SIPSorcery
webrtc-for-the-curious
SIPSorcery | webrtc-for-the-curious | |
---|---|---|
9 | 13 | |
1,309 | 1,881 | |
1.5% | 0.5% | |
8.5 | 5.2 | |
8 days ago | 6 months ago | |
C# | Python | |
GNU General Public License v3.0 or later | Creative Commons Zero v1.0 Universal |
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.
SIPSorcery
- 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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Why WebRTC is not able to capture screen when the System is signed out or logged out?
I checked another example in Sipsorcery and it works. In this example it show a basic test pattern. For screen sharing Sipsorcery uses FFMPEG. I think FFMPEG has the restriction to capture the screen when the system is signed out.
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Need help to connect to WebSocket signaling for WebRTC, C#
I was going through FFmpegFileAndDevicesTest, it is created using Sipsorcery library. In this code a websocket server is created and the PeerConnection is added to the websocket. Is there any way to connect to an external websocket server instead of creating one. The code is
- Need some help on C# WebRTC Signaling using WebSocket
- Show HN: WebRTC-Echoes: Interop for C#, C++, Python, TypeScript, Go and Servers
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?
Pion WebRTC - Pure Go implementation of the WebRTC API
web-codecs - WebCodecs is a flexible web API for encoding and decoding audio and video.
libdatachannel - C/C++ WebRTC network library featuring Data Channels, Media Transport, and WebSockets
broadcast-box - A broadcast, in a box.
aiortc - WebRTC and ORTC implementation for Python using asyncio
ws-tcp-proxy - Simple websocket tcp proxy.
janus-gateway - Janus WebRTC Server
offline-browser-communication - Demonstration of a browser connecting to Pion WebRTC without a signaling server.
webrtc-echoes - Simple useful interoperability tests for WebRTC libraries. If you are a WebRTC library developer we'd love to include you!
webrtc-rtptransport - Repository for the RTPTransport specification of the WebRTC Working Group
webrtc - A pure Rust implementation of WebRTC
direct-sockets - Direct Sockets API for the web platform