The dock opens a panel with a 12-key dialpad (typing, paste normalization,
Enter) and microphone/speaker selection. Call cards have mute, hold,
keypad (DTMF) and transfer: "Transfer now" (blind) or "Talk first"
(attended, with Complete/Cancel). An incoming call opens a dialog when
idle, or a card during a call (call waiting holds the first call). A
synthesized WebAudio ringtone and desktop notifications, auto-answer for
Call-Info answer-after=0, and a poor-connection warning from getStats.
One tab registers: a Web Lock plus a BroadcastChannel "Use here".
Found by e2e: Chrome can report canInsertDTMF=false, so DTMF falls back to
SIP INFO; after a REFER's 2xx NOTIFY the browser sends BYE itself (RFC
5589), otherwise SIP.js keeps the call up. apps/web/TESTING.md is the
12-step manual run for Chrome and Firefox (still open, roadmap note);
e2e m5-t2 automates most of it in headless Edge.
Task: M5-T2
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
iamdoubz