Signal-Calling-Service
galene
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Signal-Calling-Service | galene | |
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4 | 36 | |
410 | 859 | |
0.5% | - | |
8.6 | 8.4 | |
about 1 month ago | 12 days ago | |
Rust | Go | |
GNU Affero General Public License v3.0 | MIT License |
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.
Signal-Calling-Service
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Is async runtime (Tokio) overhead significant for a "real-time" video stream server?
I am npt sure if this is related but Signal built Signal Calling Service and according to them it worked great.
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Pyrite – open-source video conferencing
I was curious and looked through the code of Galene briefly and found the following, which may answer your question. For context, I am familiar with the Jitsi code and have written my own calling server (and written about it: https://signal.org/blog/how-to-build-encrypted-group-calls/).
Galene appears to be less mature than Jitsi. For example, it uses REMB feedback messages from the client to calculate allowable bitrates rather than calculating the bitrates itself (as Jitsi and Signal's SFU do). Worse, it appears that what it does with that information is erroneous. I could be wrong, but it looks like the bitrate allocation code (see https://github.com/jech/galene/blob/e8fbfcb9ba532f733405b1c5...) only allocates the bitrate for one of the video streams, not all of them. Perhaps the author did not realize that there is one REMB sent back for all the video streams by WebRTC rather than one per stream (see, for example, here: https://source.chromium.org/chromium/chromium/src/+/main:thi...). Further, I find the spatial layer switching code to be strange. For examples, it doesn't go down a layer unless it's 150% over the estimated allowable bitrate, which gives a lot of opportunity for inducing latency.
In short, I think Galene has a ways to go before it works as well as Jitsi (Videobridge), and thus Pyrite group calls are unlikely to work as well as Jitsi group calls (for 1:1 calls, I don't know; I didn't look into that).
Oh, and just a reminder, the SFU we use for Signal group calls is also open source: https://github.com/signalapp/Signal-Calling-Service.
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How to build large-scale end-to-end encrypted group video calls
And yeah, it uses Signal-Calling-Service written on Rust.
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An Introduction to WebRTC Simulcast
That's a well written article covering the basics of simulcast.
If you're interested in seeing an implementation of an SFU doing simulcast forwarding written in Rust, we (at Signal) recently open sourced our SFU:
https://github.com/signalapp/Signal-Calling-Service/blob/mai...
galene
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livekit-server VS galene - a user suggested alternative
2 projects | 28 Mar 2024
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Show HN: New Jitsi WebRTC Alternative: ChatGPT, File Transfer, Docker
I would like to recommend Galene: https://github.com/jech/galene
Runs in my raspberry pi, a single small executable, like in the old good times.
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Zoom terms now allow training AI on user content with no opt out
> Do you happen to know of others by any chance.
There's Galene, <https://galene.org>. It's easy to deploy, uses minimal server resources, and the server is pretty solid. The client interface is still a little awkward, though. (Full disclosure, I'm the main author.)
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Best voice and video chat?
galene - basically selfhosted zoom/jitsi
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Ask HN: FFmpeg real-time desktop streaming
What latency are you trying to do? Will the professor being communicating with the students while doing this? Will the students all have the same bandwidth, or will you want multiple renditions (low, med, high quality levels)?
If you want AV1 you will not be able to use RTMP. The protocol is orphaned/deprecated, so avoid if possible!
If I was building it this is what I would do, and my reasoning.
* For capture + encoding I would use OBS. You will want to use something that is easy for users to install configure. Professors will also have lots of custom requirements when it comes to layout etc... it will be tempting to do a ffmpeg command directly, but it will fall apart quick I believe.
* To get AV1 out of OBS I would use FFMPEG output. I would have it send RTP. RTP is used to carry video in a sub-second manner. This is the same protocol that WebRTC uses. You know have AV1 + low latency.
* Then for users to watch I would use WebRTC. That will allow them to watch in their web browser. Conceptually it will be like this https://github.com/pion/webrtc/tree/master/examples/rtp-to-w... this takes the RTP packets and puts them in the browser.
Lots of great projects exist that you could use for 'RTP -> WebRTC' like https://galene.org/ and https://livekit.io/ I would suggest checking them all out!
If you have more questions/want to talk to people in the video space always happy to chat on https://pion.ly/slack :)
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Voice/Video call for Iranians
galene
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Self-hosted chat app with chat/video?
The most lightweight all-inclusive central solution for video conferences I know is Galene. It runs in under 200 MB RAM.
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What are good self-hosted WebRTC video solutions today?
Even though the default UI is extremely simplistic, I very much like galene. It bundles all the components you need in a single binary. Even a TURN server so you don't have to fiddle with coturn. Not to mention that it's very resource efficient.
- Galène. FOSS Videoconference Server
- Galène Videoconference Server
What are some alternatives?
rtp - A Go implementation of RTP
Jitsi Meet - Jitsi Meet - Secure, Simple and Scalable Video Conferences that you use as a standalone app or embed in your web application.
pyrite - Pyrite is a web(RTC) client & management interface for Galène SFU
janus-gateway - Janus WebRTC Server
Jitsi Video Bridge - Jitsi Videobridge is a WebRTC compatible video router or SFU that lets build highly scalable video conferencing infrastructure (i.e., up to hundreds of conferences per server).
azure-ubuntu-jitsi - A private Jitsi videoconferencing set up on Azure
galene_ynh - Galène package for YunoHost
mirotalk - 🚀 WebRTC - P2P - Simple, Secure, Fast Real-Time Video Conferences Up to 4k and 60fps, compatible with all browsers and platforms.
wirow-server - A full featured self-hosted video web-conferencing platform.
room.cafe - An extremely simple video meeting, integrated whiteboard, chat and screen sharing
mirotalksfu - 🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.