mediasoup
webrtc
mediasoup | webrtc | |
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24 | 41 | |
5,895 | 3,792 | |
0.9% | 1.5% | |
9.5 | 8.6 | |
8 days ago | 9 days ago | |
C++ | Rust | |
ISC License | Apache License 2.0 |
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
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
webrtc
- VoRS: Vo(IP) Simple Alternative to Mumble
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Pure C WebRTC
I am really excited about https://github.com/sepfy/libpeer. It has examples ready for ESP32 etc....
When working on KVS I wasn't familiar with the embedded space at all. I saw 'heavyweight' embedded where you were running on Linux. Then you had RTOS/No OS at all. I wasn't prepared for these devices at all. If we can make WebRTC work in the embedded space I think it will really accelerate what developers are able to build!
Remotely driven cars, security cameras, robots in hospitals that bring iPads to infectious patients etc... Creative people are building amazing things. The WebRTC/video space needs to work harder and support them :)
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I love how diverse the WebRTC space is now. Outside of this implementation you have plenty of other options!
* https://github.com/shinyoshiaki/werift-webrtc (Typescript)
* https://github.com/pion/webrtc (Golang)
* https://github.com/webrtc-rs/webrtc (Rust)
* https://github.com/algesten/str0m (Rust)
* hhttps://github.com/sepfy/libpeer (C/Embedded)
* https://webrtc.googlesource.com/src/ (C++)
* https://github.com/sipsorcery-org/sipsorcery (C#)
* https://github.com/paullouisageneau/libdatachannel (C++)
* https://github.com/elixir-webrtc (Elixir)
* https://github.com/aiortc/aiortc (Python)
* GStreamer’s webrtcbin (C)
See https://github.com/sipsorcery/webrtc-echoes for examples of some running against each other.
- WebRTC for the Curious
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Building WebRTC server implementation for Erlang
This is not true, there are actually multiple WebRTC implementations in different languages besides the reference library: aiortc (python), libdatachannel (C++), sipsorcery (C#),webrtc-rs (rust), werift (Typescript), and Amazon Kinesis (C)
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Trying to get WebRTC ICE to work with Rust
I am trying to get WebRTC working using Rust https://github.com/webrtc-rs/webrtc
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Real-Time Video Processing with WebCodecs and Streams
I have opened an issue on GitHub [1], we can continue there.
[1] https://github.com/webrtc-rs/webrtc/issues/426
- Can you help me with Webrtc-rs and insertable streams?
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A Rust client library for interacting with Microsoft Airsim https://github.com/Sollimann/airsim-client
webrtc
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A Rust library for cross-platform video apps using WebRTC and LiveKit
webrtc.rs is a port of Pion (which we also use). It's a better fit for server-side use
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WebRTC signaling server in Rust
I want to use peer-to-peer communication and data transfer for my next side project (client-server web app). I've been doing some research and WebRTC seems to be the only option for this on the client. There are a ton of libraries and product offering for facilitating STUN/TURN servers as a service, but I'm quite interested in learning more about these protocols. That being said, I'm not the best rust programmer (part of the reason of using Rust as the server is so that I can learn more), and the signalling protocols seem rather complicated. I've looked at https://github.com/webrtc-rs/webrtc and it seems promising.
What are some alternatives?
Pion WebRTC - Pure Go implementation of the WebRTC API
aiortc - WebRTC and ORTC implementation for Python using asyncio
janus-gateway - Janus WebRTC Server
jitsi - Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.
tauri - Build smaller, faster, and more secure desktop applications with a web frontend.
peerjs - Simple peer-to-peer with WebRTC.
libdatachannel - C/C++ WebRTC network library featuring Data Channels, Media Transport, and WebSockets
webrtc-without-signaling-server - webrtc without signaling server. a stun server is still used if connecting over the internet.
opencv-python - Automated CI toolchain to produce precompiled opencv-python, opencv-python-headless, opencv-contrib-python and opencv-contrib-python-headless packages.
mirotalk - 🚀 WebRTC - P2P - Simple, Secure, Fast Real-Time Video Conferences Up to 4k and 60fps, compatible with all browsers and platforms.
Homer - HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring