CODEWITHSUNDEEP
node.jswebrtcvoip
Ran4Display
Ran4Display

Reputation: 46

WebRTC server node

I'm painfuly new to webrtc world. Have learned the essentials but when I want to build my own very simple app, I understand I'm missing a lot to start develope.

What I want?
Build a receiver audio node server that receive the audio stream and do with it something

(my main goal is sending the stream to another SIP node after adding some audio background effects, but that can wait for another time)

What the heck is my problem?
I don't know from where to start and what is the best way for me to do it right... I'm kinda lost..

There are already built webrtc servers like Kurento or Janus from one hand,
and (It sound better choice for me, but I'm not sure) some node base webrtc servers like node-webrtc, easyrtc, electron-rtc to the other hand

I also looked at the native API and it is a bit complicated and it is written in CPP that I do not master (but can learn if you guys tell me it is the best way to go)

What choice should I choose for my purpose?
I'm fairly good programmer and can cope with any lang, but still wish the easiest and best option for my purpose.

Please, if anyone was at the same phase as I'm now and passed it,
Or if you know how to help me and set me on the right track.
Please help

As you can see, I'm lost and beg for guidance.

Thank you heros of the world wide web!!

Upvotes: 1

Views: 551

Answers (1)

DavidP
DavidP

Reputation: 71

If you want someone using a browser to speak with a SIP user (or another webrtc user), then one of the easiest paths is to setup a Freeswitch server and use its 'verto' webrtc js library. See http://evoluxbr.github.io/verto-docs

The doc above will show you how to make a webrtc call to Freeswitch. You will still need to find a way for the 2nd caller to find what call/session to connect to. Ask about that on the Freeswitch users mailing list.

Upvotes: 1

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