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by jeremija
2247 days ago
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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]. [1]: https://peercalls.com [2]: https://peercalls.com/alpha [3]: https://github.com/peer-calls/peer-calls |
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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...