The APIs are flexible and easy-to-use, supporting authentication, user identity, and complex enterprise features like SSO and SCIM provisioning. Learn more →
Top 23 Rtp Open-Source Projects
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mediamtx
Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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InfluxDB
Power Real-Time Data Analytics at Scale. Get real-time insights from all types of time series data with InfluxDB. Ingest, query, and analyze billions of data points in real-time with unbounded cardinality.
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go2rtc
Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.
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lal
🔥 Golang audio/video live streaming library/client/server. support RTMP, RTSP(RTP/RTCP), HLS, HTTP[S]/WebSocket-FLV/TS, GB28181, H264/H265/AAC/G711, relay, cluster, record, HTTP Notify/API/UI. 直播
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RootEncoder
RootEncoder for Android (rtmp-rtsp-stream-client-java) is a stream encoder to push video/audio to media servers using protocols RTMP, RTSP, SRT and UDP with all code written in Java/Kotlin
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WorkOS
The modern identity platform for B2B SaaS. The APIs are flexible and easy-to-use, supporting authentication, user identity, and complex enterprise features like SSO and SCIM provisioning.
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SIPSorcery
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
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webrtc-nuts-and-bolts
A holistic way of understanding how WebRTC and its protocols run in practice, with code and detailed documentation.
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RtspClientSharp
Pure C# RTSP client for .NET Standard without external dependencies and with true async nature. I welcome contributions.
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werift-webrtc
WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP/SRTP/WEBM/MP4
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rings
Rings is a structured peer-to-peer network implementation using WebRTC, Chord DHT, and full WebAssembly (WASM) support.
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SaaSHub
SaaSHub - Software Alternatives and Reviews. SaaSHub helps you find the best software and product alternatives
Project mention: Bad audio quality when re-encoding RTSP stream from AAC to Opus in MediaMTX | /r/ffmpeg | 2023-12-09I started a discussion on the MediaMTX Github repo, where you can find more information on my problem: https://github.com/bluenviron/mediamtx/discussions/2787
Not my project but I have had great success with https://github.com/gtxaspec/wz_mini_hacks & V3 model.
The V3 models need to be downgraded to a specific firmware first and patching it exposes RSTP streams using https://github.com/AlexxIT/go2rtc. Everything doable without ever installing Wyze app on an environment air gapped environment with no internet.
Project mention: [OpenSource] I am building high performance Plex alternative in Go for Movies and TV Show | /r/golang | 2023-06-02
Hello, I'm in the process of developing a multiplayer FPS game and recently delved into ICE connectivity (STUN/TURN). Currently, my setup involves a custom matchmaking server in C++, with UDP port handling on the client side through UPnP or a fallback custom relay server. While the current approach works well, I'm exploring options to simplify the project by incorporating existing technologies. I've come across Libjuice and Libpjsip for NAT traversal. Libjuice offers a nice and simple API, but it supports only one person. Hence, I'm considering Libpjsip. I came across their ICE demo script at https://github.com/pjsip/pjproject/blob/master/pjsip-apps/src/samples/icedemo.c and I'm curious about its performance, particularly the pj_ice_strans_sendto2 function. I'm keen to understand how it compares to my current implementation with Berkeley Sockets and whether Libpjsip is a suitable choice for multiplayer P2P games. Any insights or assistance would be highly appreciated. Thanks!
Project mention: Show HN: WFB-ng – long range high speed link for drones and robotics | news.ycombinator.com | 2024-03-04
Project mention: Show HN: LinuxDAW – Quality audio software for Linux | news.ycombinator.com | 2023-12-30> Aaand, this actually makes me want one of those cards myself!
heh, sorry!
Looks like I probably only searched for Dante virtual-soundcard before, when I searched for AES67 virtual soundcard I got [1] and [2] which may be the answer to my problems (if my machine can handle that many audio streams, but it should be able to with 64 cores haha!).
Looks like that's going to be my project tomorrow.
[1] https://github.com/bondagit/aes67-linux-daemon
[2] https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/AES...
Project mention: Show HN: Bring phone calls into the browser (sip-to-WebRTC) | news.ycombinator.com | 2024-01-04
Try starting with just the Basic Example at: https://github.com/tayler6000/pyVoIP
Project mention: Show HN: Rings Network, A P2P network can running inside browser | news.ycombinator.com | 2023-12-12
NOTE:
The open source projects on this list are ordered by number of github stars.
The number of mentions indicates repo mentiontions in the last 12 Months or
since we started tracking (Dec 2020).
The latest post mention was on 2024-04-19.
Rtp related posts
- VoRS: Vo(IP) Simple Alternative to Mumble
- Pure C WebRTC
- Building WebRTC server implementation for Erlang
- Bad audio quality when re-encoding RTSP stream from AAC to Opus in MediaMTX
- Hi, anyone used PJSIP for P2P connectivity (ICE)
- Ask HN: Does this USB audio device exist?
- MediaMTX: Zero-dependency server for live video and audio streams
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A note from our sponsor - WorkOS
workos.com | 19 Apr 2024
Index
What are some of the best open-source Rtp projects? This list will help you:
| Project | Stars | |
|---|---|---|
| 1 | Pion WebRTC | 12,633 |
| 2 | mediamtx | 9,856 |
| 3 | libstreaming | 3,411 |
| 4 | go2rtc | 3,185 |
| 5 | lal | 2,613 |
| 6 | RootEncoder | 2,345 |
| 7 | pjproject | 1,824 |
| 8 | OpenHD | 1,793 |
| 9 | SIPSorcery | 1,300 |
| 10 | libsrtp | 1,171 |
| 11 | webrtc-nuts-and-bolts | 890 |
| 12 | wfb-ng | 835 |
| 13 | RtspClientSharp | 669 |
| 14 | gortsplib | 604 |
| 15 | mediadevices | 504 |
| 16 | werift-webrtc | 415 |
| 17 | rtp | 325 |
| 18 | aes67-linux-daemon | 314 |
| 19 | Arduino-AppleMIDI-Library | 290 |
| 20 | ex_webrtc | 211 |
| 21 | pyVoIP | 188 |
| 22 | rings | 180 |
| 23 | WireBug | 167 |
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