Update version to 1.0.1

.github/workflows/release.yml

@@ -2,9 +2,9 @@ name: release
 
 on:
   workflow_dispatch:
-  push:
-    tags:
-      - 'v*'
+#  push:
+#    tags:
+#      - 'v*'
 
 jobs:
   build-and-release:

README.md

@@ -27,6 +27,40 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
 - [MediaSoup](https://mediasoup.org/) framework routing idea
 - HomeKit Accessory Protocol from [@brutella](https://github.com/brutella/hap)
 
+---
+
+* [Fast start](#fast-start)
+  * [go2rtc: Binary](#go2rtc-binary)
+  * [go2rtc: Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+  * [go2rtc: Docker](#go2rtc-docker)
+* [Configuration](#configuration)
+  * [Module: Streams](#module-streams)
+    * [Source: RTSP](#source-rtsp)
+    * [Source: RTMP](#source-rtmp)
+    * [Source: HTTP](#source-http)
+    * [Source: FFmpeg](#source-ffmpeg)
+    * [Source: FFmpeg Device](#source-ffmpeg-device)
+    * [Source: Exec](#source-exec)
+    * [Source: Echo](#source-echo)
+    * [Source: HomeKit](#source-homekit)
+    * [Source: Ivideon](#source-ivideon)
+    * [Source: Hass](#source-hass)
+  * [Module: API](#module-api)
+  * [Module: RTSP](#module-rtsp)
+  * [Module: WebRTC](#module-webrtc)
+  * [Module: Ngrok](#module-ngrok)
+  * [Module: Hass](#module-hass)
+    * [From go2rtc to Hass](#from-go2rtc-to-hass)
+    * [From Hass to go2rtc](#from-hass-to-go2rtc)
+  * [Module: MP4](#module-mp4)
+  * [Module: MJPEG](#module-mjpeg)
+  * [Module: Log](#module-log)
+* [Security](#security)
+* [Codecs madness](#codecs-madness)
+* [Codecs negotiation](#codecs-negotiation)
+* [TIPS](#tips)
+* [FAQ](#faq)
+
 ## Fast start
 
 1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
@@ -36,7 +70,6 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
 
 - add your [streams](#module-streams) to [config](#configuration) file
 - setup [external access](#module-webrtc) to webrtc
-- setup [external access](#module-ngrok) to web interface
 
 **Developers:**
 
@@ -74,14 +107,14 @@ Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support
 
 ## Configuration
 
-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
+- by default go2rtc will search `go2rtc.yaml` in the current work dirrectory
 - `api` server will start on default **1984 port** (TCP)
 - `rtsp` server will start on default **8554 port** (TCP)
 - `webrtc` will use port **8555** (TCP/UDP) for connections
 - `ffmpeg` will use default transcoding options
 
+Configuration options and a complete list of settings can be found in [the wiki](https://github.com/AlexxIT/go2rtc/wiki/Configuration).
+
 Available modules:
 
 - [streams](#module-streams)
@@ -95,8 +128,6 @@ Available modules:
 - [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.
@@ -359,6 +390,8 @@ go2rtc has simple HTML page (`stream.html`) with support params in URL:
 ```yaml
 api:
   listen: ":1984"    # default ":1984", HTTP API port ("" - disabled)
+  username: "admin"  # default "", Basic auth for WebUI
+  password: "pass"   # default "", Basic auth for WebUI
   base_path: "/rtc"  # default "", API prefix for serve on suburl (/api => /rtc/api)
   static_dir: "www"  # default "", folder for static files (custom web interface)
   origin: "*"        # default "", allow CORS requests (only * supported)
@@ -366,7 +399,7 @@ api:
 
 **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
+- go2rtc doesn't provide HTTPS. Use [Nginx](https://nginx.org/) or [Ngrok](#module-ngrok) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on) for this tasks
 - you can access microphone (for 2-way audio) only with HTTPS ([read more](https://stackoverflow.com/questions/52759992/how-to-access-camera-and-microphone-in-chrome-without-https))
 - MJPEG over WebSocket plays better than native MJPEG because Chrome [bug](https://bugs.chromium.org/p/chromium/issues/detail?id=527446)
 - MP4 over WebSocket was created only for Apple iOS because it doesn't support MSE and native MP4
@@ -384,9 +417,9 @@ Password protection always disabled for localhost calls (ex. FFmpeg or Hass on s
 
 ```yaml
 rtsp:
-  listen: ":8554"  # RTSP Server TCP port, default - 8554
-  username: admin  # optional, default - disabled
-  password: pass   # optional, default - disabled
+  listen: ":8554"    # RTSP Server TCP port, default - 8554
+  username: "admin"  # optional, default - disabled
+  password: "pass"   # optional, default - disabled
 ```
 
 ### Module: WebRTC
@@ -399,7 +432,7 @@ WebRTC usually works without problems in the local network. But external access
 
 ```yaml
 webrtc:
-  listen: ":8555" # address of your local server and port (TCP/UDP)
+  listen: ":8555"  # address of your local server and port (TCP/UDP)
 ```
 
 **Static public IP**

cmd/api/ws.go

@@ -81,7 +81,9 @@ func apiWS(w http.ResponseWriter, r *http.Request) {
 	for {
 		msg := new(Message)
 		if err = ws.ReadJSON(msg); err != nil {
-			log.Trace().Err(err).Caller().Send()
+			if !websocket.IsCloseError(err, websocket.CloseNoStatusReceived) {
+				log.Trace().Err(err).Caller().Send()
+			}
 			_ = ws.Close()
 			break
 		}

cmd/app/app.go

@@ -14,7 +14,7 @@ import (
 	"time"
 )
 
-var Version = "1.0.0"
+var Version = "1.0.1"
 var UserAgent = "go2rtc/" + Version
 
 var ConfigPath string

pkg/h264/rtp.go

@@ -1,6 +1,7 @@
 package h264
 
 import (
+	"bytes"
 	"encoding/binary"
 	"github.com/AlexxIT/go2rtc/pkg/streamer"
 	"github.com/pion/rtp"
@@ -27,7 +28,8 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 			}
 
 			// Fix TP-Link Tapo TC70: sends SPS and PPS with packet.Marker = true
-			if packet.Marker {
+			// Reolink Duo 2: sends SPS with Marker and PPS without
+			if packet.Marker && len(payload) < 128 {
 				switch NALUType(payload) {
 				case NALUTypeSPS, NALUTypePPS:
 					buf = append(buf, payload...)
@@ -68,6 +70,27 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
 			if len(buf) > 0 {
 				payload = append(buf, payload...)
 				buf = buf[:0]
+			} else {
+				// some Chinese buggy cameras has single packet with SPS+PPS+IFrame separated by 00 00 00 01
+				// https://github.com/AlexxIT/WebRTC/issues/391
+				// https://github.com/AlexxIT/WebRTC/issues/392
+				for i := 0; i < len(payload); {
+					if i+4 >= len(payload) {
+						break
+					}
+
+					size := bytes.Index(payload[i+4:], []byte{0, 0, 0, 1})
+					if size < 0 {
+						if i == 0 {
+							break
+						}
+						size = len(payload) - (i + 4)
+					}
+
+					binary.BigEndian.PutUint32(payload[i:], uint32(size))
+
+					i += size + 4
+				}
 			}
 
 			//log.Printf("[AVC] %v, len: %d, ts: %10d, seq: %d", Types(payload), len(payload), packet.Timestamp, packet.SequenceNumber)

pkg/httpflv/flvio.go

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

pkg/httpflv/httpflv.go

@@ -128,8 +128,10 @@ func (c *Conn) Streams() ([]av.CodecData, error) {
 		return []av.CodecData{video, audio}, nil
 	} else if video != nil {
 		c.videoIdx = 0
+		c.audioIdx = -1
 		return []av.CodecData{video}, nil
 	} else if audio != nil {
+		c.videoIdx = -1
 		c.audioIdx = 0
 		return []av.CodecData{audio}, nil
 	}
@@ -139,17 +141,19 @@ func (c *Conn) Streams() ([]av.CodecData, error) {
 
 func (c *Conn) ReadPacket() (av.Packet, error) {
 	for {
-		tag, ts, err := flvio.ReadTag(c.reader, c.buf)
+		tag, ts, err := ReadTag(c.reader, c.buf)
 		if err != nil {
 			return av.Packet{}, err
 		}
 
 		switch tag.Type {
 		case flvio.TAG_VIDEO:
-			if tag.AVCPacketType != flvio.AVC_NALU {
+			if c.videoIdx < 0 || tag.AVCPacketType != flvio.AVC_NALU {
 				continue
 			}
 
+			//log.Printf("[FLV] %v, len: %d, ts: %10d", h264.Types(tag.Data), len(tag.Data), flvio.TsToTime(ts))
+
 			return av.Packet{
 				Idx:             c.videoIdx,
 				Data:            tag.Data,
@@ -159,7 +163,7 @@ func (c *Conn) ReadPacket() (av.Packet, error) {
 			}, nil
 
 		case flvio.TAG_AUDIO:
-			if tag.SoundFormat != flvio.SOUND_AAC || tag.AACPacketType != flvio.AAC_RAW {
+			if c.audioIdx < 0 || tag.SoundFormat != flvio.SOUND_AAC || tag.AACPacketType != flvio.AAC_RAW {
 				continue
 			}
 

pkg/mp4f/consumer.go

@@ -14,6 +14,7 @@ import (
 type Consumer struct {
 	streamer.Element
 
+	Medias     []*streamer.Media
 	UserAgent  string
 	RemoteAddr string
 
@@ -26,6 +27,10 @@ type Consumer struct {
 }
 
 func (c *Consumer) GetMedias() []*streamer.Media {
+	if c.Medias != nil {
+		return c.Medias
+	}
+
 	return []*streamer.Media{
 		{
 			Kind:      streamer.KindVideo,

pkg/rtsp/conn.go

@@ -2,7 +2,6 @@ package rtsp
 
 import (
 	"bufio"
-	"bytes"
 	"crypto/tls"
 	"encoding/binary"
 	"errors"
@@ -281,7 +280,7 @@ func (c *Conn) Options() error {
 	}
 
 	if val := res.Header.Get("Content-Base"); val != "" {
-		c.URL, err = url.Parse(val)
+		c.URL, err = urlParse(val)
 		if err != nil {
 			return err
 		}
@@ -311,7 +310,7 @@ func (c *Conn) Describe() error {
 	}
 
 	if val := res.Header.Get("Content-Base"); val != "" {
-		c.URL, err = url.Parse(val)
+		c.URL, err = urlParse(val)
 		if err != nil {
 			return err
 		}
@@ -380,7 +379,7 @@ func (c *Conn) SetupMedia(
 		}
 		rawURL += media.Control
 	}
-	trackURL, err := url.Parse(rawURL)
+	trackURL, err := urlParse(rawURL)
 	if err != nil {
 		return nil, err
 	}
@@ -882,42 +881,3 @@ func (c *Conn) bindTrack(
 
 	return track.Bind(push)
 }
-
-type RTCP struct {
-	Channel byte
-	Header  rtcp.Header
-	Packets []rtcp.Packet
-}
-
-const sdpHeader = `v=0
-o=- 0 0 IN IP4 0.0.0.0
-s=-
-t=0 0`
-
-func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
-	medias, err := streamer.UnmarshalSDP(rawSDP)
-	if err != nil {
-		// fix SDP header for some cameras
-		i := bytes.Index(rawSDP, []byte("\nm="))
-		if i > 0 {
-			rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
-			medias, err = streamer.UnmarshalSDP(rawSDP)
-		}
-		if err != nil {
-			return nil, err
-		}
-	}
-
-	// fix bug in ONVIF spec
-	// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
-	for _, media := range medias {
-		switch media.Direction {
-		case streamer.DirectionRecvonly, "":
-			media.Direction = streamer.DirectionSendonly
-		case streamer.DirectionSendonly:
-			media.Direction = streamer.DirectionRecvonly
-		}
-	}
-
-	return medias, nil
-}

pkg/rtsp/helpers.go

@@ -0,0 +1,63 @@
+package rtsp
+
+import (
+	"bytes"
+	"github.com/AlexxIT/go2rtc/pkg/streamer"
+	"github.com/pion/rtcp"
+	"net/url"
+	"strings"
+)
+
+type RTCP struct {
+	Channel byte
+	Header  rtcp.Header
+	Packets []rtcp.Packet
+}
+
+const sdpHeader = `v=0
+o=- 0 0 IN IP4 0.0.0.0
+s=-
+t=0 0`
+
+func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
+	medias, err := streamer.UnmarshalSDP(rawSDP)
+	if err != nil {
+		// fix SDP header for some cameras
+		i := bytes.Index(rawSDP, []byte("\nm="))
+		if i > 0 {
+			rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
+			medias, err = streamer.UnmarshalSDP(rawSDP)
+		}
+		if err != nil {
+			return nil, err
+		}
+	}
+
+	// fix bug in ONVIF spec
+	// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
+	for _, media := range medias {
+		switch media.Direction {
+		case streamer.DirectionRecvonly, "":
+			media.Direction = streamer.DirectionSendonly
+		case streamer.DirectionSendonly:
+			media.Direction = streamer.DirectionRecvonly
+		}
+	}
+
+	return medias, nil
+}
+
+// urlParse fix bug in URL from D-Link camera:
+// Content-Base: rtsp://::ffff:192.168.1.123/onvif/profile.1/
+func urlParse(rawURL string) (*url.URL, error) {
+	u, err := url.Parse(rawURL)
+	if err != nil && strings.HasSuffix(err.Error(), "after host") {
+		if i1 := strings.Index(rawURL, "://"); i1 > 0 {
+			if i2 := strings.IndexByte(rawURL[i1+3:], '/'); i2 > 0 {
+				return urlParse(rawURL[:i1+3+i2] + ":" + rawURL[i1+3+i2:])
+			}
+		}
+	}
+
+	return u, err
+}

pkg/rtsp/rtsp_test.go

@@ -0,0 +1,12 @@
+package rtsp
+
+import (
+	"github.com/stretchr/testify/assert"
+	"testing"
+)
+
+func TestURLParse(t *testing.T) {
+	base := "rtsp://::ffff:192.168.1.123/onvif/profile.1/"
+	_, err := urlParse(base)
+	assert.Empty(t, err)
+}

www/video-rtc.js

@@ -395,11 +395,11 @@ export class VideoRTC extends HTMLElement {
                         bufLen = 0;
                         sb.appendBuffer(data);
                     } else if (sb.buffered && sb.buffered.length) {
-                        const end = sb.buffered.end(sb.buffered.length - 1) - 5;
+                        const end = sb.buffered.end(sb.buffered.length - 1) - 15;
                         const start = sb.buffered.start(0);
                         if (end > start) {
                             sb.remove(start, end);
-                            ms.setLiveSeekableRange(end, end + 5);
+                            ms.setLiveSeekableRange(end, end + 15);
                         }
                         // console.debug("VideoRTC.buffered", start, end);
                     }