Open-source telephony core

Rooms

1. Four projects, four roles

   Read the role each project states. Do not treat them as four names for one PBX.

2. Asterisk, PJSIP, and the dialplan

   chan_pjsip is the SIP channel driver. res_pjsip reads pjsip.conf. extensions.conf is the dialplan. The license page is GPL version 2, with a stated exception for AMI, AGI, and ARI.

3. FreeSWITCH profiles, dialplan, and WebRTC

   The manual covers SIP profiles, the dialplan, codecs, media, WebRTC, and the Event Socket. There is an open source edition and an enterprise edition.

4. Kamailio as the SIP server

   Kamailio is an open source SIP server under GPLv2+. Its page lists proxy, registrar, WebSocket, and scaling in front of a PBX.

5. rtpengine and the integration map

   rtpengine proxies RTP, rewrites SDP, and can bridge ICE and SRTP. It runs on GNU/Linux and is meant to sit with Kamailio.

Ecosystem

SIP does not replace workforce management, quality monitoring, CRM, emergency policy, or a messaging registry. It is the session seam those systems hang from. Compliance notes on this page are educational and are not legal advice.

Open the map