JsSIP Harness — voip-api Phase 5 v2.0 Manual Validation

Phase 5 v2.0: valida REGISTER + INVITE + DTLS-SRTP + bridge audio entre 2 ramais same-workspace (STUN-only baseline). Cobertura TURN entra Phase 6.

2-tab smoke: abra ?slug=harness-v2-<unix>&ramal=2000&password=<senha-2000> em uma aba (caller) e ?slug=harness-v2-<unix>&ramal=2001&password=<senha-2001> em outra aba (auto-answerer). mute mic da aba inativa E use HEADPHONES antes de iniciar a chamada (evita feedback loop).

Config priority: query-string (?slug=...&ramal=...&password=...) overrides inline constants. Edit DEFAULT_SLUG, DEFAULT_RAMAL, DEFAULT_SENHA constants in the <script> tag below only if you prefer inline config; prefer the query-string for disposable test sessions to avoid committing credentials.

Open browser DevTools console (F12) for [SC1] [SC2] [SC3] [SC4] event tags.

After UA registers, Plan 09 Task 3 drives SC3 from the DevTools console using window.ua.call(...) — ua is exposed globally below.

Config source: (pending init)

SC1: WebSocket not connected yet

SC2: SIP REGISTER pending

SC3: no RTC session yet

SDP inspector

(no SDP offer captured yet)

(no SDP answer captured yet — full answer requires Phase 7 ARI originate)