webrtc-for-the-curious
broadcast-box
webrtc-for-the-curious | broadcast-box | |
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13 | 10 | |
1,884 | 524 | |
0.6% | 5.3% | |
5.2 | 8.9 | |
7 months ago | 5 days ago | |
Python | Go | |
Creative Commons Zero v1.0 Universal | MIT License |
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
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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
broadcast-box
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Show HN: Bring phone calls into the browser (sip-to-WebRTC)
> [1] https://github.com/Glimesh/broadcast-box
This is really interesting! Any latency benchmarks comparing it to gstreamer UDP, P2P primarily?
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OBS Studio 30.0
This release includes WebRTC support. I would really appreciate if people could try it! I have so many things I want to improve with broadcasting.
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* Serverless Streaming - WebRTC is P2P so you can video right into your browser. You don’t have to stand up a server anymore to stream for a small audience.
* Sub-Second Latency - Create content and interact with viewers instantly. There is something magical about having a real conversation with your viewers.
* Multitrack Input - Upload your transcodes instead of generating then server side. Give viewers multiple video tracks to see action from all sides.
* Mobility - WebRTC lets you switch networks at any time. Go from WiFi -> Mobile with zero interruptions.
* E2E Authentication - WebRTC lets you add authentication on the sender and receiver. Broadcasters could sign their video streams so viewers can be sure it wasn’t modified.
Try it out with Broadcast Box. A reference server implementation. https://github.com/glimesh/broadcast-box
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Show HN: AV1 Broadcasting from OBS
This is a PR (with builds) to add AV1 support to the WebRTC output from OBS.
We also have been working on a reference server implementation with https://github.com/Glimesh/broadcast-box/ to make it easy to self host.
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Show HN: OBS with Simulcast Support
Simulcast builds of OBS are available. Simulcast is when a client generates multiple encodes of a video. Testers and feedback would be greatly appreciated!
Design Discussion: https://github.com/obsproject/rfcs/pull/55
Use it against your favorite WHIP ingest. If you don't have one check out https://github.com/glimesh/broadcast-box. It is a reference implementation of a WHIP/WHEP server.
I hope/believe the impact of this will be much greater then WebRTC. This could benefit all broadcast workflows.
* Higher Quality Videos - Decoding and re-encoding causes quality loss. Generating all the renditions from the source video is going to be a big improvement for viewers.
* Lower Latency - Removing the additional encoding/decoding allows video to be delivered faster.
* Reduce complexity - With Simulcast setting up a streaming server becomes dramatically easier. It is much easier for an individual to afford and manage a single broadcast server. RTMP -> Transcode -> HLS gets complicated and expensive quickly.
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OBS merges WebRTC support
One of the devs commented on HN that you can use Broadcast Box handle the initial connection. That's open source so you can self-host that too
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OBS Merges WebRTC Support
Try it out today with Broadcast Box. A reference server implementation. https://github.com/glimesh/broadcast-box. The PR to add WebRTC to OBS https://github.com/obsproject/obs-studio/pull/7926
I don't have an idea on how to do cascading OBS instances. I should write some software to make this easy though. I want to make 'self hosting' as easy as possible. I think https://github.com/glimesh/broadcast-box will fill that gap for now.
For my own usage I run WebRTC servers and have them 'replicate' the video traffic. LiveKit did a write up on this with some good graphics https://blog.livekit.io/scaling-webrtc-with-distributed-mesh...
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WebRTC support being added to FFmpeg
I am really excited for this PR to land. Being able to use WebRTC for Broadcast and Playback is going to be huge.
Instead of doing fixed interval keyframe we can use receiver feedback (Massive reduction in bandwidth)
Instead of server side generated transcodes we can use Simulcast. Will be better quality AND massively reduced server load.
If anyone wants to use with Pion check out https://github.com/Glimesh/broadcast-box
Then run `ffmpeg -re -f lavfi -i testsrc=s=1280x720:r=30 -f lavfi -i sine=f=440:b=4 -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -r 25 -g 50 -acodec libopus -ar 48000 -ac 2 -f rtc -authorization "STREAM_NAME" "http://localhost:8080/whip"
What are some alternatives?
web-codecs - WebCodecs is a flexible web API for encoding and decoding audio and video.
go2rtc - Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.
ws-tcp-proxy - Simple websocket tcp proxy.
str0m - A synchronous sans I/O WebRTC implementation in Rust.
offline-browser-communication - Demonstration of a browser connecting to Pion WebRTC without a signaling server.
ffmpeg - FFmpeg Zig package
webrtc-rtptransport - Repository for the RTPTransport specification of the WebRTC Working Group
webrtc-echoes - Simple useful interoperability tests for WebRTC libraries. If you are a WebRTC library developer we'd love to include you!
direct-sockets - Direct Sockets API for the web platform
rfcs - RFCs for changes to OBS Studio (and supporting toolset)
ebook-diffuser - An end to end, customizable, ebook automation tool
obs-studio - OBS Studio - Free and open source software for live streaming and screen recording