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Top 23 Voip Open-Source Projects
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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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freeswitch
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.
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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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Kamailio
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
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mirotalksfu
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
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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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linphone-android
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
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sipvicious
SIPVicious OSS is a VoIP security testing toolset. It helps security teams, QA and developers test SIP-based VoIP systems and applications. This toolset is useful in simulating VoIP hacking attacks against PBX systems especially through identification, scanning, extension enumeration and password cracking.
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OSS Alternative to Twilio
Project mention: Welcome to mwmbl, the free, open-source and non-profit search engine | news.ycombinator.com | 2023-09-18
Multi-process server applications using a prefork model are one example. You might have one process dedicated to accepting new connections over the network and maybe doing an initial handshake, and then a bunch of worker processes handling the actual connections. Kamailio's CDP module is built like this (https://github.com/kamailio/kamailio/tree/master/src/modules/cdp)
Project mention: FreePBX – Open-Source PBX (PHP GUI for Asterisk) | news.ycombinator.com | 2024-03-11
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!
SimplePhone is a phone built using SIP.js that runs as a Docker container. Please see the documentation at https://sipjs.com/ to develop your implementation.
Mirotalk - [ Star on GitHub ]
Project mention: Baresip – An Open Source modular SIP User-Agent with audio and video support | /r/patient_hackernews | 2023-09-04
Project mention: Building a VoIP Network with Routr on DigitalOcean Kubernetes: Part I | dev.to | 2024-03-04Please see the Official Chart for many more options for your deployment.
Project mention: Issues with Caller Identification in Linphone App - Need Help with Settings | /r/VOIP | 2023-06-05
Project mention: React Native Experts (context: react-native-callkeep) | news.ycombinator.com | 2023-05-05We are trying to get some help on this issue. https://github.com/react-native-webrtc/react-native-callkeep...
Gist: The issue is related to using RNCallKeep library in a React Native app on Android. When a call is initiated while the app is in the background or closed state, the RNCallKeep UI is invoked, but the desired behavior is to open the app's screen and handle the call through the app. The RNCallKeep.backToForeground() function is used to achieve this, but it causes the audio to stop working properly when using Agora to connect VOIP calls. This issue does not occur when handling calls manually and is only happening on Android, not on iOS devices.
Let me know what you guys think. You can check out the github here and experiment with it via the first link.
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).
Voip related posts
- VoRS: Vo(IP) Simple Alternative to Mumble
- Browser-to-Browser calling with SIP.js and Routr
- Show HN: Jitsi alternative – Docker amd64/ARM64 – WebRTC self-hosted
- Pure C WebRTC
- WebRTC for the Curious
- Building WebRTC server implementation for Erlang
- What's the use of fd passing?
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A note from our sponsor - SaaSHub
www.saashub.com | 26 Apr 2024
Index
What are some of the best open-source Voip projects? This list will help you:
| Project | Stars | |
|---|---|---|
| 1 | Pion WebRTC | 12,664 |
| 2 | fonoster | 6,089 |
| 3 | Mumble | 5,966 |
| 4 | ejabberd | 5,916 |
| 5 | flutter-webrtc | 3,953 |
| 6 | webrtc | 3,792 |
| 7 | freeswitch | 2,981 |
| 8 | MadelineProto | 2,668 |
| 9 | Kamailio | 2,128 |
| 10 | Asterisk | 1,944 |
| 11 | pjproject | 1,832 |
| 12 | SIP.js | 1,821 |
| 13 | mirotalksfu | 1,819 |
| 14 | baresip | 1,611 |
| 15 | Homer | 1,511 |
| 16 | Routr | 1,328 |
| 17 | SIPSorcery | 1,309 |
| 18 | linphone-android | 1,073 |
| 19 | react-native-callkeep | 868 |
| 20 | sipvicious | 826 |
| 21 | stun | 562 |
| 22 | mediadevices | 508 |
| 23 | element-call | 504 |
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