webrtc-for-the-curious
mediasoup
webrtc-for-the-curious | mediasoup | |
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13 | 24 | |
1,884 | 5,928 | |
0.6% | 1.4% | |
5.2 | 9.5 | |
7 months ago | 3 days ago | |
Python | C++ | |
Creative Commons Zero v1.0 Universal | ISC License |
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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
mediasoup
- WebRTC for the Curious
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Implementing group video conference seems quite hard. Any tips on what I might be doing wrong ?
Given the financial restraint, i was avoiding paid API's like twilio. I started looking at mediasoup https://github.com/versatica/mediasoup, but while implementing the SFU server, its seems a lot more involved. For ex, TURN and STUN, peers negotiating different video codecs, adaptively changing the quality of video etc. Is it usually this difficult to implement a video conferencing apps ?
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STUNner Kubernetes media gateway for WebRTC
This release ships lots of new features to the already comprehensive set of them. Currently, we offer several working tutorials on how to set up STUNner with widely used WebRTC media servers and other applications that use WebRTC in Kubernetes, such as: - LiveKit - Jitsi - mediasoup - n.eko - Kurento
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Free - Self-hosted - WebRTC - alternative to Zoom, Teams, Google Meet - Real time video calls, chat, screen sharing, file sharing, collaborative whiteboard, dashboard, rooms scheduler and more!
Architecture WebRTC SFU (server with Selective Forwarding Unit). Can handle unlimited rooms without limits of time, each having 8+ users, potentially many as it is scalable. Routing is a multiparty topology, where each participant sends its media to the MiroTalk media server mediasoup and receives all other’s media from it. This version is Ideally suited for large group video conferences.
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Free Secure WebRTC P2P/SFU/C2C Video Calls, Screen Sharing, File Sharing, Chat and more.
I started the MiroTalk P2P & MiroTalk SFU projects during the pandemic period (about 1+ year ago), not knowing anything about the WebRTC. Making often the video conferences with my colleagues and not wanting to depend on Zoom, Teams, Google Meet... I decided to do some research about how it works and from there MiroTalk was born :) I Giving to everyone the chance to have its own instance of MiroTalk, which can be customized as you like and run in any cloud, vps, server. If you're just starting out, I suggest you take a look at the MiroTalk C2C (New) code, which can be a good starting point to understand how the architecture WebRTC Mesh (P2P) works. Later you can also study how the WebRTC SFU (Selective Forwarding Units - I recommend mediasoup which I personally love) or MCU (Multipoint Control Unit) architecture works. I wish you all the best!
- Jitsi: More secure, more flexible, and completely free video conferencing
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WebRTC 102: Understanding libWebrtc
The "Mediasoup" project provides a high level JavaScript/TypeScript interface to the WebRTC APIs. The core logic of this project is implemented in C++/Rust. Consider taking a look at the project if you want an easy-to-use library instead of the low-level libWebRTC APIs. A notable project to mention is the Pion/webrtc project which has a Golang implementation of the WebRTC API. Of course, we should mention the rust port WebRTC.rs. Let’s keep all the rustaceans happy too!
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Germany Forces a Microsoft 365 Ban Due to Privacy Concerns
Indeed, maddening, especially as the wonderful https://mediasoup.org/ is developed here. Europe will never have great tech companies when the answer seems to be throwing €€€ away instead of investing locally
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WebRTC P2P/SFU - Open Source - Alternative to Jitsi, Zoom, Google-Meet, Microsoft-Teams...
Hello thedominux, Thanks for your interest in MiroTalk ;) MiroTalk SFU code is: - JAVA-SCRIPT: 85.2% - HTML: 10.0% - CSS: 4.5% And has built in mediasoup server, more details about: https://mediasoup.org/
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How to Build a Video Chat App: Types, Cost, & Must-Have Features
Mediasoup
What are some alternatives?
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.
janus-gateway - Janus WebRTC Server
ws-tcp-proxy - Simple websocket tcp proxy.
jitsi - Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.
offline-browser-communication - Demonstration of a browser connecting to Pion WebRTC without a signaling server.
peerjs - Simple peer-to-peer with WebRTC.
webrtc-rtptransport - Repository for the RTPTransport specification of the WebRTC Working Group
webrtc-without-signaling-server - webrtc without signaling server. a stun server is still used if connecting over the internet.
direct-sockets - Direct Sockets API for the web platform
mirotalk - 🚀 WebRTC - P2P - Simple, Secure, Fast Real-Time Video Conferences Up to 4k and 60fps, compatible with all browsers and platforms.