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docs(wiki): document 1.4.1 (fork-winner media, native MESSAGE relay) and 1.4.2 (WebRTC hold/resume overhaul, RTCP PSFB keyframe feedback)
docs(wiki): document 1.4.0 — HEP capture, video-bundle, DTMF parity, per-Contact REGISTER, TLS 1.2 default, ice/fork tracers
Document 1.3.5: inbound WebRTC (offerer) two-way audio + SDP/ICE diagnostics
Document GABPBX 1.3.4: RFC 5626 flow-close, sip show peer summary/detail, RED (RFC 2198) hardening, rtp set debug ice
WebRTC: document the DataChannel relay (RFC 8831/8832/8841) chan_sofia now bridges WebRTC DataChannels back-to-back between two WebRTC legs (SCTP-over-DTLS bundled on the audio DTLS, DCEP OPEN/ACK proxy, PPID-preserving message relay, abortive teardown). Add a DataChannel section to WebRTC.md, a feature entry to Features.md, the standards-table rows and the datachannel=yes config, and remove DataChannel from 'not in scope'.
wiki: GABPBX 1.1.2 — feature catalogue, WebRTC, Security, Migrating Home + sidebar bumped to 1.1.2. New pages: Features (the full chan_sofia catalogue one by one), WebRTC (the browser media story), Security (the hardening posture), and Migrating-from-chan_sip. Chan-Sofia: WebRTC media marked implemented (no longer roadmap).