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Germán Luis Aracil Boned edited this page Jul 3, 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 1.5.0. It completes chan_sofia as a broad, production-grade, drop-in replacement for chan_sip, and delivers reliable two-way WebRTC media — DTLS-SRTP, an ICE-lite STUN responder, rtcp-mux, BUNDLE and RTCP keyframe feedback — so a browser can register, call, be called, hold and resume over secure WebSocket with audio and video, no pjproject and no libnice in the tree. 1.5.0 adds video bridging between a legacy SIP video phone and a WebRTC leg (H.264 passthrough), RFC 3264 o= version stickiness, and DTMF logging.

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 1.5.0

  • Video between a legacy SIP video phone and a WebRTC leg. A SIP phone that offers H.264 (RTP/AVP) can now exchange two-way video with a browser leg, with no transcoding. The answer sent to the caller is narrowed to the video codec the far/bridge leg actually negotiated (RFC 3264 §6): codecs the far leg cannot produce are dropped instead of merely reordered, so the caller does not lock its video onto a payload type the peer will never send. H.264 a=fmtp (profile-level-id / packetization-mode, RFC 6184 §8.2.2) is relayed and, when the two sides cannot agree, video fails closed to audio rather than sending a stream the peer drops. The intersection is SDP-only — no channel format state is mutated mid-call — and a keyframe request crosses the bridge in both directions (SIP INFO picture_fast_update ⇄ RTCP PLI/FIR, RFC 5168/4585/5104). The INFO answer is now bound to its own transaction (NUTAG_WITH_THIS), removing a ~32-second Timer-F cut. See WebRTC.
  • SDP o= version stickiness (RFC 3264 §8). A re-offer that repeats the same session origin with an unchanged (<=) version is treated as a no-op, so a learned symmetric-RTP remote address survives an UPDATE/re-INVITE — matching chan_sip process_sdp_o and preserving audio across NAT keepalive re-offers. The ignoresdpversion option is honoured (default no).
  • DTMF logging. sip set debug dtmf on|off logs every received DTMF digit (RFC 2833/telephone-event, SIP INFO, or inband) to the CLI and the messages log, the way chan_sip did — useful when diagnosing IVR and feature-code entry. Default off. See Chan-Sofia.

What is new in 1.4.2

  • WebRTC call hold/resume works end to end. Pressing Hold on a WebRTC phone no longer tears the call down, and video resumes by itself on unhold, exactly like audio — in every combination of WebRTC and desk-phone endpoints, on either side of the call. Five fixes: the SDP answer now mirrors the offered per-media direction (RFC 3264 §6.1); answers mirror every offered m-line using the current offer/answer transaction role (RFC 3264 §6); the fork-winner handover preserves the SDP stream identity (cname/msid, RFC 7022/8830); negotiated video payload types survive renegotiation (RFC 3264 §8.3.2); and RTCP keyframe feedback (PSFB PLI/FIR, RFC 4585/5104) is implemented end to end — advertised as a=rtcp-fb, relayed across the bridge, and transmitted SRTCP-protected over the ICE-selected tuple. See WebRTC.

What is new in 1.4.1

  • WebRTC fork-winner media fix. When a call forks to an extension registered from both a browser (wss) and a desk phone (udp) and the browser answers first, the call now carries two-way audio and video (previously the winning leg's DTLS never completed because stale media descriptors starved its socket). See WebRTC.
  • Native peer-to-peer SIP text messaging. An out-of-dialog MESSAGE between two registered users is delivered to the recipient's live devices, resolved through the same auto-created hint namespace as BLF and fanned out to every registered contact (RFC 3428). New option message_autorelay (default on); the message_context dialplan route remains available as a fallback. See Features.

What is new in 1.4.0

  • Reliable WebRTC audio. The ICE media tuple is now latched on every integrity-authenticated connectivity check, not only the nominated pair, removing intermittent one-way audio and the 10–20 s startup gap and making inbound calls to a browser come up cleanly (RFC 8445 / RFC 7675). See WebRTC.
  • Bundled WebRTC video. A new option webrtc_video_bundle (per-peer and [general], default off, requires webrtc=yes) carries audio and video over a single ICE/DTLS transport with payload-type demux and the RTP MID header extension (RFC 8843 max-bundle, RFC 8285). See WebRTC.
  • Per-Contact registration. Each Contact in a REGISTER now binds, refreshes and de-registers independently with its own ;expires (RFC 3261 §10.3), and a call to a peer whose bindings have all expired returns CHANUNAVAIL instead of routing to a stale binding. See Chan-Sofia.
  • DTMF parity with chan_sip. The negotiated DTMF mode is reflected end to end: telephone-event is advertised only for RFC 2833 modes, dtmfmode=auto resolves at SDP commit with a fax-safe detector reconfigure, and inbound SIP INFO signals are parsed. See Features.
  • TLS 1.2 accepted by default. tls_min_version now defaults to TLS 1.2 (1.2 and 1.3) instead of 1.3-only, so endpoints that cap at TLS 1.2 connect. See Security.
  • Live decrypted SIP capture. sip_capture_address streams every decrypted SIP message (udp/tcp/tls/ws/wss, with SDP) as HEP to a Homer/sipcapture server, sip_capture_file writes the same to a text file, and rtp set debug ice / sip set debug fork add targeted tracers. See Operations.

What is new in 1.3.5

  • Inbound WebRTC calls now connect. A call placed to a wss-registered browser (GABPBX as the SDP offerer) comes up with two-way audio, alongside the already-working browser-originated calls. Two RFC-grounded fixes make it work: an ICE role-conflict repair (GABPBX replies 487 Role Conflict so the browser flips to controlling and nominates a pair, RFC 8445), and deferring inbound DTLS until the answer's a=fingerprint is installed instead of failing closed (RFC 5763/8842). Rejected video tears down only its own pre-armed DTLS/ICE and never drops the audio call. See WebRTC.
  • SDP/ICE diagnostics. sip set debug sdp [on|off] traces the generated offer SDP, the WebRTC offer-gate decision, the incoming answer SDP and the offerer answer-apply path — the WSS/TLS signalling already decrypted. Every Sofia: … debug line now carries a [from-user|to-user|callid] transaction tag. See Operations.

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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