/mp4/muxer.go
+++ b/pkg/mp4/muxer.go
@@ -3,7 +3,6 @@ package mp4
 import (
 	"encoding/binary"
 	"encoding/hex"
-	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/h265"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
@@ -20,8 +19,6 @@ type Muxer struct {
 	fragIndex uint32
 	dts       []uint64
 	pts       []uint32
-	//data      []byte
-	//total     int
 }

 func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
@@ -36,8 +33,9 @@ func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
 		case streamer.CodecH264:
 			s += "avc1." + h264.GetProfileLevelID(codec.FmtpLine)
 		case streamer.CodecH265:
-			// +Safari +Chrome +Edge -iOS15 -Android13
-			s += "hvc1.1.6.L93.B0" // hev1.1.6.L93.B0
+			// 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"
 		}
@@ -97,7 +95,10 @@ func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
 		case streamer.CodecH265:
 			vps, sps, pps := h265.GetParameterSet(codec.FmtpLine)
 			if sps == nil {
-				return nil, fmt.Errorf("empty SPS: %#v", codec)
+				// 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)
@@ -182,10 +183,13 @@ func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
 	return append(FTYP(), data...), nil
 }

-//func (m *Muxer) Rewind() {
-//	m.dts = 0
-//	m.pts = 0
-//}
+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{
@@ -215,15 +219,16 @@ func (m *Muxer) Marshal(trackID byte, packet *rtp.Packet) []byte {
 	}

 	entry := mp4io.TrackFragRunEntry{
-		//Duration: 90000,
 		Size: uint32(len(packet.Payload)),
 	}

 	newTime := packet.Timestamp
 	if m.pts[trackID] > 0 {
-		//m.dts += uint64(newTime - m.pts)
 		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

--- a/pkg/mp4/segment.go
+++ b/pkg/mp4/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)
+}
--- a/pkg/mp4f/consumer.go
+++ b/pkg/mp4f/consumer.go
@@ -1,7 +1,6 @@
 package mp4f

 import (
-	"encoding/json"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/deepch/vdk/av"
@@ -89,7 +88,7 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
 			return nil
 		}

-		if !codec.IsMP4() {
+		if codec.IsRTP() {
 			wrapper := h264.RTPDepay(track)
 			push = wrapper(push)
 		}
@@ -146,19 +145,6 @@ func (c *Consumer) Init() ([]byte, error) {
 	return data, nil
 }

-func (c *Consumer) Start()  {
+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)
-}
--- a/pkg/rtmp/client.go
+++ b/pkg/rtmp/client.go
@@ -11,7 +11,8 @@ import (
 	"github.com/deepch/vdk/codec/h264parser"
 	"github.com/deepch/vdk/format/rtmp"
 	"github.com/pion/rtp"
-	"strings"
+	"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 {
@@ -41,16 +42,20 @@ func NewClient(uri string) *Client {
 }

 func (c *Client) Dial() (err error) {
-	if strings.HasPrefix(c.URI, "http") {
-		c.conn, err = httpflv.Dial(c.URI)
-	} else {
-		c.conn, err = rtmp.Dial(c.URI)
-	}
+	c.conn, err = rtmp.Dial(c.URI)
+	return
+}

+// Accept - convert http.Response to Client
+func Accept(res *http.Response) (*Client, error) {
+	conn, err := httpflv.Accept(res)
 	if err != nil {
-		return
+		return nil, err
 	}
+	return &Client{URI: res.Request.URL.String(), conn: conn}, nil
+}

+func (c *Client) Describe() (err error) {
 	// important to get SPS/PPS
 	streams, err := c.conn.Streams()
 	if err != nil {
@@ -73,7 +78,7 @@ func (c *Client) Dial() (err error) {
 				Name:        streamer.CodecH264,
 				ClockRate:   90000,
 				FmtpLine:    fmtp,
-				PayloadType: streamer.PayloadTypeMP4,
+				PayloadType: streamer.PayloadTypeRAW,
 			}

 			media := &streamer.Media{
@@ -83,9 +88,7 @@ func (c *Client) Dial() (err error) {
 			}
 			c.medias = append(c.medias, media)

-			track := &streamer.Track{
-				Codec: codec, Direction: media.Direction,
-			}
+			track := streamer.NewTrack(codec, media.Direction)
 			c.tracks = append(c.tracks, track)

 		case av.AAC:
@@ -98,7 +101,7 @@ func (c *Client) Dial() (err error) {
 				Channels:  uint16(cd.Config.ChannelConfig),
 				//  a=fmtp:97 streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1588
 				FmtpLine:    "streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=" + hex.EncodeToString(cd.ConfigBytes),
-				PayloadType: streamer.PayloadTypeMP4,
+				PayloadType: streamer.PayloadTypeRAW,
 			}

 			media := &streamer.Media{
@@ -108,9 +111,7 @@ func (c *Client) Dial() (err error) {
 			}
 			c.medias = append(c.medias, media)

-			track := &streamer.Track{
-				Codec: codec, Direction: media.Direction,
-			}
+			track := streamer.NewTrack(codec, media.Direction)
 			c.tracks = append(c.tracks, track)

 		default:
@@ -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)]

--- a/pkg/rtmp/streamer.go
+++ b/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)
 }
--- a/pkg/rtsp/conn.go
+++ b/pkg/rtsp/conn.go
@@ -9,6 +9,7 @@ import (
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/aac"
 	"github.com/AlexxIT/go2rtc/pkg/h264"
+	"github.com/AlexxIT/go2rtc/pkg/mjpeg"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/pion/rtcp"
@@ -19,6 +20,7 @@ import (
 	"net/url"
 	"strconv"
 	"strings"
+	"sync"
 	"time"
 )

@@ -46,11 +48,28 @@ 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 {
@@ -59,6 +78,7 @@ type Conn struct {
 	// public

 	Backchannel bool
+	SessionName string

 	Medias    []*streamer.Media
 	Session   string
@@ -71,6 +91,7 @@ type Conn struct {
 	conn     net.Conn
 	mode     Mode
 	state    State
+	stateMu  sync.Mutex
 	reader   *bufio.Reader
 	sequence int
 	uri      string
@@ -99,6 +120,11 @@ func NewServer(conn net.Conn) *Conn {
 	return c
 }

+func (c *Conn) Auth(username, password string) {
+	info := url.UserPassword(username, password)
+	c.auth = tcp.NewAuth(info)
+}
+
 func (c *Conn) parseURI() (err error) {
 	c.URL, err = url.Parse(c.uri)
 	if err != nil {
@@ -334,6 +360,13 @@ func (c *Conn) Setup() error {
 func (c *Conn) SetupMedia(
 	media *streamer.Media, codec *streamer.Codec,
 ) (*streamer.Track, error) {
+	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)
@@ -398,6 +431,12 @@ func (c *Conn) SetupMedia(
 		}
 	}

+	// in case the track has already been setup before
+	if codec == nil {
+		c.state = StateSetup
+		return nil, nil
+	}
+
 	// we send our `interleaved`, but camera can answer with another

 	// Transport: RTP/AVP/TCP;unicast;interleaved=10-11;ssrc=10117CB7
@@ -429,9 +468,7 @@ func (c *Conn) SetupMedia(
 		return nil, err
 	}

-	track := &streamer.Track{
-		Codec: codec, Direction: media.Direction,
-	}
+	track := streamer.NewTrack(codec, media.Direction)

 	switch track.Direction {
 	case streamer.DirectionSendonly:
@@ -451,12 +488,19 @@ 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) {
@@ -466,12 +510,14 @@ 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()
 }
@@ -492,6 +538,17 @@ func (c *Conn) Accept() error {

 		c.Fire(req)

+		if !c.auth.Validate(req) {
+			res := &tcp.Response{
+				Status: "401 Unauthorized",
+				Header: map[string][]string{"Www-Authenticate": {`Basic realm="go2rtc"`}},
+			}
+			if err = c.Response(res); err != nil {
+				return err
+			}
+			continue
+		}
+
 		// Receiver: OPTIONS > DESCRIBE > SETUP... > PLAY > TEARDOWN
 		// Sender: OPTIONS > ANNOUNCE > SETUP... > RECORD > TEARDOWN
 		switch req.Method {
@@ -519,9 +576,7 @@ func (c *Conn) Accept() error {
 			// TODO: fix someday...
 			c.channels = map[byte]*streamer.Track{}
 			for i, media := range c.Medias {
-				track := &streamer.Track{
-					Codec: media.Codecs[0], Direction: media.Direction,
-				}
+				track := streamer.NewTrack(media.Codecs[0], media.Direction)
 				c.tracks = append(c.tracks, track)
 				c.channels[byte(i<<1)] = track
 			}
@@ -564,7 +619,7 @@ 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
 			}
@@ -595,7 +650,16 @@ 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)
@@ -604,13 +668,31 @@ 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
@@ -717,12 +799,12 @@ func (c *Conn) Handle() (err error) {
 			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)
@@ -764,6 +846,8 @@ func (c *Conn) bindTrack(

 		size := packet.MarshalSize()

+		//log.Printf("[RTP] codec: %s, size: %6d, ts: %10d, pt: %2d, ssrc: %d, seq: %d, mark: %v", track.Codec.Name, len(packet.Payload), packet.Timestamp, packet.PayloadType, packet.SSRC, packet.SequenceNumber, packet.Marker)
+
 		data := make([]byte, 4+size)
 		data[0] = '$'
 		data[1] = channel
@@ -782,7 +866,7 @@ func (c *Conn) bindTrack(
 		return nil
 	}

-	if track.Codec.IsMP4() {
+	if !track.Codec.IsRTP() {
 		switch track.Codec.Name {
 		case streamer.CodecH264:
 			wrapper := h264.RTPPay(1500)
@@ -790,6 +874,9 @@ func (c *Conn) bindTrack(
 		case streamer.CodecAAC:
 			wrapper := aac.RTPPay(1500)
 			push = wrapper(push)
+		case streamer.CodecJPEG:
+			wrapper := mjpeg.RTPPay()
+			push = wrapper(push)
 		}
 	}

--- a/pkg/rtsp/streamer.go
+++ b/pkg/rtsp/streamer.go
@@ -4,7 +4,6 @@ import (
 	"encoding/json"
 	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
-	"strconv"
 )

 // Element Producer
@@ -20,9 +19,11 @@ func (c *Conn) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.
 		}
 	}

-	// can't setup new tracks from play state
-	if c.state == StatePlay {
-		return nil
+	// can't setup new tracks from play state - forcing a reconnection feature
+	switch c.state {
+	case StatePlay, StateHandle:
+		go c.Close()
+		return streamer.NewTrack(codec, media.Direction)
 	}

 	track, err := c.SetupMedia(media, codec)
@@ -86,40 +87,30 @@ 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 {
@@ -128,5 +119,6 @@ func (c *Conn) MarshalJSON() ([]byte, error) {
 	//		v[k] = track.MimeType()
 	//	}
 	//}
-	return json.Marshal(v)
+
+	return json.Marshal(info)
 }
--- a/pkg/shell/env.go
+++ b/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
+	})
+}
--- a/pkg/srtp/server.go
+++ b/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
 			}
--- a/pkg/srtp/session.go
+++ b/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
--- a/pkg/streamer/helpers.go
+++ b/pkg/streamer/helpers.go
@@ -4,22 +4,17 @@ import (
 	"strings"
 )

-const (
-	JSONType       = "type"
-	JSONRemoteAddr = "remote_addr"
-	JSONUserAgent  = "user_agent"
-	JSONReceive    = "receive"
-	JSONSend       = "send"
-)
-
-// Message - struct for data exchange in Web API
-type Message struct {
-	Type  string      `json:"type"`
-	Value interface{} `json:"value,omitempty"`
+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"`
 }

-// other
-
 func Between(s, sub1, sub2 string) string {
 	i := strings.Index(s, sub1)
 	if i < 0 {
--- a/pkg/streamer/media.go
+++ b/pkg/streamer/media.go
@@ -1,6 +1,7 @@
 package streamer

 import (
+	"encoding/json"
 	"fmt"
 	"github.com/pion/sdp/v3"
 	"strconv"
@@ -37,7 +38,7 @@ const (
 	CodecELD = "ELD" // AAC-ELD
 )

-const PayloadTypeMP4 byte = 255
+const PayloadTypeRAW byte = 255

 func GetKind(name string) string {
 	switch name {
@@ -70,6 +71,10 @@ 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
@@ -139,8 +144,8 @@ func (c *Codec) String() string {
 	return s
 }

-func (c *Codec) IsMP4() bool {
-	return c.PayloadType == PayloadTypeMP4
+func (c *Codec) IsRTP() bool {
+	return c.PayloadType != PayloadTypeRAW
 }

 func (c *Codec) Clone() *Codec {
@@ -178,8 +183,22 @@ func UnmarshalSDP(rawSDP []byte) ([]*Media, error) {
 	return medias, nil
 }

-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)

--- a/pkg/streamer/media_test.go
+++ b/pkg/streamer/media_test.go
@@ -0,0 +1,23 @@
+package streamer
+
+import (
+	"github.com/pion/sdp/v3"
+	"github.com/stretchr/testify/assert"
+	"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)
+}
--- a/pkg/streamer/track.go
+++ b/pkg/streamer/track.go
@@ -1,6 +1,7 @@
 package streamer

 import (
+	"encoding/json"
 	"fmt"
 	"github.com/pion/rtp"
 	"sync"
@@ -13,15 +14,28 @@ type Track struct {
 	Codec     *Codec
 	Direction string
 	sink      map[*Track]WriterFunc
-	sinkMu    sync.RWMutex
+	sinkMu    *sync.RWMutex
+}
+
+func NewTrack(codec *Codec, direction string) *Track {
+	return &Track{Codec: codec, Direction: direction, sinkMu: new(sync.RWMutex)}
 }

 func (t *Track) String() string {
 	s := t.Codec.String()
-	s += fmt.Sprintf(", sinks=%d", len(t.sink))
+	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 {
@@ -38,14 +52,12 @@ func (t *Track) Bind(w WriterFunc) *Track {
 		t.sink = map[*Track]WriterFunc{}
 	}

-	clone := &Track{
-		Codec: t.Codec, Direction: t.Direction, sink: t.sink,
-	}
-	t.sink[clone] = w
+	clone := *t
+	t.sink[&clone] = w

 	t.sinkMu.Unlock()

-	return clone
+	return &clone
 }

 func (t *Track) Unbind() {
@@ -55,7 +67,9 @@ func (t *Track) Unbind() {
 }

 func (t *Track) GetSink(from *Track) {
+	t.sinkMu.Lock()
 	t.sink = from.sink
+	t.sinkMu.Unlock()
 }

 func (t *Track) HasSink() bool {
--- a/pkg/tcp/auth.go
+++ b/pkg/tcp/auth.go
@@ -80,6 +80,24 @@ func (a *Auth) Write(req *Request) {
 	}
 }

+func (a *Auth) Validate(req *Request) bool {
+	if a == nil {
+		return true
+	}
+
+	header := req.Header.Get("Authorization")
+	if header == "" {
+		return false
+	}
+
+	if a.Method == AuthUnknown {
+		a.Method = AuthBasic
+		a.header = "Basic " + B64(a.user, a.pass)
+	}
+
+	return header == a.header
+}
+
 func Between(s, sub1, sub2 string) string {
 	i := strings.Index(s, sub1)
 	if i < 0 {
--- a/pkg/tcp/request.go
+++ b/pkg/tcp/request.go
@@ -0,0 +1,68 @@
+package tcp
+
+import (
+	"errors"
+	"fmt"
+	"net/http"
+	"strings"
+	"time"
+)
+
+// Do - http.Client with support Digest Authorization
+func Do(req *http.Request) (*http.Response, error) {
+	// need to create new client each time to reset timeout
+	client := http.Client{Timeout: time.Second * 5000}
+	res, err := client.Do(req)
+	if err != nil {
+		return nil, err
+	}
+
+	if res.StatusCode == http.StatusUnauthorized && req.URL.User != nil {
+		auth := res.Header.Get("WWW-Authenticate")
+		if !strings.HasPrefix(auth, "Digest") {
+			return nil, errors.New("unsupported auth: " + auth)
+		}
+
+		realm := Between(auth, `realm="`, `"`)
+		nonce := Between(auth, `nonce="`, `"`)
+		qop := Between(auth, `qop="`, `"`)
+
+		user := req.URL.User
+		username := user.Username()
+		password, _ := user.Password()
+		ha1 := HexMD5(username, realm, password)
+
+		uri := req.URL.RequestURI()
+		ha2 := HexMD5(req.Method, uri)
+
+		var header string
+
+		switch qop {
+		case "":
+			response := HexMD5(ha1, nonce, ha2)
+			header = fmt.Sprintf(
+				`Digest username="%s", realm="%s", nonce="%s", uri="%s", response="%s"`,
+				user, realm, nonce, uri, response,
+			)
+		case "auth":
+			nc := "00000001"
+			cnonce := "00000001" // TODO: random...
+			response := HexMD5(ha1, nonce, nc, cnonce, qop, ha2)
+			header = fmt.Sprintf(
+				`Digest username="%s", realm="%s", nonce="%s", uri="%s", qop=%s, nc=%s, cnonce="%s", response="%s"`,
+				username, realm, nonce, uri, qop, nc, cnonce, response,
+			)
+		default:
+			return nil, errors.New("unsupported qop: " + auth)
+		}
+
+		req.Header.Set("Authorization", header)
+
+		res, err = client.Do(req)
+		if err != nil {
+			return nil, err
+		}
+	}
+
+	return res, nil
+}
--- a/pkg/webrtc/api.go
+++ b/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)
 		}
--- a/pkg/webrtc/conn.go
+++ b/pkg/webrtc/conn.go
@@ -1,17 +1,8 @@
 package webrtc

 import (
-	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/webrtc/v3"
-	"sort"
-)
-
-const (
-	MsgTypeOffer         = "webrtc/offer"
-	MsgTypeOfferComplete = "webrtc/offer-complete"
-	MsgTypeAnswer        = "webrtc/answer"
-	MsgTypeCandidate     = "webrtc/candidate"
 )

 type Conn struct {
@@ -30,11 +21,7 @@ type Conn struct {

 func (c *Conn) Init() {
 	c.Conn.OnICECandidate(func(candidate *webrtc.ICECandidate) {
-		if candidate != nil {
-			c.Fire(&streamer.Message{
-				Type: MsgTypeCandidate, Value: candidate.ToJSON().Candidate,
-			})
-		}
+		c.Fire(candidate)
 	})

 	c.Conn.OnTrack(func(remote *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
@@ -59,7 +46,11 @@ func (c *Conn) Init() {
 			}
 		}

-		fmt.Printf("TODO: webrtc ontrack %+v\n", remote)
+		//fmt.Printf("TODO: webrtc ontrack %+v\n", remote)
+	})
+
+	c.Conn.OnDataChannel(func(channel *webrtc.DataChannel) {
+		c.Fire(channel)
 	})

 	// OK connection:
@@ -100,12 +91,23 @@ func (c *Conn) SetOffer(offer string) (err error) {
 		return
 	}
 	rawSDP := []byte(c.Conn.RemoteDescription().SDP)
-	c.medias, err = streamer.UnmarshalSDP(rawSDP)
+	medias, err := streamer.UnmarshalSDP(rawSDP)
+	if err != nil {
+		return
+	}

 	// sort medias, so video will always be before audio
-	sort.Slice(c.medias, func(i, j int) bool {
-		return c.medias[i].Kind == streamer.KindVideo
-	})
+	// and ignore application media from Hass default lovelace card
+	for _, media := range medias {
+		if media.Kind == streamer.KindVideo {
+			c.medias = append(c.medias, media)
+		}
+	}
+	for _, media := range medias {
+		if media.Kind == streamer.KindAudio {
+			c.medias = append(c.medias, media)
+		}
+	}

 	return
 }
--- a/pkg/webrtc/consumer.go
+++ b/pkg/webrtc/consumer.go
@@ -57,10 +57,10 @@ func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.
 			wrapper := h264.RTPPay(1200)
 			push = wrapper(push)

-			if codec.IsMP4() {
-				wrapper = h264.RepairAVC(track)
-			} else {
+			if codec.IsRTP() {
 				wrapper = h264.RTPDepay(track)
+			} else {
+				wrapper = h264.RepairAVC(track)
 			}
 			push = wrapper(push)

@@ -108,31 +108,17 @@ func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.

 //

-func (c *Conn) Push(msg interface{}) {
-	if msg := msg.(*streamer.Message); msg != nil {
-		if msg.Type == MsgTypeCandidate {
-			_ = c.Conn.AddICECandidate(webrtc.ICECandidateInit{
-				Candidate: msg.Value.(string),
-			})
-		}
-	}
+func (c *Conn) AddCandidate(candidate string) {
+	_ = c.Conn.AddICECandidate(webrtc.ICECandidateInit{Candidate: candidate})
 }

 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)
 }
--- a/pkg/webrtc/helper.go
+++ b/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
 	}
--- a/scripts/README.md
+++ b/scripts/README.md
@@ -2,6 +2,7 @@

 - UPX-3.96 pack broken bin for `linux_mipsel`
 - UPX-3.95 pack broken bin for `mac_amd64`
+- UPX pack broken bin for `mac_arm64`
 - UPX windows pack is recognised by anti-viruses as malicious
 - `aarch64` = `arm64`
 - `armv7` = `arm`
--- a/scripts/build.cmd
+++ b/scripts/build.cmd
@@ -3,45 +3,50 @@
@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 -mx9 -sdel %FILENAME% go2rtc.exe

@SET GOOS=linux
@SET GOARCH=amd64
@SET FILENAME=go2rtc_linux_amd64
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx %FILENAME%

@SET GOOS=linux
@SET GOARCH=386
@SET FILENAME=go2rtc_linux_i386
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx %FILENAME%

@SET GOOS=linux
@SET GOARCH=arm64
@SET FILENAME=go2rtc_linux_arm64
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx %FILENAME%

@SET GOOS=linux
@SET GOARCH=arm
@SET GOARM=7
@SET FILENAME=go2rtc_linux_arm
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx %FILENAME%

@SET GOOS=linux
@SET GOARCH=mipsle
@SET FILENAME=go2rtc_linux_mipsel
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.95 %FILENAME%
+go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx %FILENAME%

@SET GOOS=darwin
@SET GOARCH=amd64
-@SET FILENAME=go2rtc_mac_amd64
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+@SET FILENAME=go2rtc_mac_amd64.zip
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc

@SET GOOS=darwin
@SET GOARCH=arm64
-@SET FILENAME=go2rtc_mac_arm64
-go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
+@SET FILENAME=go2rtc_mac_arm64.zip
+go build -ldflags "-s -w" -trimpath && 7z a -mx9 -sdel %FILENAME% go2rtc
--- a/scripts/build_linux_amd64.cmd
+++ b/scripts/build_linux_amd64.cmd
@@ -1,5 +0,0 @@
-@SET GOOS=linux
-@SET GOARCH=amd64
-@cd ..
-del go2rtc
-go build -ldflags "-s -w" -trimpath
--- a/scripts/build_linux_arm.cmd
+++ b/scripts/build_linux_arm.cmd
@@ -1,5 +0,0 @@
-@SET GOOS=linux
-@SET GOARCH=arm
-@cd ..
-del go2rtc
-go build -ldflags "-s -w" -trimpath
--- a/scripts/build_linux_mips.cmd
+++ b/scripts/build_linux_mips.cmd
@@ -1,5 +0,0 @@
-@ECHO OFF
-@SET GOOS=linux
-@SET GOARCH=mipsle
-cd ..
-go build -ldflags "-s -w" -trimpath && upx-3.95 go2rtc
--- a/scripts/build_mac_amd64.cmd
+++ b/scripts/build_mac_amd64.cmd
@@ -1,5 +0,0 @@
-@SET GOOS=darwin
-@SET GOARCH=amd64
-@cd ..
-del go2rtc
-go build -ldflags "-s -w" -trimpath
--- a/scripts/build_win64.cmd
+++ b/scripts/build_win64.cmd
@@ -1,4 +0,0 @@
-@SET GOOS=windows
-@SET GOARCH=amd64
-cd ..
-go build -ldflags "-w -s" -trimpath
--- a/www/devices.html
+++ b/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 =
--- a/www/editor.html
+++ b/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>
--- a/www/homekit.html
+++ b/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 =
--- a/www/index.html
+++ b/www/index.html
@@ -8,6 +8,11 @@
    <title>go2rtc</title>

    <style>
+        body {
+            font-family: Arial, Helvetica, sans-serif;
+            background-color: white;
+        }
+
        table {
            background-color: white;
            text-align: left;
@@ -36,17 +41,39 @@
            text-align: center;
        }

+        label {
+            display: flex;
+            align-items: center;
+        }
+
        .header {
            padding: 5px 5px;
        }
+
+        .controls {
+            display: flex;
+            padding: 5px;
+        }
+
+        .controls > label {
+            margin-left: 10px;
+        }
    </style>
 </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>
 </div>
+<div class="controls">
+    <button>stream</button>
+    <label><input type="checkbox" name="webrtc" checked>webrtc</label>
+    <label><input type="checkbox" name="mse" checked>mse</label>
+    <label><input type="checkbox" name="mp4" checked>mp4</label>
+    <label><input type="checkbox" name="mjpeg" checked>mjpeg</label>
+</div>
 <table id="streams">
    <thead>
    <tr>
@@ -59,53 +86,87 @@
    </tbody>
 </table>
 <script>
-    const baseUrl = location.origin + location.pathname.substr(
-        0, location.pathname.lastIndexOf("/")
-    );
-
-    const links = [
-        '<a href="webrtc.html?src={name}">webrtc</a>',
-        '<a href="mse.html?src={name}">mse</a>',
-        // '<a href="video.html?src={name}">video</a>',
+    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/frame.mp4?src={name}">frame</a>',
-        `<a href="rtsp://${location.hostname}:8554/{name}">rtsp</a>`,
        '<a href="api/stream.mjpeg?src={name}">mjpeg</a>',
        '<a href="api/streams?src={name}">info</a>',
+        '<a href="#" data-name="{name}">delete</a>',
    ];

-    function reload() {
-        fetch(`${baseUrl}/api/streams`).then(r => {
-            r.json().then(data => {
-                let html = '';
+    document.querySelector("#add")
+        .addEventListener("click", () => {
+            const src = document.querySelector("#src");
+            const url = new URL("api/streams", location.href);
+            url.searchParams.set("src", src.value);
+            fetch(url, {method: "PUT"}).then(reload);
+        });

-                for (const [name, value] of Object.entries(data)) {
-                    const online = value !== null ? value.length : 0
-                    html += `<tr><td>${name || 'default'}</td><td>${online}</td><td>`;
-                    links.forEach(link => {
-                        html += link.replace('{name}', encodeURIComponent(name)) + ' ';
-                    })
-                    html += `<a href="#" onclick="deleteStream('${name}')">delete</a>`;
-                    html += `</td></tr>`;
-                }
+    document.querySelector(".controls > button")
+        .addEventListener("click", () => {
+            const url = new URL("stream.html", location.href);

-                let content = document.getElementById('streams').getElementsByTagName('tbody')[0];
-                content.innerHTML = html
+            const streams = document.querySelectorAll("#streams input");
+            streams.forEach(i => {
+                if (i.checked) url.searchParams.append("src", i.name);
            });
-        })
+
+            if (!url.searchParams.has("src")) return;
+
+            let mode = document.querySelectorAll(".controls input");
+            mode = Array.from(mode).filter(i => i.checked).map(i => i.name).join(",");
+
+            window.location.href = `${url}&mode=${mode}`;
+        });
+
+    const tbody = document.querySelector("#streams > tbody");
+    tbody.addEventListener("click", ev => {
+        if (ev.target.innerText !== "delete") return;
+
+        ev.preventDefault();
+
+        const url = new URL("api/streams", location.href);
+        url.searchParams.set("src", ev.target.dataset.name);
+        fetch(url, {method: "DELETE"}).then(reload);
+    });
+
+    function reload() {
+        const url = new URL("api/streams", location.href);
+        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.consumers ? value.consumers.length : 0;
+                const links = templates.map(link => {
+                    return link.replace("{name}", encodeURIComponent(name));
+                }).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>`;
+                tbody.appendChild(tr);
+            }
+        });
    }

-    function deleteStream(src) {
-        fetch(`${baseUrl}/api/streams?src=${encodeURIComponent(src)}`, {method: 'DELETE'}).then(reload);
-    }
+    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}`;

-    const addButton = document.querySelector('a#add');
-    addButton.onclick = () => {
-        let src = document.querySelector('input#src');
-        fetch(`${baseUrl}/api/streams?src=${encodeURIComponent(src.value)}`, {method: 'PUT'}).then(reload);
-    }
+        try {
+            const host = data.host.match(/^[^:]+/)[0];
+            const port = data.rtsp.listen.match(/[0-9]+$/)[0];
+            templates.splice(4, 0, `<a href="rtsp://${host}:${port}/{name}">rtsp</a>`);
+        } catch (e) {
+            templates.splice(4, 0, `<a href="rtsp://${location.host}:8554/{name}">rtsp</a>`);
+        }

-    reload();
+        reload();
+    });
 </script>
 </body>
 </html>
--- a/www/main.js
+++ b/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;
--- a/www/mse.html
+++ b/www/mse.html
@@ -1,99 +0,0 @@
-<!DOCTYPE html>
-<html lang="en">
-<head>
-    <meta name="viewport" content="width=device-width, initial-scale=1">
-    <title>go2rtc - MSE</title>
-    <style>
-        body {
-            margin: 0;
-            padding: 0;
-        }
-
-        html, body {
-            height: 100%;
-            width: 100%;
-        }
-
-        #video {
-            width: 100%;
-            height: 100%;
-            background: black;
-        }
-    </style>
-</head>
-<body>
-<!-- muted is important for autoplay -->
-<video id="video" autoplay controls playsinline muted></video>
-<script>
-    // support api_path
-    const baseUrl = location.origin + location.pathname.substr(
-        0, location.pathname.lastIndexOf("/")
-    );
-    const video = document.querySelector('#video');
-
-    function init() {
-        let mediaSource, sourceBuffer, queueBuffer = [];
-
-        const ws = new WebSocket(`ws${baseUrl.substr(4)}/api/ws${location.search}`);
-        ws.binaryType = "arraybuffer";
-
-        ws.onopen = () => {
-            mediaSource = new MediaSource();
-            video.src = URL.createObjectURL(mediaSource);
-            mediaSource.onsourceopen = () => {
-                mediaSource.onsourceopen = null;
-                URL.revokeObjectURL(video.src);
-                ws.send(JSON.stringify({"type": "mse"}));
-            };
-        };
-
-        ws.onmessage = ev => {
-            if (typeof ev.data === 'string') {
-                const data = JSON.parse(ev.data);
-                console.debug("ws.onmessage", data);
-
-                if (data.type === "mse") {
-                    sourceBuffer = mediaSource.addSourceBuffer(data.value);
-                    sourceBuffer.mode = "segments"; // segments or sequence
-                    sourceBuffer.onupdateend = () => {
-                        if (!sourceBuffer.updating && queueBuffer.length > 0) {
-                            try {
-                                sourceBuffer.appendBuffer(queueBuffer.shift());
-                            } catch (e) {
-                                // console.warn(e);
-                            }
-                        }
-                    }
-                }
-            } else if (sourceBuffer.updating || queueBuffer.length > 0) {
-                queueBuffer.push(ev.data);
-            } else {
-                try {
-                    sourceBuffer.appendBuffer(ev.data);
-                } catch (e) {
-                    // console.warn(e);
-                }
-            }
-
-            if (video.seekable.length > 0) {
-                const delay = video.seekable.end(video.seekable.length - 1) - video.currentTime;
-                if (delay < 1) {
-                    video.playbackRate = 1;
-                } else if (delay > 10) {
-                    video.playbackRate = 10;
-                } else if (delay > 2) {
-                    video.playbackRate = Math.floor(delay);
-                }
-            }
-        }
-
-        video.onpause = () => {
-            ws.close();
-            setTimeout(init, 0);
-        }
-    }
-
-    init();
-</script>
-</body>
-</html>
--- a/www/stream.html
+++ b/www/stream.html
@@ -0,0 +1,62 @@
+<!DOCTYPE html>
+<html lang="en">
+<head>
+    <meta name="viewport" content="width=device-width, initial-scale=1">
+    <title>go2rtc - Stream</title>
+    <style>
+        body {
+            background: black;
+            margin: 0;
+            padding: 0;
+            display: flex;
+            font-family: Arial, Helvetica, sans-serif;
+        }
+
+        html, body {
+            height: 100%;
+            width: 100%;
+        }
+
+        .flex {
+            flex-wrap: wrap;
+            align-content: flex-start;
+            align-items: flex-start;
+        }
+    </style>
+</head>
+<body>
+<script type="module" src="./video-stream.js"></script>
+<script type="module">
+    const params = new URLSearchParams(location.search);
+
+    // support multiple streams and multiple modes
+    const streams = params.getAll("src");
+    const modes = params.getAll("mode");
+    if (modes.length === 0) modes.push("");
+
+    while (modes.length > streams.length) {
+        streams.push(streams[0]);
+    }
+    while (streams.length > modes.length) {
+        modes.push(modes[0]);
+    }
+
+    if (streams.length > 1) {
+        document.body.className = "flex";
+    }
+
+    const background = params.get("background") !== "false";
+    const width = "1 0 " + (params.get("width") || "320px");
+
+    for (let i = 0; i < streams.length; i++) {
+        /** @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=" + encodeURIComponent(streams[i]), location.href);
+        document.body.appendChild(video);
+    }
+</script>
+</body>
+</html>
--- a/www/video-rtc.js
+++ b/www/video-rtc.js
@@ -0,0 +1,593 @@
+/**
+ * Video player for go2rtc streaming application.
+ *
+ * All modern web technologies are supported in almost any browser except Apple Safari.
+ *
+ * Support:
+ * - RTCPeerConnection for Safari iOS 11.0+
+ * - IntersectionObserver for Safari iOS 12.2+
+ *
+ * Doesn't support:
+ * - MediaSource for Safari iOS all
+ * - Customized built-in elements (extends HTMLVideoElement) because all Safari
+ * - Public class fields because old Safari (before 14.0)
+ */
+export class VideoRTC extends HTMLElement {
+    constructor() {
+        super();
+
+        this.DISCONNECT_TIMEOUT = 5000;
+        this.RECONNECT_TIMEOUT = 30000;
+
+        this.CODECS = [
+            "avc1.640029",      // H.264 high 4.1 (Chromecast 1st and 2nd Gen)
+            "avc1.64002A",      // H.264 high 4.2 (Chromecast 3rd Gen)
+            "avc1.640033",      // H.264 high 5.1 (Chromecast with Google TV)
+            "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
+        ];
+
+        /**
+         * [config] Supported modes (webrtc, mse, mp4, mjpeg).
+         * @type {string}
+         */
+        this.mode = "webrtc,mse,mp4,mjpeg";
+
+        /**
+         * [config] Run stream when not displayed on the screen. Default `false`.
+         * @type {boolean}
+         */
+        this.background = false;
+
+        /**
+         * [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.visibilityThreshold = 0;
+
+        /**
+         * [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"}]};
+
+        /**
+         * [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}
+         */
+        this.video = null;
+
+        /**
+         * @type {WebSocket}
+         */
+        this.ws = null;
+
+        /**
+         * @type {string|URL}
+         */
+        this.wsURL = "";
+
+        /**
+         * @type {RTCPeerConnection}
+         */
+        this.pc = null;
+
+        /**
+         * @type {number}
+         */
+        this.connectTS = 0;
+
+        /**
+         * @type {string}
+         */
+        this.mseCodecs = "";
+
+        /**
+         * [internal] Disconnect TimeoutID.
+         * @type {number}
+         */
+        this.disconnectTID = 0;
+
+        /**
+         * [internal] Reconnect TimeoutID.
+         * @type {number}
+         */
+        this.reconnectTID = 0;
+
+        /**
+         * [internal] Handler for receiving Binary from WebSocket.
+         * @type {Function}
+         */
+        this.ondata = null;
+
+        /**
+         * [internal] Handlers list for receiving JSON from WebSocket
+         * @type {Object.<string,Function>}}
+         */
+        this.onmessage = null;
+    }
+
+    /**
+     * Set video source (WebSocket URL). Support relative path.
+     * @param {string|URL} value
+     */
+    set src(value) {
+        if (typeof value !== "string") value = value.toString();
+        if (value.startsWith("http")) {
+            value = "ws" + value.substring(4);
+        } else if (value.startsWith("/")) {
+            value = "ws" + location.origin.substring(4) + value;
+        }
+
+        this.wsURL = value;
+
+        this.onconnect();
+    }
+
+    /**
+     * Play video. Support automute when autoplay blocked.
+     * https://developer.chrome.com/blog/autoplay/
+     */
+    play() {
+        this.video.play().catch(er => {
+            if (er.name === "NotAllowedError" && !this.video.muted) {
+                this.video.muted = true;
+                this.video.play().catch(() => console.debug);
+            }
+        });
+    }
+
+    /**
+     * Send message to server via WebSocket
+     * @param {Object} value
+     */
+    send(value) {
+        if (this.ws) this.ws.send(JSON.stringify(value));
+    }
+
+    codecs(type) {
+        const test = type === "mse"
+            ? codec => MediaSource.isTypeSupported(`video/mp4; codecs="${codec}"`)
+            : codec => this.video.canPlayType(`video/mp4; codecs="${codec}"`);
+        return this.CODECS.filter(test).join();
+    }
+
+    /**
+     * `CustomElement`. Invoked each time the custom element is appended into a
+     * document-connected element.
+     */
+    connectedCallback() {
+        if (this.disconnectTID) {
+            clearTimeout(this.disconnectTID);
+            this.disconnectTID = 0;
+        }
+
+        // because video autopause on disconnected from DOM
+        if (this.video) {
+            const seek = this.video.seekable;
+            if (seek.length > 0) {
+                this.video.currentTime = seek.end(seek.length - 1);
+                this.play();
+            }
+        } else {
+            this.oninit();
+        }
+
+        this.onconnect();
+    }
+
+    /**
+     * `CustomElement`. Invoked each time the custom element is disconnected from the
+     * document's DOM.
+     */
+    disconnectedCallback() {
+        if (this.background || this.disconnectTID) return;
+        if (this.wsState === WebSocket.CLOSED && this.pcState === WebSocket.CLOSED) return;
+
+        this.disconnectTID = setTimeout(() => {
+            if (this.reconnectTID) {
+                clearTimeout(this.reconnectTID);
+                this.reconnectTID = 0;
+            }
+
+            this.disconnectTID = 0;
+
+            this.ondisconnect();
+        }, this.DISCONNECT_TIMEOUT);
+    }
+
+    /**
+     * 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.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) {
+            document.addEventListener("visibilitychange", () => {
+                if (document.hidden) {
+                    this.disconnectedCallback();
+                } else if (this.isConnected) {
+                    this.connectedCallback();
+                }
+            })
+        }
+
+        if ("IntersectionObserver" in window && this.visibilityThreshold) {
+            const observer = new IntersectionObserver(entries => {
+                entries.forEach(entry => {
+                    if (!entry.isIntersecting) {
+                        this.disconnectedCallback();
+                    } else if (this.isConnected) {
+                        this.connectedCallback();
+                    }
+                });
+            }, {threshold: this.visibilityThreshold});
+            observer.observe(this);
+        }
+    }
+
+    /**
+     * 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;
+
+        // CLOSED or CONNECTING => CONNECTING
+        this.wsState = WebSocket.CONNECTING;
+
+        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));
+
+        return true;
+    }
+
+    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", () => {
+            URL.revokeObjectURL(this.video.src);
+            this.send({type: "mse", value: this.codecs("mse")});
+        }, {once: true});
+
+        this.video.src = URL.createObjectURL(ms);
+        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;
+
+                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) - 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);
+                }
+            });
+
+            const buf = new Uint8Array(2 * 1024 * 1024);
+            let bufLen = 0;
+
+            this.ondata = data => {
+                if (sb.updating || bufLen > 0) {
+                    const b = new Uint8Array(data);
+                    buf.set(b, bufLen);
+                    bufLen += b.byteLength;
+                    // console.debug("VideoRTC.buffer", b.byteLength, bufLen);
+                } else {
+                    try {
+                        sb.appendBuffer(data);
+                    } catch (e) {
+                        // console.debug(e);
+                    }
+                }
+            }
+        }
+    }
+
+    onwebrtc() {
+        const pc = new RTCPeerConnection(this.pcConfig);
+
+        /** @type {HTMLVideoElement} */
+        const video2 = document.createElement("video");
+        video2.addEventListener("loadeddata", ev => this.onpcvideo(ev), {once: true});
+
+        pc.addEventListener("icecandidate", ev => {
+            const candidate = ev.candidate ? ev.candidate.toJSON().candidate : "";
+            this.send({type: "webrtc/candidate", value: candidate});
+        });
+
+        pc.addEventListener("track", ev => {
+            // when stream already init
+            if (video2.srcObject !== null) return;
+
+            // when audio track not exist in Chrome
+            if (ev.streams.length === 0) return;
+
+            // when audio track not exist in Firefox
+            if (ev.streams[0].id[0] === '{') return;
+
+            video2.srcObject = ev.streams[0];
+        });
+
+        pc.addEventListener("connectionstatechange", () => {
+            if (pc.connectionState === "failed" || pc.connectionState === "disconnected") {
+                pc.close(); // stop next events
+
+                this.pcState = WebSocket.CLOSED;
+                this.pc = null;
+
+                this.onconnect();
+            }
+        });
+
+        this.onmessage["webrtc"] = msg => {
+            switch (msg.type) {
+                case "webrtc/candidate":
+                    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 "error":
+                    if (msg.value.indexOf("webrtc/offer") < 0) return;
+                    pc.close();
+            }
+        };
+
+        // Safari doesn't support "offerToReceiveVideo"
+        pc.addTransceiver("video", {direction: "recvonly"});
+        pc.addTransceiver("audio", {direction: "recvonly"});
+
+        pc.createOffer().then(offer => {
+            pc.setLocalDescription(offer).then(() => {
+                this.send({type: "webrtc/offer", value: offer.sdp});
+            });
+        });
+
+        this.pcState = WebSocket.CONNECTING;
+        this.pc = pc;
+    }
+
+    /**
+     * @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"});
+    }
+
+    onmp4() {
+        /** @type {HTMLCanvasElement} **/
+        const canvas = document.createElement("canvas");
+        /** @type {CanvasRenderingContext2D} */
+        let context;
+
+        /** @type {HTMLVideoElement} */
+        const video2 = document.createElement("video");
+        video2.autoplay = true;
+        video2.muted = true;
+
+        video2.addEventListener("loadeddata", ev => {
+            if (!context) {
+                canvas.width = video2.videoWidth;
+                canvas.height = video2.videoHeight;
+                context = canvas.getContext('2d');
+            }
+
+            context.drawImage(video2, 0, 0, canvas.width, canvas.height);
+
+            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")});
+    }
+
+    static btoa(buffer) {
+        const bytes = new Uint8Array(buffer);
+        const len = bytes.byteLength;
+        let binary = "";
+        for (let i = 0; i < len; i++) {
+            binary += String.fromCharCode(bytes[i]);
+        }
+        return window.btoa(binary);
+    }
+}
--- a/www/video-stream.js
+++ b/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);
--- a/www/video.html
+++ b/www/video.html
@@ -1,53 +0,0 @@
-<!DOCTYPE html>
-<html lang="en">
-<head>
-    <meta name="viewport" content="width=device-width, initial-scale=1">
-    <title>go2rtc - WebRTC</title>
-    <style>
-        body {
-            margin: 0;
-            padding: 0;
-        }
-
-        html, body {
-            height: 100%;
-            width: 100%;
-        }
-
-        #video {
-            /* video "container" size */
-            width: 100%;
-            height: 100%;
-            background: black;
-        }
-    </style>
-</head>
-<body>
-<video id="video" autoplay controls playsinline muted></video>
-<!--<video id="video" preload="auto" controls playsinline muted></video>-->
-<script>
-    const baseUrl = location.origin + location.pathname.substr(
-        0, location.pathname.lastIndexOf("/")
-    );
-    const video = document.getElementById('video');
-
-    video.oncanplay = ev => console.log(ev.type, ev);
-    video.onplaying = ev => console.log(ev.type, ev);
-    video.onwaiting = ev => console.log(ev.type, ev);
-    video.onseeking = ev => console.log(ev.type, ev);
-    video.onloadeddata = ev => console.log(ev.type, ev);
-    video.oncanplaythrough = ev => console.log(ev.type, ev);
-    // video.ondurationchange = ev => console.log(ev.type, ev);
-    // video.ontimeupdate = ev => console.log(ev.type, ev);
-    video.onplay = ev => console.log(ev.type, ev);
-    video.onpause = ev => console.log(ev.type, ev);
-    video.onsuspended = ev => console.log(ev.type, ev);
-    video.onemptied = ev => console.log(ev.type, ev);
-    video.onstalled = ev => console.log(ev.type, ev);
-
-    console.log("start");
-
-    video.src = baseUrl + "/api/stream.mp4" + location.search;
-</script>
-</body>
-</html>
--- a/www/webrtc.html
+++ b/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});
            }
Author	SHA1	Message	Date
Alexey Khit	9ddbb326b4	Update version to 1.0.0	2023-01-20 17:07:43 +03:00
Alexey Khit	a2e58d928e	Fix timezone in logs	2023-01-20 13:45:01 +03:00
Alexey Khit	3c48fb8bea	Simplify Dockerfile	2023-01-20 11:23:28 +03:00
Alexey Khit	4b0cbb5a73	Add support basic auth for API	2023-01-20 10:54:26 +03:00
Alexey Khit	e28b49ea86	Ignore errors for RTCP packets	2023-01-20 10:26:57 +03:00
Alexey Khit	5c17d8fcb6	Add support AAC audio for HTTP-FLV	2023-01-19 21:44:15 +03:00
Alexey Khit	e040fb591f	Disable CGO for git releases	2023-01-18 15:07:42 +03:00
Alexey Khit	140014f2a6	Fix info for WS/MP4	2023-01-18 15:04:06 +03:00
Alexey Khit	23f72d111e	Add Teardown handler for RTSP server (untested)	2023-01-18 12:21:54 +03:00
Alexey Khit	f9d5ab9d0a	Fix RTSP server SDP for some clients	2023-01-18 11:45:39 +03:00
Alexey Khit	8628c48db8	Add no-cache for all GET API requests	2023-01-18 10:01:00 +03:00
Alexey Khit	6e49d51c33	Update GET config API when config file not set	2023-01-18 10:00:20 +03:00
Alexey Khit	6a61b5234e	Fix HTTP-FLV support for Reolink cameras	2023-01-18 09:36:32 +03:00
Alexey Khit	7a0091777d	Fix relative config path #171	2023-01-16 11:00:04 +03:00
Alexey Khit	d23d2a7eff	Fix release binaries for mac	2023-01-16 00:40:02 +03:00
Alexey Khit	cecbe4166c	Update version to 0.1-rc.9	2023-01-16 00:06:55 +03:00
Alexey Khit	dcb457235c	Rewrite stream info API	2023-01-15 23:51:20 +03:00
Alexey Khit	bc4e032830	Update readme	2023-01-15 11:13:38 +03:00
Alexey Khit	8218cda149	Add version, config_path to web UI and fix RTSP link	2023-01-15 09:57:15 +03:00
Alexey Khit	d1e56feeb6	Update full path to config file	2023-01-15 09:55:32 +03:00
Alexey Khit	463d05dfd3	Update readme	2023-01-15 00:28:48 +03:00
Alexey Khit	a1a73f7b45	Rewrite WS+MP4 format to keyframes stream	2023-01-15 00:12:26 +03:00
Alexey Khit	39662e10af	Fix errors in JS player	2023-01-15 00:11:31 +03:00
Alexey Khit	1c830d6e60	Code refactoring	2023-01-14 22:49:12 +03:00
Alex X	2039aa60b3	Merge pull request #170 from skrashevich/config-api-patch-method PATH api/config method for merge configuration	2023-01-14 21:57:34 +03:00
Sergey Krashevich	b7016e798f	Update config.go	2023-01-14 21:27:23 +03:00
Alexey Khit	0b291f5185	Support multiple configs and config in raw yaml form	2023-01-14 21:12:17 +03:00
Alexey Khit	395304654a	Code refactoring	2023-01-14 19:15:13 +03:00
Alexey Khit	e472397705	Add general info API	2023-01-14 18:00:43 +03:00
Alexey Khit	7c1f48e0ad	Support empty default environment value	2023-01-14 17:25:05 +03:00
Alexey Khit	f4346a104f	Add support env variables in config file #143	2023-01-14 17:19:51 +03:00
Alexey Khit	030972b436	Auto build binaries on release #158	2023-01-14 14:14:23 +03:00
Alexey Khit	efddefa123	Add web config editor #153	2023-01-14 13:47:34 +03:00
Alexey Khit	3c1bdd0dab	Fix WebRTC candidate type	2023-01-14 09:45:03 +03:00
Alexey Khit	7e7e15d7c8	Update readme	2023-01-14 09:22:22 +03:00
Alex X	a1a9f77535	Merge pull request #167 from felipecrs/master Match docs with new webrtc udp fixed port	2023-01-14 09:10:46 +03:00
Alexey Khit	a06462729d	Code refactoring	2023-01-14 09:04:54 +03:00
Alex X	331c5bbcad	Merge pull request #166 from tsightler/udp-candidate-fix Fix invalid tcpType for UDP candidate	2023-01-14 08:59:25 +03:00
Felipe Santos	58a76efc8a	Match docs with new webrtc udp fixed port	2023-01-13 23:15:04 -03:00
tsightler	5e0f010885	Update helper.go	2023-01-13 18:18:39 -05:00
Alexey Khit	4ae733aa11	Update version to 0.1-rc.8	2023-01-13 22:39:24 +03:00
Alexey Khit	27d8b33b62	Fix concurrency in ivideon	2023-01-13 21:52:29 +03:00
Alexey Khit	ff8b0fbb9c	Set default 8555 port for WebRTC (UDP+TCP)	2023-01-13 21:51:48 +03:00
Alexey Khit	c6ad7ac39f	Add single UDP port for WebRTC Server	2023-01-13 21:51:48 +03:00
Alexey Khit	7a3adf17be	Fix mp4f consumer (unused)	2023-01-13 21:51:24 +03:00
Alexey Khit	94f6c07b28	Fix mjpeg client network connection	2023-01-13 18:03:54 +03:00
Alexey Khit	7b326d4753	Fix simultaneous stream reconnect and start	2023-01-13 18:03:17 +03:00
Alexey Khit	5407a3bc4b	Fix multiple requests from different consumers	2023-01-13 18:02:03 +03:00
Alexey Khit	6b24421722	Fix unblocking exec error	2023-01-13 18:01:01 +03:00
Alexey Khit	d12775a2d7	Fix unblocking exec waiter	2023-01-13 18:00:48 +03:00
Alexey Khit	6151593c08	Fix ws lock on write and close	2023-01-13 17:28:01 +03:00
Alexey Khit	dba0989c54	Fix empty streams json on stream lock	2023-01-13 13:37:36 +03:00
Alexey Khit	ba0c7d911d	Fix ffmpeg link to same stream	2023-01-13 13:36:43 +03:00
Alexey Khit	09fefca712	Remove backchannel codec from add consumer error	2023-01-13 13:35:58 +03:00
Alexey Khit	b3f177e2ec	Handle closed state for ws connection	2023-01-13 13:34:41 +03:00
Alexey Khit	228abb8fbe	Change logs msg from WRN to DBG for fail on add consumer	2023-01-13 13:33:55 +03:00
Alexey Khit	eee70c07b7	Fix closer for ivideon source	2023-01-13 13:32:48 +03:00
Alexey Khit	d92b0f29af	Fix states handle for RTSP	2023-01-13 13:32:09 +03:00
Alexey Khit	fca6c87b2c	Fix RTSP tracks list in info json	2023-01-13 13:31:22 +03:00
Alexey Khit	0601091772	Fix closer for RTSP server #163	2023-01-13 13:30:41 +03:00
Alexey Khit	89eb653d67	Update version to 0.1-rc.7	2023-01-08 23:18:52 +03:00
Alexey Khit	0e49ffdfff	Fix GetMedias for producer in reconnect state	2023-01-08 21:42:13 +03:00
Alexey Khit	bd2fc1252d	Update last error for reconnect stream	2023-01-08 21:36:28 +03:00
Alexey Khit	78ac88448c	Fix close problem ivideon client	2023-01-08 21:35:45 +03:00
Alexey Khit	4cd9757e53	Fix status info in JS player	2023-01-08 21:05:50 +03:00
Alexey Khit	f9cb6fd670	Fix wrong RTSP H264 profile for some cameras	2023-01-08 21:05:17 +03:00
Alexey Khit	57fa6a5530	Add support for simultaneous requests from different consumers	2023-01-08 20:31:00 +03:00
Alexey Khit	6906b56524	Fix double start for RTSP source	2023-01-08 20:01:38 +03:00
Alexey Khit	c9b0806c84	Add producer url to logs	2023-01-08 20:00:48 +03:00
Alexey Khit	a9d1e64f88	Fix STUN candidate in IPv6 format	2023-01-08 15:45:11 +03:00
Alex X	9e9f07f3f7	Merge pull request #150 from skrashevich/dockerfile-crossbuild Speedup container building using Golang cross-building	2023-01-06 14:06:50 +03:00
Sergey Krashevich	b51aabd3d9	Update Dockerfile	2023-01-06 11:52:09 +03:00
Alexey Khit	368562c540	Update version to 0.1-rc.6	2023-01-02 20:53:04 +03:00
Alexey Khit	6d6e7010b4	Rewrite JS player for better integration	2023-01-02 16:33:00 +03:00
Alexey Khit	4157a53dd8	Response with error on codec negotiation	2023-01-02 16:32:08 +03:00
Alexey Khit	bdf5654c01	Change WS default buffer	2023-01-02 16:31:11 +03:00
Alexey Khit	66f729aa0e	Send WS response on MJPEG or MP4 stream starts	2023-01-02 16:30:54 +03:00
Alexey Khit	96d1ef2d2c	Adds about RTSPtoWebRTC STUN server to readme	2022-12-27 15:59:08 +03:00
Alexey Khit	9739f7f416	Add auto create new stream for async webrtc	2022-12-25 11:17:14 +03:00
Alexey Khit	654fa32b3a	Fix packet size for MSE	2022-12-25 11:09:19 +03:00
Alexey Khit	db2263c7fe	Fix stream page for raw urls in src	2022-12-25 08:55:58 +03:00
Alexey Khit	e6c36f1cf7	Rename Hardware Dockerfile	2022-12-19 12:16:38 +03:00
Alexey Khit	110f90cb34	Disable JS stream background by default	2022-12-18 22:39:52 +03:00
Alexey Khit	aca3bab238	Fix Firefox WebRTC support	2022-12-18 22:38:44 +03:00
Alexey Khit	4df44645d7	Fix lags for Intel HW transcoding	2022-12-18 21:33:38 +03:00
Alexey Khit	097fdfbbb8	Rename HW engine for Raspberry	2022-12-18 10:25:04 +03:00
Alexey Khit	dc21a04da7	Fix freezing with VAAPI HW	2022-12-18 10:24:30 +03:00
Alexey Khit	db255b476a	Add hardware acceleration support to FFmpeg	2022-12-18 01:02:12 +03:00
Alexey Khit	464ea417ef	Add docker image with Hardware drivers	2022-12-18 01:00:39 +03:00
Alexey Khit	c1fac66329	Refactoring CI	2022-12-17 23:40:56 +03:00
Alex X	a6057a2eca	Merge pull request #72 from felipecrs/refactor-docker Refactor docker image and ci	2022-12-17 23:07:55 +03:00
Alexey Khit	7c69ba13b0	Fix RTP H264 with two SEI in packet	2022-12-17 22:59:06 +03:00
Alexey Khit	2b8bfe8bd9	Add support width and height params for FFmpeg	2022-12-09 22:07:57 +03:00
Alexey Khit	0bd54da456	Increase compression level for 7zip	2022-12-09 00:25:38 +03:00
Alexey Khit	9f6af1c9e4	Update connection method in JS player so it can be extended	2022-12-09 00:24:29 +03:00
Alexey Khit	c9dd0e37e4	Fix RTSP JPEG processing	2022-12-09 00:22:25 +03:00
Felipe Santos	562872beb8	Add jq	2022-12-06 10:38:18 -03:00
Felipe Santos	46a278c067	Add curl and exit on error run.sh	2022-12-06 10:38:18 -03:00
Felipe Santos	270fc7c1b6	Allow to run container without mounting /config	2022-12-06 10:38:18 -03:00
Felipe Santos	6feb635522	Delete config.yaml not used anymore	2022-12-06 10:38:18 -03:00
Felipe Santos	6f48131e4d	Remove armv6 which was never supported	2022-12-06 10:38:18 -03:00
Felipe Santos	f120db71a3	Add all platforms	2022-12-06 10:38:18 -03:00
Felipe Santos	72823af9d0	Refactor docker workflow	2022-12-06 10:38:18 -03:00
Felipe Santos	15d9d4ebf4	Use official python as base and add tini	2022-12-06 10:38:18 -03:00
Felipe Santos	b09bbd79c4	Fix VERSION	2022-12-06 10:38:18 -03:00
Felipe Santos	1830273f02	Refactor docker image	2022-12-06 10:38:18 -03:00
Alexey Khit	07f3972794	Update readme	2022-12-06 16:17:28 +03:00
Alexey Khit	4c2ebd20bc	Update version to 0.1-rc.5	2022-12-06 12:19:29 +03:00
Alexey Khit	440c7bd6e1	Recover link to 2-way audio on Web UI	2022-12-06 12:15:22 +03:00
Alexey Khit	74c3510a10	Trace to log supported MP4 codecs	2022-12-06 10:51:42 +03:00
Alexey Khit	c2748fc77b	Add check H264 High@5.1 for JS player	2022-12-06 10:48:42 +03:00
Alexey Khit	d334551591	Update main Web UI page	2022-12-06 01:34:24 +03:00
Alexey Khit	cfe20925ac	Big rewrite JS player WebSocket processing	2022-12-05 23:54:40 +03:00
Alexey Khit	5b39f78ace	Update error msg for fail codecs negotiation	2022-12-05 23:52:28 +03:00
Alexey Khit	b965c191b7	Adds errors output to API	2022-12-05 20:03:26 +03:00
Alexey Khit	7057b4846f	Code refactoring	2022-12-05 00:47:46 +03:00
Alexey Khit	a746b96adc	Remove old video.html file	2022-12-04 23:24:44 +03:00
Alexey Khit	b7718b33b8	Rewrite WS transport handler	2022-12-04 23:24:20 +03:00
Alexey Khit	69b17230f3	Collect producers codecs during negotiation	2022-12-04 23:16:34 +03:00
Alexey Khit	e2ecd909ab	Update stream HTML page	2022-12-04 22:38:44 +03:00
Alexey Khit	ea79da0d53	Rename modes to mode in JS player	2022-12-04 22:38:21 +03:00
Alexey Khit	e64919838c	Update MP4 over WebSocket support	2022-12-04 22:37:15 +03:00
Alexey Khit	162b11213d	Allow JS player URL to http	2022-12-04 20:09:46 +03:00
Alexey Khit	d27acbd7e3	Fix MSE buffering	2022-12-04 18:38:15 +03:00
Alexey Khit	a692ecd7c1	Fix H265 for WebRTC in Safari	2022-12-03 11:44:26 +03:00
Alexey Khit	98c5366ba9	Fix supported codecs check in JS player	2022-12-03 09:39:45 +03:00
Alexey Khit	3eaaa3fcfa	Add support URL as JS player src	2022-12-03 09:39:23 +03:00
Alexey Khit	7409b32836	Fix support old iOS version	2022-12-02 23:12:37 +03:00
Alexey Khit	e2cfdf8419	Support WebRTC, MSE, MSE2, MP4, MJPEG in JS player	2022-12-02 21:37:17 +03:00
Alexey Khit	4c0929d854	Support MP4 over WebSocket	2022-12-02 21:33:51 +03:00
Alexey Khit	258a0ffb91	Support MJPEG over WebSocket	2022-12-02 21:31:41 +03:00
Alexey Khit	999e81c2dd	Code refactoring for webrtc candidates	2022-12-01 23:41:12 +03:00
Alexey Khit	8c6729027b	Add feature to SETUP new RTSP tracks after PLAY	2022-12-01 23:37:11 +03:00
Alexey Khit	d3bd5eeab5	Added a blank payloader for MJPEG RTSP	2022-12-01 23:20:31 +03:00
Alexey Khit	dbbf2ea310	Update readme about Dahua bug	2022-12-01 14:43:13 +03:00
Alexey Khit	b8234e0c76	Update codecs table in readme	2022-12-01 13:46:59 +03:00
Alexey Khit	96cd753e27	Adds about RTSP server password to readme	2022-12-01 13:35:46 +03:00
Alexey Khit	c522e5bb08	Update docs about new sources	2022-12-01 13:02:29 +03:00
Alexey Khit	a16d8acc30	Add support HTTP JPEG and MJPEG sources	2022-12-01 13:01:48 +03:00
Alexey Khit	684878b4b1	Skip check RTSP auth for localhost requests	2022-11-29 21:04:38 +03:00
Alexey Khit	140a742cee	Fix MSE2 check in JS	2022-11-27 10:22:58 +03:00
Alexey Khit	1b518b94fd	Add support auth for RTSP server	2022-11-27 10:08:37 +03:00
Alexey Khit	5678121c50	Fix build for mac arm64	2022-11-27 09:57:27 +03:00
Alexey Khit	4915f12bde	Add build for win arm64	2022-11-27 09:57:04 +03:00
Alexey Khit	bb91240b95	Remove unnecessary build scripts	2022-11-27 09:54:46 +03:00
Alexey Khit	b1d5d53832	Update to go 1.19	2022-11-27 09:53:11 +03:00
Alexey Khit	31fbbf91bb	Remove websocket error on disconnect	2022-11-25 10:04:51 +03:00
Alexey Khit	a3f72fbab9	Fix websocket origin check with wrong port	2022-11-25 10:04:13 +03:00
Alexey Khit	fae59c7992	Update version to 0.1-rc.4	2022-11-24 02:00:31 +03:00
Alexey Khit	aff34f1d21	Totally rewritten MSE player	2022-11-24 01:59:48 +03:00
Alexey Khit	65e7efa775	Support codecs negotiation for MSE	2022-11-23 21:45:10 +03:00
Alexey Khit	3c3e9d282b	Update about H265 support in readme	2022-11-23 20:35:11 +03:00
Alexey Khit	bd51069086	Add support CORS for API	2022-11-23 20:34:06 +03:00
Alexey Khit	1ddf7f1a6c	Change trace log for stream.mp4	2022-11-23 12:57:11 +03:00
Alexey Khit	0e281e36d3	Fix race (concurency) for Track	2022-11-22 20:03:36 +03:00
Alexey Khit	3d6472cfb1	Update H265 preset for FFmpeg	2022-11-22 17:22:54 +03:00
Alexey Khit	7c31fa2ffd	Fix empty SPS for MSE H265	2022-11-22 17:22:26 +03:00
Alexey Khit	0ed9d2410a	Fix H265 support for MSE in Safari	2022-11-22 17:21:58 +03:00
Alexey Khit	1c89e7945e	Remove printf for webrtc ontrack	2022-11-18 09:13:24 +03:00
Alexey Khit	48635ae341	Add two locks for Track	2022-11-18 09:12:48 +03:00
Alexey Khit	fdb316910f	Fix WebRTC async connection	2022-11-16 11:26:56 +03:00
Alexey Khit	e29f2594fa	Fix multiple transcoding when track not exists	2022-11-15 16:16:22 +03:00
Alexey Khit	c3da7584b0	Add check on RTSP server requers with empty url path	2022-11-14 19:06:43 +03:00
Alexey Khit	1e247cba92	Igroner app media for WebRTC from Hass	2022-11-14 17:26:59 +03:00
Alexey Khit	01631d9eb0	Update readme	2022-11-14 14:57:43 +03:00