OpenSIPS/opensips
OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. observed · 2026-08-28
Health v2 · maintenance only
77/100
- Activity 99
- Release rhythm 35
- Longevity 100
Flags: no_releases no_license
How is this computed?
round(0.45*activity + 0.35*rhythm + 0.20*longevity); archived -> min(score, 10) — computed 2026-09-02. Adoption (stars, forks) is never an input.
- gap_med: n/a
- age_days: 4864
- days_rel: n/a
- days_push: 7
- n_releases_24m: 0
Adoption not part of the score
1516 stars · 672 forks observed · 2026-08-28
What it is AI-extracted, prompt v1, taxonomy v1, 2026-08-30, confidence not recorded
OpenSIPS is a GPL-licensed, open-source SIP server/proxy implementation written in C, forked from the SER project. It provides carrier-grade SIP signaling for voice, video, IM, and presence services, with a modular architecture of 120+ modules and a flexible routing scripting language.
Use cases
- run a SIP proxy or registrar for a VoIP platform
- build a Class 4/5 residential or wholesale trunking platform
- deploy a session border controller (SBC)
- load balance SIP traffic across media or app servers
- implement least-cost routing and failover for calls
- set up a virtual PBX or enterprise telephony backend
- handle NAT traversal for SIP and RTP traffic
- build a presence or IM signaling server
When to choose
- you need a high-throughput, carrier-grade SIP server (tens of thousands of calls per second)
- you want flexible, scriptable SIP routing logic with many integration modules
- you are building telecom/ITSP infrastructure like trunking, SBCs, or PBX front-ends
- you need support for UDP/TCP/TLS/SCTP transports, IPv4/IPv6, and AAA via RADIUS/DIAMETER
When to avoid
- you need a simple end-user softphone or PBX GUI rather than a signaling server
- your project is small-scale and a simpler PBX like Asterisk or FreeSWITCH media handling suffices
- you are not comfortable writing SIP routing scripts or managing telecom infrastructure
- you need WebRTC media processing out of the box without additional components
Facets
service · maturity stable
http-server routing middleware auth load-testing monitoring networking self-hosted backend cross-platform c sip-server voip telephony sip-proxy carrier-grade rtp presence load-balancer session-border-controller telecom linux
10 sources
- readme: https://github.com/OpenSIPS/opensips · fetched 2026-08-28 · 61a67f56cbfc
- homepage: https://opensips.org · fetched 2026-08-29 · 291d6791b193
- site_page: https://www.opensips.org/Documentation/Manual · fetched 2026-08-29 · 886cf44034c2
- site_page: https://opensips.org/About/About · fetched 2026-08-29 · da73c5da1b57
- site_page: https://opensips.org/Documentation/Manuals · fetched 2026-08-29 · 886cf44034c2
- site_page: https://blog.opensips.org/2026/07/28/the-re-design-of-the-opensips-documentation · fetched 2026-08-29 · 804aedb22624
- site_page: https://www.opensips.org/About/Version-Overview-4-0-0 · fetched 2026-08-29 · 021c4dc7dd41
- site_page: https://www.opensips.org/About/WhoIsUsing · fetched 2026-08-29 · 985dbc152b67
- site_page: https://www.opensips.org/About/Features · fetched 2026-08-29 · 494e2fbd4a0b
- site_page: https://www.opensips.org/About/PerformanceTests · fetched 2026-08-29 · e597df47d7ee
Member repositories
| Repository | Role | Health v2 |
|---|---|---|
| OpenSIPS/opensips | main | 77 |
For agents
markdown · JSON · MCP: product_card(name="OpenSIPS/opensips")
Data as of 2026-08-30T08:39:29.467469+00:00 · Report a problem