iamdoubz
ae3a723a3c
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
67 lines
4.4 KiB
Markdown
67 lines
4.4 KiB
Markdown
# Browser checks
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Dependency-free scripts that drive a headless Chromium (Edge/Chrome) over the DevTools
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protocol. They need an api serving the built web UI and a fresh database.
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```sh
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pnpm --filter @selfphoneng/web build && mkdir -p /tmp/sp && cp -r apps/web/dist /tmp/sp/public
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# fresh DB, then from /tmp/sp:
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DATABASE_URL=postgres://… INITIAL_ADMIN_EMAIL=admin@example.com INITIAL_ADMIN_PASSWORD=initial-admin-pw \
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node --env-file=<repo>/.env --conditions=source <repo>/apps/api/src/index.ts
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BROWSER=/usr/bin/chromium node scripts/e2e/m1-t4.mjs http://127.0.0.1:8080 admin@example.com initial-admin-pw /tmp/shots
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node scripts/e2e/mobile.mjs http://127.0.0.1:8080 admin@example.com correct-horse-battery-staple /tmp/shots
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```
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`m1-t4.mjs` is the M1-T4 acceptance flow: forced password change, then create/edit/delete a user.
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## M2 live loop (any Docker host, no host networking)
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`m2-live.mjs` runs the whole chain on a private Docker network: a test Asterisk with AMI published
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on `127.0.0.1:15038`, the api writing config into it, two baresip phones, and the dashboard in a
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browser. It checks that a new extension is live in Asterisk within 2 s, both phones show online,
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and the dots turn red during a call.
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```sh
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node scripts/e2e/m2-live.mjs up /tmp/sp-live-conf
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# start the api with AMI_PORT=15038 AMI_SECRET=m2-live-secret ASTERISK_CONFIG_DIR=/tmp/sp-live-conf
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node scripts/e2e/m1-t4.mjs http://127.0.0.1:8080 admin@example.com initial-admin-pw /tmp/shots
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node scripts/e2e/m2-live.mjs check http://127.0.0.1:8080 admin@example.com correct-horse-battery-staple /tmp/shots
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node scripts/e2e/m2-live.mjs down
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```
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`m3-t1.mjs` runs on the same setup (after `m1-t4.mjs`): it starts a fake voip.ms (a second Asterisk
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that accepts one sub-account) on the m2-live network, creates the trunk in the UI and waits for
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**Registered**, then presses Test. `m3-t2.mjs` (after `m3-t1.mjs`) adds a phone number routed to a user in the UI and
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checks `sp-inbound` in Asterisk, and that the extension can no longer be deleted. `m3-t3.mjs` (after
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both) checks the default outbound rules, the test box, a user's calling permission, and that
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reordering changes the include order in Asterisk. `m3-t4.mjs` (after all three) runs a tiny SMTP sink on
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127.0.0.1:2525 and a baresip phone as 101, then checks the dashboard warnings, Settings → Email and
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Emergency, and the Test notifications button (email arrives, 101 rings, red banner). Nothing dials 911. The fake provider from `m3-t1.mjs` stays up until `m2-live.mjs down`: a container that took
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over its IP would be identified as the trunk.
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`pnpm test:asterisk` (golden-file load test, `call-test.ts` with 30 s two-way audio between two
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baresip phones, and `trunk-test.ts` with registration against a fake provider) needs no api at all.
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`m3-t5.mjs` is the M3-T5 acceptance and needs its own fresh database: it signs in with the
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initial password, walks all ten wizard steps against a fake voip.ms, then proves an inbound call
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to the DID rings the new user's softphone and an outbound call leaves with the DID as caller ID.
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`m4-t2.mjs` is the M4-T2 acceptance (no browser): start m2-live with a voicemail dir
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(`m2-live.mjs up <confDir> <vmDir>`) and the api with `VOICEMAIL_DIR=<vmDir>`. It runs the SMTP
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sink (`smtp-sink.mjs`, shared with `m3-t4.mjs`), adds a user and extension 150 with voicemail,
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leaves a message from a Local channel, and checks the email arrives within 10 s with a PCM 8 kHz
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wav attached. The voicemail itself (MWI on a phone, `*97`) is covered by `pnpm test:asterisk`.
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`m4-t3.mjs <baseUrl> <adminEmail> <adminPassword> <vmDir> <shotsDir>` is the M4-T3 acceptance on the
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same setup: it adds a user with extension 160 and a baresip phone with the mwi module, leaves
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messages, then as that user in the browser plays one (marks it read) and deletes another, each
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time checking the phone's MWI goes off within 5 s. It records the "no answer" greeting with a
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fake microphone and checks a caller hears it.
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`m5-t1.mjs <baseUrl> <adminEmail> <adminPassword> <shotsDir>` is the M5-T1 check in Docker: start
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the api with `WEBRTC_WS_URL=ws://127.0.0.1:18088/ws`. `m2-live.mjs up` publishes Asterisk's HTTP
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server on 127.0.0.1:18088 and RTP 10000–10019/udp. The script sets LAN IP = the container and public
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IP = 127.0.0.1 (so ICE offers 127.0.0.1), restarts telephony, then signs in as a new user, enables
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the softphone with a fake microphone, and checks two-way audio to a baresip phone (171) in both
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directions.
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