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There are some solutions to save a raw usermedia audio stream on the server side but I want to save the webRTC encoded stream which has low channel bandwidth transmission. I think of a solution that I'm not sure about:

Connect server and client using webRTC, the stream from the client is encoded then by the browser, convert the stream to mp3/ogg for later usage on the server. I found two server side nodejs webrtc implementations :

1- licode

2- node-webrtc

Is there any other solution or better idea for my problem?

Einar Sundgren
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Sina Sharafzadeh
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3 Answers3

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You could give a try to kurento

I will just link you this post : https://stackoverflow.com/a/24960167/1032907

Community
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Rocel
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you could give https://github.com/mido22/recordOpus a try,

I basically, capture user's microphone and convert the raw pcm data into opus packets, send it to server, convert back to wav format, also provided the option of converting to mp3 and ogg using ffmpeg.

mido
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    looks great , the bandwidth is nearly the same as native solution (I couldn't handle stream bitrate), there need to be more document on the project about configurations , it seems better solution but by now kurento is more stable one until then , I hope I can help on the project , thank you :) – Sina Sharafzadeh Mar 14 '15 at 20:34
  • I know I am lagging behind in the configuration part, more than happy to take help, just fork the project and make a pull request, and it ll be done. – mido Mar 15 '15 at 01:03
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I have recently successfully set up an OpenVidu server on Ubuntu for recording video and audio, which runs the Kurento Media Server under the hood, and offers a host of convenient API's. Running the OpenVidu server with their CloudFormation config is the easiest, which takes care of SSL setup, running the docker container necessary for recording, etc.

Julian Gong
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