rtp VS Signal-Calling-Service

Compare rtp vs Signal-Calling-Service and see what are their differences.

Signal-Calling-Service

Forwards media from 1 group call device to N group call devices. (by signalapp)
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rtp Signal-Calling-Service
1 4
325 409
2.5% 0.2%
6.9 8.8
about 12 hours ago about 1 month ago
Go Rust
MIT License GNU Affero General Public License v3.0
The number of mentions indicates the total number of mentions that we've tracked plus the number of user suggested alternatives.
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
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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.

rtp

Posts with mentions or reviews of rtp. We have used some of these posts to build our list of alternatives and similar projects. The last one was on 2021-09-20.

Signal-Calling-Service

Posts with mentions or reviews of Signal-Calling-Service. We have used some of these posts to build our list of alternatives and similar projects. The last one was on 2023-03-08.
  • Is async runtime (Tokio) overhead significant for a "real-time" video stream server?
    5 projects | /r/rust | 8 Mar 2023
    I am npt sure if this is related but Signal built Signal Calling Service and according to them it worked great.
  • Pyrite – open-source video conferencing
    9 projects | news.ycombinator.com | 19 Feb 2022
    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.

  • How to build large-scale end-to-end encrypted group video calls
    1 project | /r/rust | 15 Dec 2021
    And yeah, it uses Signal-Calling-Service written on Rust.
  • An Introduction to WebRTC Simulcast
    3 projects | news.ycombinator.com | 20 Sep 2021
    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?

When comparing rtp and Signal-Calling-Service you can also consider the following projects:

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