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Germán Luis Aracil Boned edited this page Jun 24, 2026 · 33 revisions

GABPBX Wiki

GABPBX is a GPLv2 open source PBX based on the Asterisk architecture and maintained as the Germán Aracil Boned PBX project.

This wiki documents GABPBX as a system: architecture, modules, configuration, operation and implementation notes taken from the source tree.

The current release is GABPBX 1.1.2. It completes chan_sofia as a broad, production-grade, drop-in replacement for chan_sip, and adds two-way WebRTC media — DTLS-SRTP, an ICE-lite STUN responder, rtcp-mux and BUNDLE — so a browser can register, call and be called over secure WebSocket with audio and video, no pjproject and no libnice in the tree.

The headline: chan_sofia

A modern SIP channel driver built on the Sofia-SIP NUA stack. Drop-in for chan_sip (same SIP channel, sip show … CLI, SIPpeers AMI, sippeers realtime), with capabilities chan_sip never had:

  • Registration: SIP Outbound (RFC 5626), Path (RFC 3327), Service-Route (RFC 3608), GRUU.
  • Presence/voicemail: outbound PUBLISH (RFC 3903), generic outbound SUBSCRIBE, an outbound MWI watcher, solicited and unsolicited MWI, presence/BLF dialog-info.
  • Signaling: PRACK/100rel (RFC 3262), in-dialog UPDATE (RFC 3311), Q.850 Reason (RFC 3326), REFER transfer (blind and attended), out-of-dialog MESSAGE.
  • Media: SRTP/SDES (RFC 4568), DTLS-SRTP (RFC 5763/5764), WebRTC audio and two-way video, Opus passthrough, T.38 fax with a state machine.
  • Security: mutual TLS, TLS hardening, SHA-256 digest auth, a local anti-abuse SIP blacklist.
  • Diagnostics: per-call SIP history with verbose call analysis, plus a richer sip … CLI.

Start at Chan-Sofia, then Features for the full catalogue, WebRTC for the browser story, Migrating-from-chan_sip for the swap, and Security for the hardening posture.

What is new in 1.1.2

  • WebRTC media is no longer roadmap. Two WSS browsers can hold a real two-way audio call, with two-way video (VP8/H.264) on top. See WebRTC.
  • Opus relays without transcoding. Three core fixes (codec selection, the 48 kHz clock rate, and the RTP smoother) let a both-Opus call pass through cleanly. See Features → Opus.
  • The full chan_sofia SIP and media catalogue is documented one feature at a time in Features.

First Chapter

  • Chan-Sofia - the Sofia-SIP based SIP channel driver and chan_sip replacement.
  • Features - the complete chan_sofia capability catalogue, one by one.
  • WebRTC - browser SIP over WSS, DTLS-SRTP, ICE-lite, audio and video.
  • Migrating-from-chan_sip - the two-line swap and what carries over.
  • Security - the security and robustness posture.

Current Chapters

Source Policy

Documentation in this wiki follows the source code first. When a behavior is uncertain, the source file and sample configuration are authoritative.

GABPBX 1.4.2

SIP — chan_sofia

Subsystems

Realtime & data

Operations

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