Here’s a comprehensive, curated overview of open source SIP projects (Session Initiation Protocol for VoIP, video, messaging, etc.).Exactly 50 active, high-quality, distinct projects with substantial features is difficult — the ecosystem is dominated by a few mature projects, with many smaller/niche/language-specific implementations, forks, and legacy tools on GitHub. I’ve compiled the most relevant and actively maintained or historically significant ones (roughly 35–40 core projects), grouped by category.I provide detailed features and usage for the major ones and shorter notes for others.
Published by James.zhu @https://www.linkedin.com/in/james-zhu-siplab/
1-SIP agents/user agents
All projects below are open source (mostly GPL, BSD, MIT, or Apache).1. SIP Stacks & Libraries (for building Agents, UAs, Clients, and custom applications)These are the core building blocks for SIP agents/user agents.
| # | Project | Language | Key Features | Typical Usage |
|---|---|---|---|---|
| 1 | PJSIP (pjproject) | C | Full multimedia stack (SIP + audio/video codecs, NAT traversal with ICE/STUN/TURN, TLS, SRTP, presence). Highly portable (embedded to mobile/desktop). Includes pjsua CLI UA. | Building custom VoIP apps, softphones, gateways, mobile clients. Most popular general-purpose stack. |
| 2 | Sofia-SIP | C | Lightweight, RFC 3261 compliant SIP UA library. Excellent performance and standards compliance. | Core of FreeSWITCH; custom clients/servers. |
| 3 | reSIProcate | C++ | Modular, carrier-grade stack. Strong security, scalability, presence, IM. Includes repro proxy. | High-volume carrier apps, secure proxies. |
| 4 | belle-sip | C (OO API) | Modern transport/transaction/dialog layers. Clean object-oriented design. | Foundation of liblinphone and Linphone ecosystem. |
| 5 | oSIP + eXosip | C | Lightweight core (oSIP) + higher-level UA abstraction (eXosip). | Simple to medium-complexity C-based UAs. |
| 6 | libre (creytiv/re or baresip/re) | C | Complete SIP + SDP + RTP/RTCP + SRTP + STUN/TURN/ICE + DNS. Very portable. | Foundation for Baresip; lightweight RTC apps. |
| 7 | Baresip | C (modular) | Full modular SIP UA on top of libre. Audio + video support, many plugins. | Lightweight embedded/desktop softphones or as library. Extremely portable. |
| 8 | MjSIP | Java | Complete SIP stack (UA, proxy, registrar capabilities). | Java-based clients and servers. |
| 9 | JAIN-SIP (NIST jsip) | Java | Reference implementation of JAIN SIP API (JSR 32). | Enterprise Java SIP applications. |
| 10 | SIP.js | JavaScript | Modern SIP stack optimized for WebRTC. Easy browser integration. | Web-based VoIP/video clients in browsers. |
| 11 | JsSIP | JavaScript | Lightweight SIP library (often used with WebRTC). | Web phones and browser VoIP. |
| 12 | SIPSorcery | C#/.NET | Full SIP + WebRTC + RTP + ICE/STUN. Rich examples. | .NET/Windows VoIP applications and servers. |
| 13 | sipgo | Go | Fast parsing, RFC 3261/3581/6026 compliant. Optimized for performance. | High-performance Go-based SIP services/servers. |
| 14 | python-sipsimple | Python | SIP SIMPLE implementation (audio, video, IM, file transfer). | Python SIP endpoints and rich-media apps. |
| 15 | ersip | Erlang | Building blocks for SIP applications. | Erlang/OTP-based telecom systems. |
| 16+ | Others (niche/language-specific) | Various | Rust (rsipstack, libsip, sipcore, ezk), Go (gossip, gosip variants), PHP (php-sip/RTCKit), Python (pyVoIP), Dart (dart-sip-ua), Elixir (elixir-sippet), etc. | Language-specific or lightweight custom implementations. |
2-SIP Servers, Proxies, Registrars & B2BUAsThese handle routing, registration, proxying, and advanced call control.
| # | Project | Key Features | Typical Usage |
|---|---|---|---|
| 17 | Kamailio (formerly OpenSER/SER) | Extremely scalable (thousands of CPS), 100+ modules (registrar, proxy, presence, load balancing, security, WebSocket, etc.). KEMI scripting (Lua, Python, JS, Ruby). | Carrier-grade platforms, large enterprises, SIP proxies, SBCs. Most popular high-scale open source SIP server. |
| 18 | OpenSIPS | Fork of Kamailio. Strong native B2BUA, integrated routing/dialplan features, high performance. | Similar to Kamailio but preferred when built-in B2BUA or certain modules are needed. |
| 19 | Asterisk | Full PBX with powerful SIP channel driver + dialplan. IVR, conferencing, queues, many protocols/codecs. | Complete telephony systems, call centers, PBXs. Not a pure SIP server but excellent SIP support. |
| 20 | FreeSWITCH | Modular softswitch. Uses Sofia-SIP. Powerful media handling, scripting (Lua/JS), high concurrency, XML config. | VoIP platforms, conferencing servers, IVR, media-heavy applications. |
| 21 | Flexisip | SIP server suite (proxy + presence + conference + B2BUA). Good push notification support. | Unified communications platforms (pairs well with Linphone). |
| 22 | repro (reSIProcate) | SIP proxy/registrar with built-in web UI. Federated VoIP support. | Quick-setup proxies and small-to-medium deployments. |
| 23 | Sippy B2BUA (Python + Go versions) | Full Back-to-Back User Agent with advanced call control and media handling. | Wholesale routing, advanced B2BUA scenarios. |
| 24 | Routr | Lightweight Node.js SIP proxy + location server + registrar. | Simple/microservice deployments, quick prototypes. |
| 25 | Siproxd | SIP + RTP proxy focused on NAT traversal. | Small setups behind NAT/firewalls. |
| 26 | Yeti Switch | High-performance SBC with routing, billing, media proxy/transcoding, LNP support. | Wholesale VoIP carriers. |
| 27+ | Others | GNU SIP Witch, Yate, sipXecs/sipX, Clearwater (IMS components), Drachtio, nksip (Erlang framework), OpenSBC, Partysip, OpenJSIP, MySIPSwitch, SaRP, VOCAL. | Various legacy or niche proxies/registrars/SBCs. |
3-Notable Complete SIP Clients / Softphones / SDKs (Agents)These are ready-to-use agents built on the stacks above.
- Linphone + liblinphone — Full-featured cross-platform client + SDK (uses belle-sip). Excellent for custom development.
- Jitsi — Java-based client with strong encryption (ZRTP).
- Blink — Feature-rich (chat, file transfer, conferencing via MSRP).
- Baresip — Lightweight modular client (CLI or with GUI wrappers).
- PJSUA — Command-line UA from PJSIP.
- Web-based: sipML5, SIP.js-based webphones.
- Mobile: SIPdroid, Lumicall (Android, based on various stacks).
- Legacy but notable: Ekiga, Twinkle, SFLphone.
- Testing, Monitoring & Tools
- SIPp — Industry-standard SIP traffic generator/stress tester.
- sipsak — “Swiss army knife” for SIP diagnostics.
- SIPVicious — Security auditing and reconnaissance suite.
- sngrep / sipgrep — Terminal-based SIP flow viewers.
- HOMER / SIP3 — RTC traffic capture and monitoring platforms.
4-Quick Recommendations by Use Case
| Use Case | Recommended Projects |
|---|---|
| Lightweight/Embedded Agent | Baresip + libre or PJSIP |
| Web/Browser Client | SIP.js or JsSIP |
| .NET / Windows App | SIPSorcery |
| High-scale SIP Server/Proxy | Kamailio or OpenSIPS |
| Full PBX / Call Center | Asterisk or FreeSWITCH |
| Modern SDK + Client | liblinphone (belle-sip) |
| B2BUA / Advanced Routing | Sippy B2BUA or OpenSIPS/Kamailio modules |
| Go / High-performance services | sipgo |
Important Notes:
- Most projects are mature but check GitHub activity (stars, recent commits) before adopting.
- Licenses vary (check compatibility with your project).
- For production, combine a stack (e.g., PJSIP/Baresip) with a server (Kamailio/FreeSWITCH).
- Many projects support TLS, SRTP, WebSocket, IPv6, and NAT traversal out of the box.
Source:
- miconda/sip-resources (excellent categorized list)
- rtckit/awesome-rtc
- GitHub topics: sip-stack, sip-server, sip-proxy, sip-client
- Wikipedia: List of SIP software
