GET /api/me/webrtc returns the <ext>-web identity, WebSocket URL
(WEBRTC_WS_URL overrides wss://PUBLIC_HOSTNAME/ws) and STUN/TURN servers.
SoftphoneEngine wraps SIP.js 0.21.2: register with backoff reconnect,
call, answer, decline, hang up, remote audio. A minimal dock on /me
enables it (asks for the microphone on the click) and dials/answers;
M5-T2 grows it into the full panel.
Generator: ice_host_candidates keeps the LAN address
(include_local_address), so LAN browsers don't need hairpin NAT, and -web
endpoints set from_domain = selfphoneng.invalid because SIP.js drops
INVITEs whose From host starts with a digit (Docker/VPS hostnames).
e2e m5-t1 (Docker): browser registers as 170-web and has two-way audio
with a baresip phone both ways. The LAN and hotspot check on the owner's
server is still open (roadmap note).
Task: M5-T1
/api/me/voicemail lists, streams (Range, download) and marks read or
deletes through AMI VoicemailMove/VoicemailRemove, then rescans so MWI
and the list agree. /api/me/greetings takes the raw recording or file and
converts it with ffmpeg to 8 kHz PCM (media/audio.ts). My Phone gets a
layout with Home (new voicemails) and Voicemail (play marks read,
download, delete, record/upload greetings via MediaRecorder); the list
refetches on the live voicemail.new event. e2e m4-t3 checks MWI goes off
within 5 s after play and delete, and a browser-recorded greeting is what
callers hear.
Task: M4-T3
iamdoubz