Yate

Yate, which stands for Yet Another Telephony Engine, is an open source telephony engine written in C++ by Null Team, which launched the project’s website in 2004. It can power a VoIP server, a gateway to the phone network or an SS7 switch, and it is the foundation of products such as YateBTS, aimed at mobile network base stations, and YateUCN, dedicated to the core network.

The code is released under the GPL version 2, with exceptions that allow linking with proprietary software. Despite a website that is rarely refreshed, the project has not been abandoned. The yatevoip/yate repository on GitHub still receives changes, the latest dated 15 September 2026. No new packaged release has appeared for a long time, though.

What Yate is for

Yate is built on a simple idea: a central engine passes messages between modules that do not talk to each other directly. A protocol module announces an incoming call, a routing module picks the destination, a third module sets up the outgoing leg, and each one reacts only to the messages it cares about. This loose coupling makes it possible to combine very different technologies in a single instance and to add logic without touching the core, including from external scripts in PHP, Python, Perl, JavaScript or the shell.

Protocol coverage is its main strength. The documentation lists SIP, with router, session border controller and registrar functions, along with H.323, IAX, Jabber and Jingle, MGCP, SS7 signaling and ISDN. That range makes it a strong candidate when networks from different eras must be joined, such as a modern SIP setup and an ageing H.323 system or a carrier SS7 link.

Documented uses range from VoIP server and client to gateways toward the switched network, conference servers, interactive voice response, call centers, prepaid card systems and call recording. According to the history page in its documentation, the German provider sipgate announced in 2011 that it had chosen Yate for its core infrastructure, and the hardware maker Sangoma backed its Windows port as early as 2005.

Yate therefore sits in the same niche as FreeSWITCH, a full engine handling both signaling and media, but with a smaller community and a stronger lean toward legacy and mobile telephony. Its purpose differs from Kamailio and OpenSIPS, which stick to SIP signaling.

Who is it for?

Yate appeals to telecom engineers who need to interconnect mixed technologies, operators working with SS7, researchers and mobile network hobbyists drawn to YateBTS, and developers who like its message-based architecture for writing their own telephony services.

Its limits are clear. The home page of the official wiki has not changed since July 2018 and the download page since March 2023, while the website still presents 6.0.0 as the stable release even though the files on offer are 6.1.0 and the source tree now reports 6.4.1. Development continues, but at a pace of a few changes a month, mostly from a single Null Team developer. For a new project with no need for SS7 or H.323, FreeSWITCH is the safer option, and Kamailio or OpenSIPS are better suited to a high-traffic SIP front end.

Download Yate safely

The official addresses are yate.ro, docs.yate.ro and the yatevoip/yate repository on GitHub, which the website’s Git page names as the reference. The wiki’s download page offers a source tarball and a source RPM for Linux, an installer and a ZIP archive for Windows, packages for Mageia, and mentions a FreeBSD port. All of these files are version 6.1.0. To benefit from recent fixes, the most reliable route is to clone the GitHub repository and build with the usual configure, make and make install steps.

The engine is free, with no paid edition. The Ubuntu repositories listed on the wiki are run by community members rather than Null Team, and some only target very old Ubuntu releases. Since the website also serves its binaries over unencrypted HTTP, it is better to rely on the code from the official repository than on executables copied to other sites.

Frequently asked questions