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237.0 12.0 64.0 1.5 MB

RTSP Stream to WebBrowser MSE over WebSocket based MP4f segments

Go 3.77% JavaScript 96.23%

rtsptowsmp4f's Introduction

RTSPtoWSMP4f

RTSP Stream to WebBrowser MSE over WebSocket based MP4f segments

full native! not use ffmpeg or gstreamer

if you need RTSPtoWebRTC use https://github.com/deepch/RTSPtoWebRTC

Download Source

Download source

$ git clone https://github.com/deepch/RTSPtoWSMP4f

CD to Directory
```
 $ cd RTSPtoWSMP4f/
```
Test Run
```
 $ GO111MODULE=on go run *.go
```

Open Browser

open web browser http://127.0.0.1:8083 work chrome, safari, firefox

Configuration

Edit file config.json

format:

{
  "server": {
    "http_port": ":8083"
  },
  "streams": {
   "H264_AAC": {
      "url": "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov"
    }
  }
}

Limitations

Video Codecs Supported: H264 all profiles, H265 work only safari and (IE hw video card)

Audio Codecs Supported: AAC

Test

CPU usage 0.2% one core cpu intel core i7 / stream

Team

Deepch - https://github.com/deepch streaming developer

Dmitry - https://github.com/vdalex25 web developer

Other Example

Examples of working with video on golang

- You can make one-time donations via PayPal. I'll probably buy a ~~coffee~~ tea. 🍵

rtsptowsmp4f's People

Contributors

Stargazers

Watchers

rtsptowsmp4f's Issues

Audio in streaming

Is it possible to hear the streaming audio? I try to press on the unmute, but it won't allow it

"No Codec Info"

when I config the RTSP url and run ,it always says "No Codec Info". why?

SSL for RTSPtoWSMP4f

Is there a way to enable using SSL for this?

Home Assistant Integration

Is it possible to use this application to stream .mp4 files to Home Assistant?

I am currently using AlexIT/WebRTC to stream from a camera (rtsp feed) and it is fast and smooth.

To stream an mp4 I am using <video src="https://...file.mp4">. It works but is very slow and choppy.

I am looking for an mp4 alternative which is smooth and fast. I currently use javascript to build custom html to stream to the video element, so I am hopeful that I can just replace the https://file.mp4 with a connection to an RTSPtoWSMP4f server.

But I don't want to try if it is not possible. If is is doable, any tips would be greatly appreciated.

Can anyone please advise? Thanks!

There is no sound when i play the H264_AAC sample

There is no sound when I played the H264_AAC sample . And I'm sure I can hear somthing when I played this sample on VLC. It still did not work after I removed "muted" in index.tmpl.

H265,Stream Exit No Video On Stream.

2023/08/30 15:39:26 Server Start Awaiting Signal
2023/08/30 15:39:26 H265_AAC Stream Try Connect
2023/08/30 15:39:26 [OPTIONS rtsp://192.168.211.68:554/0 RTSP/1.0
CSeq: 1
User-Agent: Lavf58.76.100

]
2023/08/30 15:39:26 [RTSP/1.0 200 OK
CSeq: 1
Public: OPTIONS, DESCRIBE, PLAY, PAUSE, SETUP, TEARDOWN, SET_PARAMETER, GET_PARAMETER
Date:  Wed, Aug 30 2023 07:37:40 GMT

]
2023/08/30 15:39:26 [DESCRIBE rtsp://192.168.211.68:554/0 RTSP/1.0
CSeq: 2
Accept: application/sdp
User-Agent: Lavf58.76.100

]
2023/08/30 15:39:26 [DESCRIBE rtsp://192.168.211.68:554/0 RTSP/1.0
CSeq: 3
Accept: application/sdp
Authorization: Basic XXRtaW46TXXXMTIzNXXX
User-Agent: Lavf58.76.100

]
2023/08/30 15:39:26 [RTSP/1.0 200 OK
CSeq: 3
Content-Type: application/sdp
Content-Base: rtsp://192.168.211.68:554/0/
Content-Length: 581

v=0
o=- 1693381060347684 1693381060347684 IN IP4 0.0.0.0
s=Media Presentation
e=NONE
b=AS:5100
t=0 0
a=control:rtsp://192.168.211.68:554/0/
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
b=AS:5000
a=recvonly
a=x-dimensions:2560,1440
a=control:rtsp://192.168.211.68:554/0/track=1
a=rtpmap:96 H265/90000
m=audio 0 RTP/AVP 8
c=IN IP4 0.0.0.0
b=AS:50
a=recvonly
a=control:rtsp://192.168.211.68:554/0/track=2
a=rtpmap:8 PCMA/8000
a=Audio_Bit:64000
a=Media_header:MEDIAINFO=494D4B48010200000400050011710110401F000000FA000000000000000000000000000000000000;
a=appversion:1.0
]
2023/08/30 15:39:26 [SETUP rtsp://192.168.211.68:554/0/track=1 RTSP/1.0
CSeq: 4
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
User-Agent: Lavf58.76.100
Authorization: Basic XXRtaW46TXXXMTIzNXXX

]
2023/08/30 15:39:26 [RTSP/1.0 200 OK
CSeq: 4
Session: 0000000010913733;timeout=60
Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=6af60f63;mode="play"
Date:  Wed, Aug 30 2023 07:37:40 GMT

]
2023/08/30 15:39:26 [PLAY rtsp://192.168.211.68:554/0/ RTSP/1.0
CSeq: 5
User-Agent: Lavf58.76.100
Authorization: Basic XXRtaW46TXXXMTIzNXXX
Session: 0000000010913733

]
2023/08/30 15:39:26 [RTSP/1.0 200 OK
CSeq: 5
Session: 0000000010913733
RTP-Info: url=rtsp://192.168.211.68:554/0/track=1;seq=256;rtptime=590087160
Date:  Wed, Aug 30 2023 07:37:40 GMT

]
2023/08/30 15:39:46 [TEARDOWN rtsp://192.168.211.68:554/0/ RTSP/1.0
CSeq: 6
Authorization: Basic XXRtaW46TXXXMTIzNXXX
Session: 0000000010913733
User-Agent: Lavf58.76.100

]
2023/08/30 15:39:46 [RTSP Client RTP ReadFull <nil>]
2023/08/30 15:39:46 [RTSP Client Close <nil>]
2023/08/30 15:39:46 Stream Exit No Video On Stream

Network Issue

Hello,I can't connect to American IP,can you put the package on github? Any other packages can't also to get. Thank U.

package golang.org/x/crypto/sha3: unrecognized import path "golang.org/x/crypto/sha3": https fetch: Get "https://golang.org/x/crypto/sha3?go-get=1": dial tcp 216.239.37.1:443: i/o timeout
package google.golang.org/protobuf/encoding/prototext: unrecognized import path "google.golang.org/protobuf/encoding/prototext": https fetch: Get "https://google.golang.org/protobuf/encoding/prototext?go-get=1": dial tcp 216.239.37.1:443: i/o timeout
package google.golang.org/protobuf/encoding/protowire: unrecognized import path "google.golang.org/protobuf/encoding/protowire": https fetch: Get "https://google.golang.org/protobuf/encoding/protowire?go-get=1": dial tcp 216.239.37.1:443: i/o timeout

High Latency

Hi @deepch,

Firstable your project is very good. Congratulations for your work.

I'm trying to fix a high latency, about 5 seconds. There is a way to fine tune the initialization of a view, I think this delay is caused in the initialization of the video.

Could you indicate if possible where to make an adjustment to the source code?

Sir Can support H264_PCMALAW

that's all 。thank you for your reply 。

never play the video

It does not reproduce any video, neither the demo ones, nor my camera that I have locally.
my config.json is:

{
  "server": {
    "http_port": ":8083"
  },
  "streams": {
    "H264_AAC": {
      "on_demand": true,
      "url": "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov"
    },
    "MICRO": {
      "on_demand": true,
      "url": "rtsp://akaimp.duckdns.org/live/ch00_0"
    },
    "H265": {
      "on_demand": true,
      "url": "rtsp://171.25.232.14/37edec0a09174c639e21c9fc4cf9d9a1"
    },
    "H264_PCMALAW": {
      "on_demand": true,
      "url": "rtsp://admin:[email protected]:33556/"
    },
    "H264_AAC-2": {
      "on_demand": true,
      "url": "rtsp://admin:@172.16.255.247:554//h264Preview_01_main"
    },
    "MICRO-2": {
      "on_demand": true,
      "url": "rtsp://admin:@172.16.255.247:554//h264Preview_01_main"
    },
    "H265-2": {
      "on_demand": true,
      "url": "rtsp://admin:@172.16.255.247:554//h264Preview_01_main"
    },
    "H264_PCMALAW-2": {
      "on_demand": true,
      "url": "rtsp://admin:@172.16.255.247:554//h264Preview_01_main"
    }

  }
}

demo1 rtsp: codec data change, please call HandleCodecDataChange()

detailed log

OPTIONS rtsp://192.168.1.100:554/test RTSP/1.0
CSeq: 1

DESCRIBE rtsp://192.168.1.100:554/test RTSP/1.0
CSeq: 2
Accept: application/sdp

< v=0
o=- 0 0 IN IP4 127.0.0.1
s=Media Presentation
c=IN IP4 192.168.1.100
t=0 0
a=tool:libavformat 58.46.101
m=video 0 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1; sprop-parameter-sets=Z0LglageAIn5UA==,aM4wpIA=; profile-level-id=42E095
a=control:streamid=0

SETUP rtsp://192.168.1.100:554/test/streamid=0 RTSP/1.0
CSeq: 3
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Session: cAa2lnZMg

PLAY rtsp://192.168.1.100:554/test RTSP/1.0
CSeq: 4
Session: cAa2lnZMg
2020/06/18 14:34:54 demo1 rtsp: codec data change, please call HandleCodecDataChange()
2020/06/18 14:34:54 demo1 reconnect wait 5s
2020/06/18 14:34:59 demo1 connect rtsp://192.168.1.100/test

Could you please add an open-source license?

Hello, I would like to use your project but there is no license present. Would greatly appreciate it if you can add one so myself and others can use it.

Dependency Dashboard

This issue lists Renovate updates and detected dependencies. Read the Dependency Dashboard docs to learn more.

Open

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Detected dependencies

gomod

go.mod

go 1.15

github.com/deepch/vdk v0.0.0-20220402115557-02504b116966@02504b116966

github.com/gin-gonic/gin v1.7.7

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Delay compare to VLC

Very good job!

but I compared the rtsp streaming to vlc rtsp and after a minute or two the go service have delay that growing all the time

Memory is constantly increasing, there is no trace of release

Hello, I can play the video normally when I open it with the browser, but as time goes by, I find that the memory occupied by the browser keeps increasing without any trace of release. Less than an hour, the memory is more than 2G, can you help to solve this problem? thank you

Cannot close pull stream

I worked it with EasyDarwin.
When i visit the video，and then close the view.
the pull stream is still worked .There is any ways to del or close the channel ??

when i use the custom rtsp url, the preview screen is not play, the interface response is full

RTSP/1.0 461 client error

I use vlc to provide a rtsp ,but it's err

RTSP/1.0 461 Client error.

How to solve it,please?

I test on a Hik camera (nvr),but I meet some problems；

hi boss @deepch
I test on a Hik camera (nvr),but I meet some problems；
the same stream url. can play on the "vlc plyer",but not work on the demo html page , can you help me. my dear brother

Installation Failed

The installation instructions are incorrect or I'm not performing them as expected.
Step 1. "go get github.com/deepch/RTSPtoWSMP4f"
does not created the directory described in
Step 2. "cd src/github.com/deepch/RTSPtoWSMP4f".

How to draw rectangle on go stream?

I'm trying to draw a rectangle on the []uint8 stream.
the goal is when I'm using the RTSPtoWSMP4f I can draw a bounding box prediction of ML models on the video in the client.

in short:
[]uint8 + rectangle => videoTagHtml

Is there any way to support IOS devices

It works well on my PC chrome, but not work on IOS safari. I printed the error log on IOS safar, which says

ReferenceError: Can't find variable: MediaSource
/static/js/app.js?ver=2020-10-14%2009%3a23%3a56.286920163%20%2b0800%20CST%20m%3d%2b13.941190558:11:25 global code@http://192.168.32.229:8083/static/js/app.js?ver=2020-10-14%2009%3a23%3a56.286920163%20%2b0800%20CST%20m%3d%2b13.941190558:11:25

Camera intelligent stream, variable key frame

According to the code, currently judging whether to load the video is judged by the key frame. When the camera is set to smart streaming and the key frame interval of the RTSP video stream is uncontrollable, it can be loaded when the same client accesses for the first time, and the subsequent access to the websocket will be in the pending state until the RTSP stream has a new i frame. The waiting time for loading is very long. Is there a way to solve this?

QuotaExceededError

Hello,

first of all, thanks for your app, it works great. I've managed to integrate it with my application.
However, sometimes I get this error:

QuotaExceededError
Failed to execute 'appendBuffer' on 'SourceBuffer': The SourceBuffer is full, and cannot free space to append additional buffers.

Unfortunately, I don't know how to reproduce this, it just happens time to time...

EDIT: it's a client (javascript) error

Два видео потока на 1 HTML странице.

Здравствуйте! У меня вопрос. А можно, используя RTSPtoWSMP4f, вывести одновременно 2 видео потока в разных окнах на одну HTML страницу? Если нет, то что конкретно мешает: Js скрипт или socket, формирующий видео поток? Сейчас потоки открываются последовательно, замещая друг друга в одном окне.
Спасибо!

How to change server url

I have created a reverse ssh tunnel from home to my vps for my rtsp streams and created a systemd process on the vps to run the RTSPtoWSMP4f server. What file do I hack in order to change the default url myserver:8083 to something like
myserver:8083/homecctv in order to make it a little more hidden?

Добавление потоков на лету

Добрый день! Очень хотелось бы иметь возможность добавлять потоки в течение работы сервиса. Или, как вариант, перечитывать конфиг, без остановки сервиса

Not working using Chrome in iPhone

cannon display the video in iPhone

How to handle frame get stucked

It works great when the RTSP is locally, but when I playing it from remote via static IP the streaming get stack sometimes and I want to catch the error in order to know why and let the client (in my case using the web via Iframe) refresh or just no that something wrong in the streaming.

Not familiar with GO, please help.

BTW, hoping the new service RTSPtoWeb will solve it...

Не открывает изображение

Добрый день! Решил последовать вашему совету и попробовать RTSPtoWSMP4f для передачи видео с камеры RTSP в браузер. Столкнулся с тем, что изображение не воспроизводится ни с моей камеры, ни с ваших примеров. При том, что RTSPtoWebRTC изображение с моей камеры прекрасно воспроизводит. Сразу оговорюсь, что звук меня не интересует. Пока пытаюсь сделать все в локальной сети. Помогите, пожалуйста, разобраться в чем дело?

Вывод в консоли:

``[GIN-debug] GET / --> main.serveHTTP.func1 (3 handlers)
[GIN-debug] GET /player/:suuid --> main.serveHTTP.func2 (3 handlers)
[GIN-debug] GET /ws/:suuid --> main.serveHTTP.func3 (3 handlers)
[GIN-debug] GET /static/*filepath --> github.com/gin-gonic/gin.(*RouterGroup).createStaticHandler.func1 (3 handlers)
[GIN-debug] HEAD /static/*filepath --> github.com/gin-gonic/gin.(*RouterGroup).createStaticHandler.func1 (3 handlers)
[GIN-debug] Listening and serving HTTP on :8083
[GIN] 2021/02/16 - 08:24:40 | 200 | 9.422189ms | 192.168.1.154 | GET "/player/H265"
[GIN] 2021/02/16 - 08:24:40 | 200 | 49.387034ms | 192.168.1.154 | GET "/static/js/app.js?ver=2021-02-16%2008%3a24%3a40.54063974%20%2b0000%20UTC%20m%3d%2b11.089518110"
[GIN] 2021/02/16 - 08:24:59 | 200 | 5.75281ms | 192.168.1.154 | GET "/player/MICRO"
[GIN] 2021/02/16 - 08:24:59 | 200 | 790.057µs | 192.168.1.154 | GET "/static/js/app.js?ver=2021-02-16%2008%3a24%3a59.914388279%20%2b0000%20UTC%20m%3d%2b30.463266482"
[GIN] 2021/02/16 - 08:25:22 | 200 | 6.084966ms | 192.168.1.154 | GET "/player/H264_PCMALAW"
[GIN] 2021/02/16 - 08:25:22 | 200 | 865.179µs | 192.168.1.154 | GET "/static/js/app.js?ver=2021-02-16%2008%3a25%3a22.695737523%20%2b0000%20UTC%20m%3d%2b53.244615726"
[GIN] 2021/02/16 - 08:25:33 | 200 | 5.774351ms | 192.168.1.154 | GET "/player/H264_PCMALAW"
[GIN] 2021/02/16 - 08:25:33 | 200 | 785.724µs | 192.168.1.154 | GET "/static/js/app.js?ver=2021-02-16%2008%3a25%3a33.037778801%20%2b0000%20UTC%20m%3d%2b63.586657046"
[GIN] 2021/02/16 - 08:26:03 | 200 | 5.148621ms | 192.168.1.154 | GET "/player/H264_MY-CAM"
[GIN] 2021/02/16 - 08:26:03 | 200 | 866.971µs | 192.168.1.154 | GET "/static/js/app.js?ver=2021-02-16%2008%3a26%3a03.302493357%20%2b0000%20UTC%20m%3d%2b93.851371768"
[GIN] 2021/02/16 - 08:26:27 | 200 | 5.72881ms | 192.168.1.154 | GET "/player/H264_MY-CAM"
[GIN] 2021/02/16 - 08:26:27 | 200 | 735.434µs | 192.168.1.154 | GET "/static/js/app.js?ver=2021-02-16%2008%3a26%3a27.217189096%20%2b0000%20UTC%20m%3d%2b117.766067300"

Индикатор загрузки крутится вечно, видео не появляется

Video does not start

Hi, I'm trying to run the code to show an IP camera in my local network. I changed the config.json file to point to the IP camera (in my case, the line in the config.json is : "url": "rtsp://192.168.100.119:80/1", if I input this address to VLC it plays the video from the camera with no problems)
In the browser, it keeps the video spinning but does not show anything. The code is running in a Ubuntu server machine, and the browser in a Windows 10 machine (I tried in Firefox and Chrome).
The code outputs the following messages:
https://pastebin.com/36GvErs1

Are there any other logs that I can take a look to try to understand why the video is not working? Are there anything I can do to try to solve the issue and make it work?

Thank you!

Rewind stream?

I have run the demo with my ip camera stream. I want to rewind the running stream, but I can only rewind about 3 minutes, if I rewind more, the stream will go back to the current time. Is there any way I can extend this rewind time. Thank you!

RTSPtoWSMP4f stream can lag several hours behind source

When viewing a video stream via RTSPtoWSMP4f, things go well for a while, but when you switch tabs, the browser (chrome in particular) will try to slow all background tabs down (to try to prevent them from dragging down the system). I believe this causes the subroutines displaying the live video to slow down.

I am not familiar with the details of what you have written (by the way, thanks!), but i'm assuming it's the player part that's lagging. If so, is there a way that when it's behind, the player could drop all the frames up to the most recently received keyframe and start playing at that keyframe?

TO REPLICATE:

open chrome
open a live video stream to a camera in tab 1
- confirm that you are seeing live video
open a live video stream to a different camera in tab 2
- confirm that you are seeing live video
open a live video stream to yet another camera in tab 3
go home leaving tab 3 as the displaying tab
come back the next day. tab 3 may be in sync, but when you go to tab 1 or tab 2, they show you video of what happened several hours ago.

hello,why? is bug?

runtime error: index out of range [1] with length 1
I:/go1.13.15.windows-amd64/go/src/runtime/panic.go:75 (0x433ed9)
goPanicIndex: panic(boundsError{x: int64(x), signed: true, y: y, code: boundsIndex})
D:/go-gateway/gopath/pkg/mod/github.com/deepch/[email protected]/format/mp4f/muxer.go:256 (0x887772)
(*Muxer).WritePacket: stream := element.streams[pkt.Idx]
D:/go-gateway/RTSPtoWSMP4f/http.go:82 (0xb43422)
ws: ready, buf, _ := muxer.WritePacket(pck, false)
D:/go-gateway/gopath/pkg/mod/golang.org/x/[email protected]/websocket/server.go:89 (0xb2b524)
Server.serveWebSocket: s.Handler(conn)
D:/go-gateway/gopath/pkg/mod/golang.org/x/[email protected]/websocket/server.go:112 (0xb2b876)
Handler.ServeHTTP: s.serveWebSocket(w, req)
D:/go-gateway/RTSPtoWSMP4f/http.go:39 (0xb44511)
serveHTTP.func3: handler.ServeHTTP(c.Writer, c.Request)
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/context.go:161 (0xb076b2)
(*Context).Next: c.handlersc.index
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/recovery.go:83 (0xb23f4a)
RecoveryWithWriter.func1: c.Next()
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/context.go:161 (0xb076b2)
(*Context).Next: c.handlersc.index
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/logger.go:241 (0xb2296b)
LoggerWithConfig.func1: c.Next()
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/context.go:161 (0xb076b2)
(*Context).Next: c.handlersc.index
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/gin.go:409 (0xb149a1)
(*Engine).handleHTTPRequest: c.Next()
D:/go-gateway/gopath/pkg/mod/github.com/gin-gonic/[email protected]/gin.go:367 (0xb1444c)
(*Engine).ServeHTTP: engine.handleHTTPRequest(c)
I:/go1.13.15.windows-amd64/go/src/net/http/server.go:2831 (0x7f25f5)
serverHandler.ServeHTTP: handler.ServeHTTP(rw, req)
I:/go1.13.15.windows-amd64/go/src/net/http/server.go:1919 (0x7ecac0)
(*conn).serve: serverHandler{c.server}.ServeHTTP(w, w.req)
I:/go1.13.15.windows-amd64/go/src/runtime/asm_amd64.s:1357 (0x462fb0)
goexit: BYTE $0x90 // NOP

Reload configuration

Is it possible to reload a configuration with some POSIX signal, like SIGHUP?

How to save stream as a local file?

I want to save stream as a local file, such as mp4, but how?

Lag while playing

If you leave the player for about 30 minutes, I see a lot of lag. When you refresh it catches up again.

Is it possible to view more than one camera at a time (ex: 2x2)?

I would like to know if it is possible to view more than one camera at a time. Thanks!

Изменить качество видео

Приветствую.
Подскажите, пожалуйста, как уменьшить качество видео на стороне сервера?
Необходимо уменьшить количество трафика в сторону сервера и обратно, но скорость воспроизведения оставить прежней. Вывод, уменьшить качество видео.

AAC/PCM audio codec

I see here

https://github.com/deepch/RTSPtoWSMP4f/blob/master/stream.go#L44

that the audio is turned off.

If I turn it out to false, I hear AAC sound in my browser.
But in case PCM audio format I see message

muxer.WriteHeader fmp4: codec type=PCM_MULAW is not supported

and I even do not see video.

Is it supported to show only video stream in case of unsupported audio codecs?

In web/static/JS/app.js Using "127.0.0.1" may cause the media cannot play normally

when browser and server are not the same computer

wowza rtsp address is not supported.

The rtsp address in Demo works well. However, the bunny provided by wowza rtsp address is not supported.

app.js?ver=2020-09-2…3d%2b1.933701794:22 Uncaught DOMException: Failed to execute 'appendBuffer' on 'SourceBuffer': The HTMLMediaElement.error attribute is not null.
    at pushPacket (http://x.x.x.x:8081/static/js/app.js?ver=2020-09-27%2006%3a25%3a35.958356866%20%2b0000%20UTC%20m%3d%2b1.933701794:22:18)
    at WebSocket.ws.onmessage (http://x.x.x.x:8081/static/js/app.js?ver=2020-09-27%2006%3a25%3a35.958356866%20%2b0000%20UTC%20m%3d%2b1.933701794:83:7

My Setting

config.json Modifyed

{
  "server": {
    "http_port": ":8081"
  },
  "streams": {
    "demo": {
        "url": "rtsp://171.25.232.5/3e3019c02ad14b84bceb2b810db29918"
    },
    "wowza": {
      "url": "rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov"
    }
  }
}

OS: Linux (ubuntu 18.04)
Go Version: go1.15.2 linux/amd64

461 Unsupported Transport

Hi,
I have an issue with my encoder device (https://avonic.com/fr/produit/rec200/) who provide rtsp stream :

[GIN-debug] Listening and serving HTTP on :8083 2022/11/03 14:26:34 LIVE1 Stream Try Connect 2022/11/03 14:26:34 Server Start Awaiting Signal 2022/11/03 14:26:35 Camera send statusRTSP/1.0 461 Unsupported Transport 2022/11/03 14:26:36 LIVE1 Stream Try Connect 2022/11/03 14:26:37 Camera send statusRTSP/1.0 461 Unsupported Transport 2022/11/03 14:26:38 LIVE1 Stream Try Connect 2022/11/03 14:26:39 Camera send statusRTSP/1.0 461 Unsupported Transport
But the stream work with vlc ... any Id how can I fix that ?

Thanks in advance

RTSP Client Unauthorized 401

Hello, it is like this. My computer cannot directly access the server, so I need an interface machine to forward tcp connection to the server through nginx. The proxy port of the interface machine is 556 and the interface machine is forwarded to interface 554 of the server. RTSP Client Unauthorized 401 is prompted. How to solve this problem

Config.json

Nginx Options

Installation instructions outdated

Installation instructions are outdated:

~/tmp/r
❯ go get github.com/deepch/RTSPtoWSMP4f

go: downloading github.com/deepch/RTSPtoWSMP4f v0.0.0-20210209015721-314189c0c54c
go: downloading github.com/gin-gonic/gin v1.6.3
go: downloading github.com/deepch/vdk v0.0.0-20210209013919-981746cd6aa0
go: downloading golang.org/x/net v0.0.0-20210119194325-5f4716e94777
go: downloading github.com/mattn/go-isatty v0.0.12
go: downloading github.com/gin-contrib/sse v0.1.0
go: downloading github.com/json-iterator/go v1.1.10
go: downloading github.com/golang/protobuf v1.4.3
go: downloading github.com/ugorji/go v1.2.4
go: downloading gopkg.in/yaml.v2 v2.4.0
go: downloading github.com/go-playground/validator/v10 v10.4.1
go: downloading github.com/ugorji/go/codec v1.2.4
go: downloading golang.org/x/sys v0.0.0-20210124154548-22da62e12c0c
go: downloading google.golang.org/protobuf v1.25.0
go: downloading github.com/modern-go/concurrent v0.0.0-20180306012644-bacd9c7ef1dd
go: downloading github.com/modern-go/reflect2 v1.0.1
go: downloading github.com/go-playground/universal-translator v0.17.0
go: downloading github.com/leodido/go-urn v1.2.1
go: downloading golang.org/x/crypto v0.0.0-20201221181555-eec23a3978ad
go: downloading github.com/go-playground/locales v0.13.0

~/tmp/r 53s
❯ cd src/github.com/deepch/RTSPtoWSMP4f

cd: no such file or directory: src/github.com/deepch/RTSPtoWSMP4f

~/tmp/r
❯ ls -l

~/tmp/r

The files went to ~/go but there is no src either:

❯ ls -l ~/go
total 0
drwxr-xr-x  3 semenov  staff   96 Mar  8 12:49 bin
drwxr-xr-x  4 semenov  staff  128 Mar  8 12:48 pkg

This is go 1.16:

❯ go version
go version go1.16 darwin/arm64

H.264 NAL unit stream, no audio

...this is the result:

Would you know, why?