I'm the creator of Peer Calls [1], a peer to peer video conferencing web app using WebRTC, and it has a basic chat functionality (sending files is a little quirky). The first release was back it in 2016. Users create a room and share the link.
It's built in NodeJS/React/TypeScript, and I just recently ported the backend to Go because I wanted to build a Selective Forwarding Unit using pion/webrtc. You can test this in the alpha release on peercalls.com/alpha [2].
Would love to get more feedback and/or bug reports! Open source, available on
GitHub [3].
I discovered Peer Calls last night and think it’s an awesome project! It worked flawlessly from both my phone and desktop; it’s one of the (surprisingly few) fully featured WebRTC chat apps!
My one hope: Could it be possible to select from different audio bitrates when on a direct connection? High bandwidth audio seems like one of the huge selling points of P2P chat applications. My naive assumption is that it’s the sampleRate constraint on the audio stream, but I’m not sure how the compression works...
Check out bigbluebutton [0]. It is open source with tons of features. It however requires specific Ubuntu 16.04 version and can only be installed on that OS. It is not the best fit for very large audiences (say over 200) in one session BUT for that they are working on another load balancing solution recently [1]
I have been using this for our clients (edtech) for a while and with their 2.2 official version, they made it HTML5 only and removed flash. Works really well if you follow all the guidelines and installation instructions.
They also have a bash installer script [2] which literally does everything for you in one single script.
I use it as a student. It's very good and easy to use. I'm curious about the performance with many participants though as we only use it in classes with at most 40 persons. And of course no camera just the presentation.
[+] [-] jeremija|6 years ago|reply
It's built in NodeJS/React/TypeScript, and I just recently ported the backend to Go because I wanted to build a Selective Forwarding Unit using pion/webrtc. You can test this in the alpha release on peercalls.com/alpha [2].
Would love to get more feedback and/or bug reports! Open source, available on GitHub [3].
[1]: https://peercalls.com
[2]: https://peercalls.com/alpha
[3]: https://github.com/peer-calls/peer-calls
[+] [-] zalo|6 years ago|reply
My one hope: Could it be possible to select from different audio bitrates when on a direct connection? High bandwidth audio seems like one of the huge selling points of P2P chat applications. My naive assumption is that it’s the sampleRate constraint on the audio stream, but I’m not sure how the compression works...
[+] [-] based2|6 years ago|reply
https://jitsi.org/jitsi-meet/
https://bigbluebutton.org/
https://openmeetings.apache.org/
https://github.com/havfo/multiparty-meeting
src: https://linuxfr.org/news/organiser-des-visioconferences-de-h...
[+] [-] codegeek|6 years ago|reply
I have been using this for our clients (edtech) for a while and with their 2.2 official version, they made it HTML5 only and removed flash. Works really well if you follow all the guidelines and installation instructions.
They also have a bash installer script [2] which literally does everything for you in one single script.
[0] https://docs.bigbluebutton.org
[1] https://github.com/blindsidenetworks/scalelite
[2] https://github.com/bigbluebutton/bbb-install
[+] [-] xthetrfd|6 years ago|reply
[+] [-] unknown|6 years ago|reply
[deleted]
[+] [-] blohs|6 years ago|reply
[+] [-] mutant|6 years ago|reply
[+] [-] rvz|6 years ago|reply
[0] https://jitsi.org
[+] [-] tkjef|6 years ago|reply
best alternative i've seen so far was a webrtc setup with okta authentication. https://github.com/rdegges/chatapp
[+] [-] blohs|6 years ago|reply
[+] [-] derfabianpeter|6 years ago|reply
[1] https://www.peter.saarland/hosted-homeoffice
[+] [-] metah|6 years ago|reply