Fix busy RTSP backchannel

README.md

@@ -2,19 +2,19 @@
 
 **go2rtc** - ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
 
-- zero-dependency and zero-config small [app for all OS](#installation) (Windows, macOS, Linux, ARM)
+- zero-dependency and zero-config small [app for all OS](#go2rtc-binary) (Windows, macOS, Linux, ARM)
 - zero-delay for all supported protocols (lowest possible streaming latency)
-- zero-load on CPU for supported codecs
-- on the fly transcoding for unsupported codecs [via FFmpeg](#source-ffmpeg)
+- low CPU load for supported codecs
+- on the fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
 - multi-source 2-way [codecs negotiation](#codecs-negotiation)
-- streaming from private networks via [Ngrok or SSH-tunnels](#module-webrtc)
+- streaming from private networks via [Ngrok](#module-webrtc)
 
 **Inspired by:**
 
 - [webrtc](https://github.com/pion/webrtc) go library and whole [@pion](https://github.com/pion) team
 - series of streaming projects from [@deepch](https://github.com/deepch)
 - [rtsp-simple-server](https://github.com/aler9/rtsp-simple-server) idea from [@aler9](https://github.com/aler9)
-- [GStreamer](https://gstreamer.freedesktop.org/) multimedia framework pipeline idea
+- [GStreamer](https://gstreamer.freedesktop.org/) framework pipeline idea
 - [MediaSoup](https://mediasoup.org/) framework routing idea
 
 ## Codecs negotiation
@@ -45,7 +45,24 @@ streams:
 
 ![](codecs.svg)
 
-## Installation
+## Fast start
+
+1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+2. Open web interface [http://localhost:1984/](http://localhost:1984/)
+
+**Optionally:**
+
+- add your [streams](#module-streams) to [config](#configuration) file
+- setup [external access](#module-webrtc) to webrtc
+- setup [external access](#module-ngrok) to web interface
+- install [ffmpeg](#source-ffmpeg) for transcoding
+
+**Developers:**
+
+- write your own [web interface](#module-api)
+- integrate [web api](#module-api) into your smart home platform
+
+### go2rtc: Binary
 
 Download binary for your OS from [latest release](https://github.com/AlexxIT/go2rtc/releases/):
 
@@ -59,7 +76,24 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
 - `go2rtc_mac_amd64` - Mac with Intel
 - `go2rtc_mac_arm64` - Mac with M1
 
-Don't forget to fix the rights `chmod +x go2rtc_linux_xxx` on Linux and Mac.
+Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
+
+### go2rtc: Home Assistant Add-on
+
+[![](https://my.home-assistant.io/badges/supervisor_addon.svg)](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons)
+
+1. Install Add-On:
+    - Settings > Add-ons > Plus > Repositories > Add `https://github.com/AlexxIT/hassio-addons`
+    - go2rtc > Install > Start
+2. Setup [Integration](#module-hass)
+
+**Optionally:**
+
+- create `go2rtc.yaml` in your Home Assistant [config](https://www.home-assistant.io/docs/configuration) folder
+
+### go2rtc: Docker
+
+Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support `amd64`, `386`, `arm64`, `arm`. This container same as [Home Assistant Add-on](#go2rtc-home-assistant-add-on), but can be used separately from the Home Assistant. Container has preinstalled [FFmpeg](#source-ffmpeg) and [Ngrok](#module-ngrok) applications.
 
 ## Configuration
 
@@ -69,14 +103,14 @@ Create file `go2rtc.yaml` next to the app.
 - `api` server will start on default **1984 port**
 - `rtsp` server will start on default **8554 port**
 - `webrtc` will use random UDP port for each connection
-- `ffmpeg` will use default transcoding options (you need to install it [manually](https://ffmpeg.org/))
+- `ffmpeg` will use default transcoding options (you may install it [manually](https://ffmpeg.org/))
 
 Available modules:
 
 - [streams](#module-streams)
 - [api](#module-api) - HTTP API (important for WebRTC support)
 - [rtsp](#module-rtsp) - RTSP Server (important for FFmpeg support)
-- [webrtc](#module-webrtc) - WebRTC Server (important for external access)
+- [webrtc](#module-webrtc) - WebRTC Server
 - [ngrok](#module-ngrok) - Ngrok integration (external access for private network)
 - [ffmpeg](#source-ffmpeg) - FFmpeg integration
 - [hass](#module-hass) - Home Assistant integration
@@ -84,7 +118,7 @@ Available modules:
 
 ### Module: Streams
 
-**go2rtc** support different stream source types. You can config only one link as stream source or multiple.
+**go2rtc** support different stream source types. You can config one or multiple link as stream source.
 
 Available source types:
 
@@ -94,6 +128,8 @@ Available source types:
 - [exec](#source-exec) - advanced FFmpeg and GStreamer integration
 - [hass](#source-hass) - Home Assistant integration
 
+**PS.** You can use sources like `MJPEG`, `HLS` and others via FFmpeg integration.
+
 #### Source: RTSP
 
 - Support **RTSP and RTSPS** links with multiple video and audio tracks
@@ -130,7 +166,7 @@ streams:
 
 You can get any stream or file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.
 
-Format: `ffmpeg:{input}#{params}`. Examples:
+Format: `ffmpeg:{input}#{param1}#{param2}#{param3}`. Examples:
 
 ```yaml
 streams:
@@ -141,7 +177,7 @@ streams:
   file2: ffmpeg:~/media/BigBuckBunny.mp4#video=h264
 
   # [FILE] video will be copied, audio will be transcoded to pcmu
-  file3: ffmpeg:~/media/BigBuckBunny.mp4#video=copy&audio=pcmu
+  file3: ffmpeg:~/media/BigBuckBunny.mp4#video=copy#audio=pcmu
 
   # [HLS] video will be copied, audio will be skipped
   hls: ffmpeg:https://devstreaming-cdn.apple.com/videos/streaming/examples/bipbop_16x9/gear5/prog_index.m3u8#video=copy
@@ -150,7 +186,7 @@ streams:
   mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg?stream=half&fps=15#video=h264
 
   # [RTSP] video and audio will be copied
-  rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=copy&audio=copy
+  rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=copy#audio=copy
 ```
 
 All trascoding formats has built-in templates. But you can override them via YAML config. You can also add your own formats to config and use them with source params.
@@ -202,7 +238,7 @@ streams:
 
 ### Module: API
 
-The HTTP API is the main part for interacting with the application.
+The HTTP API is the main part for interacting with the application. Default address: `http://127.0.0.1:1984/`.
 
 - you can use WebRTC only when HTTP API enabled
 - you can disable HTTP API with `listen: ""` and use, for example, only RTSP client/server protocol
@@ -212,11 +248,15 @@ The HTTP API is the main part for interacting with the application.
 
 ```yaml
 api:
-  listen: ":1984"    # HTTP API port ("" - disabled)
-  base_path: ""      # API prefix for serve on suburl
-  static_dir: "www"  # folder for static files ("" - disabled)
+  listen: ":1984"  # HTTP API port ("" - disabled)
+  base_path: ""    # API prefix for serve on suburl
+  static_dir: ""   # folder for static files (custom web interface)
 ```
 
+**PS. go2rtc** don't provide HTTPS or password protection. Use [Nginx](https://nginx.org/) or [Ngrok](#module-ngrok) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on) for this tasks.
+
+**PS2.** You can access microphone (for 2-way audio) only with HTTPS
+
 ### Module: RTSP
 
 You can get any stream as RTSP-stream with codecs filter:
@@ -236,9 +276,9 @@ rtsp:
 
 ### Module: WebRTC
 
-WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of internet do you have.
+WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of Internet do you have.
 
-- by default, WebRTC use two random UDP ports for each connection (for video and audio)
+- by default, WebRTC use two random UDP ports for each connection (video and audio)
 - you can enable one additional TCP port for all connections and use it for external access
 
 **Static public IP**
@@ -353,14 +393,24 @@ tunnels:
 
 ### Module: Hass
 
-go2rtc compatible with Home Assistant [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration API.
+**go2rtc** compatible with Home Assistant [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration.
 
-- add integration with link to go2rtc HTTP API:
-  - Hass > Settings > Integrations > Add Integration > RTSPtoWebRTC > `http://192.168.1.123:1984/`
-- add generic camera with RTSP link:
-  - Hass > Settings > Integrations > Add Integration > Generic Camera > `rtsp://...`
-- use Picture Entity or Picture Glance lovelace card
-- open full screen card - this is should be WebRTC stream
+If you install **go2rtc** as [Hass Add-on](#go2rtc-home-assistant-add-on) - you need to use localhost IP-address, example:
+
+- `http://127.0.0.1:1984/` to web interface
+- `rtsp://127.0.0.1:8554/camera1` to RTSP streams
+
+In other cases you need to use IP-address of server with **go2rtc** application.
+
+1. Add integration with link to go2rtc HTTP API:
+   - Hass > Settings > Integrations > Add Integration > [RTSPtoWebRTC](https://my.home-assistant.io/redirect/config_flow_start/?domain=rtsp_to_webrtc) > `http://127.0.0.1:1984/`
+2. Add generic camera with RTSP link:
+   - Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://...` or `rtmp://...`
+3. Use Picture Entity or Picture Glance lovelace card
+   - you can use either direct RTSP links to cameras or take RTSP streams from **go2rtc**
+4. Open full screen card - this is should be WebRTC stream
+
+PS. Default Home Assistant lovelace cards don't support 2-way audio. You can use 2-way audio from [Add-on Web UI](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons). But you need use HTTPS to access the microphone. This is a browser restriction and cannot be avoided.
 
 ### Module: Log
 

build/hassio/Dockerfile

@@ -16,4 +16,7 @@ RUN if [ "${BUILD_ARCH}" = "aarch64" ]; then BUILD_ARCH="arm64"; \
     && curl $(curl -s "https://raw.githubusercontent.com/ngrok/docker-ngrok/main/releases.json" | jq -r ".${BUILD_ARCH}.url") -o ngrok.zip \
     && unzip ngrok
 
-CMD [ "/app/go2rtc", "-config", "/config/go2rtc.yaml" ]
+COPY run.sh /
+RUN chmod a+x /run.sh
+
+CMD [ "/run.sh" ]

build/hassio/run.sh

@@ -0,0 +1,13 @@
+#!/usr/bin/with-contenv bashio
+
+set +e
+
+while true; do
+    if [ -x /config/go2rtc ]; then
+        /config/go2rtc -config /config/go2rtc.yaml
+    else
+        /app/go2rtc -config /config/go2rtc.yaml
+    fi
+
+    sleep 5
+done

cmd/api/api.go

@@ -37,8 +37,7 @@ func Init() {
 
 	HandleFunc("/api/frame.mp4", frameHandler)
 	HandleFunc("/api/frame.raw", frameHandler)
-	HandleFunc("/api/stack", stackHandler)
-	HandleFunc("/api/stats", statsHandler)
+	HandleFunc("/api/streams", streamsHandler)
 	HandleFunc("/api/ws", apiWS)
 
 	// ensure we can listen without errors
@@ -69,16 +68,30 @@ var basePath string
 var log zerolog.Logger
 var wsHandlers = make(map[string]WSHandler)
 
-func statsHandler(w http.ResponseWriter, _ *http.Request) {
-	v := map[string]interface{}{
-		"streams": streams.All(),
+func streamsHandler(w http.ResponseWriter, r *http.Request) {
+	src := r.URL.Query().Get("src")
+
+	switch r.Method {
+	case "PUT":
+		streams.Get(src)
+		return
+	case "DELETE":
+		streams.Delete(src)
+		return
+	}
+
+	var v interface{}
+	if src != "" {
+		v = streams.Get(src)
+	} else {
+		v = streams.All()
 	}
 	data, err := json.Marshal(v)
 	if err != nil {
-		log.Error().Err(err).Msg("[api.stats] marshal")
+		log.Error().Err(err).Msg("[api.streams] marshal")
 	}
 	if _, err = w.Write(data); err != nil {
-		log.Error().Err(err).Msg("[api.stats] write")
+		log.Error().Err(err).Msg("[api.streams] write")
 	}
 }
 

cmd/api/static.go

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

cmd/debug/debug.go

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

cmd/debug/stack.go

@@ -1,4 +1,4 @@
-package api
+package debug
 
 import (
 	"bytes"

cmd/exec/exec.go

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

cmd/ffmpeg/ffmpeg.go

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

cmd/rtsp/rtsp.go

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

cmd/streams/handlers.go

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

cmd/streams/producer.go

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

cmd/streams/stream.go

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

cmd/streams/stream_test.go

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

cmd/streams/streams.go

@@ -34,6 +34,10 @@ func Get(name string) *Stream {
 	return nil
 }
 
+func Delete(name string) {
+	delete(streams, name)
+}
+
 func All() map[string]interface{} {
 	all := map[string]interface{}{}
 	for name, stream := range streams {

cmd/webrtc/webrtc.go

@@ -108,6 +108,7 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
 	// 2. AddConsumer, so we get new tracks
 	if err = stream.AddConsumer(conn); err != nil {
 		log.Warn().Err(err).Msg("[api.webrtc] add consumer")
+		_ = conn.Conn.Close()
 		ctx.Error(err)
 		return
 	}

main.go

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

pkg/rtmp/client.go

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

pkg/rtsp/conn.go

@@ -43,11 +43,15 @@ const (
 	ModeServerConsumer
 )
 
+const KeepAlive = time.Second * 25
+
 type Conn struct {
 	streamer.Element
 
 	// public
 
+	Backchannel bool
+
 	Medias    []*streamer.Media
 	Session   string
 	UserAgent string
@@ -104,6 +108,9 @@ func (c *Conn) Dial() (err error) {
 	//if c.state != StateClientInit {
 	//	panic("wrong state")
 	//}
+	if c.conn != nil && c.auth != nil {
+		c.auth.Reset()
+	}
 
 	c.conn, err = net.DialTimeout(
 		"tcp", c.URL.Host, 10*time.Second,
@@ -144,7 +151,9 @@ func (c *Conn) Request(req *tcp.Request) error {
 	}
 
 	c.sequence++
-	req.Header.Set("CSeq", strconv.Itoa(c.sequence))
+	// important to send case sensitive CSeq
+	// https://github.com/AlexxIT/go2rtc/issues/7
+	req.Header["CSeq"] = []string{strconv.Itoa(c.sequence)}
 
 	c.auth.Write(req)
 
@@ -189,7 +198,7 @@ func (c *Conn) Do(req *tcp.Request) (*tcp.Response, error) {
 	}
 
 	if res.StatusCode != http.StatusOK {
-		return nil, fmt.Errorf("wrong response on %s", req.Method)
+		return res, fmt.Errorf("wrong response on %s", req.Method)
 	}
 
 	return res, nil
@@ -254,23 +263,27 @@ func (c *Conn) Describe() error {
 		Method: MethodDescribe,
 		URL:    c.URL,
 		Header: map[string][]string{
-			"Accept":  {"application/sdp"},
-			"Require": {"www.onvif.org/ver20/backchannel"},
+			"Accept": {"application/sdp"},
 		},
 	}
 
+	if c.Backchannel {
+		req.Header.Set("Require", "www.onvif.org/ver20/backchannel")
+	}
+
 	res, err := c.Do(req)
 	if err != nil {
 		return err
 	}
 
-	// fix bug in Sonoff camera SDP "o=- 1 1 IN IP4 rom t_rtsplin"
-	// TODO: make some universal fix
-	if i := bytes.Index(res.Body, []byte("rom t_rtsplin")); i > 0 {
-		res.Body[i+3] = '_'
+	if val := res.Header.Get("Content-Base"); val != "" {
+		c.URL, err = url.Parse(val)
+		if err != nil {
+			return err
+		}
 	}
 
-	c.Medias, err = streamer.UnmarshalRTSPSDP(res.Body)
+	c.Medias, err = UnmarshalSDP(res.Body)
 	if err != nil {
 		return err
 	}
@@ -355,10 +368,23 @@ func (c *Conn) SetupMedia(
 	// we send our `interleaved`, but camera can answer with another
 
 	// Transport: RTP/AVP/TCP;unicast;interleaved=10-11;ssrc=10117CB7
+	// Transport: RTP/AVP/TCP;unicast;destination=192.168.1.123;source=192.168.10.12;interleaved=0
+	// Transport: RTP/AVP/TCP;ssrc=22345682;interleaved=0-1
 	s := res.Header.Get("Transport")
-	s, ok1, ok2 := between(s, "RTP/AVP/TCP;unicast;interleaved=", "-")
-	if !ok1 || !ok2 {
-		panic("wrong response")
+	// TODO: rewrite
+	if !strings.HasPrefix(s, "RTP/AVP/TCP;") {
+		return nil, fmt.Errorf("wrong transport: %s", s)
+	}
+
+	i := strings.Index(s, "interleaved=")
+	if i < 0 {
+		return nil, fmt.Errorf("wrong transport: %s", s)
+	}
+
+	s = s[i+len("interleaved="):]
+	i = strings.IndexAny(s, "-;")
+	if i > 0 {
+		s = s[:i]
 	}
 
 	ch, err = strconv.Atoi(s)
@@ -449,7 +475,7 @@ func (c *Conn) Accept() error {
 				return errors.New("wrong content type")
 			}
 
-			c.Medias, err = streamer.UnmarshalRTSPSDP(req.Body)
+			c.Medias, err = UnmarshalSDP(req.Body)
 			if err != nil {
 				return err
 			}
@@ -549,6 +575,7 @@ func (c *Conn) Handle() (err error) {
 	}()
 
 	//c.Fire(streamer.StatePlaying)
+	ts := time.Now().Add(KeepAlive)
 
 	for {
 		// we can read:
@@ -603,7 +630,7 @@ func (c *Conn) Handle() (err error) {
 		if channelID&1 == 0 {
 			packet := &rtp.Packet{}
 			if err = packet.Unmarshal(buf); err != nil {
-				return errors.New("wrong RTP data")
+				return
 			}
 
 			track := c.channels[channelID]
@@ -617,16 +644,27 @@ func (c *Conn) Handle() (err error) {
 			msg := &RTCP{Channel: channelID}
 
 			if err = msg.Header.Unmarshal(buf); err != nil {
-				return errors.New("wrong RTCP data")
+				return
 			}
 
 			msg.Packets, err = rtcp.Unmarshal(buf)
 			if err != nil {
-				return errors.New("wrong RTCP data")
+				return
 			}
 
 			c.Fire(msg)
 		}
+
+		// keep-alive
+		now := time.Now()
+		if now.After(ts) {
+			req := &tcp.Request{Method: MethodOptions, URL: c.URL}
+			// don't need to wait respose on this request
+			if err = c.Request(req); err != nil {
+				return err
+			}
+			ts = now.Add(KeepAlive)
+		}
 	}
 }
 
@@ -686,17 +724,35 @@ type RTCP struct {
 	Packets []rtcp.Packet
 }
 
-func between(s, sub1, sub2 string) (res string, ok1 bool, ok2 bool) {
-	i := strings.Index(s, sub1)
-	if i >= 0 {
-		ok1 = true
-		s = s[i+len(sub1):]
+const sdpHeader = `v=0
+o=- 0 0 IN IP4 0.0.0.0
+s=-
+t=0 0`
+
+func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
+	medias, err := streamer.UnmarshalSDP(rawSDP)
+	if err != nil {
+		// fix SDP header for some cameras
+		i := bytes.Index(rawSDP, []byte("\nm="))
+		if i > 0 {
+			rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
+			medias, err = streamer.UnmarshalSDP(rawSDP)
+		}
+		if err != nil {
+			return nil, err
+		}
 	}
 
-	i = strings.Index(s, sub2)
-	if i >= 0 {
-		return s[:i], ok1, true
+	// fix bug in ONVIF spec
+	// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
+	for _, media := range medias {
+		switch media.Direction {
+		case streamer.DirectionRecvonly, "":
+			media.Direction = streamer.DirectionSendonly
+		case streamer.DirectionSendonly:
+			media.Direction = streamer.DirectionRecvonly
+		}
 	}
 
-	return s, ok1, false
+	return medias, nil
 }

pkg/streamer/media.go

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

pkg/streamer/track.go

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

pkg/tcp/auth.go

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

pkg/tcp/textproto.go

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

pkg/webrtc/conn.go

@@ -1,6 +1,7 @@
 package webrtc
 
 import (
+	"fmt"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/webrtc/v3"
 )
@@ -57,7 +58,8 @@ func (c *Conn) Init() {
 			}
 		}
 
-		panic("something wrong")
+		fmt.Printf("TODO: webrtc ontrack %+v\n", remote)
+		fmt.Printf("TODO: webrtc ontrack %#v\n", remote)
 	})
 
 	c.Conn.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
@@ -128,7 +130,9 @@ func (c *Conn) GetCompleteAnswer() (answer string, err error) {
 func (c *Conn) remote() string {
 	for _, trans := range c.Conn.GetTransceivers() {
 		pair, _ := trans.Receiver().Transport().ICETransport().GetSelectedCandidatePair()
-		return pair.Remote.String()
+		if pair.Remote != nil {
+			return pair.Remote.String()
+		}
 	}
 	return ""
 }

scripts/build_linux_amd64.cmd

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

scripts/build_win64.cmd

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

www/README.md

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

www/index.html

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