A future proof, experimental WebRTC VP9 SVC SFU wit end to end encryption support
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A future proof, experimental WebRTC VP9 SVC SFU.
There are already several good production ready alternatives for implementing multiconferencing on webrtc, like Jitsi, Janus or SwitchRTC SFUs and even if you need more legacy support you can try our MCU. Our goal is to experiment and provide an early access to the functionalities that will be available in the near future that will improve drastically the performance and quality of multiconferencing services on WebRTC.
Due to the experimental nature of this functionalities we will only officially support Chrome Canary to be able to access the very latest functionalities available (sometimes even running behind a flag). We don't care about interporeability with other browsers (they will eventually catch up) nor SDP legacy support.
It is our goal to implement only the We intent to implement support the following features:
This is a moving target as new functionalities will be available on Chrome and some others will be removed, we will update our targets appropiatelly.
To enable VP9 SVC on Chrome Canary you must use the following command line:
A full version of SFrame end to end encryption is under works via insertable streams. Current implementation just uses frame counter as IV which is then inserted in the AES-GCM encrypted frame payload for emoing all required capabilities.
You just need to install all the depencencies and generate the ssl certificates:
npm install openssl req -sha256 -days 3650 -newkey rsa:1024 -nodes -new -x509 -keyout server.key -out server.cert
If you get an error like this ``` gyp verb build dir attempting to create "build" dir: /usr/local/src/medooze/sfu/nodemodules/medooze-media-server/build gyp ERR! configure error gyp ERR! stack Error: EACCES: permission denied, mkdir '/usr/local/src/medooze/sfu/nodemodules/medooze-media-server/build'
You may try instead with:
npm install --unsafe-perm ```
In order to run the sfu just:
node index.js [ip]
ipis the ICE candidate ip address used for RTP media. To test a simple web client just browse to