Fix empty remote for webrtc

README.md

@@ -5,9 +5,9 @@
 - zero-dependency and zero-config small [app for all OS](#installation) (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)
+- 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:**
 
@@ -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/):
 
@@ -61,6 +78,23 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
 
 Don't forget to fix the rights `chmod +x go2rtc_linux_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
 
 Create file `go2rtc.yaml` next to the app.
@@ -76,7 +110,7 @@ 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
@@ -202,7 +236,7 @@ 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 +246,15 @@ 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:
@@ -353,14 +391,25 @@ 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
+4. Open full screen card - this is should be WebRTC stream
+
+- you can use either direct RTSP links to cameras or take RTSP streams from **go2rtc**
+
+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
 

cmd/streams/stream.go

@@ -61,6 +61,10 @@ 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)
@@ -121,7 +125,7 @@ func (s *Stream) RemoveProducer(prod streamer.Producer) {
 }
 
 func (s *Stream) Active() bool {
-	if len(s.consumers) > 0{
+	if len(s.consumers) > 0 {
 		return true
 	}
 

pkg/rtsp/conn.go

@@ -189,7 +189,7 @@ 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
@@ -261,7 +261,21 @@ func (c *Conn) Describe() error {
 
 	res, err := c.Do(req)
 	if err != nil {
-		return err
+		if res != nil {
+			// if we have answer - give second chanse without onvif header
+			req.Header.Del("Require")
+			res, err = c.Do(req)
+		}
+		if err != nil {
+			return err
+		}
+	}
+
+	if val := res.Header.Get("Content-Base"); val != "" {
+		c.URL, err = url.Parse(val)
+		if err != nil {
+			return err
+		}
 	}
 
 	// fix bug in Sonoff camera SDP "o=- 1 1 IN IP4 rom t_rtsplin"
@@ -355,10 +369,22 @@ 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
 	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;unicast") {
+		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)

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/webrtc/conn.go

@@ -130,7 +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 ""
 }