WebRTC peer on NodeJS for saving media stream

[Riccardo Tresoldi]

Hi evryone.

I'm searching to develop a server-side application that allow a WebRTC client to "send" its real time stream to the server throw WebRTC API.

I think to use this way because record the stream at the client-side and then send it to the server asynchronously became really hard because the browser cache can't contain lot's of minutes.

I try to understand what libjingle does, but i think it doesn't do something useful for my situation.

Then I try the NodaJS way, and i found lot's of packages for WebRTC, but I can't understand if there a package that allow a server to receive a stream throw client's RTCPerrConection.

Someone can help me?

PS: I know that this is a WebRTC discussion, and not a NodeJS discussion, but I can't find a NodeJS one oriented to the WebRTC.

[Ket]

The technology you are looking for doesn't exist yet. You have to create it yourself.

[nazmus shakeeb]

Hi Riccardo,

You can easily do this without any programming. You need 2 things in server side. You don't need node.js as asterisk has built in support for websocket. I have used latest software of sipml5 and asterisk.  

1.  http://sipml5.org/

2.  http://www.asterisk.org/

 

The extension.conf of Asterisk should have the following lines.

exten => 300,1,Answer

exten => 300,2,Record(/var/lib/asterisk/sounds/en/ntest:ulaw,,20)

exten => 300,3,Wait(3)

exten => 300,4,Playback(ntest)

exten => 300,5,Hangup

  

Although you can install and configure turnserver but you don't need for this purpose.

I have made this for my children and they enjoyed it very much. 

 

Installing sipml5 is very easy. you just need to put it on webserver. Installing asterisk is not straight forward. You need SRTP and ICE support.

You will many get help from internet about installing these. Please verify your asterisk log to be sure that these are working properly.

Thanks,

Shakeeb  

[Silvia Pfeiffer]

Is this audio-only?
Silvia.

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[Nazmus Shakeeb]

Yes. only with audio.
Shakeeb

[nazmus shakeeb]

Yes. for audio only.

Shakeeb

On Monday, October 28, 2013 9:56:18 AM UTC+6, Silvia Pfeiffer wrote:

[Riccardo Tresoldi]

Thanks for the replay.

I need specially for save the video. :(

[Riccardo Tresoldi]

There's nothing that can simplify my job? like some library?

Do you think that node.js is a correct way to do this?

[tom]

Not sure if getUserMedia can provide two events on mediaStream, like

 mediaStram.onVideoData = function(Blob video){}

 mediaStram.onAudioData = function(Blob audio){}

Best regards

  Tom

[Warren McDonald]

Hi,

we have experimented with this using several existing components and some integration glue.

The prime thing you need is a conferencing server (which can record) which understands SIP over websockets and can decode VP8, The 2 opensource ones are Licode from Lynckia and Open Telepresence from Doubango. There is also NGVX, a commercial one from Next Gen Bits. 

You need to setup up a conference, with recording on, which will give you a SIP address to call using the SIPml5 client side JS library. How to manage the conference, security and recording varies between implementations. The Open Telepresence server has nice auto conference feature and a simple single file recording output, so I would try that first.

Warren  

[Nazmus Shakeeb]

You  can record video by asterisk. You can also do it using node.js. 

For Asterisk you need to modify code to record video encoded using VP8 codec.

For node.js if you don't find any available library or program you have write it.

Shakeeb  

[Shachar]

There's this project: https://github.com/helloIAmPau/node-rtc

I believed he'd love all the help he can get

[Riccardo Tresoldi]

Thanks a lot! but I don't find any documentation... I don't understand exactly what it does... ;)

Can you explain me please?

[Nazmus Shakeeb]

You can also use WebRTC native client. 

To me this is the best option. 

You can download source code of WebRTC from here 

https://code.google.com/p/webrtc/source/checkout

There is an example to make or receive call in the following directory.  

WebRTC/talk/examples/peerconnection

You need to enable auto answer and dump incoming video RTP to a file. 

Shakeeb

[Harald Alvestrand]

Tom, what you are looking for is the Recording API.

http://www.w3.org/TR/mediastream-recording/

(folllow the link to the editor's draft)

The bad news: Nobody's implemented it yet.

--

[Nazmus Shakeeb]

Hi Riccardo,

I have been able to record( also play) vp8 video at the asterisk end using the following patch.

https://github.com/meetecho/asterisk-opus

Thanks,

Shakeeb

[tash...@gmail.com]

Hi Nazmus

> You need to enable auto answer and dump incoming video RTP to a file.

I have modified the peerconnection sample so that "server_test.html" streams video to "peerconnection_client.exe".  This is working fine but I am not sure how to dump incoming video RTP to a file.

Please could you tell me how to do that?

Many thanks

[Vinay Kumar]

hi, did you find any solution for the server side?

[Riccardo Tresoldi]

Hi Kumar,

I'm so sorry, but I don't develop on WebRTC since 2 years.

Anyway, I didn't found a solution for that issue.

I had to change the requirements.

Bye!

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[Sergio Garcia Murillo]

If you want to do recording server side on node you can use my media server:

https://github.com/medooze/media-server-node

It is still in beta stage but that works.. ;)

Best regards
Sergio

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[陈超]

can i use it in mac os?

在 2017年6月27日星期二 UTC+8下午9:18:56，Sergio Garcia Murillo写道：