rtp
Signal-Calling-Service
rtp | Signal-Calling-Service | |
---|---|---|
1 | 4 | |
324 | 411 | |
1.2% | 0.2% | |
7.4 | 8.6 | |
5 days ago | 8 days ago | |
Go | Rust | |
MIT License | GNU Affero General Public License v3.0 |
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rtp
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...
What are some alternatives?
rtsp-simple-server - Also known as rtsp-simple-server. ready-to-use RTSP / RTMP / LL-HLS / WebRTC server and proxy that allows to read, publish and proxy video and audio streams. [Moved to: https://github.com/aler9/mediamtx]
galene - The Galène videoconference server
pyrite - Pyrite is a web(RTC) client & management interface for Galène SFU
Pion WebRTC - Pure Go implementation of the WebRTC API
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).
livekit-server - Scalable, high-performance WebRTC SFU. SDKs in JavaScript, React, React Native, Flutter, Swift, Kotlin, Unity/C#, Go, Ruby and Node. [Moved to: https://github.com/livekit/livekit]
azure-ubuntu-jitsi - A private Jitsi videoconferencing set up on Azure
ion - Real-Distributed RTC System by pure Go and Flutter
NHLGames - Watch NHL.tv, official NHL games streams in HD at 60fps, for free with the NHLGames application on Windows.
mediadevices - Go implementation of the MediaDevices API.
quicktime_video_hack - Record iOS device audio and video