amazon-kinesis-video-streams-webr
webrtc-for-the-curious
| amazon-kinesis-video-streams-webr | webrtc-for-the-curious | |
|---|---|---|
| 5 | 13 | |
| - | 1,884 | |
| - | 0.6% | |
| - | 5.2 | |
| - | 7 months ago | |
| Python | ||
| - | Creative Commons Zero v1.0 Universal |
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amazon-kinesis-video-streams-webr
- VoRS: Vo(IP) Simple Alternative to Mumble
- WebRTC for the Curious
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Show HN: WebRTC Nuts and Bolts, A holistic way of understanding how WebRTC runs
Yes! I have used it in production with these.
* FaceTime @ Apple https://support.apple.com/en-us/HT212619
* KVS and Chime @ AWS https://github.com/awslabs/amazon-kinesis-video-streams-webr.... Lots of security cameras and robots use it, not public though.
* Lightstream https://golightstream.com . Cloud compositing and other magic.
It is kind of amazing everywhere you will find WebRTC. Stadia, Boston Dynamics, Zoom, Meet, Security Systems, Drones etc... It is probable that you use WebRTC in production everyday :)
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WebRTC is now a W3C and IETF standard
https://github.com/awslabs/amazon-kinesis-video-streams-webr... (C/Embedded)
webrtc-for-the-curious
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Dive into Web RTC or write SFU on your own
Here I will briefly go over the basics of how Web RTC works; for those who are interested in going a little deeper, I’ll leave the link here. In order for two peers to be able to provide themselves with RTCPeerConnection, the SDP (Session Description Protocol) protocol is used. The protocol has a key-value structure and is essentially a description of a single peer (the name speaks for itself).
- WebRTC for the Curious
- Show HN: Bring phone calls into the browser (sip-to-WebRTC)
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Ask HN: What side projects landed you a job?
I have worked four jobs related to https://github.com/pion/webrtc and one for https://webrtcforthecurious.com
Two companies used Pion. The other two were just using the protocol (WebRTC)
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Replacing WebRTC: real-time latency with WebTransport and WebCodecs
For the WebRTC jargon check out https://webrtcforthecurious.com/
If that still doesn’t cover enough I would love to hear! Always trying to make it better.
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OBS Merges WebRTC Support
It is pretty easy to get a one way trip time for packets that is sub-second! You see it with conferencing and other real-time communication things.
If you are curious on the 'how' of WebRTC I wrote a Free/Open Source book that goes into the details https://webrtcforthecurious.com/. Happy to answer any particular questions you have.
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Help: I'm facing an issue developing a webrtc app
Read this thoroughly: https://webrtcforthecurious.com
What are some alternatives?
webrtc - A pure Rust implementation of WebRTC
web-codecs - WebCodecs is a flexible web API for encoding and decoding audio and video.
Pion WebRTC - Pure Go implementation of the WebRTC API
broadcast-box - A broadcast, in a box.
aiortc - WebRTC and ORTC implementation for Python using asyncio
ws-tcp-proxy - Simple websocket tcp proxy.
libdatachannel - C/C++ WebRTC network library featuring Data Channels, Media Transport, and WebSockets
offline-browser-communication - Demonstration of a browser connecting to Pion WebRTC without a signaling server.
datachannel-rs - Rust wrappers for libdatachannel
webrtc-rtptransport - Repository for the RTPTransport specification of the WebRTC Working Group
werift-webrtc - WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP/SRTP/WEBM/MP4
direct-sockets - Direct Sockets API for the web platform