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 SIP Open-Source Projects
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jitsi
Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.
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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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Kamailio
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
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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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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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libresbc
An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.
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atm0s-media-server
Decentralized, Global-Scale Media Server written in Rust (WebRTC/Whip/Whep/Rtmp/Sip)
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SaaSHub
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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
Which clients do you use? And how do you connect to a SIP trunk / DID without Asterisk?
I use WebRTC with Asterisk, and Browser Phone for the client (https://github.com/InnovateAsterisk/Browser-Phone). I don't use it much, but good enough for the rare times I have to use the phone.
Try starting with just the Basic Example at: https://github.com/tayler6000/pyVoIP
We're thrilled to share the vision behind our open source decentralized streaming project, which is currently in its alpha stages of development. Our primary objective is to build a stable and reliable foundation that will serve as the cornerstone for future growth.
Here's a snapshot of our ambitious goals:
1. Cluster: Create a decentralized media server cluster with multiple zones. Our commitment is to support a broad range of SDKs and mainstream protocols like RTMP, SIP, and SRT to ensure seamless compatibility.
2. Marketplace: Develop a sharing marketplace that will enable resource sharing and monetization. This marketplace platform will not only help our partner scale during high-demand periods but also reduce costs during lulls. With a long vision, the revenue generated from this marketplace service fees will be instrumental in funding the project's development.
3. P2P Network: Our long-term goal is to establish a peer-to-peer network where servers act as fallbacks, significantly cutting infrastructure costs and paving the way for limitless scalability.
Currently, our focus is laser-sharp on achieving Goal 1. To stay updated on our progress details of our project roadmap, please visit our github page or follow our updates at 8xFF Foundation page.
We're excited to embark on this journey and can't wait to share our progress with you. Let's build the future of media servers with us!
More details about our project https://github.com/8xFF/atm0s-media-server
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).
SIP related posts
- FreePBX – Open-Source PBX (PHP GUI for Asterisk)
- Building a VoIP Network with Routr on DigitalOcean Kubernetes: Part I
- Browser-to-Browser calling with SIP.js and Routr
- What's the use of fd passing?
- Hi, anyone used PJSIP for P2P connectivity (ICE)
- Baresip – An Open Source modular SIP User-Agent with audio and video support
- Baresip – An Open Source modular SIP User-Agent with audio and video support
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A note from our sponsor - WorkOS
workos.com | 26 Apr 2024
Index
What are some of the best open-source SIP projects? This list will help you:
| Project | Stars | |
|---|---|---|
| 1 | ejabberd | 5,916 |
| 2 | jitsi | 4,042 |
| 3 | flutter-webrtc | 3,953 |
| 4 | freeswitch | 2,981 |
| 5 | Kamailio | 2,128 |
| 6 | Asterisk | 1,944 |
| 7 | pjproject | 1,832 |
| 8 | SIP.js | 1,821 |
| 9 | mirotalksfu | 1,819 |
| 10 | baresip | 1,611 |
| 11 | Homer | 1,511 |
| 12 | Routr | 1,328 |
| 13 | SIPSorcery | 1,309 |
| 14 | linphone-android | 1,073 |
| 15 | sngrep | 940 |
| 16 | sipvicious | 826 |
| 17 | stun | 562 |
| 18 | Browser-Phone | 431 |
| 19 | libjuice | 363 |
| 20 | soup | 335 |
| 21 | libresbc | 325 |
| 22 | pyVoIP | 189 |
| 23 | atm0s-media-server | 169 |
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